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Update OpenAL-soft to 1.21.1

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This commit is contained in:
Miku AuahDark
2021-02-11 18:04:50 +08:00
parent 08f227997e
commit 904cc75ba8
358 changed files with 61122 additions and 60973 deletions
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libs/openal-soft/examples/alhrtf.c
+62 -54
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@@ -24,11 +24,15 @@
/* This file contains an example for selecting an HRTF. */
#include <stdio.h>
#include <assert.h>
#include <inttypes.h>
#include <limits.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <SDL_sound.h>
#include "sndfile.h"
#include "AL/al.h"
#include "AL/alc.h"
@@ -49,68 +53,73 @@ static LPALCRESETDEVICESOFT alcResetDeviceSOFT;
*/
static ALuint LoadSound(const char *filename)
{
Sound_Sample *sample;
ALenum err, format;
ALuint buffer;
Uint32 slen;
SNDFILE *sndfile;
SF_INFO sfinfo;
short *membuf;
sf_count_t num_frames;
ALsizei num_bytes;
/* Open the audio file */
sample = Sound_NewSampleFromFile(filename, NULL, 65536);
if(!sample)
/* Open the audio file and check that it's usable. */
sndfile = sf_open(filename, SFM_READ, &sfinfo);
if(!sndfile)
{
fprintf(stderr, "Could not open audio in %s\n", filename);
fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
return 0;
}
if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(short))/sfinfo.channels)
{
fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
sf_close(sndfile);
return 0;
}
/* Get the sound format, and figure out the OpenAL format */
if(sample->actual.channels == 1)
format = AL_NONE;
if(sfinfo.channels == 1)
format = AL_FORMAT_MONO16;
else if(sfinfo.channels == 2)
format = AL_FORMAT_STEREO16;
else if(sfinfo.channels == 3)
{
if(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_MONO8;
else if(sample->actual.format == AUDIO_S16SYS)
format = AL_FORMAT_MONO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
Sound_FreeSample(sample);
return 0;
}
if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
format = AL_FORMAT_BFORMAT2D_16;
}
else if(sample->actual.channels == 2)
else if(sfinfo.channels == 4)
{
if(sample->actual.format == AUDIO_U8)
format = AL_FORMAT_STEREO8;
else if(sample->actual.format == AUDIO_S16SYS)
format = AL_FORMAT_STEREO16;
else
{
fprintf(stderr, "Unsupported sample format: 0x%04x\n", sample->actual.format);
Sound_FreeSample(sample);
return 0;
}
if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
format = AL_FORMAT_BFORMAT3D_16;
}
else
if(!format)
{
fprintf(stderr, "Unsupported channel count: %d\n", sample->actual.channels);
Sound_FreeSample(sample);
fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
sf_close(sndfile);
return 0;
}
/* Decode the whole audio stream to a buffer. */
slen = Sound_DecodeAll(sample);
if(!sample->buffer || slen == 0)
/* Decode the whole audio file to a buffer. */
membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(short));
num_frames = sf_readf_short(sndfile, membuf, sfinfo.frames);
if(num_frames < 1)
{
fprintf(stderr, "Failed to read audio from %s\n", filename);
Sound_FreeSample(sample);
free(membuf);
sf_close(sndfile);
fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
return 0;
}
num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(short);
/* Buffer the audio data into a new buffer object, then free the data and
* close the file. */
* close the file.
*/
buffer = 0;
alGenBuffers(1, &buffer);
alBufferData(buffer, format, sample->buffer, slen, sample->actual.rate);
Sound_FreeSample(sample);
alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
free(membuf);
sf_close(sndfile);
/* Check if an error occured, and clean up if so. */
err = alGetError();
@@ -129,6 +138,7 @@ static ALuint LoadSound(const char *filename)
int main(int argc, char **argv)
{
ALCdevice *device;
ALCcontext *context;
ALboolean has_angle_ext;
ALuint source, buffer;
const char *soundname;
@@ -150,7 +160,8 @@ int main(int argc, char **argv)
if(InitAL(&argv, &argc) != 0)
return 1;
device = alcGetContextsDevice(alcGetCurrentContext());
context = alcGetCurrentContext();
device = alcGetContextsDevice(context);
if(!alcIsExtensionPresent(device, "ALC_SOFT_HRTF"))
{
fprintf(stderr, "Error: ALC_SOFT_HRTF not supported\n");
@@ -159,16 +170,16 @@ int main(int argc, char **argv)
}
/* Define a macro to help load the function pointers. */
#define LOAD_PROC(d, x) ((x) = alcGetProcAddress((d), #x))
LOAD_PROC(device, alcGetStringiSOFT);
LOAD_PROC(device, alcResetDeviceSOFT);
#define LOAD_PROC(d, T, x) ((x) = (T)alcGetProcAddress((d), #x))
LOAD_PROC(device, LPALCGETSTRINGISOFT, alcGetStringiSOFT);
LOAD_PROC(device, LPALCRESETDEVICESOFT, alcResetDeviceSOFT);
#undef LOAD_PROC
/* Check for the AL_EXT_STEREO_ANGLES extension to be able to also rotate
* stereo sources.
*/
has_angle_ext = alIsExtensionPresent("AL_EXT_STEREO_ANGLES");
printf("AL_EXT_STEREO_ANGLES%s found\n", has_angle_ext?"":" not");
printf("AL_EXT_STEREO_ANGLES %sfound\n", has_angle_ext?"":"not ");
/* Check for user-preferred HRTF */
if(strcmp(argv[0], "-hrtf") == 0)
@@ -235,14 +246,10 @@ int main(int argc, char **argv)
}
fflush(stdout);
/* Initialize SDL_sound. */
Sound_Init();
/* Load the sound into a buffer. */
buffer = LoadSound(soundname);
if(!buffer)
{
Sound_Quit();
CloseAL();
return 1;
}
@@ -252,7 +259,7 @@ int main(int argc, char **argv)
alGenSources(1, &source);
alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE);
alSource3f(source, AL_POSITION, 0.0f, 0.0f, -1.0f);
alSourcei(source, AL_BUFFER, buffer);
alSourcei(source, AL_BUFFER, (ALint)buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
@@ -261,6 +268,8 @@ int main(int argc, char **argv)
do {
al_nssleep(10000000);
alcSuspendContext(context);
/* Rotate the source around the listener by about 1/4 cycle per second,
* and keep it within -pi...+pi.
*/
@@ -279,15 +288,14 @@ int main(int argc, char **argv)
ALfloat angles[2] = { (ALfloat)(M_PI/6.0 - angle), (ALfloat)(-M_PI/6.0 - angle) };
alSourcefv(source, AL_STEREO_ANGLES, angles);
}
alcProcessContext(context);
alGetSourcei(source, AL_SOURCE_STATE, &state);
} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
/* All done. Delete resources, and close down SDL_sound and OpenAL. */
/* All done. Delete resources, and close down OpenAL. */
alDeleteSources(1, &source);
alDeleteBuffers(1, &buffer);
Sound_Quit();
CloseAL();
return 0;