Nabble - OpenAL - Dev

Re: EFX Echo effect on Linux

2009-11-19T13:28:23Z

On Thursday 19 November 2009 1:17:51 pm Pierre Proske wrote:
> How embarrassing : /
> I can't believe I didn't spot that. Sorry I'm not so familiar with CMake
> otherwise I would have solved that myself...(CMakeCache I mean)
> It works!

Not a problem. Glad you got it working. :)
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Re: EFX Echo effect on Linux

2009-11-19T13:17:51Z

How embarrassing : /
I can't believe I didn't spot that. Sorry I'm not so familiar with CMake
otherwise I would have solved that myself...(CMakeCache I mean)
It works!

Chris Robinson wrote:

> On Thursday 19 November 2009 12:43:19 pm Pierre Proske wrote:
>
>> Here's my .alsoftrc file...
>> My apologies, I actually hadn't compiled pulseaudio support into the
>> library.
>> However I just did that right now and I get the following when running
>> openal-info:
>> "ALC error Invalid Context occured"
>>
>> My cmake output is the following:
>> -- Building OpenAL with support for the following backends:
>> -- ALSA, OSS, PortAudio, WaveFile
>>
>> I'm using libpulse-dev from the Ubuntu repos, and it is version
>> 1:0.9.19-0ubuntu4
>>
>
> PulseAudio still didn't get enabled (don't confuse it with *Port*Audio,
> there). You may need to delete CMakeCache.txt before it'll notice the Pulse
> headers and lib. If it still fails to find Pulse after that, can you attach
> CMakeFiles/CMakeError.log?
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>

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Re: EFX Echo effect on Linux

2009-11-19T13:02:25Z

On Thursday 19 November 2009 12:43:19 pm Pierre Proske wrote:

> Here's my .alsoftrc file...
> My apologies, I actually hadn't compiled pulseaudio support into the
> library.
> However I just did that right now and I get the following when running
> openal-info:
> "ALC error Invalid Context occured"
>
> My cmake output is the following:
> -- Building OpenAL with support for the following backends:
> -- ALSA, OSS, PortAudio, WaveFile
>
> I'm using libpulse-dev from the Ubuntu repos, and it is version
> 1:0.9.19-0ubuntu4

PulseAudio still didn't get enabled (don't confuse it with *Port*Audio,
there). You may need to delete CMakeCache.txt before it'll notice the Pulse
headers and lib. If it still fails to find Pulse after that, can you attach
CMakeFiles/CMakeError.log?
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Re: EFX Echo effect on Linux

2009-11-19T12:43:19Z

Here's my .alsoftrc file...
My apologies, I actually hadn't compiled pulseaudio support into the
library.
However I just did that right now and I get the following when running
openal-info:
"ALC error Invalid Context occured"

My cmake output is the following:
-- Building OpenAL with support for the following backends:
-- ALSA, OSS, PortAudio, WaveFile

I'm using libpulse-dev from the Ubuntu repos, and it is version
1:0.9.19-0ubuntu4

Chris Robinson wrote:

> On Thursday 19 November 2009 9:12:32 am you wrote:
>
>> Ok I had a closer look at this and it seems that it fails to open the
>> default device, and
>> alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER) returns nothing.
>>
>> Perhaps it has something to do with my system sound configuration, but
>> I'm successfully using pulseaudio for music playback using Rhythmbox so
>> I'm not sure what to do.
>> I'd be happy to test some things if you have any suggestions.
>>
>
> What version of PulseAudio are you using, and are you sure support for it was
> enabled in OpenAL Soft? What does your ~/.alsoftrc look like? OpenAL Soft
> should say something if it fails to load the proper functions from libpulse..
> the only way it won't say anything is if libpulse itself fails to load, or
> pulseaudio was disabled (at compile time or via .alsoftrc).
>
>

# OpenAL config file. Options that are not under a block or are under the
# [general] block are for general, non-backend-specific options. Blocks may
# appear multiple times, and duplicated options will take the last value
# specified.
# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
# specific override settings in ~/.alsoftrc.
# For Windows, these settings should go into %AppData%\alsoft.ini
# The environment variable ALSOFT_CONF can be used to specify another config
# override

# Option and block names are case-insenstive. The supplied values are only
# hints and may not be honored (though generally it'll try to get as close as
# possible). Note: options that are left unset may default to app- or system-
# specified values. These are the current available settings:

## format:
# Sets the output format. Can be one of:
# AL_FORMAT_MONO8 (8-bit mono)
# AL_FORMAT_STEREO8 (8-bit stereo)
# AL_FORMAT_QUAD8 (8-bit 4-channel)
# AL_FORMAT_51CHN8 (8-bit 5.1 output)
# AL_FORMAT_61CHN8 (8-bit 6.1 output)
# AL_FORMAT_71CHN8 (8-bit 7.1 output)
# AL_FORMAT_MONO16 (16-bit mono)
# AL_FORMAT_STEREO16 (16-bit stereo)
# AL_FORMAT_QUAD16 (16-bit 4-channel)
# AL_FORMAT_51CHN16 (16-bit 5.1 output)
# AL_FORMAT_61CHN16 (16-bit 6.1 output)
# AL_FORMAT_71CHN16 (16-bit 7.1 output)
# AL_FORMAT_MONO32 (32-bit float mono)
# AL_FORMAT_STEREO32 (32-bit float stereo)
# AL_FORMAT_QUAD32 (32-bit float 4-channel)
# AL_FORMAT_51CHN32 (32-bit float 5.1 output)
# AL_FORMAT_61CHN32 (32-bit float 6.1 output)
# AL_FORMAT_71CHN32 (32-bit float 7.1 output)
#format = AL_FORMAT_STEREO16

## cf_level:
# Sets the crossfeed level for stereo output. Valid values are:
# 0 - No crossfeed
# 1 - Low crossfeed
# 2 - Middle crossfeed
# 3 - High crossfeed (virtual speakers are closer to itself)
# 4 - Low easy crossfeed
# 5 - Middle easy crossfeed
# 6 - High easy crossfeed
# Users of headphones may want to try various settings. Has no effect on non-
# stereo modes.
#cf_level = 0

## frequency:
# Sets the output frequency.
#frequency = 44100

## period_size:
# Sets the update period size, in frames. This is the number of frames needed
# for each mixing update. If the deprecated 'refresh' option is specified and
# this isn't, the value will be calculated as size = refresh/periods.
period_size = 256

## periods:
# Sets the number of update periods. Higher values create a larger mix ahead,
# which helps protect against skips when the CPU is under load, but increases
# the delay between a sound getting mixed and being heard.
#periods = 4

## sources:
# Sets the maximum number of allocatable sources. Lower values may help for
# systems with apps that try to play more sounds than the CPU can handle.
#sources = 256

## stereodup:
# Sets whether to duplicate stereo sounds on the rear and side speakers for 4+
# channel output. This can make stereo sources substantially louder than mono
# or even 4+ channel sources, but provides a "fuller" playback quality. True,
# yes, on, and non-0 values will duplicate stereo sources. 0 and anything else
# will cause stereo sounds to only play out the front speakers.
#stereodup = false

## drivers:
# Sets the backend driver list order, comma-seperated. Unknown backends and
# duplicated names are ignored, and unlisted backends won't be considered for
# use. An empty list means the default.
#drivers = alsa,oss,solaris,dsound,winmm,port,pulse,wave
drivers = pulse

## excludefx:
# Sets which effects to exclude, preventing apps from using them. This can
# help for apps that try to use effects which are too CPU intensive for the
# system to handle. Available effects are: eaxreverb,reverb,echo
#excludefx =

## slots:
# Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
# can use a non-negligible amount of CPU time if an effect is set on it even
# if no sources are feeding it, so this may help when apps use more than the
# system can handle.
#slots = 4

## sends:
# Sets the maximum number of auxiliary sends per source. The total number of
# sends possible is defined at compile time and thus can not be increased
# beyond the default (2).
#sends = 2

## layout_STEREO:
# Sets the speaker layout when using stereo output. Values are specified in
# degrees, where 0 is straight in front, negative goes left, and positive goes
# right. The values must define a circular pattern, starting with the back-
# left at the most negative, around the front to back-center. Unspecified
# speakers will remain at their default position. Available speakers are
# front-left(fl) and front-right(fr).
#layout_STEREO = fl=-90, fr=90

## laytout_QUAD:
# Sets the speaker layout when using quadriphonic output. Available speakers
# are back-left(bl), front-left(fl), front-right(fr), and back-right(br).
#layout_QUAD = bl=-135, fl=-45, fr=45, br=135

## layout_51CHN:
# Sets the speaker layout when using 5.1 output. Available speakers are back-
# left(bl), front-left(fl), front-center(fc), front-right(fr), and back-
# right(br).
#layout_51CHN = bl=-110, fl=-30, fc=0, fr=30, br=110

## layout_61CHN:
# Sets the speaker layout when using 6.1 output. Available speakers are side-
# left(sl), front-left(fl), front-center(fc), front-right(fr), side-right(sr),
# and back-center(bc).
#layout_61CHN = sl=-90, fl=-30, fc=0, fr=30, sr=90, bc=180

## layout_71CHN:
# Sets the speaker layout when using 7.1 output. Available speakers are back-
# left(bl), side-left(sl), front-left(fl), front-center(fc), front-right(fr),
# side-right(sr), and back-right(br).
#layout_71CHN = bl=-150, sl=-90, fl=-30, fc=0, fr=30, sr=90, br=150

##
## ALSA backend stuff
##
[alsa]

## device:
# Sets the device name for the default playback device.
#device = default

## capture:
# Sets the device name for the default capture device.
#capture = default

## mmap:
# Sets whether to try using mmap mode (helps reduce latencies and CPU
# consumption). If mmap isn't available, it will automatically fall back to
# non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
# and anything else will force mmap off.
#mmap = true

##
## OSS backend stuff
##
[oss]

## device:
# Sets the device name for OSS output.
#device = /dev/dsp

## capture:
# Sets the device name for OSS capture.
#capture = /dev/dsp

##
## Solaris backend stuff
##
[solaris]

## device:
# Sets the device name for Solaris output.
#device = /dev/audio

##
## DirectSound backend stuff
##
[dsound]

##
## Windows Multimedia backend stuff
##
[winmm]

##
## PortAudio backend stuff
##
[port]

## device:
# Sets the device index for output. Negative values will use the default as
# given by PortAudio itself.
#device = -1

##
## PulseAudio backend stuff
##
[pulse]

##
## Wave File Writer stuff
##
[wave]

## file:
# Sets the filename of the wave file to write to. An empty name prevents the
# backend from opening, even when explicitly requested.
# THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
#file =

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Re: EFX Echo effect on Linux

2009-11-19T10:12:19Z

On Thursday 19 November 2009 9:12:32 am you wrote:
> Ok I had a closer look at this and it seems that it fails to open the
> default device, and
> alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER) returns nothing.
>
> Perhaps it has something to do with my system sound configuration, but
> I'm successfully using pulseaudio for music playback using Rhythmbox so
> I'm not sure what to do.
> I'd be happy to test some things if you have any suggestions.

What version of PulseAudio are you using, and are you sure support for it was
enabled in OpenAL Soft? What does your ~/.alsoftrc look like? OpenAL Soft
should say something if it fails to load the proper functions from libpulse..
the only way it won't say anything is if libpulse itself fails to load, or
pulseaudio was disabled (at compile time or via .alsoftrc).
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Re: EFX Echo effect on Linux

2009-11-19T05:51:34Z

Thanks! Now have an echo effect :)

Regarding pulseaudio, when I enable it, I get no sound and all OpenAL
call return AL_INVALID_OPERATION it seems....sorry that's all the info I
can give for the moment.
Keep up the good work!

Chris Robinson wrote:

> On Wednesday 18 November 2009 9:29:15 am Pierre Proske wrote:
>
>> Hi,
>>
>> Currently I'm using OpenAL soft and firstly thanks for the great work.
>> I've got things up and running and managed to get the EFX reverb
>> working, but I tried the Echo and it crashed my application.
>>
>
> Hi, thanks. :)
>
> Odd for the crash, though. Can you paste the code that's causing the problem,
> and give a backtrace?
>
>
>> I did a quick search and came across this written by Chris Robinson on
>>
>> this list:
>>
>>> The current alext.h doesn't contain the EFX defines and types. However,
>>> you can get Creative's OpenAL SDK which includes a header that defines
>>> them.
>>>
>>> To note, though, the Echo effect isn't available without hardware drivers
>>> (or through accelerated DSound3D). The standard reverb effect and lowpass
>>> filter should always be available with EFX, though, even in software.
>>>
>> What's the status of the Echo effect on openal soft? It appears to be
>> implemented, and openal-info reports this:
>>
>> Max auxiliary sends: 2
>> Available filters:
>> Low-pass
>> Available effects:
>> EAX Reverb
>> Reverb
>> Echo
>>
>
> Yeah, I said that before the echo effect was implemented. It should fully work
> now in 1.8 and newer.
>
>
>> Additionally I can't seem to get the pulseaudio backend to work. It
>> seemed to work in the last version of OpenAL soft, I think I tried it
>> briefly. I'm on Ubuntu 9.10.
>> Any suggestions that I can try to fix this? Although I guess I won't be
>> using it but rather ALSA anyway.
>>
>
> Weird.. OpenAL Soft 1.9 had problems with the PulseAudio backend that
> prevented it from working, while 1.10 should have fixed it (1.8 would work
> with older versions of PulseAudio, but would lockup on newer versions). What
> happens with 1.10?
>
>
>> One more thing, the openal-info reports the following with the portaudio
>> backend:
>>
>> Available playback devices:
>> PortAudio Software
>> Available capture devices:
>> Segmentation fault
>>
>> Is this normal? PortAudio seems to work otherwise, although I haven't
>> tried recording anything yet.
>>
>
> The crash isn't normal, no. What seems to happen is that the alcGetString
> query for the capture device list returns NULL, since there's no capture
> devices (PortAudio doesn't provide one, and no other backends were available).
> The spec says querying for a default device will return NULL if there aren't
> any devices, but doesn't say what the device list will be in that case (NULL?
> an empty string, or two null chars?). So I don't know if OpenAL Soft is wrong
> for returning a NULL list, or if the info program is wrong for not checking
> the returned pointer.
> _______________________________________________
> Openal-devel mailing list
> Openal-devel@...
> http://opensource.creative.com/mailman/listinfo/openal-devel
>
>

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Re: EFX Echo effect on Linux

2009-11-18T19:35:04Z

Greetings,

> Ugh. It's always the smallest bugs that annoy me most when they slip by. :P I

> really need to set up some kind of automated test suite to watch for these

> kinds of breaks, which I aim to at least start on before 1.11.

>

> Thanks much for the fix. It's applied in GIT/SVN now.

Not a problem. There are a few more (less critical) ones coming along with the reverb update. I hope to have it all ready later tonight or early tomorrow.

Regards,
- Christopher

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Re: EFX Echo effect on Linux

2009-11-18T17:50:35Z

On Wednesday 18 November 2009 5:21:50 pm Christopher Fitzgerald wrote:

> I actually have a patch for this bug coming with the latest reverb update.
> It seems when converting to the new effects model there was a line dropped
> from EchoDeviceUpdate() in alcEcho.c.
>
> I can't attach the patch at the moment (it also has some other changes),
> but here's the offending code and the fix so Chris can add a patch right
> away.
>
> In EchoDeviceUpdate() the sample buffer is allocated but the new buffer is
> not assigned to state->SampleBuffer. Here's the .diff-like change:
>
> alSetError(AL_OUT_OF_MEMORY);
> return AL_FALSE;
> }
> + state->SampleBuffer = temp;
> state->BufferLength = maxlen;
> }
> for(i = 0;i < state->BufferLength;i++)

Ugh. It's always the smallest bugs that annoy me most when they slip by. :P I
really need to set up some kind of automated test suite to watch for these
kinds of breaks, which I aim to at least start on before 1.11.

Thanks much for the fix. It's applied in GIT/SVN now.
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Re: EFX Echo effect on Linux

2009-11-18T17:21:50Z

Greetings,

> Hi, thanks. :)

>

> Odd for the crash, though. Can you paste the code that's causing the problem,

> and give a backtrace?

I actually have a patch for this bug coming with the latest reverb update. It seems when converting to the new effects model there was a line dropped from EchoDeviceUpdate() in alcEcho.c.

I can't attach the patch at the moment (it also has some other changes), but here's the offending code and the fix so Chris can add a patch right away.

In EchoDeviceUpdate() the sample buffer is allocated but the new buffer is not assigned to state->SampleBuffer. Here's the .diff-like change:

       alSetError(AL_OUT_OF_MEMORY);
       return AL_FALSE;
   }
+    state->SampleBuffer = temp;
   state->BufferLength = maxlen;
}
for(i = 0;i < state->BufferLength;i++)

Regards,
- Christopher

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Re: EFX Echo effect on Linux

2009-11-18T12:30:45Z

On Wednesday 18 November 2009 9:29:15 am Pierre Proske wrote:
> Hi,
>
> Currently I'm using OpenAL soft and firstly thanks for the great work.
> I've got things up and running and managed to get the EFX reverb
> working, but I tried the Echo and it crashed my application.

Hi, thanks. :)

Odd for the crash, though. Can you paste the code that's causing the problem,
and give a backtrace?

> I did a quick search and came across this written by Chris Robinson on
>
> this list:
> > The current alext.h doesn't contain the EFX defines and types. However,
> > you can get Creative's OpenAL SDK which includes a header that defines
> > them.
> >
> > To note, though, the Echo effect isn't available without hardware drivers
> > (or through accelerated DSound3D). The standard reverb effect and lowpass
> > filter should always be available with EFX, though, even in software.
>
> What's the status of the Echo effect on openal soft? It appears to be
> implemented, and openal-info reports this:
>
> Max auxiliary sends: 2
> Available filters:
> Low-pass
> Available effects:
> EAX Reverb
> Reverb
> Echo

Yeah, I said that before the echo effect was implemented. It should fully work
now in 1.8 and newer.

> Additionally I can't seem to get the pulseaudio backend to work. It
> seemed to work in the last version of OpenAL soft, I think I tried it
> briefly. I'm on Ubuntu 9.10.
> Any suggestions that I can try to fix this? Although I guess I won't be
> using it but rather ALSA anyway.

Weird.. OpenAL Soft 1.9 had problems with the PulseAudio backend that
prevented it from working, while 1.10 should have fixed it (1.8 would work
with older versions of PulseAudio, but would lockup on newer versions). What
happens with 1.10?

> One more thing, the openal-info reports the following with the portaudio
> backend:
>
> Available playback devices:
> PortAudio Software
> Available capture devices:
> Segmentation fault
>
> Is this normal? PortAudio seems to work otherwise, although I haven't
> tried recording anything yet.

The crash isn't normal, no. What seems to happen is that the alcGetString
query for the capture device list returns NULL, since there's no capture
devices (PortAudio doesn't provide one, and no other backends were available).
The spec says querying for a default device will return NULL if there aren't
any devices, but doesn't say what the device list will be in that case (NULL?
an empty string, or two null chars?). So I don't know if OpenAL Soft is wrong
for returning a NULL list, or if the info program is wrong for not checking
the returned pointer.
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EFX Echo effect on Linux

2009-11-18T09:50:55Z

Hi,

Currently I'm using OpenAL soft and firstly thanks for the great work.
I've got things up and running and managed to get the EFX reverb
working, but I tried the Echo and it crashed my application.
I did a quick search and came across this written by Chris Robinson on
this list:
> The current alext.h doesn't contain the EFX defines and types. However, you
> can get Creative's OpenAL SDK which includes a header that defines them.
>
> To note, though, the Echo effect isn't available without hardware drivers (or
> through accelerated DSound3D). The standard reverb effect and lowpass filter
> should always be available with EFX, though, even in software.
>
What's the status of the Echo effect on openal soft? It appears to be
implemented, and openal-info reports this:

Max auxiliary sends: 2
Available filters:
Low-pass
Available effects:
EAX Reverb
Reverb
Echo

I'm using version: 1.10.622

Additionally I can't seem to get the pulseaudio backend to work. It
seemed to work in the last version of OpenAL soft, I think I tried it
briefly. I'm on Ubuntu 9.10.
Any suggestions that I can try to fix this? Although I guess I won't be
using it but rather ALSA anyway.

One more thing, the openal-info reports the following with the portaudio
backend:

Available playback devices:
PortAudio Software
Available capture devices:
Segmentation fault

Is this normal? PortAudio seems to work otherwise, although I haven't
tried recording anything yet.

thanks in advance,
Pierre

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EFX Echo effect on Linux

2009-11-18T09:29:15Z

Hi,

Currently I'm using OpenAL soft and firstly thanks for the great work.
I've got things up and running and managed to get the EFX reverb
working, but I tried the Echo and it crashed my application.
I did a quick search and came across this written by Chris Robinson on
this list:
> The current alext.h doesn't contain the EFX defines and types. However, you
> can get Creative's OpenAL SDK which includes a header that defines them.
>
> To note, though, the Echo effect isn't available without hardware drivers (or
> through accelerated DSound3D). The standard reverb effect and lowpass filter
> should always be available with EFX, though, even in software.
>
What's the status of the Echo effect on openal soft? It appears to be
implemented, and openal-info reports this:

Max auxiliary sends: 2
Available filters:
Low-pass
Available effects:
EAX Reverb
Reverb
Echo

I'm using version: 1.10.622

Additionally I can't seem to get the pulseaudio backend to work. It
seemed to work in the last version of OpenAL soft, I think I tried it
briefly. I'm on Ubuntu 9.10.
Any suggestions that I can try to fix this? Although I guess I won't be
using it but rather ALSA anyway.

One more thing, the openal-info reports the following with the portaudio
backend:

Available playback devices:
PortAudio Software
Available capture devices:
Segmentation fault

Is this normal? PortAudio seems to work otherwise, although I haven't
tried recording anything yet.

thanks in advance,
Pierre
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Re: Pulse capture (was Re: OpenAL Soft 1.10.622 isreleased!)

2009-11-10T16:56:21Z

The gstreamer pulseaudio src "plugin" captures audio fine. So there
must be some setup nuance required. I'll have a look at the pulsesrc
gstreamer piece and see how it does it.

Mike

On Tue, 2009-11-10 at 18:16 -0600, Mike Dickson wrote:

> Looking for one. I'll check that next. I think gstreamer has a pulse
> src module.
>
> So taking a seperate tack, is it possible to use the config file to use
> alsa for capture and pulseaudio for playback? How would that be set up?
> The pulseaudio alsa capture module does seem to work...
>
> Mike
>
> On Tue, 2009-11-10 at 16:05 -0800, Chris Robinson wrote:
> > On Tuesday 10 November 2009 12:49:33 pm Mike Dickson wrote:
> > > Using the test capture client source I've tracked this down to a hang in
> > > opening the capture device. The client opens the output device find and
> > > then opens the capture device. It hangs in pulse_open at line 391 at the
> > > call to ppa_threaded_mainloop_wait:
> > >
> > ...
> > >
> > > I'm a little confused by one thing.. Doesn't the call to set the
> > > state_callback need to go before the polling loop? I'm hesitant to
> > > start tinkering in here since this code is shared with the code that
> > > opens the output device and that *is* working fine..
> >
> > There's a call to ppa_context_set_state_callback before it attempts a
> > connection. The one that happens afterward is to set a different callback
> > which watches for context failures (eg. from the server being killed).
> >
> > If it hangs on ppa_threaded_mainloop_wait, that likely means the context isn't
> > fully connecting or failing, for some reason. Does pulseaudio recording work
> > with other apps?
> > _______________________________________________
> > Openal-devel mailing list
> > Openal-devel@...
> > http://opensource.creative.com/mailman/listinfo/openal-devel
>
>
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Re: Pulse capture

2009-11-10T16:37:21Z

On Tuesday 10 November 2009 4:16:45 pm Mike Dickson wrote:
> Looking for one. I'll check that next. I think gstreamer has a pulse
> src module.
>
> So taking a seperate tack, is it possible to use the config file to use
> alsa for capture and pulseaudio for playback? How would that be set up?
> The pulseaudio alsa capture module does seem to work...

There isn't a way to default to alsa for capture and pulseaudio for playback.
But you can configure alsa to use any capture device you want, by setting:
[alsa]
capture = <devicename>
in ~/.alsoftrc. You can then configure the app to use "ALSA Capture on
default".
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Re: Pulse capture (was Re: OpenAL Soft 1.10.622 isreleased!)

2009-11-10T16:16:45Z

Looking for one. I'll check that next. I think gstreamer has a pulse
src module.

So taking a seperate tack, is it possible to use the config file to use
alsa for capture and pulseaudio for playback? How would that be set up?
The pulseaudio alsa capture module does seem to work...

Mike

On Tue, 2009-11-10 at 16:05 -0800, Chris Robinson wrote:

> On Tuesday 10 November 2009 12:49:33 pm Mike Dickson wrote:
> > Using the test capture client source I've tracked this down to a hang in
> > opening the capture device. The client opens the output device find and
> > then opens the capture device. It hangs in pulse_open at line 391 at the
> > call to ppa_threaded_mainloop_wait:
> >
> ...
> >
> > I'm a little confused by one thing.. Doesn't the call to set the
> > state_callback need to go before the polling loop? I'm hesitant to
> > start tinkering in here since this code is shared with the code that
> > opens the output device and that *is* working fine..
>
> There's a call to ppa_context_set_state_callback before it attempts a
> connection. The one that happens afterward is to set a different callback
> which watches for context failures (eg. from the server being killed).
>
> If it hangs on ppa_threaded_mainloop_wait, that likely means the context isn't
> fully connecting or failing, for some reason. Does pulseaudio recording work
> with other apps?
> _______________________________________________
> Openal-devel mailing list
> Openal-devel@...
> http://opensource.creative.com/mailman/listinfo/openal-devel

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Re: Pulse capture (was Re: OpenAL Soft 1.10.622 isreleased!)

2009-11-10T16:05:51Z

On Tuesday 10 November 2009 12:49:33 pm Mike Dickson wrote:
> Using the test capture client source I've tracked this down to a hang in
> opening the capture device. The client opens the output device find and
> then opens the capture device. It hangs in pulse_open at line 391 at the
> call to ppa_threaded_mainloop_wait:
>
...
>
> I'm a little confused by one thing.. Doesn't the call to set the
> state_callback need to go before the polling loop? I'm hesitant to
> start tinkering in here since this code is shared with the code that
> opens the output device and that *is* working fine..

There's a call to ppa_context_set_state_callback before it attempts a
connection. The one that happens afterward is to set a different callback
which watches for context failures (eg. from the server being killed).

If it hangs on ppa_threaded_mainloop_wait, that likely means the context isn't
fully connecting or failing, for some reason. Does pulseaudio recording work
with other apps?
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Re: OpenAL SDK header compatibility

2009-11-10T15:54:57Z

On Tuesday 10 November 2009 7:42:41 am Daniel PEACOCK wrote:
> Hi Chris,
>
> I've made a note here. Next time we refresh the SDK we'll roll this
> change in.

Cool, thanks. :)
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Re: Pulse capture (was Re: OpenAL Soft 1.10.622 isreleased!)

2009-11-10T12:49:33Z

Using the test capture client source I've tracked this down to a hang in
opening the capture device. The client opens the output device find and
then opens the capture device. It hangs in pulse_open at line 391 at the
call to ppa_threaded_mainloop_wait:

while((state=ppa_context_get_state(data->context)) != PA_CONTEXT_READY)
{
if(!PA_CONTEXT_IS_GOOD(state))
{
AL_PRINT("Context did not get ready: %s\n",
ppa_strerror(ppa_context_errno(data->context)));

ppa_context_unref(data->context);
data->context = NULL;

ppa_threaded_mainloop_unlock(data->loop);
goto out;
}

ppa_threaded_mainloop_wait(data->loop);
ppa_threaded_mainloop_accept(data->loop);
}
ppa_context_set_state_callback(data->context, context_state_callback2, device);

I'm a little confused by one thing.. Doesn't the call to set the
state_callback need to go before the polling loop? I'm hesitant to
start tinkering in here since this code is shared with the code that
opens the output device and that *is* working fine..

Thoughts?

Mike

On Mon, 2009-11-09 at 19:57 -0500, Jason Daly wrote:

> Chris Robinson wrote:
> > On Monday 09 November 2009 5:01:46 am Mike Dickson wrote:
> >
> > > Great release!
> > >
> > > The pulseaudio support for output is working great for the apps I've
> > > tried. Much better performance overall than using the ALSA emulation.
> > > I do still have problems with pulseaudio capture, specifically with
> > > Vivox/secondlife. I'd like to help with that if possible but I'm new to
> > > the OpenAL sources so I hope an occasional question to the list is fine.
> > >
> >
> > There shouldn't be any issue about asking here. Thanks for wanting to look
> > into it. :)
> >
>
> No problem at all. That's one reason why the openal-devel list
> exists :-)
>
> --"J"
>
>
> _______________________________________________
> Openal-devel mailing list
> Openal-devel@...
> http://opensource.creative.com/mailman/listinfo/openal-devel

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Re: Pulse capture (was Re: OpenAL Soft 1.10.622isreleased!)

2009-11-10T09:03:42Z

Ok, at least with a quick run I've verified that pulse capture isn't
working. If I use the ALSA driver (which will use the ALSA/pulse
adapter) capture works (albeit time compressed, I sound like Mickey
Mouse). With native pulse capture no playback and it loops trying to
play back endlessly.

I'll start with this and then move on to the Vivox client. Thanks for
this, it will be a big help.

Mike

On Tue, 2009-11-10 at 11:08 -0500, Jason Daly wrote:

> Mike Dickson wrote:
> > Can someone point me to a simple capture client w/source that uses
> > OpenAL-Soft for the capture? Starting with Vivox SLVoice directly is
> > going to be a pain since it's binary only..
> >
>
> Here's a really simple test app that I wrote. It captures audio for
> three seconds, pauses, then captures three more seconds. At the end it
> plays back what was captured.
>
> A lot of the values are hard-coded, so you may need to tweak it to get
> what you're looking for.
>
> --"J"
>

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Re: Pulse capture (was Re: OpenAL Soft 1.10.622isreleased!)

2009-11-10T08:22:34Z

Thanks so much! This should get me pointed in the right direction.

Mike

On Tue, 2009-11-10 at 11:08 -0500, Jason Daly wrote:

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Re: Pulse capture (was Re: OpenAL Soft 1.10.622isreleased!)

2009-11-10T08:08:25Z

Mike Dickson wrote:
> Can someone point me to a simple capture client w/source that uses
> OpenAL-Soft for the capture? Starting with Vivox SLVoice directly is
> going to be a pain since it's binary only..
>

Here's a really simple test app that I wrote. It captures audio for
three seconds, pauses, then captures three more seconds. At the end it
plays back what was captured.

A lot of the values are hard-coded, so you may need to tweak it to get
what you're looking for.

--"J"

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <AL/al.h>
#include <AL/alc.h>
#include <sys/time.h>

int main(void)
{
const ALCchar * devices;
ALCchar * ptr;
ALCdevice * mainDev;
ALCcontext * mainContext;
ALCdevice * captureDev;
ALubyte captureBuffer[655360];
ALubyte *captureBufPtr;
ALint samplesAvailable;
ALint samplesCaptured;
time_t currentTime;
time_t lastTime;
ALuint buffer;
ALuint source;
ALint playState;
int i;

// Print the list of capture devices
printf("Available playback devices:\n");
devices = alcGetString(NULL, ALC_DEVICE_SPECIFIER);
ptr = devices;
while (ptr[0] != NULL)
{
printf(" %s\n", ptr);
ptr += strlen(ptr) + 1;
}

// Open a playback device and create a context first
printf("Opening playback device:\n");
mainDev = alcOpenDevice(NULL);
if (mainDev == NULL)
{
printf("Unable to open playback device!\n");
exit(1);
}
devices = alcGetString(mainDev, ALC_DEVICE_SPECIFIER);
printf(" opened device '%s'\n", devices);
mainContext = alcCreateContext(mainDev, NULL);
if (mainContext == NULL)
{
printf("Unable to create playback context!\n");
exit(1);
}
printf(" created playback context\n");

// Make the playback context current
alcMakeContextCurrent(mainContext);
alcProcessContext(mainContext);

// Print the list of capture devices
printf("Available capture devices:\n");
devices = alcGetString(NULL, ALC_CAPTURE_DEVICE_SPECIFIER);
ptr = devices;
while (ptr[0] != NULL)
{
printf(" %s\n", ptr);
ptr += strlen(ptr) + 1;
}

// Open the default device
printf("Opening capture device:\n");
captureDev = alcCaptureOpenDevice(NULL, 8000, AL_FORMAT_MONO16, 800);
if (captureDev == NULL)
{
printf(" Unable to open device!\n");
exit(1);
}
devices = alcGetString(captureDev, ALC_CAPTURE_DEVICE_SPECIFIER);
printf(" opened device %s\n", devices);

// Wait for three seconds to prompt the user
for (i = 3; i > 0; i--)
{
printf("Starting capture in %d...\r", i);
fflush(stdout);
lastTime = time(NULL);
currentTime = lastTime;
while (currentTime == lastTime)
{
currentTime = time(NULL);
usleep(100000);
}
}

printf("Starting capture NOW!\n");
fflush(stdout);
lastTime = currentTime;

// Capture (roughly) five seconds of audio
alcCaptureStart(captureDev);
samplesCaptured = 0;
captureBufPtr = captureBuffer;
while (currentTime < (lastTime + 5))
{
// Get the number of samples available
alcGetIntegerv(captureDev, ALC_CAPTURE_SAMPLES, 1, &samplesAvailable);

// Copy the samples to our capture buffer
if (samplesAvailable > 0)
{
alcCaptureSamples(captureDev, captureBufPtr, samplesAvailable);
samplesCaptured += samplesAvailable;
printf("Captured %d samples (adding %d)\r", samplesCaptured,
samplesAvailable);
fflush(stdout);

// Advance the buffer (two bytes per sample * number of samples)
captureBufPtr += samplesAvailable * 2;
}

// Wait for a bit
usleep(10000);

// Update the clock
currentTime = time(NULL);
}
printf("\nPausing capture.\n");
alcCaptureStop(captureDev);

// Wait for three seconds to prompt the user
for (i = 3; i > 0; i--)
{
printf("Resuming capture in %d...\r", i);
fflush(stdout);
lastTime = time(NULL);
currentTime = lastTime;
while (currentTime == lastTime)
{
currentTime = time(NULL);
usleep(100000);
}
}

printf("Resuming capture NOW!\n");
fflush(stdout);
lastTime = currentTime;

// Capture (roughly) five seconds of audio
alcCaptureStart(captureDev);
while (currentTime < (lastTime + 5))
{
// Get the number of samples available
alcGetIntegerv(captureDev, ALC_CAPTURE_SAMPLES, 1, &samplesAvailable);

// Copy the samples to our capture buffer
if (samplesAvailable > 0)
{
alcCaptureSamples(captureDev, captureBufPtr, samplesAvailable);
samplesCaptured += samplesAvailable;
printf("Captured %d samples (adding %d)\r", samplesCaptured,
samplesAvailable);
fflush(stdout);

// Advance the buffer (two bytes per sample * number of samples)
captureBufPtr += samplesAvailable * 2;
}

// Wait for a bit
usleep(10000);

// Update the clock
currentTime = time(NULL);
}

printf("\nDone capturing.\n");
alcCaptureStop(captureDev);

// Play back the captured data
printf("Starting playback...\n");
fflush(stdout);

// Generate an OpenAL buffer for the captured data
alGenBuffers(1, &buffer);
alGenSources(1, &source);
alBufferData(buffer, AL_FORMAT_MONO16, captureBuffer,
samplesCaptured*2, 22050);
alSourcei(source, AL_BUFFER, buffer);
alSourcePlay(source);

// Wait for the source to stop playing
playState = AL_PLAYING;
while (playState == AL_PLAYING)
{
printf(" source %d is playing...\r", source);
fflush(stdout);
alGetSourcei(source, AL_SOURCE_STATE, &playState);
usleep(100000);
}
printf("\nDone with playback.\n");
fflush(stdout);

// Shut down OpenAL
alDeleteSources(1, &source);
alDeleteBuffers(1, &buffer);
alcMakeContextCurrent(NULL);
alcCloseDevice(mainDev);
alcCaptureCloseDevice(captureDev);
}

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Re: Pulse capture (was Re: OpenAL Soft 1.10.622 isreleased!)

2009-11-10T07:52:14Z

Can someone point me to a simple capture client w/source that uses
OpenAL-Soft for the capture? Starting with Vivox SLVoice directly is
going to be a pain since it's binary only..

Mike

On Mon, 2009-11-09 at 19:57 -0500, Jason Daly wrote:

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Re: OpenAL SDK header compatibility

2009-11-10T07:42:41Z

Hi Chris,

I've made a note here. Next time we refresh the SDK we'll roll this
change in.

Dan
Creative Labs (UK) Ltd.
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unlawful. If you have received this message in error, please delete it
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at 79 Knightsbridge, London SW1X 7RB

openal-devel-bounces@... wrote on 11/10/2009 01:13:31
PM:

> Just something of note, in case there's going to be an SDK update.
> The headers
> efx.h and xram.h declare their function pointer types using __cdecl, as
> opposed to the more appropriate AL_APIENTRY. Under Windows these arethe
same
> thing, but that's not guaranteed for other OSs (non-Windows systems use
the
> system/compiler default calling convention). Additionally, the
> __cdecl keyword
> doesn't tend to be valid for non-Windows compilers.
>
> Given that OpenAL Soft supports EFX for Linux/Unix systems, it would be
very
> nice if these could be changed so developers can more easily use the
headers
> in cross-platform code.
>
> Thanks in advance. :)
> - Chris
> _______________________________________________
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> ForwardSourceID:NT00076ADE

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OpenAL SDK header compatibility

2009-11-10T05:13:31Z

Just something of note, in case there's going to be an SDK update. The headers
efx.h and xram.h declare their function pointer types using __cdecl, as
opposed to the more appropriate AL_APIENTRY. Under Windows these are the same
thing, but that's not guaranteed for other OSs (non-Windows systems use the
system/compiler default calling convention). Additionally, the __cdecl keyword
doesn't tend to be valid for non-Windows compilers.

Given that OpenAL Soft supports EFX for Linux/Unix systems, it would be very
nice if these could be changed so developers can more easily use the headers
in cross-platform code.

Thanks in advance. :)
- Chris
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Re: Pulse capture (was Re: OpenAL Soft 1.10.622 isreleased!)

2009-11-09T16:57:49Z

Chris Robinson wrote:

On Monday 09 November 2009 5:01:46 am Mike Dickson wrote:

Great release!

The pulseaudio support for output is working great for the apps I've
tried.  Much better performance overall than using the ALSA emulation.
I do still have problems with pulseaudio capture, specifically with
Vivox/secondlife.  I'd like to help with that if possible but I'm new to
the OpenAL sources so I hope an occasional question to the list is fine.


There shouldn't be any issue about asking here. Thanks for wanting to look 
into it. :)

No problem at all. That's one reason why the openal-devel list exists :-)

--"J"

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Re: Pulse capture (was Re: OpenAL Soft 1.10.622 is released!)

2009-11-09T15:50:37Z

It's the vivox client I'm using with SecondLife at least to start.
Behavior is similar to the pulse output stuff before the large patchset.
It looks like it gets connected but no data. I've only just started to
mess with it, hopefully can dig deeper tonite. I'll follow up with
findings then.

Mike

On Mon, 2009-11-09 at 15:33 -0800, Chris Robinson wrote:

> On Monday 09 November 2009 5:01:46 am Mike Dickson wrote:
> > Great release!
> >
> > The pulseaudio support for output is working great for the apps I've
> > tried. Much better performance overall than using the ALSA emulation.
> > I do still have problems with pulseaudio capture, specifically with
> > Vivox/secondlife. I'd like to help with that if possible but I'm new to
> > the OpenAL sources so I hope an occasional question to the list is fine.
>
> There shouldn't be any issue about asking here. Thanks for wanting to look
> into it. :)
>
> What's the problem you're seeing with pulseaudio capture? I'm not sure if I
> have any working microphones laying around to test it myself.. need to look
> around more.
> _______________________________________________
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> Openal-devel@...
> http://opensource.creative.com/mailman/listinfo/openal-devel

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Pulse capture (was Re: OpenAL Soft 1.10.622 is released!)

2009-11-09T15:33:26Z

On Monday 09 November 2009 5:01:46 am Mike Dickson wrote:
> Great release!
>
> The pulseaudio support for output is working great for the apps I've
> tried. Much better performance overall than using the ALSA emulation.
> I do still have problems with pulseaudio capture, specifically with
> Vivox/secondlife. I'd like to help with that if possible but I'm new to
> the OpenAL sources so I hope an occasional question to the list is fine.

There shouldn't be any issue about asking here. Thanks for wanting to look
into it. :)

What's the problem you're seeing with pulseaudio capture? I'm not sure if I
have any working microphones laying around to test it myself.. need to look
around more.
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Re: OpenAL Soft 1.10.622 is released!

2009-11-09T05:01:46Z

Great release!

The pulseaudio support for output is working great for the apps I've
tried. Much better performance overall than using the ALSA emulation.
I do still have problems with pulseaudio capture, specifically with
Vivox/secondlife. I'd like to help with that if possible but I'm new to
the OpenAL sources so I hope an occasional question to the list is fine.

Mike

On Sun, 2009-11-08 at 23:46 -0800, Chris Robinson wrote:
> May we forget about 1.9...
>
> http://kcat.strangesoft.net/openal.html
>
> Hopefully, finally, for real, fixed the PulseAudio backend this time. Also
> fixed the OSS and Solaris backends. 7.1 output should work properly again, and
> fixed a problem/potential crash when setting an AL_EFFECT_NULL effect on an
> auxiliary slot.

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OpenAL Soft 1.10.622 is released!

2009-11-08T23:46:42Z

May we forget about 1.9...

http://kcat.strangesoft.net/openal.html

Hopefully, finally, for real, fixed the PulseAudio backend this time. Also
fixed the OSS and Solaris backends. 7.1 output should work properly again, and
fixed a problem/potential crash when setting an AL_EFFECT_NULL effect on an
auxiliary slot.

In terms of features, there's only a few. Backend libraries are only loaded
when they're needed now, so unused backends won't hold their shared libs in
memory, and devices provided by them are automatically added and removed as
the libs are installed and uninstalled. Additionally, multiple contexts per
device are now supported.

Given multiple contexts, it's now possible to change certain device parameters
on the fly. Creating a new context with certain attributes (eg. ALC_FREQUENCY,
and perhaps ALC_REFRESH in the future) will change the device parameters, and
consequently affects all contexts on the device.

Now I'll try to get back to taking care of those proposed extensions. It may
be possible to see alcSuspendContext/alcProcessContext properly supported, as
well.
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OpenAL Soft 1.10 will be sooner than expected

2009-10-29T15:48:40Z

It turns out some bad issues popped up in 1.9, and I'd like to get a fixed
version out relatively soon. However, one of the issues is with PulseAudio not
playing sound. This is something I experience myself with the latest GIT
version of OpenAL Soft, however I'm not very familiar with Pulse, I can't find
a cause from reading the code and the online docs, and asking around for help
in #pulseaudio on FreeNode IRC didn't yield anything helpful. I don't know if
it's a problem with the code, or if PulseAudio is just being PulseAudio (as I
also have issues getting paplay to output sound, too).

So, if anyone reading is familiar with coding PulseAudio, it would be very
helpful if you could check out the latest GIT/SVN version of OpenAL Soft and
have a look through PulseAudio's code (in Alc/pulseaudio.c). Also, if anyone
/uses/ PulseAudio, checking out the latest GIT/SVN version and just seeing if
it woks would be helpful too (please note the PulseAudio version you use, and
any abnormal settings).

The GIT repo is here:
git://repo.or.cz/openal-soft.git
And the SVN repo is on Creative's server:
svn://connect.creativelabs.com/OpenAL

Thanks in advance!
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Re: Downloading Mac Source

2009-10-28T16:05:22Z

I don't know if the problem was ever dealt with but this :

svn co svn://connect.creativelabs.com/OpenAL/tags/MacOSX1-2_Spec1-1

still results in this:

svn: Can't open 'C:\WINDOWS\TEMP\report.tmp': Access is denied.

on my macbook...

- David

Re: AL_EXT_source_distance_model revisit

2009-10-20T10:39:59Z

On Monday 19 October 2009 3:20:30 pm you wrote:
> You never know how a developer will decide to use a published API :-)
> So while 'normally' you probably wouldn't use these 2 mechanisms
> together, almost certainly someone will, so it's important to document
> the expected behavior.

Right. And in that case, alDistanceModel will reset every source's distance
model, whether one was explicitly set on it or not. It needs to reset the
model for generated sources to maintain backwards compatibility, and I don't
personally see a reason to special-case sources that had a distance model
explicitly set.

Though if people recommends otherwise, I won't complain too much. It's just an
extra flag per source that's set in alSourcei[v] with AL_DISTANCE_MODEL, and
checked in alDistanceModel.

Or perhaps another option I just thought of.. have an alEnable/alDisable
context state for per-source distance models. When disabled (default) the
context's model is used, otherwise the source's model is. Then alDistanceModel
has no explicit effect on source state, but the enable says which should be
used. Eg:

alSourcei(sID, AL_DISTANCE_MODEL, AL_LINEAR_DISTANCE_CLAMPED);
alDistanceModel(AL_NONE);
... AL_NONE is used for all sources ...

alEnable(AL_SOURCE_DISTANCE_MODEL);
... now linear distance clamped is used for sID, inverse distance clamped for
other sources ...

alDisable(AL_SOURCE_DISTANCE_MODEL);
... now AL_NONE is used for all sources ...

alGetSourceiv(sID, AL_DISTANCE_MODEL, &val);
... gives val == AL_LINEAR_DISTANCE_CLAMPED ...
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Re: Get setting of Sound Blaster X-Fi Elite Pro

2009-10-20T01:51:18Z

Hi Rosemma,

The APIs to set and get these properties have not been published I'm
afraid, so it is not possible for you to control them from your own
application.

Dan
Creative Labs (UK) Ltd.

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"GARIO Rosemma
(C.R.F.)"
<rosemma.gario@co To
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Subject
10/19/2009 04:31 [Openal-devel] Get setting of Sound
PM Blaster X-Fi Elite Pro

Dear All,
we have the sound blaster X-Fi Elite Pro installed on PC HP xw8600 with
Windows XP 64 SO.
We would get, with an our external program written in C++, the values
visualized by Creative Console that we launch selecting from menu
creative->SoundBlaster X-fi->Creative console utility. In particular, we
need to get the values managed by equalizer, and wave and MIDI synthesizer
gains. We tried with OpenAl library manual but we didn’t find anything
which describes such possibility. We don’t have a good know-how regarding
Open Al library…
Please Could you help us??
Are there any class methods / functions which could allow us to get such
values set by creative routine, and to overwrite them?

Best Regards
Rosemma Gario

p.s.
Sorry to my bad english

Gario Rosemma
Physical Ergonomics

Tel +39 011 9083890
e-mail: rosemma.gario@...

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Get setting of Sound Blaster X-Fi Elite Pro

2009-10-19T08:31:47Z

Dear All,

we have the sound blaster X-Fi Elite Pro installed on PC HP xw8600 with Windows XP 64 SO.

We would get, with an our external program written in C++, the values visualized by Creative Console that we launch selecting from menu creative->SoundBlaster X-fi->Creative console utility. In particular, we need to get the values managed by equalizer, and wave and MIDI synthesizer gains. We tried with OpenAl library manual but we didn’t find anything which describes such possibility. We don’t have a good know-how regarding Open Al library…

Please Could you help us??

Are there any class methods / functions which could allow us to get such values set by creative routine, and to overwrite them?

Best Regards

Rosemma Gario

p.s.

Sorry to my bad english

Gario Rosemma

Physical Ergonomics

Tel +39 011 9083890

e-mail: rosemma.gario@...

----------Internet E-Mail confidentiality footer-------------------------------------------------
Le informazioni contenute in questo messaggio e-mail sono destinate esclusivamente alla persone o enti cui è stata indirizzato e può contenere informazioni riservate. E’ vietato ogni diffusione o altro utilizzo o azione basata sul presente messaggio da parte di persone o enti diverse dal destinatario, se non espressamente autorizzate dal mittente. Se avete ricevuto questo messaggio per errore, siete pregati di contattare immediatamente il mittente via e-mail e di cancellarlo da ogni vostro computer.

The information transmitted in this e-mail message is intended only for the person or entity to which it is addressed and may contain confidential information.Any retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by person or entity other than the intended recipient, if not clearly authorized by the sender, is prohibited.If you have received this communication in error, please notify the sender immediately by e-mail and delete the message from any computer.

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Re: AL_EXT_source_distance_model revisit

2009-10-15T22:41:17Z

On Thursday 15 October 2009 11:11:24 am Bob Aron wrote:
> • What happens when some of the sources are set to something other
> than the default and then the application calls alDistanceModel()?

All existing and newly-generated sources are set to the model given to
alDistanceModel, regardless if anything specific was set on the source.
Normally an app shouldn't use alDistanceModel if it's also using
AL_EXT_source_distance_model (except perhaps to set a global default on
initialization), so I don't think it really matters. Having alDistanceModel
simply reset all sources seems to be the simplest method while maintaining
backwards compatibility with code not aware of the extension. Especially since
there would be no way to "unset" a specific model to make alDistanceModel
affect it again.

> • What is the purpose of having different distance models for each
> source?

Different sounds may want different rolloff properties. For instance, some
sounds may want to follow a more natural inverse distance rolloff, while
others may want to use linear or a manual method to ensure reaching 0/min gain
at max distance (without losing EFX air absorption).

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