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https://github.com/love2d/love-android.git
synced 2026-08-19 12:14:49 +02:00
Updated openal-soft to version 1.15.1
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@@ -5,35 +5,36 @@
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# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
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# specific override settings in ~/.alsoftrc.
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# For Windows, these settings should go into %AppData%\alsoft.ini
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# The environment variable ALSOFT_CONF can be used to specify another config
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# override
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# Option and block names are case-insenstive. The supplied values are only
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# hints and may not be honored (though generally it'll try to get as close as
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# possible). Note: options that are left unset may default to app- or system-
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# specified values. These are the current available settings:
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## format:
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# Sets the output format. Can be one of:
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# AL_FORMAT_MONO8 (8-bit mono)
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# AL_FORMAT_STEREO8 (8-bit stereo)
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# AL_FORMAT_QUAD8 (8-bit 4-channel)
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# AL_FORMAT_51CHN8 (8-bit 5.1 output)
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# AL_FORMAT_61CHN8 (8-bit 6.1 output)
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# AL_FORMAT_71CHN8 (8-bit 7.1 output)
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# AL_FORMAT_MONO16 (16-bit mono)
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# AL_FORMAT_STEREO16 (16-bit stereo)
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# AL_FORMAT_QUAD16 (16-bit 4-channel)
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# AL_FORMAT_51CHN16 (16-bit 5.1 output)
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# AL_FORMAT_61CHN16 (16-bit 6.1 output)
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# AL_FORMAT_71CHN16 (16-bit 7.1 output)
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# AL_FORMAT_MONO32 (32-bit float mono)
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# AL_FORMAT_STEREO32 (32-bit float stereo)
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# AL_FORMAT_QUAD32 (32-bit float 4-channel)
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# AL_FORMAT_51CHN32 (32-bit float 5.1 output)
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# AL_FORMAT_61CHN32 (32-bit float 6.1 output)
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# AL_FORMAT_71CHN32 (32-bit float 7.1 output)
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#format = AL_FORMAT_STEREO16
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## disable-cpu-exts:
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# Disables use of the listed CPU extensions. Certain methods may utilize CPU
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# extensions when detected, and this option is useful for preventing those
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# extensions from being used. The available extensions are: sse, sse2, neon.
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# Specifying 'all' disables use of all extensions.
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#disable-cpu-exts =
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## channels:
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# Sets the output channel configuration. If left unspecified, one will try to
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# be detected from the system, and defaulting to stereo. The available values
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# are: mono, stereo, quad, surround51, surround61, surround71
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#channels = stereo
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## sample-type:
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# Sets the output sample type. Currently, all mixing is done with 32-bit float
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# and converted to the output sample type as needed. Available values are:
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# int8 - signed 8-bit int
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# uint8 - unsigned 8-bit int
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# int16 - signed 16-bit int
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# uint16 - unsigned 16-bit int
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# int32 - signed 32-bit int
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# uint32 - unsigned 32-bit int
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# float32 - 32-bit float
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#sample-type = float32
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## hrtf:
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# Enables HRTF filters. These filters provide for better sound spatialization
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@@ -42,6 +43,18 @@
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# Default is disabled since stereo speaker output quality may suffer.
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#hrtf = false
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## hrtf_tables
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# Specifies a comma-separated list of files containing HRTF data sets. The
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# listed data sets can be used in place of or in addiiton to the the built-in
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# set. The format of the files are described in hrtf.txt. The filenames may
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# contain these markers, which will be replaced as needed:
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# %r - Device sampling rate
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# %% - Percent sign (%)
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# So if this is set to "kemar-%r-diffuse.mhr", it will try to open
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# "kemar-44100-diffuse.mhr" if the device is using 44100hz output, or
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# "kemar-48000-diffuse.mhr" if the device is using 48000hz output, etc.
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#hrtf_tables =
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## cf_level:
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# Sets the crossfeed level for stereo output. Valid values are:
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# 0 - No crossfeed
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@@ -55,17 +68,25 @@
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# stereo modes.
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#cf_level = 0
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## wide-stereo:
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# Specifies that stereo sources are given a width of about 120 degrees on each
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# channel, centering on -90 (left) and +90 (right), as opposed to being points
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# placed at -30 (left) and +30 (right). This can be useful for surround-sound
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# to give stereo sources a more encompassing sound. Note that the sound's
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# overall volume will be slightly reduced to account for the extra output.
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#wide-stereo = false
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## frequency:
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# Sets the output frequency.
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#frequency = 44100
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## resampler:
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# Selects the resampler used when mixing sources. Valid values are:
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# 0 - None (nearest sample, no interpolation)
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# 1 - Linear (extrapolates samples using a linear slope between samples)
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# 2 - Cubic (extrapolates samples using a Catmull-Rom spline)
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# point - nearest sample, no interpolation
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# linear - extrapolates samples using a linear slope between samples
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# cubic - extrapolates samples using a Catmull-Rom spline
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# Specifying other values will result in using the default (linear).
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#resampler = 1
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#resampler = linear
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## rt-prio:
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# Sets real-time priority for the mixing thread. Not all drivers may use this
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@@ -78,13 +99,14 @@
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## period_size:
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# Sets the update period size, in frames. This is the number of frames needed
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# for each mixing update.
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# for each mixing update. Acceptable values range between 64 and 8192.
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#period_size = 1024
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## periods:
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# Sets the number of update periods. Higher values create a larger mix ahead,
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# which helps protect against skips when the CPU is under load, but increases
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# the delay between a sound getting mixed and being heard.
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# the delay between a sound getting mixed and being heard. Acceptable values
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# range between 2 and 16.
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#periods = 4
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## sources:
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@@ -92,16 +114,6 @@
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# systems with apps that try to play more sounds than the CPU can handle.
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#sources = 256
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## stereodup:
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# Sets whether to duplicate stereo sounds behind the listener for 4+ channel
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# output. This provides a "fuller" playback quality for surround sound output
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# modes, although each individual speaker will have a slight reduction in
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# volume to compensate for the extra output speakers. True, yes, on, and non-0
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# values will duplicate stereo sources. 0 and anything else will cause stereo
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# sounds to only play in front. This only has an effect when a suitable output
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# format is used (ie. those that contain side and/or rear speakers).
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#stereodup = true
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## drivers:
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# Sets the backend driver list order, comma-seperated. Unknown backends and
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# duplicated names are ignored. Unlisted backends won't be considered for use
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@@ -109,13 +121,13 @@
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# followed by all other available backends, while 'oss' will list OSS only).
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# Backends prepended with - won't be available for use (eg. '-oss,' will allow
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# all available backends except OSS). An empty list means the default.
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#drivers = pulse,alsa,core,oss,solaris,sndio,mmdevapi,dsound,winmm,port,opensl,null,wave
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#drivers = pulse,alsa,core,oss,solaris,sndio,qsa,mmdevapi,dsound,winmm,port,opensl,null,wave
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## excludefx:
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# Sets which effects to exclude, preventing apps from using them. This can
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# help for apps that try to use effects which are too CPU intensive for the
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# system to handle. Available effects are: eaxreverb,reverb,echo,modulator,
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# dedicated
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# system to handle. Available effects are: eaxreverb,reverb,autowah,chorus,
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# compressor,distortion,echo,equalizer,flanger,modulator,dedicated
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#excludefx =
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## slots:
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@@ -144,28 +156,50 @@
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# Channel-specific layouts may be specified to override the layout option. The
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# same speakers as the layout option are available, and the default settings
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# are shown below.
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#layout_STEREO = fl=-90, fr=90
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#layout_QUAD = fl=-45, fr=45, bl=-135, br=135
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#layout_51CHN = fl=-30, fr=30, fc=0, bl=-110, br=110
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#layout_61CHN = fl=-30, fr=30, fc=0, sl=-90, sr=90, bc=180
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#layout_71CHN = fl=-30, fr=30, fc=0, sl=-90, sr=90, bl=-150, br=150
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#layout_stereo = fl=-90, fr=90
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#layout_quad = fl=-45, fr=45, bl=-135, br=135
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#layout_surround51 = fl=-30, fr=30, fc=0, bl=-110, br=110
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#layout_surround61 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bc=180
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#layout_surround71 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bl=-150, br=150
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## default-reverb:
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# A reverb preset that applies by default to all sources on send 0
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# (applications that set their own slots on send 0 will override this).
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# Available presets are: None, Generic, PaddedCell, Room, Bathroom,
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# Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
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# CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
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# Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
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#default-reverb =
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## trap-alc-error:
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# Generates a SIGTRAP signal when an ALC device error is generated, on systems
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# that support it. This helps when debugging, while trying to find the cause
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# of a device error. On Windows, a breakpoint exception is generated.
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# Optionally, the __ALSOFT_TRAP_ALC_ERROR env var may be set before running
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# the app instead.
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#trap-alc-error = false
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## trap-al-error:
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# Generates a SIGTRAP signal when an AL context error is generated, on systems
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# that support it. This helps when debugging, while trying to find the cause
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# of a context error. On Windows, a breakpoint exception is generated.
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# Optionally, the __ALSOFT_TRAP_AL_ERROR env var may be set before running the
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# app instead.
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#trap-al-error = false
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##
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## MIDI stuff (EXPERIMENTAL)
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##
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[midi]
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## soundfont:
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# A default soundfont (sf2 format). Used when an app requests the system
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# default.
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#soundfont =
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## volume:
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# Additional attenuation applied to MIDI output, expressed in decibels. This
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# is used to help keep the mix from clipping, and so must be 0 or less. The
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# value is logarithmic, so -6 will be about half amplitude, and -12 about
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# 1/4th. The default is roughly -13.9794 (0.2, or 1/5th).
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#volume =
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##
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## Reverb effect stuff (includes EAX reverb)
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##
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@@ -184,6 +218,24 @@
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# allows a simpler effect to be used at the loss of some quality.
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#emulate-eax = false
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##
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## PulseAudio backend stuff
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##
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[pulse]
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## spawn-server:
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# Attempts to autospawn a PulseAudio server whenever needed (initializing the
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# backend, enumerating devices, etc). Setting autospawn to false in Pulse's
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# client.conf will still prevent autospawning even if this is set to true.
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#spawn-server = true
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## allow-moves:
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# Allows PulseAudio to move active streams to different devices. Note that the
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# device specifier (seen by applications) will not be updated when this
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# occurs, and neither will the AL device configuration (sample rate, format,
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# etc).
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#allow-moves = false
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##
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## ALSA backend stuff
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##
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@@ -199,16 +251,30 @@
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# device index for the requested device name.
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#device-prefix = plughw:
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## device-prefix-*:
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# Card- and device-specific prefixes may be used to override the device-prefix
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# option. The option may specify the card id (eg, device-prefix-NVidia), or
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# the card id and device index (eg, device-prefix-NVidia-0). The card id is
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# case-sensitive.
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#device-prefix- =
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## capture:
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# Sets the device name for the default capture device.
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#capture = default
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## capture-prefix:
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# Sets the prefix used by the discovered (non-default) capture devices. This
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# will be appended with "CARD=c,DEV=d", where c is the card number and d is
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# the device number for the requested device name.
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# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
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# device number for the requested device name.
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#capture-prefix = plughw:
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## capture-prefix-*:
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# Card- and device-specific prefixes may be used to override the
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# capture-prefix option. The option may specify the card id (eg,
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# capture-prefix-NVidia), or the card id and device index (eg,
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# capture-prefix-NVidia-0). The card id is case-sensitive.
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#capture-prefix- =
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## mmap:
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# Sets whether to try using mmap mode (helps reduce latencies and CPU
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# consumption). If mmap isn't available, it will automatically fall back to
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@@ -238,6 +304,19 @@
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# Sets the device name for Solaris output.
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#device = /dev/audio
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##
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## QSA backend stuff
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##
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[qsa]
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## device:
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# Sets the device name for the default playback device.
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#device = default
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## capture:
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# Sets the device name for the default capture device.
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#capture = default
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##
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## MMDevApi backend stuff
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##
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@@ -268,19 +347,6 @@
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# given by PortAudio itself.
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#capture = -1
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##
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## PulseAudio backend stuff
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##
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[pulse]
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## spawn-server:
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# Attempts to spawn a PulseAudio server when requesting to open a PulseAudio
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# device. Note that some apps may open and probe all enumerated devices on
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# startup, causing a server to spawn even if a PulseAudio device is not
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# actually selected. Setting autospawn to false in Pulse's client.conf will
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# still prevent autospawning even if this is set to true.
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#spawn-server = false
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##
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## Wave File Writer stuff
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##
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