Merge branch 'Isometric-Dev' into master

2026-08-28 19:30:51 +02:00 · 2020-08-06 15:44:55 -07:00
parent d7243e8265 902d8381cb
commit ae0648fccc
433 changed files with 172342 additions and 8485 deletions
@@ -0,0 +1,719 @@
+[Basic]
+Name=In the thick of it
+Intro=<none>
+Brief=<none>
+Win=<none>
+Lose=<none>
+Action=<none>
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+Theme=No theme
+CarryOverMoney=0
+ToCarryOver=no
+ToInherit=no
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+Official=yes
+FillSilos=no
+TruckCrate=no
+Percent=0
+
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+
+[BigOverlay]
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+
+[Map]
+Theater=SNOW
+X=25
+Y=33
+Width=54
+Height=46
+HomeCell=4386
+
+[Waypoints]
+0=8990
+1=9024
+2=5186
+3=5279
+98=4386
+99=8256
+
+[MapPack]
+1=BwAAIIH//v8f/4BXAAAggf/+XQ3/ghsAEAL++gD//wADXg2CFAAQAv76AP//7ANeEYIXAD
+2=ACgRLIURaBECACMAyBEyACgRhAAv7SAP//AAJKFv8ABUoR/xQBSh2CGgAQAv6cAP+AGQAA
+3=IIH//l0A/4IaABAC/voA//8AAl4A/qIc/4AHAAAggf/+/x//gGgAACCBAP6vFgCCAQL+fQ
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+6=X+fQAAgwYHCP59AACDCQoL/s4eAIA=
+
+[TERRAIN]
+7207=T01
+7981=T01
+6568=T01
+6818=T01
+6814=T01
+7330=T01
+9522=T01
+9009=T01
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+7226=T01
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+5810=T01
+5814=T01
+5300=T01
+4786=T01
+4917=T01
+4403=T01
+9914=T01
+
+[STRUCTURES]
+0=Neutral,ABAN01,256,7338,0,None,1,0
+1=Neutral,ABAN01,256,6442,0,None,1,0
+2=Neutral,ABAN01,256,6449,0,None,1,0
+3=Neutral,ABAN01,256,7473,0,None,1,0
+4=Neutral,ABAN02,256,8113,0,None,1,0
+5=Neutral,ABAN02,256,5555,0,None,1,0
+6=Neutral,ABAN03,256,4659,0,None,1,0
+7=Neutral,ABAN02,256,7363,0,None,1,0
+8=Neutral,ABAN02,256,6584,0,None,1,0
+9=Neutral,ABAN02,256,9266,0,None,1,0
+
+[Base]
+Player=Multi1
+Count=0
+
+[OverlayPack]
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+9=TALkSwL/YwDKANrNBv9dAMYD/lgY/4A=
+
+[Briefing]
+1=Rescue Einstein from the Headquarters inside this Soviet complex. Once
+2=found, evacuate him via the helicopter at the signal flare. Einstein and
+3=Tanya must be kept alive at all costs. Beware the Soviet's Tesla Coils.
+4=Direct Tanya to destroy the westmost power plants to take them off-line.
+
+[Digest]
+1=ty7v+aJuxNMiESmZBtqZtzZ5pIc=
+
@@ -0,0 +1,556 @@
+[General]
+ChronoTechLevel=12
+ParaTech=5
+ParabombTech=8
+SpyPlaneTech=5
+GPSTechLevel=10
+
+[HBOX]
+Power=-10
+
+[PBOX]
+Power=-10
+
+[AGUN]
+Power=-40
+
+[FTUR]
+Power=-40
+
+[MRJ]
+TechLevel=15
+Owner=soviet
+
+[MGG]
+TechLevel=15
+
+[MCV]
+TechLevel=15
+
+[LST]
+TechLevel=1
+
+[DOG]
+TechLevel=3
+
+[E4]
+TechLevel=6
+
+[THF]
+TechLevel=15
+
+[E7]
+Prerequisite=atek
+TechLevel=15
+Owner=allies
+
+[YAK]
+TechLevel=1
+
+[TRAN]
+TechLevel=15
+Owner=allies,soviet
+
+[HELI]
+TechLevel=10
+
+[HIND]
+TechLevel=10
+
+[IRON]
+TechLevel=15
+
+[ATEK]
+TechLevel=10
+
+[PDOX]
+TechLevel=15
+
+[SPEN]
+TechLevel=5
+
+[HPAD]
+TechLevel=10
+
+[MSLO]
+TechLevel=15
+Owner=soviet
+
+[AFLD]
+TechLevel=4
+
+[KENN]
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+4=USSR,TTNK,256,5706,192,Guard,None
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+13=Greece,2TNK,256,6320,0,Area Guard,None
+14=Greece,2TNK,256,6449,0,Area Guard,None
+
+[INFANTRY]
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+33=Greece,C7,256,6332,1,Area Guard,0,None
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+53=USSR,POWR,256,2747,0,None,0,0
+54=USSR,POWR,256,2105,0,None,0,0
+55=Greece,PBOX,256,3891,0,None,0,0
+56=Greece,PBOX,256,3900,0,None,0,0
+57=Greece,PBOX,256,3906,0,None,0,0
+58=Greece,PBOX,256,3886,0,None,0,0
+59=Greece,APWR,256,6925,0,None,0,0
+
+[Base]
+Player=USSR
+Count=0
+
+[OverlayPack]
+1=bQAAIIH//osZ/4EK/ncA/4EOAAL+egD/gQtgCoEM/vkA/4MM/wn+ewD/gQkQfnGH/nMA/4
+2=EK/30Ahxv/gwCDGoQJ/wwL/34AiByBCwD9/30Aix2BDACD/30AjBqBCgB/gQz/fAAJHAYA
+3=gQr/cAAMHYBWAAAggf9wAYMKCv8gAoEJ/ngA/4IJCgB8gwoLDP95ABMAIIUgAf55AP+BCS
+4=CB/noA/wB8/4AAkgEQf/96ABMA/30AlABiA/5+AP//gAARA/+CABED/uka/4A=
+
+[Briefing]
+1=A pitiful excuse for resistance has blockaded itself in this village.
+2=Stalin has decided to make an example of them. Kill them all and destroy
+3=their homes. You will have Yak aircraft to use in teaching these rebels a
+4=lesson.
+
+[Digest]
+1=9QVKWgKK7MJHlaf+Cp3h7V2cWSk=
+
@@ -0,0 +1,4 @@
+set PATH="C:\Program Files\ImageMagick-7.0.10-Q16-HDRI"
+
+magick.exe convert sno1.png -alpha set -virtual-pixel transparent +distort Affine "0,512 0,0   0,0 -87,-50  512,512 87,-50" -background none -compose plus  -bordercolor black -compose over -border 5x2 sno1_iso.png
+pause
@@ -6,7 +6,12 @@ enable_language(ASM)
 set(CMAKE_CXX_FLAGS "")
 set(CMAKE_LIBRARY_OUTPUT_DIRECTORY ${CMAKE_CURRENT_SOURCE_DIR}/../)
 set(CMAKE_RUNTIME_OUTPUT_DIRECTORY ${CMAKE_CURRENT_SOURCE_DIR}/../)
-set(CMAKE_VERBOSE_MAKEFILE ON)
+SET(CMAKE_CXX_FLAGS_DEBUG "/MTd /Od /Ob0 /Zi /DDEBUG /D_DEBUG")
+SET(CMAKE_C_FLAGS_DEBUG "/MTd /Od /Ob0 /Zi /DDEBUG /D_DEBUG")
+
+set(src_launcher
+	./launcher/launcher_main.cpp
+)

 set(src_io
 	./redalert/io/userinput.cpp
@@ -104,7 +109,7 @@ set(src_win32lib
 	./redalert/win32lib/wwstd.h
 	./redalert/win32lib/ww_win.h
 	./redalert/win32lib/_diptabl.cpp
-)
+ "redalert/ffmpegmovie.cpp" "redalert/briefing.cpp")

 #Red Alert files
 set(src_redalert
@@ -160,6 +165,8 @@ set(src_redalert
 	./redalert/carry.cpp
 	./redalert/carry.h
 	./redalert/cbn_.h
+	./redalert/crc32.cpp
+	./redalert/crc32.h
 	./redalert/ccdde.cpp
 	./redalert/ccdde.h
 	./redalert/ccfile.cpp
@@ -304,6 +311,8 @@ set(src_redalert
 	./redalert/idata.cpp
 	./redalert/infantry.cpp
 	./redalert/infantry.h
+	./redalert/tacticalcoord.cpp
+	./redalert/tacticalcoord.h
 	./redalert/ini.cpp
 	./redalert/ini.h
 	./redalert/inibin.cpp
@@ -629,21 +638,17 @@ set(src_redalert
 	./redalert/mapscript.cpp
 	./redalert/txtprnt.cpp
 	./redalert/tilesetxml.cpp
- "redalert/sidebareditor.cpp")
+	"redalert/bigoverlay.cpp"
+	"redalert/bigoverlay.h"
+ 	"redalert/sidebareditor.cpp" 
+	"redalert/rendertexture.cpp"
+	"redalert/rendertexture.h"
+	"redalert/light.cpp")

 set(src_external
 	#glew
 	./external/gl/glew.c
 	
-	#libsmacker
-	./external/libsmacker/smacker.c
-	./external/libsmacker/smacker.h
-	./external/libsmacker/smk_bitstream.c
-	./external/libsmacker/smk_bitstream.h
-	./external/libsmacker/smk_hufftree.c
-	./external/libsmacker/smk_hufftree.h
-	./external/libsmacker/smk_malloc.h
-	
 	#imgui
 	./external/imgui/imconfig.h
 	./external/imgui/imgui.cpp
@@ -719,8 +724,16 @@ set(src_external
 	./external/xml/tinyxml2.h
 )

+set(FFmpeg_LIBRARIES
+				"${CMAKE_SOURCE_DIR}/external/ffmpeg-win64/lib/avcodec.lib"
+				"${CMAKE_SOURCE_DIR}/external/ffmpeg-win64/lib/avformat.lib"
+				"${CMAKE_SOURCE_DIR}/external/ffmpeg-win64/lib/avutil.lib"
+				"${CMAKE_SOURCE_DIR}/external/ffmpeg-win64/lib/swscale.lib") 

 if (MSVC)
+# External always compiles with optimizations enabled.
+set_source_files_properties(${src_external} PROPERTIES COMPILE_FLAGS "/O2")
+
 add_definitions(-D RA_AF -DCHEAT_KEYS -DIMGUI_IMPL_OPENGL_LOADER_GLEW -DSCENARIO_EDITOR -DWINDOWS_IGNORE_PACKING_MISMATCH -DTRUE_FALSE_DEFINED -DWIN32 -DWINDOWS -DENGLISH -D_USRDLL -DREDALERT_EXPORTS)
 add_compile_options(/permissive+  /Zc:forScope- /Zp1)
 endif(MSVC)
@@ -729,15 +742,20 @@ if (UNIX)
 add_definitions(-D RA_AF -DCHEAT_KEYS -DIMGUI_IMPL_OPENGL_LOADER_GLEW -DSCENARIO_EDITOR -DTRUE_FALSE_DEFINED -DENGLISH -D_USRDLL -DREDALERT_EXPORTS)
 endif(UNIX)

-add_executable(RedAlert ${src_io} ${src_win32lib} ${src_redalert} ${src_external})
+add_library(RedAlertDLL SHARED ${src_io} ${src_win32lib} ${src_redalert} ${src_external} "redalert/light.cpp" "redalert/briefing.cpp")

 if (MSVC)
-set_target_properties(RedAlert PROPERTIES OUTPUT_NAME "RedAlert" LINK_FLAGS "/PDB:\"RedAlert.pdb\" /SUBSYSTEM:WINDOWS" RUNTIME_OUTPUT_DIRECTORY "${CMAKE_SOURCE_DIR}/../" )
-target_link_libraries(RedAlert "opengl32.lib" "winmm.lib" "Ws2_32.lib" "${CMAKE_SOURCE_DIR}/external/devil/ilu.lib" "${CMAKE_SOURCE_DIR}/external/devil/DevIL.lib" "${CMAKE_SOURCE_DIR}/external/sdl2/lib/x64/SDL2.lib" "${CMAKE_SOURCE_DIR}/external/openal/out/build/x64-Release/OpenAL32.lib")
+set_target_properties(RedAlertDLL PROPERTIES OUTPUT_NAME "RedAlertx64" LINK_FLAGS "/PDB:\"RedAlertx64.pdb\" /SUBSYSTEM:WINDOWS /DEF:${CMAKE_CURRENT_SOURCE_DIR}/exports.def" RUNTIME_OUTPUT_DIRECTORY "${CMAKE_SOURCE_DIR}/../" )
+target_link_libraries(RedAlertDLL "opengl32.lib" "winmm.lib" "Ws2_32.lib" "${CMAKE_SOURCE_DIR}/external/devil/ilu.lib" "${CMAKE_SOURCE_DIR}/external/devil/DevIL.lib" "${CMAKE_SOURCE_DIR}/external/sdl2/lib/x64/SDL2.lib" "${CMAKE_SOURCE_DIR}/external/openal/out/build/x64-Release/OpenAL32.lib" ${FFmpeg_LIBRARIES})
+
+add_executable(Launcher ${src_launcher})
+target_link_libraries(Launcher RedAlertDLL)
+set_target_properties(Launcher PROPERTIES OUTPUT_NAME "RedAlert" LINK_FLAGS "/SUBSYSTEM:WINDOWS /PDB:\"RedAlert.pdb\"")
+
 endif(MSVC)

 if (UNIX)
 set_target_properties(RedAlert PROPERTIES OUTPUT_NAME "RedAlert"  RUNTIME_OUTPUT_DIRECTORY "${CMAKE_SOURCE_DIR}/.." )
 endif(UNIX)

-target_include_directories(RedAlert PRIVATE ./external/xml/;./external/devil/;./external/;./redalert/win32lib;./external/lua/;./external/imgui;./external/dxsdk/Include;./external/ffmpeg-win32/include;./external/sdl2/include;./win32lib;./external/openal/include)
+target_include_directories(RedAlertDLL PRIVATE ./external/ffmpeg-win64/include/;./external/xml/;./external/devil/;./external/;./redalert/win32lib;./external/lua/;./external/imgui;./external/dxsdk/Include;./external/ffmpeg-win32/include;./external/sdl2/include;./win32lib;./external/openal/include)
@@ -0,0 +1,2 @@
+EXPORTS
+	CnCMain @1
@@ -0,0 +1,111 @@
+This is a FFmpeg win64 shared build by Kyle Schwarz.
+
+Zeranoe's FFmpeg Builds Home Page: <http://ffmpeg.zeranoe.com/builds/>
+
+FFmpeg version: 20151105-git-c878082
+  libavutil      54. 30.100 / 54. 30.100
+  libavcodec     56. 57.100 / 56. 57.100
+  libavformat    56. 40.101 / 56. 40.101
+  libavdevice    56.  4.100 / 56.  4.100
+  libavfilter     5. 32.100 /  5. 32.100
+  libswscale      3.  1.101 /  3.  1.101
+  libswresample   1.  2.101 /  1.  2.101
+  libpostproc    53.  3.100 / 53.  3.100
+
+This FFmpeg build was configured with:
+  --disable-static
+  --enable-shared
+  --enable-gpl
+  --enable-version3
+  --disable-w32threads
+  --enable-avisynth
+  --enable-bzlib
+  --enable-fontconfig
+  --enable-frei0r
+  --enable-gnutls
+  --enable-iconv
+  --enable-libass
+  --enable-libbluray
+  --enable-libbs2b
+  --enable-libcaca
+  --enable-libdcadec
+  --enable-libfreetype
+  --enable-libgme
+  --enable-libgsm
+  --enable-libilbc
+  --enable-libmodplug
+  --enable-libmp3lame
+  --enable-libopencore-amrnb
+  --enable-libopencore-amrwb
+  --enable-libopenjpeg
+  --enable-libopus
+  --enable-librtmp
+  --enable-libschroedinger
+  --enable-libsoxr
+  --enable-libspeex
+  --enable-libtheora
+  --enable-libtwolame
+  --enable-libvidstab
+  --enable-libvo-aacenc
+  --enable-libvo-amrwbenc
+  --enable-libvorbis
+  --enable-libvpx
+  --enable-libwavpack
+  --enable-libwebp
+  --enable-libx264
+  --enable-libx265
+  --enable-libxavs
+  --enable-libxvid
+  --enable-lzma
+  --enable-decklink
+  --enable-zlib
+
+This build was compiled with the following external libraries:
+  bzip2 1.0.6 <http://bzip.org/>
+  Fontconfig 2.11.94 <http://freedesktop.org/wiki/Software/fontconfig>
+  Frei0r 20130909-git-10d8360 <http://frei0r.dyne.org/>
+  GnuTLS 3.3.16 <http://gnutls.org/>
+  libiconv 1.14 <http://gnu.org/software/libiconv/>
+  libass 0.13.0 <https://github.com/libass/libass>
+  libbluray 0.9.0 <http://videolan.org/developers/libbluray.html>
+  libbs2b 3.1.0 <http://bs2b.sourceforge.net/>
+  libcaca 0.99.beta18 <http://caca.zoy.org/wiki/libcaca>
+  dcadec 20150506-git-98fb3b6 <https://github.com/foo86/dcadec>
+  FreeType 2.6.1 <http://freetype.sourceforge.net/>
+  Game Music Emu 0.6.0 <https://bitbucket.org/mpyne/game-music-emu/wiki/Home>
+  GSM 1.0.13-4 <http://packages.debian.org/source/squeeze/libgsm>
+  iLBC 20141214-git-ef04ebe <https://github.com/dekkers/libilbc/>
+  Modplug-XMMS 0.8.8.5 <http://modplug-xmms.sourceforge.net/>
+  LAME 3.99.5 <http://lame.sourceforge.net/>
+  OpenCORE AMR 0.1.3 <http://sourceforge.net/projects/opencore-amr/>
+  OpenJPEG 1.5.2 <https://github.com/uclouvain/openjpeg>
+  Opus 1.1 <http://opus-codec.org/>
+  RTMPDump 20140707-git-a1900c3 <http://rtmpdump.mplayerhq.hu/>
+  Schroedinger 1.0.11 <http://diracvideo.org/>
+  libsoxr 0.1.2 <http://sourceforge.net/projects/soxr/>
+  Speex 1.2rc2 <http://speex.org/>
+  Theora 1.1.1 <http://theora.org/>
+  TwoLAME 0.3.13 <http://twolame.org/>
+  vid.stab 0.98 <http://public.hronopik.de/vid.stab/>
+  VisualOn AAC 0.1.3 <https://github.com/mstorsjo/vo-aacenc>
+  VisualOn AMR-WB 0.1.2 <https://github.com/mstorsjo/vo-amrwbenc>
+  Vorbis 1.3.5 <http://vorbis.com/>
+  vpx 1.4.0 <http://webmproject.org/>
+  WavPack 4.75.0 <http://wavpack.com/>
+  WebP 0.4.3 <https://developers.google.com/speed/webp/>
+  x264 20151011-git-7599210 <http://videolan.org/developers/x264.html>
+  x265 1.8 <http://x265.org/>
+  XAVS svn-r55 <http://xavs.sourceforge.net/>
+  Xvid 1.3.4 <http://xvid.org/>
+  XZ Utils 5.2.2 <http://tukaani.org/xz>
+  zlib 1.2.8 <http://zlib.net/>
+
+The source code for this FFmpeg build can be found at: <http://ffmpeg.zeranoe.com/builds/source/>
+
+This build was compiled on Debian 8.2 (64-bit): <http://www.debian.org/>
+
+GCC 5.2.0 was used to compile this FFmpeg build: <http://gcc.gnu.org/>
+
+This build was compiled using the MinGW-w64 toolchain: <http://mingw-w64.sourceforge.net/>
+
+Licenses for each library can be found in the 'licenses' folder.
@@ -0,0 +1,785 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Developer Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Developer Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Developers-Guide" href="#Developers-Guide">1 Developers Guide</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Notes-for-external-developers" href="#Notes-for-external-developers">1.1 Notes for external developers</a></li>
+    <li><a name="toc-Contributing" href="#Contributing">1.2 Contributing</a></li>
+    <li><a name="toc-Coding-Rules-1" href="#Coding-Rules-1">1.3 Coding Rules</a>
+    <ul class="no-bullet">
+      <li><a name="toc-Code-formatting-conventions" href="#Code-formatting-conventions">1.3.1 Code formatting conventions</a></li>
+      <li><a name="toc-Comments" href="#Comments">1.3.2 Comments</a></li>
+      <li><a name="toc-C-language-features" href="#C-language-features">1.3.3 C language features</a></li>
+      <li><a name="toc-Naming-conventions" href="#Naming-conventions">1.3.4 Naming conventions</a></li>
+      <li><a name="toc-Miscellaneous-conventions" href="#Miscellaneous-conventions">1.3.5 Miscellaneous conventions</a></li>
+      <li><a name="toc-Editor-configuration" href="#Editor-configuration">1.3.6 Editor configuration</a></li>
+    </ul></li>
+    <li><a name="toc-Development-Policy" href="#Development-Policy">1.4 Development Policy</a></li>
+    <li><a name="toc-Submitting-patches-1" href="#Submitting-patches-1">1.5 Submitting patches</a></li>
+    <li><a name="toc-New-codecs-or-formats-checklist" href="#New-codecs-or-formats-checklist">1.6 New codecs or formats checklist</a></li>
+    <li><a name="toc-patch-submission-checklist" href="#patch-submission-checklist">1.7 patch submission checklist</a></li>
+    <li><a name="toc-Patch-review-process" href="#Patch-review-process">1.8 Patch review process</a></li>
+    <li><a name="toc-Regression-tests-1" href="#Regression-tests-1">1.9 Regression tests</a>
+    <ul class="no-bullet">
+      <li><a name="toc-Adding-files-to-the-fate_002dsuite-dataset" href="#Adding-files-to-the-fate_002dsuite-dataset">1.9.1 Adding files to the fate-suite dataset</a></li>
+      <li><a name="toc-Visualizing-Test-Coverage" href="#Visualizing-Test-Coverage">1.9.2 Visualizing Test Coverage</a></li>
+      <li><a name="toc-Using-Valgrind" href="#Using-Valgrind">1.9.3 Using Valgrind</a></li>
+    </ul></li>
+    <li><a name="toc-Release-process-1" href="#Release-process-1">1.10 Release process</a>
+    <ul class="no-bullet">
+      <li><a name="toc-Criteria-for-Point-Releases-1" href="#Criteria-for-Point-Releases-1">1.10.1 Criteria for Point Releases</a></li>
+      <li><a name="toc-Release-Checklist" href="#Release-Checklist">1.10.2 Release Checklist</a></li>
+    </ul></li>
+  </ul></li>
+</ul>
+</div>
+
+
+<a name="Developers-Guide"></a>
+<h2 class="chapter">1 Developers Guide<span class="pull-right"><a class="anchor hidden-xs" href="#Developers-Guide" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Developers-Guide" aria-hidden="true">TOC</a></span></h2>
+
+<a name="Notes-for-external-developers"></a>
+<h3 class="section">1.1 Notes for external developers<span class="pull-right"><a class="anchor hidden-xs" href="#Notes-for-external-developers" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Notes-for-external-developers" aria-hidden="true">TOC</a></span></h3>
+
+<p>This document is mostly useful for internal FFmpeg developers.
+External developers who need to use the API in their application should
+refer to the API doxygen documentation in the public headers, and
+check the examples in <samp>doc/examples</samp> and in the source code to
+see how the public API is employed.
+</p>
+<p>You can use the FFmpeg libraries in your commercial program, but you
+are encouraged to <em>publish any patch you make</em>. In this case the
+best way to proceed is to send your patches to the ffmpeg-devel
+mailing list following the guidelines illustrated in the remainder of
+this document.
+</p>
+<p>For more detailed legal information about the use of FFmpeg in
+external programs read the <samp>LICENSE</samp> file in the source tree and
+consult <a href="https://ffmpeg.org/legal.html">https://ffmpeg.org/legal.html</a>.
+</p>
+<a name="Contributing"></a>
+<h3 class="section">1.2 Contributing<span class="pull-right"><a class="anchor hidden-xs" href="#Contributing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Contributing" aria-hidden="true">TOC</a></span></h3>
+
+<p>There are 3 ways by which code gets into ffmpeg.
+</p><ul>
+<li> Submitting Patches to the main developer mailing list
+      see <a href="#Submitting-patches">Submitting patches</a> for details.
+</li><li> Directly committing changes to the main tree.
+</li><li> Committing changes to a git clone, for example on github.com or
+      gitorious.org. And asking us to merge these changes.
+</li></ul>
+
+<p>Whichever way, changes should be reviewed by the maintainer of the code
+before they are committed. And they should follow the <a href="#Coding-Rules">Coding Rules</a>.
+The developer making the commit and the author are responsible for their changes
+and should try to fix issues their commit causes.
+</p>
+<a name="Coding-Rules"></a><a name="Coding-Rules-1"></a>
+<h3 class="section">1.3 Coding Rules<span class="pull-right"><a class="anchor hidden-xs" href="#Coding-Rules-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Coding-Rules-1" aria-hidden="true">TOC</a></span></h3>
+
+<a name="Code-formatting-conventions"></a>
+<h4 class="subsection">1.3.1 Code formatting conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Code-formatting-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Code-formatting-conventions" aria-hidden="true">TOC</a></span></h4>
+
+<p>There are the following guidelines regarding the indentation in files:
+</p>
+<ul>
+<li> Indent size is 4.
+
+</li><li> The TAB character is forbidden outside of Makefiles as is any
+form of trailing whitespace. Commits containing either will be
+rejected by the git repository.
+
+</li><li> You should try to limit your code lines to 80 characters; however, do so if
+and only if this improves readability.
+</li></ul>
+<p>The presentation is one inspired by &rsquo;indent -i4 -kr -nut&rsquo;.
+</p>
+<p>The main priority in FFmpeg is simplicity and small code size in order to
+minimize the bug count.
+</p>
+<a name="Comments"></a>
+<h4 class="subsection">1.3.2 Comments<span class="pull-right"><a class="anchor hidden-xs" href="#Comments" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Comments" aria-hidden="true">TOC</a></span></h4>
+<p>Use the JavaDoc/Doxygen  format (see examples below) so that code documentation
+can be generated automatically. All nontrivial functions should have a comment
+above them explaining what the function does, even if it is just one sentence.
+All structures and their member variables should be documented, too.
+</p>
+<p>Avoid Qt-style and similar Doxygen syntax with <code>!</code> in it, i.e. replace
+<code>//!</code> with <code>///</code> and similar.  Also @ syntax should be employed
+for markup commands, i.e. use <code>@param</code> and not <code>\param</code>.
+</p>
+<div class="example">
+<pre class="example">/**
+ * @file
+ * MPEG codec.
+ * @author ...
+ */
+
+/**
+ * Summary sentence.
+ * more text ...
+ * ...
+ */
+typedef struct Foobar {
+    int var1; /**&lt; var1 description */
+    int var2; ///&lt; var2 description
+    /** var3 description */
+    int var3;
+} Foobar;
+
+/**
+ * Summary sentence.
+ * more text ...
+ * ...
+ * @param my_parameter description of my_parameter
+ * @return return value description
+ */
+int myfunc(int my_parameter)
+...
+</pre></div>
+
+<a name="C-language-features"></a>
+<h4 class="subsection">1.3.3 C language features<span class="pull-right"><a class="anchor hidden-xs" href="#C-language-features" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-C-language-features" aria-hidden="true">TOC</a></span></h4>
+
+<p>FFmpeg is programmed in the ISO C90 language with a few additional
+features from ISO C99, namely:
+</p>
+<ul>
+<li> the &lsquo;<samp>inline</samp>&rsquo; keyword;
+
+</li><li> &lsquo;<samp>//</samp>&rsquo; comments;
+
+</li><li> designated struct initializers (&lsquo;<samp>struct s x = { .i = 17 };</samp>&rsquo;);
+
+</li><li> compound literals (&lsquo;<samp>x = (struct s) { 17, 23 };</samp>&rsquo;).
+</li></ul>
+
+<p>These features are supported by all compilers we care about, so we will not
+accept patches to remove their use unless they absolutely do not impair
+clarity and performance.
+</p>
+<p>All code must compile with recent versions of GCC and a number of other
+currently supported compilers. To ensure compatibility, please do not use
+additional C99 features or GCC extensions. Especially watch out for:
+</p>
+<ul>
+<li> mixing statements and declarations;
+
+</li><li> &lsquo;<samp>long long</samp>&rsquo; (use &lsquo;<samp>int64_t</samp>&rsquo; instead);
+
+</li><li> &lsquo;<samp>__attribute__</samp>&rsquo; not protected by &lsquo;<samp>#ifdef __GNUC__</samp>&rsquo; or similar;
+
+</li><li> GCC statement expressions (&lsquo;<samp>(x = ({ int y = 4; y; })</samp>&rsquo;).
+</li></ul>
+
+<a name="Naming-conventions"></a>
+<h4 class="subsection">1.3.4 Naming conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Naming-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Naming-conventions" aria-hidden="true">TOC</a></span></h4>
+<p>All names should be composed with underscores (_), not CamelCase. For example,
+&lsquo;<samp>avfilter_get_video_buffer</samp>&rsquo; is an acceptable function name and
+&lsquo;<samp>AVFilterGetVideo</samp>&rsquo; is not. The exception from this are type names, like
+for example structs and enums; they should always be in CamelCase.
+</p>
+<p>There are the following conventions for naming variables and functions:
+</p>
+<ul>
+<li> For local variables no prefix is required.
+
+</li><li> For file-scope variables and functions declared as <code>static</code>, no prefix
+is required.
+
+</li><li> For variables and functions visible outside of file scope, but only used
+internally by a library, an <code>ff_</code> prefix should be used,
+e.g. &lsquo;<samp>ff_w64_demuxer</samp>&rsquo;.
+
+</li><li> For variables and functions visible outside of file scope, used internally
+across multiple libraries, use <code>avpriv_</code> as prefix, for example,
+&lsquo;<samp>avpriv_aac_parse_header</samp>&rsquo;.
+
+</li><li> Each library has its own prefix for public symbols, in addition to the
+commonly used <code>av_</code> (<code>avformat_</code> for libavformat,
+<code>avcodec_</code> for libavcodec, <code>swr_</code> for libswresample, etc).
+Check the existing code and choose names accordingly.
+Note that some symbols without these prefixes are also exported for
+retro-compatibility reasons. These exceptions are declared in the
+<code>lib&lt;name&gt;/lib&lt;name&gt;.v</code> files.
+</li></ul>
+
+<p>Furthermore, name space reserved for the system should not be invaded.
+Identifiers ending in <code>_t</code> are reserved by
+<a href="http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02">POSIX</a>.
+Also avoid names starting with <code>__</code> or <code>_</code> followed by an uppercase
+letter as they are reserved by the C standard. Names starting with <code>_</code>
+are reserved at the file level and may not be used for externally visible
+symbols. If in doubt, just avoid names starting with <code>_</code> altogether.
+</p>
+<a name="Miscellaneous-conventions"></a>
+<h4 class="subsection">1.3.5 Miscellaneous conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Miscellaneous-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Miscellaneous-conventions" aria-hidden="true">TOC</a></span></h4>
+
+<ul>
+<li> fprintf and printf are forbidden in libavformat and libavcodec,
+please use av_log() instead.
+
+</li><li> Casts should be used only when necessary. Unneeded parentheses
+should also be avoided if they don&rsquo;t make the code easier to understand.
+</li></ul>
+
+<a name="Editor-configuration"></a>
+<h4 class="subsection">1.3.6 Editor configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Editor-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Editor-configuration" aria-hidden="true">TOC</a></span></h4>
+<p>In order to configure Vim to follow FFmpeg formatting conventions, paste
+the following snippet into your <samp>.vimrc</samp>:
+</p><div class="example">
+<pre class="example">&quot; indentation rules for FFmpeg: 4 spaces, no tabs
+set expandtab
+set shiftwidth=4
+set softtabstop=4
+set cindent
+set cinoptions=(0
+&quot; Allow tabs in Makefiles.
+autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
+&quot; Trailing whitespace and tabs are forbidden, so highlight them.
+highlight ForbiddenWhitespace ctermbg=red guibg=red
+match ForbiddenWhitespace /\s\+$\|\t/
+&quot; Do not highlight spaces at the end of line while typing on that line.
+autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@&lt;!$/
+</pre></div>
+
+<p>For Emacs, add these roughly equivalent lines to your <samp>.emacs.d/init.el</samp>:
+</p><div class="lisp">
+<pre class="lisp">(c-add-style &quot;ffmpeg&quot;
+             '(&quot;k&amp;r&quot;
+               (c-basic-offset . 4)
+               (indent-tabs-mode . nil)
+               (show-trailing-whitespace . t)
+               (c-offsets-alist
+                (statement-cont . (c-lineup-assignments +)))
+               )
+             )
+(setq c-default-style &quot;ffmpeg&quot;)
+</pre></div>
+
+<a name="Development-Policy"></a>
+<h3 class="section">1.4 Development Policy<span class="pull-right"><a class="anchor hidden-xs" href="#Development-Policy" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Development-Policy" aria-hidden="true">TOC</a></span></h3>
+
+<ol>
+<li> Contributions should be licensed under the
+<a href="http://www.gnu.org/licenses/lgpl-2.1.html">LGPL 2.1</a>,
+including an &quot;or any later version&quot; clause, or, if you prefer
+a gift-style license, the
+<a href="http://opensource.org/licenses/isc-license.txt">ISC</a> or
+<a href="http://mit-license.org/">MIT</a> license.
+<a href="http://www.gnu.org/licenses/gpl-2.0.html">GPL 2</a> including
+an &quot;or any later version&quot; clause is also acceptable, but LGPL is
+preferred.
+If you add a new file, give it a proper license header. Do not copy and
+paste it from a random place, use an existing file as template.
+
+</li><li> You must not commit code which breaks FFmpeg! (Meaning unfinished but
+enabled code which breaks compilation or compiles but does not work or
+breaks the regression tests)
+You can commit unfinished stuff (for testing etc), but it must be disabled
+(#ifdef etc) by default so it does not interfere with other developers&rsquo;
+work.
+
+</li><li> The commit message should have a short first line in the form of
+a &lsquo;<samp>topic: short description</samp>&rsquo; as a header, separated by a newline
+from the body consisting of an explanation of why the change is necessary.
+If the commit fixes a known bug on the bug tracker, the commit message
+should include its bug ID. Referring to the issue on the bug tracker does
+not exempt you from writing an excerpt of the bug in the commit message.
+
+</li><li> You do not have to over-test things. If it works for you, and you think it
+should work for others, then commit. If your code has problems
+(portability, triggers compiler bugs, unusual environment etc) they will be
+reported and eventually fixed.
+
+</li><li> Do not commit unrelated changes together, split them into self-contained
+pieces. Also do not forget that if part B depends on part A, but A does not
+depend on B, then A can and should be committed first and separate from B.
+Keeping changes well split into self-contained parts makes reviewing and
+understanding them on the commit log mailing list easier. This also helps
+in case of debugging later on.
+Also if you have doubts about splitting or not splitting, do not hesitate to
+ask/discuss it on the developer mailing list.
+
+</li><li> Do not change behavior of the programs (renaming options etc) or public
+API or ABI without first discussing it on the ffmpeg-devel mailing list.
+Do not remove functionality from the code. Just improve!
+
+<p>Note: Redundant code can be removed.
+</p>
+</li><li> Do not commit changes to the build system (Makefiles, configure script)
+which change behavior, defaults etc, without asking first. The same
+applies to compiler warning fixes, trivial looking fixes and to code
+maintained by other developers. We usually have a reason for doing things
+the way we do. Send your changes as patches to the ffmpeg-devel mailing
+list, and if the code maintainers say OK, you may commit. This does not
+apply to files you wrote and/or maintain.
+
+</li><li> We refuse source indentation and other cosmetic changes if they are mixed
+with functional changes, such commits will be rejected and removed. Every
+developer has his own indentation style, you should not change it. Of course
+if you (re)write something, you can use your own style, even though we would
+prefer if the indentation throughout FFmpeg was consistent (Many projects
+force a given indentation style - we do not.). If you really need to make
+indentation changes (try to avoid this), separate them strictly from real
+changes.
+
+<p>NOTE: If you had to put if(){ .. } over a large (&gt; 5 lines) chunk of code,
+then either do NOT change the indentation of the inner part within (do not
+move it to the right)! or do so in a separate commit
+</p>
+</li><li> Always fill out the commit log message. Describe in a few lines what you
+changed and why. You can refer to mailing list postings if you fix a
+particular bug. Comments such as &quot;fixed!&quot; or &quot;Changed it.&quot; are unacceptable.
+Recommended format:
+
+<div class="example">
+<pre class="example">area changed: Short 1 line description
+
+details describing what and why and giving references.
+</pre></div>
+
+</li><li> Make sure the author of the commit is set correctly. (see git commit &ndash;author)
+If you apply a patch, send an
+answer to ffmpeg-devel (or wherever you got the patch from) saying that
+you applied the patch.
+
+</li><li> When applying patches that have been discussed (at length) on the mailing
+list, reference the thread in the log message.
+
+</li><li> Do NOT commit to code actively maintained by others without permission.
+Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
+timeframe (12h for build failures and security fixes, 3 days small changes,
+1 week for big patches) then commit your patch if you think it is OK.
+Also note, the maintainer can simply ask for more time to review!
+
+</li><li> Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
+are sent there and reviewed by all the other developers. Bugs and possible
+improvements or general questions regarding commits are discussed there. We
+expect you to react if problems with your code are uncovered.
+
+</li><li> Update the documentation if you change behavior or add features. If you are
+unsure how best to do this, send a patch to ffmpeg-devel, the documentation
+maintainer(s) will review and commit your stuff.
+
+</li><li> Try to keep important discussions and requests (also) on the public
+developer mailing list, so that all developers can benefit from them.
+
+</li><li> Never write to unallocated memory, never write over the end of arrays,
+always check values read from some untrusted source before using them
+as array index or other risky things.
+
+</li><li> Remember to check if you need to bump versions for the specific libav*
+parts (libavutil, libavcodec, libavformat) you are changing. You need
+to change the version integer.
+Incrementing the first component means no backward compatibility to
+previous versions (e.g. removal of a function from the public API).
+Incrementing the second component means backward compatible change
+(e.g. addition of a function to the public API or extension of an
+existing data structure).
+Incrementing the third component means a noteworthy binary compatible
+change (e.g. encoder bug fix that matters for the decoder). The third
+component always starts at 100 to distinguish FFmpeg from Libav.
+
+</li><li> Compiler warnings indicate potential bugs or code with bad style. If a type of
+warning always points to correct and clean code, that warning should
+be disabled, not the code changed.
+Thus the remaining warnings can either be bugs or correct code.
+If it is a bug, the bug has to be fixed. If it is not, the code should
+be changed to not generate a warning unless that causes a slowdown
+or obfuscates the code.
+
+</li><li> Make sure that no parts of the codebase that you maintain are missing from the
+<samp>MAINTAINERS</samp> file. If something that you want to maintain is missing add it with
+your name after it.
+If at some point you no longer want to maintain some code, then please help
+finding a new maintainer and also don&rsquo;t forget updating the <samp>MAINTAINERS</samp> file.
+</li></ol>
+
+<p>We think our rules are not too hard. If you have comments, contact us.
+</p>
+<a name="Submitting-patches"></a><a name="Submitting-patches-1"></a>
+<h3 class="section">1.5 Submitting patches<span class="pull-right"><a class="anchor hidden-xs" href="#Submitting-patches-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Submitting-patches-1" aria-hidden="true">TOC</a></span></h3>
+
+<p>First, read the <a href="#Coding-Rules">Coding Rules</a> above if you did not yet, in particular
+the rules regarding patch submission.
+</p>
+<p>When you submit your patch, please use <code>git format-patch</code> or
+<code>git send-email</code>. We cannot read other diffs :-)
+</p>
+<p>Also please do not submit a patch which contains several unrelated changes.
+Split it into separate, self-contained pieces. This does not mean splitting
+file by file. Instead, make the patch as small as possible while still
+keeping it as a logical unit that contains an individual change, even
+if it spans multiple files. This makes reviewing your patches much easier
+for us and greatly increases your chances of getting your patch applied.
+</p>
+<p>Use the patcheck tool of FFmpeg to check your patch.
+The tool is located in the tools directory.
+</p>
+<p>Run the <a href="#Regression-tests">Regression tests</a> before submitting a patch in order to verify
+it does not cause unexpected problems.
+</p>
+<p>It also helps quite a bit if you tell us what the patch does (for example
+&rsquo;replaces lrint by lrintf&rsquo;), and why (for example &rsquo;*BSD isn&rsquo;t C99 compliant
+and has no lrint()&rsquo;)
+</p>
+<p>Also please if you send several patches, send each patch as a separate mail,
+do not attach several unrelated patches to the same mail.
+</p>
+<p>Patches should be posted to the
+<a href="https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel">ffmpeg-devel</a>
+mailing list. Use <code>git send-email</code> when possible since it will properly
+send patches without requiring extra care. If you cannot, then send patches
+as base64-encoded attachments, so your patch is not trashed during
+transmission.
+</p>
+<p>Your patch will be reviewed on the mailing list. You will likely be asked
+to make some changes and are expected to send in an improved version that
+incorporates the requests from the review. This process may go through
+several iterations. Once your patch is deemed good enough, some developer
+will pick it up and commit it to the official FFmpeg tree.
+</p>
+<p>Give us a few days to react. But if some time passes without reaction,
+send a reminder by email. Your patch should eventually be dealt with.
+</p>
+
+<a name="New-codecs-or-formats-checklist"></a>
+<h3 class="section">1.6 New codecs or formats checklist<span class="pull-right"><a class="anchor hidden-xs" href="#New-codecs-or-formats-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-New-codecs-or-formats-checklist" aria-hidden="true">TOC</a></span></h3>
+
+<ol>
+<li> Did you use av_cold for codec initialization and close functions?
+
+</li><li> Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
+AVInputFormat/AVOutputFormat struct?
+
+</li><li> Did you bump the minor version number (and reset the micro version
+number) in <samp>libavcodec/version.h</samp> or <samp>libavformat/version.h</samp>?
+
+</li><li> Did you register it in <samp>allcodecs.c</samp> or <samp>allformats.c</samp>?
+
+</li><li> Did you add the AVCodecID to <samp>avcodec.h</samp>?
+When adding new codec IDs, also add an entry to the codec descriptor
+list in <samp>libavcodec/codec_desc.c</samp>.
+
+</li><li> If it has a FourCC, did you add it to <samp>libavformat/riff.c</samp>,
+even if it is only a decoder?
+
+</li><li> Did you add a rule to compile the appropriate files in the Makefile?
+Remember to do this even if you&rsquo;re just adding a format to a file that is
+already being compiled by some other rule, like a raw demuxer.
+
+</li><li> Did you add an entry to the table of supported formats or codecs in
+<samp>doc/general.texi</samp>?
+
+</li><li> Did you add an entry in the Changelog?
+
+</li><li> If it depends on a parser or a library, did you add that dependency in
+configure?
+
+</li><li> Did you <code>git add</code> the appropriate files before committing?
+
+</li><li> Did you make sure it compiles standalone, i.e. with
+<code>configure --disable-everything --enable-decoder=foo</code>
+(or <code>--enable-demuxer</code> or whatever your component is)?
+</li></ol>
+
+
+<a name="patch-submission-checklist"></a>
+<h3 class="section">1.7 patch submission checklist<span class="pull-right"><a class="anchor hidden-xs" href="#patch-submission-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-patch-submission-checklist" aria-hidden="true">TOC</a></span></h3>
+
+<ol>
+<li> Does <code>make fate</code> pass with the patch applied?
+
+</li><li> Was the patch generated with git format-patch or send-email?
+
+</li><li> Did you sign off your patch? (git commit -s)
+See <a href="http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches">http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches</a> for the meaning
+of sign off.
+
+</li><li> Did you provide a clear git commit log message?
+
+</li><li> Is the patch against latest FFmpeg git master branch?
+
+</li><li> Are you subscribed to ffmpeg-devel?
+(the list is subscribers only due to spam)
+
+</li><li> Have you checked that the changes are minimal, so that the same cannot be
+achieved with a smaller patch and/or simpler final code?
+
+</li><li> If the change is to speed critical code, did you benchmark it?
+
+</li><li> If you did any benchmarks, did you provide them in the mail?
+
+</li><li> Have you checked that the patch does not introduce buffer overflows or
+other security issues?
+
+</li><li> Did you test your decoder or demuxer against damaged data? If no, see
+tools/trasher, the noise bitstream filter, and
+<a href="http://caca.zoy.org/wiki/zzuf">zzuf</a>. Your decoder or demuxer
+should not crash, end in a (near) infinite loop, or allocate ridiculous
+amounts of memory when fed damaged data.
+
+</li><li> Did you test your decoder or demuxer against sample files?
+Samples may be obtained at <a href="https://samples.ffmpeg.org">https://samples.ffmpeg.org</a>.
+
+</li><li> Does the patch not mix functional and cosmetic changes?
+
+</li><li> Did you add tabs or trailing whitespace to the code? Both are forbidden.
+
+</li><li> Is the patch attached to the email you send?
+
+</li><li> Is the mime type of the patch correct? It should be text/x-diff or
+text/x-patch or at least text/plain and not application/octet-stream.
+
+</li><li> If the patch fixes a bug, did you provide a verbose analysis of the bug?
+
+</li><li> If the patch fixes a bug, did you provide enough information, including
+a sample, so the bug can be reproduced and the fix can be verified?
+Note please do not attach samples &gt;100k to mails but rather provide a
+URL, you can upload to ftp://upload.ffmpeg.org
+
+</li><li> Did you provide a verbose summary about what the patch does change?
+
+</li><li> Did you provide a verbose explanation why it changes things like it does?
+
+</li><li> Did you provide a verbose summary of the user visible advantages and
+disadvantages if the patch is applied?
+
+</li><li> Did you provide an example so we can verify the new feature added by the
+patch easily?
+
+</li><li> If you added a new file, did you insert a license header? It should be
+taken from FFmpeg, not randomly copied and pasted from somewhere else.
+
+</li><li> You should maintain alphabetical order in alphabetically ordered lists as
+long as doing so does not break API/ABI compatibility.
+
+</li><li> Lines with similar content should be aligned vertically when doing so
+improves readability.
+
+</li><li> Consider to add a regression test for your code.
+
+</li><li> If you added YASM code please check that things still work with &ndash;disable-yasm
+
+</li><li> Make sure you check the return values of function and return appropriate
+error codes. Especially memory allocation functions like <code>av_malloc()</code>
+are notoriously left unchecked, which is a serious problem.
+
+</li><li> Test your code with valgrind and or Address Sanitizer to ensure it&rsquo;s free
+of leaks, out of array accesses, etc.
+</li></ol>
+
+<a name="Patch-review-process"></a>
+<h3 class="section">1.8 Patch review process<span class="pull-right"><a class="anchor hidden-xs" href="#Patch-review-process" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Patch-review-process" aria-hidden="true">TOC</a></span></h3>
+
+<p>All patches posted to ffmpeg-devel will be reviewed, unless they contain a
+clear note that the patch is not for the git master branch.
+Reviews and comments will be posted as replies to the patch on the
+mailing list. The patch submitter then has to take care of every comment,
+that can be by resubmitting a changed patch or by discussion. Resubmitted
+patches will themselves be reviewed like any other patch. If at some point
+a patch passes review with no comments then it is approved, that can for
+simple and small patches happen immediately while large patches will generally
+have to be changed and reviewed many times before they are approved.
+After a patch is approved it will be committed to the repository.
+</p>
+<p>We will review all submitted patches, but sometimes we are quite busy so
+especially for large patches this can take several weeks.
+</p>
+<p>If you feel that the review process is too slow and you are willing to try to
+take over maintainership of the area of code you change then just clone
+git master and maintain the area of code there. We will merge each area from
+where its best maintained.
+</p>
+<p>When resubmitting patches, please do not make any significant changes
+not related to the comments received during review. Such patches will
+be rejected. Instead, submit significant changes or new features as
+separate patches.
+</p>
+<p>Everyone is welcome to review patches. Also if you are waiting for your patch
+to be reviewed, please consider helping to review other patches, that is a great
+way to get everyone&rsquo;s patches reviewed sooner.
+</p>
+<a name="Regression-tests"></a><a name="Regression-tests-1"></a>
+<h3 class="section">1.9 Regression tests<span class="pull-right"><a class="anchor hidden-xs" href="#Regression-tests-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Regression-tests-1" aria-hidden="true">TOC</a></span></h3>
+
+<p>Before submitting a patch (or committing to the repository), you should at least
+test that you did not break anything.
+</p>
+<p>Running &rsquo;make fate&rsquo; accomplishes this, please see <a href="fate.html">fate.html</a> for details.
+</p>
+<p>[Of course, some patches may change the results of the regression tests. In
+this case, the reference results of the regression tests shall be modified
+accordingly].
+</p>
+<a name="Adding-files-to-the-fate_002dsuite-dataset"></a>
+<h4 class="subsection">1.9.1 Adding files to the fate-suite dataset<span class="pull-right"><a class="anchor hidden-xs" href="#Adding-files-to-the-fate_002dsuite-dataset" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Adding-files-to-the-fate_002dsuite-dataset" aria-hidden="true">TOC</a></span></h4>
+
+<p>When there is no muxer or encoder available to generate test media for a
+specific test then the media has to be included in the fate-suite.
+First please make sure that the sample file is as small as possible to test the
+respective decoder or demuxer sufficiently. Large files increase network
+bandwidth and disk space requirements.
+Once you have a working fate test and fate sample, provide in the commit
+message or introductory message for the patch series that you post to
+the ffmpeg-devel mailing list, a direct link to download the sample media.
+</p>
+
+<a name="Visualizing-Test-Coverage"></a>
+<h4 class="subsection">1.9.2 Visualizing Test Coverage<span class="pull-right"><a class="anchor hidden-xs" href="#Visualizing-Test-Coverage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Visualizing-Test-Coverage" aria-hidden="true">TOC</a></span></h4>
+
+<p>The FFmpeg build system allows visualizing the test coverage in an easy
+manner with the coverage tools <code>gcov</code>/<code>lcov</code>.  This involves
+the following steps:
+</p>
+<ol>
+<li> Configure to compile with instrumentation enabled:
+    <code>configure --toolchain=gcov</code>.
+
+</li><li> Run your test case, either manually or via FATE. This can be either
+    the full FATE regression suite, or any arbitrary invocation of any
+    front-end tool provided by FFmpeg, in any combination.
+
+</li><li> Run <code>make lcov</code> to generate coverage data in HTML format.
+
+</li><li> View <code>lcov/index.html</code> in your preferred HTML viewer.
+</li></ol>
+
+<p>You can use the command <code>make lcov-reset</code> to reset the coverage
+measurements. You will need to rerun <code>make lcov</code> after running a
+new test.
+</p>
+<a name="Using-Valgrind"></a>
+<h4 class="subsection">1.9.3 Using Valgrind<span class="pull-right"><a class="anchor hidden-xs" href="#Using-Valgrind" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-Valgrind" aria-hidden="true">TOC</a></span></h4>
+
+<p>The configure script provides a shortcut for using valgrind to spot bugs
+related to memory handling. Just add the option
+<code>--toolchain=valgrind-memcheck</code> or <code>--toolchain=valgrind-massif</code>
+to your configure line, and reasonable defaults will be set for running
+FATE under the supervision of either the <strong>memcheck</strong> or the
+<strong>massif</strong> tool of the valgrind suite.
+</p>
+<p>In case you need finer control over how valgrind is invoked, use the
+<code>--target-exec='valgrind &lt;your_custom_valgrind_options&gt;</code> option in
+your configure line instead.
+</p>
+<a name="Release-process"></a><a name="Release-process-1"></a>
+<h3 class="section">1.10 Release process<span class="pull-right"><a class="anchor hidden-xs" href="#Release-process-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Release-process-1" aria-hidden="true">TOC</a></span></h3>
+
+<p>FFmpeg maintains a set of <strong>release branches</strong>, which are the
+recommended deliverable for system integrators and distributors (such as
+Linux distributions, etc.). At regular times, a <strong>release
+manager</strong> prepares, tests and publishes tarballs on the
+<a href="https://ffmpeg.org">https://ffmpeg.org</a> website.
+</p>
+<p>There are two kinds of releases:
+</p>
+<ol>
+<li> <strong>Major releases</strong> always include the latest and greatest
+features and functionality.
+
+</li><li> <strong>Point releases</strong> are cut from <strong>release</strong> branches,
+which are named <code>release/X</code>, with <code>X</code> being the release
+version number.
+</li></ol>
+
+<p>Note that we promise to our users that shared libraries from any FFmpeg
+release never break programs that have been <strong>compiled</strong> against
+previous versions of <strong>the same release series</strong> in any case!
+</p>
+<p>However, from time to time, we do make API changes that require adaptations
+in applications. Such changes are only allowed in (new) major releases and
+require further steps such as bumping library version numbers and/or
+adjustments to the symbol versioning file. Please discuss such changes
+on the <strong>ffmpeg-devel</strong> mailing list in time to allow forward planning.
+</p>
+<a name="Criteria-for-Point-Releases"></a><a name="Criteria-for-Point-Releases-1"></a>
+<h4 class="subsection">1.10.1 Criteria for Point Releases<span class="pull-right"><a class="anchor hidden-xs" href="#Criteria-for-Point-Releases-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Criteria-for-Point-Releases-1" aria-hidden="true">TOC</a></span></h4>
+
+<p>Changes that match the following criteria are valid candidates for
+inclusion into a point release:
+</p>
+<ol>
+<li> Fixes a security issue, preferably identified by a <strong>CVE
+number</strong> issued by <a href="http://cve.mitre.org/">http://cve.mitre.org/</a>.
+
+</li><li> Fixes a documented bug in <a href="https://trac.ffmpeg.org">https://trac.ffmpeg.org</a>.
+
+</li><li> Improves the included documentation.
+
+</li><li> Retains both source code and binary compatibility with previous
+point releases of the same release branch.
+</li></ol>
+
+<p>The order for checking the rules is (1 OR 2 OR 3) AND 4.
+</p>
+
+<a name="Release-Checklist"></a>
+<h4 class="subsection">1.10.2 Release Checklist<span class="pull-right"><a class="anchor hidden-xs" href="#Release-Checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Release-Checklist" aria-hidden="true">TOC</a></span></h4>
+
+<p>The release process involves the following steps:
+</p>
+<ol>
+<li> Ensure that the <samp>RELEASE</samp> file contains the version number for
+the upcoming release.
+
+</li><li> Add the release at <a href="https://trac.ffmpeg.org/admin/ticket/versions">https://trac.ffmpeg.org/admin/ticket/versions</a>.
+
+</li><li> Announce the intent to do a release to the mailing list.
+
+</li><li> Make sure all relevant security fixes have been backported. See
+<a href="https://ffmpeg.org/security.html">https://ffmpeg.org/security.html</a>.
+
+</li><li> Ensure that the FATE regression suite still passes in the release
+branch on at least <strong>i386</strong> and <strong>amd64</strong>
+(cf. <a href="#Regression-tests">Regression tests</a>).
+
+</li><li> Prepare the release tarballs in <code>bz2</code> and <code>gz</code> formats, and
+supplementing files that contain <code>gpg</code> signatures
+
+</li><li> Publish the tarballs at <a href="https://ffmpeg.org/releases">https://ffmpeg.org/releases</a>. Create and
+push an annotated tag in the form <code>nX</code>, with <code>X</code>
+containing the version number.
+
+</li><li> Propose and send a patch to the <strong>ffmpeg-devel</strong> mailing list
+with a news entry for the website.
+
+</li><li> Publish the news entry.
+
+</li><li> Send announcement to the mailing list.
+</li></ol>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,46 @@
+# use pkg-config for getting CFLAGS and LDLIBS
+FFMPEG_LIBS=    libavdevice                        \
+                libavformat                        \
+                libavfilter                        \
+                libavcodec                         \
+                libswresample                      \
+                libswscale                         \
+                libavutil                          \
+
+CFLAGS += -Wall -g
+CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
+LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
+
+EXAMPLES=       avio_dir_cmd                       \
+                avio_reading                       \
+                decoding_encoding                  \
+                demuxing_decoding                  \
+                extract_mvs                        \
+                filtering_video                    \
+                filtering_audio                    \
+                http_multiclient                   \
+                metadata                           \
+                muxing                             \
+                remuxing                           \
+                resampling_audio                   \
+                scaling_video                      \
+                transcode_aac                      \
+                transcoding                        \
+
+OBJS=$(addsuffix .o,$(EXAMPLES))
+
+# the following examples make explicit use of the math library
+avcodec:           LDLIBS += -lm
+decoding_encoding: LDLIBS += -lm
+muxing:            LDLIBS += -lm
+resampling_audio:  LDLIBS += -lm
+
+.phony: all clean-test clean
+
+all: $(OBJS) $(EXAMPLES)
+
+clean-test:
+	$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
+
+clean: clean-test
+	$(RM) $(EXAMPLES) $(OBJS)
@@ -0,0 +1,23 @@
+FFmpeg examples README
+----------------------
+
+Both following use cases rely on pkg-config and make, thus make sure
+that you have them installed and working on your system.
+
+
+Method 1: build the installed examples in a generic read/write user directory
+
+Copy to a read/write user directory and just use "make", it will link
+to the libraries on your system, assuming the PKG_CONFIG_PATH is
+correctly configured.
+
+Method 2: build the examples in-tree
+
+Assuming you are in the source FFmpeg checkout directory, you need to build
+FFmpeg (no need to make install in any prefix). Then just run "make examples".
+This will build the examples using the FFmpeg build system. You can clean those
+examples using "make examplesclean"
+
+If you want to try the dedicated Makefile examples (to emulate the first
+method), go into doc/examples and run a command such as
+PKG_CONFIG_PATH=pc-uninstalled make.
@@ -0,0 +1,180 @@
+/*
+ * Copyright (c) 2014 Lukasz Marek
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+
+static const char *type_string(int type)
+{
+    switch (type) {
+    case AVIO_ENTRY_DIRECTORY:
+        return "<DIR>";
+    case AVIO_ENTRY_FILE:
+        return "<FILE>";
+    case AVIO_ENTRY_BLOCK_DEVICE:
+        return "<BLOCK DEVICE>";
+    case AVIO_ENTRY_CHARACTER_DEVICE:
+        return "<CHARACTER DEVICE>";
+    case AVIO_ENTRY_NAMED_PIPE:
+        return "<PIPE>";
+    case AVIO_ENTRY_SYMBOLIC_LINK:
+        return "<LINK>";
+    case AVIO_ENTRY_SOCKET:
+        return "<SOCKET>";
+    case AVIO_ENTRY_SERVER:
+        return "<SERVER>";
+    case AVIO_ENTRY_SHARE:
+        return "<SHARE>";
+    case AVIO_ENTRY_WORKGROUP:
+        return "<WORKGROUP>";
+    case AVIO_ENTRY_UNKNOWN:
+    default:
+        break;
+    }
+    return "<UNKNOWN>";
+}
+
+static int list_op(const char *input_dir)
+{
+    AVIODirEntry *entry = NULL;
+    AVIODirContext *ctx = NULL;
+    int cnt, ret;
+    char filemode[4], uid_and_gid[20];
+
+    if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
+        goto fail;
+    }
+
+    cnt = 0;
+    for (;;) {
+        if ((ret = avio_read_dir(ctx, &entry)) < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
+            goto fail;
+        }
+        if (!entry)
+            break;
+        if (entry->filemode == -1) {
+            snprintf(filemode, 4, "???");
+        } else {
+            snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
+        }
+        snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
+        if (cnt == 0)
+            av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
+                   "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
+                   "ACCESSED", "STATUS_CHANGED");
+        av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
+               type_string(entry->type),
+               entry->size,
+               entry->name,
+               uid_and_gid,
+               filemode,
+               entry->modification_timestamp,
+               entry->access_timestamp,
+               entry->status_change_timestamp);
+        avio_free_directory_entry(&entry);
+        cnt++;
+    };
+
+  fail:
+    avio_close_dir(&ctx);
+    return ret;
+}
+
+static int del_op(const char *url)
+{
+    int ret = avpriv_io_delete(url);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
+    return ret;
+}
+
+static int move_op(const char *src, const char *dst)
+{
+    int ret = avpriv_io_move(src, dst);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
+    return ret;
+}
+
+
+static void usage(const char *program_name)
+{
+    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
+            "API example program to show how to manipulate resources "
+            "accessed through AVIOContext.\n"
+            "OPERATIONS:\n"
+            "list      list content of the directory\n"
+            "move      rename content in directory\n"
+            "del       delete content in directory\n",
+            program_name);
+}
+
+int main(int argc, char *argv[])
+{
+    const char *op = NULL;
+    int ret;
+
+    av_log_set_level(AV_LOG_DEBUG);
+
+    if (argc < 2) {
+        usage(argv[0]);
+        return 1;
+    }
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+    avformat_network_init();
+
+    op = argv[1];
+    if (strcmp(op, "list") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = list_op(argv[2]);
+        }
+    } else if (strcmp(op, "del") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = del_op(argv[2]);
+        }
+    } else if (strcmp(op, "move") == 0) {
+        if (argc < 4) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = move_op(argv[2], argv[3]);
+        }
+    } else {
+        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
+        ret = AVERROR(EINVAL);
+    }
+
+    avformat_network_deinit();
+
+    return ret < 0 ? 1 : 0;
+}
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2014 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat AVIOContext API example.
+ *
+ * Make libavformat demuxer access media content through a custom
+ * AVIOContext read callback.
+ * @example avio_reading.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+#include <libavutil/file.h>
+
+struct buffer_data {
+    uint8_t *ptr;
+    size_t size; ///< size left in the buffer
+};
+
+static int read_packet(void *opaque, uint8_t *buf, int buf_size)
+{
+    struct buffer_data *bd = (struct buffer_data *)opaque;
+    buf_size = FFMIN(buf_size, bd->size);
+
+    printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
+
+    /* copy internal buffer data to buf */
+    memcpy(buf, bd->ptr, buf_size);
+    bd->ptr  += buf_size;
+    bd->size -= buf_size;
+
+    return buf_size;
+}
+
+int main(int argc, char *argv[])
+{
+    AVFormatContext *fmt_ctx = NULL;
+    AVIOContext *avio_ctx = NULL;
+    uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
+    size_t buffer_size, avio_ctx_buffer_size = 4096;
+    char *input_filename = NULL;
+    int ret = 0;
+    struct buffer_data bd = { 0 };
+
+    if (argc != 2) {
+        fprintf(stderr, "usage: %s input_file\n"
+                "API example program to show how to read from a custom buffer "
+                "accessed through AVIOContext.\n", argv[0]);
+        return 1;
+    }
+    input_filename = argv[1];
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+
+    /* slurp file content into buffer */
+    ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
+    if (ret < 0)
+        goto end;
+
+    /* fill opaque structure used by the AVIOContext read callback */
+    bd.ptr  = buffer;
+    bd.size = buffer_size;
+
+    if (!(fmt_ctx = avformat_alloc_context())) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
+    if (!avio_ctx_buffer) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
+                                  0, &bd, &read_packet, NULL, NULL);
+    if (!avio_ctx) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    fmt_ctx->pb = avio_ctx;
+
+    ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open input\n");
+        goto end;
+    }
+
+    ret = avformat_find_stream_info(fmt_ctx, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        goto end;
+    }
+
+    av_dump_format(fmt_ctx, 0, input_filename, 0);
+
+end:
+    avformat_close_input(&fmt_ctx);
+    /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
+    if (avio_ctx) {
+        av_freep(&avio_ctx->buffer);
+        av_freep(&avio_ctx);
+    }
+    av_file_unmap(buffer, buffer_size);
+
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}
@@ -0,0 +1,665 @@
+/*
+ * Copyright (c) 2001 Fabrice Bellard
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavcodec API use example.
+ *
+ * @example decoding_encoding.c
+ * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
+ * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
+ * format handling
+ */
+
+#include <math.h>
+
+#include <libavutil/opt.h>
+#include <libavcodec/avcodec.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/common.h>
+#include <libavutil/imgutils.h>
+#include <libavutil/mathematics.h>
+#include <libavutil/samplefmt.h>
+
+#define INBUF_SIZE 4096
+#define AUDIO_INBUF_SIZE 20480
+#define AUDIO_REFILL_THRESH 4096
+
+/* check that a given sample format is supported by the encoder */
+static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
+{
+    const enum AVSampleFormat *p = codec->sample_fmts;
+
+    while (*p != AV_SAMPLE_FMT_NONE) {
+        if (*p == sample_fmt)
+            return 1;
+        p++;
+    }
+    return 0;
+}
+
+/* just pick the highest supported samplerate */
+static int select_sample_rate(AVCodec *codec)
+{
+    const int *p;
+    int best_samplerate = 0;
+
+    if (!codec->supported_samplerates)
+        return 44100;
+
+    p = codec->supported_samplerates;
+    while (*p) {
+        best_samplerate = FFMAX(*p, best_samplerate);
+        p++;
+    }
+    return best_samplerate;
+}
+
+/* select layout with the highest channel count */
+static int select_channel_layout(AVCodec *codec)
+{
+    const uint64_t *p;
+    uint64_t best_ch_layout = 0;
+    int best_nb_channels   = 0;
+
+    if (!codec->channel_layouts)
+        return AV_CH_LAYOUT_STEREO;
+
+    p = codec->channel_layouts;
+    while (*p) {
+        int nb_channels = av_get_channel_layout_nb_channels(*p);
+
+        if (nb_channels > best_nb_channels) {
+            best_ch_layout    = *p;
+            best_nb_channels = nb_channels;
+        }
+        p++;
+    }
+    return best_ch_layout;
+}
+
+/*
+ * Audio encoding example
+ */
+static void audio_encode_example(const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    AVFrame *frame;
+    AVPacket pkt;
+    int i, j, k, ret, got_output;
+    int buffer_size;
+    FILE *f;
+    uint16_t *samples;
+    float t, tincr;
+
+    printf("Encode audio file %s\n", filename);
+
+    /* find the MP2 encoder */
+    codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate audio codec context\n");
+        exit(1);
+    }
+
+    /* put sample parameters */
+    c->bit_rate = 64000;
+
+    /* check that the encoder supports s16 pcm input */
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+    if (!check_sample_fmt(codec, c->sample_fmt)) {
+        fprintf(stderr, "Encoder does not support sample format %s",
+                av_get_sample_fmt_name(c->sample_fmt));
+        exit(1);
+    }
+
+    /* select other audio parameters supported by the encoder */
+    c->sample_rate    = select_sample_rate(codec);
+    c->channel_layout = select_channel_layout(codec);
+    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "wb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    /* frame containing input raw audio */
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate audio frame\n");
+        exit(1);
+    }
+
+    frame->nb_samples     = c->frame_size;
+    frame->format         = c->sample_fmt;
+    frame->channel_layout = c->channel_layout;
+
+    /* the codec gives us the frame size, in samples,
+     * we calculate the size of the samples buffer in bytes */
+    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
+                                             c->sample_fmt, 0);
+    if (buffer_size < 0) {
+        fprintf(stderr, "Could not get sample buffer size\n");
+        exit(1);
+    }
+    samples = av_malloc(buffer_size);
+    if (!samples) {
+        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
+                buffer_size);
+        exit(1);
+    }
+    /* setup the data pointers in the AVFrame */
+    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                                   (const uint8_t*)samples, buffer_size, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not setup audio frame\n");
+        exit(1);
+    }
+
+    /* encode a single tone sound */
+    t = 0;
+    tincr = 2 * M_PI * 440.0 / c->sample_rate;
+    for (i = 0; i < 200; i++) {
+        av_init_packet(&pkt);
+        pkt.data = NULL; // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        for (j = 0; j < c->frame_size; j++) {
+            samples[2*j] = (int)(sin(t) * 10000);
+
+            for (k = 1; k < c->channels; k++)
+                samples[2*j + k] = samples[2*j];
+            t += tincr;
+        }
+        /* encode the samples */
+        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding audio frame\n");
+            exit(1);
+        }
+        if (got_output) {
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* get the delayed frames */
+    for (got_output = 1; got_output; i++) {
+        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+    fclose(f);
+
+    av_freep(&samples);
+    av_frame_free(&frame);
+    avcodec_close(c);
+    av_free(c);
+}
+
+/*
+ * Audio decoding.
+ */
+static void audio_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int len;
+    FILE *f, *outfile;
+    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    AVPacket avpkt;
+    AVFrame *decoded_frame = NULL;
+
+    av_init_packet(&avpkt);
+
+    printf("Decode audio file %s to %s\n", filename, outfilename);
+
+    /* find the mpeg audio decoder */
+    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate audio codec context\n");
+        exit(1);
+    }
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+    outfile = fopen(outfilename, "wb");
+    if (!outfile) {
+        av_free(c);
+        exit(1);
+    }
+
+    /* decode until eof */
+    avpkt.data = inbuf;
+    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
+
+    while (avpkt.size > 0) {
+        int i, ch;
+        int got_frame = 0;
+
+        if (!decoded_frame) {
+            if (!(decoded_frame = av_frame_alloc())) {
+                fprintf(stderr, "Could not allocate audio frame\n");
+                exit(1);
+            }
+        }
+
+        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
+        if (len < 0) {
+            fprintf(stderr, "Error while decoding\n");
+            exit(1);
+        }
+        if (got_frame) {
+            /* if a frame has been decoded, output it */
+            int data_size = av_get_bytes_per_sample(c->sample_fmt);
+            if (data_size < 0) {
+                /* This should not occur, checking just for paranoia */
+                fprintf(stderr, "Failed to calculate data size\n");
+                exit(1);
+            }
+            for (i=0; i<decoded_frame->nb_samples; i++)
+                for (ch=0; ch<c->channels; ch++)
+                    fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
+        }
+        avpkt.size -= len;
+        avpkt.data += len;
+        avpkt.dts =
+        avpkt.pts = AV_NOPTS_VALUE;
+        if (avpkt.size < AUDIO_REFILL_THRESH) {
+            /* Refill the input buffer, to avoid trying to decode
+             * incomplete frames. Instead of this, one could also use
+             * a parser, or use a proper container format through
+             * libavformat. */
+            memmove(inbuf, avpkt.data, avpkt.size);
+            avpkt.data = inbuf;
+            len = fread(avpkt.data + avpkt.size, 1,
+                        AUDIO_INBUF_SIZE - avpkt.size, f);
+            if (len > 0)
+                avpkt.size += len;
+        }
+    }
+
+    fclose(outfile);
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_frame_free(&decoded_frame);
+}
+
+/*
+ * Video encoding example
+ */
+static void video_encode_example(const char *filename, int codec_id)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int i, ret, x, y, got_output;
+    FILE *f;
+    AVFrame *frame;
+    AVPacket pkt;
+    uint8_t endcode[] = { 0, 0, 1, 0xb7 };
+
+    printf("Encode video file %s\n", filename);
+
+    /* find the mpeg1 video encoder */
+    codec = avcodec_find_encoder(codec_id);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate video codec context\n");
+        exit(1);
+    }
+
+    /* put sample parameters */
+    c->bit_rate = 400000;
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* frames per second */
+    c->time_base = (AVRational){1,25};
+    /* emit one intra frame every ten frames
+     * check frame pict_type before passing frame
+     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
+     * then gop_size is ignored and the output of encoder
+     * will always be I frame irrespective to gop_size
+     */
+    c->gop_size = 10;
+    c->max_b_frames = 1;
+    c->pix_fmt = AV_PIX_FMT_YUV420P;
+
+    if (codec_id == AV_CODEC_ID_H264)
+        av_opt_set(c->priv_data, "preset", "slow", 0);
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "wb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate video frame\n");
+        exit(1);
+    }
+    frame->format = c->pix_fmt;
+    frame->width  = c->width;
+    frame->height = c->height;
+
+    /* the image can be allocated by any means and av_image_alloc() is
+     * just the most convenient way if av_malloc() is to be used */
+    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
+                         c->pix_fmt, 32);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate raw picture buffer\n");
+        exit(1);
+    }
+
+    /* encode 1 second of video */
+    for (i = 0; i < 25; i++) {
+        av_init_packet(&pkt);
+        pkt.data = NULL;    // packet data will be allocated by the encoder
+        pkt.size = 0;
+
+        fflush(stdout);
+        /* prepare a dummy image */
+        /* Y */
+        for (y = 0; y < c->height; y++) {
+            for (x = 0; x < c->width; x++) {
+                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
+            }
+        }
+
+        /* Cb and Cr */
+        for (y = 0; y < c->height/2; y++) {
+            for (x = 0; x < c->width/2; x++) {
+                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
+                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
+            }
+        }
+
+        frame->pts = i;
+
+        /* encode the image */
+        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* get the delayed frames */
+    for (got_output = 1; got_output; i++) {
+        fflush(stdout);
+
+        ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
+        if (ret < 0) {
+            fprintf(stderr, "Error encoding frame\n");
+            exit(1);
+        }
+
+        if (got_output) {
+            printf("Write frame %3d (size=%5d)\n", i, pkt.size);
+            fwrite(pkt.data, 1, pkt.size, f);
+            av_packet_unref(&pkt);
+        }
+    }
+
+    /* add sequence end code to have a real mpeg file */
+    fwrite(endcode, 1, sizeof(endcode), f);
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_freep(&frame->data[0]);
+    av_frame_free(&frame);
+    printf("\n");
+}
+
+/*
+ * Video decoding example
+ */
+
+static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
+                     char *filename)
+{
+    FILE *f;
+    int i;
+
+    f = fopen(filename,"w");
+    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
+    for (i = 0; i < ysize; i++)
+        fwrite(buf + i * wrap, 1, xsize, f);
+    fclose(f);
+}
+
+static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
+                              AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
+{
+    int len, got_frame;
+    char buf[1024];
+
+    len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
+    if (len < 0) {
+        fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
+        return len;
+    }
+    if (got_frame) {
+        printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
+        fflush(stdout);
+
+        /* the picture is allocated by the decoder, no need to free it */
+        snprintf(buf, sizeof(buf), outfilename, *frame_count);
+        pgm_save(frame->data[0], frame->linesize[0],
+                 frame->width, frame->height, buf);
+        (*frame_count)++;
+    }
+    if (pkt->data) {
+        pkt->size -= len;
+        pkt->data += len;
+    }
+    return 0;
+}
+
+static void video_decode_example(const char *outfilename, const char *filename)
+{
+    AVCodec *codec;
+    AVCodecContext *c= NULL;
+    int frame_count;
+    FILE *f;
+    AVFrame *frame;
+    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    AVPacket avpkt;
+
+    av_init_packet(&avpkt);
+
+    /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
+    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+
+    printf("Decode video file %s to %s\n", filename, outfilename);
+
+    /* find the mpeg1 video decoder */
+    codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
+    if (!codec) {
+        fprintf(stderr, "Codec not found\n");
+        exit(1);
+    }
+
+    c = avcodec_alloc_context3(codec);
+    if (!c) {
+        fprintf(stderr, "Could not allocate video codec context\n");
+        exit(1);
+    }
+
+    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
+        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
+
+    /* For some codecs, such as msmpeg4 and mpeg4, width and height
+       MUST be initialized there because this information is not
+       available in the bitstream. */
+
+    /* open it */
+    if (avcodec_open2(c, codec, NULL) < 0) {
+        fprintf(stderr, "Could not open codec\n");
+        exit(1);
+    }
+
+    f = fopen(filename, "rb");
+    if (!f) {
+        fprintf(stderr, "Could not open %s\n", filename);
+        exit(1);
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate video frame\n");
+        exit(1);
+    }
+
+    frame_count = 0;
+    for (;;) {
+        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
+        if (avpkt.size == 0)
+            break;
+
+        /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
+           and this is the only method to use them because you cannot
+           know the compressed data size before analysing it.
+
+           BUT some other codecs (msmpeg4, mpeg4) are inherently frame
+           based, so you must call them with all the data for one
+           frame exactly. You must also initialize 'width' and
+           'height' before initializing them. */
+
+        /* NOTE2: some codecs allow the raw parameters (frame size,
+           sample rate) to be changed at any frame. We handle this, so
+           you should also take care of it */
+
+        /* here, we use a stream based decoder (mpeg1video), so we
+           feed decoder and see if it could decode a frame */
+        avpkt.data = inbuf;
+        while (avpkt.size > 0)
+            if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
+                exit(1);
+    }
+
+    /* some codecs, such as MPEG, transmit the I and P frame with a
+       latency of one frame. You must do the following to have a
+       chance to get the last frame of the video */
+    avpkt.data = NULL;
+    avpkt.size = 0;
+    decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
+
+    fclose(f);
+
+    avcodec_close(c);
+    av_free(c);
+    av_frame_free(&frame);
+    printf("\n");
+}
+
+int main(int argc, char **argv)
+{
+    const char *output_type;
+
+    /* register all the codecs */
+    avcodec_register_all();
+
+    if (argc < 2) {
+        printf("usage: %s output_type\n"
+               "API example program to decode/encode a media stream with libavcodec.\n"
+               "This program generates a synthetic stream and encodes it to a file\n"
+               "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
+               "The encoded stream is then decoded and written to a raw data output.\n"
+               "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
+               argv[0]);
+        return 1;
+    }
+    output_type = argv[1];
+
+    if (!strcmp(output_type, "h264")) {
+        video_encode_example("test.h264", AV_CODEC_ID_H264);
+    } else if (!strcmp(output_type, "mp2")) {
+        audio_encode_example("test.mp2");
+        audio_decode_example("test.pcm", "test.mp2");
+    } else if (!strcmp(output_type, "mpg")) {
+        video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
+        video_decode_example("test%02d.pgm", "test.mpg");
+    } else {
+        fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
+                output_type);
+        return 1;
+    }
+
+    return 0;
+}
@@ -0,0 +1,383 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * Demuxing and decoding example.
+ *
+ * Show how to use the libavformat and libavcodec API to demux and
+ * decode audio and video data.
+ * @example demuxing_decoding.c
+ */
+
+#include <libavutil/imgutils.h>
+#include <libavutil/samplefmt.h>
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+
+static AVFormatContext *fmt_ctx = NULL;
+static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
+static int width, height;
+static enum AVPixelFormat pix_fmt;
+static AVStream *video_stream = NULL, *audio_stream = NULL;
+static const char *src_filename = NULL;
+static const char *video_dst_filename = NULL;
+static const char *audio_dst_filename = NULL;
+static FILE *video_dst_file = NULL;
+static FILE *audio_dst_file = NULL;
+
+static uint8_t *video_dst_data[4] = {NULL};
+static int      video_dst_linesize[4];
+static int video_dst_bufsize;
+
+static int video_stream_idx = -1, audio_stream_idx = -1;
+static AVFrame *frame = NULL;
+static AVPacket pkt;
+static int video_frame_count = 0;
+static int audio_frame_count = 0;
+
+/* Enable or disable frame reference counting. You are not supposed to support
+ * both paths in your application but pick the one most appropriate to your
+ * needs. Look for the use of refcount in this example to see what are the
+ * differences of API usage between them. */
+static int refcount = 0;
+
+static int decode_packet(int *got_frame, int cached)
+{
+    int ret = 0;
+    int decoded = pkt.size;
+
+    *got_frame = 0;
+
+    if (pkt.stream_index == video_stream_idx) {
+        /* decode video frame */
+        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
+            return ret;
+        }
+
+        if (*got_frame) {
+
+            if (frame->width != width || frame->height != height ||
+                frame->format != pix_fmt) {
+                /* To handle this change, one could call av_image_alloc again and
+                 * decode the following frames into another rawvideo file. */
+                fprintf(stderr, "Error: Width, height and pixel format have to be "
+                        "constant in a rawvideo file, but the width, height or "
+                        "pixel format of the input video changed:\n"
+                        "old: width = %d, height = %d, format = %s\n"
+                        "new: width = %d, height = %d, format = %s\n",
+                        width, height, av_get_pix_fmt_name(pix_fmt),
+                        frame->width, frame->height,
+                        av_get_pix_fmt_name(frame->format));
+                return -1;
+            }
+
+            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
+                   cached ? "(cached)" : "",
+                   video_frame_count++, frame->coded_picture_number,
+                   av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
+
+            /* copy decoded frame to destination buffer:
+             * this is required since rawvideo expects non aligned data */
+            av_image_copy(video_dst_data, video_dst_linesize,
+                          (const uint8_t **)(frame->data), frame->linesize,
+                          pix_fmt, width, height);
+
+            /* write to rawvideo file */
+            fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
+        }
+    } else if (pkt.stream_index == audio_stream_idx) {
+        /* decode audio frame */
+        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
+            return ret;
+        }
+        /* Some audio decoders decode only part of the packet, and have to be
+         * called again with the remainder of the packet data.
+         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
+         * Also, some decoders might over-read the packet. */
+        decoded = FFMIN(ret, pkt.size);
+
+        if (*got_frame) {
+            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
+            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
+                   cached ? "(cached)" : "",
+                   audio_frame_count++, frame->nb_samples,
+                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
+
+            /* Write the raw audio data samples of the first plane. This works
+             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
+             * most audio decoders output planar audio, which uses a separate
+             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
+             * In other words, this code will write only the first audio channel
+             * in these cases.
+             * You should use libswresample or libavfilter to convert the frame
+             * to packed data. */
+            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
+        }
+    }
+
+    /* If we use frame reference counting, we own the data and need
+     * to de-reference it when we don't use it anymore */
+    if (*got_frame && refcount)
+        av_frame_unref(frame);
+
+    return decoded;
+}
+
+static int open_codec_context(int *stream_idx,
+                              AVFormatContext *fmt_ctx, enum AVMediaType type)
+{
+    int ret, stream_index;
+    AVStream *st;
+    AVCodecContext *dec_ctx = NULL;
+    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;
+
+    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
+                av_get_media_type_string(type), src_filename);
+        return ret;
+    } else {
+        stream_index = ret;
+        st = fmt_ctx->streams[stream_index];
+
+        /* find decoder for the stream */
+        dec_ctx = st->codec;
+        dec = avcodec_find_decoder(dec_ctx->codec_id);
+        if (!dec) {
+            fprintf(stderr, "Failed to find %s codec\n",
+                    av_get_media_type_string(type));
+            return AVERROR(EINVAL);
+        }
+
+        /* Init the decoders, with or without reference counting */
+        av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0);
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
+            fprintf(stderr, "Failed to open %s codec\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+        *stream_idx = stream_index;
+    }
+
+    return 0;
+}
+
+static int get_format_from_sample_fmt(const char **fmt,
+                                      enum AVSampleFormat sample_fmt)
+{
+    int i;
+    struct sample_fmt_entry {
+        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
+    } sample_fmt_entries[] = {
+        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
+        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
+        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
+        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
+        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
+    };
+    *fmt = NULL;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
+        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
+        if (sample_fmt == entry->sample_fmt) {
+            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
+            return 0;
+        }
+    }
+
+    fprintf(stderr,
+            "sample format %s is not supported as output format\n",
+            av_get_sample_fmt_name(sample_fmt));
+    return -1;
+}
+
+int main (int argc, char **argv)
+{
+    int ret = 0, got_frame;
+
+    if (argc != 4 && argc != 5) {
+        fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n"
+                "API example program to show how to read frames from an input file.\n"
+                "This program reads frames from a file, decodes them, and writes decoded\n"
+                "video frames to a rawvideo file named video_output_file, and decoded\n"
+                "audio frames to a rawaudio file named audio_output_file.\n\n"
+                "If the -refcount option is specified, the program use the\n"
+                "reference counting frame system which allows keeping a copy of\n"
+                "the data for longer than one decode call.\n"
+                "\n", argv[0]);
+        exit(1);
+    }
+    if (argc == 5 && !strcmp(argv[1], "-refcount")) {
+        refcount = 1;
+        argv++;
+    }
+    src_filename = argv[1];
+    video_dst_filename = argv[2];
+    audio_dst_filename = argv[3];
+
+    /* register all formats and codecs */
+    av_register_all();
+
+    /* open input file, and allocate format context */
+    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
+        fprintf(stderr, "Could not open source file %s\n", src_filename);
+        exit(1);
+    }
+
+    /* retrieve stream information */
+    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        exit(1);
+    }
+
+    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+        video_stream = fmt_ctx->streams[video_stream_idx];
+        video_dec_ctx = video_stream->codec;
+
+        video_dst_file = fopen(video_dst_filename, "wb");
+        if (!video_dst_file) {
+            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
+            ret = 1;
+            goto end;
+        }
+
+        /* allocate image where the decoded image will be put */
+        width = video_dec_ctx->width;
+        height = video_dec_ctx->height;
+        pix_fmt = video_dec_ctx->pix_fmt;
+        ret = av_image_alloc(video_dst_data, video_dst_linesize,
+                             width, height, pix_fmt, 1);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate raw video buffer\n");
+            goto end;
+        }
+        video_dst_bufsize = ret;
+    }
+
+    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
+        audio_stream = fmt_ctx->streams[audio_stream_idx];
+        audio_dec_ctx = audio_stream->codec;
+        audio_dst_file = fopen(audio_dst_filename, "wb");
+        if (!audio_dst_file) {
+            fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
+            ret = 1;
+            goto end;
+        }
+    }
+
+    /* dump input information to stderr */
+    av_dump_format(fmt_ctx, 0, src_filename, 0);
+
+    if (!audio_stream && !video_stream) {
+        fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
+        ret = 1;
+        goto end;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate frame\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* initialize packet, set data to NULL, let the demuxer fill it */
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
+    if (video_stream)
+        printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
+    if (audio_stream)
+        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
+
+    /* read frames from the file */
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_packet_unref(&orig_pkt);
+    }
+
+    /* flush cached frames */
+    pkt.data = NULL;
+    pkt.size = 0;
+    do {
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
+
+    printf("Demuxing succeeded.\n");
+
+    if (video_stream) {
+        printf("Play the output video file with the command:\n"
+               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
+               av_get_pix_fmt_name(pix_fmt), width, height,
+               video_dst_filename);
+    }
+
+    if (audio_stream) {
+        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
+        int n_channels = audio_dec_ctx->channels;
+        const char *fmt;
+
+        if (av_sample_fmt_is_planar(sfmt)) {
+            const char *packed = av_get_sample_fmt_name(sfmt);
+            printf("Warning: the sample format the decoder produced is planar "
+                   "(%s). This example will output the first channel only.\n",
+                   packed ? packed : "?");
+            sfmt = av_get_packed_sample_fmt(sfmt);
+            n_channels = 1;
+        }
+
+        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
+            goto end;
+
+        printf("Play the output audio file with the command:\n"
+               "ffplay -f %s -ac %d -ar %d %s\n",
+               fmt, n_channels, audio_dec_ctx->sample_rate,
+               audio_dst_filename);
+    }
+
+end:
+    avcodec_close(video_dec_ctx);
+    avcodec_close(audio_dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    if (video_dst_file)
+        fclose(video_dst_file);
+    if (audio_dst_file)
+        fclose(audio_dst_file);
+    av_frame_free(&frame);
+    av_free(video_dst_data[0]);
+
+    return ret < 0;
+}
@@ -0,0 +1,185 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ * Copyright (c) 2014 Clément Bœsch
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <libavutil/motion_vector.h>
+#include <libavformat/avformat.h>
+
+static AVFormatContext *fmt_ctx = NULL;
+static AVCodecContext *video_dec_ctx = NULL;
+static AVStream *video_stream = NULL;
+static const char *src_filename = NULL;
+
+static int video_stream_idx = -1;
+static AVFrame *frame = NULL;
+static AVPacket pkt;
+static int video_frame_count = 0;
+
+static int decode_packet(int *got_frame, int cached)
+{
+    int decoded = pkt.size;
+
+    *got_frame = 0;
+
+    if (pkt.stream_index == video_stream_idx) {
+        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
+            return ret;
+        }
+
+        if (*got_frame) {
+            int i;
+            AVFrameSideData *sd;
+
+            video_frame_count++;
+            sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
+            if (sd) {
+                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
+                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
+                    const AVMotionVector *mv = &mvs[i];
+                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
+                           video_frame_count, mv->source,
+                           mv->w, mv->h, mv->src_x, mv->src_y,
+                           mv->dst_x, mv->dst_y, mv->flags);
+                }
+            }
+        }
+    }
+
+    return decoded;
+}
+
+static int open_codec_context(int *stream_idx,
+                              AVFormatContext *fmt_ctx, enum AVMediaType type)
+{
+    int ret;
+    AVStream *st;
+    AVCodecContext *dec_ctx = NULL;
+    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;
+
+    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
+                av_get_media_type_string(type), src_filename);
+        return ret;
+    } else {
+        *stream_idx = ret;
+        st = fmt_ctx->streams[*stream_idx];
+
+        /* find decoder for the stream */
+        dec_ctx = st->codec;
+        dec = avcodec_find_decoder(dec_ctx->codec_id);
+        if (!dec) {
+            fprintf(stderr, "Failed to find %s codec\n",
+                    av_get_media_type_string(type));
+            return AVERROR(EINVAL);
+        }
+
+        /* Init the video decoder */
+        av_dict_set(&opts, "flags2", "+export_mvs", 0);
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
+            fprintf(stderr, "Failed to open %s codec\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+    }
+
+    return 0;
+}
+
+int main(int argc, char **argv)
+{
+    int ret = 0, got_frame;
+
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
+        exit(1);
+    }
+    src_filename = argv[1];
+
+    av_register_all();
+
+    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
+        fprintf(stderr, "Could not open source file %s\n", src_filename);
+        exit(1);
+    }
+
+    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        exit(1);
+    }
+
+    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+        video_stream = fmt_ctx->streams[video_stream_idx];
+        video_dec_ctx = video_stream->codec;
+    }
+
+    av_dump_format(fmt_ctx, 0, src_filename, 0);
+
+    if (!video_stream) {
+        fprintf(stderr, "Could not find video stream in the input, aborting\n");
+        ret = 1;
+        goto end;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate frame\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
+
+    /* initialize packet, set data to NULL, let the demuxer fill it */
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
+    /* read frames from the file */
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_packet_unref(&orig_pkt);
+    }
+
+    /* flush cached frames */
+    pkt.data = NULL;
+    pkt.size = 0;
+    do {
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
+
+end:
+    avcodec_close(video_dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    return ret < 0;
+}
@@ -0,0 +1,365 @@
+/*
+ * copyright (c) 2013 Andrew Kelley
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * libavfilter API usage example.
+ *
+ * @example filter_audio.c
+ * This example will generate a sine wave audio,
+ * pass it through a simple filter chain, and then compute the MD5 checksum of
+ * the output data.
+ *
+ * The filter chain it uses is:
+ * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
+ *
+ * abuffer: This provides the endpoint where you can feed the decoded samples.
+ * volume: In this example we hardcode it to 0.90.
+ * aformat: This converts the samples to the samplefreq, channel layout,
+ *          and sample format required by the audio device.
+ * abuffersink: This provides the endpoint where you can read the samples after
+ *              they have passed through the filter chain.
+ */
+
+#include <inttypes.h>
+#include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/md5.h"
+#include "libavutil/mem.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+
+#include "libavfilter/avfilter.h"
+#include "libavfilter/buffersink.h"
+#include "libavfilter/buffersrc.h"
+
+#define INPUT_SAMPLERATE     48000
+#define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
+#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
+
+#define VOLUME_VAL 0.90
+
+static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
+                             AVFilterContext **sink)
+{
+    AVFilterGraph *filter_graph;
+    AVFilterContext *abuffer_ctx;
+    AVFilter        *abuffer;
+    AVFilterContext *volume_ctx;
+    AVFilter        *volume;
+    AVFilterContext *aformat_ctx;
+    AVFilter        *aformat;
+    AVFilterContext *abuffersink_ctx;
+    AVFilter        *abuffersink;
+
+    AVDictionary *options_dict = NULL;
+    uint8_t options_str[1024];
+    uint8_t ch_layout[64];
+
+    int err;
+
+    /* Create a new filtergraph, which will contain all the filters. */
+    filter_graph = avfilter_graph_alloc();
+    if (!filter_graph) {
+        fprintf(stderr, "Unable to create filter graph.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Create the abuffer filter;
+     * it will be used for feeding the data into the graph. */
+    abuffer = avfilter_get_by_name("abuffer");
+    if (!abuffer) {
+        fprintf(stderr, "Could not find the abuffer filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
+    if (!abuffer_ctx) {
+        fprintf(stderr, "Could not allocate the abuffer instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Set the filter options through the AVOptions API. */
+    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
+    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
+    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
+    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
+    av_opt_set_int(abuffer_ctx, "sample_rate",    INPUT_SAMPLERATE,                     AV_OPT_SEARCH_CHILDREN);
+
+    /* Now initialize the filter; we pass NULL options, since we have already
+     * set all the options above. */
+    err = avfilter_init_str(abuffer_ctx, NULL);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the abuffer filter.\n");
+        return err;
+    }
+
+    /* Create volume filter. */
+    volume = avfilter_get_by_name("volume");
+    if (!volume) {
+        fprintf(stderr, "Could not find the volume filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
+    if (!volume_ctx) {
+        fprintf(stderr, "Could not allocate the volume instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* A different way of passing the options is as key/value pairs in a
+     * dictionary. */
+    av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
+    err = avfilter_init_dict(volume_ctx, &options_dict);
+    av_dict_free(&options_dict);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the volume filter.\n");
+        return err;
+    }
+
+    /* Create the aformat filter;
+     * it ensures that the output is of the format we want. */
+    aformat = avfilter_get_by_name("aformat");
+    if (!aformat) {
+        fprintf(stderr, "Could not find the aformat filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
+    if (!aformat_ctx) {
+        fprintf(stderr, "Could not allocate the aformat instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* A third way of passing the options is in a string of the form
+     * key1=value1:key2=value2.... */
+    snprintf(options_str, sizeof(options_str),
+             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
+             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
+             (uint64_t)AV_CH_LAYOUT_STEREO);
+    err = avfilter_init_str(aformat_ctx, options_str);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
+        return err;
+    }
+
+    /* Finally create the abuffersink filter;
+     * it will be used to get the filtered data out of the graph. */
+    abuffersink = avfilter_get_by_name("abuffersink");
+    if (!abuffersink) {
+        fprintf(stderr, "Could not find the abuffersink filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
+    if (!abuffersink_ctx) {
+        fprintf(stderr, "Could not allocate the abuffersink instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* This filter takes no options. */
+    err = avfilter_init_str(abuffersink_ctx, NULL);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the abuffersink instance.\n");
+        return err;
+    }
+
+    /* Connect the filters;
+     * in this simple case the filters just form a linear chain. */
+    err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
+    if (err >= 0)
+        err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
+    if (err >= 0)
+        err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
+    if (err < 0) {
+        fprintf(stderr, "Error connecting filters\n");
+        return err;
+    }
+
+    /* Configure the graph. */
+    err = avfilter_graph_config(filter_graph, NULL);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
+        return err;
+    }
+
+    *graph = filter_graph;
+    *src   = abuffer_ctx;
+    *sink  = abuffersink_ctx;
+
+    return 0;
+}
+
+/* Do something useful with the filtered data: this simple
+ * example just prints the MD5 checksum of each plane to stdout. */
+static int process_output(struct AVMD5 *md5, AVFrame *frame)
+{
+    int planar     = av_sample_fmt_is_planar(frame->format);
+    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
+    int planes     = planar ? channels : 1;
+    int bps        = av_get_bytes_per_sample(frame->format);
+    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
+    int i, j;
+
+    for (i = 0; i < planes; i++) {
+        uint8_t checksum[16];
+
+        av_md5_init(md5);
+        av_md5_sum(checksum, frame->extended_data[i], plane_size);
+
+        fprintf(stdout, "plane %d: 0x", i);
+        for (j = 0; j < sizeof(checksum); j++)
+            fprintf(stdout, "%02X", checksum[j]);
+        fprintf(stdout, "\n");
+    }
+    fprintf(stdout, "\n");
+
+    return 0;
+}
+
+/* Construct a frame of audio data to be filtered;
+ * this simple example just synthesizes a sine wave. */
+static int get_input(AVFrame *frame, int frame_num)
+{
+    int err, i, j;
+
+#define FRAME_SIZE 1024
+
+    /* Set up the frame properties and allocate the buffer for the data. */
+    frame->sample_rate    = INPUT_SAMPLERATE;
+    frame->format         = INPUT_FORMAT;
+    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
+    frame->nb_samples     = FRAME_SIZE;
+    frame->pts            = frame_num * FRAME_SIZE;
+
+    err = av_frame_get_buffer(frame, 0);
+    if (err < 0)
+        return err;
+
+    /* Fill the data for each channel. */
+    for (i = 0; i < 5; i++) {
+        float *data = (float*)frame->extended_data[i];
+
+        for (j = 0; j < frame->nb_samples; j++)
+            data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
+    }
+
+    return 0;
+}
+
+int main(int argc, char *argv[])
+{
+    struct AVMD5 *md5;
+    AVFilterGraph *graph;
+    AVFilterContext *src, *sink;
+    AVFrame *frame;
+    uint8_t errstr[1024];
+    float duration;
+    int err, nb_frames, i;
+
+    if (argc < 2) {
+        fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
+        return 1;
+    }
+
+    duration  = atof(argv[1]);
+    nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
+    if (nb_frames <= 0) {
+        fprintf(stderr, "Invalid duration: %s\n", argv[1]);
+        return 1;
+    }
+
+    avfilter_register_all();
+
+    /* Allocate the frame we will be using to store the data. */
+    frame  = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Error allocating the frame\n");
+        return 1;
+    }
+
+    md5 = av_md5_alloc();
+    if (!md5) {
+        fprintf(stderr, "Error allocating the MD5 context\n");
+        return 1;
+    }
+
+    /* Set up the filtergraph. */
+    err = init_filter_graph(&graph, &src, &sink);
+    if (err < 0) {
+        fprintf(stderr, "Unable to init filter graph:");
+        goto fail;
+    }
+
+    /* the main filtering loop */
+    for (i = 0; i < nb_frames; i++) {
+        /* get an input frame to be filtered */
+        err = get_input(frame, i);
+        if (err < 0) {
+            fprintf(stderr, "Error generating input frame:");
+            goto fail;
+        }
+
+        /* Send the frame to the input of the filtergraph. */
+        err = av_buffersrc_add_frame(src, frame);
+        if (err < 0) {
+            av_frame_unref(frame);
+            fprintf(stderr, "Error submitting the frame to the filtergraph:");
+            goto fail;
+        }
+
+        /* Get all the filtered output that is available. */
+        while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
+            /* now do something with our filtered frame */
+            err = process_output(md5, frame);
+            if (err < 0) {
+                fprintf(stderr, "Error processing the filtered frame:");
+                goto fail;
+            }
+            av_frame_unref(frame);
+        }
+
+        if (err == AVERROR(EAGAIN)) {
+            /* Need to feed more frames in. */
+            continue;
+        } else if (err == AVERROR_EOF) {
+            /* Nothing more to do, finish. */
+            break;
+        } else if (err < 0) {
+            /* An error occurred. */
+            fprintf(stderr, "Error filtering the data:");
+            goto fail;
+        }
+    }
+
+    avfilter_graph_free(&graph);
+    av_frame_free(&frame);
+    av_freep(&md5);
+
+    return 0;
+
+fail:
+    av_strerror(err, errstr, sizeof(errstr));
+    fprintf(stderr, "%s\n", errstr);
+    return 1;
+}
@@ -0,0 +1,295 @@
+/*
+ * Copyright (c) 2010 Nicolas George
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2012 Clément Bœsch
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * API example for audio decoding and filtering
+ * @example filtering_audio.c
+ */
+
+#include <unistd.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/buffersink.h>
+#include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
+
+static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
+static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
+
+static AVFormatContext *fmt_ctx;
+static AVCodecContext *dec_ctx;
+AVFilterContext *buffersink_ctx;
+AVFilterContext *buffersrc_ctx;
+AVFilterGraph *filter_graph;
+static int audio_stream_index = -1;
+
+static int open_input_file(const char *filename)
+{
+    int ret;
+    AVCodec *dec;
+
+    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
+        return ret;
+    }
+
+    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
+        return ret;
+    }
+
+    /* select the audio stream */
+    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
+        return ret;
+    }
+    audio_stream_index = ret;
+    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
+
+    /* init the audio decoder */
+    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+static int init_filters(const char *filters_descr)
+{
+    char args[512];
+    int ret = 0;
+    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
+    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
+    AVFilterInOut *outputs = avfilter_inout_alloc();
+    AVFilterInOut *inputs  = avfilter_inout_alloc();
+    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    static const int out_sample_rates[] = { 8000, -1 };
+    const AVFilterLink *outlink;
+    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
+
+    filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
+    if (!dec_ctx->channel_layout)
+        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
+    snprintf(args, sizeof(args),
+            "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
+             time_base.num, time_base.den, dec_ctx->sample_rate,
+             av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
+    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
+                                       args, NULL, filter_graph);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
+        goto end;
+    }
+
+    /* buffer audio sink: to terminate the filter chain. */
+    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
+                                       NULL, NULL, filter_graph);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
+        goto end;
+    }
+
+    /*
+     * Set the endpoints for the filter graph. The filter_graph will
+     * be linked to the graph described by filters_descr.
+     */
+
+    /*
+     * The buffer source output must be connected to the input pad of
+     * the first filter described by filters_descr; since the first
+     * filter input label is not specified, it is set to "in" by
+     * default.
+     */
+    outputs->name       = av_strdup("in");
+    outputs->filter_ctx = buffersrc_ctx;
+    outputs->pad_idx    = 0;
+    outputs->next       = NULL;
+
+    /*
+     * The buffer sink input must be connected to the output pad of
+     * the last filter described by filters_descr; since the last
+     * filter output label is not specified, it is set to "out" by
+     * default.
+     */
+    inputs->name       = av_strdup("out");
+    inputs->filter_ctx = buffersink_ctx;
+    inputs->pad_idx    = 0;
+    inputs->next       = NULL;
+
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
+                                        &inputs, &outputs, NULL)) < 0)
+        goto end;
+
+    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
+        goto end;
+
+    /* Print summary of the sink buffer
+     * Note: args buffer is reused to store channel layout string */
+    outlink = buffersink_ctx->inputs[0];
+    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
+    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
+           (int)outlink->sample_rate,
+           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
+           args);
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
+}
+
+static void print_frame(const AVFrame *frame)
+{
+    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
+    const uint16_t *p     = (uint16_t*)frame->data[0];
+    const uint16_t *p_end = p + n;
+
+    while (p < p_end) {
+        fputc(*p    & 0xff, stdout);
+        fputc(*p>>8 & 0xff, stdout);
+        p++;
+    }
+    fflush(stdout);
+}
+
+int main(int argc, char **argv)
+{
+    int ret;
+    AVPacket packet0, packet;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
+    int got_frame;
+
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
+        exit(1);
+    }
+
+    av_register_all();
+    avfilter_register_all();
+
+    if ((ret = open_input_file(argv[1])) < 0)
+        goto end;
+    if ((ret = init_filters(filter_descr)) < 0)
+        goto end;
+
+    /* read all packets */
+    packet0.data = NULL;
+    packet.data = NULL;
+    while (1) {
+        if (!packet0.data) {
+            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+                break;
+            packet0 = packet;
+        }
+
+        if (packet.stream_index == audio_stream_index) {
+            got_frame = 0;
+            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
+                continue;
+            }
+            packet.size -= ret;
+            packet.data += ret;
+
+            if (got_frame) {
+                /* push the audio data from decoded frame into the filtergraph */
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
+                    break;
+                }
+
+                /* pull filtered audio from the filtergraph */
+                while (1) {
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                        break;
+                    if (ret < 0)
+                        goto end;
+                    print_frame(filt_frame);
+                    av_frame_unref(filt_frame);
+                }
+            }
+
+            if (packet.size <= 0)
+                av_packet_unref(&packet0);
+        } else {
+            /* discard non-wanted packets */
+            av_packet_unref(&packet0);
+        }
+    }
+end:
+    avfilter_graph_free(&filter_graph);
+    avcodec_close(dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    exit(0);
+}
@@ -0,0 +1,280 @@
+/*
+ * Copyright (c) 2010 Nicolas George
+ * Copyright (c) 2011 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * API example for decoding and filtering
+ * @example filtering_video.c
+ */
+
+#define _XOPEN_SOURCE 600 /* for usleep */
+#include <unistd.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/buffersink.h>
+#include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
+
+const char *filter_descr = "scale=78:24,transpose=cclock";
+/* other way:
+   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
+ */
+
+static AVFormatContext *fmt_ctx;
+static AVCodecContext *dec_ctx;
+AVFilterContext *buffersink_ctx;
+AVFilterContext *buffersrc_ctx;
+AVFilterGraph *filter_graph;
+static int video_stream_index = -1;
+static int64_t last_pts = AV_NOPTS_VALUE;
+
+static int open_input_file(const char *filename)
+{
+    int ret;
+    AVCodec *dec;
+
+    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
+        return ret;
+    }
+
+    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
+        return ret;
+    }
+
+    /* select the video stream */
+    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
+        return ret;
+    }
+    video_stream_index = ret;
+    dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
+
+    /* init the video decoder */
+    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+static int init_filters(const char *filters_descr)
+{
+    char args[512];
+    int ret = 0;
+    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
+    AVFilterInOut *outputs = avfilter_inout_alloc();
+    AVFilterInOut *inputs  = avfilter_inout_alloc();
+    AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
+    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
+
+    filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* buffer video source: the decoded frames from the decoder will be inserted here. */
+    snprintf(args, sizeof(args),
+            "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
+            dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
+            time_base.num, time_base.den,
+            dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
+
+    ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
+                                       args, NULL, filter_graph);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
+        goto end;
+    }
+
+    /* buffer video sink: to terminate the filter chain. */
+    ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
+                                       NULL, NULL, filter_graph);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
+                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
+        goto end;
+    }
+
+    /*
+     * Set the endpoints for the filter graph. The filter_graph will
+     * be linked to the graph described by filters_descr.
+     */
+
+    /*
+     * The buffer source output must be connected to the input pad of
+     * the first filter described by filters_descr; since the first
+     * filter input label is not specified, it is set to "in" by
+     * default.
+     */
+    outputs->name       = av_strdup("in");
+    outputs->filter_ctx = buffersrc_ctx;
+    outputs->pad_idx    = 0;
+    outputs->next       = NULL;
+
+    /*
+     * The buffer sink input must be connected to the output pad of
+     * the last filter described by filters_descr; since the last
+     * filter output label is not specified, it is set to "out" by
+     * default.
+     */
+    inputs->name       = av_strdup("out");
+    inputs->filter_ctx = buffersink_ctx;
+    inputs->pad_idx    = 0;
+    inputs->next       = NULL;
+
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
+                                    &inputs, &outputs, NULL)) < 0)
+        goto end;
+
+    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
+        goto end;
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
+}
+
+static void display_frame(const AVFrame *frame, AVRational time_base)
+{
+    int x, y;
+    uint8_t *p0, *p;
+    int64_t delay;
+
+    if (frame->pts != AV_NOPTS_VALUE) {
+        if (last_pts != AV_NOPTS_VALUE) {
+            /* sleep roughly the right amount of time;
+             * usleep is in microseconds, just like AV_TIME_BASE. */
+            delay = av_rescale_q(frame->pts - last_pts,
+                                 time_base, AV_TIME_BASE_Q);
+            if (delay > 0 && delay < 1000000)
+                usleep(delay);
+        }
+        last_pts = frame->pts;
+    }
+
+    /* Trivial ASCII grayscale display. */
+    p0 = frame->data[0];
+    puts("\033c");
+    for (y = 0; y < frame->height; y++) {
+        p = p0;
+        for (x = 0; x < frame->width; x++)
+            putchar(" .-+#"[*(p++) / 52]);
+        putchar('\n');
+        p0 += frame->linesize[0];
+    }
+    fflush(stdout);
+}
+
+int main(int argc, char **argv)
+{
+    int ret;
+    AVPacket packet;
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
+    int got_frame;
+
+    if (!frame || !filt_frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s file\n", argv[0]);
+        exit(1);
+    }
+
+    av_register_all();
+    avfilter_register_all();
+
+    if ((ret = open_input_file(argv[1])) < 0)
+        goto end;
+    if ((ret = init_filters(filter_descr)) < 0)
+        goto end;
+
+    /* read all packets */
+    while (1) {
+        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+            break;
+
+        if (packet.stream_index == video_stream_index) {
+            got_frame = 0;
+            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
+                break;
+            }
+
+            if (got_frame) {
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
+
+                /* push the decoded frame into the filtergraph */
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
+                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
+                    break;
+                }
+
+                /* pull filtered frames from the filtergraph */
+                while (1) {
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                        break;
+                    if (ret < 0)
+                        goto end;
+                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
+                    av_frame_unref(filt_frame);
+                }
+                av_frame_unref(frame);
+            }
+        }
+        av_packet_unref(&packet);
+    }
+end:
+    avfilter_graph_free(&filter_graph);
+    avcodec_close(dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    exit(0);
+}
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2015 Stephan Holljes
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat multi-client network API usage example.
+ *
+ * @example http_multiclient.c
+ * This example will serve a file without decoding or demuxing it over http.
+ * Multiple clients can connect and will receive the same file.
+ */
+
+#include <libavformat/avformat.h>
+#include <libavutil/opt.h>
+#include <unistd.h>
+
+void process_client(AVIOContext *client, const char *in_uri)
+{
+    AVIOContext *input = NULL;
+    uint8_t buf[1024];
+    int ret, n, reply_code;
+    char *resource = NULL;
+    while ((ret = avio_handshake(client)) > 0) {
+        av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
+        // check for strlen(resource) is necessary, because av_opt_get()
+        // may return empty string.
+        if (resource && strlen(resource))
+            break;
+    }
+    if (ret < 0)
+        goto end;
+    av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
+    if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
+        reply_code = 200;
+    } else {
+        reply_code = AVERROR_HTTP_NOT_FOUND;
+    }
+    if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
+        av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
+        goto end;
+    }
+    av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
+
+    while ((ret = avio_handshake(client)) > 0);
+
+    if (ret < 0)
+        goto end;
+
+    fprintf(stderr, "Handshake performed.\n");
+    if (reply_code != 200)
+        goto end;
+    fprintf(stderr, "Opening input file.\n");
+    if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
+        av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
+               av_err2str(ret));
+        goto end;
+    }
+    for(;;) {
+        n = avio_read(input, buf, sizeof(buf));
+        if (n < 0) {
+            if (n == AVERROR_EOF)
+                break;
+            av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
+                   av_err2str(n));
+            break;
+        }
+        avio_write(client, buf, n);
+        avio_flush(client);
+    }
+end:
+    fprintf(stderr, "Flushing client\n");
+    avio_flush(client);
+    fprintf(stderr, "Closing client\n");
+    avio_close(client);
+    fprintf(stderr, "Closing input\n");
+    avio_close(input);
+}
+
+int main(int argc, char **argv)
+{
+    av_log_set_level(AV_LOG_TRACE);
+    AVDictionary *options = NULL;
+    AVIOContext *client = NULL, *server = NULL;
+    const char *in_uri, *out_uri;
+    int ret, pid;
+    if (argc < 3) {
+        printf("usage: %s input http://hostname[:port]\n"
+               "API example program to serve http to multiple clients.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_uri = argv[1];
+    out_uri = argv[2];
+
+    av_register_all();
+    avformat_network_init();
+
+    if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
+        fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
+        fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    fprintf(stderr, "Entering main loop.\n");
+    for(;;) {
+        if ((ret = avio_accept(server, &client)) < 0)
+            goto end;
+        fprintf(stderr, "Accepted client, forking process.\n");
+        // XXX: Since we don't reap our children and don't ignore signals
+        //      this produces zombie processes.
+        pid = fork();
+        if (pid < 0) {
+            perror("Fork failed");
+            ret = AVERROR(errno);
+            goto end;
+        }
+        if (pid == 0) {
+            fprintf(stderr, "In child.\n");
+            process_client(client, in_uri);
+            avio_close(server);
+            exit(0);
+        }
+        if (pid > 0)
+            avio_close(client);
+    }
+end:
+    avio_close(server);
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+    return 0;
+}
@@ -0,0 +1,56 @@
+/*
+ * Copyright (c) 2011 Reinhard Tartler
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * Shows how the metadata API can be used in application programs.
+ * @example metadata.c
+ */
+
+#include <stdio.h>
+
+#include <libavformat/avformat.h>
+#include <libavutil/dict.h>
+
+int main (int argc, char **argv)
+{
+    AVFormatContext *fmt_ctx = NULL;
+    AVDictionaryEntry *tag = NULL;
+    int ret;
+
+    if (argc != 2) {
+        printf("usage: %s <input_file>\n"
+               "example program to demonstrate the use of the libavformat metadata API.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    av_register_all();
+    if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
+        return ret;
+
+    while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
+        printf("%s=%s\n", tag->key, tag->value);
+
+    avformat_close_input(&fmt_ctx);
+    return 0;
+}
@@ -0,0 +1,651 @@
+/*
+ * Copyright (c) 2003 Fabrice Bellard
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat API example.
+ *
+ * Output a media file in any supported libavformat format. The default
+ * codecs are used.
+ * @example muxing.c
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+
+#include <libavutil/avassert.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/opt.h>
+#include <libavutil/mathematics.h>
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+#include <libswscale/swscale.h>
+#include <libswresample/swresample.h>
+
+#define STREAM_DURATION   10.0
+#define STREAM_FRAME_RATE 25 /* 25 images/s */
+#define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
+
+#define SCALE_FLAGS SWS_BICUBIC
+
+// a wrapper around a single output AVStream
+typedef struct OutputStream {
+    AVStream *st;
+
+    /* pts of the next frame that will be generated */
+    int64_t next_pts;
+    int samples_count;
+
+    AVFrame *frame;
+    AVFrame *tmp_frame;
+
+    float t, tincr, tincr2;
+
+    struct SwsContext *sws_ctx;
+    struct SwrContext *swr_ctx;
+} OutputStream;
+
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
+{
+    /* rescale output packet timestamp values from codec to stream timebase */
+    av_packet_rescale_ts(pkt, *time_base, st->time_base);
+    pkt->stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    log_packet(fmt_ctx, pkt);
+    return av_interleaved_write_frame(fmt_ctx, pkt);
+}
+
+/* Add an output stream. */
+static void add_stream(OutputStream *ost, AVFormatContext *oc,
+                       AVCodec **codec,
+                       enum AVCodecID codec_id)
+{
+    AVCodecContext *c;
+    int i;
+
+    /* find the encoder */
+    *codec = avcodec_find_encoder(codec_id);
+    if (!(*codec)) {
+        fprintf(stderr, "Could not find encoder for '%s'\n",
+                avcodec_get_name(codec_id));
+        exit(1);
+    }
+
+    ost->st = avformat_new_stream(oc, *codec);
+    if (!ost->st) {
+        fprintf(stderr, "Could not allocate stream\n");
+        exit(1);
+    }
+    ost->st->id = oc->nb_streams-1;
+    c = ost->st->codec;
+
+    switch ((*codec)->type) {
+    case AVMEDIA_TYPE_AUDIO:
+        c->sample_fmt  = (*codec)->sample_fmts ?
+            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
+        c->bit_rate    = 64000;
+        c->sample_rate = 44100;
+        if ((*codec)->supported_samplerates) {
+            c->sample_rate = (*codec)->supported_samplerates[0];
+            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
+                if ((*codec)->supported_samplerates[i] == 44100)
+                    c->sample_rate = 44100;
+            }
+        }
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
+        c->channel_layout = AV_CH_LAYOUT_STEREO;
+        if ((*codec)->channel_layouts) {
+            c->channel_layout = (*codec)->channel_layouts[0];
+            for (i = 0; (*codec)->channel_layouts[i]; i++) {
+                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
+                    c->channel_layout = AV_CH_LAYOUT_STEREO;
+            }
+        }
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
+        ost->st->time_base = (AVRational){ 1, c->sample_rate };
+        break;
+
+    case AVMEDIA_TYPE_VIDEO:
+        c->codec_id = codec_id;
+
+        c->bit_rate = 400000;
+        /* Resolution must be a multiple of two. */
+        c->width    = 352;
+        c->height   = 288;
+        /* timebase: This is the fundamental unit of time (in seconds) in terms
+         * of which frame timestamps are represented. For fixed-fps content,
+         * timebase should be 1/framerate and timestamp increments should be
+         * identical to 1. */
+        ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
+        c->time_base       = ost->st->time_base;
+
+        c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
+        c->pix_fmt       = STREAM_PIX_FMT;
+        if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
+            /* just for testing, we also add B frames */
+            c->max_b_frames = 2;
+        }
+        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
+            /* Needed to avoid using macroblocks in which some coeffs overflow.
+             * This does not happen with normal video, it just happens here as
+             * the motion of the chroma plane does not match the luma plane. */
+            c->mb_decision = 2;
+        }
+    break;
+
+    default:
+        break;
+    }
+
+    /* Some formats want stream headers to be separate. */
+    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
+        c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+}
+
+/**************************************************************/
+/* audio output */
+
+static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
+                                  uint64_t channel_layout,
+                                  int sample_rate, int nb_samples)
+{
+    AVFrame *frame = av_frame_alloc();
+    int ret;
+
+    if (!frame) {
+        fprintf(stderr, "Error allocating an audio frame\n");
+        exit(1);
+    }
+
+    frame->format = sample_fmt;
+    frame->channel_layout = channel_layout;
+    frame->sample_rate = sample_rate;
+    frame->nb_samples = nb_samples;
+
+    if (nb_samples) {
+        ret = av_frame_get_buffer(frame, 0);
+        if (ret < 0) {
+            fprintf(stderr, "Error allocating an audio buffer\n");
+            exit(1);
+        }
+    }
+
+    return frame;
+}
+
+static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+{
+    AVCodecContext *c;
+    int nb_samples;
+    int ret;
+    AVDictionary *opt = NULL;
+
+    c = ost->st->codec;
+
+    /* open it */
+    av_dict_copy(&opt, opt_arg, 0);
+    ret = avcodec_open2(c, codec, &opt);
+    av_dict_free(&opt);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    /* init signal generator */
+    ost->t     = 0;
+    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
+    /* increment frequency by 110 Hz per second */
+    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
+
+    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
+        nb_samples = 10000;
+    else
+        nb_samples = c->frame_size;
+
+    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
+                                       c->sample_rate, nb_samples);
+    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
+                                       c->sample_rate, nb_samples);
+
+    /* create resampler context */
+        ost->swr_ctx = swr_alloc();
+        if (!ost->swr_ctx) {
+            fprintf(stderr, "Could not allocate resampler context\n");
+            exit(1);
+        }
+
+        /* set options */
+        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+
+        /* initialize the resampling context */
+        if ((ret = swr_init(ost->swr_ctx)) < 0) {
+            fprintf(stderr, "Failed to initialize the resampling context\n");
+            exit(1);
+        }
+}
+
+/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
+ * 'nb_channels' channels. */
+static AVFrame *get_audio_frame(OutputStream *ost)
+{
+    AVFrame *frame = ost->tmp_frame;
+    int j, i, v;
+    int16_t *q = (int16_t*)frame->data[0];
+
+    /* check if we want to generate more frames */
+    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
+        return NULL;
+
+    for (j = 0; j <frame->nb_samples; j++) {
+        v = (int)(sin(ost->t) * 10000);
+        for (i = 0; i < ost->st->codec->channels; i++)
+            *q++ = v;
+        ost->t     += ost->tincr;
+        ost->tincr += ost->tincr2;
+    }
+
+    frame->pts = ost->next_pts;
+    ost->next_pts  += frame->nb_samples;
+
+    return frame;
+}
+
+/*
+ * encode one audio frame and send it to the muxer
+ * return 1 when encoding is finished, 0 otherwise
+ */
+static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
+{
+    AVCodecContext *c;
+    AVPacket pkt = { 0 }; // data and size must be 0;
+    AVFrame *frame;
+    int ret;
+    int got_packet;
+    int dst_nb_samples;
+
+    av_init_packet(&pkt);
+    c = ost->st->codec;
+
+    frame = get_audio_frame(ost);
+
+    if (frame) {
+        /* convert samples from native format to destination codec format, using the resampler */
+            /* compute destination number of samples */
+            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
+                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
+            av_assert0(dst_nb_samples == frame->nb_samples);
+
+        /* when we pass a frame to the encoder, it may keep a reference to it
+         * internally;
+         * make sure we do not overwrite it here
+         */
+        ret = av_frame_make_writable(ost->frame);
+        if (ret < 0)
+            exit(1);
+
+            /* convert to destination format */
+            ret = swr_convert(ost->swr_ctx,
+                              ost->frame->data, dst_nb_samples,
+                              (const uint8_t **)frame->data, frame->nb_samples);
+            if (ret < 0) {
+                fprintf(stderr, "Error while converting\n");
+                exit(1);
+            }
+            frame = ost->frame;
+
+        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        ost->samples_count += dst_nb_samples;
+    }
+
+    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    if (got_packet) {
+        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error while writing audio frame: %s\n",
+                    av_err2str(ret));
+            exit(1);
+        }
+    }
+
+    return (frame || got_packet) ? 0 : 1;
+}
+
+/**************************************************************/
+/* video output */
+
+static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
+{
+    AVFrame *picture;
+    int ret;
+
+    picture = av_frame_alloc();
+    if (!picture)
+        return NULL;
+
+    picture->format = pix_fmt;
+    picture->width  = width;
+    picture->height = height;
+
+    /* allocate the buffers for the frame data */
+    ret = av_frame_get_buffer(picture, 32);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate frame data.\n");
+        exit(1);
+    }
+
+    return picture;
+}
+
+static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+{
+    int ret;
+    AVCodecContext *c = ost->st->codec;
+    AVDictionary *opt = NULL;
+
+    av_dict_copy(&opt, opt_arg, 0);
+
+    /* open the codec */
+    ret = avcodec_open2(c, codec, &opt);
+    av_dict_free(&opt);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    /* allocate and init a re-usable frame */
+    ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
+    if (!ost->frame) {
+        fprintf(stderr, "Could not allocate video frame\n");
+        exit(1);
+    }
+
+    /* If the output format is not YUV420P, then a temporary YUV420P
+     * picture is needed too. It is then converted to the required
+     * output format. */
+    ost->tmp_frame = NULL;
+    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
+        ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
+        if (!ost->tmp_frame) {
+            fprintf(stderr, "Could not allocate temporary picture\n");
+            exit(1);
+        }
+    }
+}
+
+/* Prepare a dummy image. */
+static void fill_yuv_image(AVFrame *pict, int frame_index,
+                           int width, int height)
+{
+    int x, y, i, ret;
+
+    /* when we pass a frame to the encoder, it may keep a reference to it
+     * internally;
+     * make sure we do not overwrite it here
+     */
+    ret = av_frame_make_writable(pict);
+    if (ret < 0)
+        exit(1);
+
+    i = frame_index;
+
+    /* Y */
+    for (y = 0; y < height; y++)
+        for (x = 0; x < width; x++)
+            pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
+
+    /* Cb and Cr */
+    for (y = 0; y < height / 2; y++) {
+        for (x = 0; x < width / 2; x++) {
+            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
+            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
+        }
+    }
+}
+
+static AVFrame *get_video_frame(OutputStream *ost)
+{
+    AVCodecContext *c = ost->st->codec;
+
+    /* check if we want to generate more frames */
+    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
+                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
+        return NULL;
+
+    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
+        /* as we only generate a YUV420P picture, we must convert it
+         * to the codec pixel format if needed */
+        if (!ost->sws_ctx) {
+            ost->sws_ctx = sws_getContext(c->width, c->height,
+                                          AV_PIX_FMT_YUV420P,
+                                          c->width, c->height,
+                                          c->pix_fmt,
+                                          SCALE_FLAGS, NULL, NULL, NULL);
+            if (!ost->sws_ctx) {
+                fprintf(stderr,
+                        "Could not initialize the conversion context\n");
+                exit(1);
+            }
+        }
+        fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
+        sws_scale(ost->sws_ctx,
+                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
+                  0, c->height, ost->frame->data, ost->frame->linesize);
+    } else {
+        fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
+    }
+
+    ost->frame->pts = ost->next_pts++;
+
+    return ost->frame;
+}
+
+/*
+ * encode one video frame and send it to the muxer
+ * return 1 when encoding is finished, 0 otherwise
+ */
+static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
+{
+    int ret;
+    AVCodecContext *c;
+    AVFrame *frame;
+    int got_packet = 0;
+    AVPacket pkt = { 0 };
+
+    c = ost->st->codec;
+
+    frame = get_video_frame(ost);
+
+    av_init_packet(&pkt);
+
+    /* encode the image */
+    ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+    if (ret < 0) {
+        fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    if (got_packet) {
+        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+    } else {
+        ret = 0;
+    }
+
+    if (ret < 0) {
+        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
+    return (frame || got_packet) ? 0 : 1;
+}
+
+static void close_stream(AVFormatContext *oc, OutputStream *ost)
+{
+    avcodec_close(ost->st->codec);
+    av_frame_free(&ost->frame);
+    av_frame_free(&ost->tmp_frame);
+    sws_freeContext(ost->sws_ctx);
+    swr_free(&ost->swr_ctx);
+}
+
+/**************************************************************/
+/* media file output */
+
+int main(int argc, char **argv)
+{
+    OutputStream video_st = { 0 }, audio_st = { 0 };
+    const char *filename;
+    AVOutputFormat *fmt;
+    AVFormatContext *oc;
+    AVCodec *audio_codec, *video_codec;
+    int ret;
+    int have_video = 0, have_audio = 0;
+    int encode_video = 0, encode_audio = 0;
+    AVDictionary *opt = NULL;
+
+    /* Initialize libavcodec, and register all codecs and formats. */
+    av_register_all();
+
+    if (argc < 2) {
+        printf("usage: %s output_file\n"
+               "API example program to output a media file with libavformat.\n"
+               "This program generates a synthetic audio and video stream, encodes and\n"
+               "muxes them into a file named output_file.\n"
+               "The output format is automatically guessed according to the file extension.\n"
+               "Raw images can also be output by using '%%d' in the filename.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    filename = argv[1];
+    if (argc > 3 && !strcmp(argv[2], "-flags")) {
+        av_dict_set(&opt, argv[2]+1, argv[3], 0);
+    }
+
+    /* allocate the output media context */
+    avformat_alloc_output_context2(&oc, NULL, NULL, filename);
+    if (!oc) {
+        printf("Could not deduce output format from file extension: using MPEG.\n");
+        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
+    }
+    if (!oc)
+        return 1;
+
+    fmt = oc->oformat;
+
+    /* Add the audio and video streams using the default format codecs
+     * and initialize the codecs. */
+    if (fmt->video_codec != AV_CODEC_ID_NONE) {
+        add_stream(&video_st, oc, &video_codec, fmt->video_codec);
+        have_video = 1;
+        encode_video = 1;
+    }
+    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
+        add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
+        have_audio = 1;
+        encode_audio = 1;
+    }
+
+    /* Now that all the parameters are set, we can open the audio and
+     * video codecs and allocate the necessary encode buffers. */
+    if (have_video)
+        open_video(oc, video_codec, &video_st, opt);
+
+    if (have_audio)
+        open_audio(oc, audio_codec, &audio_st, opt);
+
+    av_dump_format(oc, 0, filename, 1);
+
+    /* open the output file, if needed */
+    if (!(fmt->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            fprintf(stderr, "Could not open '%s': %s\n", filename,
+                    av_err2str(ret));
+            return 1;
+        }
+    }
+
+    /* Write the stream header, if any. */
+    ret = avformat_write_header(oc, &opt);
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred when opening output file: %s\n",
+                av_err2str(ret));
+        return 1;
+    }
+
+    while (encode_video || encode_audio) {
+        /* select the stream to encode */
+        if (encode_video &&
+            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
+                                            audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
+            encode_video = !write_video_frame(oc, &video_st);
+        } else {
+            encode_audio = !write_audio_frame(oc, &audio_st);
+        }
+    }
+
+    /* Write the trailer, if any. The trailer must be written before you
+     * close the CodecContexts open when you wrote the header; otherwise
+     * av_write_trailer() may try to use memory that was freed on
+     * av_codec_close(). */
+    av_write_trailer(oc);
+
+    /* Close each codec. */
+    if (have_video)
+        close_stream(oc, &video_st);
+    if (have_audio)
+        close_stream(oc, &audio_st);
+
+    if (!(fmt->flags & AVFMT_NOFILE))
+        /* Close the output file. */
+        avio_closep(&oc->pb);
+
+    /* free the stream */
+    avformat_free_context(oc);
+
+    return 0;
+}
@@ -0,0 +1,487 @@
+/*
+ * Copyright (c) 2015 Anton Khirnov
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * Intel QSV-accelerated H.264 decoding example.
+ *
+ * @example qsvdec.c
+ * This example shows how to do QSV-accelerated H.264 decoding with output
+ * frames in the VA-API video surfaces.
+ */
+
+#include "config.h"
+
+#include <stdio.h>
+
+#include <mfx/mfxvideo.h>
+
+#include <va/va.h>
+#include <va/va_x11.h>
+#include <X11/Xlib.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+#include "libavcodec/qsv.h"
+
+#include "libavutil/error.h"
+#include "libavutil/mem.h"
+
+typedef struct DecodeContext {
+    mfxSession mfx_session;
+    VADisplay va_dpy;
+
+    VASurfaceID *surfaces;
+    mfxMemId    *surface_ids;
+    int         *surface_used;
+    int       nb_surfaces;
+
+    mfxFrameInfo frame_info;
+} DecodeContext;
+
+static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
+                             mfxFrameAllocResponse *resp)
+{
+    DecodeContext *decode = pthis;
+    int err, i;
+
+    if (decode->surfaces) {
+        fprintf(stderr, "Multiple allocation requests.\n");
+        return MFX_ERR_MEMORY_ALLOC;
+    }
+    if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
+        fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
+        return MFX_ERR_UNSUPPORTED;
+    }
+    if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
+        req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
+        req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
+        fprintf(stderr, "Unsupported surface properties.\n");
+        return MFX_ERR_UNSUPPORTED;
+    }
+
+    decode->surfaces     = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
+    decode->surface_ids  = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
+    decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
+    if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
+        goto fail;
+
+    err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
+                           req->Info.Width, req->Info.Height,
+                           decode->surfaces, req->NumFrameSuggested,
+                           NULL, 0);
+    if (err != VA_STATUS_SUCCESS) {
+        fprintf(stderr, "Error allocating VA surfaces\n");
+        goto fail;
+    }
+    decode->nb_surfaces = req->NumFrameSuggested;
+
+    for (i = 0; i < decode->nb_surfaces; i++)
+        decode->surface_ids[i] = &decode->surfaces[i];
+
+    resp->mids           = decode->surface_ids;
+    resp->NumFrameActual = decode->nb_surfaces;
+
+    decode->frame_info = req->Info;
+
+    return MFX_ERR_NONE;
+fail:
+    av_freep(&decode->surfaces);
+    av_freep(&decode->surface_ids);
+    av_freep(&decode->surface_used);
+
+    return MFX_ERR_MEMORY_ALLOC;
+}
+
+static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
+{
+    return MFX_ERR_NONE;
+}
+
+static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
+{
+    return MFX_ERR_UNSUPPORTED;
+}
+
+static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
+{
+    return MFX_ERR_UNSUPPORTED;
+}
+
+static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
+{
+    *hdl = mid;
+    return MFX_ERR_NONE;
+}
+
+static void free_surfaces(DecodeContext *decode)
+{
+    if (decode->surfaces)
+        vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
+    av_freep(&decode->surfaces);
+    av_freep(&decode->surface_ids);
+    av_freep(&decode->surface_used);
+    decode->nb_surfaces = 0;
+}
+
+static void free_buffer(void *opaque, uint8_t *data)
+{
+    int *used = opaque;
+    *used = 0;
+    av_freep(&data);
+}
+
+static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
+{
+    DecodeContext *decode = avctx->opaque;
+
+    mfxFrameSurface1 *surf;
+    AVBufferRef *surf_buf;
+    int idx;
+
+    for (idx = 0; idx < decode->nb_surfaces; idx++) {
+        if (!decode->surface_used[idx])
+            break;
+    }
+    if (idx == decode->nb_surfaces) {
+        fprintf(stderr, "No free surfaces\n");
+        return AVERROR(ENOMEM);
+    }
+
+    surf = av_mallocz(sizeof(*surf));
+    if (!surf)
+        return AVERROR(ENOMEM);
+    surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
+                                &decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
+    if (!surf_buf) {
+        av_freep(&surf);
+        return AVERROR(ENOMEM);
+    }
+
+    surf->Info       = decode->frame_info;
+    surf->Data.MemId = &decode->surfaces[idx];
+
+    frame->buf[0]  = surf_buf;
+    frame->data[3] = (uint8_t*)surf;
+
+    decode->surface_used[idx] = 1;
+
+    return 0;
+}
+
+static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
+{
+    while (*pix_fmts != AV_PIX_FMT_NONE) {
+        if (*pix_fmts == AV_PIX_FMT_QSV) {
+            if (!avctx->hwaccel_context) {
+                DecodeContext *decode = avctx->opaque;
+                AVQSVContext *qsv = av_qsv_alloc_context();
+                if (!qsv)
+                    return AV_PIX_FMT_NONE;
+
+                qsv->session   = decode->mfx_session;
+                qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
+
+                avctx->hwaccel_context = qsv;
+            }
+
+            return AV_PIX_FMT_QSV;
+        }
+
+        pix_fmts++;
+    }
+
+    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
+
+    return AV_PIX_FMT_NONE;
+}
+
+static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
+                         AVFrame *frame, AVPacket *pkt,
+                         AVIOContext *output_ctx)
+{
+    int ret = 0;
+    int got_frame = 1;
+
+    while (pkt->size > 0 || (!pkt->data && got_frame)) {
+        ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error during decoding\n");
+            return ret;
+        }
+
+        pkt->data += ret;
+        pkt->size -= ret;
+
+        /* A real program would do something useful with the decoded frame here.
+         * We just retrieve the raw data and write it to a file, which is rather
+         * useless but pedagogic. */
+        if (got_frame) {
+            mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
+            VASurfaceID    surface = *(VASurfaceID*)surf->Data.MemId;
+
+            VAImageFormat img_fmt = {
+                .fourcc         = VA_FOURCC_NV12,
+                .byte_order     = VA_LSB_FIRST,
+                .bits_per_pixel = 8,
+                .depth          = 8,
+            };
+
+            VAImage img;
+
+            VAStatus err;
+            uint8_t *data;
+            int i, j;
+
+            img.buf      = VA_INVALID_ID;
+            img.image_id = VA_INVALID_ID;
+
+            err = vaCreateImage(decode->va_dpy, &img_fmt,
+                                frame->width, frame->height, &img);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error creating an image: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            err = vaGetImage(decode->va_dpy, surface, 0, 0,
+                             frame->width, frame->height,
+                             img.image_id);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error getting an image: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error mapping the image buffer: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            for (i = 0; i < img.num_planes; i++)
+                for (j = 0; j < (img.height >> (i > 0)); j++)
+                    avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
+
+fail:
+            if (img.buf != VA_INVALID_ID)
+                vaUnmapBuffer(decode->va_dpy, img.buf);
+            if (img.image_id != VA_INVALID_ID)
+                vaDestroyImage(decode->va_dpy, img.image_id);
+            av_frame_unref(frame);
+
+            if (ret < 0)
+                return ret;
+        }
+    }
+
+    return 0;
+}
+
+int main(int argc, char **argv)
+{
+    AVFormatContext *input_ctx = NULL;
+    AVStream *video_st = NULL;
+    AVCodecContext *decoder_ctx = NULL;
+    const AVCodec *decoder;
+
+    AVPacket pkt = { 0 };
+    AVFrame *frame = NULL;
+
+    DecodeContext decode = { NULL };
+
+    Display *dpy = NULL;
+    int va_ver_major, va_ver_minor;
+
+    mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
+    mfxVersion mfx_ver = { { 1, 1 } };
+
+    mfxFrameAllocator frame_allocator = {
+        .pthis = &decode,
+        .Alloc = frame_alloc,
+        .Lock  = frame_lock,
+        .Unlock = frame_unlock,
+        .GetHDL = frame_get_hdl,
+        .Free   = frame_free,
+    };
+
+    AVIOContext *output_ctx = NULL;
+
+    int ret, i, err;
+
+    av_register_all();
+
+    if (argc < 3) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+        return 1;
+    }
+
+    /* open the input file */
+    ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
+        goto finish;
+    }
+
+    /* find the first H.264 video stream */
+    for (i = 0; i < input_ctx->nb_streams; i++) {
+        AVStream *st = input_ctx->streams[i];
+
+        if (st->codec->codec_id == AV_CODEC_ID_H264 && !video_st)
+            video_st = st;
+        else
+            st->discard = AVDISCARD_ALL;
+    }
+    if (!video_st) {
+        fprintf(stderr, "No H.264 video stream in the input file\n");
+        goto finish;
+    }
+
+    /* initialize VA-API */
+    dpy = XOpenDisplay(NULL);
+    if (!dpy) {
+        fprintf(stderr, "Cannot open the X display\n");
+        goto finish;
+    }
+    decode.va_dpy = vaGetDisplay(dpy);
+    if (!decode.va_dpy) {
+        fprintf(stderr, "Cannot open the VA display\n");
+        goto finish;
+    }
+
+    err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
+    if (err != VA_STATUS_SUCCESS) {
+        fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
+        goto finish;
+    }
+    fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
+
+    /* initialize an MFX session */
+    err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
+    if (err != MFX_ERR_NONE) {
+        fprintf(stderr, "Error initializing an MFX session\n");
+        goto finish;
+    }
+
+    MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
+    MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
+
+    /* initialize the decoder */
+    decoder = avcodec_find_decoder_by_name("h264_qsv");
+    if (!decoder) {
+        fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
+        goto finish;
+    }
+
+    decoder_ctx = avcodec_alloc_context3(decoder);
+    if (!decoder_ctx) {
+        ret = AVERROR(ENOMEM);
+        goto finish;
+    }
+    decoder_ctx->codec_id = AV_CODEC_ID_H264;
+    if (video_st->codec->extradata_size) {
+        decoder_ctx->extradata = av_mallocz(video_st->codec->extradata_size +
+                                            AV_INPUT_BUFFER_PADDING_SIZE);
+        if (!decoder_ctx->extradata) {
+            ret = AVERROR(ENOMEM);
+            goto finish;
+        }
+        memcpy(decoder_ctx->extradata, video_st->codec->extradata,
+               video_st->codec->extradata_size);
+        decoder_ctx->extradata_size = video_st->codec->extradata_size;
+    }
+    decoder_ctx->refcounted_frames = 1;
+
+    decoder_ctx->opaque      = &decode;
+    decoder_ctx->get_buffer2 = get_buffer;
+    decoder_ctx->get_format  = get_format;
+
+    ret = avcodec_open2(decoder_ctx, NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Error opening the decoder: ");
+        goto finish;
+    }
+
+    /* open the output stream */
+    ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
+    if (ret < 0) {
+        fprintf(stderr, "Error opening the output context: ");
+        goto finish;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        ret = AVERROR(ENOMEM);
+        goto finish;
+    }
+
+    /* actual decoding */
+    while (ret >= 0) {
+        ret = av_read_frame(input_ctx, &pkt);
+        if (ret < 0)
+            break;
+
+        if (pkt.stream_index == video_st->index)
+            ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
+
+        av_packet_unref(&pkt);
+    }
+
+    /* flush the decoder */
+    pkt.data = NULL;
+    pkt.size = 0;
+    ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
+
+finish:
+    if (ret < 0) {
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "%s\n", buf);
+    }
+
+    avformat_close_input(&input_ctx);
+
+    av_frame_free(&frame);
+
+    if (decoder_ctx)
+        av_freep(&decoder_ctx->hwaccel_context);
+    avcodec_free_context(&decoder_ctx);
+
+    free_surfaces(&decode);
+
+    if (decode.mfx_session)
+        MFXClose(decode.mfx_session);
+    if (decode.va_dpy)
+        vaTerminate(decode.va_dpy);
+    if (dpy)
+        XCloseDisplay(dpy);
+
+    avio_close(output_ctx);
+
+    return ret;
+}
@@ -0,0 +1,165 @@
+/*
+ * Copyright (c) 2013 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat/libavcodec demuxing and muxing API example.
+ *
+ * Remux streams from one container format to another.
+ * @example remuxing.c
+ */
+
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           tag,
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+int main(int argc, char **argv)
+{
+    AVOutputFormat *ofmt = NULL;
+    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
+    AVPacket pkt;
+    const char *in_filename, *out_filename;
+    int ret, i;
+
+    if (argc < 3) {
+        printf("usage: %s input output\n"
+               "API example program to remux a media file with libavformat and libavcodec.\n"
+               "The output format is guessed according to the file extension.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_filename  = argv[1];
+    out_filename = argv[2];
+
+    av_register_all();
+
+    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
+        fprintf(stderr, "Could not open input file '%s'", in_filename);
+        goto end;
+    }
+
+    if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
+        fprintf(stderr, "Failed to retrieve input stream information");
+        goto end;
+    }
+
+    av_dump_format(ifmt_ctx, 0, in_filename, 0);
+
+    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
+    if (!ofmt_ctx) {
+        fprintf(stderr, "Could not create output context\n");
+        ret = AVERROR_UNKNOWN;
+        goto end;
+    }
+
+    ofmt = ofmt_ctx->oformat;
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        AVStream *in_stream = ifmt_ctx->streams[i];
+        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
+        if (!out_stream) {
+            fprintf(stderr, "Failed allocating output stream\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
+        if (ret < 0) {
+            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
+            goto end;
+        }
+        out_stream->codec->codec_tag = 0;
+        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+    }
+    av_dump_format(ofmt_ctx, 0, out_filename, 1);
+
+    if (!(ofmt->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            fprintf(stderr, "Could not open output file '%s'", out_filename);
+            goto end;
+        }
+    }
+
+    ret = avformat_write_header(ofmt_ctx, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred when opening output file\n");
+        goto end;
+    }
+
+    while (1) {
+        AVStream *in_stream, *out_stream;
+
+        ret = av_read_frame(ifmt_ctx, &pkt);
+        if (ret < 0)
+            break;
+
+        in_stream  = ifmt_ctx->streams[pkt.stream_index];
+        out_stream = ofmt_ctx->streams[pkt.stream_index];
+
+        log_packet(ifmt_ctx, &pkt, "in");
+
+        /* copy packet */
+        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
+        pkt.pos = -1;
+        log_packet(ofmt_ctx, &pkt, "out");
+
+        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error muxing packet\n");
+            break;
+        }
+        av_packet_unref(&pkt);
+    }
+
+    av_write_trailer(ofmt_ctx);
+end:
+
+    avformat_close_input(&ifmt_ctx);
+
+    /* close output */
+    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
+        avio_closep(&ofmt_ctx->pb);
+    avformat_free_context(ofmt_ctx);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}
@@ -0,0 +1,214 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @example resampling_audio.c
+ * libswresample API use example.
+ */
+
+#include <libavutil/opt.h>
+#include <libavutil/channel_layout.h>
+#include <libavutil/samplefmt.h>
+#include <libswresample/swresample.h>
+
+static int get_format_from_sample_fmt(const char **fmt,
+                                      enum AVSampleFormat sample_fmt)
+{
+    int i;
+    struct sample_fmt_entry {
+        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
+    } sample_fmt_entries[] = {
+        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
+        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
+        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
+        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
+        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
+    };
+    *fmt = NULL;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
+        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
+        if (sample_fmt == entry->sample_fmt) {
+            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
+            return 0;
+        }
+    }
+
+    fprintf(stderr,
+            "Sample format %s not supported as output format\n",
+            av_get_sample_fmt_name(sample_fmt));
+    return AVERROR(EINVAL);
+}
+
+/**
+ * Fill dst buffer with nb_samples, generated starting from t.
+ */
+static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
+{
+    int i, j;
+    double tincr = 1.0 / sample_rate, *dstp = dst;
+    const double c = 2 * M_PI * 440.0;
+
+    /* generate sin tone with 440Hz frequency and duplicated channels */
+    for (i = 0; i < nb_samples; i++) {
+        *dstp = sin(c * *t);
+        for (j = 1; j < nb_channels; j++)
+            dstp[j] = dstp[0];
+        dstp += nb_channels;
+        *t += tincr;
+    }
+}
+
+int main(int argc, char **argv)
+{
+    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
+    int src_rate = 48000, dst_rate = 44100;
+    uint8_t **src_data = NULL, **dst_data = NULL;
+    int src_nb_channels = 0, dst_nb_channels = 0;
+    int src_linesize, dst_linesize;
+    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
+    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
+    const char *dst_filename = NULL;
+    FILE *dst_file;
+    int dst_bufsize;
+    const char *fmt;
+    struct SwrContext *swr_ctx;
+    double t;
+    int ret;
+
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s output_file\n"
+                "API example program to show how to resample an audio stream with libswresample.\n"
+                "This program generates a series of audio frames, resamples them to a specified "
+                "output format and rate and saves them to an output file named output_file.\n",
+            argv[0]);
+        exit(1);
+    }
+    dst_filename = argv[1];
+
+    dst_file = fopen(dst_filename, "wb");
+    if (!dst_file) {
+        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
+        exit(1);
+    }
+
+    /* create resampler context */
+    swr_ctx = swr_alloc();
+    if (!swr_ctx) {
+        fprintf(stderr, "Could not allocate resampler context\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    /* set options */
+    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
+    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
+
+    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
+    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
+    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
+
+    /* initialize the resampling context */
+    if ((ret = swr_init(swr_ctx)) < 0) {
+        fprintf(stderr, "Failed to initialize the resampling context\n");
+        goto end;
+    }
+
+    /* allocate source and destination samples buffers */
+
+    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
+    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
+                                             src_nb_samples, src_sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
+        goto end;
+    }
+
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = dst_nb_samples =
+        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+
+    /* buffer is going to be directly written to a rawaudio file, no alignment */
+    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
+    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
+                                             dst_nb_samples, dst_sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        goto end;
+    }
+
+    t = 0;
+    do {
+        /* generate synthetic audio */
+        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
+
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
+                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_freep(&dst_data[0]);
+            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
+                                   dst_nb_samples, dst_sample_fmt, 1);
+            if (ret < 0)
+                break;
+            max_dst_nb_samples = dst_nb_samples;
+        }
+
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            goto end;
+        }
+        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
+                                                 ret, dst_sample_fmt, 1);
+        if (dst_bufsize < 0) {
+            fprintf(stderr, "Could not get sample buffer size\n");
+            goto end;
+        }
+        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
+        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
+    } while (t < 10);
+
+    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
+        goto end;
+    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
+            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
+            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
+
+end:
+    fclose(dst_file);
+
+    if (src_data)
+        av_freep(&src_data[0]);
+    av_freep(&src_data);
+
+    if (dst_data)
+        av_freep(&dst_data[0]);
+    av_freep(&dst_data);
+
+    swr_free(&swr_ctx);
+    return ret < 0;
+}
@@ -0,0 +1,140 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libswscale API use example.
+ * @example scaling_video.c
+ */
+
+#include <libavutil/imgutils.h>
+#include <libavutil/parseutils.h>
+#include <libswscale/swscale.h>
+
+static void fill_yuv_image(uint8_t *data[4], int linesize[4],
+                           int width, int height, int frame_index)
+{
+    int x, y;
+
+    /* Y */
+    for (y = 0; y < height; y++)
+        for (x = 0; x < width; x++)
+            data[0][y * linesize[0] + x] = x + y + frame_index * 3;
+
+    /* Cb and Cr */
+    for (y = 0; y < height / 2; y++) {
+        for (x = 0; x < width / 2; x++) {
+            data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
+            data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
+        }
+    }
+}
+
+int main(int argc, char **argv)
+{
+    uint8_t *src_data[4], *dst_data[4];
+    int src_linesize[4], dst_linesize[4];
+    int src_w = 320, src_h = 240, dst_w, dst_h;
+    enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
+    const char *dst_size = NULL;
+    const char *dst_filename = NULL;
+    FILE *dst_file;
+    int dst_bufsize;
+    struct SwsContext *sws_ctx;
+    int i, ret;
+
+    if (argc != 3) {
+        fprintf(stderr, "Usage: %s output_file output_size\n"
+                "API example program to show how to scale an image with libswscale.\n"
+                "This program generates a series of pictures, rescales them to the given "
+                "output_size and saves them to an output file named output_file\n."
+                "\n", argv[0]);
+        exit(1);
+    }
+    dst_filename = argv[1];
+    dst_size     = argv[2];
+
+    if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
+        fprintf(stderr,
+                "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
+                dst_size);
+        exit(1);
+    }
+
+    dst_file = fopen(dst_filename, "wb");
+    if (!dst_file) {
+        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
+        exit(1);
+    }
+
+    /* create scaling context */
+    sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
+                             dst_w, dst_h, dst_pix_fmt,
+                             SWS_BILINEAR, NULL, NULL, NULL);
+    if (!sws_ctx) {
+        fprintf(stderr,
+                "Impossible to create scale context for the conversion "
+                "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
+                av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
+                av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
+        ret = AVERROR(EINVAL);
+        goto end;
+    }
+
+    /* allocate source and destination image buffers */
+    if ((ret = av_image_alloc(src_data, src_linesize,
+                              src_w, src_h, src_pix_fmt, 16)) < 0) {
+        fprintf(stderr, "Could not allocate source image\n");
+        goto end;
+    }
+
+    /* buffer is going to be written to rawvideo file, no alignment */
+    if ((ret = av_image_alloc(dst_data, dst_linesize,
+                              dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
+        fprintf(stderr, "Could not allocate destination image\n");
+        goto end;
+    }
+    dst_bufsize = ret;
+
+    for (i = 0; i < 100; i++) {
+        /* generate synthetic video */
+        fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
+
+        /* convert to destination format */
+        sws_scale(sws_ctx, (const uint8_t * const*)src_data,
+                  src_linesize, 0, src_h, dst_data, dst_linesize);
+
+        /* write scaled image to file */
+        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
+    }
+
+    fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
+           "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
+           av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
+
+end:
+    fclose(dst_file);
+    av_freep(&src_data[0]);
+    av_freep(&dst_data[0]);
+    sws_freeContext(sws_ctx);
+    return ret < 0;
+}
@@ -0,0 +1,770 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * simple audio converter
+ *
+ * @example transcode_aac.c
+ * Convert an input audio file to AAC in an MP4 container using FFmpeg.
+ * @author Andreas Unterweger (dustsigns@gmail.com)
+ */
+
+#include <stdio.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/frame.h"
+#include "libavutil/opt.h"
+
+#include "libswresample/swresample.h"
+
+/** The output bit rate in kbit/s */
+#define OUTPUT_BIT_RATE 96000
+/** The number of output channels */
+#define OUTPUT_CHANNELS 2
+
+/**
+ * Convert an error code into a text message.
+ * @param error Error code to be converted
+ * @return Corresponding error text (not thread-safe)
+ */
+static const char *get_error_text(const int error)
+{
+    static char error_buffer[255];
+    av_strerror(error, error_buffer, sizeof(error_buffer));
+    return error_buffer;
+}
+
+/** Open an input file and the required decoder. */
+static int open_input_file(const char *filename,
+                           AVFormatContext **input_format_context,
+                           AVCodecContext **input_codec_context)
+{
+    AVCodec *input_codec;
+    int error;
+
+    /** Open the input file to read from it. */
+    if ((error = avformat_open_input(input_format_context, filename, NULL,
+                                     NULL)) < 0) {
+        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
+                filename, get_error_text(error));
+        *input_format_context = NULL;
+        return error;
+    }
+
+    /** Get information on the input file (number of streams etc.). */
+    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
+        fprintf(stderr, "Could not open find stream info (error '%s')\n",
+                get_error_text(error));
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /** Make sure that there is only one stream in the input file. */
+    if ((*input_format_context)->nb_streams != 1) {
+        fprintf(stderr, "Expected one audio input stream, but found %d\n",
+                (*input_format_context)->nb_streams);
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /** Find a decoder for the audio stream. */
+    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
+        fprintf(stderr, "Could not find input codec\n");
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /** Open the decoder for the audio stream to use it later. */
+    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
+                               input_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open input codec (error '%s')\n",
+                get_error_text(error));
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /** Save the decoder context for easier access later. */
+    *input_codec_context = (*input_format_context)->streams[0]->codec;
+
+    return 0;
+}
+
+/**
+ * Open an output file and the required encoder.
+ * Also set some basic encoder parameters.
+ * Some of these parameters are based on the input file's parameters.
+ */
+static int open_output_file(const char *filename,
+                            AVCodecContext *input_codec_context,
+                            AVFormatContext **output_format_context,
+                            AVCodecContext **output_codec_context)
+{
+    AVIOContext *output_io_context = NULL;
+    AVStream *stream               = NULL;
+    AVCodec *output_codec          = NULL;
+    int error;
+
+    /** Open the output file to write to it. */
+    if ((error = avio_open(&output_io_context, filename,
+                           AVIO_FLAG_WRITE)) < 0) {
+        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
+                filename, get_error_text(error));
+        return error;
+    }
+
+    /** Create a new format context for the output container format. */
+    if (!(*output_format_context = avformat_alloc_context())) {
+        fprintf(stderr, "Could not allocate output format context\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /** Associate the output file (pointer) with the container format context. */
+    (*output_format_context)->pb = output_io_context;
+
+    /** Guess the desired container format based on the file extension. */
+    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
+                                                              NULL))) {
+        fprintf(stderr, "Could not find output file format\n");
+        goto cleanup;
+    }
+
+    av_strlcpy((*output_format_context)->filename, filename,
+               sizeof((*output_format_context)->filename));
+
+    /** Find the encoder to be used by its name. */
+    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
+        fprintf(stderr, "Could not find an AAC encoder.\n");
+        goto cleanup;
+    }
+
+    /** Create a new audio stream in the output file container. */
+    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
+        fprintf(stderr, "Could not create new stream\n");
+        error = AVERROR(ENOMEM);
+        goto cleanup;
+    }
+
+    /** Save the encoder context for easier access later. */
+    *output_codec_context = stream->codec;
+
+    /**
+     * Set the basic encoder parameters.
+     * The input file's sample rate is used to avoid a sample rate conversion.
+     */
+    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
+    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
+    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
+    (*output_codec_context)->sample_fmt     = output_codec->sample_fmts[0];
+    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
+
+    /** Allow the use of the experimental AAC encoder */
+    (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+
+    /** Set the sample rate for the container. */
+    stream->time_base.den = input_codec_context->sample_rate;
+    stream->time_base.num = 1;
+
+    /**
+     * Some container formats (like MP4) require global headers to be present
+     * Mark the encoder so that it behaves accordingly.
+     */
+    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
+        (*output_codec_context)->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+
+    /** Open the encoder for the audio stream to use it later. */
+    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open output codec (error '%s')\n",
+                get_error_text(error));
+        goto cleanup;
+    }
+
+    return 0;
+
+cleanup:
+    avio_closep(&(*output_format_context)->pb);
+    avformat_free_context(*output_format_context);
+    *output_format_context = NULL;
+    return error < 0 ? error : AVERROR_EXIT;
+}
+
+/** Initialize one data packet for reading or writing. */
+static void init_packet(AVPacket *packet)
+{
+    av_init_packet(packet);
+    /** Set the packet data and size so that it is recognized as being empty. */
+    packet->data = NULL;
+    packet->size = 0;
+}
+
+/** Initialize one audio frame for reading from the input file */
+static int init_input_frame(AVFrame **frame)
+{
+    if (!(*frame = av_frame_alloc())) {
+        fprintf(stderr, "Could not allocate input frame\n");
+        return AVERROR(ENOMEM);
+    }
+    return 0;
+}
+
+/**
+ * Initialize the audio resampler based on the input and output codec settings.
+ * If the input and output sample formats differ, a conversion is required
+ * libswresample takes care of this, but requires initialization.
+ */
+static int init_resampler(AVCodecContext *input_codec_context,
+                          AVCodecContext *output_codec_context,
+                          SwrContext **resample_context)
+{
+        int error;
+
+        /**
+         * Create a resampler context for the conversion.
+         * Set the conversion parameters.
+         * Default channel layouts based on the number of channels
+         * are assumed for simplicity (they are sometimes not detected
+         * properly by the demuxer and/or decoder).
+         */
+        *resample_context = swr_alloc_set_opts(NULL,
+                                              av_get_default_channel_layout(output_codec_context->channels),
+                                              output_codec_context->sample_fmt,
+                                              output_codec_context->sample_rate,
+                                              av_get_default_channel_layout(input_codec_context->channels),
+                                              input_codec_context->sample_fmt,
+                                              input_codec_context->sample_rate,
+                                              0, NULL);
+        if (!*resample_context) {
+            fprintf(stderr, "Could not allocate resample context\n");
+            return AVERROR(ENOMEM);
+        }
+        /**
+        * Perform a sanity check so that the number of converted samples is
+        * not greater than the number of samples to be converted.
+        * If the sample rates differ, this case has to be handled differently
+        */
+        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
+
+        /** Open the resampler with the specified parameters. */
+        if ((error = swr_init(*resample_context)) < 0) {
+            fprintf(stderr, "Could not open resample context\n");
+            swr_free(resample_context);
+            return error;
+        }
+    return 0;
+}
+
+/** Initialize a FIFO buffer for the audio samples to be encoded. */
+static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
+{
+    /** Create the FIFO buffer based on the specified output sample format. */
+    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
+                                      output_codec_context->channels, 1))) {
+        fprintf(stderr, "Could not allocate FIFO\n");
+        return AVERROR(ENOMEM);
+    }
+    return 0;
+}
+
+/** Write the header of the output file container. */
+static int write_output_file_header(AVFormatContext *output_format_context)
+{
+    int error;
+    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
+        fprintf(stderr, "Could not write output file header (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+    return 0;
+}
+
+/** Decode one audio frame from the input file. */
+static int decode_audio_frame(AVFrame *frame,
+                              AVFormatContext *input_format_context,
+                              AVCodecContext *input_codec_context,
+                              int *data_present, int *finished)
+{
+    /** Packet used for temporary storage. */
+    AVPacket input_packet;
+    int error;
+    init_packet(&input_packet);
+
+    /** Read one audio frame from the input file into a temporary packet. */
+    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
+        /** If we are at the end of the file, flush the decoder below. */
+        if (error == AVERROR_EOF)
+            *finished = 1;
+        else {
+            fprintf(stderr, "Could not read frame (error '%s')\n",
+                    get_error_text(error));
+            return error;
+        }
+    }
+
+    /**
+     * Decode the audio frame stored in the temporary packet.
+     * The input audio stream decoder is used to do this.
+     * If we are at the end of the file, pass an empty packet to the decoder
+     * to flush it.
+     */
+    if ((error = avcodec_decode_audio4(input_codec_context, frame,
+                                       data_present, &input_packet)) < 0) {
+        fprintf(stderr, "Could not decode frame (error '%s')\n",
+                get_error_text(error));
+        av_packet_unref(&input_packet);
+        return error;
+    }
+
+    /**
+     * If the decoder has not been flushed completely, we are not finished,
+     * so that this function has to be called again.
+     */
+    if (*finished && *data_present)
+        *finished = 0;
+    av_packet_unref(&input_packet);
+    return 0;
+}
+
+/**
+ * Initialize a temporary storage for the specified number of audio samples.
+ * The conversion requires temporary storage due to the different format.
+ * The number of audio samples to be allocated is specified in frame_size.
+ */
+static int init_converted_samples(uint8_t ***converted_input_samples,
+                                  AVCodecContext *output_codec_context,
+                                  int frame_size)
+{
+    int error;
+
+    /**
+     * Allocate as many pointers as there are audio channels.
+     * Each pointer will later point to the audio samples of the corresponding
+     * channels (although it may be NULL for interleaved formats).
+     */
+    if (!(*converted_input_samples = calloc(output_codec_context->channels,
+                                            sizeof(**converted_input_samples)))) {
+        fprintf(stderr, "Could not allocate converted input sample pointers\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /**
+     * Allocate memory for the samples of all channels in one consecutive
+     * block for convenience.
+     */
+    if ((error = av_samples_alloc(*converted_input_samples, NULL,
+                                  output_codec_context->channels,
+                                  frame_size,
+                                  output_codec_context->sample_fmt, 0)) < 0) {
+        fprintf(stderr,
+                "Could not allocate converted input samples (error '%s')\n",
+                get_error_text(error));
+        av_freep(&(*converted_input_samples)[0]);
+        free(*converted_input_samples);
+        return error;
+    }
+    return 0;
+}
+
+/**
+ * Convert the input audio samples into the output sample format.
+ * The conversion happens on a per-frame basis, the size of which is specified
+ * by frame_size.
+ */
+static int convert_samples(const uint8_t **input_data,
+                           uint8_t **converted_data, const int frame_size,
+                           SwrContext *resample_context)
+{
+    int error;
+
+    /** Convert the samples using the resampler. */
+    if ((error = swr_convert(resample_context,
+                             converted_data, frame_size,
+                             input_data    , frame_size)) < 0) {
+        fprintf(stderr, "Could not convert input samples (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+
+    return 0;
+}
+
+/** Add converted input audio samples to the FIFO buffer for later processing. */
+static int add_samples_to_fifo(AVAudioFifo *fifo,
+                               uint8_t **converted_input_samples,
+                               const int frame_size)
+{
+    int error;
+
+    /**
+     * Make the FIFO as large as it needs to be to hold both,
+     * the old and the new samples.
+     */
+    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
+        fprintf(stderr, "Could not reallocate FIFO\n");
+        return error;
+    }
+
+    /** Store the new samples in the FIFO buffer. */
+    if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
+                            frame_size) < frame_size) {
+        fprintf(stderr, "Could not write data to FIFO\n");
+        return AVERROR_EXIT;
+    }
+    return 0;
+}
+
+/**
+ * Read one audio frame from the input file, decodes, converts and stores
+ * it in the FIFO buffer.
+ */
+static int read_decode_convert_and_store(AVAudioFifo *fifo,
+                                         AVFormatContext *input_format_context,
+                                         AVCodecContext *input_codec_context,
+                                         AVCodecContext *output_codec_context,
+                                         SwrContext *resampler_context,
+                                         int *finished)
+{
+    /** Temporary storage of the input samples of the frame read from the file. */
+    AVFrame *input_frame = NULL;
+    /** Temporary storage for the converted input samples. */
+    uint8_t **converted_input_samples = NULL;
+    int data_present;
+    int ret = AVERROR_EXIT;
+
+    /** Initialize temporary storage for one input frame. */
+    if (init_input_frame(&input_frame))
+        goto cleanup;
+    /** Decode one frame worth of audio samples. */
+    if (decode_audio_frame(input_frame, input_format_context,
+                           input_codec_context, &data_present, finished))
+        goto cleanup;
+    /**
+     * If we are at the end of the file and there are no more samples
+     * in the decoder which are delayed, we are actually finished.
+     * This must not be treated as an error.
+     */
+    if (*finished && !data_present) {
+        ret = 0;
+        goto cleanup;
+    }
+    /** If there is decoded data, convert and store it */
+    if (data_present) {
+        /** Initialize the temporary storage for the converted input samples. */
+        if (init_converted_samples(&converted_input_samples, output_codec_context,
+                                   input_frame->nb_samples))
+            goto cleanup;
+
+        /**
+         * Convert the input samples to the desired output sample format.
+         * This requires a temporary storage provided by converted_input_samples.
+         */
+        if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
+                            input_frame->nb_samples, resampler_context))
+            goto cleanup;
+
+        /** Add the converted input samples to the FIFO buffer for later processing. */
+        if (add_samples_to_fifo(fifo, converted_input_samples,
+                                input_frame->nb_samples))
+            goto cleanup;
+        ret = 0;
+    }
+    ret = 0;
+
+cleanup:
+    if (converted_input_samples) {
+        av_freep(&converted_input_samples[0]);
+        free(converted_input_samples);
+    }
+    av_frame_free(&input_frame);
+
+    return ret;
+}
+
+/**
+ * Initialize one input frame for writing to the output file.
+ * The frame will be exactly frame_size samples large.
+ */
+static int init_output_frame(AVFrame **frame,
+                             AVCodecContext *output_codec_context,
+                             int frame_size)
+{
+    int error;
+
+    /** Create a new frame to store the audio samples. */
+    if (!(*frame = av_frame_alloc())) {
+        fprintf(stderr, "Could not allocate output frame\n");
+        return AVERROR_EXIT;
+    }
+
+    /**
+     * Set the frame's parameters, especially its size and format.
+     * av_frame_get_buffer needs this to allocate memory for the
+     * audio samples of the frame.
+     * Default channel layouts based on the number of channels
+     * are assumed for simplicity.
+     */
+    (*frame)->nb_samples     = frame_size;
+    (*frame)->channel_layout = output_codec_context->channel_layout;
+    (*frame)->format         = output_codec_context->sample_fmt;
+    (*frame)->sample_rate    = output_codec_context->sample_rate;
+
+    /**
+     * Allocate the samples of the created frame. This call will make
+     * sure that the audio frame can hold as many samples as specified.
+     */
+    if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
+        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
+                get_error_text(error));
+        av_frame_free(frame);
+        return error;
+    }
+
+    return 0;
+}
+
+/** Global timestamp for the audio frames */
+static int64_t pts = 0;
+
+/** Encode one frame worth of audio to the output file. */
+static int encode_audio_frame(AVFrame *frame,
+                              AVFormatContext *output_format_context,
+                              AVCodecContext *output_codec_context,
+                              int *data_present)
+{
+    /** Packet used for temporary storage. */
+    AVPacket output_packet;
+    int error;
+    init_packet(&output_packet);
+
+    /** Set a timestamp based on the sample rate for the container. */
+    if (frame) {
+        frame->pts = pts;
+        pts += frame->nb_samples;
+    }
+
+    /**
+     * Encode the audio frame and store it in the temporary packet.
+     * The output audio stream encoder is used to do this.
+     */
+    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
+                                       frame, data_present)) < 0) {
+        fprintf(stderr, "Could not encode frame (error '%s')\n",
+                get_error_text(error));
+        av_packet_unref(&output_packet);
+        return error;
+    }
+
+    /** Write one audio frame from the temporary packet to the output file. */
+    if (*data_present) {
+        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
+            fprintf(stderr, "Could not write frame (error '%s')\n",
+                    get_error_text(error));
+            av_packet_unref(&output_packet);
+            return error;
+        }
+
+        av_packet_unref(&output_packet);
+    }
+
+    return 0;
+}
+
+/**
+ * Load one audio frame from the FIFO buffer, encode and write it to the
+ * output file.
+ */
+static int load_encode_and_write(AVAudioFifo *fifo,
+                                 AVFormatContext *output_format_context,
+                                 AVCodecContext *output_codec_context)
+{
+    /** Temporary storage of the output samples of the frame written to the file. */
+    AVFrame *output_frame;
+    /**
+     * Use the maximum number of possible samples per frame.
+     * If there is less than the maximum possible frame size in the FIFO
+     * buffer use this number. Otherwise, use the maximum possible frame size
+     */
+    const int frame_size = FFMIN(av_audio_fifo_size(fifo),
+                                 output_codec_context->frame_size);
+    int data_written;
+
+    /** Initialize temporary storage for one output frame. */
+    if (init_output_frame(&output_frame, output_codec_context, frame_size))
+        return AVERROR_EXIT;
+
+    /**
+     * Read as many samples from the FIFO buffer as required to fill the frame.
+     * The samples are stored in the frame temporarily.
+     */
+    if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
+        fprintf(stderr, "Could not read data from FIFO\n");
+        av_frame_free(&output_frame);
+        return AVERROR_EXIT;
+    }
+
+    /** Encode one frame worth of audio samples. */
+    if (encode_audio_frame(output_frame, output_format_context,
+                           output_codec_context, &data_written)) {
+        av_frame_free(&output_frame);
+        return AVERROR_EXIT;
+    }
+    av_frame_free(&output_frame);
+    return 0;
+}
+
+/** Write the trailer of the output file container. */
+static int write_output_file_trailer(AVFormatContext *output_format_context)
+{
+    int error;
+    if ((error = av_write_trailer(output_format_context)) < 0) {
+        fprintf(stderr, "Could not write output file trailer (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+    return 0;
+}
+
+/** Convert an audio file to an AAC file in an MP4 container. */
+int main(int argc, char **argv)
+{
+    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
+    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
+    SwrContext *resample_context = NULL;
+    AVAudioFifo *fifo = NULL;
+    int ret = AVERROR_EXIT;
+
+    if (argc < 3) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+        exit(1);
+    }
+
+    /** Register all codecs and formats so that they can be used. */
+    av_register_all();
+    /** Open the input file for reading. */
+    if (open_input_file(argv[1], &input_format_context,
+                        &input_codec_context))
+        goto cleanup;
+    /** Open the output file for writing. */
+    if (open_output_file(argv[2], input_codec_context,
+                         &output_format_context, &output_codec_context))
+        goto cleanup;
+    /** Initialize the resampler to be able to convert audio sample formats. */
+    if (init_resampler(input_codec_context, output_codec_context,
+                       &resample_context))
+        goto cleanup;
+    /** Initialize the FIFO buffer to store audio samples to be encoded. */
+    if (init_fifo(&fifo, output_codec_context))
+        goto cleanup;
+    /** Write the header of the output file container. */
+    if (write_output_file_header(output_format_context))
+        goto cleanup;
+
+    /**
+     * Loop as long as we have input samples to read or output samples
+     * to write; abort as soon as we have neither.
+     */
+    while (1) {
+        /** Use the encoder's desired frame size for processing. */
+        const int output_frame_size = output_codec_context->frame_size;
+        int finished                = 0;
+
+        /**
+         * Make sure that there is one frame worth of samples in the FIFO
+         * buffer so that the encoder can do its work.
+         * Since the decoder's and the encoder's frame size may differ, we
+         * need to FIFO buffer to store as many frames worth of input samples
+         * that they make up at least one frame worth of output samples.
+         */
+        while (av_audio_fifo_size(fifo) < output_frame_size) {
+            /**
+             * Decode one frame worth of audio samples, convert it to the
+             * output sample format and put it into the FIFO buffer.
+             */
+            if (read_decode_convert_and_store(fifo, input_format_context,
+                                              input_codec_context,
+                                              output_codec_context,
+                                              resample_context, &finished))
+                goto cleanup;
+
+            /**
+             * If we are at the end of the input file, we continue
+             * encoding the remaining audio samples to the output file.
+             */
+            if (finished)
+                break;
+        }
+
+        /**
+         * If we have enough samples for the encoder, we encode them.
+         * At the end of the file, we pass the remaining samples to
+         * the encoder.
+         */
+        while (av_audio_fifo_size(fifo) >= output_frame_size ||
+               (finished && av_audio_fifo_size(fifo) > 0))
+            /**
+             * Take one frame worth of audio samples from the FIFO buffer,
+             * encode it and write it to the output file.
+             */
+            if (load_encode_and_write(fifo, output_format_context,
+                                      output_codec_context))
+                goto cleanup;
+
+        /**
+         * If we are at the end of the input file and have encoded
+         * all remaining samples, we can exit this loop and finish.
+         */
+        if (finished) {
+            int data_written;
+            /** Flush the encoder as it may have delayed frames. */
+            do {
+                if (encode_audio_frame(NULL, output_format_context,
+                                       output_codec_context, &data_written))
+                    goto cleanup;
+            } while (data_written);
+            break;
+        }
+    }
+
+    /** Write the trailer of the output file container. */
+    if (write_output_file_trailer(output_format_context))
+        goto cleanup;
+    ret = 0;
+
+cleanup:
+    if (fifo)
+        av_audio_fifo_free(fifo);
+    swr_free(&resample_context);
+    if (output_codec_context)
+        avcodec_close(output_codec_context);
+    if (output_format_context) {
+        avio_closep(&output_format_context->pb);
+        avformat_free_context(output_format_context);
+    }
+    if (input_codec_context)
+        avcodec_close(input_codec_context);
+    if (input_format_context)
+        avformat_close_input(&input_format_context);
+
+    return ret;
+}
@@ -0,0 +1,582 @@
+/*
+ * Copyright (c) 2010 Nicolas George
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2014 Andrey Utkin
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * API example for demuxing, decoding, filtering, encoding and muxing
+ * @example transcoding.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavfilter/avfiltergraph.h>
+#include <libavfilter/buffersink.h>
+#include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
+#include <libavutil/pixdesc.h>
+
+static AVFormatContext *ifmt_ctx;
+static AVFormatContext *ofmt_ctx;
+typedef struct FilteringContext {
+    AVFilterContext *buffersink_ctx;
+    AVFilterContext *buffersrc_ctx;
+    AVFilterGraph *filter_graph;
+} FilteringContext;
+static FilteringContext *filter_ctx;
+
+static int open_input_file(const char *filename)
+{
+    int ret;
+    unsigned int i;
+
+    ifmt_ctx = NULL;
+    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
+        return ret;
+    }
+
+    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
+        return ret;
+    }
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        AVStream *stream;
+        AVCodecContext *codec_ctx;
+        stream = ifmt_ctx->streams[i];
+        codec_ctx = stream->codec;
+        /* Reencode video & audio and remux subtitles etc. */
+        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
+                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
+            /* Open decoder */
+            ret = avcodec_open2(codec_ctx,
+                    avcodec_find_decoder(codec_ctx->codec_id), NULL);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
+                return ret;
+            }
+        }
+    }
+
+    av_dump_format(ifmt_ctx, 0, filename, 0);
+    return 0;
+}
+
+static int open_output_file(const char *filename)
+{
+    AVStream *out_stream;
+    AVStream *in_stream;
+    AVCodecContext *dec_ctx, *enc_ctx;
+    AVCodec *encoder;
+    int ret;
+    unsigned int i;
+
+    ofmt_ctx = NULL;
+    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
+    if (!ofmt_ctx) {
+        av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
+        return AVERROR_UNKNOWN;
+    }
+
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        out_stream = avformat_new_stream(ofmt_ctx, NULL);
+        if (!out_stream) {
+            av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
+            return AVERROR_UNKNOWN;
+        }
+
+        in_stream = ifmt_ctx->streams[i];
+        dec_ctx = in_stream->codec;
+        enc_ctx = out_stream->codec;
+
+        if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
+                || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
+            /* in this example, we choose transcoding to same codec */
+            encoder = avcodec_find_encoder(dec_ctx->codec_id);
+            if (!encoder) {
+                av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
+                return AVERROR_INVALIDDATA;
+            }
+
+            /* In this example, we transcode to same properties (picture size,
+             * sample rate etc.). These properties can be changed for output
+             * streams easily using filters */
+            if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
+                enc_ctx->height = dec_ctx->height;
+                enc_ctx->width = dec_ctx->width;
+                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
+                /* take first format from list of supported formats */
+                enc_ctx->pix_fmt = encoder->pix_fmts[0];
+                /* video time_base can be set to whatever is handy and supported by encoder */
+                enc_ctx->time_base = dec_ctx->time_base;
+            } else {
+                enc_ctx->sample_rate = dec_ctx->sample_rate;
+                enc_ctx->channel_layout = dec_ctx->channel_layout;
+                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
+                /* take first format from list of supported formats */
+                enc_ctx->sample_fmt = encoder->sample_fmts[0];
+                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
+            }
+
+            /* Third parameter can be used to pass settings to encoder */
+            ret = avcodec_open2(enc_ctx, encoder, NULL);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
+                return ret;
+            }
+        } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
+            av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
+            return AVERROR_INVALIDDATA;
+        } else {
+            /* if this stream must be remuxed */
+            ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
+                    ifmt_ctx->streams[i]->codec);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
+                return ret;
+            }
+        }
+
+        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+            enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+
+    }
+    av_dump_format(ofmt_ctx, 0, filename, 1);
+
+    if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
+            return ret;
+        }
+    }
+
+    /* init muxer, write output file header */
+    ret = avformat_write_header(ofmt_ctx, NULL);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
+        AVCodecContext *enc_ctx, const char *filter_spec)
+{
+    char args[512];
+    int ret = 0;
+    AVFilter *buffersrc = NULL;
+    AVFilter *buffersink = NULL;
+    AVFilterContext *buffersrc_ctx = NULL;
+    AVFilterContext *buffersink_ctx = NULL;
+    AVFilterInOut *outputs = avfilter_inout_alloc();
+    AVFilterInOut *inputs  = avfilter_inout_alloc();
+    AVFilterGraph *filter_graph = avfilter_graph_alloc();
+
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
+        buffersrc = avfilter_get_by_name("buffer");
+        buffersink = avfilter_get_by_name("buffersink");
+        if (!buffersrc || !buffersink) {
+            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        snprintf(args, sizeof(args),
+                "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
+                dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
+                dec_ctx->time_base.num, dec_ctx->time_base.den,
+                dec_ctx->sample_aspect_ratio.num,
+                dec_ctx->sample_aspect_ratio.den);
+
+        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
+                args, NULL, filter_graph);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
+            goto end;
+        }
+
+        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
+                NULL, NULL, filter_graph);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
+            goto end;
+        }
+
+        ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
+                (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
+                AV_OPT_SEARCH_CHILDREN);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
+            goto end;
+        }
+    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
+        buffersrc = avfilter_get_by_name("abuffer");
+        buffersink = avfilter_get_by_name("abuffersink");
+        if (!buffersrc || !buffersink) {
+            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        if (!dec_ctx->channel_layout)
+            dec_ctx->channel_layout =
+                av_get_default_channel_layout(dec_ctx->channels);
+        snprintf(args, sizeof(args),
+                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
+                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
+                av_get_sample_fmt_name(dec_ctx->sample_fmt),
+                dec_ctx->channel_layout);
+        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
+                args, NULL, filter_graph);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
+            goto end;
+        }
+
+        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
+                NULL, NULL, filter_graph);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
+            goto end;
+        }
+
+        ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
+                (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
+                AV_OPT_SEARCH_CHILDREN);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
+            goto end;
+        }
+
+        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
+                (uint8_t*)&enc_ctx->channel_layout,
+                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
+            goto end;
+        }
+
+        ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
+                (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
+                AV_OPT_SEARCH_CHILDREN);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
+            goto end;
+        }
+    } else {
+        ret = AVERROR_UNKNOWN;
+        goto end;
+    }
+
+    /* Endpoints for the filter graph. */
+    outputs->name       = av_strdup("in");
+    outputs->filter_ctx = buffersrc_ctx;
+    outputs->pad_idx    = 0;
+    outputs->next       = NULL;
+
+    inputs->name       = av_strdup("out");
+    inputs->filter_ctx = buffersink_ctx;
+    inputs->pad_idx    = 0;
+    inputs->next       = NULL;
+
+    if (!outputs->name || !inputs->name) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
+                    &inputs, &outputs, NULL)) < 0)
+        goto end;
+
+    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
+        goto end;
+
+    /* Fill FilteringContext */
+    fctx->buffersrc_ctx = buffersrc_ctx;
+    fctx->buffersink_ctx = buffersink_ctx;
+    fctx->filter_graph = filter_graph;
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
+}
+
+static int init_filters(void)
+{
+    const char *filter_spec;
+    unsigned int i;
+    int ret;
+    filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
+    if (!filter_ctx)
+        return AVERROR(ENOMEM);
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        filter_ctx[i].buffersrc_ctx  = NULL;
+        filter_ctx[i].buffersink_ctx = NULL;
+        filter_ctx[i].filter_graph   = NULL;
+        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
+                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
+            continue;
+
+
+        if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
+            filter_spec = "null"; /* passthrough (dummy) filter for video */
+        else
+            filter_spec = "anull"; /* passthrough (dummy) filter for audio */
+        ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
+                ofmt_ctx->streams[i]->codec, filter_spec);
+        if (ret)
+            return ret;
+    }
+    return 0;
+}
+
+static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
+    int ret;
+    int got_frame_local;
+    AVPacket enc_pkt;
+    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
+        (ifmt_ctx->streams[stream_index]->codec->codec_type ==
+         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
+
+    if (!got_frame)
+        got_frame = &got_frame_local;
+
+    av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
+    /* encode filtered frame */
+    enc_pkt.data = NULL;
+    enc_pkt.size = 0;
+    av_init_packet(&enc_pkt);
+    ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
+            filt_frame, got_frame);
+    av_frame_free(&filt_frame);
+    if (ret < 0)
+        return ret;
+    if (!(*got_frame))
+        return 0;
+
+    /* prepare packet for muxing */
+    enc_pkt.stream_index = stream_index;
+    av_packet_rescale_ts(&enc_pkt,
+                         ofmt_ctx->streams[stream_index]->codec->time_base,
+                         ofmt_ctx->streams[stream_index]->time_base);
+
+    av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
+    /* mux encoded frame */
+    ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
+    return ret;
+}
+
+static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
+{
+    int ret;
+    AVFrame *filt_frame;
+
+    av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
+    /* push the decoded frame into the filtergraph */
+    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
+            frame, 0);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
+        return ret;
+    }
+
+    /* pull filtered frames from the filtergraph */
+    while (1) {
+        filt_frame = av_frame_alloc();
+        if (!filt_frame) {
+            ret = AVERROR(ENOMEM);
+            break;
+        }
+        av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
+        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
+                filt_frame);
+        if (ret < 0) {
+            /* if no more frames for output - returns AVERROR(EAGAIN)
+             * if flushed and no more frames for output - returns AVERROR_EOF
+             * rewrite retcode to 0 to show it as normal procedure completion
+             */
+            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                ret = 0;
+            av_frame_free(&filt_frame);
+            break;
+        }
+
+        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
+        ret = encode_write_frame(filt_frame, stream_index, NULL);
+        if (ret < 0)
+            break;
+    }
+
+    return ret;
+}
+
+static int flush_encoder(unsigned int stream_index)
+{
+    int ret;
+    int got_frame;
+
+    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
+                AV_CODEC_CAP_DELAY))
+        return 0;
+
+    while (1) {
+        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
+        ret = encode_write_frame(NULL, stream_index, &got_frame);
+        if (ret < 0)
+            break;
+        if (!got_frame)
+            return 0;
+    }
+    return ret;
+}
+
+int main(int argc, char **argv)
+{
+    int ret;
+    AVPacket packet = { .data = NULL, .size = 0 };
+    AVFrame *frame = NULL;
+    enum AVMediaType type;
+    unsigned int stream_index;
+    unsigned int i;
+    int got_frame;
+    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
+
+    if (argc != 3) {
+        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
+        return 1;
+    }
+
+    av_register_all();
+    avfilter_register_all();
+
+    if ((ret = open_input_file(argv[1])) < 0)
+        goto end;
+    if ((ret = open_output_file(argv[2])) < 0)
+        goto end;
+    if ((ret = init_filters()) < 0)
+        goto end;
+
+    /* read all packets */
+    while (1) {
+        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
+            break;
+        stream_index = packet.stream_index;
+        type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
+        av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
+                stream_index);
+
+        if (filter_ctx[stream_index].filter_graph) {
+            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
+            frame = av_frame_alloc();
+            if (!frame) {
+                ret = AVERROR(ENOMEM);
+                break;
+            }
+            av_packet_rescale_ts(&packet,
+                                 ifmt_ctx->streams[stream_index]->time_base,
+                                 ifmt_ctx->streams[stream_index]->codec->time_base);
+            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
+                avcodec_decode_audio4;
+            ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
+                    &got_frame, &packet);
+            if (ret < 0) {
+                av_frame_free(&frame);
+                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
+                break;
+            }
+
+            if (got_frame) {
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
+                ret = filter_encode_write_frame(frame, stream_index);
+                av_frame_free(&frame);
+                if (ret < 0)
+                    goto end;
+            } else {
+                av_frame_free(&frame);
+            }
+        } else {
+            /* remux this frame without reencoding */
+            av_packet_rescale_ts(&packet,
+                                 ifmt_ctx->streams[stream_index]->time_base,
+                                 ofmt_ctx->streams[stream_index]->time_base);
+
+            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
+            if (ret < 0)
+                goto end;
+        }
+        av_packet_unref(&packet);
+    }
+
+    /* flush filters and encoders */
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        /* flush filter */
+        if (!filter_ctx[i].filter_graph)
+            continue;
+        ret = filter_encode_write_frame(NULL, i);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
+            goto end;
+        }
+
+        /* flush encoder */
+        ret = flush_encoder(i);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
+            goto end;
+        }
+    }
+
+    av_write_trailer(ofmt_ctx);
+end:
+    av_packet_unref(&packet);
+    av_frame_free(&frame);
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        avcodec_close(ifmt_ctx->streams[i]->codec);
+        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
+            avcodec_close(ofmt_ctx->streams[i]->codec);
+        if (filter_ctx && filter_ctx[i].filter_graph)
+            avfilter_graph_free(&filter_ctx[i].filter_graph);
+    }
+    av_free(filter_ctx);
+    avformat_close_input(&ifmt_ctx);
+    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
+        avio_closep(&ofmt_ctx->pb);
+    avformat_free_context(ofmt_ctx);
+
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
+
+    return ret ? 1 : 0;
+}
@@ -0,0 +1,756 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      FFmpeg FAQ
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      FFmpeg FAQ
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-General-Questions" href="#General-Questions">1 General Questions</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" href="#Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f">1.1 Why doesn&rsquo;t FFmpeg support feature [xyz]?</a></li>
+    <li><a name="toc-FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" href="#FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f">1.2 FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?</a></li>
+    <li><a name="toc-I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" href="#I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e">1.3 I cannot read this file although this format seems to be supported by ffmpeg.</a></li>
+    <li><a name="toc-Which-codecs-are-supported-by-Windows_003f" href="#Which-codecs-are-supported-by-Windows_003f">1.4 Which codecs are supported by Windows?</a></li>
+  </ul></li>
+  <li><a name="toc-Compilation" href="#Compilation">2 Compilation</a>
+  <ul class="no-bullet">
+    <li><a name="toc-error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" href="#error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027">2.1 <code>error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'</code></a></li>
+    <li><a name="toc-I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" href="#I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f">2.2 I have installed this library with my distro&rsquo;s package manager. Why does <code>configure</code> not see it?</a></li>
+    <li><a name="toc-How-do-I-make-pkg_002dconfig-find-my-libraries_003f" href="#How-do-I-make-pkg_002dconfig-find-my-libraries_003f">2.3 How do I make <code>pkg-config</code> find my libraries?</a></li>
+    <li><a name="toc-How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" href="#How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f">2.4 How do I use <code>pkg-config</code> when cross-compiling?</a></li>
+  </ul></li>
+  <li><a name="toc-Usage" href="#Usage">3 Usage</a>
+  <ul class="no-bullet">
+    <li><a name="toc-ffmpeg-does-not-work_003b-what-is-wrong_003f" href="#ffmpeg-does-not-work_003b-what-is-wrong_003f">3.1 ffmpeg does not work; what is wrong?</a></li>
+    <li><a name="toc-How-do-I-encode-single-pictures-into-movies_003f" href="#How-do-I-encode-single-pictures-into-movies_003f">3.2 How do I encode single pictures into movies?</a></li>
+    <li><a name="toc-How-do-I-encode-movie-to-single-pictures_003f" href="#How-do-I-encode-movie-to-single-pictures_003f">3.3 How do I encode movie to single pictures?</a></li>
+    <li><a name="toc-Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" href="#Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f">3.4 Why do I see a slight quality degradation with multithreaded MPEG* encoding?</a></li>
+    <li><a name="toc-How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" href="#How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f">3.5 How can I read from the standard input or write to the standard output?</a></li>
+    <li><a name="toc-_002df-jpeg-doesn_0027t-work_002e" href="#g_t_002df-jpeg-doesn_0027t-work_002e">3.6 -f jpeg doesn&rsquo;t work.</a></li>
+    <li><a name="toc-Why-can-I-not-change-the-frame-rate_003f" href="#Why-can-I-not-change-the-frame-rate_003f">3.7 Why can I not change the frame rate?</a></li>
+    <li><a name="toc-How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" href="#How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f">3.8 How do I encode Xvid or DivX video with ffmpeg?</a></li>
+    <li><a name="toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f">3.9 Which are good parameters for encoding high quality MPEG-4?</a></li>
+    <li><a name="toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f">3.10 Which are good parameters for encoding high quality MPEG-1/MPEG-2?</a></li>
+    <li><a name="toc-Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" href="#Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f">3.11 Interlaced video looks very bad when encoded with ffmpeg, what is wrong?</a></li>
+    <li><a name="toc-How-can-I-read-DirectShow-files_003f" href="#How-can-I-read-DirectShow-files_003f">3.12 How can I read DirectShow files?</a></li>
+    <li><a name="toc-How-can-I-join-video-files_003f" href="#How-can-I-join-video-files_003f">3.13 How can I join video files?</a></li>
+    <li><a name="toc-How-can-I-concatenate-video-files_003f" href="#How-can-I-concatenate-video-files_003f">3.14 How can I concatenate video files?</a>
+    <ul class="no-bullet">
+      <li><a name="toc-Concatenating-using-the-concat-filter" href="#Concatenating-using-the-concat-filter">3.14.1 Concatenating using the concat <em>filter</em></a></li>
+      <li><a name="toc-Concatenating-using-the-concat-demuxer" href="#Concatenating-using-the-concat-demuxer">3.14.2 Concatenating using the concat <em>demuxer</em></a></li>
+      <li><a name="toc-Concatenating-using-the-concat-protocol-_0028file-level_0029" href="#Concatenating-using-the-concat-protocol-_0028file-level_0029">3.14.3 Concatenating using the concat <em>protocol</em> (file level)</a></li>
+      <li><a name="toc-Concatenating-using-raw-audio-and-video" href="#Concatenating-using-raw-audio-and-video">3.14.4 Concatenating using raw audio and video</a></li>
+    </ul></li>
+    <li><a name="toc-Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" href="#Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e">3.15 Using <samp>-f lavfi</samp>, audio becomes mono for no apparent reason.</a></li>
+    <li><a name="toc-Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" href="#Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f">3.16 Why does FFmpeg not see the subtitles in my VOB file?</a></li>
+    <li><a name="toc-Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" href="#Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f">3.17 Why was the <code>ffmpeg</code> <samp>-sameq</samp> option removed? What to use instead?</a></li>
+    <li><a name="toc-I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f" href="#I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f">3.18 I have a stretched video, why does scaling does not fix it?</a></li>
+  </ul></li>
+  <li><a name="toc-Development" href="#Development">4 Development</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" href="#Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f">4.1 Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?</a></li>
+    <li><a name="toc-Can-you-support-my-C-compiler-XXX_003f" href="#Can-you-support-my-C-compiler-XXX_003f">4.2 Can you support my C compiler XXX?</a></li>
+    <li><a name="toc-Is-Microsoft-Visual-C_002b_002b-supported_003f" href="#Is-Microsoft-Visual-C_002b_002b-supported_003f">4.3 Is Microsoft Visual C++ supported?</a></li>
+    <li><a name="toc-Can-you-add-automake_002c-libtool-or-autoconf-support_003f" href="#Can-you-add-automake_002c-libtool-or-autoconf-support_003f">4.4 Can you add automake, libtool or autoconf support?</a></li>
+    <li><a name="toc-Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" href="#Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f">4.5 Why not rewrite FFmpeg in object-oriented C++?</a></li>
+    <li><a name="toc-Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" href="#Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f">4.6 Why are the ffmpeg programs devoid of debugging symbols?</a></li>
+    <li><a name="toc-I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" href="#I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f">4.7 I do not like the LGPL, can I contribute code under the GPL instead?</a></li>
+    <li><a name="toc-I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" href="#I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e">4.8 I&rsquo;m using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.</a></li>
+    <li><a name="toc-I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" href="#I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e">4.9 I&rsquo;m using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.</a></li>
+    <li><a name="toc-I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" href="#I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope">4.10 I&rsquo;m using libavutil from within my C++ application but the compiler complains about &rsquo;UINT64_C&rsquo; was not declared in this scope</a></li>
+    <li><a name="toc-I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" href="#I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f">4.11 I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?</a></li>
+    <li><a name="toc-Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" href="#Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f">4.12 Where is the documentation about ffv1, msmpeg4, asv1, 4xm?</a></li>
+    <li><a name="toc-How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" href="#How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f">4.13 How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?</a></li>
+    <li><a name="toc-AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" href="#AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e">4.14 AVStream.r_frame_rate is wrong, it is much larger than the frame rate.</a></li>
+    <li><a name="toc-Why-is-make-fate-not-running-all-tests_003f" href="#Why-is-make-fate-not-running-all-tests_003f">4.15 Why is <code>make fate</code> not running all tests?</a></li>
+    <li><a name="toc-Why-is-make-fate-not-finding-the-samples_003f" href="#Why-is-make-fate-not-finding-the-samples_003f">4.16 Why is <code>make fate</code> not finding the samples?</a></li>
+  </ul></li>
+</ul>
+</div>
+
+
+<a name="General-Questions"></a>
+<h2 class="chapter">1 General Questions<span class="pull-right"><a class="anchor hidden-xs" href="#General-Questions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-General-Questions" aria-hidden="true">TOC</a></span></h2>
+
+<a name="Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f"></a>
+<h3 class="section">1.1 Why doesn&rsquo;t FFmpeg support feature [xyz]?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Because no one has taken on that task yet. FFmpeg development is
+driven by the tasks that are important to the individual developers.
+If there is a feature that is important to you, the best way to get
+it implemented is to undertake the task yourself or sponsor a developer.
+</p>
+<a name="FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f"></a>
+<h3 class="section">1.2 FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?<span class="pull-right"><a class="anchor hidden-xs" href="#FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>No. Windows DLLs are not portable, bloated and often slow.
+Moreover FFmpeg strives to support all codecs natively.
+A DLL loader is not conducive to that goal.
+</p>
+<a name="I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e"></a>
+<h3 class="section">1.3 I cannot read this file although this format seems to be supported by ffmpeg.<span class="pull-right"><a class="anchor hidden-xs" href="#I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" aria-hidden="true">TOC</a></span></h3>
+
+<p>Even if ffmpeg can read the container format, it may not support all its
+codecs. Please consult the supported codec list in the ffmpeg
+documentation.
+</p>
+<a name="Which-codecs-are-supported-by-Windows_003f"></a>
+<h3 class="section">1.4 Which codecs are supported by Windows?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-codecs-are-supported-by-Windows_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-codecs-are-supported-by-Windows_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Windows does not support standard formats like MPEG very well, unless you
+install some additional codecs.
+</p>
+<p>The following list of video codecs should work on most Windows systems:
+</p><dl compact="compact">
+<dt><samp>msmpeg4v2</samp></dt>
+<dd><p>.avi/.asf
+</p></dd>
+<dt><samp>msmpeg4</samp></dt>
+<dd><p>.asf only
+</p></dd>
+<dt><samp>wmv1</samp></dt>
+<dd><p>.asf only
+</p></dd>
+<dt><samp>wmv2</samp></dt>
+<dd><p>.asf only
+</p></dd>
+<dt><samp>mpeg4</samp></dt>
+<dd><p>Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
+</p></dd>
+<dt><samp>mpeg1video</samp></dt>
+<dd><p>.mpg only
+</p></dd>
+</dl>
+<p>Note, ASF files often have .wmv or .wma extensions in Windows. It should also
+be mentioned that Microsoft claims a patent on the ASF format, and may sue
+or threaten users who create ASF files with non-Microsoft software. It is
+strongly advised to avoid ASF where possible.
+</p>
+<p>The following list of audio codecs should work on most Windows systems:
+</p><dl compact="compact">
+<dt><samp>adpcm_ima_wav</samp></dt>
+<dt><samp>adpcm_ms</samp></dt>
+<dt><samp>pcm_s16le</samp></dt>
+<dd><p>always
+</p></dd>
+<dt><samp>libmp3lame</samp></dt>
+<dd><p>If some MP3 codec like LAME is installed.
+</p></dd>
+</dl>
+
+
+<a name="Compilation"></a>
+<h2 class="chapter">2 Compilation<span class="pull-right"><a class="anchor hidden-xs" href="#Compilation" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Compilation" aria-hidden="true">TOC</a></span></h2>
+
+<a name="error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027"></a>
+<h3 class="section">2.1 <code>error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'</code><span class="pull-right"><a class="anchor hidden-xs" href="#error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" aria-hidden="true">TOC</a></span></h3>
+
+<p>This is a bug in gcc. Do not report it to us. Instead, please report it to
+the gcc developers. Note that we will not add workarounds for gcc bugs.
+</p>
+<p>Also note that (some of) the gcc developers believe this is not a bug or
+not a bug they should fix:
+<a href="http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203">http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203</a>.
+Then again, some of them do not know the difference between an undecidable
+problem and an NP-hard problem...
+</p>
+<a name="I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f"></a>
+<h3 class="section">2.2 I have installed this library with my distro&rsquo;s package manager. Why does <code>configure</code> not see it?<span class="pull-right"><a class="anchor hidden-xs" href="#I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Distributions usually split libraries in several packages. The main package
+contains the files necessary to run programs using the library. The
+development package contains the files necessary to build programs using the
+library. Sometimes, docs and/or data are in a separate package too.
+</p>
+<p>To build FFmpeg, you need to install the development package. It is usually
+called <samp>libfoo-dev</samp> or <samp>libfoo-devel</samp>. You can remove it after the
+build is finished, but be sure to keep the main package.
+</p>
+<a name="How-do-I-make-pkg_002dconfig-find-my-libraries_003f"></a>
+<h3 class="section">2.3 How do I make <code>pkg-config</code> find my libraries?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-make-pkg_002dconfig-find-my-libraries_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-make-pkg_002dconfig-find-my-libraries_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Somewhere along with your libraries, there is a <samp>.pc</samp> file (or several)
+in a <samp>pkgconfig</samp> directory. You need to set environment variables to
+point <code>pkg-config</code> to these files.
+</p>
+<p>If you need to <em>add</em> directories to <code>pkg-config</code>&rsquo;s search list
+(typical use case: library installed separately), add it to
+<code>$PKG_CONFIG_PATH</code>:
+</p>
+<div class="example">
+<pre class="example">export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
+</pre></div>
+
+<p>If you need to <em>replace</em> <code>pkg-config</code>&rsquo;s search list
+(typical use case: cross-compiling), set it in
+<code>$PKG_CONFIG_LIBDIR</code>:
+</p>
+<div class="example">
+<pre class="example">export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
+</pre></div>
+
+<p>If you need to know the library&rsquo;s internal dependencies (typical use: static
+linking), add the <code>--static</code> option to <code>pkg-config</code>:
+</p>
+<div class="example">
+<pre class="example">./configure --pkg-config-flags=--static
+</pre></div>
+
+<a name="How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f"></a>
+<h3 class="section">2.4 How do I use <code>pkg-config</code> when cross-compiling?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>The best way is to install <code>pkg-config</code> in your cross-compilation
+environment. It will automatically use the cross-compilation libraries.
+</p>
+<p>You can also use <code>pkg-config</code> from the host environment by
+specifying explicitly <code>--pkg-config=pkg-config</code> to <code>configure</code>.
+In that case, you must point <code>pkg-config</code> to the correct directories
+using the <code>PKG_CONFIG_LIBDIR</code>, as explained in the previous entry.
+</p>
+<p>As an intermediate solution, you can place in your cross-compilation
+environment a script that calls the host <code>pkg-config</code> with
+<code>PKG_CONFIG_LIBDIR</code> set. That script can look like that:
+</p>
+<div class="example">
+<pre class="example">#!/bin/sh
+PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
+export PKG_CONFIG_LIBDIR
+exec /usr/bin/pkg-config &quot;$@&quot;
+</pre></div>
+
+<a name="Usage"></a>
+<h2 class="chapter">3 Usage<span class="pull-right"><a class="anchor hidden-xs" href="#Usage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Usage" aria-hidden="true">TOC</a></span></h2>
+
+<a name="ffmpeg-does-not-work_003b-what-is-wrong_003f"></a>
+<h3 class="section">3.1 ffmpeg does not work; what is wrong?<span class="pull-right"><a class="anchor hidden-xs" href="#ffmpeg-does-not-work_003b-what-is-wrong_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-ffmpeg-does-not-work_003b-what-is-wrong_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Try a <code>make distclean</code> in the ffmpeg source directory before the build.
+If this does not help see
+(<a href="https://ffmpeg.org/bugreports.html">https://ffmpeg.org/bugreports.html</a>).
+</p>
+<a name="How-do-I-encode-single-pictures-into-movies_003f"></a>
+<h3 class="section">3.2 How do I encode single pictures into movies?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-single-pictures-into-movies_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-single-pictures-into-movies_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>First, rename your pictures to follow a numerical sequence.
+For example, img1.jpg, img2.jpg, img3.jpg,...
+Then you may run:
+</p>
+<div class="example">
+<pre class="example">ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
+</pre></div>
+
+<p>Notice that &lsquo;<samp>%d</samp>&rsquo; is replaced by the image number.
+</p>
+<p><samp>img%03d.jpg</samp> means the sequence <samp>img001.jpg</samp>, <samp>img002.jpg</samp>, etc.
+</p>
+<p>Use the <samp>-start_number</samp> option to declare a starting number for
+the sequence. This is useful if your sequence does not start with
+<samp>img001.jpg</samp> but is still in a numerical order. The following
+example will start with <samp>img100.jpg</samp>:
+</p>
+<div class="example">
+<pre class="example">ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
+</pre></div>
+
+<p>If you have large number of pictures to rename, you can use the
+following command to ease the burden. The command, using the bourne
+shell syntax, symbolically links all files in the current directory
+that match <code>*jpg</code> to the <samp>/tmp</samp> directory in the sequence of
+<samp>img001.jpg</samp>, <samp>img002.jpg</samp> and so on.
+</p>
+<div class="example">
+<pre class="example">x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s &quot;$i&quot; /tmp/img&quot;$counter&quot;.jpg; x=$(($x+1)); done
+</pre></div>
+
+<p>If you want to sequence them by oldest modified first, substitute
+<code>$(ls -r -t *jpg)</code> in place of <code>*jpg</code>.
+</p>
+<p>Then run:
+</p>
+<div class="example">
+<pre class="example">ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
+</pre></div>
+
+<p>The same logic is used for any image format that ffmpeg reads.
+</p>
+<p>You can also use <code>cat</code> to pipe images to ffmpeg:
+</p>
+<div class="example">
+<pre class="example">cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
+</pre></div>
+
+<a name="How-do-I-encode-movie-to-single-pictures_003f"></a>
+<h3 class="section">3.3 How do I encode movie to single pictures?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-movie-to-single-pictures_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-movie-to-single-pictures_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Use:
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i movie.mpg movie%d.jpg
+</pre></div>
+
+<p>The <samp>movie.mpg</samp> used as input will be converted to
+<samp>movie1.jpg</samp>, <samp>movie2.jpg</samp>, etc...
+</p>
+<p>Instead of relying on file format self-recognition, you may also use
+</p><dl compact="compact">
+<dt><samp>-c:v ppm</samp></dt>
+<dt><samp>-c:v png</samp></dt>
+<dt><samp>-c:v mjpeg</samp></dt>
+</dl>
+<p>to force the encoding.
+</p>
+<p>Applying that to the previous example:
+</p><div class="example">
+<pre class="example">ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
+</pre></div>
+
+<p>Beware that there is no &quot;jpeg&quot; codec. Use &quot;mjpeg&quot; instead.
+</p>
+<a name="Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f"></a>
+<h3 class="section">3.4 Why do I see a slight quality degradation with multithreaded MPEG* encoding?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>For multithreaded MPEG* encoding, the encoded slices must be independent,
+otherwise thread n would practically have to wait for n-1 to finish, so it&rsquo;s
+quite logical that there is a small reduction of quality. This is not a bug.
+</p>
+<a name="How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f"></a>
+<h3 class="section">3.5 How can I read from the standard input or write to the standard output?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Use <samp>-</samp> as file name.
+</p>
+<a name="g_t_002df-jpeg-doesn_0027t-work_002e"></a>
+<h3 class="section">3.6 -f jpeg doesn&rsquo;t work.<span class="pull-right"><a class="anchor hidden-xs" href="#_002df-jpeg-doesn_0027t-work_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-_002df-jpeg-doesn_0027t-work_002e" aria-hidden="true">TOC</a></span></h3>
+
+<p>Try &rsquo;-f image2 test%d.jpg&rsquo;.
+</p>
+<a name="Why-can-I-not-change-the-frame-rate_003f"></a>
+<h3 class="section">3.7 Why can I not change the frame rate?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-can-I-not-change-the-frame-rate_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-can-I-not-change-the-frame-rate_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
+Choose a different codec with the -c:v command line option.
+</p>
+<a name="How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f"></a>
+<h3 class="section">3.8 How do I encode Xvid or DivX video with ffmpeg?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
+standard (note that there are many other coding formats that use this
+same standard). Thus, use &rsquo;-c:v mpeg4&rsquo; to encode in these formats. The
+default fourcc stored in an MPEG-4-coded file will be &rsquo;FMP4&rsquo;. If you want
+a different fourcc, use the &rsquo;-vtag&rsquo; option. E.g., &rsquo;-vtag xvid&rsquo; will
+force the fourcc &rsquo;xvid&rsquo; to be stored as the video fourcc rather than the
+default.
+</p>
+<a name="Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f"></a>
+<h3 class="section">3.9 Which are good parameters for encoding high quality MPEG-4?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>&rsquo;-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2&rsquo;,
+things to try: &rsquo;-bf 2&rsquo;, &rsquo;-flags qprd&rsquo;, &rsquo;-flags mv0&rsquo;, &rsquo;-flags skiprd&rsquo;.
+</p>
+<a name="Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f"></a>
+<h3 class="section">3.10 Which are good parameters for encoding high quality MPEG-1/MPEG-2?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>&rsquo;-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2&rsquo;
+but beware the &rsquo;-g 100&rsquo; might cause problems with some decoders.
+Things to try: &rsquo;-bf 2&rsquo;, &rsquo;-flags qprd&rsquo;, &rsquo;-flags mv0&rsquo;, &rsquo;-flags skiprd.
+</p>
+<a name="Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f"></a>
+<h3 class="section">3.11 Interlaced video looks very bad when encoded with ffmpeg, what is wrong?<span class="pull-right"><a class="anchor hidden-xs" href="#Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>You should use &rsquo;-flags +ilme+ildct&rsquo; and maybe &rsquo;-flags +alt&rsquo; for interlaced
+material, and try &rsquo;-top 0/1&rsquo; if the result looks really messed-up.
+</p>
+<a name="How-can-I-read-DirectShow-files_003f"></a>
+<h3 class="section">3.12 How can I read DirectShow files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-read-DirectShow-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-read-DirectShow-files_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>If you have built FFmpeg with <code>./configure --enable-avisynth</code>
+(only possible on MinGW/Cygwin platforms),
+then you may use any file that DirectShow can read as input.
+</p>
+<p>Just create an &quot;input.avs&quot; text file with this single line ...
+</p><div class="example">
+<pre class="example">DirectShowSource(&quot;C:\path to your file\yourfile.asf&quot;)
+</pre></div>
+<p>... and then feed that text file to ffmpeg:
+</p><div class="example">
+<pre class="example">ffmpeg -i input.avs
+</pre></div>
+
+<p>For ANY other help on AviSynth, please visit the
+<a href="http://www.avisynth.org/">AviSynth homepage</a>.
+</p>
+<a name="How-can-I-join-video-files_003f"></a>
+<h3 class="section">3.13 How can I join video files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-join-video-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-join-video-files_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>To &quot;join&quot; video files is quite ambiguous. The following list explains the
+different kinds of &quot;joining&quot; and points out how those are addressed in
+FFmpeg. To join video files may mean:
+</p>
+<ul>
+<li> To put them one after the other: this is called to <em>concatenate</em> them
+(in short: concat) and is addressed
+<a href="#How-can-I-concatenate-video-files">in this very faq</a>.
+
+</li><li> To put them together in the same file, to let the user choose between the
+different versions (example: different audio languages): this is called to
+<em>multiplex</em> them together (in short: mux), and is done by simply
+invoking ffmpeg with several <samp>-i</samp> options.
+
+</li><li> For audio, to put all channels together in a single stream (example: two
+mono streams into one stereo stream): this is sometimes called to
+<em>merge</em> them, and can be done using the
+<a href="https://ffmpeg.org/ffmpeg-filters.html#amerge"><code>amerge</code></a> filter.
+
+</li><li> For audio, to play one on top of the other: this is called to <em>mix</em>
+them, and can be done by first merging them into a single stream and then
+using the <a href="https://ffmpeg.org/ffmpeg-filters.html#pan"><code>pan</code></a> filter to mix
+the channels at will.
+
+</li><li> For video, to display both together, side by side or one on top of a part of
+the other; it can be done using the
+<a href="https://ffmpeg.org/ffmpeg-filters.html#overlay"><code>overlay</code></a> video filter.
+
+</li></ul>
+
+<a name="How-can-I-concatenate-video-files"></a><a name="How-can-I-concatenate-video-files_003f"></a>
+<h3 class="section">3.14 How can I concatenate video files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-concatenate-video-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-concatenate-video-files_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>There are several solutions, depending on the exact circumstances.
+</p>
+<a name="Concatenating-using-the-concat-filter"></a>
+<h4 class="subsection">3.14.1 Concatenating using the concat <em>filter</em><span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-filter" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-filter" aria-hidden="true">TOC</a></span></h4>
+
+<p>FFmpeg has a <a href="https://ffmpeg.org/ffmpeg-filters.html#concat"><code>concat</code></a> filter designed specifically for that, with examples in the
+documentation. This operation is recommended if you need to re-encode.
+</p>
+<a name="Concatenating-using-the-concat-demuxer"></a>
+<h4 class="subsection">3.14.2 Concatenating using the concat <em>demuxer</em><span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-demuxer" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-demuxer" aria-hidden="true">TOC</a></span></h4>
+
+<p>FFmpeg has a <a href="https://www.ffmpeg.org/ffmpeg-formats.html#concat"><code>concat</code></a> demuxer which you can use when you want to avoid a re-encode and
+your format doesn&rsquo;t support file level concatenation.
+</p>
+<a name="Concatenating-using-the-concat-protocol-_0028file-level_0029"></a>
+<h4 class="subsection">3.14.3 Concatenating using the concat <em>protocol</em> (file level)<span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-protocol-_0028file-level_0029" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-protocol-_0028file-level_0029" aria-hidden="true">TOC</a></span></h4>
+
+<p>FFmpeg has a <a href="https://ffmpeg.org/ffmpeg-protocols.html#concat"><code>concat</code></a> protocol designed specifically for that, with examples in the
+documentation.
+</p>
+<p>A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
+video by merely concatenating the files containing them.
+</p>
+<p>Hence you may concatenate your multimedia files by first transcoding them to
+these privileged formats, then using the humble <code>cat</code> command (or the
+equally humble <code>copy</code> under Windows), and finally transcoding back to your
+format of choice.
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
+ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
+cat intermediate1.mpg intermediate2.mpg &gt; intermediate_all.mpg
+ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
+</pre></div>
+
+<p>Additionally, you can use the <code>concat</code> protocol instead of <code>cat</code> or
+<code>copy</code> which will avoid creation of a potentially huge intermediate file.
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
+ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
+ffmpeg -i concat:&quot;intermediate1.mpg|intermediate2.mpg&quot; -c copy intermediate_all.mpg
+ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
+</pre></div>
+
+<p>Note that you may need to escape the character &quot;|&quot; which is special for many
+shells.
+</p>
+<p>Another option is usage of named pipes, should your platform support it:
+</p>
+<div class="example">
+<pre class="example">mkfifo intermediate1.mpg
+mkfifo intermediate2.mpg
+ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg &lt; /dev/null &amp;
+ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg &lt; /dev/null &amp;
+cat intermediate1.mpg intermediate2.mpg |\
+ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
+</pre></div>
+
+<a name="Concatenating-using-raw-audio-and-video"></a>
+<h4 class="subsection">3.14.4 Concatenating using raw audio and video<span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-raw-audio-and-video" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-raw-audio-and-video" aria-hidden="true">TOC</a></span></h4>
+
+<p>Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
+allow concatenation, and the transcoding step is almost lossless.
+When using multiple yuv4mpegpipe(s), the first line needs to be discarded
+from all but the first stream. This can be accomplished by piping through
+<code>tail</code> as seen below. Note that when piping through <code>tail</code> you
+must use command grouping, <code>{  ;}</code>, to background properly.
+</p>
+<p>For example, let&rsquo;s say we want to concatenate two FLV files into an
+output.flv file:
+</p>
+<div class="example">
+<pre class="example">mkfifo temp1.a
+mkfifo temp1.v
+mkfifo temp2.a
+mkfifo temp2.v
+mkfifo all.a
+mkfifo all.v
+ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - &gt; temp1.a &lt; /dev/null &amp;
+ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - &gt; temp2.a &lt; /dev/null &amp;
+ffmpeg -i input1.flv -an -f yuv4mpegpipe - &gt; temp1.v &lt; /dev/null &amp;
+{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - &lt; /dev/null | tail -n +2 &gt; temp2.v ; } &amp;
+cat temp1.a temp2.a &gt; all.a &amp;
+cat temp1.v temp2.v &gt; all.v &amp;
+ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
+       -f yuv4mpegpipe -i all.v \
+       -y output.flv
+rm temp[12].[av] all.[av]
+</pre></div>
+
+<a name="Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e"></a>
+<h3 class="section">3.15 Using <samp>-f lavfi</samp>, audio becomes mono for no apparent reason.<span class="pull-right"><a class="anchor hidden-xs" href="#Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" aria-hidden="true">TOC</a></span></h3>
+
+<p>Use <samp>-dumpgraph -</samp> to find out exactly where the channel layout is
+lost.
+</p>
+<p>Most likely, it is through <code>auto-inserted aresample</code>. Try to understand
+why the converting filter was needed at that place.
+</p>
+<p>Just before the output is a likely place, as <samp>-f lavfi</samp> currently
+only support packed S16.
+</p>
+<p>Then insert the correct <code>aformat</code> explicitly in the filtergraph,
+specifying the exact format.
+</p>
+<div class="example">
+<pre class="example">aformat=sample_fmts=s16:channel_layouts=stereo
+</pre></div>
+
+<a name="Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f"></a>
+<h3 class="section">3.16 Why does FFmpeg not see the subtitles in my VOB file?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>VOB and a few other formats do not have a global header that describes
+everything present in the file. Instead, applications are supposed to scan
+the file to see what it contains. Since VOB files are frequently large, only
+the beginning is scanned. If the subtitles happen only later in the file,
+they will not be initially detected.
+</p>
+<p>Some applications, including the <code>ffmpeg</code> command-line tool, can only
+work with streams that were detected during the initial scan; streams that
+are detected later are ignored.
+</p>
+<p>The size of the initial scan is controlled by two options: <code>probesize</code>
+(default ~5 Mo) and <code>analyzeduration</code> (default 5,000,000 µs = 5 s). For
+the subtitle stream to be detected, both values must be large enough.
+</p>
+<a name="Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f"></a>
+<h3 class="section">3.17 Why was the <code>ffmpeg</code> <samp>-sameq</samp> option removed? What to use instead?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>The <samp>-sameq</samp> option meant &quot;same quantizer&quot;, and made sense only in a
+very limited set of cases. Unfortunately, a lot of people mistook it for
+&quot;same quality&quot; and used it in places where it did not make sense: it had
+roughly the expected visible effect, but achieved it in a very inefficient
+way.
+</p>
+<p>Each encoder has its own set of options to set the quality-vs-size balance,
+use the options for the encoder you are using to set the quality level to a
+point acceptable for your tastes. The most common options to do that are
+<samp>-qscale</samp> and <samp>-qmax</samp>, but you should peruse the documentation
+of the encoder you chose.
+</p>
+<a name="I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f"></a>
+<h3 class="section">3.18 I have a stretched video, why does scaling does not fix it?<span class="pull-right"><a class="anchor hidden-xs" href="#I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-have-a-stretched-video_002c-why-does-scaling-does-not-fix-it_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>A lot of video codecs and formats can store the <em>aspect ratio</em> of the
+video: this is the ratio between the width and the height of either the full
+image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
+ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
+SAR.
+</p>
+<p>Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
+of video standards, especially from the analogic-numeric transition era, use
+non-square pixels.
+</p>
+<p>Most processing filters in FFmpeg handle the aspect ratio to avoid
+stretching the image: cropping adjusts the DAR to keep the SAR constant,
+scaling adjusts the SAR to keep the DAR constant.
+</p>
+<p>If you want to stretch, or “unstretch”, the image, you need to override the
+information with the
+<a href="https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar"><code>setdar or setsar filters</code></a>.
+</p>
+<p>Do not forget to examine carefully the original video to check whether the
+stretching comes from the image or from the aspect ratio information.
+</p>
+<p>For example, to fix a badly encoded EGA capture, use the following commands,
+either the first one to upscale to square pixels or the second one to set
+the correct aspect ratio or the third one to avoid transcoding (may not work
+depending on the format / codec / player / phase of the moon):
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
+ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
+ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
+</pre></div>
+
+<a name="Development"></a>
+<h2 class="chapter">4 Development<span class="pull-right"><a class="anchor hidden-xs" href="#Development" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Development" aria-hidden="true">TOC</a></span></h2>
+
+<a name="Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f"></a>
+<h3 class="section">4.1 Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?<span class="pull-right"><a class="anchor hidden-xs" href="#Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Yes. Check the <samp>doc/examples</samp> directory in the source
+repository, also available online at:
+<a href="https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples">https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples</a>.
+</p>
+<p>Examples are also installed by default, usually in
+<code>$PREFIX/share/ffmpeg/examples</code>.
+</p>
+<p>Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
+examine the source code for one of the many open source projects that
+already incorporate FFmpeg at (<a href="projects.html">projects.html</a>).
+</p>
+<a name="Can-you-support-my-C-compiler-XXX_003f"></a>
+<h3 class="section">4.2 Can you support my C compiler XXX?<span class="pull-right"><a class="anchor hidden-xs" href="#Can-you-support-my-C-compiler-XXX_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Can-you-support-my-C-compiler-XXX_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>It depends. If your compiler is C99-compliant, then patches to support
+it are likely to be welcome if they do not pollute the source code
+with <code>#ifdef</code>s related to the compiler.
+</p>
+<a name="Is-Microsoft-Visual-C_002b_002b-supported_003f"></a>
+<h3 class="section">4.3 Is Microsoft Visual C++ supported?<span class="pull-right"><a class="anchor hidden-xs" href="#Is-Microsoft-Visual-C_002b_002b-supported_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Is-Microsoft-Visual-C_002b_002b-supported_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Yes. Please see the <a href="platform.html">Microsoft Visual C++</a>
+section in the FFmpeg documentation.
+</p>
+<a name="Can-you-add-automake_002c-libtool-or-autoconf-support_003f"></a>
+<h3 class="section">4.4 Can you add automake, libtool or autoconf support?<span class="pull-right"><a class="anchor hidden-xs" href="#Can-you-add-automake_002c-libtool-or-autoconf-support_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Can-you-add-automake_002c-libtool-or-autoconf-support_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>No. These tools are too bloated and they complicate the build.
+</p>
+<a name="Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f"></a>
+<h3 class="section">4.5 Why not rewrite FFmpeg in object-oriented C++?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>FFmpeg is already organized in a highly modular manner and does not need to
+be rewritten in a formal object language. Further, many of the developers
+favor straight C; it works for them. For more arguments on this matter,
+read <a href="http://www.tux.org/lkml/#s15">&quot;Programming Religion&quot;</a>.
+</p>
+<a name="Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f"></a>
+<h3 class="section">4.6 Why are the ffmpeg programs devoid of debugging symbols?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>The build process creates <code>ffmpeg_g</code>, <code>ffplay_g</code>, etc. which
+contain full debug information. Those binaries are stripped to create
+<code>ffmpeg</code>, <code>ffplay</code>, etc. If you need the debug information, use
+the *_g versions.
+</p>
+<a name="I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f"></a>
+<h3 class="section">4.7 I do not like the LGPL, can I contribute code under the GPL instead?<span class="pull-right"><a class="anchor hidden-xs" href="#I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Yes, as long as the code is optional and can easily and cleanly be placed
+under #if CONFIG_GPL without breaking anything. So, for example, a new codec
+or filter would be OK under GPL while a bug fix to LGPL code would not.
+</p>
+<a name="I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e"></a>
+<h3 class="section">4.8 I&rsquo;m using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" aria-hidden="true">TOC</a></span></h3>
+
+<p>FFmpeg builds static libraries by default. In static libraries, dependencies
+are not handled. That has two consequences. First, you must specify the
+libraries in dependency order: <code>-lavdevice</code> must come before
+<code>-lavformat</code>, <code>-lavutil</code> must come after everything else, etc.
+Second, external libraries that are used in FFmpeg have to be specified too.
+</p>
+<p>An easy way to get the full list of required libraries in dependency order
+is to use <code>pkg-config</code>.
+</p>
+<div class="example">
+<pre class="example">c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
+</pre></div>
+
+<p>See <samp>doc/example/Makefile</samp> and <samp>doc/example/pc-uninstalled</samp> for
+more details.
+</p>
+<a name="I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e"></a>
+<h3 class="section">4.9 I&rsquo;m using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" aria-hidden="true">TOC</a></span></h3>
+
+<p>FFmpeg is a pure C project, so to use the libraries within your C++ application
+you need to explicitly state that you are using a C library. You can do this by
+encompassing your FFmpeg includes using <code>extern &quot;C&quot;</code>.
+</p>
+<p>See <a href="http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3">http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3</a>
+</p>
+<a name="I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope"></a>
+<h3 class="section">4.10 I&rsquo;m using libavutil from within my C++ application but the compiler complains about &rsquo;UINT64_C&rsquo; was not declared in this scope<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" aria-hidden="true">TOC</a></span></h3>
+
+<p>FFmpeg is a pure C project using C99 math features, in order to enable C++
+to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
+</p>
+<a name="I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f"></a>
+<h3 class="section">4.11 I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?<span class="pull-right"><a class="anchor hidden-xs" href="#I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>You have to create a custom AVIOContext using <code>avio_alloc_context</code>,
+see <samp>libavformat/aviobuf.c</samp> in FFmpeg and <samp>libmpdemux/demux_lavf.c</samp> in MPlayer or MPlayer2 sources.
+</p>
+<a name="Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f"></a>
+<h3 class="section">4.12 Where is the documentation about ffv1, msmpeg4, asv1, 4xm?<span class="pull-right"><a class="anchor hidden-xs" href="#Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>see <a href="https://www.ffmpeg.org/~michael/">https://www.ffmpeg.org/~michael/</a>
+</p>
+<a name="How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f"></a>
+<h3 class="section">4.13 How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Even if peculiar since it is network oriented, RTP is a container like any
+other. You have to <em>demux</em> RTP before feeding the payload to libavcodec.
+In this specific case please look at RFC 4629 to see how it should be done.
+</p>
+<a name="AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e"></a>
+<h3 class="section">4.14 AVStream.r_frame_rate is wrong, it is much larger than the frame rate.<span class="pull-right"><a class="anchor hidden-xs" href="#AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" aria-hidden="true">TOC</a></span></h3>
+
+<p><code>r_frame_rate</code> is NOT the average frame rate, it is the smallest frame rate
+that can accurately represent all timestamps. So no, it is not
+wrong if it is larger than the average!
+For example, if you have mixed 25 and 30 fps content, then <code>r_frame_rate</code>
+will be 150 (it is the least common multiple).
+If you are looking for the average frame rate, see <code>AVStream.avg_frame_rate</code>.
+</p>
+<a name="Why-is-make-fate-not-running-all-tests_003f"></a>
+<h3 class="section">4.15 Why is <code>make fate</code> not running all tests?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-is-make-fate-not-running-all-tests_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-is-make-fate-not-running-all-tests_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Make sure you have the fate-suite samples and the <code>SAMPLES</code> Make variable
+or <code>FATE_SAMPLES</code> environment variable or the <code>--samples</code>
+<code>configure</code> option is set to the right path.
+</p>
+<a name="Why-is-make-fate-not-finding-the-samples_003f"></a>
+<h3 class="section">4.16 Why is <code>make fate</code> not finding the samples?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-is-make-fate-not-finding-the-samples_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-is-make-fate-not-finding-the-samples_003f" aria-hidden="true">TOC</a></span></h3>
+
+<p>Do you happen to have a <code>~</code> character in the samples path to indicate a
+home directory? The value is used in ways where the shell cannot expand it,
+causing FATE to not find files. Just replace <code>~</code> by the full path.
+</p>
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,288 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      FFmpeg Automated Testing Environment
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      FFmpeg Automated Testing Environment
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="Top"></a>
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Introduction" href="#Introduction">1 Introduction</a></li>
+  <li><a name="toc-Using-FATE-from-your-FFmpeg-source-directory" href="#Using-FATE-from-your-FFmpeg-source-directory">2 Using FATE from your FFmpeg source directory</a></li>
+  <li><a name="toc-Submitting-the-results-to-the-FFmpeg-result-aggregation-server" href="#Submitting-the-results-to-the-FFmpeg-result-aggregation-server">3 Submitting the results to the FFmpeg result aggregation server</a></li>
+  <li><a name="toc-FATE-makefile-targets-and-variables" href="#FATE-makefile-targets-and-variables">4 FATE makefile targets and variables</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Makefile-targets" href="#Makefile-targets">4.1 Makefile targets</a></li>
+    <li><a name="toc-Makefile-variables" href="#Makefile-variables">4.2 Makefile variables</a></li>
+    <li><a name="toc-Examples" href="#Examples">4.3 Examples</a></li>
+  </ul></li>
+</ul>
+</div>
+
+
+<a name="Introduction"></a>
+<h2 class="chapter">1 Introduction<span class="pull-right"><a class="anchor hidden-xs" href="#Introduction" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Introduction" aria-hidden="true">TOC</a></span></h2>
+
+<p>FATE is an extended regression suite on the client-side and a means
+for results aggregation and presentation on the server-side.
+</p>
+<p>The first part of this document explains how you can use FATE from
+your FFmpeg source directory to test your ffmpeg binary. The second
+part describes how you can run FATE to submit the results to FFmpeg&rsquo;s
+FATE server.
+</p>
+<p>In any way you can have a look at the publicly viewable FATE results
+by visiting this website:
+</p>
+<p><a href="http://fate.ffmpeg.org/">http://fate.ffmpeg.org/</a>
+</p>
+<p>This is especially recommended for all people contributing source
+code to FFmpeg, as it can be seen if some test on some platform broke
+with their recent contribution. This usually happens on the platforms
+the developers could not test on.
+</p>
+<p>The second part of this document describes how you can run FATE to
+submit your results to FFmpeg&rsquo;s FATE server. If you want to submit your
+results be sure to check that your combination of CPU, OS and compiler
+is not already listed on the above mentioned website.
+</p>
+<p>In the third part you can find a comprehensive listing of FATE makefile
+targets and variables.
+</p>
+
+<a name="Using-FATE-from-your-FFmpeg-source-directory"></a>
+<h2 class="chapter">2 Using FATE from your FFmpeg source directory<span class="pull-right"><a class="anchor hidden-xs" href="#Using-FATE-from-your-FFmpeg-source-directory" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-FATE-from-your-FFmpeg-source-directory" aria-hidden="true">TOC</a></span></h2>
+
+<p>If you want to run FATE on your machine you need to have the samples
+in place. You can get the samples via the build target fate-rsync.
+Use this command from the top-level source directory:
+</p>
+<div class="example">
+<pre class="example">make fate-rsync SAMPLES=fate-suite/
+make fate       SAMPLES=fate-suite/
+</pre></div>
+
+<p>The above commands set the samples location by passing a makefile
+variable via command line. It is also possible to set the samples
+location at source configuration time by invoking configure with
+<samp>--samples=&lt;path to the samples directory&gt;</samp>. Afterwards you can
+invoke the makefile targets without setting the <var>SAMPLES</var> makefile
+variable. This is illustrated by the following commands:
+</p>
+<div class="example">
+<pre class="example">./configure --samples=fate-suite/
+make fate-rsync
+make fate
+</pre></div>
+
+<p>Yet another way to tell FATE about the location of the sample
+directory is by making sure the environment variable FATE_SAMPLES
+contains the path to your samples directory. This can be achieved
+by e.g. putting that variable in your shell profile or by setting
+it in your interactive session.
+</p>
+<div class="example">
+<pre class="example">FATE_SAMPLES=fate-suite/ make fate
+</pre></div>
+
+<div class="info">
+<p>Do not put a &rsquo;~&rsquo; character in the samples path to indicate a home
+directory. Because of shell nuances, this will cause FATE to fail.
+</p></div>
+<p>To use a custom wrapper to run the test, pass <samp>--target-exec</samp> to
+<code>configure</code> or set the <var>TARGET_EXEC</var> Make variable.
+</p>
+
+<a name="Submitting-the-results-to-the-FFmpeg-result-aggregation-server"></a>
+<h2 class="chapter">3 Submitting the results to the FFmpeg result aggregation server<span class="pull-right"><a class="anchor hidden-xs" href="#Submitting-the-results-to-the-FFmpeg-result-aggregation-server" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Submitting-the-results-to-the-FFmpeg-result-aggregation-server" aria-hidden="true">TOC</a></span></h2>
+
+<p>To submit your results to the server you should run fate through the
+shell script <samp>tests/fate.sh</samp> from the FFmpeg sources. This script needs
+to be invoked with a configuration file as its first argument.
+</p>
+<div class="example">
+<pre class="example">tests/fate.sh /path/to/fate_config
+</pre></div>
+
+<p>A configuration file template with comments describing the individual
+configuration variables can be found at <samp>doc/fate_config.sh.template</samp>.
+</p>
+<p>The mentioned configuration template is also available here:
+</p><pre class="verbatim">slot=                                    # some unique identifier
+repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
+#branch=release/2.6                       # the branch to test
+samples=                                 # path to samples directory
+workdir=                                 # directory in which to do all the work
+#fate_recv=&quot;ssh -T fate@fate.ffmpeg.org&quot; # command to submit report
+comment=                                 # optional description
+build_only=     # set to &quot;yes&quot; for a compile-only instance that skips tests
+
+# the following are optional and map to configure options
+arch=
+cpu=
+cross_prefix=
+as=
+cc=
+ld=
+target_os=
+sysroot=
+target_exec=
+target_path=
+target_samples=
+extra_cflags=
+extra_ldflags=
+extra_libs=
+extra_conf=     # extra configure options not covered above
+
+#make=          # name of GNU make if not 'make'
+makeopts=       # extra options passed to 'make'
+#tar=           # command to create a tar archive from its arguments on stdout,
+                # defaults to 'tar c'
+</pre>
+<p>Create a configuration that suits your needs, based on the configuration
+template. The <code>slot</code> configuration variable can be any string that is not
+yet used, but it is suggested that you name it adhering to the following
+pattern &lsquo;<samp><var>arch</var>-<var>os</var>-<var>compiler</var>-<var>compiler version</var></samp>&rsquo;. The
+configuration file itself will be sourced in a shell script, therefore all
+shell features may be used. This enables you to setup the environment as you
+need it for your build.
+</p>
+<p>For your first test runs the <code>fate_recv</code> variable should be empty or
+commented out. This will run everything as normal except that it will omit
+the submission of the results to the server. The following files should be
+present in $workdir as specified in the configuration file:
+</p>
+<ul>
+<li> configure.log
+    </li><li> compile.log
+    </li><li> test.log
+    </li><li> report
+    </li><li> version
+</li></ul>
+
+<p>When you have everything working properly you can create an SSH key pair
+and send the public key to the FATE server administrator who can be contacted
+at the email address <a href="mailto:fate-admin@ffmpeg.org">fate-admin@ffmpeg.org</a>.
+</p>
+<p>Configure your SSH client to use public key authentication with that key
+when connecting to the FATE server. Also do not forget to check the identity
+of the server and to accept its host key. This can usually be achieved by
+running your SSH client manually and killing it after you accepted the key.
+The FATE server&rsquo;s fingerprint is:
+</p>
+<dl compact="compact">
+<dt>&lsquo;<samp>RSA</samp>&rsquo;</dt>
+<dd><p>d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
+</p></dd>
+<dt>&lsquo;<samp>ECDSA</samp>&rsquo;</dt>
+<dd><p>76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
+</p></dd>
+</dl>
+
+<p>If you have problems connecting to the FATE server, it may help to try out
+the <code>ssh</code> command with one or more <samp>-v</samp> options. You should
+get detailed output concerning your SSH configuration and the authentication
+process.
+</p>
+<p>The only thing left is to automate the execution of the fate.sh script and
+the synchronisation of the samples directory.
+</p>
+
+<a name="FATE-makefile-targets-and-variables"></a>
+<h2 class="chapter">4 FATE makefile targets and variables<span class="pull-right"><a class="anchor hidden-xs" href="#FATE-makefile-targets-and-variables" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FATE-makefile-targets-and-variables" aria-hidden="true">TOC</a></span></h2>
+
+<a name="Makefile-targets"></a>
+<h3 class="section">4.1 Makefile targets<span class="pull-right"><a class="anchor hidden-xs" href="#Makefile-targets" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Makefile-targets" aria-hidden="true">TOC</a></span></h3>
+
+<dl compact="compact">
+<dt><samp>fate-rsync</samp></dt>
+<dd><p>Download/synchronize sample files to the configured samples directory.
+</p>
+</dd>
+<dt><samp>fate-list</samp></dt>
+<dd><p>Will list all fate/regression test targets.
+</p>
+</dd>
+<dt><samp>fate</samp></dt>
+<dd><p>Run the FATE test suite (requires the fate-suite dataset).
+</p></dd>
+</dl>
+
+<a name="Makefile-variables"></a>
+<h3 class="section">4.2 Makefile variables<span class="pull-right"><a class="anchor hidden-xs" href="#Makefile-variables" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Makefile-variables" aria-hidden="true">TOC</a></span></h3>
+
+<dl compact="compact">
+<dt><code>V</code></dt>
+<dd><p>Verbosity level, can be set to 0, 1 or 2.
+    </p><ul>
+<li> 0: show just the test arguments
+        </li><li> 1: show just the command used in the test
+        </li><li> 2: show everything
+    </li></ul>
+
+</dd>
+<dt><code>SAMPLES</code></dt>
+<dd><p>Specify or override the path to the FATE samples at make time, it has a
+meaning only while running the regression tests.
+</p>
+</dd>
+<dt><code>THREADS</code></dt>
+<dd><p>Specify how many threads to use while running regression tests, it is
+quite useful to detect thread-related regressions.
+</p>
+</dd>
+<dt><code>THREAD_TYPE</code></dt>
+<dd><p>Specify which threading strategy test, either &lsquo;<samp>slice</samp>&rsquo; or &lsquo;<samp>frame</samp>&rsquo;,
+by default &lsquo;<samp>slice+frame</samp>&rsquo;
+</p>
+</dd>
+<dt><code>CPUFLAGS</code></dt>
+<dd><p>Specify CPU flags.
+</p>
+</dd>
+<dt><code>TARGET_EXEC</code></dt>
+<dd><p>Specify or override the wrapper used to run the tests.
+The <code>TARGET_EXEC</code> option provides a way to run FATE wrapped in
+<code>valgrind</code>, <code>qemu-user</code> or <code>wine</code> or on remote targets
+through <code>ssh</code>.
+</p>
+</dd>
+<dt><code>GEN</code></dt>
+<dd><p>Set to &lsquo;<samp>1</samp>&rsquo; to generate the missing or mismatched references.
+</p></dd>
+</dl>
+
+<a name="Examples"></a>
+<h3 class="section">4.3 Examples<span class="pull-right"><a class="anchor hidden-xs" href="#Examples" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Examples" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
+</pre></div>
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,284 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      FFmpeg Bitstream Filters Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
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+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      FFmpeg Bitstream Filters Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-Bitstream-Filters" href="#Bitstream-Filters">2 Bitstream Filters</a>
+  <ul class="no-bullet">
+    <li><a name="toc-aac_005fadtstoasc" href="#aac_005fadtstoasc">2.1 aac_adtstoasc</a></li>
+    <li><a name="toc-chomp" href="#chomp">2.2 chomp</a></li>
+    <li><a name="toc-dump_005fextra" href="#dump_005fextra">2.3 dump_extra</a></li>
+    <li><a name="toc-h264_005fmp4toannexb" href="#h264_005fmp4toannexb">2.4 h264_mp4toannexb</a></li>
+    <li><a name="toc-imxdump" href="#imxdump">2.5 imxdump</a></li>
+    <li><a name="toc-mjpeg2jpeg" href="#mjpeg2jpeg">2.6 mjpeg2jpeg</a></li>
+    <li><a name="toc-mjpega_005fdump_005fheader" href="#mjpega_005fdump_005fheader">2.7 mjpega_dump_header</a></li>
+    <li><a name="toc-movsub" href="#movsub">2.8 movsub</a></li>
+    <li><a name="toc-mp3_005fheader_005fdecompress" href="#mp3_005fheader_005fdecompress">2.9 mp3_header_decompress</a></li>
+    <li><a name="toc-mpeg4_005funpack_005fbframes" href="#mpeg4_005funpack_005fbframes">2.10 mpeg4_unpack_bframes</a></li>
+    <li><a name="toc-noise" href="#noise">2.11 noise</a></li>
+    <li><a name="toc-remove_005fextra" href="#remove_005fextra">2.12 remove_extra</a></li>
+  </ul></li>
+  <li><a name="toc-See-Also" href="#See-Also">3 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">4 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>This document describes the bitstream filters provided by the
+libavcodec library.
+</p>
+<p>A bitstream filter operates on the encoded stream data, and performs
+bitstream level modifications without performing decoding.
+</p>
+
+<a name="Bitstream-Filters"></a>
+<h2 class="chapter">2 Bitstream Filters<span class="pull-right"><a class="anchor hidden-xs" href="#Bitstream-Filters" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Bitstream-Filters" aria-hidden="true">TOC</a></span></h2>
+
+<p>When you configure your FFmpeg build, all the supported bitstream
+filters are enabled by default. You can list all available ones using
+the configure option <code>--list-bsfs</code>.
+</p>
+<p>You can disable all the bitstream filters using the configure option
+<code>--disable-bsfs</code>, and selectively enable any bitstream filter using
+the option <code>--enable-bsf=BSF</code>, or you can disable a particular
+bitstream filter using the option <code>--disable-bsf=BSF</code>.
+</p>
+<p>The option <code>-bsfs</code> of the ff* tools will display the list of
+all the supported bitstream filters included in your build.
+</p>
+<p>The ff* tools have a -bsf option applied per stream, taking a
+comma-separated list of filters, whose parameters follow the filter
+name after a &rsquo;=&rsquo;.
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
+</pre></div>
+
+<p>Below is a description of the currently available bitstream filters,
+with their parameters, if any.
+</p>
+<a name="aac_005fadtstoasc"></a>
+<h3 class="section">2.1 aac_adtstoasc<span class="pull-right"><a class="anchor hidden-xs" href="#aac_005fadtstoasc" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-aac_005fadtstoasc" aria-hidden="true">TOC</a></span></h3>
+
+<p>Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
+bitstream filter.
+</p>
+<p>This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
+ADTS header and removes the ADTS header.
+</p>
+<p>This is required for example when copying an AAC stream from a raw
+ADTS AAC container to a FLV or a MOV/MP4 file.
+</p>
+<a name="chomp"></a>
+<h3 class="section">2.2 chomp<span class="pull-right"><a class="anchor hidden-xs" href="#chomp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-chomp" aria-hidden="true">TOC</a></span></h3>
+
+<p>Remove zero padding at the end of a packet.
+</p>
+<a name="dump_005fextra"></a>
+<h3 class="section">2.3 dump_extra<span class="pull-right"><a class="anchor hidden-xs" href="#dump_005fextra" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-dump_005fextra" aria-hidden="true">TOC</a></span></h3>
+
+<p>Add extradata to the beginning of the filtered packets.
+</p>
+<p>The additional argument specifies which packets should be filtered.
+It accepts the values:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>a</samp>&rsquo;</dt>
+<dd><p>add extradata to all key packets, but only if <var>local_header</var> is
+set in the <samp>flags2</samp> codec context field
+</p>
+</dd>
+<dt>&lsquo;<samp>k</samp>&rsquo;</dt>
+<dd><p>add extradata to all key packets
+</p>
+</dd>
+<dt>&lsquo;<samp>e</samp>&rsquo;</dt>
+<dd><p>add extradata to all packets
+</p></dd>
+</dl>
+
+<p>If not specified it is assumed &lsquo;<samp>k</samp>&rsquo;.
+</p>
+<p>For example the following <code>ffmpeg</code> command forces a global
+header (thus disabling individual packet headers) in the H.264 packets
+generated by the <code>libx264</code> encoder, but corrects them by adding
+the header stored in extradata to the key packets:
+</p><div class="example">
+<pre class="example">ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
+</pre></div>
+
+<a name="h264_005fmp4toannexb"></a>
+<h3 class="section">2.4 h264_mp4toannexb<span class="pull-right"><a class="anchor hidden-xs" href="#h264_005fmp4toannexb" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-h264_005fmp4toannexb" aria-hidden="true">TOC</a></span></h3>
+
+<p>Convert an H.264 bitstream from length prefixed mode to start code
+prefixed mode (as defined in the Annex B of the ITU-T H.264
+specification).
+</p>
+<p>This is required by some streaming formats, typically the MPEG-2
+transport stream format (&quot;mpegts&quot;).
+</p>
+<p>For example to remux an MP4 file containing an H.264 stream to mpegts
+format with <code>ffmpeg</code>, you can use the command:
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
+</pre></div>
+
+<a name="imxdump"></a>
+<h3 class="section">2.5 imxdump<span class="pull-right"><a class="anchor hidden-xs" href="#imxdump" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-imxdump" aria-hidden="true">TOC</a></span></h3>
+
+<p>Modifies the bitstream to fit in MOV and to be usable by the Final Cut
+Pro decoder. This filter only applies to the mpeg2video codec, and is
+likely not needed for Final Cut Pro 7 and newer with the appropriate
+<samp>-tag:v</samp>.
+</p>
+<p>For example, to remux 30 MB/sec NTSC IMX to MOV:
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
+</pre></div>
+
+<a name="mjpeg2jpeg"></a>
+<h3 class="section">2.6 mjpeg2jpeg<span class="pull-right"><a class="anchor hidden-xs" href="#mjpeg2jpeg" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mjpeg2jpeg" aria-hidden="true">TOC</a></span></h3>
+
+<p>Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
+</p>
+<p>MJPEG is a video codec wherein each video frame is essentially a
+JPEG image. The individual frames can be extracted without loss,
+e.g. by
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
+</pre></div>
+
+<p>Unfortunately, these chunks are incomplete JPEG images, because
+they lack the DHT segment required for decoding. Quoting from
+<a href="http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml">http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml</a>:
+</p>
+<p>Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
+commented that &quot;MJPEG, or at least the MJPEG in AVIs having the
+MJPG fourcc, is restricted JPEG with a fixed &ndash; and *omitted* &ndash;
+Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
+and it must use basic Huffman encoding, not arithmetic or
+progressive. . . . You can indeed extract the MJPEG frames and
+decode them with a regular JPEG decoder, but you have to prepend
+the DHT segment to them, or else the decoder won&rsquo;t have any idea
+how to decompress the data. The exact table necessary is given in
+the OpenDML spec.&quot;
+</p>
+<p>This bitstream filter patches the header of frames extracted from an MJPEG
+stream (carrying the AVI1 header ID and lacking a DHT segment) to
+produce fully qualified JPEG images.
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
+exiftran -i -9 frame*.jpg
+ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
+</pre></div>
+
+<a name="mjpega_005fdump_005fheader"></a>
+<h3 class="section">2.7 mjpega_dump_header<span class="pull-right"><a class="anchor hidden-xs" href="#mjpega_005fdump_005fheader" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mjpega_005fdump_005fheader" aria-hidden="true">TOC</a></span></h3>
+
+<a name="movsub"></a>
+<h3 class="section">2.8 movsub<span class="pull-right"><a class="anchor hidden-xs" href="#movsub" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-movsub" aria-hidden="true">TOC</a></span></h3>
+
+<a name="mp3_005fheader_005fdecompress"></a>
+<h3 class="section">2.9 mp3_header_decompress<span class="pull-right"><a class="anchor hidden-xs" href="#mp3_005fheader_005fdecompress" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mp3_005fheader_005fdecompress" aria-hidden="true">TOC</a></span></h3>
+
+<a name="mpeg4_005funpack_005fbframes"></a>
+<h3 class="section">2.10 mpeg4_unpack_bframes<span class="pull-right"><a class="anchor hidden-xs" href="#mpeg4_005funpack_005fbframes" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mpeg4_005funpack_005fbframes" aria-hidden="true">TOC</a></span></h3>
+
+<p>Unpack DivX-style packed B-frames.
+</p>
+<p>DivX-style packed B-frames are not valid MPEG-4 and were only a
+workaround for the broken Video for Windows subsystem.
+They use more space, can cause minor AV sync issues, require more
+CPU power to decode (unless the player has some decoded picture queue
+to compensate the 2,0,2,0 frame per packet style) and cause
+trouble if copied into a standard container like mp4 or mpeg-ps/ts,
+because MPEG-4 decoders may not be able to decode them, since they are
+not valid MPEG-4.
+</p>
+<p>For example to fix an AVI file containing an MPEG-4 stream with
+DivX-style packed B-frames using <code>ffmpeg</code>, you can use the command:
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
+</pre></div>
+
+<a name="noise"></a>
+<h3 class="section">2.11 noise<span class="pull-right"><a class="anchor hidden-xs" href="#noise" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-noise" aria-hidden="true">TOC</a></span></h3>
+
+<p>Damages the contents of packets without damaging the container. Can be
+used for fuzzing or testing error resilience/concealment.
+</p>
+<p>Parameters:
+A numeral string, whose value is related to how often output bytes will
+be modified. Therefore, values below or equal to 0 are forbidden, and
+the lower the more frequent bytes will be modified, with 1 meaning
+every byte is modified.
+</p>
+<div class="example">
+<pre class="example">ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
+</pre></div>
+<p>applies the modification to every byte.
+</p>
+<a name="remove_005fextra"></a>
+<h3 class="section">2.12 remove_extra<span class="pull-right"><a class="anchor hidden-xs" href="#remove_005fextra" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-remove_005fextra" aria-hidden="true">TOC</a></span></h3>
+
+
+<a name="See-Also"></a>
+<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="libavcodec.html">libavcodec</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,358 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      FFmpeg Resampler Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      FFmpeg Resampler Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-Resampler-Options" href="#Resampler-Options">2 Resampler Options</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">3 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">4 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg resampler provides a high-level interface to the
+libswresample library audio resampling utilities. In particular it
+allows one to perform audio resampling, audio channel layout rematrixing,
+and convert audio format and packing layout.
+</p>
+
+<a name="Resampler-Options"></a>
+<h2 class="chapter">2 Resampler Options<span class="pull-right"><a class="anchor hidden-xs" href="#Resampler-Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Resampler-Options" aria-hidden="true">TOC</a></span></h2>
+
+<p>The audio resampler supports the following named options.
+</p>
+<p>Options may be set by specifying -<var>option</var> <var>value</var> in the
+FFmpeg tools, <var>option</var>=<var>value</var> for the aresample filter,
+by setting the value explicitly in the
+<code>SwrContext</code> options or using the <samp>libavutil/opt.h</samp> API for
+programmatic use.
+</p>
+<dl compact="compact">
+<dt><samp>ich, in_channel_count</samp></dt>
+<dd><p>Set the number of input channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+<samp>in_channel_layout</samp> is set.
+</p>
+</dd>
+<dt><samp>och, out_channel_count</samp></dt>
+<dd><p>Set the number of output channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+<samp>out_channel_layout</samp> is set.
+</p>
+</dd>
+<dt><samp>uch, used_channel_count</samp></dt>
+<dd><p>Set the number of used input channels. Default value is 0. This option is
+only used for special remapping.
+</p>
+</dd>
+<dt><samp>isr, in_sample_rate</samp></dt>
+<dd><p>Set the input sample rate. Default value is 0.
+</p>
+</dd>
+<dt><samp>osr, out_sample_rate</samp></dt>
+<dd><p>Set the output sample rate. Default value is 0.
+</p>
+</dd>
+<dt><samp>isf, in_sample_fmt</samp></dt>
+<dd><p>Specify the input sample format. It is set by default to <code>none</code>.
+</p>
+</dd>
+<dt><samp>osf, out_sample_fmt</samp></dt>
+<dd><p>Specify the output sample format. It is set by default to <code>none</code>.
+</p>
+</dd>
+<dt><samp>tsf, internal_sample_fmt</samp></dt>
+<dd><p>Set the internal sample format. Default value is <code>none</code>.
+This will automatically be chosen when it is not explicitly set.
+</p>
+</dd>
+<dt><samp>icl, in_channel_layout</samp></dt>
+<dt><samp>ocl, out_channel_layout</samp></dt>
+<dd><p>Set the input/output channel layout.
+</p>
+<p>See <a href="ffmpeg-utils.html#channel-layout-syntax">(ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual</a>
+for the required syntax.
+</p>
+</dd>
+<dt><samp>clev, center_mix_level</samp></dt>
+<dd><p>Set the center mix level. It is a value expressed in deciBel, and must be
+in the interval [-32,32].
+</p>
+</dd>
+<dt><samp>slev, surround_mix_level</samp></dt>
+<dd><p>Set the surround mix level. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
+</p>
+</dd>
+<dt><samp>lfe_mix_level</samp></dt>
+<dd><p>Set LFE mix into non LFE level. It is used when there is a LFE input but no
+LFE output. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
+</p>
+</dd>
+<dt><samp>rmvol, rematrix_volume</samp></dt>
+<dd><p>Set rematrix volume. Default value is 1.0.
+</p>
+</dd>
+<dt><samp>rematrix_maxval</samp></dt>
+<dd><p>Set maximum output value for rematrixing.
+This can be used to prevent clipping vs. preventing volumn reduction
+A value of 1.0 prevents cliping.
+</p>
+</dd>
+<dt><samp>flags, swr_flags</samp></dt>
+<dd><p>Set flags used by the converter. Default value is 0.
+</p>
+<p>It supports the following individual flags:
+</p><dl compact="compact">
+<dt><samp>res</samp></dt>
+<dd><p>force resampling, this flag forces resampling to be used even when the
+input and output sample rates match.
+</p></dd>
+</dl>
+
+</dd>
+<dt><samp>dither_scale</samp></dt>
+<dd><p>Set the dither scale. Default value is 1.
+</p>
+</dd>
+<dt><samp>dither_method</samp></dt>
+<dd><p>Set dither method. Default value is 0.
+</p>
+<p>Supported values:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>rectangular</samp>&rsquo;</dt>
+<dd><p>select rectangular dither
+</p></dd>
+<dt>&lsquo;<samp>triangular</samp>&rsquo;</dt>
+<dd><p>select triangular dither
+</p></dd>
+<dt>&lsquo;<samp>triangular_hp</samp>&rsquo;</dt>
+<dd><p>select triangular dither with high pass
+</p></dd>
+<dt>&lsquo;<samp>lipshitz</samp>&rsquo;</dt>
+<dd><p>select Lipshitz noise shaping dither.
+</p></dd>
+<dt>&lsquo;<samp>shibata</samp>&rsquo;</dt>
+<dd><p>select Shibata noise shaping dither.
+</p></dd>
+<dt>&lsquo;<samp>low_shibata</samp>&rsquo;</dt>
+<dd><p>select low Shibata noise shaping dither.
+</p></dd>
+<dt>&lsquo;<samp>high_shibata</samp>&rsquo;</dt>
+<dd><p>select high Shibata noise shaping dither.
+</p></dd>
+<dt>&lsquo;<samp>f_weighted</samp>&rsquo;</dt>
+<dd><p>select f-weighted noise shaping dither
+</p></dd>
+<dt>&lsquo;<samp>modified_e_weighted</samp>&rsquo;</dt>
+<dd><p>select modified-e-weighted noise shaping dither
+</p></dd>
+<dt>&lsquo;<samp>improved_e_weighted</samp>&rsquo;</dt>
+<dd><p>select improved-e-weighted noise shaping dither
+</p>
+</dd>
+</dl>
+
+</dd>
+<dt><samp>resampler</samp></dt>
+<dd><p>Set resampling engine. Default value is swr.
+</p>
+<p>Supported values:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>swr</samp>&rsquo;</dt>
+<dd><p>select the native SW Resampler; filter options precision and cheby are not
+applicable in this case.
+</p></dd>
+<dt>&lsquo;<samp>soxr</samp>&rsquo;</dt>
+<dd><p>select the SoX Resampler (where available); compensation, and filter options
+filter_size, phase_shift, filter_type &amp; kaiser_beta, are not applicable in this
+case.
+</p></dd>
+</dl>
+
+</dd>
+<dt><samp>filter_size</samp></dt>
+<dd><p>For swr only, set resampling filter size, default value is 32.
+</p>
+</dd>
+<dt><samp>phase_shift</samp></dt>
+<dd><p>For swr only, set resampling phase shift, default value is 10, and must be in
+the interval [0,30].
+</p>
+</dd>
+<dt><samp>linear_interp</samp></dt>
+<dd><p>Use linear interpolation if set to 1, default value is 0.
+</p>
+</dd>
+<dt><samp>cutoff</samp></dt>
+<dd><p>Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
+value between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr
+(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
+</p>
+</dd>
+<dt><samp>precision</samp></dt>
+<dd><p>For soxr only, the precision in bits to which the resampled signal will be
+calculated.  The default value of 20 (which, with suitable dithering, is
+appropriate for a destination bit-depth of 16) gives SoX&rsquo;s &rsquo;High Quality&rsquo;; a
+value of 28 gives SoX&rsquo;s &rsquo;Very High Quality&rsquo;.
+</p>
+</dd>
+<dt><samp>cheby</samp></dt>
+<dd><p>For soxr only, selects passband rolloff none (Chebyshev) &amp; higher-precision
+approximation for &rsquo;irrational&rsquo; ratios. Default value is 0.
+</p>
+</dd>
+<dt><samp>async</samp></dt>
+<dd><p>For swr only, simple 1 parameter audio sync to timestamps using stretching,
+squeezing, filling and trimming. Setting this to 1 will enable filling and
+trimming, larger values represent the maximum amount in samples that the data
+may be stretched or squeezed for each second.
+Default value is 0, thus no compensation is applied to make the samples match
+the audio timestamps.
+</p>
+</dd>
+<dt><samp>first_pts</samp></dt>
+<dd><p>For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
+This allows for padding/trimming at the start of stream. By default, no
+assumption is made about the first frame&rsquo;s expected pts, so no padding or
+trimming is done. For example, this could be set to 0 to pad the beginning with
+silence if an audio stream starts after the video stream or to trim any samples
+with a negative pts due to encoder delay.
+</p>
+</dd>
+<dt><samp>min_comp</samp></dt>
+<dd><p>For swr only, set the minimum difference between timestamps and audio data (in
+seconds) to trigger stretching/squeezing/filling or trimming of the
+data to make it match the timestamps. The default is that
+stretching/squeezing/filling and trimming is disabled
+(<samp>min_comp</samp> = <code>FLT_MAX</code>).
+</p>
+</dd>
+<dt><samp>min_hard_comp</samp></dt>
+<dd><p>For swr only, set the minimum difference between timestamps and audio data (in
+seconds) to trigger adding/dropping samples to make it match the
+timestamps.  This option effectively is a threshold to select between
+hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
+all compensation is by default disabled through <samp>min_comp</samp>.
+The default is 0.1.
+</p>
+</dd>
+<dt><samp>comp_duration</samp></dt>
+<dd><p>For swr only, set duration (in seconds) over which data is stretched/squeezed
+to make it match the timestamps. Must be a non-negative double float value,
+default value is 1.0.
+</p>
+</dd>
+<dt><samp>max_soft_comp</samp></dt>
+<dd><p>For swr only, set maximum factor by which data is stretched/squeezed to make it
+match the timestamps. Must be a non-negative double float value, default value
+is 0.
+</p>
+</dd>
+<dt><samp>matrix_encoding</samp></dt>
+<dd><p>Select matrixed stereo encoding.
+</p>
+<p>It accepts the following values:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>none</samp>&rsquo;</dt>
+<dd><p>select none
+</p></dd>
+<dt>&lsquo;<samp>dolby</samp>&rsquo;</dt>
+<dd><p>select Dolby
+</p></dd>
+<dt>&lsquo;<samp>dplii</samp>&rsquo;</dt>
+<dd><p>select Dolby Pro Logic II
+</p></dd>
+</dl>
+
+<p>Default value is <code>none</code>.
+</p>
+</dd>
+<dt><samp>filter_type</samp></dt>
+<dd><p>For swr only, select resampling filter type. This only affects resampling
+operations.
+</p>
+<p>It accepts the following values:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>cubic</samp>&rsquo;</dt>
+<dd><p>select cubic
+</p></dd>
+<dt>&lsquo;<samp>blackman_nuttall</samp>&rsquo;</dt>
+<dd><p>select Blackman Nuttall windowed sinc
+</p></dd>
+<dt>&lsquo;<samp>kaiser</samp>&rsquo;</dt>
+<dd><p>select Kaiser windowed sinc
+</p></dd>
+</dl>
+
+</dd>
+<dt><samp>kaiser_beta</samp></dt>
+<dd><p>For swr only, set Kaiser window beta value. Must be an integer in the
+interval [2,16], default value is 9.
+</p>
+</dd>
+<dt><samp>output_sample_bits</samp></dt>
+<dd><p>For swr only, set number of used output sample bits for dithering. Must be an integer in the
+interval [0,64], default value is 0, which means it&rsquo;s not used.
+</p>
+</dd>
+</dl>
+
+
+<a name="See-Also"></a>
+<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="libswresample.html">libswresample</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,252 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      FFmpeg Scaler Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      FFmpeg Scaler Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-Scaler-Options" href="#Scaler-Options">2 Scaler Options</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">3 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">4 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg rescaler provides a high-level interface to the libswscale
+library image conversion utilities. In particular it allows one to perform
+image rescaling and pixel format conversion.
+</p>
+
+<a name="scaler_005foptions"></a><a name="Scaler-Options"></a>
+<h2 class="chapter">2 Scaler Options<span class="pull-right"><a class="anchor hidden-xs" href="#Scaler-Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Scaler-Options" aria-hidden="true">TOC</a></span></h2>
+
+<p>The video scaler supports the following named options.
+</p>
+<p>Options may be set by specifying -<var>option</var> <var>value</var> in the
+FFmpeg tools. For programmatic use, they can be set explicitly in the
+<code>SwsContext</code> options or through the <samp>libavutil/opt.h</samp> API.
+</p>
+<dl compact="compact">
+<dd>
+<a name="sws_005fflags"></a></dd>
+<dt><samp>sws_flags</samp></dt>
+<dd><p>Set the scaler flags. This is also used to set the scaling
+algorithm. Only a single algorithm should be selected.
+</p>
+<p>It accepts the following values:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>fast_bilinear</samp>&rsquo;</dt>
+<dd><p>Select fast bilinear scaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>bilinear</samp>&rsquo;</dt>
+<dd><p>Select bilinear scaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>bicubic</samp>&rsquo;</dt>
+<dd><p>Select bicubic scaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>experimental</samp>&rsquo;</dt>
+<dd><p>Select experimental scaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>neighbor</samp>&rsquo;</dt>
+<dd><p>Select nearest neighbor rescaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>area</samp>&rsquo;</dt>
+<dd><p>Select averaging area rescaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>bicublin</samp>&rsquo;</dt>
+<dd><p>Select bicubic scaling algorithm for the luma component, bilinear for
+chroma components.
+</p>
+</dd>
+<dt>&lsquo;<samp>gauss</samp>&rsquo;</dt>
+<dd><p>Select Gaussian rescaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>sinc</samp>&rsquo;</dt>
+<dd><p>Select sinc rescaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>lanczos</samp>&rsquo;</dt>
+<dd><p>Select Lanczos rescaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>spline</samp>&rsquo;</dt>
+<dd><p>Select natural bicubic spline rescaling algorithm.
+</p>
+</dd>
+<dt>&lsquo;<samp>print_info</samp>&rsquo;</dt>
+<dd><p>Enable printing/debug logging.
+</p>
+</dd>
+<dt>&lsquo;<samp>accurate_rnd</samp>&rsquo;</dt>
+<dd><p>Enable accurate rounding.
+</p>
+</dd>
+<dt>&lsquo;<samp>full_chroma_int</samp>&rsquo;</dt>
+<dd><p>Enable full chroma interpolation.
+</p>
+</dd>
+<dt>&lsquo;<samp>full_chroma_inp</samp>&rsquo;</dt>
+<dd><p>Select full chroma input.
+</p>
+</dd>
+<dt>&lsquo;<samp>bitexact</samp>&rsquo;</dt>
+<dd><p>Enable bitexact output.
+</p></dd>
+</dl>
+
+</dd>
+<dt><samp>srcw</samp></dt>
+<dd><p>Set source width.
+</p>
+</dd>
+<dt><samp>srch</samp></dt>
+<dd><p>Set source height.
+</p>
+</dd>
+<dt><samp>dstw</samp></dt>
+<dd><p>Set destination width.
+</p>
+</dd>
+<dt><samp>dsth</samp></dt>
+<dd><p>Set destination height.
+</p>
+</dd>
+<dt><samp>src_format</samp></dt>
+<dd><p>Set source pixel format (must be expressed as an integer).
+</p>
+</dd>
+<dt><samp>dst_format</samp></dt>
+<dd><p>Set destination pixel format (must be expressed as an integer).
+</p>
+</dd>
+<dt><samp>src_range</samp></dt>
+<dd><p>Select source range.
+</p>
+</dd>
+<dt><samp>dst_range</samp></dt>
+<dd><p>Select destination range.
+</p>
+</dd>
+<dt><samp>param0, param1</samp></dt>
+<dd><p>Set scaling algorithm parameters. The specified values are specific of
+some scaling algorithms and ignored by others. The specified values
+are floating point number values.
+</p>
+</dd>
+<dt><samp>sws_dither</samp></dt>
+<dd><p>Set the dithering algorithm. Accepts one of the following
+values. Default value is &lsquo;<samp>auto</samp>&rsquo;.
+</p>
+<dl compact="compact">
+<dt>&lsquo;<samp>auto</samp>&rsquo;</dt>
+<dd><p>automatic choice
+</p>
+</dd>
+<dt>&lsquo;<samp>none</samp>&rsquo;</dt>
+<dd><p>no dithering
+</p>
+</dd>
+<dt>&lsquo;<samp>bayer</samp>&rsquo;</dt>
+<dd><p>bayer dither
+</p>
+</dd>
+<dt>&lsquo;<samp>ed</samp>&rsquo;</dt>
+<dd><p>error diffusion dither
+</p>
+</dd>
+<dt>&lsquo;<samp>a_dither</samp>&rsquo;</dt>
+<dd><p>arithmetic dither, based using addition
+</p>
+</dd>
+<dt>&lsquo;<samp>x_dither</samp>&rsquo;</dt>
+<dd><p>arithmetic dither, based using xor (more random/less apparent patterning that
+a_dither).
+</p>
+</dd>
+</dl>
+
+</dd>
+<dt><samp>alphablend</samp></dt>
+<dd><p>Set the alpha blending to use when the input has alpha but the output does not.
+Default value is &lsquo;<samp>none</samp>&rsquo;.
+</p>
+<dl compact="compact">
+<dt>&lsquo;<samp>uniform_color</samp>&rsquo;</dt>
+<dd><p>Blend onto a uniform background color
+</p>
+</dd>
+<dt>&lsquo;<samp>checkerboard</samp>&rsquo;</dt>
+<dd><p>Blend onto a checkerboard
+</p>
+</dd>
+<dt>&lsquo;<samp>none</samp>&rsquo;</dt>
+<dd><p>No blending
+</p>
+</dd>
+</dl>
+
+</dd>
+</dl>
+
+
+<a name="See-Also"></a>
+<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="libswscale.html">libswscale</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,818 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      ffplay Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      ffplay Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Synopsis" href="#Synopsis">1 Synopsis</a></li>
+  <li><a name="toc-Description" href="#Description">2 Description</a></li>
+  <li><a name="toc-Options" href="#Options">3 Options</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Stream-specifiers-1" href="#Stream-specifiers-1">3.1 Stream specifiers</a></li>
+    <li><a name="toc-Generic-options" href="#Generic-options">3.2 Generic options</a></li>
+    <li><a name="toc-AVOptions" href="#AVOptions">3.3 AVOptions</a></li>
+    <li><a name="toc-Main-options" href="#Main-options">3.4 Main options</a></li>
+    <li><a name="toc-Advanced-options" href="#Advanced-options">3.5 Advanced options</a></li>
+    <li><a name="toc-While-playing" href="#While-playing">3.6 While playing</a></li>
+  </ul></li>
+  <li><a name="toc-See-Also" href="#See-Also">4 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">5 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Synopsis"></a>
+<h2 class="chapter">1 Synopsis<span class="pull-right"><a class="anchor hidden-xs" href="#Synopsis" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Synopsis" aria-hidden="true">TOC</a></span></h2>
+
+<p>ffplay [<var>options</var>] [<samp>input_file</samp>]
+</p>
+<a name="Description"></a>
+<h2 class="chapter">2 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>FFplay is a very simple and portable media player using the FFmpeg
+libraries and the SDL library. It is mostly used as a testbed for the
+various FFmpeg APIs.
+</p>
+<a name="Options"></a>
+<h2 class="chapter">3 Options<span class="pull-right"><a class="anchor hidden-xs" href="#Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Options" aria-hidden="true">TOC</a></span></h2>
+
+<p>All the numerical options, if not specified otherwise, accept a string
+representing a number as input, which may be followed by one of the SI
+unit prefixes, for example: &rsquo;K&rsquo;, &rsquo;M&rsquo;, or &rsquo;G&rsquo;.
+</p>
+<p>If &rsquo;i&rsquo; is appended to the SI unit prefix, the complete prefix will be
+interpreted as a unit prefix for binary multiples, which are based on
+powers of 1024 instead of powers of 1000. Appending &rsquo;B&rsquo; to the SI unit
+prefix multiplies the value by 8. This allows using, for example:
+&rsquo;KB&rsquo;, &rsquo;MiB&rsquo;, &rsquo;G&rsquo; and &rsquo;B&rsquo; as number suffixes.
+</p>
+<p>Options which do not take arguments are boolean options, and set the
+corresponding value to true. They can be set to false by prefixing
+the option name with &quot;no&quot;. For example using &quot;-nofoo&quot;
+will set the boolean option with name &quot;foo&quot; to false.
+</p>
+<a name="Stream-specifiers"></a><a name="Stream-specifiers-1"></a>
+<h3 class="section">3.1 Stream specifiers<span class="pull-right"><a class="anchor hidden-xs" href="#Stream-specifiers-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Stream-specifiers-1" aria-hidden="true">TOC</a></span></h3>
+<p>Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
+are used to precisely specify which stream(s) a given option belongs to.
+</p>
+<p>A stream specifier is a string generally appended to the option name and
+separated from it by a colon. E.g. <code>-codec:a:1 ac3</code> contains the
+<code>a:1</code> stream specifier, which matches the second audio stream. Therefore, it
+would select the ac3 codec for the second audio stream.
+</p>
+<p>A stream specifier can match several streams, so that the option is applied to all
+of them. E.g. the stream specifier in <code>-b:a 128k</code> matches all audio
+streams.
+</p>
+<p>An empty stream specifier matches all streams. For example, <code>-codec copy</code>
+or <code>-codec: copy</code> would copy all the streams without reencoding.
+</p>
+<p>Possible forms of stream specifiers are:
+</p><dl compact="compact">
+<dt><samp><var>stream_index</var></samp></dt>
+<dd><p>Matches the stream with this index. E.g. <code>-threads:1 4</code> would set the
+thread count for the second stream to 4.
+</p></dd>
+<dt><samp><var>stream_type</var>[:<var>stream_index</var>]</samp></dt>
+<dd><p><var>stream_type</var> is one of following: &rsquo;v&rsquo; or &rsquo;V&rsquo; for video, &rsquo;a&rsquo; for audio, &rsquo;s&rsquo;
+for subtitle, &rsquo;d&rsquo; for data, and &rsquo;t&rsquo; for attachments. &rsquo;v&rsquo; matches all video
+streams, &rsquo;V&rsquo; only matches video streams which are not attached pictures, video
+thumbnails or cover arts.  If <var>stream_index</var> is given, then it matches
+stream number <var>stream_index</var> of this type. Otherwise, it matches all
+streams of this type.
+</p></dd>
+<dt><samp>p:<var>program_id</var>[:<var>stream_index</var>]</samp></dt>
+<dd><p>If <var>stream_index</var> is given, then it matches the stream with number <var>stream_index</var>
+in the program with the id <var>program_id</var>. Otherwise, it matches all streams in the
+program.
+</p></dd>
+<dt><samp>#<var>stream_id</var> or i:<var>stream_id</var></samp></dt>
+<dd><p>Match the stream by stream id (e.g. PID in MPEG-TS container).
+</p></dd>
+<dt><samp>m:<var>key</var>[:<var>value</var>]</samp></dt>
+<dd><p>Matches streams with the metadata tag <var>key</var> having the specified value. If
+<var>value</var> is not given, matches streams that contain the given tag with any
+value.
+</p></dd>
+<dt><samp>u</samp></dt>
+<dd><p>Matches streams with usable configuration, the codec must be defined and the
+essential information such as video dimension or audio sample rate must be present.
+</p>
+<p>Note that in <code>ffmpeg</code>, matching by metadata will only work properly for
+input files.
+</p></dd>
+</dl>
+
+<a name="Generic-options"></a>
+<h3 class="section">3.2 Generic options<span class="pull-right"><a class="anchor hidden-xs" href="#Generic-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Generic-options" aria-hidden="true">TOC</a></span></h3>
+
+<p>These options are shared amongst the ff* tools.
+</p>
+<dl compact="compact">
+<dt><samp>-L</samp></dt>
+<dd><p>Show license.
+</p>
+</dd>
+<dt><samp>-h, -?, -help, --help [<var>arg</var>]</samp></dt>
+<dd><p>Show help. An optional parameter may be specified to print help about a specific
+item. If no argument is specified, only basic (non advanced) tool
+options are shown.
+</p>
+<p>Possible values of <var>arg</var> are:
+</p><dl compact="compact">
+<dt><samp>long</samp></dt>
+<dd><p>Print advanced tool options in addition to the basic tool options.
+</p>
+</dd>
+<dt><samp>full</samp></dt>
+<dd><p>Print complete list of options, including shared and private options
+for encoders, decoders, demuxers, muxers, filters, etc.
+</p>
+</dd>
+<dt><samp>decoder=<var>decoder_name</var></samp></dt>
+<dd><p>Print detailed information about the decoder named <var>decoder_name</var>. Use the
+<samp>-decoders</samp> option to get a list of all decoders.
+</p>
+</dd>
+<dt><samp>encoder=<var>encoder_name</var></samp></dt>
+<dd><p>Print detailed information about the encoder named <var>encoder_name</var>. Use the
+<samp>-encoders</samp> option to get a list of all encoders.
+</p>
+</dd>
+<dt><samp>demuxer=<var>demuxer_name</var></samp></dt>
+<dd><p>Print detailed information about the demuxer named <var>demuxer_name</var>. Use the
+<samp>-formats</samp> option to get a list of all demuxers and muxers.
+</p>
+</dd>
+<dt><samp>muxer=<var>muxer_name</var></samp></dt>
+<dd><p>Print detailed information about the muxer named <var>muxer_name</var>. Use the
+<samp>-formats</samp> option to get a list of all muxers and demuxers.
+</p>
+</dd>
+<dt><samp>filter=<var>filter_name</var></samp></dt>
+<dd><p>Print detailed information about the filter name <var>filter_name</var>. Use the
+<samp>-filters</samp> option to get a list of all filters.
+</p></dd>
+</dl>
+
+</dd>
+<dt><samp>-version</samp></dt>
+<dd><p>Show version.
+</p>
+</dd>
+<dt><samp>-formats</samp></dt>
+<dd><p>Show available formats (including devices).
+</p>
+</dd>
+<dt><samp>-devices</samp></dt>
+<dd><p>Show available devices.
+</p>
+</dd>
+<dt><samp>-codecs</samp></dt>
+<dd><p>Show all codecs known to libavcodec.
+</p>
+<p>Note that the term &rsquo;codec&rsquo; is used throughout this documentation as a shortcut
+for what is more correctly called a media bitstream format.
+</p>
+</dd>
+<dt><samp>-decoders</samp></dt>
+<dd><p>Show available decoders.
+</p>
+</dd>
+<dt><samp>-encoders</samp></dt>
+<dd><p>Show all available encoders.
+</p>
+</dd>
+<dt><samp>-bsfs</samp></dt>
+<dd><p>Show available bitstream filters.
+</p>
+</dd>
+<dt><samp>-protocols</samp></dt>
+<dd><p>Show available protocols.
+</p>
+</dd>
+<dt><samp>-filters</samp></dt>
+<dd><p>Show available libavfilter filters.
+</p>
+</dd>
+<dt><samp>-pix_fmts</samp></dt>
+<dd><p>Show available pixel formats.
+</p>
+</dd>
+<dt><samp>-sample_fmts</samp></dt>
+<dd><p>Show available sample formats.
+</p>
+</dd>
+<dt><samp>-layouts</samp></dt>
+<dd><p>Show channel names and standard channel layouts.
+</p>
+</dd>
+<dt><samp>-colors</samp></dt>
+<dd><p>Show recognized color names.
+</p>
+</dd>
+<dt><samp>-sources <var>device</var>[,<var>opt1</var>=<var>val1</var>[,<var>opt2</var>=<var>val2</var>]...]</samp></dt>
+<dd><p>Show autodetected sources of the intput device.
+Some devices may provide system-dependent source names that cannot be autodetected.
+The returned list cannot be assumed to be always complete.
+</p><div class="example">
+<pre class="example">ffmpeg -sources pulse,server=192.168.0.4
+</pre></div>
+
+</dd>
+<dt><samp>-sinks <var>device</var>[,<var>opt1</var>=<var>val1</var>[,<var>opt2</var>=<var>val2</var>]...]</samp></dt>
+<dd><p>Show autodetected sinks of the output device.
+Some devices may provide system-dependent sink names that cannot be autodetected.
+The returned list cannot be assumed to be always complete.
+</p><div class="example">
+<pre class="example">ffmpeg -sinks pulse,server=192.168.0.4
+</pre></div>
+
+</dd>
+<dt><samp>-loglevel [repeat+]<var>loglevel</var> | -v [repeat+]<var>loglevel</var></samp></dt>
+<dd><p>Set the logging level used by the library.
+Adding &quot;repeat+&quot; indicates that repeated log output should not be compressed
+to the first line and the &quot;Last message repeated n times&quot; line will be
+omitted. &quot;repeat&quot; can also be used alone.
+If &quot;repeat&quot; is used alone, and with no prior loglevel set, the default
+loglevel will be used. If multiple loglevel parameters are given, using
+&rsquo;repeat&rsquo; will not change the loglevel.
+<var>loglevel</var> is a string or a number containing one of the following values:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>quiet, -8</samp>&rsquo;</dt>
+<dd><p>Show nothing at all; be silent.
+</p></dd>
+<dt>&lsquo;<samp>panic, 0</samp>&rsquo;</dt>
+<dd><p>Only show fatal errors which could lead the process to crash, such as
+and assert failure. This is not currently used for anything.
+</p></dd>
+<dt>&lsquo;<samp>fatal, 8</samp>&rsquo;</dt>
+<dd><p>Only show fatal errors. These are errors after which the process absolutely
+cannot continue after.
+</p></dd>
+<dt>&lsquo;<samp>error, 16</samp>&rsquo;</dt>
+<dd><p>Show all errors, including ones which can be recovered from.
+</p></dd>
+<dt>&lsquo;<samp>warning, 24</samp>&rsquo;</dt>
+<dd><p>Show all warnings and errors. Any message related to possibly
+incorrect or unexpected events will be shown.
+</p></dd>
+<dt>&lsquo;<samp>info, 32</samp>&rsquo;</dt>
+<dd><p>Show informative messages during processing. This is in addition to
+warnings and errors. This is the default value.
+</p></dd>
+<dt>&lsquo;<samp>verbose, 40</samp>&rsquo;</dt>
+<dd><p>Same as <code>info</code>, except more verbose.
+</p></dd>
+<dt>&lsquo;<samp>debug, 48</samp>&rsquo;</dt>
+<dd><p>Show everything, including debugging information.
+</p></dd>
+<dt>&lsquo;<samp>trace, 56</samp>&rsquo;</dt>
+</dl>
+
+<p>By default the program logs to stderr, if coloring is supported by the
+terminal, colors are used to mark errors and warnings. Log coloring
+can be disabled setting the environment variable
+<code>AV_LOG_FORCE_NOCOLOR</code> or <code>NO_COLOR</code>, or can be forced setting
+the environment variable <code>AV_LOG_FORCE_COLOR</code>.
+The use of the environment variable <code>NO_COLOR</code> is deprecated and
+will be dropped in a following FFmpeg version.
+</p>
+</dd>
+<dt><samp>-report</samp></dt>
+<dd><p>Dump full command line and console output to a file named
+<code><var>program</var>-<var>YYYYMMDD</var>-<var>HHMMSS</var>.log</code> in the current
+directory.
+This file can be useful for bug reports.
+It also implies <code>-loglevel verbose</code>.
+</p>
+<p>Setting the environment variable <code>FFREPORT</code> to any value has the
+same effect. If the value is a &rsquo;:&rsquo;-separated key=value sequence, these
+options will affect the report; option values must be escaped if they
+contain special characters or the options delimiter &rsquo;:&rsquo; (see the
+&ldquo;Quoting and escaping&rdquo; section in the ffmpeg-utils manual).
+</p>
+<p>The following options are recognized:
+</p><dl compact="compact">
+<dt><samp>file</samp></dt>
+<dd><p>set the file name to use for the report; <code>%p</code> is expanded to the name
+of the program, <code>%t</code> is expanded to a timestamp, <code>%%</code> is expanded
+to a plain <code>%</code>
+</p></dd>
+<dt><samp>level</samp></dt>
+<dd><p>set the log verbosity level using a numerical value (see <code>-loglevel</code>).
+</p></dd>
+</dl>
+
+<p>For example, to output a report to a file named <samp>ffreport.log</samp>
+using a log level of <code>32</code> (alias for log level <code>info</code>):
+</p>
+<div class="example">
+<pre class="example">FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
+</pre></div>
+
+<p>Errors in parsing the environment variable are not fatal, and will not
+appear in the report.
+</p>
+</dd>
+<dt><samp>-hide_banner</samp></dt>
+<dd><p>Suppress printing banner.
+</p>
+<p>All FFmpeg tools will normally show a copyright notice, build options
+and library versions. This option can be used to suppress printing
+this information.
+</p>
+</dd>
+<dt><samp>-cpuflags flags (<em>global</em>)</samp></dt>
+<dd><p>Allows setting and clearing cpu flags. This option is intended
+for testing. Do not use it unless you know what you&rsquo;re doing.
+</p><div class="example">
+<pre class="example">ffmpeg -cpuflags -sse+mmx ...
+ffmpeg -cpuflags mmx ...
+ffmpeg -cpuflags 0 ...
+</pre></div>
+<p>Possible flags for this option are:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>x86</samp>&rsquo;</dt>
+<dd><dl compact="compact">
+<dt>&lsquo;<samp>mmx</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>mmxext</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>sse</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>sse2</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>sse2slow</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>sse3</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>sse3slow</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>ssse3</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>atom</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>sse4.1</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>sse4.2</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>avx</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>avx2</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>xop</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>fma3</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>fma4</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>3dnow</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>3dnowext</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>bmi1</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>bmi2</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>cmov</samp>&rsquo;</dt>
+</dl>
+</dd>
+<dt>&lsquo;<samp>ARM</samp>&rsquo;</dt>
+<dd><dl compact="compact">
+<dt>&lsquo;<samp>armv5te</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>armv6</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>armv6t2</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>vfp</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>vfpv3</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>neon</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>setend</samp>&rsquo;</dt>
+</dl>
+</dd>
+<dt>&lsquo;<samp>AArch64</samp>&rsquo;</dt>
+<dd><dl compact="compact">
+<dt>&lsquo;<samp>armv8</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>vfp</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>neon</samp>&rsquo;</dt>
+</dl>
+</dd>
+<dt>&lsquo;<samp>PowerPC</samp>&rsquo;</dt>
+<dd><dl compact="compact">
+<dt>&lsquo;<samp>altivec</samp>&rsquo;</dt>
+</dl>
+</dd>
+<dt>&lsquo;<samp>Specific Processors</samp>&rsquo;</dt>
+<dd><dl compact="compact">
+<dt>&lsquo;<samp>pentium2</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>pentium3</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>pentium4</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>k6</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>k62</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>athlon</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>athlonxp</samp>&rsquo;</dt>
+<dt>&lsquo;<samp>k8</samp>&rsquo;</dt>
+</dl>
+</dd>
+</dl>
+
+</dd>
+<dt><samp>-opencl_bench</samp></dt>
+<dd><p>This option is used to benchmark all available OpenCL devices and print the
+results. This option is only available when FFmpeg has been compiled with
+<code>--enable-opencl</code>.
+</p>
+<p>When FFmpeg is configured with <code>--enable-opencl</code>, the options for the
+global OpenCL context are set via <samp>-opencl_options</samp>. See the
+&quot;OpenCL Options&quot; section in the ffmpeg-utils manual for the complete list of
+supported options. Amongst others, these options include the ability to select
+a specific platform and device to run the OpenCL code on. By default, FFmpeg
+will run on the first device of the first platform. While the options for the
+global OpenCL context provide flexibility to the user in selecting the OpenCL
+device of their choice, most users would probably want to select the fastest
+OpenCL device for their system.
+</p>
+<p>This option assists the selection of the most efficient configuration by
+identifying the appropriate device for the user&rsquo;s system. The built-in
+benchmark is run on all the OpenCL devices and the performance is measured for
+each device. The devices in the results list are sorted based on their
+performance with the fastest device listed first. The user can subsequently
+invoke <code>ffmpeg</code> using the device deemed most appropriate via
+<samp>-opencl_options</samp> to obtain the best performance for the OpenCL
+accelerated code.
+</p>
+<p>Typical usage to use the fastest OpenCL device involve the following steps.
+</p>
+<p>Run the command:
+</p><div class="example">
+<pre class="example">ffmpeg -opencl_bench
+</pre></div>
+<p>Note down the platform ID (<var>pidx</var>) and device ID (<var>didx</var>) of the first
+i.e. fastest device in the list.
+Select the platform and device using the command:
+</p><div class="example">
+<pre class="example">ffmpeg -opencl_options platform_idx=<var>pidx</var>:device_idx=<var>didx</var> ...
+</pre></div>
+
+</dd>
+<dt><samp>-opencl_options options (<em>global</em>)</samp></dt>
+<dd><p>Set OpenCL environment options. This option is only available when
+FFmpeg has been compiled with <code>--enable-opencl</code>.
+</p>
+<p><var>options</var> must be a list of <var>key</var>=<var>value</var> option pairs
+separated by &rsquo;:&rsquo;. See the &ldquo;OpenCL Options&rdquo; section in the
+ffmpeg-utils manual for the list of supported options.
+</p></dd>
+</dl>
+
+<a name="AVOptions"></a>
+<h3 class="section">3.3 AVOptions<span class="pull-right"><a class="anchor hidden-xs" href="#AVOptions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-AVOptions" aria-hidden="true">TOC</a></span></h3>
+
+<p>These options are provided directly by the libavformat, libavdevice and
+libavcodec libraries. To see the list of available AVOptions, use the
+<samp>-help</samp> option. They are separated into two categories:
+</p><dl compact="compact">
+<dt><samp>generic</samp></dt>
+<dd><p>These options can be set for any container, codec or device. Generic options
+are listed under AVFormatContext options for containers/devices and under
+AVCodecContext options for codecs.
+</p></dd>
+<dt><samp>private</samp></dt>
+<dd><p>These options are specific to the given container, device or codec. Private
+options are listed under their corresponding containers/devices/codecs.
+</p></dd>
+</dl>
+
+<p>For example to write an ID3v2.3 header instead of a default ID3v2.4 to
+an MP3 file, use the <samp>id3v2_version</samp> private option of the MP3
+muxer:
+</p><div class="example">
+<pre class="example">ffmpeg -i input.flac -id3v2_version 3 out.mp3
+</pre></div>
+
+<p>All codec AVOptions are per-stream, and thus a stream specifier
+should be attached to them.
+</p>
+<p>Note: the <samp>-nooption</samp> syntax cannot be used for boolean
+AVOptions, use <samp>-option 0</samp>/<samp>-option 1</samp>.
+</p>
+<p>Note: the old undocumented way of specifying per-stream AVOptions by
+prepending v/a/s to the options name is now obsolete and will be
+removed soon.
+</p>
+<a name="Main-options"></a>
+<h3 class="section">3.4 Main options<span class="pull-right"><a class="anchor hidden-xs" href="#Main-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Main-options" aria-hidden="true">TOC</a></span></h3>
+
+<dl compact="compact">
+<dt><samp>-x <var>width</var></samp></dt>
+<dd><p>Force displayed width.
+</p></dd>
+<dt><samp>-y <var>height</var></samp></dt>
+<dd><p>Force displayed height.
+</p></dd>
+<dt><samp>-s <var>size</var></samp></dt>
+<dd><p>Set frame size (WxH or abbreviation), needed for videos which do
+not contain a header with the frame size like raw YUV.  This option
+has been deprecated in favor of private options, try -video_size.
+</p></dd>
+<dt><samp>-fs</samp></dt>
+<dd><p>Start in fullscreen mode.
+</p></dd>
+<dt><samp>-an</samp></dt>
+<dd><p>Disable audio.
+</p></dd>
+<dt><samp>-vn</samp></dt>
+<dd><p>Disable video.
+</p></dd>
+<dt><samp>-sn</samp></dt>
+<dd><p>Disable subtitles.
+</p></dd>
+<dt><samp>-ss <var>pos</var></samp></dt>
+<dd><p>Seek to <var>pos</var>. Note that in most formats it is not possible to seek
+exactly, so <code>ffplay</code> will seek to the nearest seek point to
+<var>pos</var>.
+</p>
+<p><var>pos</var> must be a time duration specification,
+see <a href="ffmpeg-utils.html#time-duration-syntax">(ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual</a>.
+</p></dd>
+<dt><samp>-t <var>duration</var></samp></dt>
+<dd><p>Play <var>duration</var> seconds of audio/video.
+</p>
+<p><var>duration</var> must be a time duration specification,
+see <a href="ffmpeg-utils.html#time-duration-syntax">(ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual</a>.
+</p></dd>
+<dt><samp>-bytes</samp></dt>
+<dd><p>Seek by bytes.
+</p></dd>
+<dt><samp>-nodisp</samp></dt>
+<dd><p>Disable graphical display.
+</p></dd>
+<dt><samp>-f <var>fmt</var></samp></dt>
+<dd><p>Force format.
+</p></dd>
+<dt><samp>-window_title <var>title</var></samp></dt>
+<dd><p>Set window title (default is the input filename).
+</p></dd>
+<dt><samp>-loop <var>number</var></samp></dt>
+<dd><p>Loops movie playback &lt;number&gt; times. 0 means forever.
+</p></dd>
+<dt><samp>-showmode <var>mode</var></samp></dt>
+<dd><p>Set the show mode to use.
+Available values for <var>mode</var> are:
+</p><dl compact="compact">
+<dt>&lsquo;<samp>0, video</samp>&rsquo;</dt>
+<dd><p>show video
+</p></dd>
+<dt>&lsquo;<samp>1, waves</samp>&rsquo;</dt>
+<dd><p>show audio waves
+</p></dd>
+<dt>&lsquo;<samp>2, rdft</samp>&rsquo;</dt>
+<dd><p>show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
+</p></dd>
+</dl>
+
+<p>Default value is &quot;video&quot;, if video is not present or cannot be played
+&quot;rdft&quot; is automatically selected.
+</p>
+<p>You can interactively cycle through the available show modes by
+pressing the key <tt class="key">w</tt>.
+</p>
+</dd>
+<dt><samp>-vf <var>filtergraph</var></samp></dt>
+<dd><p>Create the filtergraph specified by <var>filtergraph</var> and use it to
+filter the video stream.
+</p>
+<p><var>filtergraph</var> is a description of the filtergraph to apply to
+the stream, and must have a single video input and a single video
+output. In the filtergraph, the input is associated to the label
+<code>in</code>, and the output to the label <code>out</code>. See the
+ffmpeg-filters manual for more information about the filtergraph
+syntax.
+</p>
+<p>You can specify this parameter multiple times and cycle through the specified
+filtergraphs along with the show modes by pressing the key <tt class="key">w</tt>.
+</p>
+</dd>
+<dt><samp>-af <var>filtergraph</var></samp></dt>
+<dd><p><var>filtergraph</var> is a description of the filtergraph to apply to
+the input audio.
+Use the option &quot;-filters&quot; to show all the available filters (including
+sources and sinks).
+</p>
+</dd>
+<dt><samp>-i <var>input_file</var></samp></dt>
+<dd><p>Read <var>input_file</var>.
+</p></dd>
+</dl>
+
+<a name="Advanced-options"></a>
+<h3 class="section">3.5 Advanced options<span class="pull-right"><a class="anchor hidden-xs" href="#Advanced-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Advanced-options" aria-hidden="true">TOC</a></span></h3>
+<dl compact="compact">
+<dt><samp>-pix_fmt <var>format</var></samp></dt>
+<dd><p>Set pixel format.
+This option has been deprecated in favor of private options, try -pixel_format.
+</p>
+</dd>
+<dt><samp>-stats</samp></dt>
+<dd><p>Print several playback statistics, in particular show the stream
+duration, the codec parameters, the current position in the stream and
+the audio/video synchronisation drift. It is on by default, to
+explicitly disable it you need to specify <code>-nostats</code>.
+</p>
+</dd>
+<dt><samp>-fast</samp></dt>
+<dd><p>Non-spec-compliant optimizations.
+</p></dd>
+<dt><samp>-genpts</samp></dt>
+<dd><p>Generate pts.
+</p></dd>
+<dt><samp>-sync <var>type</var></samp></dt>
+<dd><p>Set the master clock to audio (<code>type=audio</code>), video
+(<code>type=video</code>) or external (<code>type=ext</code>). Default is audio. The
+master clock is used to control audio-video synchronization. Most media
+players use audio as master clock, but in some cases (streaming or high
+quality broadcast) it is necessary to change that. This option is mainly
+used for debugging purposes.
+</p></dd>
+<dt><samp>-ast <var>audio_stream_specifier</var></samp></dt>
+<dd><p>Select the desired audio stream using the given stream specifier. The stream
+specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
+is not specified, the &quot;best&quot; audio stream is selected in the program of the
+already selected video stream.
+</p></dd>
+<dt><samp>-vst <var>video_stream_specifier</var></samp></dt>
+<dd><p>Select the desired video stream using the given stream specifier. The stream
+specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
+is not specified, the &quot;best&quot; video stream is selected.
+</p></dd>
+<dt><samp>-sst <var>subtitle_stream_specifier</var></samp></dt>
+<dd><p>Select the desired subtitle stream using the given stream specifier. The stream
+specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
+is not specified, the &quot;best&quot; subtitle stream is selected in the program of the
+already selected video or audio stream.
+</p></dd>
+<dt><samp>-autoexit</samp></dt>
+<dd><p>Exit when video is done playing.
+</p></dd>
+<dt><samp>-exitonkeydown</samp></dt>
+<dd><p>Exit if any key is pressed.
+</p></dd>
+<dt><samp>-exitonmousedown</samp></dt>
+<dd><p>Exit if any mouse button is pressed.
+</p>
+</dd>
+<dt><samp>-codec:<var>media_specifier</var> <var>codec_name</var></samp></dt>
+<dd><p>Force a specific decoder implementation for the stream identified by
+<var>media_specifier</var>, which can assume the values <code>a</code> (audio),
+<code>v</code> (video), and <code>s</code> subtitle.
+</p>
+</dd>
+<dt><samp>-acodec <var>codec_name</var></samp></dt>
+<dd><p>Force a specific audio decoder.
+</p>
+</dd>
+<dt><samp>-vcodec <var>codec_name</var></samp></dt>
+<dd><p>Force a specific video decoder.
+</p>
+</dd>
+<dt><samp>-scodec <var>codec_name</var></samp></dt>
+<dd><p>Force a specific subtitle decoder.
+</p>
+</dd>
+<dt><samp>-autorotate</samp></dt>
+<dd><p>Automatically rotate the video according to file metadata. Enabled by
+default, use <samp>-noautorotate</samp> to disable it.
+</p>
+</dd>
+<dt><samp>-framedrop</samp></dt>
+<dd><p>Drop video frames if video is out of sync. Enabled by default if the master
+clock is not set to video. Use this option to enable frame dropping for all
+master clock sources, use <samp>-noframedrop</samp> to disable it.
+</p>
+</dd>
+<dt><samp>-infbuf</samp></dt>
+<dd><p>Do not limit the input buffer size, read as much data as possible from the
+input as soon as possible. Enabled by default for realtime streams, where data
+may be dropped if not read in time. Use this option to enable infinite buffers
+for all inputs, use <samp>-noinfbuf</samp> to disable it.
+</p>
+</dd>
+</dl>
+
+<a name="While-playing"></a>
+<h3 class="section">3.6 While playing<span class="pull-right"><a class="anchor hidden-xs" href="#While-playing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-While-playing" aria-hidden="true">TOC</a></span></h3>
+
+<dl compact="compact">
+<dt><tt class="key">q, ESC</tt></dt>
+<dd><p>Quit.
+</p>
+</dd>
+<dt><tt class="key">f</tt></dt>
+<dd><p>Toggle full screen.
+</p>
+</dd>
+<dt><tt class="key">p, SPC</tt></dt>
+<dd><p>Pause.
+</p>
+</dd>
+<dt><tt class="key">m</tt></dt>
+<dd><p>Toggle mute.
+</p>
+</dd>
+<dt><tt class="key">9, 0</tt></dt>
+<dd><p>Decrease and increase volume respectively.
+</p>
+</dd>
+<dt><tt class="key">/, *</tt></dt>
+<dd><p>Decrease and increase volume respectively.
+</p>
+</dd>
+<dt><tt class="key">a</tt></dt>
+<dd><p>Cycle audio channel in the current program.
+</p>
+</dd>
+<dt><tt class="key">v</tt></dt>
+<dd><p>Cycle video channel.
+</p>
+</dd>
+<dt><tt class="key">t</tt></dt>
+<dd><p>Cycle subtitle channel in the current program.
+</p>
+</dd>
+<dt><tt class="key">c</tt></dt>
+<dd><p>Cycle program.
+</p>
+</dd>
+<dt><tt class="key">w</tt></dt>
+<dd><p>Cycle video filters or show modes.
+</p>
+</dd>
+<dt><tt class="key">s</tt></dt>
+<dd><p>Step to the next frame.
+</p>
+<p>Pause if the stream is not already paused, step to the next video
+frame, and pause.
+</p>
+</dd>
+<dt><tt class="key">left/right</tt></dt>
+<dd><p>Seek backward/forward 10 seconds.
+</p>
+</dd>
+<dt><tt class="key">down/up</tt></dt>
+<dd><p>Seek backward/forward 1 minute.
+</p>
+</dd>
+<dt><tt class="key">page down/page up</tt></dt>
+<dd><p>Seek to the previous/next chapter.
+or if there are no chapters
+Seek backward/forward 10 minutes.
+</p>
+</dd>
+<dt><tt class="key">mouse click</tt></dt>
+<dd><p>Seek to percentage in file corresponding to fraction of width.
+</p>
+</dd>
+</dl>
+
+
+
+<a name="See-Also"></a>
+<h2 class="chapter">4 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffplay-all.html">ffmpeg-all</a>,
+<a href="ffmpeg.html">ffmpeg</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="ffmpeg-utils.html">ffmpeg-utils</a>,
+<a href="ffmpeg-scaler.html">ffmpeg-scaler</a>,
+<a href="ffmpeg-resampler.html">ffmpeg-resampler</a>,
+<a href="ffmpeg-codecs.html">ffmpeg-codecs</a>,
+<a href="ffmpeg-bitstream-filters.html">ffmpeg-bitstream-filters</a>,
+<a href="ffmpeg-formats.html">ffmpeg-formats</a>,
+<a href="ffmpeg-devices.html">ffmpeg-devices</a>,
+<a href="ffmpeg-protocols.html">ffmpeg-protocols</a>,
+<a href="ffmpeg-filters.html">ffmpeg-filters</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">5 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,497 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Using Git to develop FFmpeg
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Using Git to develop FFmpeg
+      </h1>
+<div align="center">
+</div>
+
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Introduction" href="#Introduction">1 Introduction</a></li>
+  <li><a name="toc-Basics-Usage" href="#Basics-Usage">2 Basics Usage</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Get-Git" href="#Get-Git">2.1 Get Git</a></li>
+    <li><a name="toc-Cloning-the-source-tree" href="#Cloning-the-source-tree">2.2 Cloning the source tree</a></li>
+    <li><a name="toc-Updating-the-source-tree-to-the-latest-revision-1" href="#Updating-the-source-tree-to-the-latest-revision-1">2.3 Updating the source tree to the latest revision</a></li>
+    <li><a name="toc-Rebasing-your-local-branches" href="#Rebasing-your-local-branches">2.4 Rebasing your local branches</a></li>
+    <li><a name="toc-Adding_002fremoving-files_002fdirectories" href="#Adding_002fremoving-files_002fdirectories">2.5 Adding/removing files/directories</a></li>
+    <li><a name="toc-Showing-modifications" href="#Showing-modifications">2.6 Showing modifications</a></li>
+    <li><a name="toc-Inspecting-the-changelog" href="#Inspecting-the-changelog">2.7 Inspecting the changelog</a></li>
+    <li><a name="toc-Checking-source-tree-status" href="#Checking-source-tree-status">2.8 Checking source tree status</a></li>
+    <li><a name="toc-Committing" href="#Committing">2.9 Committing</a></li>
+    <li><a name="toc-Preparing-a-patchset" href="#Preparing-a-patchset">2.10 Preparing a patchset</a></li>
+    <li><a name="toc-Sending-patches-for-review" href="#Sending-patches-for-review">2.11 Sending patches for review</a></li>
+    <li><a name="toc-Renaming_002fmoving_002fcopying-files-or-contents-of-files" href="#Renaming_002fmoving_002fcopying-files-or-contents-of-files">2.12 Renaming/moving/copying files or contents of files</a></li>
+  </ul></li>
+  <li><a name="toc-Git-configuration" href="#Git-configuration">3 Git configuration</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Personal-Git-installation" href="#Personal-Git-installation">3.1 Personal Git installation</a></li>
+    <li><a name="toc-Repository-configuration" href="#Repository-configuration">3.2 Repository configuration</a></li>
+  </ul></li>
+  <li><a name="toc-FFmpeg-specific" href="#FFmpeg-specific">4 FFmpeg specific</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Reverting-broken-commits" href="#Reverting-broken-commits">4.1 Reverting broken commits</a></li>
+    <li><a name="toc-Pushing-changes-to-remote-trees" href="#Pushing-changes-to-remote-trees">4.2 Pushing changes to remote trees</a></li>
+    <li><a name="toc-Finding-a-specific-svn-revision" href="#Finding-a-specific-svn-revision">4.3 Finding a specific svn revision</a></li>
+  </ul></li>
+  <li><a name="toc-Pre_002dpush-checklist" href="#Pre_002dpush-checklist">5 Pre-push checklist</a></li>
+  <li><a name="toc-Server-Issues" href="#Server-Issues">6 Server Issues</a></li>
+</ul>
+</div>
+
+
+<a name="Introduction"></a>
+<h2 class="chapter">1 Introduction<span class="pull-right"><a class="anchor hidden-xs" href="#Introduction" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Introduction" aria-hidden="true">TOC</a></span></h2>
+
+<p>This document aims in giving some quick references on a set of useful Git
+commands. You should always use the extensive and detailed documentation
+provided directly by Git:
+</p>
+<div class="example">
+<pre class="example">git --help
+man git
+</pre></div>
+
+<p>shows you the available subcommands,
+</p>
+<div class="example">
+<pre class="example">git &lt;command&gt; --help
+man git-&lt;command&gt;
+</pre></div>
+
+<p>shows information about the subcommand &lt;command&gt;.
+</p>
+<p>Additional information could be found on the
+<a href="http://gitref.org">Git Reference</a> website.
+</p>
+<p>For more information about the Git project, visit the
+<a href="http://git-scm.com/">Git website</a>.
+</p>
+<p>Consult these resources whenever you have problems, they are quite exhaustive.
+</p>
+<p>What follows now is a basic introduction to Git and some FFmpeg-specific
+guidelines to ease the contribution to the project.
+</p>
+<a name="Basics-Usage"></a>
+<h2 class="chapter">2 Basics Usage<span class="pull-right"><a class="anchor hidden-xs" href="#Basics-Usage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Basics-Usage" aria-hidden="true">TOC</a></span></h2>
+
+<a name="Get-Git"></a>
+<h3 class="section">2.1 Get Git<span class="pull-right"><a class="anchor hidden-xs" href="#Get-Git" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Get-Git" aria-hidden="true">TOC</a></span></h3>
+
+<p>You can get Git from <a href="http://git-scm.com/">http://git-scm.com/</a>
+Most distribution and operating system provide a package for it.
+</p>
+
+<a name="Cloning-the-source-tree"></a>
+<h3 class="section">2.2 Cloning the source tree<span class="pull-right"><a class="anchor hidden-xs" href="#Cloning-the-source-tree" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Cloning-the-source-tree" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git clone git://source.ffmpeg.org/ffmpeg &lt;target&gt;
+</pre></div>
+
+<p>This will put the FFmpeg sources into the directory <var>&lt;target&gt;</var>.
+</p>
+<div class="example">
+<pre class="example">git clone git@source.ffmpeg.org:ffmpeg &lt;target&gt;
+</pre></div>
+
+<p>This will put the FFmpeg sources into the directory <var>&lt;target&gt;</var> and let
+you push back your changes to the remote repository.
+</p>
+<p>Make sure that you do not have Windows line endings in your checkouts,
+otherwise you may experience spurious compilation failures. One way to
+achieve this is to run
+</p>
+<div class="example">
+<pre class="example">git config --global core.autocrlf false
+</pre></div>
+
+
+<a name="Updating-the-source-tree-to-the-latest-revision"></a><a name="Updating-the-source-tree-to-the-latest-revision-1"></a>
+<h3 class="section">2.3 Updating the source tree to the latest revision<span class="pull-right"><a class="anchor hidden-xs" href="#Updating-the-source-tree-to-the-latest-revision-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Updating-the-source-tree-to-the-latest-revision-1" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git pull (--rebase)
+</pre></div>
+
+<p>pulls in the latest changes from the tracked branch. The tracked branch
+can be remote. By default the master branch tracks the branch master in
+the remote origin.
+</p>
+<div class="warning">
+<p><code>--rebase</code> (see below) is recommended.
+</p></div>
+<a name="Rebasing-your-local-branches"></a>
+<h3 class="section">2.4 Rebasing your local branches<span class="pull-right"><a class="anchor hidden-xs" href="#Rebasing-your-local-branches" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Rebasing-your-local-branches" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git pull --rebase
+</pre></div>
+
+<p>fetches the changes from the main repository and replays your local commits
+over it. This is required to keep all your local changes at the top of
+FFmpeg&rsquo;s master tree. The master tree will reject pushes with merge commits.
+</p>
+
+<a name="Adding_002fremoving-files_002fdirectories"></a>
+<h3 class="section">2.5 Adding/removing files/directories<span class="pull-right"><a class="anchor hidden-xs" href="#Adding_002fremoving-files_002fdirectories" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Adding_002fremoving-files_002fdirectories" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git add [-A] &lt;filename/dirname&gt;
+git rm [-r] &lt;filename/dirname&gt;
+</pre></div>
+
+<p>Git needs to get notified of all changes you make to your working
+directory that makes files appear or disappear.
+Line moves across files are automatically tracked.
+</p>
+
+<a name="Showing-modifications"></a>
+<h3 class="section">2.6 Showing modifications<span class="pull-right"><a class="anchor hidden-xs" href="#Showing-modifications" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Showing-modifications" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git diff &lt;filename(s)&gt;
+</pre></div>
+
+<p>will show all local modifications in your working directory as unified diff.
+</p>
+
+<a name="Inspecting-the-changelog"></a>
+<h3 class="section">2.7 Inspecting the changelog<span class="pull-right"><a class="anchor hidden-xs" href="#Inspecting-the-changelog" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Inspecting-the-changelog" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git log &lt;filename(s)&gt;
+</pre></div>
+
+<p>You may also use the graphical tools like <code>gitview</code> or <code>gitk</code>
+or the web interface available at <a href="http://source.ffmpeg.org/">http://source.ffmpeg.org/</a>.
+</p>
+<a name="Checking-source-tree-status"></a>
+<h3 class="section">2.8 Checking source tree status<span class="pull-right"><a class="anchor hidden-xs" href="#Checking-source-tree-status" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Checking-source-tree-status" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git status
+</pre></div>
+
+<p>detects all the changes you made and lists what actions will be taken in case
+of a commit (additions, modifications, deletions, etc.).
+</p>
+
+<a name="Committing"></a>
+<h3 class="section">2.9 Committing<span class="pull-right"><a class="anchor hidden-xs" href="#Committing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Committing" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git diff --check
+</pre></div>
+
+<p>to double check your changes before committing them to avoid trouble later
+on. All experienced developers do this on each and every commit, no matter
+how small.
+</p>
+<p>Every one of them has been saved from looking like a fool by this many times.
+It&rsquo;s very easy for stray debug output or cosmetic modifications to slip in,
+please avoid problems through this extra level of scrutiny.
+</p>
+<p>For cosmetics-only commits you should get (almost) empty output from
+</p>
+<div class="example">
+<pre class="example">git diff -w -b &lt;filename(s)&gt;
+</pre></div>
+
+<p>Also check the output of
+</p>
+<div class="example">
+<pre class="example">git status
+</pre></div>
+
+<p>to make sure you don&rsquo;t have untracked files or deletions.
+</p>
+<div class="example">
+<pre class="example">git add [-i|-p|-A] &lt;filenames/dirnames&gt;
+</pre></div>
+
+<p>Make sure you have told Git your name and email address
+</p>
+<div class="example">
+<pre class="example">git config --global user.name &quot;My Name&quot;
+git config --global user.email my@email.invalid
+</pre></div>
+
+<p>Use <samp>--global</samp> to set the global configuration for all your Git checkouts.
+</p>
+<p>Git will select the changes to the files for commit. Optionally you can use
+the interactive or the patch mode to select hunk by hunk what should be
+added to the commit.
+</p>
+
+<div class="example">
+<pre class="example">git commit
+</pre></div>
+
+<p>Git will commit the selected changes to your current local branch.
+</p>
+<p>You will be prompted for a log message in an editor, which is either
+set in your personal configuration file through
+</p>
+<div class="example">
+<pre class="example">git config --global core.editor
+</pre></div>
+
+<p>or set by one of the following environment variables:
+<var>GIT_EDITOR</var>, <var>VISUAL</var> or <var>EDITOR</var>.
+</p>
+<p>Log messages should be concise but descriptive. Explain why you made a change,
+what you did will be obvious from the changes themselves most of the time.
+Saying just &quot;bug fix&quot; or &quot;10l&quot; is bad. Remember that people of varying skill
+levels look at and educate themselves while reading through your code. Don&rsquo;t
+include filenames in log messages, Git provides that information.
+</p>
+<p>Possibly make the commit message have a terse, descriptive first line, an
+empty line and then a full description. The first line will be used to name
+the patch by <code>git format-patch</code>.
+</p>
+<a name="Preparing-a-patchset"></a>
+<h3 class="section">2.10 Preparing a patchset<span class="pull-right"><a class="anchor hidden-xs" href="#Preparing-a-patchset" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Preparing-a-patchset" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git format-patch &lt;commit&gt; [-o directory]
+</pre></div>
+
+<p>will generate a set of patches for each commit between <var>&lt;commit&gt;</var> and
+current <var>HEAD</var>. E.g.
+</p>
+<div class="example">
+<pre class="example">git format-patch origin/master
+</pre></div>
+
+<p>will generate patches for all commits on current branch which are not
+present in upstream.
+A useful shortcut is also
+</p>
+<div class="example">
+<pre class="example">git format-patch -n
+</pre></div>
+
+<p>which will generate patches from last <var>n</var> commits.
+By default the patches are created in the current directory.
+</p>
+<a name="Sending-patches-for-review"></a>
+<h3 class="section">2.11 Sending patches for review<span class="pull-right"><a class="anchor hidden-xs" href="#Sending-patches-for-review" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Sending-patches-for-review" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git send-email &lt;commit list|directory&gt;
+</pre></div>
+
+<p>will send the patches created by <code>git format-patch</code> or directly
+generates them. All the email fields can be configured in the global/local
+configuration or overridden by command line.
+Note that this tool must often be installed separately (e.g. <var>git-email</var>
+package on Debian-based distros).
+</p>
+
+<a name="Renaming_002fmoving_002fcopying-files-or-contents-of-files"></a>
+<h3 class="section">2.12 Renaming/moving/copying files or contents of files<span class="pull-right"><a class="anchor hidden-xs" href="#Renaming_002fmoving_002fcopying-files-or-contents-of-files" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Renaming_002fmoving_002fcopying-files-or-contents-of-files" aria-hidden="true">TOC</a></span></h3>
+
+<p>Git automatically tracks such changes, making those normal commits.
+</p>
+<div class="example">
+<pre class="example">mv/cp path/file otherpath/otherfile
+git add [-A] .
+git commit
+</pre></div>
+
+
+<a name="Git-configuration"></a>
+<h2 class="chapter">3 Git configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Git-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Git-configuration" aria-hidden="true">TOC</a></span></h2>
+
+<p>In order to simplify a few workflows, it is advisable to configure both
+your personal Git installation and your local FFmpeg repository.
+</p>
+<a name="Personal-Git-installation"></a>
+<h3 class="section">3.1 Personal Git installation<span class="pull-right"><a class="anchor hidden-xs" href="#Personal-Git-installation" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Personal-Git-installation" aria-hidden="true">TOC</a></span></h3>
+
+<p>Add the following to your <samp>~/.gitconfig</samp> to help <code>git send-email</code>
+and <code>git format-patch</code> detect renames:
+</p>
+<div class="example">
+<pre class="example">[diff]
+        renames = copy
+</pre></div>
+
+<a name="Repository-configuration"></a>
+<h3 class="section">3.2 Repository configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Repository-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Repository-configuration" aria-hidden="true">TOC</a></span></h3>
+
+<p>In order to have <code>git send-email</code> automatically send patches
+to the ffmpeg-devel mailing list, add the following stanza
+to <samp>/path/to/ffmpeg/repository/.git/config</samp>:
+</p>
+<div class="example">
+<pre class="example">[sendemail]
+        to = ffmpeg-devel@ffmpeg.org
+</pre></div>
+
+<a name="FFmpeg-specific"></a>
+<h2 class="chapter">4 FFmpeg specific<span class="pull-right"><a class="anchor hidden-xs" href="#FFmpeg-specific" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FFmpeg-specific" aria-hidden="true">TOC</a></span></h2>
+
+<a name="Reverting-broken-commits"></a>
+<h3 class="section">4.1 Reverting broken commits<span class="pull-right"><a class="anchor hidden-xs" href="#Reverting-broken-commits" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Reverting-broken-commits" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git reset &lt;commit&gt;
+</pre></div>
+
+<p><code>git reset</code> will uncommit the changes till <var>&lt;commit&gt;</var> rewriting
+the current branch history.
+</p>
+<div class="example">
+<pre class="example">git commit --amend
+</pre></div>
+
+<p>allows one to amend the last commit details quickly.
+</p>
+<div class="example">
+<pre class="example">git rebase -i origin/master
+</pre></div>
+
+<p>will replay local commits over the main repository allowing to edit, merge
+or remove some of them in the process.
+</p>
+<div class="info">
+<p><code>git reset</code>, <code>git commit --amend</code> and <code>git rebase</code>
+rewrite history, so you should use them ONLY on your local or topic branches.
+The main repository will reject those changes.
+</p></div>
+<div class="example">
+<pre class="example">git revert &lt;commit&gt;
+</pre></div>
+
+<p><code>git revert</code> will generate a revert commit. This will not make the
+faulty commit disappear from the history.
+</p>
+<a name="Pushing-changes-to-remote-trees"></a>
+<h3 class="section">4.2 Pushing changes to remote trees<span class="pull-right"><a class="anchor hidden-xs" href="#Pushing-changes-to-remote-trees" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Pushing-changes-to-remote-trees" aria-hidden="true">TOC</a></span></h3>
+
+<div class="example">
+<pre class="example">git push origin master --dry-run
+</pre></div>
+
+<p>Will simulate a push of the local master branch to the default remote
+(<var>origin</var>). And list which branches and ranges or commits would have been
+pushed.
+Git will prevent you from pushing changes if the local and remote trees are
+out of sync. Refer to <a href="#Updating-the-source-tree-to-the-latest-revision">Updating the source tree to the latest revision</a>.
+</p>
+<div class="example">
+<pre class="example">git remote add &lt;name&gt; &lt;url&gt;
+</pre></div>
+
+<p>Will add additional remote with a name reference, it is useful if you want
+to push your local branch for review on a remote host.
+</p>
+<div class="example">
+<pre class="example">git push &lt;remote&gt; &lt;refspec&gt;
+</pre></div>
+
+<p>Will push the changes to the <var>&lt;remote&gt;</var> repository.
+Omitting <var>&lt;refspec&gt;</var> makes <code>git push</code> update all the remote
+branches matching the local ones.
+</p>
+<a name="Finding-a-specific-svn-revision"></a>
+<h3 class="section">4.3 Finding a specific svn revision<span class="pull-right"><a class="anchor hidden-xs" href="#Finding-a-specific-svn-revision" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Finding-a-specific-svn-revision" aria-hidden="true">TOC</a></span></h3>
+
+<p>Since version 1.7.1 Git supports &lsquo;<samp>:/foo</samp>&rsquo; syntax for specifying commits
+based on a regular expression. see man gitrevisions
+</p>
+<div class="example">
+<pre class="example">git show :/'as revision 23456'
+</pre></div>
+
+<p>will show the svn changeset &lsquo;<samp>r23456</samp>&rsquo;. With older Git versions searching in
+the <code>git log</code> output is the easiest option (especially if a pager with
+search capabilities is used).
+</p>
+<p>This commit can be checked out with
+</p>
+<div class="example">
+<pre class="example">git checkout -b svn_23456 :/'as revision 23456'
+</pre></div>
+
+<p>or for Git &lt; 1.7.1 with
+</p>
+<div class="example">
+<pre class="example">git checkout -b svn_23456 $SHA1
+</pre></div>
+
+<p>where <var>$SHA1</var> is the commit hash from the <code>git log</code> output.
+</p>
+
+<a name="Pre_002dpush-checklist"></a>
+<h2 class="chapter">5 Pre-push checklist<span class="pull-right"><a class="anchor hidden-xs" href="#Pre_002dpush-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Pre_002dpush-checklist" aria-hidden="true">TOC</a></span></h2>
+
+<p>Once you have a set of commits that you feel are ready for pushing,
+work through the following checklist to doublecheck everything is in
+proper order. This list tries to be exhaustive. In case you are just
+pushing a typo in a comment, some of the steps may be unnecessary.
+Apply your common sense, but if in doubt, err on the side of caution.
+</p>
+<p>First, make sure that the commits and branches you are going to push
+match what you want pushed and that nothing is missing, extraneous or
+wrong. You can see what will be pushed by running the git push command
+with <samp>--dry-run</samp> first. And then inspecting the commits listed with
+<code>git log -p 1234567..987654</code>. The <code>git status</code> command
+may help in finding local changes that have been forgotten to be added.
+</p>
+<p>Next let the code pass through a full run of our testsuite.
+</p>
+<ul>
+<li> <code>make distclean</code>
+</li><li> <code>/path/to/ffmpeg/configure</code>
+</li><li> <code>make fate</code>
+</li><li> if fate fails due to missing samples run <code>make fate-rsync</code> and retry
+</li></ul>
+
+<p>Make sure all your changes have been checked before pushing them, the
+testsuite only checks against regressions and that only to some extend. It does
+obviously not check newly added features/code to be working unless you have
+added a test for that (which is recommended).
+</p>
+<p>Also note that every single commit should pass the test suite, not just
+the result of a series of patches.
+</p>
+<p>Once everything passed, push the changes to your public ffmpeg clone and post a
+merge request to ffmpeg-devel. You can also push them directly but this is not
+recommended.
+</p>
+<a name="Server-Issues"></a>
+<h2 class="chapter">6 Server Issues<span class="pull-right"><a class="anchor hidden-xs" href="#Server-Issues" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Server-Issues" aria-hidden="true">TOC</a></span></h2>
+
+<p>Contact the project admins at <a href="mailto:root@ffmpeg.org">root@ffmpeg.org</a> if you have technical
+problems with the Git server.
+</p>
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,77 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Libavcodec Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Libavcodec Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The libavcodec library provides a generic encoding/decoding framework
+and contains multiple decoders and encoders for audio, video and
+subtitle streams, and several bitstream filters.
+</p>
+<p>The shared architecture provides various services ranging from bit
+stream I/O to DSP optimizations, and makes it suitable for
+implementing robust and fast codecs as well as for experimentation.
+</p>
+
+<a name="See-Also"></a>
+<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="ffmpeg-codecs.html">ffmpeg-codecs</a>, <a href="ffmpeg-bitstream-filters.html">bitstream-filters</a>,
+<a href="libavutil.html">libavutil</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,74 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Libavdevice Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Libavdevice Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The libavdevice library provides a generic framework for grabbing from
+and rendering to many common multimedia input/output devices, and
+supports several input and output devices, including Video4Linux2,
+VfW, DShow, and ALSA.
+</p>
+
+<a name="See-Also"></a>
+<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="ffmpeg-devices.html">ffmpeg-devices</a>,
+<a href="libavutil.html">libavutil</a>, <a href="libavcodec.html">libavcodec</a>, <a href="libavformat.html">libavformat</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,73 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Libavfilter Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Libavfilter Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The libavfilter library provides a generic audio/video filtering
+framework containing several filters, sources and sinks.
+</p>
+
+<a name="See-Also"></a>
+<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="ffmpeg-filters.html">ffmpeg-filters</a>,
+<a href="libavutil.html">libavutil</a>, <a href="libswscale.html">libswscale</a>, <a href="libswresample.html">libswresample</a>,
+<a href="libavcodec.html">libavcodec</a>, <a href="libavformat.html">libavformat</a>, <a href="libavdevice.html">libavdevice</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,77 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Libavformat Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Libavformat Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The libavformat library provides a generic framework for multiplexing
+and demultiplexing (muxing and demuxing) audio, video and subtitle
+streams. It encompasses multiple muxers and demuxers for multimedia
+container formats.
+</p>
+<p>It also supports several input and output protocols to access a media
+resource.
+</p>
+
+<a name="See-Also"></a>
+<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="ffmpeg-formats.html">ffmpeg-formats</a>, <a href="ffmpeg-protocols.html">ffmpeg-protocols</a>,
+<a href="libavutil.html">libavutil</a>, <a href="libavcodec.html">libavcodec</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,96 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Libavutil Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Libavutil Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The libavutil library is a utility library to aid portable
+multimedia programming. It contains safe portable string functions,
+random number generators, data structures, additional mathematics
+functions, cryptography and multimedia related functionality (like
+enumerations for pixel and sample formats). It is not a library for
+code needed by both libavcodec and libavformat.
+</p>
+<p>The goals for this library is to be:
+</p>
+<dl compact="compact">
+<dt><strong>Modular</strong></dt>
+<dd><p>It should have few interdependencies and the possibility of disabling individual
+parts during <code>./configure</code>.
+</p>
+</dd>
+<dt><strong>Small</strong></dt>
+<dd><p>Both sources and objects should be small.
+</p>
+</dd>
+<dt><strong>Efficient</strong></dt>
+<dd><p>It should have low CPU and memory usage.
+</p>
+</dd>
+<dt><strong>Useful</strong></dt>
+<dd><p>It should avoid useless features that almost no one needs.
+</p></dd>
+</dl>
+
+
+<a name="See-Also"></a>
+<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="ffmpeg-utils.html">ffmpeg-utils</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,96 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Libswresample Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Libswresample Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The libswresample library performs highly optimized audio resampling,
+rematrixing and sample format conversion operations.
+</p>
+<p>Specifically, this library performs the following conversions:
+</p>
+<ul>
+<li> <em>Resampling</em>: is the process of changing the audio rate, for
+example from a high sample rate of 44100Hz to 8000Hz. Audio
+conversion from high to low sample rate is a lossy process. Several
+resampling options and algorithms are available.
+
+</li><li> <em>Format conversion</em>: is the process of converting the type of
+samples, for example from 16-bit signed samples to unsigned 8-bit or
+float samples. It also handles packing conversion, when passing from
+packed layout (all samples belonging to distinct channels interleaved
+in the same buffer), to planar layout (all samples belonging to the
+same channel stored in a dedicated buffer or &quot;plane&quot;).
+
+</li><li> <em>Rematrixing</em>: is the process of changing the channel layout, for
+example from stereo to mono. When the input channels cannot be mapped
+to the output streams, the process is lossy, since it involves
+different gain factors and mixing.
+</li></ul>
+
+<p>Various other audio conversions (e.g. stretching and padding) are
+enabled through dedicated options.
+</p>
+
+<a name="See-Also"></a>
+<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="ffmpeg-resampler.html">ffmpeg-resampler</a>,
+<a href="libavutil.html">libavutil</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,90 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Libswscale Documentation
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Libswscale Documentation
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
+  <li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+
+<p>The libswscale library performs highly optimized image scaling and
+colorspace and pixel format conversion operations.
+</p>
+<p>Specifically, this library performs the following conversions:
+</p>
+<ul>
+<li> <em>Rescaling</em>: is the process of changing the video size. Several
+rescaling options and algorithms are available. This is usually a
+lossy process.
+
+</li><li> <em>Pixel format conversion</em>: is the process of converting the image
+format and colorspace of the image, for example from planar YUV420P to
+RGB24 packed. It also handles packing conversion, that is converts
+from packed layout (all pixels belonging to distinct planes
+interleaved in the same buffer), to planar layout (all samples
+belonging to the same plane stored in a dedicated buffer or &quot;plane&quot;).
+
+<p>This is usually a lossy process in case the source and destination
+colorspaces differ.
+</p></li></ul>
+
+
+<a name="See-Also"></a>
+<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
+
+<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
+<a href="ffmpeg-scaler.html">ffmpeg-scaler</a>,
+<a href="libavutil.html">libavutil</a>
+</p>
+
+<a name="Authors"></a>
+<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
+
+<p>The FFmpeg developers.
+</p>
+<p>For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+<code>git log</code> in the FFmpeg source directory, or browsing the
+online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
+</p>
+<p>Maintainers for the specific components are listed in the file
+<samp>MAINTAINERS</samp> in the source code tree.
+</p>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,212 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      NUT
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      NUT
+      </h1>
+<div align="center">
+</div>
+
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Description" href="#Description">1 Description</a></li>
+  <li><a name="toc-Modes" href="#Modes">2 Modes</a>
+  <ul class="no-bullet">
+    <li><a name="toc-BROADCAST" href="#BROADCAST">2.1 BROADCAST</a></li>
+    <li><a name="toc-PIPE" href="#PIPE">2.2 PIPE</a></li>
+  </ul></li>
+  <li><a name="toc-Container_002dspecific-codec-tags" href="#Container_002dspecific-codec-tags">3 Container-specific codec tags</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Generic-raw-YUVA-formats" href="#Generic-raw-YUVA-formats">3.1 Generic raw YUVA formats</a></li>
+    <li><a name="toc-Raw-Audio" href="#Raw-Audio">3.2 Raw Audio</a></li>
+    <li><a name="toc-Subtitles" href="#Subtitles">3.3 Subtitles</a></li>
+    <li><a name="toc-Raw-Data" href="#Raw-Data">3.4 Raw Data</a></li>
+    <li><a name="toc-Codecs" href="#Codecs">3.5 Codecs</a></li>
+  </ul></li>
+</ul>
+</div>
+
+
+<a name="Description"></a>
+<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
+<p>NUT is a low overhead generic container format. It stores audio, video,
+subtitle and user-defined streams in a simple, yet efficient, way.
+</p>
+<p>It was created by a group of FFmpeg and MPlayer developers in 2003
+and was finalized in 2008.
+</p>
+<p>The official nut specification is at svn://svn.mplayerhq.hu/nut
+In case of any differences between this text and the official specification,
+the official specification shall prevail.
+</p>
+<a name="Modes"></a>
+<h2 class="chapter">2 Modes<span class="pull-right"><a class="anchor hidden-xs" href="#Modes" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Modes" aria-hidden="true">TOC</a></span></h2>
+<p>NUT has some variants signaled by using the flags field in its main header.
+</p>
+<table>
+<tr><td width="40%">BROADCAST</td><td width="40%">Extend the syncpoint to report the sender wallclock</td></tr>
+<tr><td width="40%">PIPE</td><td width="40%">Omit completely the syncpoint</td></tr>
+</table>
+
+<a name="BROADCAST"></a>
+<h3 class="section">2.1 BROADCAST<span class="pull-right"><a class="anchor hidden-xs" href="#BROADCAST" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-BROADCAST" aria-hidden="true">TOC</a></span></h3>
+
+<p>The BROADCAST variant provides a secondary time reference to facilitate
+detecting endpoint latency and network delays.
+It assumes all the endpoint clocks are syncronized.
+To be used in real-time scenarios.
+</p>
+<a name="PIPE"></a>
+<h3 class="section">2.2 PIPE<span class="pull-right"><a class="anchor hidden-xs" href="#PIPE" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-PIPE" aria-hidden="true">TOC</a></span></h3>
+
+<p>The PIPE variant assumes NUT is used as non-seekable intermediate container,
+by not using syncpoint removes unneeded overhead and reduces the overall
+memory usage.
+</p>
+<a name="Container_002dspecific-codec-tags"></a>
+<h2 class="chapter">3 Container-specific codec tags<span class="pull-right"><a class="anchor hidden-xs" href="#Container_002dspecific-codec-tags" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Container_002dspecific-codec-tags" aria-hidden="true">TOC</a></span></h2>
+
+<a name="Generic-raw-YUVA-formats"></a>
+<h3 class="section">3.1 Generic raw YUVA formats<span class="pull-right"><a class="anchor hidden-xs" href="#Generic-raw-YUVA-formats" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Generic-raw-YUVA-formats" aria-hidden="true">TOC</a></span></h3>
+
+<p>Since many exotic planar YUVA pixel formats are not considered by
+the AVI/QuickTime FourCC lists, the following scheme is adopted for
+representing them.
+</p>
+<p>The first two bytes can contain the values:
+Y1 = only Y
+Y2 = Y+A
+Y3 = YUV
+Y4 = YUVA
+</p>
+<p>The third byte represents the width and height chroma subsampling
+values for the UV planes, that is the amount to shift the luma
+width/height right to find the chroma width/height.
+</p>
+<p>The fourth byte is the number of bits used (8, 16, ...).
+</p>
+<p>If the order of bytes is inverted, that means that each component has
+to be read big-endian.
+</p>
+<a name="Raw-Audio"></a>
+<h3 class="section">3.2 Raw Audio<span class="pull-right"><a class="anchor hidden-xs" href="#Raw-Audio" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Raw-Audio" aria-hidden="true">TOC</a></span></h3>
+
+<table>
+<tr><td width="40%">ALAW</td><td width="40%">A-LAW</td></tr>
+<tr><td width="40%">ULAW</td><td width="40%">MU-LAW</td></tr>
+<tr><td width="40%">P&lt;type&gt;&lt;interleaving&gt;&lt;bits&gt;</td><td width="40%">little-endian PCM</td></tr>
+<tr><td width="40%">&lt;bits&gt;&lt;interleaving&gt;&lt;type&gt;P</td><td width="40%">big-endian PCM</td></tr>
+</table>
+
+<p>&lt;type&gt; is S for signed integer, U for unsigned integer, F for IEEE float
+&lt;interleaving&gt; is D for default, P is for planar.
+&lt;bits&gt; is 8/16/24/32
+</p>
+<div class="example">
+<pre class="example">PFD[32]   would for example be signed 32 bit little-endian IEEE float
+</pre></div>
+
+<a name="Subtitles"></a>
+<h3 class="section">3.3 Subtitles<span class="pull-right"><a class="anchor hidden-xs" href="#Subtitles" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Subtitles" aria-hidden="true">TOC</a></span></h3>
+
+<table>
+<tr><td width="40%">UTF8</td><td width="40%">Raw UTF-8</td></tr>
+<tr><td width="40%">SSA[0]</td><td width="40%">SubStation Alpha</td></tr>
+<tr><td width="40%">DVDS</td><td width="40%">DVD subtitles</td></tr>
+<tr><td width="40%">DVBS</td><td width="40%">DVB subtitles</td></tr>
+</table>
+
+<a name="Raw-Data"></a>
+<h3 class="section">3.4 Raw Data<span class="pull-right"><a class="anchor hidden-xs" href="#Raw-Data" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Raw-Data" aria-hidden="true">TOC</a></span></h3>
+
+<table>
+<tr><td width="40%">UTF8</td><td width="40%">Raw UTF-8</td></tr>
+</table>
+
+<a name="Codecs"></a>
+<h3 class="section">3.5 Codecs<span class="pull-right"><a class="anchor hidden-xs" href="#Codecs" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Codecs" aria-hidden="true">TOC</a></span></h3>
+
+<table>
+<tr><td width="40%">3IV1</td><td width="40%">non-compliant MPEG-4 generated by old 3ivx</td></tr>
+<tr><td width="40%">ASV1</td><td width="40%">Asus Video</td></tr>
+<tr><td width="40%">ASV2</td><td width="40%">Asus Video 2</td></tr>
+<tr><td width="40%">CVID</td><td width="40%">Cinepak</td></tr>
+<tr><td width="40%">CYUV</td><td width="40%">Creative YUV</td></tr>
+<tr><td width="40%">DIVX</td><td width="40%">non-compliant MPEG-4 generated by old DivX</td></tr>
+<tr><td width="40%">DUCK</td><td width="40%">Truemotion 1</td></tr>
+<tr><td width="40%">FFV1</td><td width="40%">FFmpeg video 1</td></tr>
+<tr><td width="40%">FFVH</td><td width="40%">FFmpeg Huffyuv</td></tr>
+<tr><td width="40%">H261</td><td width="40%">ITU H.261</td></tr>
+<tr><td width="40%">H262</td><td width="40%">ITU H.262</td></tr>
+<tr><td width="40%">H263</td><td width="40%">ITU H.263</td></tr>
+<tr><td width="40%">H264</td><td width="40%">ITU H.264</td></tr>
+<tr><td width="40%">HFYU</td><td width="40%">Huffyuv</td></tr>
+<tr><td width="40%">I263</td><td width="40%">Intel H.263</td></tr>
+<tr><td width="40%">IV31</td><td width="40%">Indeo 3.1</td></tr>
+<tr><td width="40%">IV32</td><td width="40%">Indeo 3.2</td></tr>
+<tr><td width="40%">IV50</td><td width="40%">Indeo 5.0</td></tr>
+<tr><td width="40%">LJPG</td><td width="40%">ITU JPEG (lossless)</td></tr>
+<tr><td width="40%">MJLS</td><td width="40%">ITU JPEG-LS</td></tr>
+<tr><td width="40%">MJPG</td><td width="40%">ITU JPEG</td></tr>
+<tr><td width="40%">MPG4</td><td width="40%">MS MPEG-4v1 (not ISO MPEG-4)</td></tr>
+<tr><td width="40%">MP42</td><td width="40%">MS MPEG-4v2</td></tr>
+<tr><td width="40%">MP43</td><td width="40%">MS MPEG-4v3</td></tr>
+<tr><td width="40%">MP4V</td><td width="40%">ISO MPEG-4 Part 2 Video (from old encoders)</td></tr>
+<tr><td width="40%">mpg1</td><td width="40%">ISO MPEG-1 Video</td></tr>
+<tr><td width="40%">mpg2</td><td width="40%">ISO MPEG-2 Video</td></tr>
+<tr><td width="40%">MRLE</td><td width="40%">MS RLE</td></tr>
+<tr><td width="40%">MSVC</td><td width="40%">MS Video 1</td></tr>
+<tr><td width="40%">RT21</td><td width="40%">Indeo 2.1</td></tr>
+<tr><td width="40%">RV10</td><td width="40%">RealVideo 1.0</td></tr>
+<tr><td width="40%">RV20</td><td width="40%">RealVideo 2.0</td></tr>
+<tr><td width="40%">RV30</td><td width="40%">RealVideo 3.0</td></tr>
+<tr><td width="40%">RV40</td><td width="40%">RealVideo 4.0</td></tr>
+<tr><td width="40%">SNOW</td><td width="40%">FFmpeg Snow</td></tr>
+<tr><td width="40%">SVQ1</td><td width="40%">Sorenson Video 1</td></tr>
+<tr><td width="40%">SVQ3</td><td width="40%">Sorenson Video 3</td></tr>
+<tr><td width="40%">theo</td><td width="40%">Xiph Theora</td></tr>
+<tr><td width="40%">TM20</td><td width="40%">Truemotion 2.0</td></tr>
+<tr><td width="40%">UMP4</td><td width="40%">non-compliant MPEG-4 generated by UB Video MPEG-4</td></tr>
+<tr><td width="40%">VCR1</td><td width="40%">ATI VCR1</td></tr>
+<tr><td width="40%">VP30</td><td width="40%">VP 3.0</td></tr>
+<tr><td width="40%">VP31</td><td width="40%">VP 3.1</td></tr>
+<tr><td width="40%">VP50</td><td width="40%">VP 5.0</td></tr>
+<tr><td width="40%">VP60</td><td width="40%">VP 6.0</td></tr>
+<tr><td width="40%">VP61</td><td width="40%">VP 6.1</td></tr>
+<tr><td width="40%">VP62</td><td width="40%">VP 6.2</td></tr>
+<tr><td width="40%">VP70</td><td width="40%">VP 7.0</td></tr>
+<tr><td width="40%">WMV1</td><td width="40%">MS WMV7</td></tr>
+<tr><td width="40%">WMV2</td><td width="40%">MS WMV8</td></tr>
+<tr><td width="40%">WMV3</td><td width="40%">MS WMV9</td></tr>
+<tr><td width="40%">WV1F</td><td width="40%">non-compliant MPEG-4 generated by ?</td></tr>
+<tr><td width="40%">WVC1</td><td width="40%">VC-1</td></tr>
+<tr><td width="40%">XVID</td><td width="40%">non-compliant MPEG-4 generated by old Xvid</td></tr>
+<tr><td width="40%">XVIX</td><td width="40%">non-compliant MPEG-4 generated by old Xvid with interlacing bug</td></tr>
+</table>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,441 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
+<html>
+<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
+  <head>
+    <meta charset="utf-8">
+    <title>
+      Platform Specific Information
+    </title>
+    <meta name="viewport" content="width=device-width,initial-scale=1.0">
+    <link rel="stylesheet" type="text/css" href="bootstrap.min.css">
+    <link rel="stylesheet" type="text/css" href="style.min.css">
+  </head>
+  <body>
+    <div style="width: 95%; margin: auto">
+      <h1>
+      Platform Specific Information
+      </h1>
+<div align="center">
+</div>
+
+
+<a name="SEC_Top"></a>
+
+<a name="SEC_Contents"></a>
+<h2 class="contents-heading">Table of Contents</h2>
+
+<div class="contents">
+
+<ul class="no-bullet">
+  <li><a name="toc-Unix_002dlike" href="#Unix_002dlike">1 Unix-like</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Advanced-linking-configuration" href="#Advanced-linking-configuration">1.1 Advanced linking configuration</a></li>
+    <li><a name="toc-BSD" href="#BSD">1.2 BSD</a></li>
+    <li><a name="toc-_0028Open_0029Solaris" href="#g_t_0028Open_0029Solaris">1.3 (Open)Solaris</a></li>
+    <li><a name="toc-Darwin-_0028Mac-OS-X_002c-iPhone_0029" href="#Darwin-_0028Mac-OS-X_002c-iPhone_0029">1.4 Darwin (Mac OS X, iPhone)</a></li>
+  </ul></li>
+  <li><a name="toc-DOS" href="#DOS">2 DOS</a></li>
+  <li><a name="toc-OS_002f2" href="#OS_002f2">3 OS/2</a></li>
+  <li><a name="toc-Windows" href="#Windows">4 Windows</a>
+  <ul class="no-bullet">
+    <li><a name="toc-Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" href="#Native-Windows-compilation-using-MinGW-or-MinGW_002dw64">4.1 Native Windows compilation using MinGW or MinGW-w64</a></li>
+    <li><a name="toc-Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" href="#Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows">4.2 Microsoft Visual C++ or Intel C++ Compiler for Windows</a>
+    <ul class="no-bullet">
+      <li><a name="toc-Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" href="#Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b">4.2.1 Linking to FFmpeg with Microsoft Visual C++</a></li>
+    </ul></li>
+    <li><a name="toc-Cross-compilation-for-Windows-with-Linux-1" href="#Cross-compilation-for-Windows-with-Linux-1">4.3 Cross compilation for Windows with Linux</a></li>
+    <li><a name="toc-Compilation-under-Cygwin" href="#Compilation-under-Cygwin">4.4 Compilation under Cygwin</a></li>
+    <li><a name="toc-Crosscompilation-for-Windows-under-Cygwin" href="#Crosscompilation-for-Windows-under-Cygwin">4.5 Crosscompilation for Windows under Cygwin</a></li>
+  </ul></li>
+  <li><a name="toc-Plan-9" href="#Plan-9">5 Plan 9</a></li>
+</ul>
+</div>
+
+
+<a name="Unix_002dlike"></a>
+<h2 class="chapter">1 Unix-like<span class="pull-right"><a class="anchor hidden-xs" href="#Unix_002dlike" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Unix_002dlike" aria-hidden="true">TOC</a></span></h2>
+
+<p>Some parts of FFmpeg cannot be built with version 2.15 of the GNU
+assembler which is still provided by a few AMD64 distributions. To
+make sure your compiler really uses the required version of gas
+after a binutils upgrade, run:
+</p>
+<div class="example">
+<pre class="example">$(gcc -print-prog-name=as) --version
+</pre></div>
+
+<p>If not, then you should install a different compiler that has no
+hard-coded path to gas. In the worst case pass <code>--disable-asm</code>
+to configure.
+</p>
+<a name="Advanced-linking-configuration"></a>
+<h3 class="section">1.1 Advanced linking configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Advanced-linking-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Advanced-linking-configuration" aria-hidden="true">TOC</a></span></h3>
+
+<p>If you compiled FFmpeg libraries statically and you want to use them to
+build your own shared library, you may need to force PIC support (with
+<code>--enable-pic</code> during FFmpeg configure) and add the following option
+to your project LDFLAGS:
+</p>
+<div class="example">
+<pre class="example">-Wl,-Bsymbolic
+</pre></div>
+
+<p>If your target platform requires position independent binaries, you should
+pass the correct linking flag (e.g. <code>-pie</code>) to <code>--extra-ldexeflags</code>.
+</p>
+<a name="BSD"></a>
+<h3 class="section">1.2 BSD<span class="pull-right"><a class="anchor hidden-xs" href="#BSD" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-BSD" aria-hidden="true">TOC</a></span></h3>
+
+<p>BSD make will not build FFmpeg, you need to install and use GNU Make
+(<code>gmake</code>).
+</p>
+<a name="g_t_0028Open_0029Solaris"></a>
+<h3 class="section">1.3 (Open)Solaris<span class="pull-right"><a class="anchor hidden-xs" href="#_0028Open_0029Solaris" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-_0028Open_0029Solaris" aria-hidden="true">TOC</a></span></h3>
+
+<p>GNU Make is required to build FFmpeg, so you have to invoke (<code>gmake</code>),
+standard Solaris Make will not work. When building with a non-c99 front-end
+(gcc, generic suncc) add either <code>--extra-libs=/usr/lib/values-xpg6.o</code>
+or <code>--extra-libs=/usr/lib/64/values-xpg6.o</code> to the configure options
+since the libc is not c99-compliant by default. The probes performed by
+configure may raise an exception leading to the death of configure itself
+due to a bug in the system shell. Simply invoke a different shell such as
+bash directly to work around this:
+</p>
+<div class="example">
+<pre class="example">bash ./configure
+</pre></div>
+
+<a name="Darwin"></a><a name="Darwin-_0028Mac-OS-X_002c-iPhone_0029"></a>
+<h3 class="section">1.4 Darwin (Mac OS X, iPhone)<span class="pull-right"><a class="anchor hidden-xs" href="#Darwin-_0028Mac-OS-X_002c-iPhone_0029" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Darwin-_0028Mac-OS-X_002c-iPhone_0029" aria-hidden="true">TOC</a></span></h3>
+
+<p>The toolchain provided with Xcode is sufficient to build the basic
+unacelerated code.
+</p>
+<p>Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
+<a href="https://github.com/FFmpeg/gas-preprocessor">https://github.com/FFmpeg/gas-preprocessor</a> or
+<a href="https://github.com/yuvi/gas-preprocessor">https://github.com/yuvi/gas-preprocessor</a>(currently outdated) to build the optimized
+assembly functions. Put the Perl script somewhere
+in your PATH, FFmpeg&rsquo;s configure will pick it up automatically.
+</p>
+<p>Mac OS X on amd64 and x86 requires <code>yasm</code> to build most of the
+optimized assembly functions. <a href="http://www.finkproject.org/">Fink</a>,
+<a href="http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml">Gentoo Prefix</a>,
+<a href="https://mxcl.github.com/homebrew/">Homebrew</a>
+or <a href="http://www.macports.org">MacPorts</a> can easily provide it.
+</p>
+
+<a name="DOS"></a>
+<h2 class="chapter">2 DOS<span class="pull-right"><a class="anchor hidden-xs" href="#DOS" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-DOS" aria-hidden="true">TOC</a></span></h2>
+
+<p>Using a cross-compiler is preferred for various reasons.
+<a href="http://www.delorie.com/howto/djgpp/linux-x-djgpp.html">http://www.delorie.com/howto/djgpp/linux-x-djgpp.html</a>
+</p>
+
+<a name="OS_002f2"></a>
+<h2 class="chapter">3 OS/2<span class="pull-right"><a class="anchor hidden-xs" href="#OS_002f2" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-OS_002f2" aria-hidden="true">TOC</a></span></h2>
+
+<p>For information about compiling FFmpeg on OS/2 see
+<a href="http://www.edm2.com/index.php/FFmpeg">http://www.edm2.com/index.php/FFmpeg</a>.
+</p>
+
+<a name="Windows"></a>
+<h2 class="chapter">4 Windows<span class="pull-right"><a class="anchor hidden-xs" href="#Windows" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Windows" aria-hidden="true">TOC</a></span></h2>
+
+<p>To get help and instructions for building FFmpeg under Windows, check out
+the FFmpeg Windows Help Forum at <a href="http://ffmpeg.zeranoe.com/forum/">http://ffmpeg.zeranoe.com/forum/</a>.
+</p>
+<a name="Native-Windows-compilation-using-MinGW-or-MinGW_002dw64"></a>
+<h3 class="section">4.1 Native Windows compilation using MinGW or MinGW-w64<span class="pull-right"><a class="anchor hidden-xs" href="#Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" aria-hidden="true">TOC</a></span></h3>
+
+<p>FFmpeg can be built to run natively on Windows using the MinGW-w64
+toolchain. Install the latest versions of MSYS2 and MinGW-w64 from
+<a href="http://msys2.github.io/">http://msys2.github.io/</a> and/or <a href="http://mingw-w64.sourceforge.net/">http://mingw-w64.sourceforge.net/</a>.
+You can find detailed installation instructions in the download section and
+the FAQ.
+</p>
+<p>Notes:
+</p>
+<ul>
+<li> Building natively using MSYS2 can be sped up by disabling implicit rules
+in the Makefile by calling <code>make -r</code> instead of plain <code>make</code>. This
+speed up is close to non-existent for normal one-off builds and is only
+noticeable when running make for a second time (for example during
+<code>make install</code>).
+
+</li><li> In order to compile FFplay, you must have the MinGW development library
+of <a href="http://www.libsdl.org/">SDL</a> and <code>pkg-config</code> installed.
+
+</li><li> By using <code>./configure --enable-shared</code> when configuring FFmpeg,
+you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
+libavformat) as DLLs.
+
+</li></ul>
+
+<a name="Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows"></a>
+<h3 class="section">4.2 Microsoft Visual C++ or Intel C++ Compiler for Windows<span class="pull-right"><a class="anchor hidden-xs" href="#Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" aria-hidden="true">TOC</a></span></h3>
+
+<p>FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
+and wrapper, or with MSVC 2013 and ICL natively.
+</p>
+<p>You will need the following prerequisites:
+</p>
+<ul>
+<li> <a href="https://github.com/libav/c99-to-c89/">C99-to-C89 Converter &amp; Wrapper</a>
+(if using MSVC 2012 or earlier)
+</li><li> <a href="http://code.google.com/p/msinttypes/">msinttypes</a>
+(if using MSVC 2012 or earlier)
+</li><li> <a href="http://msys2.github.io/">MSYS2</a>
+</li><li> <a href="http://yasm.tortall.net/">YASM</a>
+(Also available via MSYS2&rsquo;s package manager.)
+</li></ul>
+
+<p>To set up a proper environment in MSYS2, you need to run <code>msys_shell.bat</code> from
+the Visual Studio or Intel Compiler command prompt.
+</p>
+<p>Place <code>yasm.exe</code> somewhere in your <code>PATH</code>. If using MSVC 2012 or
+earlier, place <code>c99wrap.exe</code> and <code>c99conv.exe</code> somewhere in your
+<code>PATH</code> as well.
+</p>
+<p>Next, make sure any other headers and libs you want to use, such as zlib, are
+located in a spot that the compiler can see. Do so by modifying the <code>LIB</code>
+and <code>INCLUDE</code> environment variables to include the <strong>Windows-style</strong>
+paths to these directories. Alternatively, you can try and use the
+<code>--extra-cflags</code>/<code>--extra-ldflags</code> configure options. If using MSVC
+2012 or earlier, place <code>inttypes.h</code> somewhere the compiler can see too.
+</p>
+<p>Finally, run:
+</p>
+<div class="example">
+<pre class="example">For MSVC:
+./configure --toolchain=msvc
+
+For ICL:
+./configure --toolchain=icl
+
+make
+make install
+</pre></div>
+
+<p>If you wish to compile shared libraries, add <code>--enable-shared</code> to your
+configure options. Note that due to the way MSVC and ICL handle DLL imports and
+exports, you cannot compile static and shared libraries at the same time, and
+enabling shared libraries will automatically disable the static ones.
+</p>
+<p>Notes:
+</p>
+<ul>
+<li> If you wish to build with zlib support, you will have to grab a compatible
+zlib binary from somewhere, with an MSVC import lib, or if you wish to link
+statically, you can follow the instructions below to build a compatible
+<code>zlib.lib</code> with MSVC. Regardless of which method you use, you must still
+follow step 3, or compilation will fail.
+<ol>
+<li> Grab the <a href="http://zlib.net/">zlib sources</a>.
+</li><li> Edit <code>win32/Makefile.msc</code> so that it uses -MT instead of -MD, since
+this is how FFmpeg is built as well.
+</li><li> Edit <code>zconf.h</code> and remove its inclusion of <code>unistd.h</code>. This gets
+erroneously included when building FFmpeg.
+</li><li> Run <code>nmake -f win32/Makefile.msc</code>.
+</li><li> Move <code>zlib.lib</code>, <code>zconf.h</code>, and <code>zlib.h</code> to somewhere MSVC
+can see.
+</li></ol>
+
+</li><li> FFmpeg has been tested with the following on i686 and x86_64:
+<ul>
+<li> Visual Studio 2010 Pro and Express
+</li><li> Visual Studio 2012 Pro and Express
+</li><li> Visual Studio 2013 Pro and Express
+</li><li> Intel Composer XE 2013
+</li><li> Intel Composer XE 2013 SP1
+</li></ul>
+<p>Anything else is not officially supported.
+</p>
+</li></ul>
+
+<a name="Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b"></a>
+<h4 class="subsection">4.2.1 Linking to FFmpeg with Microsoft Visual C++<span class="pull-right"><a class="anchor hidden-xs" href="#Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" aria-hidden="true">TOC</a></span></h4>
+
+<p>If you plan to link with MSVC-built static libraries, you will need
+to make sure you have <code>Runtime Library</code> set to
+<code>Multi-threaded (/MT)</code> in your project&rsquo;s settings.
+</p>
+<p>You will need to define <code>inline</code> to something MSVC understands:
+</p><div class="example">
+<pre class="example">#define inline __inline
+</pre></div>
+
+<p>Also note, that as stated in <strong>Microsoft Visual C++</strong>, you will need
+an MSVC-compatible <a href="http://code.google.com/p/msinttypes/">inttypes.h</a>.
+</p>
+<p>If you plan on using import libraries created by dlltool, you must
+set <code>References</code> to <code>No (/OPT:NOREF)</code> under the linker optimization
+settings, otherwise the resulting binaries will fail during runtime.
+This is not required when using import libraries generated by <code>lib.exe</code>.
+This issue is reported upstream at
+<a href="http://sourceware.org/bugzilla/show_bug.cgi?id=12633">http://sourceware.org/bugzilla/show_bug.cgi?id=12633</a>.
+</p>
+<p>To create import libraries that work with the <code>/OPT:REF</code> option
+(which is enabled by default in Release mode), follow these steps:
+</p>
+<ol>
+<li> Open the <em>Visual Studio Command Prompt</em>.
+
+<p>Alternatively, in a normal command line prompt, call <samp>vcvars32.bat</samp>
+which sets up the environment variables for the Visual C++ tools
+(the standard location for this file is something like
+<samp>C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat</samp>).
+</p>
+</li><li> Enter the <samp>bin</samp> directory where the created LIB and DLL files
+are stored.
+
+</li><li> Generate new import libraries with <code>lib.exe</code>:
+
+<div class="example">
+<pre class="example">lib /machine:i386 /def:..\lib\foo-version.def  /out:foo.lib
+</pre></div>
+
+<p>Replace <code>foo-version</code> and <code>foo</code> with the respective library names.
+</p>
+</li></ol>
+
+<a name="Cross-compilation-for-Windows-with-Linux"></a><a name="Cross-compilation-for-Windows-with-Linux-1"></a>
+<h3 class="section">4.3 Cross compilation for Windows with Linux<span class="pull-right"><a class="anchor hidden-xs" href="#Cross-compilation-for-Windows-with-Linux-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Cross-compilation-for-Windows-with-Linux-1" aria-hidden="true">TOC</a></span></h3>
+
+<p>You must use the MinGW cross compilation tools available at
+<a href="http://www.mingw.org/">http://www.mingw.org/</a>.
+</p>
+<p>Then configure FFmpeg with the following options:
+</p><div class="example">
+<pre class="example">./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
+</pre></div>
+<p>(you can change the cross-prefix according to the prefix chosen for the
+MinGW tools).
+</p>
+<p>Then you can easily test FFmpeg with <a href="http://www.winehq.com/">Wine</a>.
+</p>
+<a name="Compilation-under-Cygwin"></a>
+<h3 class="section">4.4 Compilation under Cygwin<span class="pull-right"><a class="anchor hidden-xs" href="#Compilation-under-Cygwin" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Compilation-under-Cygwin" aria-hidden="true">TOC</a></span></h3>
+
+<p>Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
+llrint() in its C library.
+</p>
+<p>Install your Cygwin with all the &quot;Base&quot; packages, plus the
+following &quot;Devel&quot; ones:
+</p><div class="example">
+<pre class="example">binutils, gcc4-core, make, git, mingw-runtime, texinfo
+</pre></div>
+
+<p>In order to run FATE you will also need the following &quot;Utils&quot; packages:
+</p><div class="example">
+<pre class="example">diffutils
+</pre></div>
+
+<p>If you want to build FFmpeg with additional libraries, download Cygwin
+&quot;Devel&quot; packages for Ogg and Vorbis from any Cygwin packages repository:
+</p><div class="example">
+<pre class="example">libogg-devel, libvorbis-devel
+</pre></div>
+
+<p>These library packages are only available from
+<a href="http://sourceware.org/cygwinports/">Cygwin Ports</a>:
+</p>
+<div class="example">
+<pre class="example">yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
+libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
+</pre></div>
+
+<p>The recommendation for x264 is to build it from source, as it evolves too
+quickly for Cygwin Ports to be up to date.
+</p>
+<a name="Crosscompilation-for-Windows-under-Cygwin"></a>
+<h3 class="section">4.5 Crosscompilation for Windows under Cygwin<span class="pull-right"><a class="anchor hidden-xs" href="#Crosscompilation-for-Windows-under-Cygwin" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Crosscompilation-for-Windows-under-Cygwin" aria-hidden="true">TOC</a></span></h3>
+
+<p>With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
+</p>
+<p>Just install your Cygwin as explained before, plus these additional
+&quot;Devel&quot; packages:
+</p><div class="example">
+<pre class="example">gcc-mingw-core, mingw-runtime, mingw-zlib
+</pre></div>
+
+<p>and add some special flags to your configure invocation.
+</p>
+<p>For a static build run
+</p><div class="example">
+<pre class="example">./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
+</pre></div>
+
+<p>and for a build with shared libraries
+</p><div class="example">
+<pre class="example">./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
+</pre></div>
+
+<a name="Plan-9"></a>
+<h2 class="chapter">5 Plan 9<span class="pull-right"><a class="anchor hidden-xs" href="#Plan-9" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Plan-9" aria-hidden="true">TOC</a></span></h2>
+
+<p>The native <a href="http://plan9.bell-labs.com/plan9/">Plan 9</a> compiler
+does not implement all the C99 features needed by FFmpeg so the gcc
+port must be used.  Furthermore, a few items missing from the C
+library and shell environment need to be fixed.
+</p>
+<ul>
+<li> GNU awk, grep, make, and sed
+
+<p>Working packages of these tools can be found at
+<a href="http://code.google.com/p/ports2plan9/downloads/list">ports2plan9</a>.
+They can be installed with <a href="http://9front.org/">9front&rsquo;s</a> <code>pkg</code>
+utility by setting <code>pkgpath</code> to
+<code>http://ports2plan9.googlecode.com/files/</code>.
+</p>
+</li><li> Missing/broken <code>head</code> and <code>printf</code> commands
+
+<p>Replacements adequate for building FFmpeg can be found in the
+<code>compat/plan9</code> directory.  Place these somewhere they will be
+found by the shell.  These are not full implementations of the
+commands and are <em>not</em> suitable for general use.
+</p>
+</li><li> Missing C99 <code>stdint.h</code> and <code>inttypes.h</code>
+
+<p>Replacement headers are available from
+<a href="http://code.google.com/p/plan9front/issues/detail?id=152">http://code.google.com/p/plan9front/issues/detail?id=152</a>.
+</p>
+</li><li> Missing or non-standard library functions
+
+<p>Some functions in the C library are missing or incomplete.  The
+<code><a href="http://ports2plan9.googlecode.com/files/gcc-apelibs-1207.tbz">gcc-apelibs-1207</a></code> package from
+<a href="http://code.google.com/p/ports2plan9/downloads/list">ports2plan9</a>
+includes an updated C library, but installing the full package gives
+unusable executables.  Instead, keep the files from <code>gccbin.tgz</code>
+under <code>/386/lib/gnu</code>.  From the <code>libc.a</code> archive in the
+<code>gcc-apelibs-1207</code> package, extract the following object files and
+turn them into a library:
+</p>
+<ul>
+<li> <code>strerror.o</code>
+</li><li> <code>strtoll.o</code>
+</li><li> <code>snprintf.o</code>
+</li><li> <code>vsnprintf.o</code>
+</li><li> <code>vfprintf.o</code>
+</li><li> <code>_IO_getc.o</code>
+</li><li> <code>_IO_putc.o</code>
+</li></ul>
+
+<p>Use the <code>--extra-libs</code> option of <code>configure</code> to inform the
+build system of this library.
+</p>
+</li><li> FPU exceptions enabled by default
+
+<p>Unlike most other systems, Plan 9 enables FPU exceptions by default.
+These must be disabled before calling any FFmpeg functions.  While the
+included tools will do this automatically, other users of the
+libraries must do it themselves.
+</p>
+</li></ul>
+
+
+      <p style="font-size: small;">
+        This document was generated using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
+      </p>
+    </div>
+  </body>
+</html>
@@ -0,0 +1,84 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AVDCT_H
+#define AVCODEC_AVDCT_H
+
+#include "libavutil/opt.h"
+
+/**
+ * AVDCT context.
+ * @note function pointers can be NULL if the specific features have been
+ *       disabled at build time.
+ */
+typedef struct AVDCT {
+    const AVClass *av_class;
+
+    void (*idct)(int16_t *block /* align 16 */);
+
+    /**
+     * IDCT input permutation.
+     * Several optimized IDCTs need a permutated input (relative to the
+     * normal order of the reference IDCT).
+     * This permutation must be performed before the idct_put/add.
+     * Note, normally this can be merged with the zigzag/alternate scan<br>
+     * An example to avoid confusion:
+     * - (->decode coeffs -> zigzag reorder -> dequant -> reference IDCT -> ...)
+     * - (x -> reference DCT -> reference IDCT -> x)
+     * - (x -> reference DCT -> simple_mmx_perm = idct_permutation
+     *    -> simple_idct_mmx -> x)
+     * - (-> decode coeffs -> zigzag reorder -> simple_mmx_perm -> dequant
+     *    -> simple_idct_mmx -> ...)
+     */
+    uint8_t idct_permutation[64];
+
+    void (*fdct)(int16_t *block /* align 16 */);
+
+
+    /**
+     * DCT algorithm.
+     * must use AVOptions to set this field.
+     */
+    int dct_algo;
+
+    /**
+     * IDCT algorithm.
+     * must use AVOptions to set this field.
+     */
+    int idct_algo;
+
+    void (*get_pixels)(int16_t *block /* align 16 */,
+                       const uint8_t *pixels /* align 8 */,
+                       ptrdiff_t line_size);
+
+    int bits_per_sample;
+} AVDCT;
+
+/**
+ * Allocates a AVDCT context.
+ * This needs to be initialized with avcodec_dct_init() after optionally
+ * configuring it with AVOptions.
+ *
+ * To free it use av_free()
+ */
+AVDCT *avcodec_dct_alloc(void);
+int avcodec_dct_init(AVDCT *);
+
+const AVClass *avcodec_dct_get_class(void);
+
+#endif /* AVCODEC_AVDCT_H */
@@ -0,0 +1,118 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AVFFT_H
+#define AVCODEC_AVFFT_H
+
+/**
+ * @file
+ * @ingroup lavc_fft
+ * FFT functions
+ */
+
+/**
+ * @defgroup lavc_fft FFT functions
+ * @ingroup lavc_misc
+ *
+ * @{
+ */
+
+typedef float FFTSample;
+
+typedef struct FFTComplex {
+    FFTSample re, im;
+} FFTComplex;
+
+typedef struct FFTContext FFTContext;
+
+/**
+ * Set up a complex FFT.
+ * @param nbits           log2 of the length of the input array
+ * @param inverse         if 0 perform the forward transform, if 1 perform the inverse
+ */
+FFTContext *av_fft_init(int nbits, int inverse);
+
+/**
+ * Do the permutation needed BEFORE calling ff_fft_calc().
+ */
+void av_fft_permute(FFTContext *s, FFTComplex *z);
+
+/**
+ * Do a complex FFT with the parameters defined in av_fft_init(). The
+ * input data must be permuted before. No 1.0/sqrt(n) normalization is done.
+ */
+void av_fft_calc(FFTContext *s, FFTComplex *z);
+
+void av_fft_end(FFTContext *s);
+
+FFTContext *av_mdct_init(int nbits, int inverse, double scale);
+void av_imdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input);
+void av_imdct_half(FFTContext *s, FFTSample *output, const FFTSample *input);
+void av_mdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input);
+void av_mdct_end(FFTContext *s);
+
+/* Real Discrete Fourier Transform */
+
+enum RDFTransformType {
+    DFT_R2C,
+    IDFT_C2R,
+    IDFT_R2C,
+    DFT_C2R,
+};
+
+typedef struct RDFTContext RDFTContext;
+
+/**
+ * Set up a real FFT.
+ * @param nbits           log2 of the length of the input array
+ * @param trans           the type of transform
+ */
+RDFTContext *av_rdft_init(int nbits, enum RDFTransformType trans);
+void av_rdft_calc(RDFTContext *s, FFTSample *data);
+void av_rdft_end(RDFTContext *s);
+
+/* Discrete Cosine Transform */
+
+typedef struct DCTContext DCTContext;
+
+enum DCTTransformType {
+    DCT_II = 0,
+    DCT_III,
+    DCT_I,
+    DST_I,
+};
+
+/**
+ * Set up DCT.
+ *
+ * @param nbits           size of the input array:
+ *                        (1 << nbits)     for DCT-II, DCT-III and DST-I
+ *                        (1 << nbits) + 1 for DCT-I
+ * @param type            the type of transform
+ *
+ * @note the first element of the input of DST-I is ignored
+ */
+DCTContext *av_dct_init(int nbits, enum DCTTransformType type);
+void av_dct_calc(DCTContext *s, FFTSample *data);
+void av_dct_end (DCTContext *s);
+
+/**
+ * @}
+ */
+
+#endif /* AVCODEC_AVFFT_H */
@@ -0,0 +1,112 @@
+/*
+ * Direct3D11 HW acceleration
+ *
+ * copyright (c) 2009 Laurent Aimar
+ * copyright (c) 2015 Steve Lhomme
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_D3D11VA_H
+#define AVCODEC_D3D11VA_H
+
+/**
+ * @file
+ * @ingroup lavc_codec_hwaccel_d3d11va
+ * Public libavcodec D3D11VA header.
+ */
+
+#if !defined(_WIN32_WINNT) || _WIN32_WINNT < 0x0602
+#undef _WIN32_WINNT
+#define _WIN32_WINNT 0x0602
+#endif
+
+#include <stdint.h>
+#include <d3d11.h>
+
+/**
+ * @defgroup lavc_codec_hwaccel_d3d11va Direct3D11
+ * @ingroup lavc_codec_hwaccel
+ *
+ * @{
+ */
+
+#define FF_DXVA2_WORKAROUND_SCALING_LIST_ZIGZAG 1 ///< Work around for Direct3D11 and old UVD/UVD+ ATI video cards
+#define FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO    2 ///< Work around for Direct3D11 and old Intel GPUs with ClearVideo interface
+
+/**
+ * This structure is used to provides the necessary configurations and data
+ * to the Direct3D11 FFmpeg HWAccel implementation.
+ *
+ * The application must make it available as AVCodecContext.hwaccel_context.
+ *
+ * Use av_d3d11va_alloc_context() exclusively to allocate an AVD3D11VAContext.
+ */
+typedef struct AVD3D11VAContext {
+    /**
+     * D3D11 decoder object
+     */
+    ID3D11VideoDecoder *decoder;
+
+    /**
+      * D3D11 VideoContext
+      */
+    ID3D11VideoContext *video_context;
+
+    /**
+     * D3D11 configuration used to create the decoder
+     */
+    D3D11_VIDEO_DECODER_CONFIG *cfg;
+
+    /**
+     * The number of surface in the surface array
+     */
+    unsigned surface_count;
+
+    /**
+     * The array of Direct3D surfaces used to create the decoder
+     */
+    ID3D11VideoDecoderOutputView **surface;
+
+    /**
+     * A bit field configuring the workarounds needed for using the decoder
+     */
+    uint64_t workaround;
+
+    /**
+     * Private to the FFmpeg AVHWAccel implementation
+     */
+    unsigned report_id;
+
+    /**
+      * Mutex to access video_context
+      */
+    HANDLE  context_mutex;
+} AVD3D11VAContext;
+
+/**
+ * Allocate an AVD3D11VAContext.
+ *
+ * @return Newly-allocated AVD3D11VAContext or NULL on failure.
+ */
+AVD3D11VAContext *av_d3d11va_alloc_context(void);
+
+/**
+ * @}
+ */
+
+#endif /* AVCODEC_D3D11VA_H */
@@ -0,0 +1,83 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_DV_PROFILE_H
+#define AVCODEC_DV_PROFILE_H
+
+#include <stdint.h>
+
+#include "libavutil/pixfmt.h"
+#include "libavutil/rational.h"
+#include "avcodec.h"
+
+/* minimum number of bytes to read from a DV stream in order to
+ * determine the profile */
+#define DV_PROFILE_BYTES (6 * 80) /* 6 DIF blocks */
+
+
+/*
+ * AVDVProfile is used to express the differences between various
+ * DV flavors. For now it's primarily used for differentiating
+ * 525/60 and 625/50, but the plans are to use it for various
+ * DV specs as well (e.g. SMPTE314M vs. IEC 61834).
+ */
+typedef struct AVDVProfile {
+    int              dsf;                   /* value of the dsf in the DV header */
+    int              video_stype;           /* stype for VAUX source pack */
+    int              frame_size;            /* total size of one frame in bytes */
+    int              difseg_size;           /* number of DIF segments per DIF channel */
+    int              n_difchan;             /* number of DIF channels per frame */
+    AVRational       time_base;             /* 1/framerate */
+    int              ltc_divisor;           /* FPS from the LTS standpoint */
+    int              height;                /* picture height in pixels */
+    int              width;                 /* picture width in pixels */
+    AVRational       sar[2];                /* sample aspect ratios for 4:3 and 16:9 */
+    enum AVPixelFormat pix_fmt;             /* picture pixel format */
+    int              bpm;                   /* blocks per macroblock */
+    const uint8_t   *block_sizes;           /* AC block sizes, in bits */
+    int              audio_stride;          /* size of audio_shuffle table */
+    int              audio_min_samples[3];  /* min amount of audio samples */
+                                            /* for 48kHz, 44.1kHz and 32kHz */
+    int              audio_samples_dist[5]; /* how many samples are supposed to be */
+                                            /* in each frame in a 5 frames window */
+    const uint8_t  (*audio_shuffle)[9];     /* PCM shuffling table */
+} AVDVProfile;
+
+/**
+ * Get a DV profile for the provided compressed frame.
+ *
+ * @param sys the profile used for the previous frame, may be NULL
+ * @param frame the compressed data buffer
+ * @param buf_size size of the buffer in bytes
+ * @return the DV profile for the supplied data or NULL on failure
+ */
+const AVDVProfile *av_dv_frame_profile(const AVDVProfile *sys,
+                                       const uint8_t *frame, unsigned buf_size);
+
+/**
+ * Get a DV profile for the provided stream parameters.
+ */
+const AVDVProfile *av_dv_codec_profile(int width, int height, enum AVPixelFormat pix_fmt);
+
+/**
+ * Get a DV profile for the provided stream parameters.
+ * The frame rate is used as a best-effort parameter.
+ */
+const AVDVProfile *av_dv_codec_profile2(int width, int height, enum AVPixelFormat pix_fmt, AVRational frame_rate);
+
+#endif /* AVCODEC_DV_PROFILE_H */
@@ -0,0 +1,93 @@
+/*
+ * DXVA2 HW acceleration
+ *
+ * copyright (c) 2009 Laurent Aimar
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_DXVA_H
+#define AVCODEC_DXVA_H
+
+/**
+ * @file
+ * @ingroup lavc_codec_hwaccel_dxva2
+ * Public libavcodec DXVA2 header.
+ */
+
+#if !defined(_WIN32_WINNT) || _WIN32_WINNT < 0x0602
+#undef _WIN32_WINNT
+#define _WIN32_WINNT 0x0602
+#endif
+
+#include <stdint.h>
+#include <d3d9.h>
+#include <dxva2api.h>
+
+/**
+ * @defgroup lavc_codec_hwaccel_dxva2 DXVA2
+ * @ingroup lavc_codec_hwaccel
+ *
+ * @{
+ */
+
+#define FF_DXVA2_WORKAROUND_SCALING_LIST_ZIGZAG 1 ///< Work around for DXVA2 and old UVD/UVD+ ATI video cards
+#define FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO    2 ///< Work around for DXVA2 and old Intel GPUs with ClearVideo interface
+
+/**
+ * This structure is used to provides the necessary configurations and data
+ * to the DXVA2 FFmpeg HWAccel implementation.
+ *
+ * The application must make it available as AVCodecContext.hwaccel_context.
+ */
+struct dxva_context {
+    /**
+     * DXVA2 decoder object
+     */
+    IDirectXVideoDecoder *decoder;
+
+    /**
+     * DXVA2 configuration used to create the decoder
+     */
+    const DXVA2_ConfigPictureDecode *cfg;
+
+    /**
+     * The number of surface in the surface array
+     */
+    unsigned surface_count;
+
+    /**
+     * The array of Direct3D surfaces used to create the decoder
+     */
+    LPDIRECT3DSURFACE9 *surface;
+
+    /**
+     * A bit field configuring the workarounds needed for using the decoder
+     */
+    uint64_t workaround;
+
+    /**
+     * Private to the FFmpeg AVHWAccel implementation
+     */
+    unsigned report_id;
+};
+
+/**
+ * @}
+ */
+
+#endif /* AVCODEC_DXVA_H */
@@ -0,0 +1,107 @@
+/*
+ * Intel MediaSDK QSV public API
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_QSV_H
+#define AVCODEC_QSV_H
+
+#include <mfx/mfxvideo.h>
+
+#include "libavutil/buffer.h"
+
+/**
+ * This struct is used for communicating QSV parameters between libavcodec and
+ * the caller. It is managed by the caller and must be assigned to
+ * AVCodecContext.hwaccel_context.
+ * - decoding: hwaccel_context must be set on return from the get_format()
+ *             callback
+ * - encoding: hwaccel_context must be set before avcodec_open2()
+ */
+typedef struct AVQSVContext {
+    /**
+     * If non-NULL, the session to use for encoding or decoding.
+     * Otherwise, libavcodec will try to create an internal session.
+     */
+    mfxSession session;
+
+    /**
+     * The IO pattern to use.
+     */
+    int iopattern;
+
+    /**
+     * Extra buffers to pass to encoder or decoder initialization.
+     */
+    mfxExtBuffer **ext_buffers;
+    int         nb_ext_buffers;
+
+    /**
+     * Encoding only. If this field is set to non-zero by the caller, libavcodec
+     * will create an mfxExtOpaqueSurfaceAlloc extended buffer and pass it to
+     * the encoder initialization. This only makes sense if iopattern is also
+     * set to MFX_IOPATTERN_IN_OPAQUE_MEMORY.
+     *
+     * The number of allocated opaque surfaces will be the sum of the number
+     * required by the encoder and the user-provided value nb_opaque_surfaces.
+     * The array of the opaque surfaces will be exported to the caller through
+     * the opaque_surfaces field.
+     */
+    int opaque_alloc;
+
+    /**
+     * Encoding only, and only if opaque_alloc is set to non-zero. Before
+     * calling avcodec_open2(), the caller should set this field to the number
+     * of extra opaque surfaces to allocate beyond what is required by the
+     * encoder.
+     *
+     * On return from avcodec_open2(), this field will be set by libavcodec to
+     * the total number of allocated opaque surfaces.
+     */
+    int nb_opaque_surfaces;
+
+    /**
+     * Encoding only, and only if opaque_alloc is set to non-zero. On return
+     * from avcodec_open2(), this field will be used by libavcodec to export the
+     * array of the allocated opaque surfaces to the caller, so they can be
+     * passed to other parts of the pipeline.
+     *
+     * The buffer reference exported here is owned and managed by libavcodec,
+     * the callers should make their own reference with av_buffer_ref() and free
+     * it with av_buffer_unref() when it is no longer needed.
+     *
+     * The buffer data is an nb_opaque_surfaces-sized array of mfxFrameSurface1.
+     */
+    AVBufferRef *opaque_surfaces;
+
+    /**
+     * Encoding only, and only if opaque_alloc is set to non-zero. On return
+     * from avcodec_open2(), this field will be set to the surface type used in
+     * the opaque allocation request.
+     */
+    int opaque_alloc_type;
+} AVQSVContext;
+
+/**
+ * Allocate a new context.
+ *
+ * It must be freed by the caller with av_free().
+ */
+AVQSVContext *av_qsv_alloc_context(void);
+
+#endif /* AVCODEC_QSV_H */
@@ -0,0 +1,189 @@
+/*
+ * Video Acceleration API (shared data between FFmpeg and the video player)
+ * HW decode acceleration for MPEG-2, MPEG-4, H.264 and VC-1
+ *
+ * Copyright (C) 2008-2009 Splitted-Desktop Systems
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_VAAPI_H
+#define AVCODEC_VAAPI_H
+
+/**
+ * @file
+ * @ingroup lavc_codec_hwaccel_vaapi
+ * Public libavcodec VA API header.
+ */
+
+#include <stdint.h>
+#include "libavutil/attributes.h"
+#include "version.h"
+
+/**
+ * @defgroup lavc_codec_hwaccel_vaapi VA API Decoding
+ * @ingroup lavc_codec_hwaccel
+ * @{
+ */
+
+/**
+ * This structure is used to share data between the FFmpeg library and
+ * the client video application.
+ * This shall be zero-allocated and available as
+ * AVCodecContext.hwaccel_context. All user members can be set once
+ * during initialization or through each AVCodecContext.get_buffer()
+ * function call. In any case, they must be valid prior to calling
+ * decoding functions.
+ */
+struct vaapi_context {
+    /**
+     * Window system dependent data
+     *
+     * - encoding: unused
+     * - decoding: Set by user
+     */
+    void *display;
+
+    /**
+     * Configuration ID
+     *
+     * - encoding: unused
+     * - decoding: Set by user
+     */
+    uint32_t config_id;
+
+    /**
+     * Context ID (video decode pipeline)
+     *
+     * - encoding: unused
+     * - decoding: Set by user
+     */
+    uint32_t context_id;
+
+#if FF_API_VAAPI_CONTEXT
+    /**
+     * VAPictureParameterBuffer ID
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    uint32_t pic_param_buf_id;
+
+    /**
+     * VAIQMatrixBuffer ID
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    uint32_t iq_matrix_buf_id;
+
+    /**
+     * VABitPlaneBuffer ID (for VC-1 decoding)
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    uint32_t bitplane_buf_id;
+
+    /**
+     * Slice parameter/data buffer IDs
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    uint32_t *slice_buf_ids;
+
+    /**
+     * Number of effective slice buffer IDs to send to the HW
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    unsigned int n_slice_buf_ids;
+
+    /**
+     * Size of pre-allocated slice_buf_ids
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    unsigned int slice_buf_ids_alloc;
+
+    /**
+     * Pointer to VASliceParameterBuffers
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    void *slice_params;
+
+    /**
+     * Size of a VASliceParameterBuffer element
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    unsigned int slice_param_size;
+
+    /**
+     * Size of pre-allocated slice_params
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    unsigned int slice_params_alloc;
+
+    /**
+     * Number of slices currently filled in
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    unsigned int slice_count;
+
+    /**
+     * Pointer to slice data buffer base
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    const uint8_t *slice_data;
+
+    /**
+     * Current size of slice data
+     *
+     * - encoding: unused
+     * - decoding: Set by libavcodec
+     */
+    attribute_deprecated
+    uint32_t slice_data_size;
+#endif
+};
+
+/* @} */
+
+#endif /* AVCODEC_VAAPI_H */
@@ -0,0 +1,230 @@
+/*
+ * VDA HW acceleration
+ *
+ * copyright (c) 2011 Sebastien Zwickert
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_VDA_H
+#define AVCODEC_VDA_H
+
+/**
+ * @file
+ * @ingroup lavc_codec_hwaccel_vda
+ * Public libavcodec VDA header.
+ */
+
+#include "libavcodec/avcodec.h"
+
+#include <stdint.h>
+
+// emmintrin.h is unable to compile with -std=c99 -Werror=missing-prototypes
+// http://openradar.appspot.com/8026390
+#undef __GNUC_STDC_INLINE__
+
+#define Picture QuickdrawPicture
+#include <VideoDecodeAcceleration/VDADecoder.h>
+#undef Picture
+
+#include "libavcodec/version.h"
+
+// extra flags not defined in VDADecoder.h
+enum {
+    kVDADecodeInfo_Asynchronous = 1UL << 0,
+    kVDADecodeInfo_FrameDropped = 1UL << 1
+};
+
+/**
+ * @defgroup lavc_codec_hwaccel_vda VDA
+ * @ingroup lavc_codec_hwaccel
+ *
+ * @{
+ */
+
+/**
+ * This structure is used to provide the necessary configurations and data
+ * to the VDA FFmpeg HWAccel implementation.
+ *
+ * The application must make it available as AVCodecContext.hwaccel_context.
+ */
+struct vda_context {
+    /**
+     * VDA decoder object.
+     *
+     * - encoding: unused
+     * - decoding: Set/Unset by libavcodec.
+     */
+    VDADecoder          decoder;
+
+    /**
+     * The Core Video pixel buffer that contains the current image data.
+     *
+     * encoding: unused
+     * decoding: Set by libavcodec. Unset by user.
+     */
+    CVPixelBufferRef    cv_buffer;
+
+    /**
+     * Use the hardware decoder in synchronous mode.
+     *
+     * encoding: unused
+     * decoding: Set by user.
+     */
+    int                 use_sync_decoding;
+
+    /**
+     * The frame width.
+     *
+     * - encoding: unused
+     * - decoding: Set/Unset by user.
+     */
+    int                 width;
+
+    /**
+     * The frame height.
+     *
+     * - encoding: unused
+     * - decoding: Set/Unset by user.
+     */
+    int                 height;
+
+    /**
+     * The frame format.
+     *
+     * - encoding: unused
+     * - decoding: Set/Unset by user.
+     */
+    int                 format;
+
+    /**
+     * The pixel format for output image buffers.
+     *
+     * - encoding: unused
+     * - decoding: Set/Unset by user.
+     */
+    OSType              cv_pix_fmt_type;
+
+    /**
+     * unused
+     */
+    uint8_t             *priv_bitstream;
+
+    /**
+     * unused
+     */
+    int                 priv_bitstream_size;
+
+    /**
+     * unused
+     */
+    int                 priv_allocated_size;
+
+    /**
+     * Use av_buffer to manage buffer.
+     * When the flag is set, the CVPixelBuffers returned by the decoder will
+     * be released automatically, so you have to retain them if necessary.
+     * Not setting this flag may cause memory leak.
+     *
+     * encoding: unused
+     * decoding: Set by user.
+     */
+    int                 use_ref_buffer;
+};
+
+/** Create the video decoder. */
+int ff_vda_create_decoder(struct vda_context *vda_ctx,
+                          uint8_t *extradata,
+                          int extradata_size);
+
+/** Destroy the video decoder. */
+int ff_vda_destroy_decoder(struct vda_context *vda_ctx);
+
+/**
+ * This struct holds all the information that needs to be passed
+ * between the caller and libavcodec for initializing VDA decoding.
+ * Its size is not a part of the public ABI, it must be allocated with
+ * av_vda_alloc_context() and freed with av_free().
+ */
+typedef struct AVVDAContext {
+    /**
+     * VDA decoder object. Created and freed by the caller.
+     */
+    VDADecoder decoder;
+
+    /**
+     * The output callback that must be passed to VDADecoderCreate.
+     * Set by av_vda_alloc_context().
+     */
+    VDADecoderOutputCallback output_callback;
+
+    /**
+     * CVPixelBuffer Format Type that VDA will use for decoded frames; set by
+     * the caller.
+     */
+    OSType cv_pix_fmt_type;
+} AVVDAContext;
+
+/**
+ * Allocate and initialize a VDA context.
+ *
+ * This function should be called from the get_format() callback when the caller
+ * selects the AV_PIX_FMT_VDA format. The caller must then create the decoder
+ * object (using the output callback provided by libavcodec) that will be used
+ * for VDA-accelerated decoding.
+ *
+ * When decoding with VDA is finished, the caller must destroy the decoder
+ * object and free the VDA context using av_free().
+ *
+ * @return the newly allocated context or NULL on failure
+ */
+AVVDAContext *av_vda_alloc_context(void);
+
+/**
+ * This is a convenience function that creates and sets up the VDA context using
+ * an internal implementation.
+ *
+ * @param avctx the corresponding codec context
+ *
+ * @return >= 0 on success, a negative AVERROR code on failure
+ */
+int av_vda_default_init(AVCodecContext *avctx);
+
+/**
+ * This is a convenience function that creates and sets up the VDA context using
+ * an internal implementation.
+ *
+ * @param avctx the corresponding codec context
+ * @param vdactx the VDA context to use
+ *
+ * @return >= 0 on success, a negative AVERROR code on failure
+ */
+int av_vda_default_init2(AVCodecContext *avctx, AVVDAContext *vdactx);
+
+/**
+ * This function must be called to free the VDA context initialized with
+ * av_vda_default_init().
+ *
+ * @param avctx the corresponding codec context
+ */
+void av_vda_default_free(AVCodecContext *avctx);
+
+/**
+ * @}
+ */
+
+#endif /* AVCODEC_VDA_H */
@@ -0,0 +1,253 @@
+/*
+ * The Video Decode and Presentation API for UNIX (VDPAU) is used for
+ * hardware-accelerated decoding of MPEG-1/2, H.264 and VC-1.
+ *
+ * Copyright (C) 2008 NVIDIA
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_VDPAU_H
+#define AVCODEC_VDPAU_H
+
+/**
+ * @file
+ * @ingroup lavc_codec_hwaccel_vdpau
+ * Public libavcodec VDPAU header.
+ */
+
+
+/**
+ * @defgroup lavc_codec_hwaccel_vdpau VDPAU Decoder and Renderer
+ * @ingroup lavc_codec_hwaccel
+ *
+ * VDPAU hardware acceleration has two modules
+ * - VDPAU decoding
+ * - VDPAU presentation
+ *
+ * The VDPAU decoding module parses all headers using FFmpeg
+ * parsing mechanisms and uses VDPAU for the actual decoding.
+ *
+ * As per the current implementation, the actual decoding
+ * and rendering (API calls) are done as part of the VDPAU
+ * presentation (vo_vdpau.c) module.
+ *
+ * @{
+ */
+
+#include <vdpau/vdpau.h>
+#include <vdpau/vdpau_x11.h>
+#include "libavutil/avconfig.h"
+#include "libavutil/attributes.h"
+
+#include "avcodec.h"
+#include "version.h"
+
+#if FF_API_BUFS_VDPAU
+union AVVDPAUPictureInfo {
+    VdpPictureInfoH264        h264;
+    VdpPictureInfoMPEG1Or2    mpeg;
+    VdpPictureInfoVC1          vc1;
+    VdpPictureInfoMPEG4Part2 mpeg4;
+};
+#endif
+
+struct AVCodecContext;
+struct AVFrame;
+
+typedef int (*AVVDPAU_Render2)(struct AVCodecContext *, struct AVFrame *,
+                               const VdpPictureInfo *, uint32_t,
+                               const VdpBitstreamBuffer *);
+
+/**
+ * This structure is used to share data between the libavcodec library and
+ * the client video application.
+ * The user shall allocate the structure via the av_alloc_vdpau_hwaccel
+ * function and make it available as
+ * AVCodecContext.hwaccel_context. Members can be set by the user once
+ * during initialization or through each AVCodecContext.get_buffer()
+ * function call. In any case, they must be valid prior to calling
+ * decoding functions.
+ *
+ * The size of this structure is not a part of the public ABI and must not
+ * be used outside of libavcodec. Use av_vdpau_alloc_context() to allocate an
+ * AVVDPAUContext.
+ */
+typedef struct AVVDPAUContext {
+    /**
+     * VDPAU decoder handle
+     *
+     * Set by user.
+     */
+    VdpDecoder decoder;
+
+    /**
+     * VDPAU decoder render callback
+     *
+     * Set by the user.
+     */
+    VdpDecoderRender *render;
+
+#if FF_API_BUFS_VDPAU
+    /**
+     * VDPAU picture information
+     *
+     * Set by libavcodec.
+     */
+    attribute_deprecated
+    union AVVDPAUPictureInfo info;
+
+    /**
+     * Allocated size of the bitstream_buffers table.
+     *
+     * Set by libavcodec.
+     */
+    attribute_deprecated
+    int bitstream_buffers_allocated;
+
+    /**
+     * Useful bitstream buffers in the bitstream buffers table.
+     *
+     * Set by libavcodec.
+     */
+    attribute_deprecated
+    int bitstream_buffers_used;
+
+   /**
+     * Table of bitstream buffers.
+     * The user is responsible for freeing this buffer using av_freep().
+     *
+     * Set by libavcodec.
+     */
+    attribute_deprecated
+    VdpBitstreamBuffer *bitstream_buffers;
+#endif
+    AVVDPAU_Render2 render2;
+} AVVDPAUContext;
+
+/**
+ * @brief allocation function for AVVDPAUContext
+ *
+ * Allows extending the struct without breaking API/ABI
+ */
+AVVDPAUContext *av_alloc_vdpaucontext(void);
+
+AVVDPAU_Render2 av_vdpau_hwaccel_get_render2(const AVVDPAUContext *);
+void av_vdpau_hwaccel_set_render2(AVVDPAUContext *, AVVDPAU_Render2);
+
+/**
+ * Associate a VDPAU device with a codec context for hardware acceleration.
+ * This function is meant to be called from the get_format() codec callback,
+ * or earlier. It can also be called after avcodec_flush_buffers() to change
+ * the underlying VDPAU device mid-stream (e.g. to recover from non-transparent
+ * display preemption).
+ *
+ * @note get_format() must return AV_PIX_FMT_VDPAU if this function completes
+ * successfully.
+ *
+ * @param avctx decoding context whose get_format() callback is invoked
+ * @param device VDPAU device handle to use for hardware acceleration
+ * @param get_proc_address VDPAU device driver
+ * @param flags zero of more OR'd AV_HWACCEL_FLAG_* flags
+ *
+ * @return 0 on success, an AVERROR code on failure.
+ */
+int av_vdpau_bind_context(AVCodecContext *avctx, VdpDevice device,
+                          VdpGetProcAddress *get_proc_address, unsigned flags);
+
+/**
+ * Gets the parameters to create an adequate VDPAU video surface for the codec
+ * context using VDPAU hardware decoding acceleration.
+ *
+ * @note Behavior is undefined if the context was not successfully bound to a
+ * VDPAU device using av_vdpau_bind_context().
+ *
+ * @param avctx the codec context being used for decoding the stream
+ * @param type storage space for the VDPAU video surface chroma type
+ *              (or NULL to ignore)
+ * @param width storage space for the VDPAU video surface pixel width
+ *              (or NULL to ignore)
+ * @param height storage space for the VDPAU video surface pixel height
+ *              (or NULL to ignore)
+ *
+ * @return 0 on success, a negative AVERROR code on failure.
+ */
+int av_vdpau_get_surface_parameters(AVCodecContext *avctx, VdpChromaType *type,
+                                    uint32_t *width, uint32_t *height);
+
+/**
+ * Allocate an AVVDPAUContext.
+ *
+ * @return Newly-allocated AVVDPAUContext or NULL on failure.
+ */
+AVVDPAUContext *av_vdpau_alloc_context(void);
+
+#if FF_API_VDPAU_PROFILE
+/**
+ * Get a decoder profile that should be used for initializing a VDPAU decoder.
+ * Should be called from the AVCodecContext.get_format() callback.
+ *
+ * @deprecated Use av_vdpau_bind_context() instead.
+ *
+ * @param avctx the codec context being used for decoding the stream
+ * @param profile a pointer into which the result will be written on success.
+ *                The contents of profile are undefined if this function returns
+ *                an error.
+ *
+ * @return 0 on success (non-negative), a negative AVERROR on failure.
+ */
+attribute_deprecated
+int av_vdpau_get_profile(AVCodecContext *avctx, VdpDecoderProfile *profile);
+#endif
+
+#if FF_API_CAP_VDPAU
+/** @brief The videoSurface is used for rendering. */
+#define FF_VDPAU_STATE_USED_FOR_RENDER 1
+
+/**
+ * @brief The videoSurface is needed for reference/prediction.
+ * The codec manipulates this.
+ */
+#define FF_VDPAU_STATE_USED_FOR_REFERENCE 2
+
+/**
+ * @brief This structure is used as a callback between the FFmpeg
+ * decoder (vd_) and presentation (vo_) module.
+ * This is used for defining a video frame containing surface,
+ * picture parameter, bitstream information etc which are passed
+ * between the FFmpeg decoder and its clients.
+ */
+struct vdpau_render_state {
+    VdpVideoSurface surface; ///< Used as rendered surface, never changed.
+
+    int state; ///< Holds FF_VDPAU_STATE_* values.
+
+    /** picture parameter information for all supported codecs */
+    union AVVDPAUPictureInfo info;
+
+    /** Describe size/location of the compressed video data.
+        Set to 0 when freeing bitstream_buffers. */
+    int bitstream_buffers_allocated;
+    int bitstream_buffers_used;
+    /** The user is responsible for freeing this buffer using av_freep(). */
+    VdpBitstreamBuffer *bitstream_buffers;
+};
+#endif
+
+/* @}*/
+
+#endif /* AVCODEC_VDPAU_H */
@@ -0,0 +1,198 @@
+/*
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_VERSION_H
+#define AVCODEC_VERSION_H
+
+/**
+ * @file
+ * @ingroup libavc
+ * Libavcodec version macros.
+ */
+
+#include "libavutil/version.h"
+
+#define LIBAVCODEC_VERSION_MAJOR  57
+#define LIBAVCODEC_VERSION_MINOR  14
+#define LIBAVCODEC_VERSION_MICRO 100
+
+#define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
+                                               LIBAVCODEC_VERSION_MINOR, \
+                                               LIBAVCODEC_VERSION_MICRO)
+#define LIBAVCODEC_VERSION      AV_VERSION(LIBAVCODEC_VERSION_MAJOR,    \
+                                           LIBAVCODEC_VERSION_MINOR,    \
+                                           LIBAVCODEC_VERSION_MICRO)
+#define LIBAVCODEC_BUILD        LIBAVCODEC_VERSION_INT
+
+#define LIBAVCODEC_IDENT        "Lavc" AV_STRINGIFY(LIBAVCODEC_VERSION)
+
+/**
+ * FF_API_* defines may be placed below to indicate public API that will be
+ * dropped at a future version bump. The defines themselves are not part of
+ * the public API and may change, break or disappear at any time.
+ *
+ * @note, when bumping the major version it is recommended to manually
+ * disable each FF_API_* in its own commit instead of disabling them all
+ * at once through the bump. This improves the git bisect-ability of the change.
+ */
+
+#ifndef FF_API_VIMA_DECODER
+#define FF_API_VIMA_DECODER     (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_AUDIO_CONVERT
+#define FF_API_AUDIO_CONVERT     (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_AVCODEC_RESAMPLE
+#define FF_API_AVCODEC_RESAMPLE  FF_API_AUDIO_CONVERT
+#endif
+#ifndef FF_API_MISSING_SAMPLE
+#define FF_API_MISSING_SAMPLE    (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_LOWRES
+#define FF_API_LOWRES            (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_CAP_VDPAU
+#define FF_API_CAP_VDPAU         (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_BUFS_VDPAU
+#define FF_API_BUFS_VDPAU        (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_VOXWARE
+#define FF_API_VOXWARE           (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_SET_DIMENSIONS
+#define FF_API_SET_DIMENSIONS    (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_DEBUG_MV
+#define FF_API_DEBUG_MV          (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_AC_VLC
+#define FF_API_AC_VLC            (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_OLD_MSMPEG4
+#define FF_API_OLD_MSMPEG4       (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_ASPECT_EXTENDED
+#define FF_API_ASPECT_EXTENDED   (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_ARCH_ALPHA
+#define FF_API_ARCH_ALPHA        (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_XVMC
+#define FF_API_XVMC              (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_ERROR_RATE
+#define FF_API_ERROR_RATE        (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_QSCALE_TYPE
+#define FF_API_QSCALE_TYPE       (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_MB_TYPE
+#define FF_API_MB_TYPE           (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_MAX_BFRAMES
+#define FF_API_MAX_BFRAMES       (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_NEG_LINESIZES
+#define FF_API_NEG_LINESIZES     (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_EMU_EDGE
+#define FF_API_EMU_EDGE          (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_ARCH_SH4
+#define FF_API_ARCH_SH4          (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_ARCH_SPARC
+#define FF_API_ARCH_SPARC        (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_UNUSED_MEMBERS
+#define FF_API_UNUSED_MEMBERS    (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_IDCT_XVIDMMX
+#define FF_API_IDCT_XVIDMMX      (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_INPUT_PRESERVED
+#define FF_API_INPUT_PRESERVED   (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_NORMALIZE_AQP
+#define FF_API_NORMALIZE_AQP     (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_GMC
+#define FF_API_GMC               (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_MV0
+#define FF_API_MV0               (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_CODEC_NAME
+#define FF_API_CODEC_NAME        (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_AFD
+#define FF_API_AFD               (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_VISMV
+/* XXX: don't forget to drop the -vismv documentation */
+#define FF_API_VISMV             (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_AUDIOENC_DELAY
+#define FF_API_AUDIOENC_DELAY    (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_VAAPI_CONTEXT
+#define FF_API_VAAPI_CONTEXT     (LIBAVCODEC_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_AVCTX_TIMEBASE
+#define FF_API_AVCTX_TIMEBASE    (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_MPV_OPT
+#define FF_API_MPV_OPT           (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_STREAM_CODEC_TAG
+#define FF_API_STREAM_CODEC_TAG  (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_QUANT_BIAS
+#define FF_API_QUANT_BIAS        (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_RC_STRATEGY
+#define FF_API_RC_STRATEGY       (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_CODED_FRAME
+#define FF_API_CODED_FRAME       (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_MOTION_EST
+#define FF_API_MOTION_EST        (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_WITHOUT_PREFIX
+#define FF_API_WITHOUT_PREFIX    (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_SIDEDATA_ONLY_PKT
+#define FF_API_SIDEDATA_ONLY_PKT (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_VDPAU_PROFILE
+#define FF_API_VDPAU_PROFILE     (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_CONVERGENCE_DURATION
+#define FF_API_CONVERGENCE_DURATION (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_AVPICTURE
+#define FF_API_AVPICTURE         (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+#ifndef FF_API_AVPACKET_OLD_API
+#define FF_API_AVPACKET_OLD_API (LIBAVCODEC_VERSION_MAJOR < 59)
+#endif
+
+#endif /* AVCODEC_VERSION_H */
@@ -0,0 +1,126 @@
+/*
+ * Videotoolbox hardware acceleration
+ *
+ * copyright (c) 2012 Sebastien Zwickert
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_VIDEOTOOLBOX_H
+#define AVCODEC_VIDEOTOOLBOX_H
+
+/**
+ * @file
+ * @ingroup lavc_codec_hwaccel_videotoolbox
+ * Public libavcodec Videotoolbox header.
+ */
+
+#include <stdint.h>
+
+#define Picture QuickdrawPicture
+#include <VideoToolbox/VideoToolbox.h>
+#undef Picture
+
+#include "libavcodec/avcodec.h"
+
+/**
+ * This struct holds all the information that needs to be passed
+ * between the caller and libavcodec for initializing Videotoolbox decoding.
+ * Its size is not a part of the public ABI, it must be allocated with
+ * av_videotoolbox_alloc_context() and freed with av_free().
+ */
+typedef struct AVVideotoolboxContext {
+    /**
+     * Videotoolbox decompression session object.
+     * Created and freed the caller.
+     */
+    VTDecompressionSessionRef session;
+
+    /**
+     * The output callback that must be passed to the session.
+     * Set by av_videottoolbox_default_init()
+     */
+    VTDecompressionOutputCallback output_callback;
+
+    /**
+     * CVPixelBuffer Format Type that Videotoolbox will use for decoded frames.
+     * set by the caller.
+     */
+    OSType cv_pix_fmt_type;
+
+    /**
+     * CoreMedia Format Description that Videotoolbox will use to create the decompression session.
+     * Set by the caller.
+     */
+    CMVideoFormatDescriptionRef cm_fmt_desc;
+
+    /**
+     * CoreMedia codec type that Videotoolbox will use to create the decompression session.
+     * Set by the caller.
+     */
+    int cm_codec_type;
+} AVVideotoolboxContext;
+
+/**
+ * Allocate and initialize a Videotoolbox context.
+ *
+ * This function should be called from the get_format() callback when the caller
+ * selects the AV_PIX_FMT_VIDETOOLBOX format. The caller must then create
+ * the decoder object (using the output callback provided by libavcodec) that
+ * will be used for Videotoolbox-accelerated decoding.
+ *
+ * When decoding with Videotoolbox is finished, the caller must destroy the decoder
+ * object and free the Videotoolbox context using av_free().
+ *
+ * @return the newly allocated context or NULL on failure
+ */
+AVVideotoolboxContext *av_videotoolbox_alloc_context(void);
+
+/**
+ * This is a convenience function that creates and sets up the Videotoolbox context using
+ * an internal implementation.
+ *
+ * @param avctx the corresponding codec context
+ *
+ * @return >= 0 on success, a negative AVERROR code on failure
+ */
+int av_videotoolbox_default_init(AVCodecContext *avctx);
+
+/**
+ * This is a convenience function that creates and sets up the Videotoolbox context using
+ * an internal implementation.
+ *
+ * @param avctx the corresponding codec context
+ * @param vtctx the Videotoolbox context to use
+ *
+ * @return >= 0 on success, a negative AVERROR code on failure
+ */
+int av_videotoolbox_default_init2(AVCodecContext *avctx, AVVideotoolboxContext *vtctx);
+
+/**
+ * This function must be called to free the Videotoolbox context initialized with
+ * av_videotoolbox_default_init().
+ *
+ * @param avctx the corresponding codec context
+ */
+void av_videotoolbox_default_free(AVCodecContext *avctx);
+
+/**
+ * @}
+ */
+
+#endif /* AVCODEC_VIDEOTOOLBOX_H */
@@ -0,0 +1,78 @@
+/*
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * A public API for Vorbis parsing
+ *
+ * Determines the duration for each packet.
+ */
+
+#ifndef AVCODEC_VORBIS_PARSE_H
+#define AVCODEC_VORBIS_PARSE_H
+
+#include <stdint.h>
+
+typedef struct AVVorbisParseContext AVVorbisParseContext;
+
+/**
+ * Allocate and initialize the Vorbis parser using headers in the extradata.
+ *
+ * @param avctx codec context
+ * @param s     Vorbis parser context
+ */
+AVVorbisParseContext *av_vorbis_parse_init(const uint8_t *extradata,
+                                           int extradata_size);
+
+/**
+ * Free the parser and everything associated with it.
+ */
+void av_vorbis_parse_free(AVVorbisParseContext **s);
+
+#define VORBIS_FLAG_HEADER  0x00000001
+#define VORBIS_FLAG_COMMENT 0x00000002
+#define VORBIS_FLAG_SETUP   0x00000004
+
+/**
+ * Get the duration for a Vorbis packet.
+ *
+ * If @p flags is @c NULL,
+ * special frames are considered invalid.
+ *
+ * @param s        Vorbis parser context
+ * @param buf      buffer containing a Vorbis frame
+ * @param buf_size size of the buffer
+ * @param flags    flags for special frames
+ */
+int av_vorbis_parse_frame_flags(AVVorbisParseContext *s, const uint8_t *buf,
+                                int buf_size, int *flags);
+
+/**
+ * Get the duration for a Vorbis packet.
+ *
+ * @param s        Vorbis parser context
+ * @param buf      buffer containing a Vorbis frame
+ * @param buf_size size of the buffer
+ */
+int av_vorbis_parse_frame(AVVorbisParseContext *s, const uint8_t *buf,
+                          int buf_size);
+
+void av_vorbis_parse_reset(AVVorbisParseContext *s);
+
+#endif /* AVCODEC_VORBIS_PARSE_H */
@@ -0,0 +1,170 @@
+/*
+ * Copyright (C) 2003 Ivan Kalvachev
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_XVMC_H
+#define AVCODEC_XVMC_H
+
+/**
+ * @file
+ * @ingroup lavc_codec_hwaccel_xvmc
+ * Public libavcodec XvMC header.
+ */
+
+#include <X11/extensions/XvMC.h>
+
+#include "libavutil/attributes.h"
+#include "version.h"
+#include "avcodec.h"
+
+/**
+ * @defgroup lavc_codec_hwaccel_xvmc XvMC
+ * @ingroup lavc_codec_hwaccel
+ *
+ * @{
+ */
+
+#define AV_XVMC_ID                    0x1DC711C0  /**< special value to ensure that regular pixel routines haven't corrupted the struct
+                                                       the number is 1337 speak for the letters IDCT MCo (motion compensation) */
+
+struct attribute_deprecated xvmc_pix_fmt {
+    /** The field contains the special constant value AV_XVMC_ID.
+        It is used as a test that the application correctly uses the API,
+        and that there is no corruption caused by pixel routines.
+        - application - set during initialization
+        - libavcodec  - unchanged
+    */
+    int             xvmc_id;
+
+    /** Pointer to the block array allocated by XvMCCreateBlocks().
+        The array has to be freed by XvMCDestroyBlocks().
+        Each group of 64 values represents one data block of differential
+        pixel information (in MoCo mode) or coefficients for IDCT.
+        - application - set the pointer during initialization
+        - libavcodec  - fills coefficients/pixel data into the array
+    */
+    short*          data_blocks;
+
+    /** Pointer to the macroblock description array allocated by
+        XvMCCreateMacroBlocks() and freed by XvMCDestroyMacroBlocks().
+        - application - set the pointer during initialization
+        - libavcodec  - fills description data into the array
+    */
+    XvMCMacroBlock* mv_blocks;
+
+    /** Number of macroblock descriptions that can be stored in the mv_blocks
+        array.
+        - application - set during initialization
+        - libavcodec  - unchanged
+    */
+    int             allocated_mv_blocks;
+
+    /** Number of blocks that can be stored at once in the data_blocks array.
+        - application - set during initialization
+        - libavcodec  - unchanged
+    */
+    int             allocated_data_blocks;
+
+    /** Indicate that the hardware would interpret data_blocks as IDCT
+        coefficients and perform IDCT on them.
+        - application - set during initialization
+        - libavcodec  - unchanged
+    */
+    int             idct;
+
+    /** In MoCo mode it indicates that intra macroblocks are assumed to be in
+        unsigned format; same as the XVMC_INTRA_UNSIGNED flag.
+        - application - set during initialization
+        - libavcodec  - unchanged
+    */
+    int             unsigned_intra;
+
+    /** Pointer to the surface allocated by XvMCCreateSurface().
+        It has to be freed by XvMCDestroySurface() on application exit.
+        It identifies the frame and its state on the video hardware.
+        - application - set during initialization
+        - libavcodec  - unchanged
+    */
+    XvMCSurface*    p_surface;
+
+/** Set by the decoder before calling ff_draw_horiz_band(),
+    needed by the XvMCRenderSurface function. */
+//@{
+    /** Pointer to the surface used as past reference
+        - application - unchanged
+        - libavcodec  - set
+    */
+    XvMCSurface*    p_past_surface;
+
+    /** Pointer to the surface used as future reference
+        - application - unchanged
+        - libavcodec  - set
+    */
+    XvMCSurface*    p_future_surface;
+
+    /** top/bottom field or frame
+        - application - unchanged
+        - libavcodec  - set
+    */
+    unsigned int    picture_structure;
+
+    /** XVMC_SECOND_FIELD - 1st or 2nd field in the sequence
+        - application - unchanged
+        - libavcodec  - set
+    */
+    unsigned int    flags;
+//}@
+
+    /** Number of macroblock descriptions in the mv_blocks array
+        that have already been passed to the hardware.
+        - application - zeroes it on get_buffer().
+                        A successful ff_draw_horiz_band() may increment it
+                        with filled_mb_block_num or zero both.
+        - libavcodec  - unchanged
+    */
+    int             start_mv_blocks_num;
+
+    /** Number of new macroblock descriptions in the mv_blocks array (after
+        start_mv_blocks_num) that are filled by libavcodec and have to be
+        passed to the hardware.
+        - application - zeroes it on get_buffer() or after successful
+                        ff_draw_horiz_band().
+        - libavcodec  - increment with one of each stored MB
+    */
+    int             filled_mv_blocks_num;
+
+    /** Number of the next free data block; one data block consists of
+        64 short values in the data_blocks array.
+        All blocks before this one have already been claimed by placing their
+        position into the corresponding block description structure field,
+        that are part of the mv_blocks array.
+        - application - zeroes it on get_buffer().
+                        A successful ff_draw_horiz_band() may zero it together
+                        with start_mb_blocks_num.
+        - libavcodec  - each decoded macroblock increases it by the number
+                        of coded blocks it contains.
+    */
+    int             next_free_data_block_num;
+};
+
+/**
+ * @}
+ */
+
+#endif /* AVCODEC_XVMC_H */
@@ -0,0 +1,509 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVDEVICE_AVDEVICE_H
+#define AVDEVICE_AVDEVICE_H
+
+#include "version.h"
+
+/**
+ * @file
+ * @ingroup lavd
+ * Main libavdevice API header
+ */
+
+/**
+ * @defgroup lavd Special devices muxing/demuxing library
+ * @{
+ * Libavdevice is a complementary library to @ref libavf "libavformat". It
+ * provides various "special" platform-specific muxers and demuxers, e.g. for
+ * grabbing devices, audio capture and playback etc. As a consequence, the
+ * (de)muxers in libavdevice are of the AVFMT_NOFILE type (they use their own
+ * I/O functions). The filename passed to avformat_open_input() often does not
+ * refer to an actually existing file, but has some special device-specific
+ * meaning - e.g. for x11grab it is the display name.
+ *
+ * To use libavdevice, simply call avdevice_register_all() to register all
+ * compiled muxers and demuxers. They all use standard libavformat API.
+ * @}
+ */
+
+#include "libavutil/log.h"
+#include "libavutil/opt.h"
+#include "libavutil/dict.h"
+#include "libavformat/avformat.h"
+
+/**
+ * Return the LIBAVDEVICE_VERSION_INT constant.
+ */
+unsigned avdevice_version(void);
+
+/**
+ * Return the libavdevice build-time configuration.
+ */
+const char *avdevice_configuration(void);
+
+/**
+ * Return the libavdevice license.
+ */
+const char *avdevice_license(void);
+
+/**
+ * Initialize libavdevice and register all the input and output devices.
+ * @warning This function is not thread safe.
+ */
+void avdevice_register_all(void);
+
+/**
+ * Audio input devices iterator.
+ *
+ * If d is NULL, returns the first registered input audio/video device,
+ * if d is non-NULL, returns the next registered input audio/video device after d
+ * or NULL if d is the last one.
+ */
+AVInputFormat *av_input_audio_device_next(AVInputFormat  *d);
+
+/**
+ * Video input devices iterator.
+ *
+ * If d is NULL, returns the first registered input audio/video device,
+ * if d is non-NULL, returns the next registered input audio/video device after d
+ * or NULL if d is the last one.
+ */
+AVInputFormat *av_input_video_device_next(AVInputFormat  *d);
+
+/**
+ * Audio output devices iterator.
+ *
+ * If d is NULL, returns the first registered output audio/video device,
+ * if d is non-NULL, returns the next registered output audio/video device after d
+ * or NULL if d is the last one.
+ */
+AVOutputFormat *av_output_audio_device_next(AVOutputFormat *d);
+
+/**
+ * Video output devices iterator.
+ *
+ * If d is NULL, returns the first registered output audio/video device,
+ * if d is non-NULL, returns the next registered output audio/video device after d
+ * or NULL if d is the last one.
+ */
+AVOutputFormat *av_output_video_device_next(AVOutputFormat *d);
+
+typedef struct AVDeviceRect {
+    int x;      /**< x coordinate of top left corner */
+    int y;      /**< y coordinate of top left corner */
+    int width;  /**< width */
+    int height; /**< height */
+} AVDeviceRect;
+
+/**
+ * Message types used by avdevice_app_to_dev_control_message().
+ */
+enum AVAppToDevMessageType {
+    /**
+     * Dummy message.
+     */
+    AV_APP_TO_DEV_NONE = MKBETAG('N','O','N','E'),
+
+    /**
+     * Window size change message.
+     *
+     * Message is sent to the device every time the application changes the size
+     * of the window device renders to.
+     * Message should also be sent right after window is created.
+     *
+     * data: AVDeviceRect: new window size.
+     */
+    AV_APP_TO_DEV_WINDOW_SIZE = MKBETAG('G','E','O','M'),
+
+    /**
+     * Repaint request message.
+     *
+     * Message is sent to the device when window has to be repainted.
+     *
+     * data: AVDeviceRect: area required to be repainted.
+     *       NULL: whole area is required to be repainted.
+     */
+    AV_APP_TO_DEV_WINDOW_REPAINT = MKBETAG('R','E','P','A'),
+
+    /**
+     * Request pause/play.
+     *
+     * Application requests pause/unpause playback.
+     * Mostly usable with devices that have internal buffer.
+     * By default devices are not paused.
+     *
+     * data: NULL
+     */
+    AV_APP_TO_DEV_PAUSE        = MKBETAG('P', 'A', 'U', ' '),
+    AV_APP_TO_DEV_PLAY         = MKBETAG('P', 'L', 'A', 'Y'),
+    AV_APP_TO_DEV_TOGGLE_PAUSE = MKBETAG('P', 'A', 'U', 'T'),
+
+    /**
+     * Volume control message.
+     *
+     * Set volume level. It may be device-dependent if volume
+     * is changed per stream or system wide. Per stream volume
+     * change is expected when possible.
+     *
+     * data: double: new volume with range of 0.0 - 1.0.
+     */
+    AV_APP_TO_DEV_SET_VOLUME = MKBETAG('S', 'V', 'O', 'L'),
+
+    /**
+     * Mute control messages.
+     *
+     * Change mute state. It may be device-dependent if mute status
+     * is changed per stream or system wide. Per stream mute status
+     * change is expected when possible.
+     *
+     * data: NULL.
+     */
+    AV_APP_TO_DEV_MUTE        = MKBETAG(' ', 'M', 'U', 'T'),
+    AV_APP_TO_DEV_UNMUTE      = MKBETAG('U', 'M', 'U', 'T'),
+    AV_APP_TO_DEV_TOGGLE_MUTE = MKBETAG('T', 'M', 'U', 'T'),
+
+    /**
+     * Get volume/mute messages.
+     *
+     * Force the device to send AV_DEV_TO_APP_VOLUME_LEVEL_CHANGED or
+     * AV_DEV_TO_APP_MUTE_STATE_CHANGED command respectively.
+     *
+     * data: NULL.
+     */
+    AV_APP_TO_DEV_GET_VOLUME = MKBETAG('G', 'V', 'O', 'L'),
+    AV_APP_TO_DEV_GET_MUTE   = MKBETAG('G', 'M', 'U', 'T'),
+};
+
+/**
+ * Message types used by avdevice_dev_to_app_control_message().
+ */
+enum AVDevToAppMessageType {
+    /**
+     * Dummy message.
+     */
+    AV_DEV_TO_APP_NONE = MKBETAG('N','O','N','E'),
+
+    /**
+     * Create window buffer message.
+     *
+     * Device requests to create a window buffer. Exact meaning is device-
+     * and application-dependent. Message is sent before rendering first
+     * frame and all one-shot initializations should be done here.
+     * Application is allowed to ignore preferred window buffer size.
+     *
+     * @note: Application is obligated to inform about window buffer size
+     *        with AV_APP_TO_DEV_WINDOW_SIZE message.
+     *
+     * data: AVDeviceRect: preferred size of the window buffer.
+     *       NULL: no preferred size of the window buffer.
+     */
+    AV_DEV_TO_APP_CREATE_WINDOW_BUFFER = MKBETAG('B','C','R','E'),
+
+    /**
+     * Prepare window buffer message.
+     *
+     * Device requests to prepare a window buffer for rendering.
+     * Exact meaning is device- and application-dependent.
+     * Message is sent before rendering of each frame.
+     *
+     * data: NULL.
+     */
+    AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER = MKBETAG('B','P','R','E'),
+
+    /**
+     * Display window buffer message.
+     *
+     * Device requests to display a window buffer.
+     * Message is sent when new frame is ready to be displayed.
+     * Usually buffers need to be swapped in handler of this message.
+     *
+     * data: NULL.
+     */
+    AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER = MKBETAG('B','D','I','S'),
+
+    /**
+     * Destroy window buffer message.
+     *
+     * Device requests to destroy a window buffer.
+     * Message is sent when device is about to be destroyed and window
+     * buffer is not required anymore.
+     *
+     * data: NULL.
+     */
+    AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER = MKBETAG('B','D','E','S'),
+
+    /**
+     * Buffer fullness status messages.
+     *
+     * Device signals buffer overflow/underflow.
+     *
+     * data: NULL.
+     */
+    AV_DEV_TO_APP_BUFFER_OVERFLOW = MKBETAG('B','O','F','L'),
+    AV_DEV_TO_APP_BUFFER_UNDERFLOW = MKBETAG('B','U','F','L'),
+
+    /**
+     * Buffer readable/writable.
+     *
+     * Device informs that buffer is readable/writable.
+     * When possible, device informs how many bytes can be read/write.
+     *
+     * @warning Device may not inform when number of bytes than can be read/write changes.
+     *
+     * data: int64_t: amount of bytes available to read/write.
+     *       NULL: amount of bytes available to read/write is not known.
+     */
+    AV_DEV_TO_APP_BUFFER_READABLE = MKBETAG('B','R','D',' '),
+    AV_DEV_TO_APP_BUFFER_WRITABLE = MKBETAG('B','W','R',' '),
+
+    /**
+     * Mute state change message.
+     *
+     * Device informs that mute state has changed.
+     *
+     * data: int: 0 for not muted state, non-zero for muted state.
+     */
+    AV_DEV_TO_APP_MUTE_STATE_CHANGED = MKBETAG('C','M','U','T'),
+
+    /**
+     * Volume level change message.
+     *
+     * Device informs that volume level has changed.
+     *
+     * data: double: new volume with range of 0.0 - 1.0.
+     */
+    AV_DEV_TO_APP_VOLUME_LEVEL_CHANGED = MKBETAG('C','V','O','L'),
+};
+
+/**
+ * Send control message from application to device.
+ *
+ * @param s         device context.
+ * @param type      message type.
+ * @param data      message data. Exact type depends on message type.
+ * @param data_size size of message data.
+ * @return >= 0 on success, negative on error.
+ *         AVERROR(ENOSYS) when device doesn't implement handler of the message.
+ */
+int avdevice_app_to_dev_control_message(struct AVFormatContext *s,
+                                        enum AVAppToDevMessageType type,
+                                        void *data, size_t data_size);
+
+/**
+ * Send control message from device to application.
+ *
+ * @param s         device context.
+ * @param type      message type.
+ * @param data      message data. Can be NULL.
+ * @param data_size size of message data.
+ * @return >= 0 on success, negative on error.
+ *         AVERROR(ENOSYS) when application doesn't implement handler of the message.
+ */
+int avdevice_dev_to_app_control_message(struct AVFormatContext *s,
+                                        enum AVDevToAppMessageType type,
+                                        void *data, size_t data_size);
+
+/**
+ * Following API allows user to probe device capabilities (supported codecs,
+ * pixel formats, sample formats, resolutions, channel counts, etc).
+ * It is build on top op AVOption API.
+ * Queried capabilities make it possible to set up converters of video or audio
+ * parameters that fit to the device.
+ *
+ * List of capabilities that can be queried:
+ *  - Capabilities valid for both audio and video devices:
+ *    - codec:          supported audio/video codecs.
+ *                      type: AV_OPT_TYPE_INT (AVCodecID value)
+ *  - Capabilities valid for audio devices:
+ *    - sample_format:  supported sample formats.
+ *                      type: AV_OPT_TYPE_INT (AVSampleFormat value)
+ *    - sample_rate:    supported sample rates.
+ *                      type: AV_OPT_TYPE_INT
+ *    - channels:       supported number of channels.
+ *                      type: AV_OPT_TYPE_INT
+ *    - channel_layout: supported channel layouts.
+ *                      type: AV_OPT_TYPE_INT64
+ *  - Capabilities valid for video devices:
+ *    - pixel_format:   supported pixel formats.
+ *                      type: AV_OPT_TYPE_INT (AVPixelFormat value)
+ *    - window_size:    supported window sizes (describes size of the window size presented to the user).
+ *                      type: AV_OPT_TYPE_IMAGE_SIZE
+ *    - frame_size:     supported frame sizes (describes size of provided video frames).
+ *                      type: AV_OPT_TYPE_IMAGE_SIZE
+ *    - fps:            supported fps values
+ *                      type: AV_OPT_TYPE_RATIONAL
+ *
+ * Value of the capability may be set by user using av_opt_set() function
+ * and AVDeviceCapabilitiesQuery object. Following queries will
+ * limit results to the values matching already set capabilities.
+ * For example, setting a codec may impact number of formats or fps values
+ * returned during next query. Setting invalid value may limit results to zero.
+ *
+ * Example of the usage basing on opengl output device:
+ *
+ * @code
+ *  AVFormatContext *oc = NULL;
+ *  AVDeviceCapabilitiesQuery *caps = NULL;
+ *  AVOptionRanges *ranges;
+ *  int ret;
+ *
+ *  if ((ret = avformat_alloc_output_context2(&oc, NULL, "opengl", NULL)) < 0)
+ *      goto fail;
+ *  if (avdevice_capabilities_create(&caps, oc, NULL) < 0)
+ *      goto fail;
+ *
+ *  //query codecs
+ *  if (av_opt_query_ranges(&ranges, caps, "codec", AV_OPT_MULTI_COMPONENT_RANGE)) < 0)
+ *      goto fail;
+ *  //pick codec here and set it
+ *  av_opt_set(caps, "codec", AV_CODEC_ID_RAWVIDEO, 0);
+ *
+ *  //query format
+ *  if (av_opt_query_ranges(&ranges, caps, "pixel_format", AV_OPT_MULTI_COMPONENT_RANGE)) < 0)
+ *      goto fail;
+ *  //pick format here and set it
+ *  av_opt_set(caps, "pixel_format", AV_PIX_FMT_YUV420P, 0);
+ *
+ *  //query and set more capabilities
+ *
+ * fail:
+ *  //clean up code
+ *  avdevice_capabilities_free(&query, oc);
+ *  avformat_free_context(oc);
+ * @endcode
+ */
+
+/**
+ * Structure describes device capabilities.
+ *
+ * It is used by devices in conjunction with av_device_capabilities AVOption table
+ * to implement capabilities probing API based on AVOption API. Should not be used directly.
+ */
+typedef struct AVDeviceCapabilitiesQuery {
+    const AVClass *av_class;
+    AVFormatContext *device_context;
+    enum AVCodecID codec;
+    enum AVSampleFormat sample_format;
+    enum AVPixelFormat pixel_format;
+    int sample_rate;
+    int channels;
+    int64_t channel_layout;
+    int window_width;
+    int window_height;
+    int frame_width;
+    int frame_height;
+    AVRational fps;
+} AVDeviceCapabilitiesQuery;
+
+/**
+ * AVOption table used by devices to implement device capabilities API. Should not be used by a user.
+ */
+extern const AVOption av_device_capabilities[];
+
+/**
+ * Initialize capabilities probing API based on AVOption API.
+ *
+ * avdevice_capabilities_free() must be called when query capabilities API is
+ * not used anymore.
+ *
+ * @param[out] caps      Device capabilities data. Pointer to a NULL pointer must be passed.
+ * @param s              Context of the device.
+ * @param device_options An AVDictionary filled with device-private options.
+ *                       On return this parameter will be destroyed and replaced with a dict
+ *                       containing options that were not found. May be NULL.
+ *                       The same options must be passed later to avformat_write_header() for output
+ *                       devices or avformat_open_input() for input devices, or at any other place
+ *                       that affects device-private options.
+ *
+ * @return >= 0 on success, negative otherwise.
+ */
+int avdevice_capabilities_create(AVDeviceCapabilitiesQuery **caps, AVFormatContext *s,
+                                 AVDictionary **device_options);
+
+/**
+ * Free resources created by avdevice_capabilities_create()
+ *
+ * @param caps Device capabilities data to be freed.
+ * @param s    Context of the device.
+ */
+void avdevice_capabilities_free(AVDeviceCapabilitiesQuery **caps, AVFormatContext *s);
+
+/**
+ * Structure describes basic parameters of the device.
+ */
+typedef struct AVDeviceInfo {
+    char *device_name;                   /**< device name, format depends on device */
+    char *device_description;            /**< human friendly name */
+} AVDeviceInfo;
+
+/**
+ * List of devices.
+ */
+typedef struct AVDeviceInfoList {
+    AVDeviceInfo **devices;              /**< list of autodetected devices */
+    int nb_devices;                      /**< number of autodetected devices */
+    int default_device;                  /**< index of default device or -1 if no default */
+} AVDeviceInfoList;
+
+/**
+ * List devices.
+ *
+ * Returns available device names and their parameters.
+ *
+ * @note: Some devices may accept system-dependent device names that cannot be
+ *        autodetected. The list returned by this function cannot be assumed to
+ *        be always completed.
+ *
+ * @param s                device context.
+ * @param[out] device_list list of autodetected devices.
+ * @return count of autodetected devices, negative on error.
+ */
+int avdevice_list_devices(struct AVFormatContext *s, AVDeviceInfoList **device_list);
+
+/**
+ * Convenient function to free result of avdevice_list_devices().
+ *
+ * @param devices device list to be freed.
+ */
+void avdevice_free_list_devices(AVDeviceInfoList **device_list);
+
+/**
+ * List devices.
+ *
+ * Returns available device names and their parameters.
+ * These are convinient wrappers for avdevice_list_devices().
+ * Device context is allocated and deallocated internally.
+ *
+ * @param device           device format. May be NULL if device name is set.
+ * @param device_name      device name. May be NULL if device format is set.
+ * @param device_options   An AVDictionary filled with device-private options. May be NULL.
+ *                         The same options must be passed later to avformat_write_header() for output
+ *                         devices or avformat_open_input() for input devices, or at any other place
+ *                         that affects device-private options.
+ * @param[out] device_list list of autodetected devices
+ * @return count of autodetected devices, negative on error.
+ * @note device argument takes precedence over device_name when both are set.
+ */
+int avdevice_list_input_sources(struct AVInputFormat *device, const char *device_name,
+                                AVDictionary *device_options, AVDeviceInfoList **device_list);
+int avdevice_list_output_sinks(struct AVOutputFormat *device, const char *device_name,
+                               AVDictionary *device_options, AVDeviceInfoList **device_list);
+
+#endif /* AVDEVICE_AVDEVICE_H */
@@ -0,0 +1,50 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVDEVICE_VERSION_H
+#define AVDEVICE_VERSION_H
+
+/**
+ * @file
+ * @ingroup lavd
+ * Libavdevice version macros
+ */
+
+#include "libavutil/version.h"
+
+#define LIBAVDEVICE_VERSION_MAJOR  57
+#define LIBAVDEVICE_VERSION_MINOR   0
+#define LIBAVDEVICE_VERSION_MICRO 100
+
+#define LIBAVDEVICE_VERSION_INT AV_VERSION_INT(LIBAVDEVICE_VERSION_MAJOR, \
+                                               LIBAVDEVICE_VERSION_MINOR, \
+                                               LIBAVDEVICE_VERSION_MICRO)
+#define LIBAVDEVICE_VERSION     AV_VERSION(LIBAVDEVICE_VERSION_MAJOR, \
+                                           LIBAVDEVICE_VERSION_MINOR, \
+                                           LIBAVDEVICE_VERSION_MICRO)
+#define LIBAVDEVICE_BUILD       LIBAVDEVICE_VERSION_INT
+
+#define LIBAVDEVICE_IDENT       "Lavd" AV_STRINGIFY(LIBAVDEVICE_VERSION)
+
+/**
+ * FF_API_* defines may be placed below to indicate public API that will be
+ * dropped at a future version bump. The defines themselves are not part of
+ * the public API and may change, break or disappear at any time.
+ */
+
+#endif /* AVDEVICE_VERSION_H */
@@ -0,0 +1,28 @@
+/*
+ * Filter graphs
+ * copyright (c) 2007 Bobby Bingham
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFILTER_AVFILTERGRAPH_H
+#define AVFILTER_AVFILTERGRAPH_H
+
+#include "avfilter.h"
+#include "libavutil/log.h"
+
+#endif /* AVFILTER_AVFILTERGRAPH_H */
@@ -0,0 +1,147 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFILTER_BUFFERSINK_H
+#define AVFILTER_BUFFERSINK_H
+
+/**
+ * @file
+ * @ingroup lavfi_buffersink
+ * memory buffer sink API for audio and video
+ */
+
+#include "avfilter.h"
+
+/**
+ * @defgroup lavfi_buffersink Buffer sink API
+ * @ingroup lavfi
+ * @{
+ */
+
+/**
+ * Get a frame with filtered data from sink and put it in frame.
+ *
+ * @param ctx    pointer to a buffersink or abuffersink filter context.
+ * @param frame  pointer to an allocated frame that will be filled with data.
+ *               The data must be freed using av_frame_unref() / av_frame_free()
+ * @param flags  a combination of AV_BUFFERSINK_FLAG_* flags
+ *
+ * @return  >= 0 in for success, a negative AVERROR code for failure.
+ */
+int av_buffersink_get_frame_flags(AVFilterContext *ctx, AVFrame *frame, int flags);
+
+/**
+ * Tell av_buffersink_get_buffer_ref() to read video/samples buffer
+ * reference, but not remove it from the buffer. This is useful if you
+ * need only to read a video/samples buffer, without to fetch it.
+ */
+#define AV_BUFFERSINK_FLAG_PEEK 1
+
+/**
+ * Tell av_buffersink_get_buffer_ref() not to request a frame from its input.
+ * If a frame is already buffered, it is read (and removed from the buffer),
+ * but if no frame is present, return AVERROR(EAGAIN).
+ */
+#define AV_BUFFERSINK_FLAG_NO_REQUEST 2
+
+/**
+ * Struct to use for initializing a buffersink context.
+ */
+typedef struct {
+    const enum AVPixelFormat *pixel_fmts; ///< list of allowed pixel formats, terminated by AV_PIX_FMT_NONE
+} AVBufferSinkParams;
+
+/**
+ * Create an AVBufferSinkParams structure.
+ *
+ * Must be freed with av_free().
+ */
+AVBufferSinkParams *av_buffersink_params_alloc(void);
+
+/**
+ * Struct to use for initializing an abuffersink context.
+ */
+typedef struct {
+    const enum AVSampleFormat *sample_fmts; ///< list of allowed sample formats, terminated by AV_SAMPLE_FMT_NONE
+    const int64_t *channel_layouts;         ///< list of allowed channel layouts, terminated by -1
+    const int *channel_counts;              ///< list of allowed channel counts, terminated by -1
+    int all_channel_counts;                 ///< if not 0, accept any channel count or layout
+    int *sample_rates;                      ///< list of allowed sample rates, terminated by -1
+} AVABufferSinkParams;
+
+/**
+ * Create an AVABufferSinkParams structure.
+ *
+ * Must be freed with av_free().
+ */
+AVABufferSinkParams *av_abuffersink_params_alloc(void);
+
+/**
+ * Set the frame size for an audio buffer sink.
+ *
+ * All calls to av_buffersink_get_buffer_ref will return a buffer with
+ * exactly the specified number of samples, or AVERROR(EAGAIN) if there is
+ * not enough. The last buffer at EOF will be padded with 0.
+ */
+void av_buffersink_set_frame_size(AVFilterContext *ctx, unsigned frame_size);
+
+/**
+ * Get the frame rate of the input.
+ */
+AVRational av_buffersink_get_frame_rate(AVFilterContext *ctx);
+
+/**
+ * Get a frame with filtered data from sink and put it in frame.
+ *
+ * @param ctx pointer to a context of a buffersink or abuffersink AVFilter.
+ * @param frame pointer to an allocated frame that will be filled with data.
+ *              The data must be freed using av_frame_unref() / av_frame_free()
+ *
+ * @return
+ *         - >= 0 if a frame was successfully returned.
+ *         - AVERROR(EAGAIN) if no frames are available at this point; more
+ *           input frames must be added to the filtergraph to get more output.
+ *         - AVERROR_EOF if there will be no more output frames on this sink.
+ *         - A different negative AVERROR code in other failure cases.
+ */
+int av_buffersink_get_frame(AVFilterContext *ctx, AVFrame *frame);
+
+/**
+ * Same as av_buffersink_get_frame(), but with the ability to specify the number
+ * of samples read. This function is less efficient than
+ * av_buffersink_get_frame(), because it copies the data around.
+ *
+ * @param ctx pointer to a context of the abuffersink AVFilter.
+ * @param frame pointer to an allocated frame that will be filled with data.
+ *              The data must be freed using av_frame_unref() / av_frame_free()
+ *              frame will contain exactly nb_samples audio samples, except at
+ *              the end of stream, when it can contain less than nb_samples.
+ *
+ * @return The return codes have the same meaning as for
+ *         av_buffersink_get_samples().
+ *
+ * @warning do not mix this function with av_buffersink_get_frame(). Use only one or
+ * the other with a single sink, not both.
+ */
+int av_buffersink_get_samples(AVFilterContext *ctx, AVFrame *frame, int nb_samples);
+
+/**
+ * @}
+ */
+
+#endif /* AVFILTER_BUFFERSINK_H */
@@ -0,0 +1,129 @@
+/*
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFILTER_BUFFERSRC_H
+#define AVFILTER_BUFFERSRC_H
+
+/**
+ * @file
+ * @ingroup lavfi_buffersrc
+ * Memory buffer source API.
+ */
+
+#include "libavcodec/avcodec.h"
+#include "avfilter.h"
+
+/**
+ * @defgroup lavfi_buffersrc Buffer source API
+ * @ingroup lavfi
+ * @{
+ */
+
+enum {
+
+    /**
+     * Do not check for format changes.
+     */
+    AV_BUFFERSRC_FLAG_NO_CHECK_FORMAT = 1,
+
+    /**
+     * Immediately push the frame to the output.
+     */
+    AV_BUFFERSRC_FLAG_PUSH = 4,
+
+    /**
+     * Keep a reference to the frame.
+     * If the frame if reference-counted, create a new reference; otherwise
+     * copy the frame data.
+     */
+    AV_BUFFERSRC_FLAG_KEEP_REF = 8,
+
+};
+
+/**
+ * Get the number of failed requests.
+ *
+ * A failed request is when the request_frame method is called while no
+ * frame is present in the buffer.
+ * The number is reset when a frame is added.
+ */
+unsigned av_buffersrc_get_nb_failed_requests(AVFilterContext *buffer_src);
+
+/**
+ * Add a frame to the buffer source.
+ *
+ * @param ctx   an instance of the buffersrc filter
+ * @param frame frame to be added. If the frame is reference counted, this
+ * function will make a new reference to it. Otherwise the frame data will be
+ * copied.
+ *
+ * @return 0 on success, a negative AVERROR on error
+ *
+ * This function is equivalent to av_buffersrc_add_frame_flags() with the
+ * AV_BUFFERSRC_FLAG_KEEP_REF flag.
+ */
+av_warn_unused_result
+int av_buffersrc_write_frame(AVFilterContext *ctx, const AVFrame *frame);
+
+/**
+ * Add a frame to the buffer source.
+ *
+ * @param ctx   an instance of the buffersrc filter
+ * @param frame frame to be added. If the frame is reference counted, this
+ * function will take ownership of the reference(s) and reset the frame.
+ * Otherwise the frame data will be copied. If this function returns an error,
+ * the input frame is not touched.
+ *
+ * @return 0 on success, a negative AVERROR on error.
+ *
+ * @note the difference between this function and av_buffersrc_write_frame() is
+ * that av_buffersrc_write_frame() creates a new reference to the input frame,
+ * while this function takes ownership of the reference passed to it.
+ *
+ * This function is equivalent to av_buffersrc_add_frame_flags() without the
+ * AV_BUFFERSRC_FLAG_KEEP_REF flag.
+ */
+av_warn_unused_result
+int av_buffersrc_add_frame(AVFilterContext *ctx, AVFrame *frame);
+
+/**
+ * Add a frame to the buffer source.
+ *
+ * By default, if the frame is reference-counted, this function will take
+ * ownership of the reference(s) and reset the frame. This can be controlled
+ * using the flags.
+ *
+ * If this function returns an error, the input frame is not touched.
+ *
+ * @param buffer_src  pointer to a buffer source context
+ * @param frame       a frame, or NULL to mark EOF
+ * @param flags       a combination of AV_BUFFERSRC_FLAG_*
+ * @return            >= 0 in case of success, a negative AVERROR code
+ *                    in case of failure
+ */
+av_warn_unused_result
+int av_buffersrc_add_frame_flags(AVFilterContext *buffer_src,
+                                 AVFrame *frame, int flags);
+
+
+/**
+ * @}
+ */
+
+#endif /* AVFILTER_BUFFERSRC_H */
@@ -0,0 +1,71 @@
+/*
+ * Version macros.
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFILTER_VERSION_H
+#define AVFILTER_VERSION_H
+
+/**
+ * @file
+ * @ingroup lavfi
+ * Libavfilter version macros
+ */
+
+#include "libavutil/version.h"
+
+#define LIBAVFILTER_VERSION_MAJOR   6
+#define LIBAVFILTER_VERSION_MINOR  14
+#define LIBAVFILTER_VERSION_MICRO 101
+
+#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
+                                               LIBAVFILTER_VERSION_MINOR, \
+                                               LIBAVFILTER_VERSION_MICRO)
+#define LIBAVFILTER_VERSION     AV_VERSION(LIBAVFILTER_VERSION_MAJOR,   \
+                                           LIBAVFILTER_VERSION_MINOR,   \
+                                           LIBAVFILTER_VERSION_MICRO)
+#define LIBAVFILTER_BUILD       LIBAVFILTER_VERSION_INT
+
+#define LIBAVFILTER_IDENT       "Lavfi" AV_STRINGIFY(LIBAVFILTER_VERSION)
+
+/**
+ * FF_API_* defines may be placed below to indicate public API that will be
+ * dropped at a future version bump. The defines themselves are not part of
+ * the public API and may change, break or disappear at any time.
+ */
+
+#ifndef FF_API_OLD_FILTER_OPTS
+#define FF_API_OLD_FILTER_OPTS              (LIBAVFILTER_VERSION_MAJOR < 7)
+#endif
+#ifndef FF_API_OLD_FILTER_OPTS_ERROR
+#define FF_API_OLD_FILTER_OPTS_ERROR        (LIBAVFILTER_VERSION_MAJOR < 7)
+#endif
+#ifndef FF_API_AVFILTER_OPEN
+#define FF_API_AVFILTER_OPEN                (LIBAVFILTER_VERSION_MAJOR < 7)
+#endif
+#ifndef FF_API_AVFILTER_INIT_FILTER
+#define FF_API_AVFILTER_INIT_FILTER         (LIBAVFILTER_VERSION_MAJOR < 7)
+#endif
+#ifndef FF_API_OLD_FILTER_REGISTER
+#define FF_API_OLD_FILTER_REGISTER          (LIBAVFILTER_VERSION_MAJOR < 7)
+#endif
+#ifndef FF_API_NOCONST_GET_NAME
+#define FF_API_NOCONST_GET_NAME             (LIBAVFILTER_VERSION_MAJOR < 7)
+#endif
+
+#endif /* AVFILTER_VERSION_H */
@@ -0,0 +1,680 @@
+/*
+ * copyright (c) 2001 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#ifndef AVFORMAT_AVIO_H
+#define AVFORMAT_AVIO_H
+
+/**
+ * @file
+ * @ingroup lavf_io
+ * Buffered I/O operations
+ */
+
+#include <stdint.h>
+
+#include "libavutil/common.h"
+#include "libavutil/dict.h"
+#include "libavutil/log.h"
+
+#include "libavformat/version.h"
+
+#define AVIO_SEEKABLE_NORMAL 0x0001 /**< Seeking works like for a local file */
+
+/**
+ * Callback for checking whether to abort blocking functions.
+ * AVERROR_EXIT is returned in this case by the interrupted
+ * function. During blocking operations, callback is called with
+ * opaque as parameter. If the callback returns 1, the
+ * blocking operation will be aborted.
+ *
+ * No members can be added to this struct without a major bump, if
+ * new elements have been added after this struct in AVFormatContext
+ * or AVIOContext.
+ */
+typedef struct AVIOInterruptCB {
+    int (*callback)(void*);
+    void *opaque;
+} AVIOInterruptCB;
+
+/**
+ * Directory entry types.
+ */
+enum AVIODirEntryType {
+    AVIO_ENTRY_UNKNOWN,
+    AVIO_ENTRY_BLOCK_DEVICE,
+    AVIO_ENTRY_CHARACTER_DEVICE,
+    AVIO_ENTRY_DIRECTORY,
+    AVIO_ENTRY_NAMED_PIPE,
+    AVIO_ENTRY_SYMBOLIC_LINK,
+    AVIO_ENTRY_SOCKET,
+    AVIO_ENTRY_FILE,
+    AVIO_ENTRY_SERVER,
+    AVIO_ENTRY_SHARE,
+    AVIO_ENTRY_WORKGROUP,
+};
+
+/**
+ * Describes single entry of the directory.
+ *
+ * Only name and type fields are guaranteed be set.
+ * Rest of fields are protocol or/and platform dependent and might be unknown.
+ */
+typedef struct AVIODirEntry {
+    char *name;                           /**< Filename */
+    int type;                             /**< Type of the entry */
+    int utf8;                             /**< Set to 1 when name is encoded with UTF-8, 0 otherwise.
+                                               Name can be encoded with UTF-8 even though 0 is set. */
+    int64_t size;                         /**< File size in bytes, -1 if unknown. */
+    int64_t modification_timestamp;       /**< Time of last modification in microseconds since unix
+                                               epoch, -1 if unknown. */
+    int64_t access_timestamp;             /**< Time of last access in microseconds since unix epoch,
+                                               -1 if unknown. */
+    int64_t status_change_timestamp;      /**< Time of last status change in microseconds since unix
+                                               epoch, -1 if unknown. */
+    int64_t user_id;                      /**< User ID of owner, -1 if unknown. */
+    int64_t group_id;                     /**< Group ID of owner, -1 if unknown. */
+    int64_t filemode;                     /**< Unix file mode, -1 if unknown. */
+} AVIODirEntry;
+
+typedef struct AVIODirContext {
+    struct URLContext *url_context;
+} AVIODirContext;
+
+/**
+ * Bytestream IO Context.
+ * New fields can be added to the end with minor version bumps.
+ * Removal, reordering and changes to existing fields require a major
+ * version bump.
+ * sizeof(AVIOContext) must not be used outside libav*.
+ *
+ * @note None of the function pointers in AVIOContext should be called
+ *       directly, they should only be set by the client application
+ *       when implementing custom I/O. Normally these are set to the
+ *       function pointers specified in avio_alloc_context()
+ */
+typedef struct AVIOContext {
+    /**
+     * A class for private options.
+     *
+     * If this AVIOContext is created by avio_open2(), av_class is set and
+     * passes the options down to protocols.
+     *
+     * If this AVIOContext is manually allocated, then av_class may be set by
+     * the caller.
+     *
+     * warning -- this field can be NULL, be sure to not pass this AVIOContext
+     * to any av_opt_* functions in that case.
+     */
+    const AVClass *av_class;
+    unsigned char *buffer;  /**< Start of the buffer. */
+    int buffer_size;        /**< Maximum buffer size */
+    unsigned char *buf_ptr; /**< Current position in the buffer */
+    unsigned char *buf_end; /**< End of the data, may be less than
+                                 buffer+buffer_size if the read function returned
+                                 less data than requested, e.g. for streams where
+                                 no more data has been received yet. */
+    void *opaque;           /**< A private pointer, passed to the read/write/seek/...
+                                 functions. */
+    int (*read_packet)(void *opaque, uint8_t *buf, int buf_size);
+    int (*write_packet)(void *opaque, uint8_t *buf, int buf_size);
+    int64_t (*seek)(void *opaque, int64_t offset, int whence);
+    int64_t pos;            /**< position in the file of the current buffer */
+    int must_flush;         /**< true if the next seek should flush */
+    int eof_reached;        /**< true if eof reached */
+    int write_flag;         /**< true if open for writing */
+    int max_packet_size;
+    unsigned long checksum;
+    unsigned char *checksum_ptr;
+    unsigned long (*update_checksum)(unsigned long checksum, const uint8_t *buf, unsigned int size);
+    int error;              /**< contains the error code or 0 if no error happened */
+    /**
+     * Pause or resume playback for network streaming protocols - e.g. MMS.
+     */
+    int (*read_pause)(void *opaque, int pause);
+    /**
+     * Seek to a given timestamp in stream with the specified stream_index.
+     * Needed for some network streaming protocols which don't support seeking
+     * to byte position.
+     */
+    int64_t (*read_seek)(void *opaque, int stream_index,
+                         int64_t timestamp, int flags);
+    /**
+     * A combination of AVIO_SEEKABLE_ flags or 0 when the stream is not seekable.
+     */
+    int seekable;
+
+    /**
+     * max filesize, used to limit allocations
+     * This field is internal to libavformat and access from outside is not allowed.
+     */
+    int64_t maxsize;
+
+    /**
+     * avio_read and avio_write should if possible be satisfied directly
+     * instead of going through a buffer, and avio_seek will always
+     * call the underlying seek function directly.
+     */
+    int direct;
+
+    /**
+     * Bytes read statistic
+     * This field is internal to libavformat and access from outside is not allowed.
+     */
+    int64_t bytes_read;
+
+    /**
+     * seek statistic
+     * This field is internal to libavformat and access from outside is not allowed.
+     */
+    int seek_count;
+
+    /**
+     * writeout statistic
+     * This field is internal to libavformat and access from outside is not allowed.
+     */
+    int writeout_count;
+
+    /**
+     * Original buffer size
+     * used internally after probing and ensure seekback to reset the buffer size
+     * This field is internal to libavformat and access from outside is not allowed.
+     */
+    int orig_buffer_size;
+
+    /**
+     * Threshold to favor readahead over seek.
+     * This is current internal only, do not use from outside.
+     */
+    int short_seek_threshold;
+} AVIOContext;
+
+/* unbuffered I/O */
+
+/**
+ * Return the name of the protocol that will handle the passed URL.
+ *
+ * NULL is returned if no protocol could be found for the given URL.
+ *
+ * @return Name of the protocol or NULL.
+ */
+const char *avio_find_protocol_name(const char *url);
+
+/**
+ * Return AVIO_FLAG_* access flags corresponding to the access permissions
+ * of the resource in url, or a negative value corresponding to an
+ * AVERROR code in case of failure. The returned access flags are
+ * masked by the value in flags.
+ *
+ * @note This function is intrinsically unsafe, in the sense that the
+ * checked resource may change its existence or permission status from
+ * one call to another. Thus you should not trust the returned value,
+ * unless you are sure that no other processes are accessing the
+ * checked resource.
+ */
+int avio_check(const char *url, int flags);
+
+/**
+ * Move or rename a resource.
+ *
+ * @note url_src and url_dst should share the same protocol and authority.
+ *
+ * @param url_src url to resource to be moved
+ * @param url_dst new url to resource if the operation succeeded
+ * @return >=0 on success or negative on error.
+ */
+int avpriv_io_move(const char *url_src, const char *url_dst);
+
+/**
+ * Delete a resource.
+ *
+ * @param url resource to be deleted.
+ * @return >=0 on success or negative on error.
+ */
+int avpriv_io_delete(const char *url);
+
+/**
+ * Open directory for reading.
+ *
+ * @param s       directory read context. Pointer to a NULL pointer must be passed.
+ * @param url     directory to be listed.
+ * @param options A dictionary filled with protocol-private options. On return
+ *                this parameter will be destroyed and replaced with a dictionary
+ *                containing options that were not found. May be NULL.
+ * @return >=0 on success or negative on error.
+ */
+int avio_open_dir(AVIODirContext **s, const char *url, AVDictionary **options);
+
+/**
+ * Get next directory entry.
+ *
+ * Returned entry must be freed with avio_free_directory_entry(). In particular
+ * it may outlive AVIODirContext.
+ *
+ * @param s         directory read context.
+ * @param[out] next next entry or NULL when no more entries.
+ * @return >=0 on success or negative on error. End of list is not considered an
+ *             error.
+ */
+int avio_read_dir(AVIODirContext *s, AVIODirEntry **next);
+
+/**
+ * Close directory.
+ *
+ * @note Entries created using avio_read_dir() are not deleted and must be
+ * freeded with avio_free_directory_entry().
+ *
+ * @param s         directory read context.
+ * @return >=0 on success or negative on error.
+ */
+int avio_close_dir(AVIODirContext **s);
+
+/**
+ * Free entry allocated by avio_read_dir().
+ *
+ * @param entry entry to be freed.
+ */
+void avio_free_directory_entry(AVIODirEntry **entry);
+
+/**
+ * Allocate and initialize an AVIOContext for buffered I/O. It must be later
+ * freed with av_free().
+ *
+ * @param buffer Memory block for input/output operations via AVIOContext.
+ *        The buffer must be allocated with av_malloc() and friends.
+ *        It may be freed and replaced with a new buffer by libavformat.
+ *        AVIOContext.buffer holds the buffer currently in use,
+ *        which must be later freed with av_free().
+ * @param buffer_size The buffer size is very important for performance.
+ *        For protocols with fixed blocksize it should be set to this blocksize.
+ *        For others a typical size is a cache page, e.g. 4kb.
+ * @param write_flag Set to 1 if the buffer should be writable, 0 otherwise.
+ * @param opaque An opaque pointer to user-specific data.
+ * @param read_packet  A function for refilling the buffer, may be NULL.
+ * @param write_packet A function for writing the buffer contents, may be NULL.
+ *        The function may not change the input buffers content.
+ * @param seek A function for seeking to specified byte position, may be NULL.
+ *
+ * @return Allocated AVIOContext or NULL on failure.
+ */
+AVIOContext *avio_alloc_context(
+                  unsigned char *buffer,
+                  int buffer_size,
+                  int write_flag,
+                  void *opaque,
+                  int (*read_packet)(void *opaque, uint8_t *buf, int buf_size),
+                  int (*write_packet)(void *opaque, uint8_t *buf, int buf_size),
+                  int64_t (*seek)(void *opaque, int64_t offset, int whence));
+
+void avio_w8(AVIOContext *s, int b);
+void avio_write(AVIOContext *s, const unsigned char *buf, int size);
+void avio_wl64(AVIOContext *s, uint64_t val);
+void avio_wb64(AVIOContext *s, uint64_t val);
+void avio_wl32(AVIOContext *s, unsigned int val);
+void avio_wb32(AVIOContext *s, unsigned int val);
+void avio_wl24(AVIOContext *s, unsigned int val);
+void avio_wb24(AVIOContext *s, unsigned int val);
+void avio_wl16(AVIOContext *s, unsigned int val);
+void avio_wb16(AVIOContext *s, unsigned int val);
+
+/**
+ * Write a NULL-terminated string.
+ * @return number of bytes written.
+ */
+int avio_put_str(AVIOContext *s, const char *str);
+
+/**
+ * Convert an UTF-8 string to UTF-16LE and write it.
+ * @param s the AVIOContext
+ * @param str NULL-terminated UTF-8 string
+ *
+ * @return number of bytes written.
+ */
+int avio_put_str16le(AVIOContext *s, const char *str);
+
+/**
+ * Convert an UTF-8 string to UTF-16BE and write it.
+ * @param s the AVIOContext
+ * @param str NULL-terminated UTF-8 string
+ *
+ * @return number of bytes written.
+ */
+int avio_put_str16be(AVIOContext *s, const char *str);
+
+/**
+ * Passing this as the "whence" parameter to a seek function causes it to
+ * return the filesize without seeking anywhere. Supporting this is optional.
+ * If it is not supported then the seek function will return <0.
+ */
+#define AVSEEK_SIZE 0x10000
+
+/**
+ * Oring this flag as into the "whence" parameter to a seek function causes it to
+ * seek by any means (like reopening and linear reading) or other normally unreasonable
+ * means that can be extremely slow.
+ * This may be ignored by the seek code.
+ */
+#define AVSEEK_FORCE 0x20000
+
+/**
+ * fseek() equivalent for AVIOContext.
+ * @return new position or AVERROR.
+ */
+int64_t avio_seek(AVIOContext *s, int64_t offset, int whence);
+
+/**
+ * Skip given number of bytes forward
+ * @return new position or AVERROR.
+ */
+int64_t avio_skip(AVIOContext *s, int64_t offset);
+
+/**
+ * ftell() equivalent for AVIOContext.
+ * @return position or AVERROR.
+ */
+static av_always_inline int64_t avio_tell(AVIOContext *s)
+{
+    return avio_seek(s, 0, SEEK_CUR);
+}
+
+/**
+ * Get the filesize.
+ * @return filesize or AVERROR
+ */
+int64_t avio_size(AVIOContext *s);
+
+/**
+ * feof() equivalent for AVIOContext.
+ * @return non zero if and only if end of file
+ */
+int avio_feof(AVIOContext *s);
+#if FF_API_URL_FEOF
+/**
+ * @deprecated use avio_feof()
+ */
+attribute_deprecated
+int url_feof(AVIOContext *s);
+#endif
+
+/** @warning currently size is limited */
+int avio_printf(AVIOContext *s, const char *fmt, ...) av_printf_format(2, 3);
+
+/**
+ * Force flushing of buffered data.
+ *
+ * For write streams, force the buffered data to be immediately written to the output,
+ * without to wait to fill the internal buffer.
+ *
+ * For read streams, discard all currently buffered data, and advance the
+ * reported file position to that of the underlying stream. This does not
+ * read new data, and does not perform any seeks.
+ */
+void avio_flush(AVIOContext *s);
+
+/**
+ * Read size bytes from AVIOContext into buf.
+ * @return number of bytes read or AVERROR
+ */
+int avio_read(AVIOContext *s, unsigned char *buf, int size);
+
+/**
+ * @name Functions for reading from AVIOContext
+ * @{
+ *
+ * @note return 0 if EOF, so you cannot use it if EOF handling is
+ *       necessary
+ */
+int          avio_r8  (AVIOContext *s);
+unsigned int avio_rl16(AVIOContext *s);
+unsigned int avio_rl24(AVIOContext *s);
+unsigned int avio_rl32(AVIOContext *s);
+uint64_t     avio_rl64(AVIOContext *s);
+unsigned int avio_rb16(AVIOContext *s);
+unsigned int avio_rb24(AVIOContext *s);
+unsigned int avio_rb32(AVIOContext *s);
+uint64_t     avio_rb64(AVIOContext *s);
+/**
+ * @}
+ */
+
+/**
+ * Read a string from pb into buf. The reading will terminate when either
+ * a NULL character was encountered, maxlen bytes have been read, or nothing
+ * more can be read from pb. The result is guaranteed to be NULL-terminated, it
+ * will be truncated if buf is too small.
+ * Note that the string is not interpreted or validated in any way, it
+ * might get truncated in the middle of a sequence for multi-byte encodings.
+ *
+ * @return number of bytes read (is always <= maxlen).
+ * If reading ends on EOF or error, the return value will be one more than
+ * bytes actually read.
+ */
+int avio_get_str(AVIOContext *pb, int maxlen, char *buf, int buflen);
+
+/**
+ * Read a UTF-16 string from pb and convert it to UTF-8.
+ * The reading will terminate when either a null or invalid character was
+ * encountered or maxlen bytes have been read.
+ * @return number of bytes read (is always <= maxlen)
+ */
+int avio_get_str16le(AVIOContext *pb, int maxlen, char *buf, int buflen);
+int avio_get_str16be(AVIOContext *pb, int maxlen, char *buf, int buflen);
+
+
+/**
+ * @name URL open modes
+ * The flags argument to avio_open must be one of the following
+ * constants, optionally ORed with other flags.
+ * @{
+ */
+#define AVIO_FLAG_READ  1                                      /**< read-only */
+#define AVIO_FLAG_WRITE 2                                      /**< write-only */
+#define AVIO_FLAG_READ_WRITE (AVIO_FLAG_READ|AVIO_FLAG_WRITE)  /**< read-write pseudo flag */
+/**
+ * @}
+ */
+
+/**
+ * Use non-blocking mode.
+ * If this flag is set, operations on the context will return
+ * AVERROR(EAGAIN) if they can not be performed immediately.
+ * If this flag is not set, operations on the context will never return
+ * AVERROR(EAGAIN).
+ * Note that this flag does not affect the opening/connecting of the
+ * context. Connecting a protocol will always block if necessary (e.g. on
+ * network protocols) but never hang (e.g. on busy devices).
+ * Warning: non-blocking protocols is work-in-progress; this flag may be
+ * silently ignored.
+ */
+#define AVIO_FLAG_NONBLOCK 8
+
+/**
+ * Use direct mode.
+ * avio_read and avio_write should if possible be satisfied directly
+ * instead of going through a buffer, and avio_seek will always
+ * call the underlying seek function directly.
+ */
+#define AVIO_FLAG_DIRECT 0x8000
+
+/**
+ * Create and initialize a AVIOContext for accessing the
+ * resource indicated by url.
+ * @note When the resource indicated by url has been opened in
+ * read+write mode, the AVIOContext can be used only for writing.
+ *
+ * @param s Used to return the pointer to the created AVIOContext.
+ * In case of failure the pointed to value is set to NULL.
+ * @param url resource to access
+ * @param flags flags which control how the resource indicated by url
+ * is to be opened
+ * @return >= 0 in case of success, a negative value corresponding to an
+ * AVERROR code in case of failure
+ */
+int avio_open(AVIOContext **s, const char *url, int flags);
+
+/**
+ * Create and initialize a AVIOContext for accessing the
+ * resource indicated by url.
+ * @note When the resource indicated by url has been opened in
+ * read+write mode, the AVIOContext can be used only for writing.
+ *
+ * @param s Used to return the pointer to the created AVIOContext.
+ * In case of failure the pointed to value is set to NULL.
+ * @param url resource to access
+ * @param flags flags which control how the resource indicated by url
+ * is to be opened
+ * @param int_cb an interrupt callback to be used at the protocols level
+ * @param options  A dictionary filled with protocol-private options. On return
+ * this parameter will be destroyed and replaced with a dict containing options
+ * that were not found. May be NULL.
+ * @return >= 0 in case of success, a negative value corresponding to an
+ * AVERROR code in case of failure
+ */
+int avio_open2(AVIOContext **s, const char *url, int flags,
+               const AVIOInterruptCB *int_cb, AVDictionary **options);
+
+/**
+ * Close the resource accessed by the AVIOContext s and free it.
+ * This function can only be used if s was opened by avio_open().
+ *
+ * The internal buffer is automatically flushed before closing the
+ * resource.
+ *
+ * @return 0 on success, an AVERROR < 0 on error.
+ * @see avio_closep
+ */
+int avio_close(AVIOContext *s);
+
+/**
+ * Close the resource accessed by the AVIOContext *s, free it
+ * and set the pointer pointing to it to NULL.
+ * This function can only be used if s was opened by avio_open().
+ *
+ * The internal buffer is automatically flushed before closing the
+ * resource.
+ *
+ * @return 0 on success, an AVERROR < 0 on error.
+ * @see avio_close
+ */
+int avio_closep(AVIOContext **s);
+
+
+/**
+ * Open a write only memory stream.
+ *
+ * @param s new IO context
+ * @return zero if no error.
+ */
+int avio_open_dyn_buf(AVIOContext **s);
+
+/**
+ * Return the written size and a pointer to the buffer. The buffer
+ * must be freed with av_free().
+ * Padding of AV_INPUT_BUFFER_PADDING_SIZE is added to the buffer.
+ *
+ * @param s IO context
+ * @param pbuffer pointer to a byte buffer
+ * @return the length of the byte buffer
+ */
+int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer);
+
+/**
+ * Iterate through names of available protocols.
+ *
+ * @param opaque A private pointer representing current protocol.
+ *        It must be a pointer to NULL on first iteration and will
+ *        be updated by successive calls to avio_enum_protocols.
+ * @param output If set to 1, iterate over output protocols,
+ *               otherwise over input protocols.
+ *
+ * @return A static string containing the name of current protocol or NULL
+ */
+const char *avio_enum_protocols(void **opaque, int output);
+
+/**
+ * Pause and resume playing - only meaningful if using a network streaming
+ * protocol (e.g. MMS).
+ *
+ * @param h     IO context from which to call the read_pause function pointer
+ * @param pause 1 for pause, 0 for resume
+ */
+int     avio_pause(AVIOContext *h, int pause);
+
+/**
+ * Seek to a given timestamp relative to some component stream.
+ * Only meaningful if using a network streaming protocol (e.g. MMS.).
+ *
+ * @param h IO context from which to call the seek function pointers
+ * @param stream_index The stream index that the timestamp is relative to.
+ *        If stream_index is (-1) the timestamp should be in AV_TIME_BASE
+ *        units from the beginning of the presentation.
+ *        If a stream_index >= 0 is used and the protocol does not support
+ *        seeking based on component streams, the call will fail.
+ * @param timestamp timestamp in AVStream.time_base units
+ *        or if there is no stream specified then in AV_TIME_BASE units.
+ * @param flags Optional combination of AVSEEK_FLAG_BACKWARD, AVSEEK_FLAG_BYTE
+ *        and AVSEEK_FLAG_ANY. The protocol may silently ignore
+ *        AVSEEK_FLAG_BACKWARD and AVSEEK_FLAG_ANY, but AVSEEK_FLAG_BYTE will
+ *        fail if used and not supported.
+ * @return >= 0 on success
+ * @see AVInputFormat::read_seek
+ */
+int64_t avio_seek_time(AVIOContext *h, int stream_index,
+                       int64_t timestamp, int flags);
+
+/* Avoid a warning. The header can not be included because it breaks c++. */
+struct AVBPrint;
+
+/**
+ * Read contents of h into print buffer, up to max_size bytes, or up to EOF.
+ *
+ * @return 0 for success (max_size bytes read or EOF reached), negative error
+ * code otherwise
+ */
+int avio_read_to_bprint(AVIOContext *h, struct AVBPrint *pb, size_t max_size);
+
+/**
+ * Accept and allocate a client context on a server context.
+ * @param  s the server context
+ * @param  c the client context, must be unallocated
+ * @return   >= 0 on success or a negative value corresponding
+ *           to an AVERROR on failure
+ */
+int avio_accept(AVIOContext *s, AVIOContext **c);
+
+/**
+ * Perform one step of the protocol handshake to accept a new client.
+ * This function must be called on a client returned by avio_accept() before
+ * using it as a read/write context.
+ * It is separate from avio_accept() because it may block.
+ * A step of the handshake is defined by places where the application may
+ * decide to change the proceedings.
+ * For example, on a protocol with a request header and a reply header, each
+ * one can constitute a step because the application may use the parameters
+ * from the request to change parameters in the reply; or each individual
+ * chunk of the request can constitute a step.
+ * If the handshake is already finished, avio_handshake() does nothing and
+ * returns 0 immediately.
+ *
+ * @param  c the client context to perform the handshake on
+ * @return   0   on a complete and successful handshake
+ *           > 0 if the handshake progressed, but is not complete
+ *           < 0 for an AVERROR code
+ */
+int avio_handshake(AVIOContext *c);
+#endif /* AVFORMAT_AVIO_H */
@@ -0,0 +1,75 @@
+/*
+ * Version macros.
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFORMAT_VERSION_H
+#define AVFORMAT_VERSION_H
+
+/**
+ * @file
+ * @ingroup libavf
+ * Libavformat version macros
+ */
+
+#include "libavutil/version.h"
+
+#define LIBAVFORMAT_VERSION_MAJOR  57
+#define LIBAVFORMAT_VERSION_MINOR  14
+#define LIBAVFORMAT_VERSION_MICRO 100
+
+#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
+                                               LIBAVFORMAT_VERSION_MINOR, \
+                                               LIBAVFORMAT_VERSION_MICRO)
+#define LIBAVFORMAT_VERSION     AV_VERSION(LIBAVFORMAT_VERSION_MAJOR,   \
+                                           LIBAVFORMAT_VERSION_MINOR,   \
+                                           LIBAVFORMAT_VERSION_MICRO)
+#define LIBAVFORMAT_BUILD       LIBAVFORMAT_VERSION_INT
+
+#define LIBAVFORMAT_IDENT       "Lavf" AV_STRINGIFY(LIBAVFORMAT_VERSION)
+
+/**
+ * FF_API_* defines may be placed below to indicate public API that will be
+ * dropped at a future version bump. The defines themselves are not part of
+ * the public API and may change, break or disappear at any time.
+ *
+ * @note, when bumping the major version it is recommended to manually
+ * disable each FF_API_* in its own commit instead of disabling them all
+ * at once through the bump. This improves the git bisect-ability of the change.
+ *
+ */
+#ifndef FF_API_LAVF_BITEXACT
+#define FF_API_LAVF_BITEXACT            (LIBAVFORMAT_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_LAVF_FRAC
+#define FF_API_LAVF_FRAC                (LIBAVFORMAT_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_LAVF_CODEC_TB
+#define FF_API_LAVF_CODEC_TB            (LIBAVFORMAT_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_URL_FEOF
+#define FF_API_URL_FEOF                 (LIBAVFORMAT_VERSION_MAJOR < 58)
+#endif
+#ifndef FF_API_LAVF_FMT_RAWPICTURE
+#define FF_API_LAVF_FMT_RAWPICTURE      (LIBAVFORMAT_VERSION_MAJOR < 58)
+#endif
+
+#ifndef FF_API_R_FRAME_RATE
+#define FF_API_R_FRAME_RATE            1
+#endif
+#endif /* AVFORMAT_VERSION_H */
@@ -0,0 +1,55 @@
+/*
+ * copyright (c) 2006 Mans Rullgard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_ADLER32_H
+#define AVUTIL_ADLER32_H
+
+#include <stdint.h>
+#include "attributes.h"
+
+/**
+ * @file
+ * Public header for libavutil Adler32 hasher
+ *
+ * @defgroup lavu_adler32 Adler32
+ * @ingroup lavu_crypto
+ * @{
+ */
+
+/**
+ * Calculate the Adler32 checksum of a buffer.
+ *
+ * Passing the return value to a subsequent av_adler32_update() call
+ * allows the checksum of multiple buffers to be calculated as though
+ * they were concatenated.
+ *
+ * @param adler initial checksum value
+ * @param buf   pointer to input buffer
+ * @param len   size of input buffer
+ * @return      updated checksum
+ */
+unsigned long av_adler32_update(unsigned long adler, const uint8_t *buf,
+                                unsigned int len) av_pure;
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_ADLER32_H */
@@ -0,0 +1,65 @@
+/*
+ * copyright (c) 2007 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_AES_H
+#define AVUTIL_AES_H
+
+#include <stdint.h>
+
+#include "attributes.h"
+#include "version.h"
+
+/**
+ * @defgroup lavu_aes AES
+ * @ingroup lavu_crypto
+ * @{
+ */
+
+extern const int av_aes_size;
+
+struct AVAES;
+
+/**
+ * Allocate an AVAES context.
+ */
+struct AVAES *av_aes_alloc(void);
+
+/**
+ * Initialize an AVAES context.
+ * @param key_bits 128, 192 or 256
+ * @param decrypt 0 for encryption, 1 for decryption
+ */
+int av_aes_init(struct AVAES *a, const uint8_t *key, int key_bits, int decrypt);
+
+/**
+ * Encrypt or decrypt a buffer using a previously initialized context.
+ * @param count number of 16 byte blocks
+ * @param dst destination array, can be equal to src
+ * @param src source array, can be equal to dst
+ * @param iv initialization vector for CBC mode, if NULL then ECB will be used
+ * @param decrypt 0 for encryption, 1 for decryption
+ */
+void av_aes_crypt(struct AVAES *a, uint8_t *dst, const uint8_t *src, int count, uint8_t *iv, int decrypt);
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_AES_H */
@@ -0,0 +1,168 @@
+/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Macro definitions for various function/variable attributes
+ */
+
+#ifndef AVUTIL_ATTRIBUTES_H
+#define AVUTIL_ATTRIBUTES_H
+
+#ifdef __GNUC__
+#    define AV_GCC_VERSION_AT_LEAST(x,y) (__GNUC__ > (x) || __GNUC__ == (x) && __GNUC_MINOR__ >= (y))
+#    define AV_GCC_VERSION_AT_MOST(x,y)  (__GNUC__ < (x) || __GNUC__ == (x) && __GNUC_MINOR__ <= (y))
+#else
+#    define AV_GCC_VERSION_AT_LEAST(x,y) 0
+#    define AV_GCC_VERSION_AT_MOST(x,y)  0
+#endif
+
+#ifndef av_always_inline
+#if AV_GCC_VERSION_AT_LEAST(3,1)
+#    define av_always_inline __attribute__((always_inline)) inline
+#elif defined(_MSC_VER)
+#    define av_always_inline __forceinline
+#else
+#    define av_always_inline inline
+#endif
+#endif
+
+#ifndef av_extern_inline
+#if defined(__ICL) && __ICL >= 1210 || defined(__GNUC_STDC_INLINE__)
+#    define av_extern_inline extern inline
+#else
+#    define av_extern_inline inline
+#endif
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(3,4)
+#    define av_warn_unused_result __attribute__((warn_unused_result))
+#else
+#    define av_warn_unused_result
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(3,1)
+#    define av_noinline __attribute__((noinline))
+#elif defined(_MSC_VER)
+#    define av_noinline __declspec(noinline)
+#else
+#    define av_noinline
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(3,1)
+#    define av_pure __attribute__((pure))
+#else
+#    define av_pure
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(2,6)
+#    define av_const __attribute__((const))
+#else
+#    define av_const
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(4,3)
+#    define av_cold __attribute__((cold))
+#else
+#    define av_cold
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(4,1) && !defined(__llvm__)
+#    define av_flatten __attribute__((flatten))
+#else
+#    define av_flatten
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(3,1)
+#    define attribute_deprecated __attribute__((deprecated))
+#elif defined(_MSC_VER)
+#    define attribute_deprecated __declspec(deprecated)
+#else
+#    define attribute_deprecated
+#endif
+
+/**
+ * Disable warnings about deprecated features
+ * This is useful for sections of code kept for backward compatibility and
+ * scheduled for removal.
+ */
+#ifndef AV_NOWARN_DEPRECATED
+#if AV_GCC_VERSION_AT_LEAST(4,6)
+#    define AV_NOWARN_DEPRECATED(code) \
+        _Pragma("GCC diagnostic push") \
+        _Pragma("GCC diagnostic ignored \"-Wdeprecated-declarations\"") \
+        code \
+        _Pragma("GCC diagnostic pop")
+#elif defined(_MSC_VER)
+#    define AV_NOWARN_DEPRECATED(code) \
+        __pragma(warning(push)) \
+        __pragma(warning(disable : 4996)) \
+        code; \
+        __pragma(warning(pop))
+#else
+#    define AV_NOWARN_DEPRECATED(code) code
+#endif
+#endif
+
+
+#if defined(__GNUC__)
+#    define av_unused __attribute__((unused))
+#else
+#    define av_unused
+#endif
+
+/**
+ * Mark a variable as used and prevent the compiler from optimizing it
+ * away.  This is useful for variables accessed only from inline
+ * assembler without the compiler being aware.
+ */
+#if AV_GCC_VERSION_AT_LEAST(3,1)
+#    define av_used __attribute__((used))
+#else
+#    define av_used
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(3,3)
+#   define av_alias __attribute__((may_alias))
+#else
+#   define av_alias
+#endif
+
+#if defined(__GNUC__) && !defined(__INTEL_COMPILER) && !defined(__clang__)
+#    define av_uninit(x) x=x
+#else
+#    define av_uninit(x) x
+#endif
+
+#ifdef __GNUC__
+#    define av_builtin_constant_p __builtin_constant_p
+#    define av_printf_format(fmtpos, attrpos) __attribute__((__format__(__printf__, fmtpos, attrpos)))
+#else
+#    define av_builtin_constant_p(x) 0
+#    define av_printf_format(fmtpos, attrpos)
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(2,5)
+#    define av_noreturn __attribute__((noreturn))
+#else
+#    define av_noreturn
+#endif
+
+#endif /* AVUTIL_ATTRIBUTES_H */
@@ -0,0 +1,170 @@
+/*
+ * Audio FIFO
+ * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Audio FIFO Buffer
+ */
+
+#ifndef AVUTIL_AUDIO_FIFO_H
+#define AVUTIL_AUDIO_FIFO_H
+
+#include "avutil.h"
+#include "fifo.h"
+#include "samplefmt.h"
+
+/**
+ * @addtogroup lavu_audio
+ * @{
+ *
+ * @defgroup lavu_audiofifo Audio FIFO Buffer
+ * @{
+ */
+
+/**
+ * Context for an Audio FIFO Buffer.
+ *
+ * - Operates at the sample level rather than the byte level.
+ * - Supports multiple channels with either planar or packed sample format.
+ * - Automatic reallocation when writing to a full buffer.
+ */
+typedef struct AVAudioFifo AVAudioFifo;
+
+/**
+ * Free an AVAudioFifo.
+ *
+ * @param af  AVAudioFifo to free
+ */
+void av_audio_fifo_free(AVAudioFifo *af);
+
+/**
+ * Allocate an AVAudioFifo.
+ *
+ * @param sample_fmt  sample format
+ * @param channels    number of channels
+ * @param nb_samples  initial allocation size, in samples
+ * @return            newly allocated AVAudioFifo, or NULL on error
+ */
+AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
+                                 int nb_samples);
+
+/**
+ * Reallocate an AVAudioFifo.
+ *
+ * @param af          AVAudioFifo to reallocate
+ * @param nb_samples  new allocation size, in samples
+ * @return            0 if OK, or negative AVERROR code on failure
+ */
+av_warn_unused_result
+int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples);
+
+/**
+ * Write data to an AVAudioFifo.
+ *
+ * The AVAudioFifo will be reallocated automatically if the available space
+ * is less than nb_samples.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param af          AVAudioFifo to write to
+ * @param data        audio data plane pointers
+ * @param nb_samples  number of samples to write
+ * @return            number of samples actually written, or negative AVERROR
+ *                    code on failure. If successful, the number of samples
+ *                    actually written will always be nb_samples.
+ */
+int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples);
+
+/**
+ * Peek data from an AVAudioFifo.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param af          AVAudioFifo to read from
+ * @param data        audio data plane pointers
+ * @param nb_samples  number of samples to peek
+ * @return            number of samples actually peek, or negative AVERROR code
+ *                    on failure. The number of samples actually peek will not
+ *                    be greater than nb_samples, and will only be less than
+ *                    nb_samples if av_audio_fifo_size is less than nb_samples.
+ */
+int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples);
+
+/**
+ * Read data from an AVAudioFifo.
+ *
+ * @see enum AVSampleFormat
+ * The documentation for AVSampleFormat describes the data layout.
+ *
+ * @param af          AVAudioFifo to read from
+ * @param data        audio data plane pointers
+ * @param nb_samples  number of samples to read
+ * @return            number of samples actually read, or negative AVERROR code
+ *                    on failure. The number of samples actually read will not
+ *                    be greater than nb_samples, and will only be less than
+ *                    nb_samples if av_audio_fifo_size is less than nb_samples.
+ */
+int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples);
+
+/**
+ * Drain data from an AVAudioFifo.
+ *
+ * Removes the data without reading it.
+ *
+ * @param af          AVAudioFifo to drain
+ * @param nb_samples  number of samples to drain
+ * @return            0 if OK, or negative AVERROR code on failure
+ */
+int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples);
+
+/**
+ * Reset the AVAudioFifo buffer.
+ *
+ * This empties all data in the buffer.
+ *
+ * @param af  AVAudioFifo to reset
+ */
+void av_audio_fifo_reset(AVAudioFifo *af);
+
+/**
+ * Get the current number of samples in the AVAudioFifo available for reading.
+ *
+ * @param af  the AVAudioFifo to query
+ * @return    number of samples available for reading
+ */
+int av_audio_fifo_size(AVAudioFifo *af);
+
+/**
+ * Get the current number of samples in the AVAudioFifo available for writing.
+ *
+ * @param af  the AVAudioFifo to query
+ * @return    number of samples available for writing
+ */
+int av_audio_fifo_space(AVAudioFifo *af);
+
+/**
+ * @}
+ * @}
+ */
+
+#endif /* AVUTIL_AUDIO_FIFO_H */
@@ -0,0 +1,66 @@
+/*
+ * copyright (c) 2010 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * simple assert() macros that are a bit more flexible than ISO C assert().
+ * @author Michael Niedermayer <michaelni@gmx.at>
+ */
+
+#ifndef AVUTIL_AVASSERT_H
+#define AVUTIL_AVASSERT_H
+
+#include <stdlib.h>
+#include "avutil.h"
+#include "log.h"
+
+/**
+ * assert() equivalent, that is always enabled.
+ */
+#define av_assert0(cond) do {                                           \
+    if (!(cond)) {                                                      \
+        av_log(NULL, AV_LOG_PANIC, "Assertion %s failed at %s:%d\n",    \
+               AV_STRINGIFY(cond), __FILE__, __LINE__);                 \
+        abort();                                                        \
+    }                                                                   \
+} while (0)
+
+
+/**
+ * assert() equivalent, that does not lie in speed critical code.
+ * These asserts() thus can be enabled without fearing speedloss.
+ */
+#if defined(ASSERT_LEVEL) && ASSERT_LEVEL > 0
+#define av_assert1(cond) av_assert0(cond)
+#else
+#define av_assert1(cond) ((void)0)
+#endif
+
+
+/**
+ * assert() equivalent, that does lie in speed critical code.
+ */
+#if defined(ASSERT_LEVEL) && ASSERT_LEVEL > 1
+#define av_assert2(cond) av_assert0(cond)
+#else
+#define av_assert2(cond) ((void)0)
+#endif
+
+#endif /* AVUTIL_AVASSERT_H */
@@ -0,0 +1,7 @@
+/* Generated by ffconf */
+#ifndef AVUTIL_AVCONFIG_H
+#define AVUTIL_AVCONFIG_H
+#define AV_HAVE_BIGENDIAN 0
+#define AV_HAVE_FAST_UNALIGNED 1
+#define AV_HAVE_INCOMPATIBLE_LIBAV_ABI 0
+#endif /* AVUTIL_AVCONFIG_H */
@@ -0,0 +1,397 @@
+/*
+ * Copyright (c) 2007 Mans Rullgard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_AVSTRING_H
+#define AVUTIL_AVSTRING_H
+
+#include <stddef.h>
+#include <stdint.h>
+#include "attributes.h"
+
+/**
+ * @addtogroup lavu_string
+ * @{
+ */
+
+/**
+ * Return non-zero if pfx is a prefix of str. If it is, *ptr is set to
+ * the address of the first character in str after the prefix.
+ *
+ * @param str input string
+ * @param pfx prefix to test
+ * @param ptr updated if the prefix is matched inside str
+ * @return non-zero if the prefix matches, zero otherwise
+ */
+int av_strstart(const char *str, const char *pfx, const char **ptr);
+
+/**
+ * Return non-zero if pfx is a prefix of str independent of case. If
+ * it is, *ptr is set to the address of the first character in str
+ * after the prefix.
+ *
+ * @param str input string
+ * @param pfx prefix to test
+ * @param ptr updated if the prefix is matched inside str
+ * @return non-zero if the prefix matches, zero otherwise
+ */
+int av_stristart(const char *str, const char *pfx, const char **ptr);
+
+/**
+ * Locate the first case-independent occurrence in the string haystack
+ * of the string needle.  A zero-length string needle is considered to
+ * match at the start of haystack.
+ *
+ * This function is a case-insensitive version of the standard strstr().
+ *
+ * @param haystack string to search in
+ * @param needle   string to search for
+ * @return         pointer to the located match within haystack
+ *                 or a null pointer if no match
+ */
+char *av_stristr(const char *haystack, const char *needle);
+
+/**
+ * Locate the first occurrence of the string needle in the string haystack
+ * where not more than hay_length characters are searched. A zero-length
+ * string needle is considered to match at the start of haystack.
+ *
+ * This function is a length-limited version of the standard strstr().
+ *
+ * @param haystack   string to search in
+ * @param needle     string to search for
+ * @param hay_length length of string to search in
+ * @return           pointer to the located match within haystack
+ *                   or a null pointer if no match
+ */
+char *av_strnstr(const char *haystack, const char *needle, size_t hay_length);
+
+/**
+ * Copy the string src to dst, but no more than size - 1 bytes, and
+ * null-terminate dst.
+ *
+ * This function is the same as BSD strlcpy().
+ *
+ * @param dst destination buffer
+ * @param src source string
+ * @param size size of destination buffer
+ * @return the length of src
+ *
+ * @warning since the return value is the length of src, src absolutely
+ * _must_ be a properly 0-terminated string, otherwise this will read beyond
+ * the end of the buffer and possibly crash.
+ */
+size_t av_strlcpy(char *dst, const char *src, size_t size);
+
+/**
+ * Append the string src to the string dst, but to a total length of
+ * no more than size - 1 bytes, and null-terminate dst.
+ *
+ * This function is similar to BSD strlcat(), but differs when
+ * size <= strlen(dst).
+ *
+ * @param dst destination buffer
+ * @param src source string
+ * @param size size of destination buffer
+ * @return the total length of src and dst
+ *
+ * @warning since the return value use the length of src and dst, these
+ * absolutely _must_ be a properly 0-terminated strings, otherwise this
+ * will read beyond the end of the buffer and possibly crash.
+ */
+size_t av_strlcat(char *dst, const char *src, size_t size);
+
+/**
+ * Append output to a string, according to a format. Never write out of
+ * the destination buffer, and always put a terminating 0 within
+ * the buffer.
+ * @param dst destination buffer (string to which the output is
+ *  appended)
+ * @param size total size of the destination buffer
+ * @param fmt printf-compatible format string, specifying how the
+ *  following parameters are used
+ * @return the length of the string that would have been generated
+ *  if enough space had been available
+ */
+size_t av_strlcatf(char *dst, size_t size, const char *fmt, ...) av_printf_format(3, 4);
+
+/**
+ * Get the count of continuous non zero chars starting from the beginning.
+ *
+ * @param len maximum number of characters to check in the string, that
+ *            is the maximum value which is returned by the function
+ */
+static inline size_t av_strnlen(const char *s, size_t len)
+{
+    size_t i;
+    for (i = 0; i < len && s[i]; i++)
+        ;
+    return i;
+}
+
+/**
+ * Print arguments following specified format into a large enough auto
+ * allocated buffer. It is similar to GNU asprintf().
+ * @param fmt printf-compatible format string, specifying how the
+ *            following parameters are used.
+ * @return the allocated string
+ * @note You have to free the string yourself with av_free().
+ */
+char *av_asprintf(const char *fmt, ...) av_printf_format(1, 2);
+
+/**
+ * Convert a number to a av_malloced string.
+ */
+char *av_d2str(double d);
+
+/**
+ * Unescape the given string until a non escaped terminating char,
+ * and return the token corresponding to the unescaped string.
+ *
+ * The normal \ and ' escaping is supported. Leading and trailing
+ * whitespaces are removed, unless they are escaped with '\' or are
+ * enclosed between ''.
+ *
+ * @param buf the buffer to parse, buf will be updated to point to the
+ * terminating char
+ * @param term a 0-terminated list of terminating chars
+ * @return the malloced unescaped string, which must be av_freed by
+ * the user, NULL in case of allocation failure
+ */
+char *av_get_token(const char **buf, const char *term);
+
+/**
+ * Split the string into several tokens which can be accessed by
+ * successive calls to av_strtok().
+ *
+ * A token is defined as a sequence of characters not belonging to the
+ * set specified in delim.
+ *
+ * On the first call to av_strtok(), s should point to the string to
+ * parse, and the value of saveptr is ignored. In subsequent calls, s
+ * should be NULL, and saveptr should be unchanged since the previous
+ * call.
+ *
+ * This function is similar to strtok_r() defined in POSIX.1.
+ *
+ * @param s the string to parse, may be NULL
+ * @param delim 0-terminated list of token delimiters, must be non-NULL
+ * @param saveptr user-provided pointer which points to stored
+ * information necessary for av_strtok() to continue scanning the same
+ * string. saveptr is updated to point to the next character after the
+ * first delimiter found, or to NULL if the string was terminated
+ * @return the found token, or NULL when no token is found
+ */
+char *av_strtok(char *s, const char *delim, char **saveptr);
+
+/**
+ * Locale-independent conversion of ASCII isdigit.
+ */
+static inline av_const int av_isdigit(int c)
+{
+    return c >= '0' && c <= '9';
+}
+
+/**
+ * Locale-independent conversion of ASCII isgraph.
+ */
+static inline av_const int av_isgraph(int c)
+{
+    return c > 32 && c < 127;
+}
+
+/**
+ * Locale-independent conversion of ASCII isspace.
+ */
+static inline av_const int av_isspace(int c)
+{
+    return c == ' ' || c == '\f' || c == '\n' || c == '\r' || c == '\t' ||
+           c == '\v';
+}
+
+/**
+ * Locale-independent conversion of ASCII characters to uppercase.
+ */
+static inline av_const int av_toupper(int c)
+{
+    if (c >= 'a' && c <= 'z')
+        c ^= 0x20;
+    return c;
+}
+
+/**
+ * Locale-independent conversion of ASCII characters to lowercase.
+ */
+static inline av_const int av_tolower(int c)
+{
+    if (c >= 'A' && c <= 'Z')
+        c ^= 0x20;
+    return c;
+}
+
+/**
+ * Locale-independent conversion of ASCII isxdigit.
+ */
+static inline av_const int av_isxdigit(int c)
+{
+    c = av_tolower(c);
+    return av_isdigit(c) || (c >= 'a' && c <= 'f');
+}
+
+/**
+ * Locale-independent case-insensitive compare.
+ * @note This means only ASCII-range characters are case-insensitive
+ */
+int av_strcasecmp(const char *a, const char *b);
+
+/**
+ * Locale-independent case-insensitive compare.
+ * @note This means only ASCII-range characters are case-insensitive
+ */
+int av_strncasecmp(const char *a, const char *b, size_t n);
+
+
+/**
+ * Thread safe basename.
+ * @param path the path, on DOS both \ and / are considered separators.
+ * @return pointer to the basename substring.
+ */
+const char *av_basename(const char *path);
+
+/**
+ * Thread safe dirname.
+ * @param path the path, on DOS both \ and / are considered separators.
+ * @return the path with the separator replaced by the string terminator or ".".
+ * @note the function may change the input string.
+ */
+const char *av_dirname(char *path);
+
+/**
+ * Match instances of a name in a comma-separated list of names.
+ * @param name  Name to look for.
+ * @param names List of names.
+ * @return 1 on match, 0 otherwise.
+ */
+int av_match_name(const char *name, const char *names);
+
+/**
+ * Append path component to the existing path.
+ * Path separator '/' is placed between when needed.
+ * Resulting string have to be freed with av_free().
+ * @param path      base path
+ * @param component component to be appended
+ * @return new path or NULL on error.
+ */
+char *av_append_path_component(const char *path, const char *component);
+
+enum AVEscapeMode {
+    AV_ESCAPE_MODE_AUTO,      ///< Use auto-selected escaping mode.
+    AV_ESCAPE_MODE_BACKSLASH, ///< Use backslash escaping.
+    AV_ESCAPE_MODE_QUOTE,     ///< Use single-quote escaping.
+};
+
+/**
+ * Consider spaces special and escape them even in the middle of the
+ * string.
+ *
+ * This is equivalent to adding the whitespace characters to the special
+ * characters lists, except it is guaranteed to use the exact same list
+ * of whitespace characters as the rest of libavutil.
+ */
+#define AV_ESCAPE_FLAG_WHITESPACE (1 << 0)
+
+/**
+ * Escape only specified special characters.
+ * Without this flag, escape also any characters that may be considered
+ * special by av_get_token(), such as the single quote.
+ */
+#define AV_ESCAPE_FLAG_STRICT (1 << 1)
+
+/**
+ * Escape string in src, and put the escaped string in an allocated
+ * string in *dst, which must be freed with av_free().
+ *
+ * @param dst           pointer where an allocated string is put
+ * @param src           string to escape, must be non-NULL
+ * @param special_chars string containing the special characters which
+ *                      need to be escaped, can be NULL
+ * @param mode          escape mode to employ, see AV_ESCAPE_MODE_* macros.
+ *                      Any unknown value for mode will be considered equivalent to
+ *                      AV_ESCAPE_MODE_BACKSLASH, but this behaviour can change without
+ *                      notice.
+ * @param flags         flags which control how to escape, see AV_ESCAPE_FLAG_ macros
+ * @return the length of the allocated string, or a negative error code in case of error
+ * @see av_bprint_escape()
+ */
+av_warn_unused_result
+int av_escape(char **dst, const char *src, const char *special_chars,
+              enum AVEscapeMode mode, int flags);
+
+#define AV_UTF8_FLAG_ACCEPT_INVALID_BIG_CODES          1 ///< accept codepoints over 0x10FFFF
+#define AV_UTF8_FLAG_ACCEPT_NON_CHARACTERS             2 ///< accept non-characters - 0xFFFE and 0xFFFF
+#define AV_UTF8_FLAG_ACCEPT_SURROGATES                 4 ///< accept UTF-16 surrogates codes
+#define AV_UTF8_FLAG_EXCLUDE_XML_INVALID_CONTROL_CODES 8 ///< exclude control codes not accepted by XML
+
+#define AV_UTF8_FLAG_ACCEPT_ALL \
+    AV_UTF8_FLAG_ACCEPT_INVALID_BIG_CODES|AV_UTF8_FLAG_ACCEPT_NON_CHARACTERS|AV_UTF8_FLAG_ACCEPT_SURROGATES
+
+/**
+ * Read and decode a single UTF-8 code point (character) from the
+ * buffer in *buf, and update *buf to point to the next byte to
+ * decode.
+ *
+ * In case of an invalid byte sequence, the pointer will be updated to
+ * the next byte after the invalid sequence and the function will
+ * return an error code.
+ *
+ * Depending on the specified flags, the function will also fail in
+ * case the decoded code point does not belong to a valid range.
+ *
+ * @note For speed-relevant code a carefully implemented use of
+ * GET_UTF8() may be preferred.
+ *
+ * @param codep   pointer used to return the parsed code in case of success.
+ *                The value in *codep is set even in case the range check fails.
+ * @param bufp    pointer to the address the first byte of the sequence
+ *                to decode, updated by the function to point to the
+ *                byte next after the decoded sequence
+ * @param buf_end pointer to the end of the buffer, points to the next
+ *                byte past the last in the buffer. This is used to
+ *                avoid buffer overreads (in case of an unfinished
+ *                UTF-8 sequence towards the end of the buffer).
+ * @param flags   a collection of AV_UTF8_FLAG_* flags
+ * @return >= 0 in case a sequence was successfully read, a negative
+ * value in case of invalid sequence
+ */
+av_warn_unused_result
+int av_utf8_decode(int32_t *codep, const uint8_t **bufp, const uint8_t *buf_end,
+                   unsigned int flags);
+
+/**
+ * Check if a name is in a list.
+ * @returns 0 if not found, or the 1 based index where it has been found in the
+ *            list.
+ */
+int av_match_list(const char *name, const char *list, char separator);
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_AVSTRING_H */
@@ -0,0 +1,343 @@
+/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_AVUTIL_H
+#define AVUTIL_AVUTIL_H
+
+/**
+ * @file
+ * external API header
+ */
+
+/**
+ * @mainpage
+ *
+ * @section ffmpeg_intro Introduction
+ *
+ * This document describes the usage of the different libraries
+ * provided by FFmpeg.
+ *
+ * @li @ref libavc "libavcodec" encoding/decoding library
+ * @li @ref lavfi "libavfilter" graph-based frame editing library
+ * @li @ref libavf "libavformat" I/O and muxing/demuxing library
+ * @li @ref lavd "libavdevice" special devices muxing/demuxing library
+ * @li @ref lavu "libavutil" common utility library
+ * @li @ref lswr "libswresample" audio resampling, format conversion and mixing
+ * @li @ref lpp  "libpostproc" post processing library
+ * @li @ref libsws "libswscale" color conversion and scaling library
+ *
+ * @section ffmpeg_versioning Versioning and compatibility
+ *
+ * Each of the FFmpeg libraries contains a version.h header, which defines a
+ * major, minor and micro version number with the
+ * <em>LIBRARYNAME_VERSION_{MAJOR,MINOR,MICRO}</em> macros. The major version
+ * number is incremented with backward incompatible changes - e.g. removing
+ * parts of the public API, reordering public struct members, etc. The minor
+ * version number is incremented for backward compatible API changes or major
+ * new features - e.g. adding a new public function or a new decoder. The micro
+ * version number is incremented for smaller changes that a calling program
+ * might still want to check for - e.g. changing behavior in a previously
+ * unspecified situation.
+ *
+ * FFmpeg guarantees backward API and ABI compatibility for each library as long
+ * as its major version number is unchanged. This means that no public symbols
+ * will be removed or renamed. Types and names of the public struct members and
+ * values of public macros and enums will remain the same (unless they were
+ * explicitly declared as not part of the public API). Documented behavior will
+ * not change.
+ *
+ * In other words, any correct program that works with a given FFmpeg snapshot
+ * should work just as well without any changes with any later snapshot with the
+ * same major versions. This applies to both rebuilding the program against new
+ * FFmpeg versions or to replacing the dynamic FFmpeg libraries that a program
+ * links against.
+ *
+ * However, new public symbols may be added and new members may be appended to
+ * public structs whose size is not part of public ABI (most public structs in
+ * FFmpeg). New macros and enum values may be added. Behavior in undocumented
+ * situations may change slightly (and be documented). All those are accompanied
+ * by an entry in doc/APIchanges and incrementing either the minor or micro
+ * version number.
+ */
+
+/**
+ * @defgroup lavu Common utility functions
+ *
+ * @brief
+ * libavutil contains the code shared across all the other FFmpeg
+ * libraries
+ *
+ * @note In order to use the functions provided by avutil you must include
+ * the specific header.
+ *
+ * @{
+ *
+ * @defgroup lavu_crypto Crypto and Hashing
+ *
+ * @{
+ * @}
+ *
+ * @defgroup lavu_math Maths
+ * @{
+ *
+ * @}
+ *
+ * @defgroup lavu_string String Manipulation
+ *
+ * @{
+ *
+ * @}
+ *
+ * @defgroup lavu_mem Memory Management
+ *
+ * @{
+ *
+ * @}
+ *
+ * @defgroup lavu_data Data Structures
+ * @{
+ *
+ * @}
+ *
+ * @defgroup lavu_audio Audio related
+ *
+ * @{
+ *
+ * @}
+ *
+ * @defgroup lavu_error Error Codes
+ *
+ * @{
+ *
+ * @}
+ *
+ * @defgroup lavu_log Logging Facility
+ *
+ * @{
+ *
+ * @}
+ *
+ * @defgroup lavu_misc Other
+ *
+ * @{
+ *
+ * @defgroup preproc_misc Preprocessor String Macros
+ *
+ * @{
+ *
+ * @}
+ *
+ * @defgroup version_utils Library Version Macros
+ *
+ * @{
+ *
+ * @}
+ */
+
+
+/**
+ * @addtogroup lavu_ver
+ * @{
+ */
+
+/**
+ * Return the LIBAVUTIL_VERSION_INT constant.
+ */
+unsigned avutil_version(void);
+
+/**
+ * Return an informative version string. This usually is the actual release
+ * version number or a git commit description. This string has no fixed format
+ * and can change any time. It should never be parsed by code.
+ */
+const char *av_version_info(void);
+
+/**
+ * Return the libavutil build-time configuration.
+ */
+const char *avutil_configuration(void);
+
+/**
+ * Return the libavutil license.
+ */
+const char *avutil_license(void);
+
+/**
+ * @}
+ */
+
+/**
+ * @addtogroup lavu_media Media Type
+ * @brief Media Type
+ */
+
+enum AVMediaType {
+    AVMEDIA_TYPE_UNKNOWN = -1,  ///< Usually treated as AVMEDIA_TYPE_DATA
+    AVMEDIA_TYPE_VIDEO,
+    AVMEDIA_TYPE_AUDIO,
+    AVMEDIA_TYPE_DATA,          ///< Opaque data information usually continuous
+    AVMEDIA_TYPE_SUBTITLE,
+    AVMEDIA_TYPE_ATTACHMENT,    ///< Opaque data information usually sparse
+    AVMEDIA_TYPE_NB
+};
+
+/**
+ * Return a string describing the media_type enum, NULL if media_type
+ * is unknown.
+ */
+const char *av_get_media_type_string(enum AVMediaType media_type);
+
+/**
+ * @defgroup lavu_const Constants
+ * @{
+ *
+ * @defgroup lavu_enc Encoding specific
+ *
+ * @note those definition should move to avcodec
+ * @{
+ */
+
+#define FF_LAMBDA_SHIFT 7
+#define FF_LAMBDA_SCALE (1<<FF_LAMBDA_SHIFT)
+#define FF_QP2LAMBDA 118 ///< factor to convert from H.263 QP to lambda
+#define FF_LAMBDA_MAX (256*128-1)
+
+#define FF_QUALITY_SCALE FF_LAMBDA_SCALE //FIXME maybe remove
+
+/**
+ * @}
+ * @defgroup lavu_time Timestamp specific
+ *
+ * FFmpeg internal timebase and timestamp definitions
+ *
+ * @{
+ */
+
+/**
+ * @brief Undefined timestamp value
+ *
+ * Usually reported by demuxer that work on containers that do not provide
+ * either pts or dts.
+ */
+
+#define AV_NOPTS_VALUE          ((int64_t)UINT64_C(0x8000000000000000))
+
+/**
+ * Internal time base represented as integer
+ */
+
+#define AV_TIME_BASE            1000000
+
+/**
+ * Internal time base represented as fractional value
+ */
+
+#define AV_TIME_BASE_Q          (AVRational){1, AV_TIME_BASE}
+
+/**
+ * @}
+ * @}
+ * @defgroup lavu_picture Image related
+ *
+ * AVPicture types, pixel formats and basic image planes manipulation.
+ *
+ * @{
+ */
+
+enum AVPictureType {
+    AV_PICTURE_TYPE_NONE = 0, ///< Undefined
+    AV_PICTURE_TYPE_I,     ///< Intra
+    AV_PICTURE_TYPE_P,     ///< Predicted
+    AV_PICTURE_TYPE_B,     ///< Bi-dir predicted
+    AV_PICTURE_TYPE_S,     ///< S(GMC)-VOP MPEG4
+    AV_PICTURE_TYPE_SI,    ///< Switching Intra
+    AV_PICTURE_TYPE_SP,    ///< Switching Predicted
+    AV_PICTURE_TYPE_BI,    ///< BI type
+};
+
+/**
+ * Return a single letter to describe the given picture type
+ * pict_type.
+ *
+ * @param[in] pict_type the picture type @return a single character
+ * representing the picture type, '?' if pict_type is unknown
+ */
+char av_get_picture_type_char(enum AVPictureType pict_type);
+
+/**
+ * @}
+ */
+
+#include "common.h"
+#include "error.h"
+#include "rational.h"
+#include "version.h"
+#include "macros.h"
+#include "mathematics.h"
+#include "log.h"
+#include "pixfmt.h"
+
+/**
+ * Return x default pointer in case p is NULL.
+ */
+static inline void *av_x_if_null(const void *p, const void *x)
+{
+    return (void *)(intptr_t)(p ? p : x);
+}
+
+/**
+ * Compute the length of an integer list.
+ *
+ * @param elsize  size in bytes of each list element (only 1, 2, 4 or 8)
+ * @param term    list terminator (usually 0 or -1)
+ * @param list    pointer to the list
+ * @return  length of the list, in elements, not counting the terminator
+ */
+unsigned av_int_list_length_for_size(unsigned elsize,
+                                     const void *list, uint64_t term) av_pure;
+
+/**
+ * Compute the length of an integer list.
+ *
+ * @param term  list terminator (usually 0 or -1)
+ * @param list  pointer to the list
+ * @return  length of the list, in elements, not counting the terminator
+ */
+#define av_int_list_length(list, term) \
+    av_int_list_length_for_size(sizeof(*(list)), list, term)
+
+/**
+ * Open a file using a UTF-8 filename.
+ * The API of this function matches POSIX fopen(), errors are returned through
+ * errno.
+ */
+FILE *av_fopen_utf8(const char *path, const char *mode);
+
+/**
+ * Return the fractional representation of the internal time base.
+ */
+AVRational av_get_time_base_q(void);
+
+/**
+ * @}
+ * @}
+ */
+
+#endif /* AVUTIL_AVUTIL_H */
@@ -0,0 +1,67 @@
+/*
+ * Copyright (c) 2006 Ryan Martell. (rdm4@martellventures.com)
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_BASE64_H
+#define AVUTIL_BASE64_H
+
+#include <stdint.h>
+
+/**
+ * @defgroup lavu_base64 Base64
+ * @ingroup lavu_crypto
+ * @{
+ */
+
+
+/**
+ * Decode a base64-encoded string.
+ *
+ * @param out      buffer for decoded data
+ * @param in       null-terminated input string
+ * @param out_size size in bytes of the out buffer, must be at
+ *                 least 3/4 of the length of in
+ * @return         number of bytes written, or a negative value in case of
+ *                 invalid input
+ */
+int av_base64_decode(uint8_t *out, const char *in, int out_size);
+
+/**
+ * Encode data to base64 and null-terminate.
+ *
+ * @param out      buffer for encoded data
+ * @param out_size size in bytes of the out buffer (including the
+ *                 null terminator), must be at least AV_BASE64_SIZE(in_size)
+ * @param in       input buffer containing the data to encode
+ * @param in_size  size in bytes of the in buffer
+ * @return         out or NULL in case of error
+ */
+char *av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size);
+
+/**
+ * Calculate the output size needed to base64-encode x bytes to a
+ * null-terminated string.
+ */
+#define AV_BASE64_SIZE(x)  (((x)+2) / 3 * 4 + 1)
+
+ /**
+  * @}
+  */
+
+#endif /* AVUTIL_BASE64_H */
@@ -0,0 +1,82 @@
+/*
+ * Blowfish algorithm
+ * Copyright (c) 2012 Samuel Pitoiset
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_BLOWFISH_H
+#define AVUTIL_BLOWFISH_H
+
+#include <stdint.h>
+
+/**
+ * @defgroup lavu_blowfish Blowfish
+ * @ingroup lavu_crypto
+ * @{
+ */
+
+#define AV_BF_ROUNDS 16
+
+typedef struct AVBlowfish {
+    uint32_t p[AV_BF_ROUNDS + 2];
+    uint32_t s[4][256];
+} AVBlowfish;
+
+/**
+ * Allocate an AVBlowfish context.
+ */
+AVBlowfish *av_blowfish_alloc(void);
+
+/**
+ * Initialize an AVBlowfish context.
+ *
+ * @param ctx an AVBlowfish context
+ * @param key a key
+ * @param key_len length of the key
+ */
+void av_blowfish_init(struct AVBlowfish *ctx, const uint8_t *key, int key_len);
+
+/**
+ * Encrypt or decrypt a buffer using a previously initialized context.
+ *
+ * @param ctx an AVBlowfish context
+ * @param xl left four bytes halves of input to be encrypted
+ * @param xr right four bytes halves of input to be encrypted
+ * @param decrypt 0 for encryption, 1 for decryption
+ */
+void av_blowfish_crypt_ecb(struct AVBlowfish *ctx, uint32_t *xl, uint32_t *xr,
+                           int decrypt);
+
+/**
+ * Encrypt or decrypt a buffer using a previously initialized context.
+ *
+ * @param ctx an AVBlowfish context
+ * @param dst destination array, can be equal to src
+ * @param src source array, can be equal to dst
+ * @param count number of 8 byte blocks
+ * @param iv initialization vector for CBC mode, if NULL ECB will be used
+ * @param decrypt 0 for encryption, 1 for decryption
+ */
+void av_blowfish_crypt(struct AVBlowfish *ctx, uint8_t *dst, const uint8_t *src,
+                       int count, uint8_t *iv, int decrypt);
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_BLOWFISH_H */
@@ -0,0 +1,219 @@
+/*
+ * Copyright (c) 2012 Nicolas George
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_BPRINT_H
+#define AVUTIL_BPRINT_H
+
+#include <stdarg.h>
+
+#include "attributes.h"
+#include "avstring.h"
+
+/**
+ * Define a structure with extra padding to a fixed size
+ * This helps ensuring binary compatibility with future versions.
+ */
+
+#define FF_PAD_STRUCTURE(name, size, ...) \
+struct ff_pad_helper_##name { __VA_ARGS__ }; \
+typedef struct name { \
+    __VA_ARGS__ \
+    char reserved_padding[size - sizeof(struct ff_pad_helper_##name)]; \
+} name;
+
+/**
+ * Buffer to print data progressively
+ *
+ * The string buffer grows as necessary and is always 0-terminated.
+ * The content of the string is never accessed, and thus is
+ * encoding-agnostic and can even hold binary data.
+ *
+ * Small buffers are kept in the structure itself, and thus require no
+ * memory allocation at all (unless the contents of the buffer is needed
+ * after the structure goes out of scope). This is almost as lightweight as
+ * declaring a local "char buf[512]".
+ *
+ * The length of the string can go beyond the allocated size: the buffer is
+ * then truncated, but the functions still keep account of the actual total
+ * length.
+ *
+ * In other words, buf->len can be greater than buf->size and records the
+ * total length of what would have been to the buffer if there had been
+ * enough memory.
+ *
+ * Append operations do not need to be tested for failure: if a memory
+ * allocation fails, data stop being appended to the buffer, but the length
+ * is still updated. This situation can be tested with
+ * av_bprint_is_complete().
+ *
+ * The size_max field determines several possible behaviours:
+ *
+ * size_max = -1 (= UINT_MAX) or any large value will let the buffer be
+ * reallocated as necessary, with an amortized linear cost.
+ *
+ * size_max = 0 prevents writing anything to the buffer: only the total
+ * length is computed. The write operations can then possibly be repeated in
+ * a buffer with exactly the necessary size
+ * (using size_init = size_max = len + 1).
+ *
+ * size_max = 1 is automatically replaced by the exact size available in the
+ * structure itself, thus ensuring no dynamic memory allocation. The
+ * internal buffer is large enough to hold a reasonable paragraph of text,
+ * such as the current paragraph.
+ */
+
+FF_PAD_STRUCTURE(AVBPrint, 1024,
+    char *str;         /**< string so far */
+    unsigned len;      /**< length so far */
+    unsigned size;     /**< allocated memory */
+    unsigned size_max; /**< maximum allocated memory */
+    char reserved_internal_buffer[1];
+)
+
+/**
+ * Convenience macros for special values for av_bprint_init() size_max
+ * parameter.
+ */
+#define AV_BPRINT_SIZE_UNLIMITED  ((unsigned)-1)
+#define AV_BPRINT_SIZE_AUTOMATIC  1
+#define AV_BPRINT_SIZE_COUNT_ONLY 0
+
+/**
+ * Init a print buffer.
+ *
+ * @param buf        buffer to init
+ * @param size_init  initial size (including the final 0)
+ * @param size_max   maximum size;
+ *                   0 means do not write anything, just count the length;
+ *                   1 is replaced by the maximum value for automatic storage;
+ *                   any large value means that the internal buffer will be
+ *                   reallocated as needed up to that limit; -1 is converted to
+ *                   UINT_MAX, the largest limit possible.
+ *                   Check also AV_BPRINT_SIZE_* macros.
+ */
+void av_bprint_init(AVBPrint *buf, unsigned size_init, unsigned size_max);
+
+/**
+ * Init a print buffer using a pre-existing buffer.
+ *
+ * The buffer will not be reallocated.
+ *
+ * @param buf     buffer structure to init
+ * @param buffer  byte buffer to use for the string data
+ * @param size    size of buffer
+ */
+void av_bprint_init_for_buffer(AVBPrint *buf, char *buffer, unsigned size);
+
+/**
+ * Append a formatted string to a print buffer.
+ */
+void av_bprintf(AVBPrint *buf, const char *fmt, ...) av_printf_format(2, 3);
+
+/**
+ * Append a formatted string to a print buffer.
+ */
+void av_vbprintf(AVBPrint *buf, const char *fmt, va_list vl_arg);
+
+/**
+ * Append char c n times to a print buffer.
+ */
+void av_bprint_chars(AVBPrint *buf, char c, unsigned n);
+
+/**
+ * Append data to a print buffer.
+ *
+ * param buf  bprint buffer to use
+ * param data pointer to data
+ * param size size of data
+ */
+void av_bprint_append_data(AVBPrint *buf, const char *data, unsigned size);
+
+struct tm;
+/**
+ * Append a formatted date and time to a print buffer.
+ *
+ * param buf  bprint buffer to use
+ * param fmt  date and time format string, see strftime()
+ * param tm   broken-down time structure to translate
+ *
+ * @note due to poor design of the standard strftime function, it may
+ * produce poor results if the format string expands to a very long text and
+ * the bprint buffer is near the limit stated by the size_max option.
+ */
+void av_bprint_strftime(AVBPrint *buf, const char *fmt, const struct tm *tm);
+
+/**
+ * Allocate bytes in the buffer for external use.
+ *
+ * @param[in]  buf          buffer structure
+ * @param[in]  size         required size
+ * @param[out] mem          pointer to the memory area
+ * @param[out] actual_size  size of the memory area after allocation;
+ *                          can be larger or smaller than size
+ */
+void av_bprint_get_buffer(AVBPrint *buf, unsigned size,
+                          unsigned char **mem, unsigned *actual_size);
+
+/**
+ * Reset the string to "" but keep internal allocated data.
+ */
+void av_bprint_clear(AVBPrint *buf);
+
+/**
+ * Test if the print buffer is complete (not truncated).
+ *
+ * It may have been truncated due to a memory allocation failure
+ * or the size_max limit (compare size and size_max if necessary).
+ */
+static inline int av_bprint_is_complete(const AVBPrint *buf)
+{
+    return buf->len < buf->size;
+}
+
+/**
+ * Finalize a print buffer.
+ *
+ * The print buffer can no longer be used afterwards,
+ * but the len and size fields are still valid.
+ *
+ * @arg[out] ret_str  if not NULL, used to return a permanent copy of the
+ *                    buffer contents, or NULL if memory allocation fails;
+ *                    if NULL, the buffer is discarded and freed
+ * @return  0 for success or error code (probably AVERROR(ENOMEM))
+ */
+int av_bprint_finalize(AVBPrint *buf, char **ret_str);
+
+/**
+ * Escape the content in src and append it to dstbuf.
+ *
+ * @param dstbuf        already inited destination bprint buffer
+ * @param src           string containing the text to escape
+ * @param special_chars string containing the special characters which
+ *                      need to be escaped, can be NULL
+ * @param mode          escape mode to employ, see AV_ESCAPE_MODE_* macros.
+ *                      Any unknown value for mode will be considered equivalent to
+ *                      AV_ESCAPE_MODE_BACKSLASH, but this behaviour can change without
+ *                      notice.
+ * @param flags         flags which control how to escape, see AV_ESCAPE_FLAG_* macros
+ */
+void av_bprint_escape(AVBPrint *dstbuf, const char *src, const char *special_chars,
+                      enum AVEscapeMode mode, int flags);
+
+#endif /* AVUTIL_BPRINT_H */
@@ -0,0 +1,109 @@
+/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * byte swapping routines
+ */
+
+#ifndef AVUTIL_BSWAP_H
+#define AVUTIL_BSWAP_H
+
+#include <stdint.h>
+#include "libavutil/avconfig.h"
+#include "attributes.h"
+
+#ifdef HAVE_AV_CONFIG_H
+
+#include "config.h"
+
+#if   ARCH_AARCH64
+#   include "aarch64/bswap.h"
+#elif ARCH_ARM
+#   include "arm/bswap.h"
+#elif ARCH_AVR32
+#   include "avr32/bswap.h"
+#elif ARCH_SH4
+#   include "sh4/bswap.h"
+#elif ARCH_X86
+#   include "x86/bswap.h"
+#endif
+
+#endif /* HAVE_AV_CONFIG_H */
+
+#define AV_BSWAP16C(x) (((x) << 8 & 0xff00)  | ((x) >> 8 & 0x00ff))
+#define AV_BSWAP32C(x) (AV_BSWAP16C(x) << 16 | AV_BSWAP16C((x) >> 16))
+#define AV_BSWAP64C(x) (AV_BSWAP32C(x) << 32 | AV_BSWAP32C((x) >> 32))
+
+#define AV_BSWAPC(s, x) AV_BSWAP##s##C(x)
+
+#ifndef av_bswap16
+static av_always_inline av_const uint16_t av_bswap16(uint16_t x)
+{
+    x= (x>>8) | (x<<8);
+    return x;
+}
+#endif
+
+#ifndef av_bswap32
+static av_always_inline av_const uint32_t av_bswap32(uint32_t x)
+{
+    return AV_BSWAP32C(x);
+}
+#endif
+
+#ifndef av_bswap64
+static inline uint64_t av_const av_bswap64(uint64_t x)
+{
+    return (uint64_t)av_bswap32(x) << 32 | av_bswap32(x >> 32);
+}
+#endif
+
+// be2ne ... big-endian to native-endian
+// le2ne ... little-endian to native-endian
+
+#if AV_HAVE_BIGENDIAN
+#define av_be2ne16(x) (x)
+#define av_be2ne32(x) (x)
+#define av_be2ne64(x) (x)
+#define av_le2ne16(x) av_bswap16(x)
+#define av_le2ne32(x) av_bswap32(x)
+#define av_le2ne64(x) av_bswap64(x)
+#define AV_BE2NEC(s, x) (x)
+#define AV_LE2NEC(s, x) AV_BSWAPC(s, x)
+#else
+#define av_be2ne16(x) av_bswap16(x)
+#define av_be2ne32(x) av_bswap32(x)
+#define av_be2ne64(x) av_bswap64(x)
+#define av_le2ne16(x) (x)
+#define av_le2ne32(x) (x)
+#define av_le2ne64(x) (x)
+#define AV_BE2NEC(s, x) AV_BSWAPC(s, x)
+#define AV_LE2NEC(s, x) (x)
+#endif
+
+#define AV_BE2NE16C(x) AV_BE2NEC(16, x)
+#define AV_BE2NE32C(x) AV_BE2NEC(32, x)
+#define AV_BE2NE64C(x) AV_BE2NEC(64, x)
+#define AV_LE2NE16C(x) AV_LE2NEC(16, x)
+#define AV_LE2NE32C(x) AV_LE2NEC(32, x)
+#define AV_LE2NE64C(x) AV_LE2NEC(64, x)
+
+#endif /* AVUTIL_BSWAP_H */
@@ -0,0 +1,274 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * @ingroup lavu_buffer
+ * refcounted data buffer API
+ */
+
+#ifndef AVUTIL_BUFFER_H
+#define AVUTIL_BUFFER_H
+
+#include <stdint.h>
+
+/**
+ * @defgroup lavu_buffer AVBuffer
+ * @ingroup lavu_data
+ *
+ * @{
+ * AVBuffer is an API for reference-counted data buffers.
+ *
+ * There are two core objects in this API -- AVBuffer and AVBufferRef. AVBuffer
+ * represents the data buffer itself; it is opaque and not meant to be accessed
+ * by the caller directly, but only through AVBufferRef. However, the caller may
+ * e.g. compare two AVBuffer pointers to check whether two different references
+ * are describing the same data buffer. AVBufferRef represents a single
+ * reference to an AVBuffer and it is the object that may be manipulated by the
+ * caller directly.
+ *
+ * There are two functions provided for creating a new AVBuffer with a single
+ * reference -- av_buffer_alloc() to just allocate a new buffer, and
+ * av_buffer_create() to wrap an existing array in an AVBuffer. From an existing
+ * reference, additional references may be created with av_buffer_ref().
+ * Use av_buffer_unref() to free a reference (this will automatically free the
+ * data once all the references are freed).
+ *
+ * The convention throughout this API and the rest of FFmpeg is such that the
+ * buffer is considered writable if there exists only one reference to it (and
+ * it has not been marked as read-only). The av_buffer_is_writable() function is
+ * provided to check whether this is true and av_buffer_make_writable() will
+ * automatically create a new writable buffer when necessary.
+ * Of course nothing prevents the calling code from violating this convention,
+ * however that is safe only when all the existing references are under its
+ * control.
+ *
+ * @note Referencing and unreferencing the buffers is thread-safe and thus
+ * may be done from multiple threads simultaneously without any need for
+ * additional locking.
+ *
+ * @note Two different references to the same buffer can point to different
+ * parts of the buffer (i.e. their AVBufferRef.data will not be equal).
+ */
+
+/**
+ * A reference counted buffer type. It is opaque and is meant to be used through
+ * references (AVBufferRef).
+ */
+typedef struct AVBuffer AVBuffer;
+
+/**
+ * A reference to a data buffer.
+ *
+ * The size of this struct is not a part of the public ABI and it is not meant
+ * to be allocated directly.
+ */
+typedef struct AVBufferRef {
+    AVBuffer *buffer;
+
+    /**
+     * The data buffer. It is considered writable if and only if
+     * this is the only reference to the buffer, in which case
+     * av_buffer_is_writable() returns 1.
+     */
+    uint8_t *data;
+    /**
+     * Size of data in bytes.
+     */
+    int      size;
+} AVBufferRef;
+
+/**
+ * Allocate an AVBuffer of the given size using av_malloc().
+ *
+ * @return an AVBufferRef of given size or NULL when out of memory
+ */
+AVBufferRef *av_buffer_alloc(int size);
+
+/**
+ * Same as av_buffer_alloc(), except the returned buffer will be initialized
+ * to zero.
+ */
+AVBufferRef *av_buffer_allocz(int size);
+
+/**
+ * Always treat the buffer as read-only, even when it has only one
+ * reference.
+ */
+#define AV_BUFFER_FLAG_READONLY (1 << 0)
+
+/**
+ * Create an AVBuffer from an existing array.
+ *
+ * If this function is successful, data is owned by the AVBuffer. The caller may
+ * only access data through the returned AVBufferRef and references derived from
+ * it.
+ * If this function fails, data is left untouched.
+ * @param data   data array
+ * @param size   size of data in bytes
+ * @param free   a callback for freeing this buffer's data
+ * @param opaque parameter to be got for processing or passed to free
+ * @param flags  a combination of AV_BUFFER_FLAG_*
+ *
+ * @return an AVBufferRef referring to data on success, NULL on failure.
+ */
+AVBufferRef *av_buffer_create(uint8_t *data, int size,
+                              void (*free)(void *opaque, uint8_t *data),
+                              void *opaque, int flags);
+
+/**
+ * Default free callback, which calls av_free() on the buffer data.
+ * This function is meant to be passed to av_buffer_create(), not called
+ * directly.
+ */
+void av_buffer_default_free(void *opaque, uint8_t *data);
+
+/**
+ * Create a new reference to an AVBuffer.
+ *
+ * @return a new AVBufferRef referring to the same AVBuffer as buf or NULL on
+ * failure.
+ */
+AVBufferRef *av_buffer_ref(AVBufferRef *buf);
+
+/**
+ * Free a given reference and automatically free the buffer if there are no more
+ * references to it.
+ *
+ * @param buf the reference to be freed. The pointer is set to NULL on return.
+ */
+void av_buffer_unref(AVBufferRef **buf);
+
+/**
+ * @return 1 if the caller may write to the data referred to by buf (which is
+ * true if and only if buf is the only reference to the underlying AVBuffer).
+ * Return 0 otherwise.
+ * A positive answer is valid until av_buffer_ref() is called on buf.
+ */
+int av_buffer_is_writable(const AVBufferRef *buf);
+
+/**
+ * @return the opaque parameter set by av_buffer_create.
+ */
+void *av_buffer_get_opaque(const AVBufferRef *buf);
+
+int av_buffer_get_ref_count(const AVBufferRef *buf);
+
+/**
+ * Create a writable reference from a given buffer reference, avoiding data copy
+ * if possible.
+ *
+ * @param buf buffer reference to make writable. On success, buf is either left
+ *            untouched, or it is unreferenced and a new writable AVBufferRef is
+ *            written in its place. On failure, buf is left untouched.
+ * @return 0 on success, a negative AVERROR on failure.
+ */
+int av_buffer_make_writable(AVBufferRef **buf);
+
+/**
+ * Reallocate a given buffer.
+ *
+ * @param buf  a buffer reference to reallocate. On success, buf will be
+ *             unreferenced and a new reference with the required size will be
+ *             written in its place. On failure buf will be left untouched. *buf
+ *             may be NULL, then a new buffer is allocated.
+ * @param size required new buffer size.
+ * @return 0 on success, a negative AVERROR on failure.
+ *
+ * @note the buffer is actually reallocated with av_realloc() only if it was
+ * initially allocated through av_buffer_realloc(NULL) and there is only one
+ * reference to it (i.e. the one passed to this function). In all other cases
+ * a new buffer is allocated and the data is copied.
+ */
+int av_buffer_realloc(AVBufferRef **buf, int size);
+
+/**
+ * @}
+ */
+
+/**
+ * @defgroup lavu_bufferpool AVBufferPool
+ * @ingroup lavu_data
+ *
+ * @{
+ * AVBufferPool is an API for a lock-free thread-safe pool of AVBuffers.
+ *
+ * Frequently allocating and freeing large buffers may be slow. AVBufferPool is
+ * meant to solve this in cases when the caller needs a set of buffers of the
+ * same size (the most obvious use case being buffers for raw video or audio
+ * frames).
+ *
+ * At the beginning, the user must call av_buffer_pool_init() to create the
+ * buffer pool. Then whenever a buffer is needed, call av_buffer_pool_get() to
+ * get a reference to a new buffer, similar to av_buffer_alloc(). This new
+ * reference works in all aspects the same way as the one created by
+ * av_buffer_alloc(). However, when the last reference to this buffer is
+ * unreferenced, it is returned to the pool instead of being freed and will be
+ * reused for subsequent av_buffer_pool_get() calls.
+ *
+ * When the caller is done with the pool and no longer needs to allocate any new
+ * buffers, av_buffer_pool_uninit() must be called to mark the pool as freeable.
+ * Once all the buffers are released, it will automatically be freed.
+ *
+ * Allocating and releasing buffers with this API is thread-safe as long as
+ * either the default alloc callback is used, or the user-supplied one is
+ * thread-safe.
+ */
+
+/**
+ * The buffer pool. This structure is opaque and not meant to be accessed
+ * directly. It is allocated with av_buffer_pool_init() and freed with
+ * av_buffer_pool_uninit().
+ */
+typedef struct AVBufferPool AVBufferPool;
+
+/**
+ * Allocate and initialize a buffer pool.
+ *
+ * @param size size of each buffer in this pool
+ * @param alloc a function that will be used to allocate new buffers when the
+ * pool is empty. May be NULL, then the default allocator will be used
+ * (av_buffer_alloc()).
+ * @return newly created buffer pool on success, NULL on error.
+ */
+AVBufferPool *av_buffer_pool_init(int size, AVBufferRef* (*alloc)(int size));
+
+/**
+ * Mark the pool as being available for freeing. It will actually be freed only
+ * once all the allocated buffers associated with the pool are released. Thus it
+ * is safe to call this function while some of the allocated buffers are still
+ * in use.
+ *
+ * @param pool pointer to the pool to be freed. It will be set to NULL.
+ * @see av_buffer_pool_can_uninit()
+ */
+void av_buffer_pool_uninit(AVBufferPool **pool);
+
+/**
+ * Allocate a new AVBuffer, reusing an old buffer from the pool when available.
+ * This function may be called simultaneously from multiple threads.
+ *
+ * @return a reference to the new buffer on success, NULL on error.
+ */
+AVBufferRef *av_buffer_pool_get(AVBufferPool *pool);
+
+/**
+ * @}
+ */
+
+#endif /* AVUTIL_BUFFER_H */
--- a/Show More
+++ b/Show More