Re: [pulseaudio-discuss] review+pull-request: Passthrough support

2011-04-20 Thread Kelly Anderson 

On 04/19/2011 02:03 PM, pl bossart wrote:

I finally managed to figure out why eac3 wouldn't work in passthrough
mode. Apparently my HDMI receiver will only lock on eac3 data if the
AES0 control is set to 0x06 (non-audio, ie iec61937).

ffmpeg -acodec copy -i csi_miami_5.1_256_spx.eac3 -f spdif outfile.pcm
aplay -Dhdmi:AES0=0x6 -fs16 -c2 -r192000 outfile.pcm

the same file plays well through pulseaudio if I modifed default.conf to
[Mapping hdmi-stereo]
device-strings = hdmi:%f,AES0=0x06
channel-map = left,right
priority = 4
direction = output

Somehow we need to find a way to set this AES0 byte when opening the
iec958 or hdmi device in the passthrough code, and reset it back to
0x04 for PCM playback (should be fairly easy to do by looking at the
code of iecset in alsa-utils). Beats me why this was not needed for
plain ac3, but setting the audio bit is actually the correct thing to
do.



So far I haven't been able to get anything except stereo working with 
passthrough.  It seems to me that selecting multiple channels in 
passthrough mode would be very similar to the necessary AES0 changes, so 
maybe we can get that fixed too.


I've basically hit a wall by only being able to passthrough 192khz 
stereo.  When the passthrough data rate exceeds the available rate, my 
code clips the data packets.  Which seems to work OK when the excess is 
maybe 200 bytes or so.  On some HD DTS I'm clipping more like 2K per 
packet which results in some very nasty sounds emanating from the speakers.


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Re: [pulseaudio-discuss] [WIP] Passthrough support

2011-03-24 Thread Kelly Anderson 

Arun,

I've been trying to get higher bit rates to passthrough with no luck.  
Is setting the channel count supported with passthrough (yet)?


It seems that 384k sample rates aren't supported directly in alsa, I did 
some patching to no avail yet.


In any case if the channel count can be specified with passthrough the 
following should work.


paplay --raw --channels=2 --rate=192000 --passthrough 
File.dts.spdif192khz ( this works).


paplay --raw --channels=4 --rate=192000 --passthrough 
File.dts.spdif384khz ( this fails).


To passthrough dolby true-hd it looks like it'll be necessary for more 
than two channels to work.


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Re: [pulseaudio-discuss] [WIP] Passthrough support

2011-03-24 Thread Kelly Anderson 

On 03/24/11 18:58, Dark Shadow wrote:

On Thu, Mar 24, 2011 at 10:50 AM, Anssi Hannulaanssi.hann...@iki.fi  wrote:

On 24.03.2011 16:18, pl bossart wrote:

It seems that 384k sample rates aren't supported directly in alsa, I did
some patching to no avail yet.

In any case if the channel count can be specified with passthrough the
following should work.

paplay --raw --channels=2 --rate=192000 --passthrough File.dts.spdif192khz (
this works).

paplay --raw --channels=4 --rate=192000 --passthrough File.dts.spdif384khz (
this fails).

To passthrough dolby true-hd it looks like it'll be necessary for more than
two channels to work.

There was a thread on dts-hd in alsa-devel at some point. Anssi
(cc:ed) contributed some patches for HDMI and provided the information
below on ffmpeg configurations.
You may want to try at the alsa level before trying with pulseaudio to
make sure your setup is correct. I tend to believe you have to go for
8ch @ 192kHz to make this work based on my limited understanding of
HBR.

Indeed for HBR you need to always specify 8 channels and use rate to
control the final rate (i.e. you either use normal 2 channel
passthrough or HBR 8 channel passthrough).

For example to passthrough the abovementioned 384 kHz stream you need to
use 8 channels and rate of 96000. However, I think 384kHz DTS bitstream
is generally *not* supported by A/V receivers, so you probably want to
use 768kHz (8 channels, 192kHz).

(note: I haven't tested whether HBR works with pulseaudio or not)



The DTS-HD part is not merged yet (patch is in ffmpeg-devel@), but the
TrueHD and E-AC-3 support is already there in ffmpeg trunk.

The ffmpeg commandline to use is:
ffmpeg -i input.file -f spdif output.spdif

For DTS-HD files, to get full passthrough (i.e. not only core), a
-dtshd_rate parameter is needed, which sets the output IEC958 rate.
ffmpeg -i input.file -f spdif -dtshd_rate 192000 output.spdif
ffmpeg -i input.file -f spdif -dtshd_rate 768000 output.spdif
192000Hz is enough for streams that have a bitrate below 6.144Mbps, which
means all DTS-HD High Resolution Audio files and even many of the DTS-HD
Master Audio (the latter are lossless VBR).

To play the spdif files back, I use
aplay -D hdmi:CARD=$CARDNAME,DEV=$DEVICENUM,AES0=0x06 -c $CHANCOUNT -r
$RATE file.spdif

- replacing $CARDNAME with the card name
- replacing $DEVICENUM with 0..3 depending on card and hdmi port (for
non-zero DEVICENUM you'll need a patch from alsa git:
http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=e6d5dcf1f625984605d362338d71162de45a6c60
)
- set $CHANCOUNT and $RATE as per below
  - rate 192000 and channels 2 for IEC958 rate 192 kHz (for e.g. 48 kHz
E-AC-3, and DTS-HD when the IEC958 rate was set to 192000 in ffmpeg)
  - rate 192000 and channels 8 for IEC958 rate 768 kHz (for most TrueHD
files, and for DTS-HD when the rate was set to 768000)
- note that having the 0x02 bit (non-pcm) set in AES0 is mandatory when
$CHANCOUNT is larger than 2, as ALSA uses it to determine whether to use
HBR or not. The additional 0x04 (non-copyright) I use above is not
mandatory, but is the alsa default so I kept it.


--
Anssi Hannula
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Would anyone know where I could get a hold of some DTS-HD samples in
192Khz and 384kHz for testing?


You can extract the dts-hd tracks from your mkv's with mkvextract.  You 
can make them with spdifer (part of AudioFilter).




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Re: [pulseaudio-discuss] [WIP] Passthrough support

2011-03-24 Thread Kelly Anderson 

On 03/24/11 19:35, Dark Shadow wrote:

On Thu, Mar 24, 2011 at 7:19 PM, Kelly Anderson
ke...@silka.with-linux.com  wrote:

On 03/24/11 18:58, Dark Shadow wrote:

On Thu, Mar 24, 2011 at 10:50 AM, Anssi Hannulaanssi.hann...@iki.fi
  wrote:

On 24.03.2011 16:18, pl bossart wrote:

It seems that 384k sample rates aren't supported directly in alsa, I
did
some patching to no avail yet.

In any case if the channel count can be specified with passthrough the
following should work.

paplay --raw --channels=2 --rate=192000 --passthrough
File.dts.spdif192khz (
this works).

paplay --raw --channels=4 --rate=192000 --passthrough
File.dts.spdif384khz (
this fails).

To passthrough dolby true-hd it looks like it'll be necessary for more
than
two channels to work.

There was a thread on dts-hd in alsa-devel at some point. Anssi
(cc:ed) contributed some patches for HDMI and provided the information
below on ffmpeg configurations.
You may want to try at the alsa level before trying with pulseaudio to
make sure your setup is correct. I tend to believe you have to go for
8ch @ 192kHz to make this work based on my limited understanding of
HBR.

Indeed for HBR you need to always specify 8 channels and use rate to
control the final rate (i.e. you either use normal 2 channel
passthrough or HBR 8 channel passthrough).

For example to passthrough the abovementioned 384 kHz stream you need to
use 8 channels and rate of 96000. However, I think 384kHz DTS bitstream
is generally *not* supported by A/V receivers, so you probably want to
use 768kHz (8 channels, 192kHz).

(note: I haven't tested whether HBR works with pulseaudio or not)



The DTS-HD part is not merged yet (patch is in ffmpeg-devel@), but the
TrueHD and E-AC-3 support is already there in ffmpeg trunk.

The ffmpeg commandline to use is:
ffmpeg -i input.file -f spdif output.spdif

For DTS-HD files, to get full passthrough (i.e. not only core), a
-dtshd_rate parameter is needed, which sets the output IEC958 rate.
ffmpeg -i input.file -f spdif -dtshd_rate 192000 output.spdif
ffmpeg -i input.file -f spdif -dtshd_rate 768000 output.spdif
192000Hz is enough for streams that have a bitrate below 6.144Mbps,
which
means all DTS-HD High Resolution Audio files and even many of the DTS-HD
Master Audio (the latter are lossless VBR).

To play the spdif files back, I use
aplay -D hdmi:CARD=$CARDNAME,DEV=$DEVICENUM,AES0=0x06 -c $CHANCOUNT -r
$RATE file.spdif

- replacing $CARDNAME with the card name
- replacing $DEVICENUM with 0..3 depending on card and hdmi port (for
non-zero DEVICENUM you'll need a patch from alsa git:

http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=e6d5dcf1f625984605d362338d71162de45a6c60
)
- set $CHANCOUNT and $RATE as per below
  - rate 192000 and channels 2 for IEC958 rate 192 kHz (for e.g. 48 kHz
E-AC-3, and DTS-HD when the IEC958 rate was set to 192000 in ffmpeg)
  - rate 192000 and channels 8 for IEC958 rate 768 kHz (for most TrueHD
files, and for DTS-HD when the rate was set to 768000)
- note that having the 0x02 bit (non-pcm) set in AES0 is mandatory when
$CHANCOUNT is larger than 2, as ALSA uses it to determine whether to use
HBR or not. The additional 0x04 (non-copyright) I use above is not
mandatory, but is the alsa default so I kept it.

--
Anssi Hannula
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Would anyone know where I could get a hold of some DTS-HD samples in
192Khz and 384kHz for testing?

You can extract the dts-hd tracks from your mkv's with mkvextract.  You can
make them with spdifer (part of AudioFilter).



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Unless I am missing something all the movies I play say 48kHz on my
receivers OSD when played through my PS3 so I thought that was the
best they have, and higher was not used on average Blu-Ray's.


The original samples are taken at 48khz and then compressed into dts or 
dts-hd.  AudioFilter is muxing the dts/dts-hd packets into zero padded 
packets that contain the dts/dts-hd stream.  A typical dts-hd stream is 
going to have packets around the 8k size.  If it is muxed at 192khz 
packets that exceed it's size limit will be clipped.  In my experience I 
was seeing maybe 200 bytes dropped on some packets.  Now if you mux it 
at 384khz you effectively double the size of the packets that can be 
sent to the receiver and you don't have to resort to clipping.


In a nutshell the 192khz and 384khz rates are spdif muxed rates not the 
original sample rate.


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Re: [pulseaudio-discuss] Pulseaudio passthrough branch doesn't read /etc/pulse/default.pa

2011-03-23 Thread Kelly Anderson 

On 03/23/11 17:20, Colin Guthrie wrote:

'Twas brillig, and Maarten Bosmans at 23/03/11 18:36 did gyre and gimble:

2011/3/21 Dark Shadowshadowofdarkn...@gmail.com:

On Sun, Mar 20, 2011 at 5:58 PM, Dark Shadowshadowofdarkn...@gmail.com  wrote:

I have successfully bitstreamed DTS-HD from xbmc through PA but in
using the passthrough git branch it doesn't read that config file
which causes problems for me forcing a sink.

How can I fix this since I have to kill pulseaudio then restart it
with pulseaudio --load=module-alsa-sink device=hdmi:1,3 to get
hdmi audio working


I am running Ubuntu 11.04 Alpha.


I figured it out, I copied default.pa into ~/.pulse and it works now.
Not preferential in a multi-user environment but it is fine for now.
With the feature of lasting through a distro upgrade.

That's a workaround, but pulse still should read the system-wide
startup script (note that that is different from the config file) What
does pulseaudio --dump-config give?

Am I missing something here... default.pa is the startup script and if
it exists in ~/.pulse it overrides the one in /etc/pulse/. The
system-wide one won't be read at all.

Even with the daemon.conf file, the same logic is true (even if it would
in theory be possible to read both, with the local file overriding the
system wide one on a setting-by-setting basis)

Perhaps I'm misunderstanding tho'

Col

I'm pretty sure what he means is that daemon.conf isn't being read from 
/etc/pulse when he doesn't have a ~.pulse/daemon.conf.


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[pulseaudio-discuss] XBMC - PulseAudio - passthrough ready for testing

2011-03-20 Thread Kelly Anderson 
Not sure how many PulseAudio developers are interested, but here's my 
post in the Xbmc developer's forum.


OK, I've been burnin the midnight oil so I'm ready for others to test 
the results.


Download/build/install in order.

git://git.collabora.co.uk/git/user/arun/pulseaudio.git

git://github.com/cbxbiker61/audiofilter.git

git://github.com/cbxbiker61/xbmc.git

Configure xbmc with appropriate devices selected in System 
Settings/Audio, pulse audio devices show up with (pulse) extension.

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Re: [pulseaudio-discuss] [WIP] Passthrough support

2011-03-19 Thread Kelly Anderson 

On 03/16/11 10:48, Arun Raghavan wrote:

On Wed, 2011-03-16 at 11:19 +0530, Arun Raghavan wrote:
[...]

It will work, but not with paplay, only with the extended API when you
create a non-PCM stream. I will rationalise the check for passthrough
sink inputs in a bit, but for now, could you change the checks (there

I've pushed a more comprehensive fix, so you shouldn't need to change
that line and using paplay should work for you.

-- Arun


Arun,

Hey, I just got around to testing without the following patch.  It 
doesn't work.  I also didn't see any recent checkins in the passthrough 
tree that would seem to affect this.  Is it possible the fix didn't get 
pushed?


--- ./src/modules/alsa/alsa-sink.c.orig 2011-03-16 01:17:02.061953124 -0600
+++ ./src/modules/alsa/alsa-sink.c  2011-03-16 01:19:07.492460278 -0600
@@ -1058,7 +1058,7 @@ static int sink_process_msg(pa_msgobject
 pa_sink_input *i = PA_SINK_INPUT(data);
 int r = 0;

-if (PA_LIKELY(pa_format_info_is_pcm(i-format)))
+if (PA_LIKELY(!(i-flags  PA_SINK_INPUT_PASSTHROUGH)) )
 break;

 u-old_rate = u-sink-sample_spec.rate;
@@ -1084,7 +1084,7 @@ static int sink_process_msg(pa_msgobject
 pa_sink_input *i = PA_SINK_INPUT(data);
 int r = 0;

-if (PA_LIKELY(pa_format_info_is_pcm(i-format)))
+if (PA_LIKELY(!(i-flags  PA_SINK_INPUT_PASSTHROUGH)) )
 break;

 /* Passthrough format, see if we need to reset sink sample 
rate */



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Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success

2011-03-18 Thread Kelly Anderson 
Good news :)  I've just watched District 9 and Master and Commander in 
full dts-hd.  A few minor glitches (a couple of pops, I think it might 
be buffer underruns),  other than that it worked great!


One thing I did notice with Master and Commander was that with standard 
dts the dialog was not nearly as clear as it was with dts-hd.  dts-hd 
has about 4 times the bandwidth.


I'll be putting up the newer code sometime tomorrow.  I've been working 
pretty long hours on it and now I just want to relax a bit.


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Re: [pulseaudio-discuss] [WIP] Passthrough support

2011-03-15 Thread Kelly Anderson 

On 03/09/11 03:10, Arun Raghavan wrote:

Hello all,
Based on the RFC I posted earlier, I've implemented basic passthrough
support in some usable shape. It's up on the passthrough branch of each
of:

PulseAudio:
http://git.collabora.co.uk/?p=user/arun/pulseaudio.git;a=shortlog;h=refs/heads/passthrough
gst-plugins-base:
http://git.collabora.co.uk/?p=user/arun/gst-plugins-base.git;a=shortlog;h=refs/heads/passthrough
gst-plugins-good:
http://git.collabora.co.uk/?p=user/arun/gst-plugins-good.git;a=shortlog;h=refs/heads/passthrough



Hello,

I'm wanting to clone the passthrough head but I haven't been able to 
figure out which git option I need.  So far All I've been able to get is 
master.


git ls-remote git://git.collabora.co.uk/git/user/arun/pulseaudio.git

dda564f50b55340ff4bfbaa8d6d6fc6427f764f4HEAD
dda564f50b55340ff4bfbaa8d6d6fc6427f764f4refs/heads/master
b42400816031c17cdf6a63f446510e13d0ab9828
refs/heads/module-echo-cancel

aa78853d728172f7fa564b2782f6e84661bfcd15refs/heads/no-ramping
4cd90d9e32ca9a23e3c0f7615974ea0c55ff3e49refs/heads/orc
a5b33c7122ee80dc6906fd51f7aa39322322f24brefs/heads/passthrough
a4e9e9b631841eac5646b925c198d236c92d8aabrefs/heads/passthrough-bt
3a6dc95735e79f791788cda73a0fc36e46cb43earefs/heads/sq-aec

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Re: [pulseaudio-discuss] [WIP] Passthrough support

2011-03-15 Thread Kelly Anderson 

On 03/15/11 22:04, Arun Raghavan wrote:

On Tue, 2011-03-15 at 18:50 -0600, Kelly Anderson wrote:

On 03/09/11 03:10, Arun Raghavan wrote:

Hello all,
Based on the RFC I posted earlier, I've implemented basic passthrough
support in some usable shape. It's up on the passthrough branch of each
of:

PulseAudio:
http://git.collabora.co.uk/?p=user/arun/pulseaudio.git;a=shortlog;h=refs/heads/passthrough
gst-plugins-base:
http://git.collabora.co.uk/?p=user/arun/gst-plugins-base.git;a=shortlog;h=refs/heads/passthrough
gst-plugins-good:
http://git.collabora.co.uk/?p=user/arun/gst-plugins-good.git;a=shortlog;h=refs/heads/passthrough





Hey Arun,

I've built and installed the passthrough branch.

Isn't changing sample rates supported yet?  So far I can only 
passthrough sample rates that match the default-sample-rate in 
daemon.conf.  Changeable rates is critical to being able to support for 
example dts-hd at 192khz, when standard dts will be 48khz.


I'm using paplay for basic functionality checks.

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Re: [pulseaudio-discuss] [WIP] Passthrough support

2011-03-15 Thread Kelly Anderson 

On 03/15/11 23:13, Kelly Anderson wrote:

On 03/15/11 22:04, Arun Raghavan wrote:

On Tue, 2011-03-15 at 18:50 -0600, Kelly Anderson wrote:

On 03/09/11 03:10, Arun Raghavan wrote:

Hello all,
Based on the RFC I posted earlier, I've implemented basic passthrough
support in some usable shape. It's up on the passthrough branch of 
each

of:

PulseAudio:
http://git.collabora.co.uk/?p=user/arun/pulseaudio.git;a=shortlog;h=refs/heads/passthrough 


gst-plugins-base:
http://git.collabora.co.uk/?p=user/arun/gst-plugins-base.git;a=shortlog;h=refs/heads/passthrough 


gst-plugins-good:
http://git.collabora.co.uk/?p=user/arun/gst-plugins-good.git;a=shortlog;h=refs/heads/passthrough 







Hey Arun,

I've built and installed the passthrough branch.

Isn't changing sample rates supported yet?  So far I can only 
passthrough sample rates that match the default-sample-rate in 
daemon.conf.  Changeable rates is critical to being able to support 
for example dts-hd at 192khz, when standard dts will be 48khz.


I'm using paplay for basic functionality checks.


I forgot to mention that I patched alsa-sink.c with extended rates as per:

--- ./src/modules/alsa/alsa-sink.c.orig2011-03-15 22:15:20.701558916 
-0600

+++ ./src/modules/alsa/alsa-sink.c2011-03-15 22:37:57.437394161 -0600
@@ -1436,6 +1436,7 @@ static pa_idxset* sink_get_formats(pa_si
 {
 pa_idxset *ret = pa_idxset_new(NULL, NULL);
 pa_format_info *f;
+const char *dev_nam = ((struct userdata*)s-userdata)-device_name;

 pa_assert(s);

@@ -1444,12 +1445,12 @@ static pa_idxset* sink_get_formats(pa_si
 pa_idxset_put(ret, f, NULL);

 /* FIXME: is there no better way to check for this? */
-if (strncmp(iec958, ((struct userdata*)s-userdata)-device_name, 
6) == 0) {
+if ( ! strncmp(iec958, dev_nam, 6) || ! strncmp(hdmi, dev_nam, 
4) ) {
 /* FIXME: Eventually, we want the list of supported formats to 
be set
  * as properties by the GUI based on what the user says is 
supported by

  * the receiver */
 /* FIXME: How do we figure out supported rates? :( */
-unsigned int i, rates[] = { 32000, 44100, 48000 };
+unsigned int i, rates[] = { 32000, 44100, 48000, 96000, 192000, 
384000 };


 /* FIXME: Make only one format_info once we've resolved how to do
  * lists/ranges in proplists */



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Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success

2011-03-14 Thread Kelly Anderson 

On 03/06/11 14:54, pl bossart wrote:

I've got pulseaudio xbmc passthrough working.  I've rolled it together with
a previously windows c++ library called Ac3Filter.  I've been tweeking up
Ac3Filter quite a bit.  I'm using the Ac3Filter library to Mux spdif.  It
was a well written library/tools that was relatively easy to get ported to
linux.

I'm watching Avatar in wonderful DTS surround sound now.

All you need is a 3D TV now...
Seriously, this is good. Can you send a link to this ported AC3Filter
code? I thought it was too Windows/DirectX-oriented to be used. I
didn't look too much since it's GPL, and LGPL is preferred in terms of
integration with proprietary components.
Thanks,
-Pierre
I've update AudioFilter and the Xbmc AudiFilter patch.  It now passes 
the core dts-5.1 portion of dts-hd audio streams.


Supposedly ffmpeg can mux a dts-hd stream that passes through the full 
dts-hd stream.  I want to analyse what ffmpeg is doing to the hd portion 
of the stream so I can implement it.  If anyone knows of the ffmpeg 
command that will mux a dts-hd stream for spdif let me know.


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Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success

2011-03-08 Thread Kelly Anderson 

On 03/06/2011 02:54 PM, pl bossart wrote:

All you need is a 3D TV now...
Seriously, this is good. Can you send a link to this ported AC3Filter
code? I thought it was too Windows/DirectX-oriented to be used. I
didn't look too much since it's GPL, and LGPL is preferred in terms of
integration with proprietary components.
Thanks,
-Pierre


Pierre,

OK, I've cleaned the code up and it works really well now (for my 
purposes).  It's really efficient with processing DTS files, I haven't 
profiled Ac3 yet.  I've rolled it into my copy of xbmc and all seems to 
be working really well.  I'll be putting patches for xbmc up as well as 
soon as I've prep'd them.


You could do me a favor if you would complete sink_pulse.{cpp,h}.  I've 
been working on other things and that's really on the back burner for 
me.  Once the SinkPulse is done, we'll be able to play raw DTS/AC3 files 
pretty much the same way paplay does for other formats.  At some point 
we might even want to roll AudioFilter into paplay.


I renamed Ac3Filter to AudioFilter.  It really does a bunch more than 
just Ac3 and I've renamed interfaces, etc.


I'm still renaming files and moving the around a bit, but the headers 
should be pretty stable.


http://silka.with-linux.com/audiofilter/

I'll probably put the code in a git repository once I've got the file 
renaming out of the way.


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Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success

2011-03-08 Thread Kelly Anderson 

On 03/08/2011 03:32 PM, Kelly Anderson wrote:

On 03/06/2011 02:54 PM, pl bossart wrote:

All you need is a 3D TV now...
Seriously, this is good. Can you send a link to this ported AC3Filter
code? I thought it was too Windows/DirectX-oriented to be used. I
didn't look too much since it's GPL, and LGPL is preferred in terms of
integration with proprietary components.
Thanks,
-Pierre


Pierre,

OK, I've cleaned the code up and it works really well now (for my 
purposes).  It's really efficient with processing DTS files, I haven't 
profiled Ac3 yet.  I've rolled it into my copy of xbmc and all seems 
to be working really well.  I'll be putting patches for xbmc up as 
well as soon as I've prep'd them.


You could do me a favor if you would complete sink_pulse.{cpp,h}.  
I've been working on other things and that's really on the back burner 
for me.  Once the SinkPulse is done, we'll be able to play raw DTS/AC3 
files pretty much the same way paplay does for other formats.  At some 
point we might even want to roll AudioFilter into paplay.


I renamed Ac3Filter to AudioFilter.  It really does a bunch more than 
just Ac3 and I've renamed interfaces, etc.


I'm still renaming files and moving the around a bit, but the headers 
should be pretty stable.


http://silka.with-linux.com/audiofilter/

I'll probably put the code in a git repository once I've got the file 
renaming out of the way.


OK,  I've just put up the patches necessary to get passthrough working 
with Nvidia hdmi on XBMC.


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Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success

2011-03-06 Thread Kelly Anderson 

On 03/06/2011 02:54 PM, pl bossart wrote:

I've got pulseaudio xbmc passthrough working.  I've rolled it together with
a previously windows c++ library called Ac3Filter.  I've been tweeking up
Ac3Filter quite a bit.  I'm using the Ac3Filter library to Mux spdif.  It
was a well written library/tools that was relatively easy to get ported to
linux.

I'm watching Avatar in wonderful DTS surround sound now.

All you need is a 3D TV now...
Seriously, this is good. Can you send a link to this ported AC3Filter
code? I thought it was too Windows/DirectX-oriented to be used. I
didn't look too much since it's GPL, and LGPL is preferred in terms of
integration with proprietary components.
Thanks,
-Pierre


I'm putting the finishing touches on it, give me a few days.  I'm 
cleaning up the code and putting it in it's own namespace so it'll 
integrate with large projects without conflict.


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[pulseaudio-discuss] Nvidia hdmi passthrough patch

2011-03-04 Thread Kelly Anderson 

Hey,

Here's a patch that I created for git that adds passthrough support for 
the Nvidia hdmi port.  Pretty much just copy/paste from the iec958 sections.



--- ./src/modules/alsa/alsa-sink.c.orig2011-02-16 22:58:11.0 
-0700

+++ ./src/modules/alsa/alsa-sink.c2011-02-17 02:53:21.452857569 -0700
@@ -1708,7 +1708,8 @@ static int setup_mixer(struct userdata *
 }

 /* FIXME: need automatic detection rather than hard-coded path */
-if (!strcmp(u-mixer_path-name, iec958-passthrough-output)) {
+if (!strcmp(u-mixer_path-name, iec958-passthrough-output)
+|| !strcmp(u-mixer_path-name, hdmi-passthrough-output)) {
 u-sink-flags |= PA_SINK_PASSTHROUGH;
 } else {
 u-sink-flags = ~PA_SINK_PASSTHROUGH;
--- ./src/modules/alsa/mixer/profile-sets/default.conf.orig
2011-02-16 22:58:11.0 -0700
+++ ./src/modules/alsa/mixer/profile-sets/default.conf2011-02-17 
02:46:25.401973394 -0700

@@ -141,6 +141,25 @@ channel-map = left,right
 priority = 4
 direction = output

+[Mapping hdmi-passthrough]
+device-strings = hdmi:%f
+channel-map = left,right
+direction = output
+paths-output = hdmi-passthrough-output
+priority = 5
+
+[Mapping hdmi-ac3-surround-40]
+device-strings = a52:%f
+channel-map = front-left,front-right,rear-left,rear-right
+priority = 2
+direction = output
+
+[Mapping hdmi-ac3-surround-51]
+device-strings = a52:%f
+channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe
+priority = 3
+direction = output
+
 ; An example for defining multiple-sink profiles
 #[Profile output:analog-stereo+output:iec958-stereo+input:analog-stereo]
 #description = Foobar
--- ./src/modules/alsa/mixer/paths/hdmi-passthrough-output.conf.orig
2011-02-17 02:40:11.766291419 -0700
+++ ./src/modules/alsa/mixer/paths/hdmi-passthrough-output.conf
2011-02-17 02:39:48.168179164 -0700

@@ -0,0 +1,19 @@
+# This file is part of PulseAudio.
+#
+# PulseAudio 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.
+#
+# PulseAudio 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
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# along with PulseAudio; if not, write to the Free Software Foundation,
+# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+
+
+[Element HDMI]
+switch = mute
--- ./src/modules/alsa/mixer/paths/hdmi-stereo-output.conf.orig
2011-02-17 02:40:18.921630373 -0700
+++ ./src/modules/alsa/mixer/paths/hdmi-stereo-output.conf2011-02-17 
02:39:55.903542803 -0700

@@ -0,0 +1,19 @@
+# This file is part of PulseAudio.
+#
+# PulseAudio 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.
+#
+# PulseAudio 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
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# along with PulseAudio; if not, write to the Free Software Foundation,
+# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+
+
+[Element HDMI]
+switch = mute
--- ./src/modules/alsa/alsa-mixer.c.orig2011-02-16 
22:58:11.0 -0700

+++ ./src/modules/alsa/alsa-mixer.c2011-02-17 02:38:08.117560106 -0700
@@ -1888,7 +1888,9 @@ static int path_verify(pa_alsa_path *p)
 { analog-output-mono, N_(Analog Mono Output) },
 { analog-output-speaker,  N_(Analog Speakers) },
 { iec958-stereo-output,   N_(Digital Output (IEC958)) },
-{ iec958-passthrough-output,  N_(Digital Passthrough 
(IEC958)) }
+{ iec958-passthrough-output,  N_(Digital Passthrough 
(IEC958)) },

+{ hdmi-stereo-output,  N_(Digital Output (HDMI)) },
+{ hdmi-passthrough-output,  N_(Digital Passthrough (HDMI)) }
 };

 pa_alsa_element *e;
@@ -2914,7 +2916,10 @@ static int mapping_verify(pa_alsa_mappin
 { iec958-passthrough, N_(Digital Passthrough  (IEC958)) },
 { iec958-ac3-surround-40, N_(Digital Surround 4.0 
(IEC958/AC3)) },
 { iec958-ac3-surround-51, N_(Digital Surround 5.1 
(IEC958/AC3)) },

-{ hdmi-stereo,N_(Digital Stereo (HDMI)) }
+{ hdmi-stereo,N_(Digital Stereo (HDMI)) },
+{ hdmi-passthrough, N_(Digital Passthrough  (HDMI)) },
+{ hdmi-ac3-surround-40, N_(Digital Surround 4.0 (HDMI/AC3)) },
+

[pulseaudio-discuss] Nvidia hdmi passthrough

2011-03-04 Thread Kelly Anderson 
diff --git a/src/Makefile.am b/src/Makefile.am
index 24e2f82..bb501fb 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -1131,7 +1131,9 @@ dist_alsapaths_DATA = \
modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf \
modules/alsa/mixer/paths/analog-output-mono.conf \
modules/alsa/mixer/paths/iec958-stereo-output.conf \
-   modules/alsa/mixer/paths/iec958-passthrough-output.conf
+   modules/alsa/mixer/paths/iec958-passthrough-output.conf \
+   modules/alsa/mixer/paths/hdmi-stereo-output.conf \
+   modules/alsa/mixer/paths/hdmi-passthrough-output.conf
 
 endif
 
diff --git a/src/modules/alsa/alsa-mixer.c b/src/modules/alsa/alsa-mixer.c
index fa07674..ad28c41 100644
--- a/src/modules/alsa/alsa-mixer.c
+++ b/src/modules/alsa/alsa-mixer.c
@@ -1952,7 +1952,9 @@ static int path_verify(pa_alsa_path *p) {
 { analog-output-mono, N_(Analog Mono Output) },
 { analog-output-speaker,  N_(Analog Speakers) },
 { iec958-stereo-output,   N_(Digital Output (IEC958)) },
-{ iec958-passthrough-output,  N_(Digital Passthrough (IEC958)) }
+{ iec958-passthrough-output,  N_(Digital Passthrough (IEC958)) },
+{ hdmi-stereo-output,  N_(Digital Output (HDMI)) },
+{ hdmi-passthrough-output,  N_(Digital Passthrough (HDMI)) }
 };
 
 pa_alsa_element *e;
@@ -2995,7 +2997,10 @@ static int mapping_verify(pa_alsa_mapping *m, const 
pa_channel_map *bonus) {
 { iec958-passthrough, N_(Digital Passthrough  (IEC958)) },
 { iec958-ac3-surround-40, N_(Digital Surround 4.0 (IEC958/AC3)) },
 { iec958-ac3-surround-51, N_(Digital Surround 5.1 (IEC958/AC3)) },
-{ hdmi-stereo,N_(Digital Stereo (HDMI)) }
+{ hdmi-stereo,N_(Digital Stereo (HDMI)) },
+{ hdmi-passthrough, N_(Digital Passthrough  (HDMI)) },
+{ hdmi-ac3-surround-40, N_(Digital Surround 4.0 (HDMI/AC3)) },
+{ hdmi-ac3-surround-51, N_(Digital Surround 5.1 (HDMI/AC3)) }
 };
 
 pa_assert(m);
@@ -3119,6 +3124,7 @@ static int profile_verify(pa_alsa_profile *p) {
 { output:analog-mono+input:analog-mono, N_(Analog Mono Duplex) 
},
 { output:analog-stereo+input:analog-stereo, N_(Analog Stereo 
Duplex) },
 { output:iec958-stereo+input:iec958-stereo, N_(Digital Stereo 
Duplex (IEC958)) },
+{ output:hdmi-stereo+input:hdmi-stereo, N_(Digital Stereo Duplex 
(HDMI)) },
 { off,  N_(Off) }
 };
 
diff --git a/src/modules/alsa/alsa-sink.c b/src/modules/alsa/alsa-sink.c
index 9ed4d4d..e10b99d 100644
--- a/src/modules/alsa/alsa-sink.c
+++ b/src/modules/alsa/alsa-sink.c
@@ -1708,7 +1708,8 @@ static int setup_mixer(struct userdata *u, pa_bool_t 
ignore_dB, pa_bool_t sync_v
 }
 
 /* FIXME: need automatic detection rather than hard-coded path */
-if (!strcmp(u-mixer_path-name, iec958-passthrough-output)) {
+if (!strcmp(u-mixer_path-name, iec958-passthrough-output)
+|| !strcmp(u-mixer_path-name, hdmi-passthrough-output)) {
 u-sink-flags |= PA_SINK_PASSTHROUGH;
 } else {
 u-sink-flags = ~PA_SINK_PASSTHROUGH;
diff --git a/src/modules/alsa/mixer/paths/hdmi-passthrough-output.conf 
b/src/modules/alsa/mixer/paths/hdmi-passthrough-output.conf
new file mode 100644
index 000..fe58f68
--- /dev/null
+++ b/src/modules/alsa/mixer/paths/hdmi-passthrough-output.conf
@@ -0,0 +1,19 @@
+# This file is part of PulseAudio.
+#
+# PulseAudio 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.
+#
+# PulseAudio 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
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# along with PulseAudio; if not, write to the Free Software Foundation,
+# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+
+
+[Element HDMI]
+switch = mute
diff --git a/src/modules/alsa/mixer/paths/hdmi-stereo-output.conf 
b/src/modules/alsa/mixer/paths/hdmi-stereo-output.conf
new file mode 100644
index 000..fe58f68
--- /dev/null
+++ b/src/modules/alsa/mixer/paths/hdmi-stereo-output.conf
@@ -0,0 +1,19 @@
+# This file is part of PulseAudio.
+#
+# PulseAudio 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.
+#
+# PulseAudio is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the

Re: [pulseaudio-discuss] Nvidia hdmi passthrough

2011-03-04 Thread Kelly Anderson 

I finally got git send-email to work.  Had to change my sendmail config.

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[pulseaudio-discuss] pulseaudio passthrough samplerate

2011-03-03 Thread Kelly Anderson 

Hey,

I've been hacking on xbmc - pulseaudio passthrough and I'm getting there 
a little bit at a time.


It seems that at the moment that if the daemon.conf default-sample-rate 
is set to 44100 then all I can passthrough is 44100 and if it's 48000 
all that works are 48000 samples.  Is that a limitation in the current 
pulse implementation?


I expected that this would be set from the pa_sample_spec, but that 
seems to be ignored currently.


I'm using paplay --rate ? --raw --passthrough for testing.

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[pulseaudio-discuss] pulseaudio xbmc passthrough success

2011-03-03 Thread Kelly Anderson 

Yahoo!

I've got pulseaudio xbmc passthrough working.  I've rolled it together 
with a previously windows c++ library called Ac3Filter.  I've been 
tweeking up Ac3Filter quite a bit.  I'm using the Ac3Filter library to 
Mux spdif.  It was a well written library/tools that was relatively easy 
to get ported to linux.


I'm watching Avatar in wonderful DTS surround sound now.

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Re: [pulseaudio-discuss] pulseaudio xbmc passthrough success

2011-03-03 Thread Kelly Anderson 
Forgot to mention.  I'm getting Hi-def video/audio and only using 8% 
cpu.  That's just awesome.  Vdpau is handling the video decode.


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Re: [pulseaudio-discuss] pavucontrol. Also Nvidia HDMI

2011-02-23 Thread Kelly Anderson 

On 02/21/11 17:46, Christopher K. wrote:

My second question is about sound over hdmi with an Nvidia GTX460
(NVC4 /GF104 family):

aplay -L gives
hdmi:CARD=NVidia,DEV=0   -  pcm3p/
hdmi:CARD=NVidia,DEV=1   -  pcm7p/
hdmi:CARD=NVidia,DEV=2   -  pcm8p/
hdmi:CARD=NVidia,DEV=3   -  pcm9p/

but only pcm7p actually does anything.

So currently I use
load-module module-alsa-sink device=hw:1,7
to add a sink, but PA automatically adds pcm3p as GF104 High
Definition Audio Controller Digital Stereo (HDMI),
so I effectively have two nvidia-hdmi sinks, one of which does
absolutely nothing.

Is there any way to disable the automatically added sink OR define
which device to chose as default?


What you want to do is add probe_mask to your snd-hda-intel module 
options.  If you google nvidia hdmi probe_mask you can find an explanation.


For example I have in /etc/modprobe.d/snd_hda_intel.conf:

options snd-hda-intel probe_mask=0x102

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Re: [pulseaudio-discuss] [RFC] Passthrough support

2011-02-18 Thread Kelly Anderson 

On 02/18/11 02:41, Arun Raghavan wrote:

Hello!

On Tue, 2011-02-15 at 22:02 +0530, Arun Raghavan wrote:

Hey folks,
I've put up a draft proposed API changes to get passthrough support to
the point where things Just Work™ for at least the common cases, based
on previous discussion on-/off-list:

http://pulseaudio.org/wiki/PassthroughSupport

Please review/comment. Once we have some consensus, I'll send in patches
to actually get this done.

I've updated this page with the changes discussed so far (and translated
them into the actual API changes that we need). Please take a look and
let me know if this looks acceptable.

Cheers,
Arun

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The alsa driver has creates a /proc entry that lists the coding types 
available for a particular device.  It'd be nice if you would use that 
for the list of supported coding types (at least for a default).  I'm 
not sure the list would be automatically updated if you unplugged a 
device and plugged in a new device.  The comprehensive list of supported 
coding types is in alsa-kernel/pci/hda/hda-eld.c.


cat /proc/asound/card0/eld#1.0

monitor_present 1
eld_valid   1
monitor_nameTX-SR608

connection_type HDMI
eld_version [0x2] CEA-861D or below
edid_version[0x3] CEA-861-B, C or D
manufacture_id  0xcb3d
product_id  0xa62
port_id 0x2
support_hdcp0
support_ai  0
audio_sync_delay0
speakers[0x4f] FL/FR LFE FC RL/RR RLC/RRC
sad_count   8
sad0_coding_type[0x1] LPCM
sad0_channels   2
sad0_rates  [0x1ee0] 44100 48000 88200 176400 192000 384000
sad0_bits   [0xe] 16 20 24
sad1_coding_type[0x1] LPCM
sad1_channels   8
sad1_rates  [0x1ee0] 44100 48000 88200 176400 192000 384000
sad1_bits   [0xe] 16 20 24
sad2_coding_type[0x2] AC-3
sad2_channels   8
sad2_rates  [0xe0] 44100 48000 88200
sad2_max_bitrate64
sad3_coding_type[0x7] DTS
sad3_channels   8
sad3_rates  [0xc0] 48000 88200
sad3_max_bitrate1536000
sad4_coding_type[0x9] DSD (One Bit Audio)
sad4_channels   6
sad4_rates  [0x40] 48000
sad5_coding_type[0xa] E-AC-3/DD+ (Dolby Digital Plus)
sad5_channels   8
sad5_rates  [0xc0] 48000 88200
sad6_coding_type[0xb] DTS-HD
sad6_channels   8
sad6_rates  [0x1ec0] 48000 88200 176400 192000 384000
sad7_coding_type[0xc] MLP (Dolby TrueHD)
sad7_channels   8
sad7_rates  [0x1480] 88200 192000


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Re: [pulseaudio-discuss] [RFC] Passthrough support

2011-02-18 Thread Kelly Anderson 

On 02/18/11 06:26, Arun Raghavan wrote:

On Fri, 2011-02-18 at 03:42 -0700, Kelly Anderson wrote:
[...]

The alsa driver has creates a /proc entry that lists the coding types
available for a particular device.  It'd be nice if you would use that
for the list of supported coding types (at least for a default).  I'm
not sure the list would be automatically updated if you unplugged a
device and plugged in a new device.  The comprehensive list of supported
coding types is in alsa-kernel/pci/hda/hda-eld.c.

I don't know how it works for HDMI, but if it does update that list
based on the receiver, that would be ideal for us, since we should be
able to probe that while routing.


Well I know for a fact with hdmi it updates the list when the kernel 
boots.  I'm not sure it gets updated if you switch devices after the 
kernel is up.  I'd say in over 90% of the cases it would be correct.


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[pulseaudio-discuss] HDMI passthrough request for help

2011-02-18 Thread Kelly Anderson 

Hey,

With all of the discussions going on about passthrough I thought someone 
on the list might be able to give me some insight into getting 
passthrough working properly with xbmc.


First of all I patched pulseaudio git with some updates to treat 
hdmi-passthrough as it does iec958-passthrough.  That seems to work.  I 
can use paplay to pass a dts-wav file through to the receiver and the 
receiver triggers into dts mode and plays it.  A native dts file 
stripped from and mkv movie file on the other hand doesn't play.


I made some minor changes to the Xbmc pulseaudio driver so that it does 
not ignore passthrough requests.  Xbmc's pulseaudio driver assumes 
(without patching) that passthrough is not available for pulseaudio.  
Now I can select the pulseaudio passthrough device.  Xbmc has code in it 
to manipulate dts streams.


The problem is that I can start a movie, Xbmc will select the dts track 
and play, but the receiver doesn't go into dts mode, but instead just 
buzzes the speakers.


Any ideas?

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Re: [pulseaudio-discuss] HDMI passthrough request for help

2011-02-18 Thread Kelly Anderson 

On 02/18/11 07:41, pl bossart wrote:

First of all I patched pulseaudio git with some updates to treat
hdmi-passthrough as it does iec958-passthrough.  That seems to work.  I can
use paplay to pass a dts-wav file through to the receiver and the receiver
triggers into dts mode and plays it.  A native dts file stripped from and
mkv movie file on the other hand doesn't play.

I made some minor changes to the Xbmc pulseaudio driver so that it does not
ignore passthrough requests.  Xbmc's pulseaudio driver assumes (without
patching) that passthrough is not available for pulseaudio.  Now I can
select the pulseaudio passthrough device.  Xbmc has code in it to manipulate
dts streams.

The problem is that I can start a movie, Xbmc will select the dts track and
play, but the receiver doesn't go into dts mode, but instead just buzzes the
speakers.

Did you actually add the PASSTHROUGH flag when you create the pa_stream?
When the receiver doesn't lock it's typically when the data was
modified by the volume control.
Another possibility is mismatch between frequencies. Make sure the
sink and the stream sampling frequencies match.


Yeah, I added the passthrough flag.  The code dealing with volume 
control has a conditonal around it to ignore requests when using 
passthrough.  I'll look a bit more a sample frequencies.  I did change 
the default pulse frequency to 48000 in daemon.conf, not sure if that 
was necessary or not.



Note that we are completely reworking this part, you may want to wait
1-2 months before the feature becomes more stable.
-Pierre
Yeah, I know.  But I need it now! hehe, I've got quite a few blue-rays 
that have been put into mkv's and I just bought a very nice surround 
sound system.


Actually I though working on it now would be beneficial both for pulse 
and xbmc.  Xbmc should be able to rock out as soon as pulse is 
official.  I might uncover something useful to pulse while hacking xbmc.



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Re: [pulseaudio-discuss] pulseaudio-0.9.21, alsa-1.0.23, kde-4.4.5, consolekit, dbus

2010-09-13 Thread Kelly Anderson 

 On 09/12/10 19:04, sibu xolo wrote:

pcm.pulse {
 type pulse
}
ctl.pulse {
 type pulse
}
pcm.!default {
 type pulse
}
ctl.!default {
 type pulse
}




Try the following, it should allow alsamixer to control the default device

pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
#pcm.!default {
#type pulse
#}
#ctl.!default {
#type pulse
#}



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Re: [pulseaudio-discuss] PulseAudio 0.9.21 deadlocking when accessed through OpenAl 1.10.622 (AlienArena)

2010-01-05 Thread Kelly Anderson 

On 01/05/2010 12:38 PM, Lennart Poettering wrote:

On Wed, 23.12.09 13:42, Kelly Anderson (ke...@silka.with-linux.com) wrote:

   

Here's an strace (with prints sprinkled in the OpenAl pulse code) which
demonstrates the deadlock.  A patch to undo the commit is at the end.
 

Does
http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=978d33b609969c3b9bbbd759e0f11aaf856c80cf

Fix the problem?

   

volatile does not fix the problem.  It still deadlocks unless I
revert the original commit.
 

I am quite sure that
http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=c5fdbeab85dfe69ebaec92969ec2ce77b105f1df
fixes the problem.

Could you please verify that?

Thanks,

Lennart

   
It does not fix the deadlock.  I still need to revert the patch that 
initiated the problem.


Here is a copy of the function in OpenAl 1.10.622 with a comment on when 
and where the deadlock occurs.


static ALCboolean pulse_open(ALCdevice *device, const ALCchar 
*device_name) //{{{

{
pulse_data *data = ppa_xmalloc0(sizeof(pulse_data));
pa_context_state_t state;

AL_PRINT(pulse_open()\n);
if(ppa_get_binary_name(data-path_name, sizeof(data-path_name)))
data-context_name = ppa_path_get_filename(data-path_name);
else
data-context_name = OpenAL Soft;

AL_PRINT(starting pa_threaded_mainloop_new()\n);
if(!(data-loop = ppa_threaded_mainloop_new()))
{
AL_PRINT(pa_threaded_mainloop_new() failed!\n);
goto out;
}

AL_PRINT(starting pa_threaded_mainloop_start()\n);
if(ppa_threaded_mainloop_start(data-loop)  0)
{
AL_PRINT(pa_threaded_mainloop_start() failed\n);
goto out;
}

AL_PRINT(starting pa_threaded_mainloop_lock()\n);
ppa_threaded_mainloop_lock(data-loop);
device-ExtraData = data;

AL_PRINT(starting pa_context_new()\n);
data-context = 
ppa_context_new(ppa_threaded_mainloop_get_api(data-loop), 
data-context_name);

if(!data-context)
{
AL_PRINT(pa_context_new() failed: %s\n,
 ppa_strerror(ppa_context_errno(data-context)));

ppa_threaded_mainloop_unlock(data-loop);
goto out;
}

AL_PRINT(starting pa_context_set_state_callback()\n);
ppa_context_set_state_callback(data-context, 
context_state_callback, device);


AL_PRINT(starting pa_context_connect()\n);
if(ppa_context_connect(data-context, NULL, PA_CONTEXT_NOAUTOSPAWN, 
NULL)  0)

{
AL_PRINT(Context did not connect: %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;
}

AL_PRINT(starting pa_context_get_state()\n);
for (;;)
{
state = ppa_context_get_state(data-context);
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;
}

if (state == PA_CONTEXT_READY)
break;
AL_PRINT(starting pa_threaded_mainloop_wait(), state=%s\n
, getContextState(state));
ppa_threaded_mainloop_wait(data-loop);
-  It consistently deadlocks here the second time through the 
loop **


AL_PRINT(starting pa_threaded_mainloop_accept()\n);
ppa_threaded_mainloop_accept(data-loop);
}
AL_PRINT(starting pa_context_set_state_callback()\n);
ppa_context_set_state_callback(data-context, 
context_state_callback2, device);


device-szDeviceName = strdup(device_name);

AL_PRINT(starting pa_threaded_mainloop_unlock()\n);
ppa_threaded_mainloop_unlock(data-loop);
return ALC_TRUE;

out:
if(data-loop)
{
ppa_threaded_mainloop_stop(data-loop);
ppa_threaded_mainloop_free(data-loop);
}

device-ExtraData = NULL;
ppa_xfree(data);
return ALC_FALSE;
} //}}}



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[pulseaudio-discuss] PulseAudio 0.9.21 deadlocking when accessed through OpenAl 1.10.622 (AlienArena)

2009-12-23 Thread Kelly Anderson 
AlienArena was deadlocking trying to go through OpenAl which was set to 
use PulseAudio.  After quite a bit of debugging (entering prints in 
OpenAl pulse code), I finally found a solution with the help of a SUSE 
bug report.


http://lists.opensuse.org/opensuse-bugs/2009-10/msg10497.html

which points at this commit.

http://git.0pointer.de/?p=pulseaudio.git;a=commitdiff;h=4f5e2b745ea357e2b5c815ff33a556505a7d1f18

Here's an strace (with prints sprinkled in the OpenAl pulse code) which 
demonstrates the deadlock.  A patch to undo the commit is at the end.


write(3, pDeviceName:PulseAudio Software\n..., 32) = 32
write(3, qalcOpenDevice:PulseAudio Softwar..., 35) = 35
mmap(NULL, 864256, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 
0) = 0x7f07b822d000

brk(0x626f000)  = 0x626f000
write(2, AL lib: pulseaudio.c:356: pulse_o..., 39AL lib: 
pulseaudio.c:356: pulse_open()

) = 39
readlink(/proc/self/exe, /usr/share/games/AlienArena/crx..., 99) = 31
write(2, AL lib: pulseaudio.c:362: startin..., 62AL lib: 
pulseaudio.c:362: starting pa_threaded_mainloop_new()

) = 62
pipe([4, 5])= 0
fcntl(4, F_GETFL)   = 0 (flags O_RDONLY)
fcntl(4, F_SETFL, O_RDONLY|O_NONBLOCK)  = 0
fcntl(5, F_GETFL)   = 0x1 (flags O_WRONLY)
fcntl(5, F_SETFL, O_WRONLY|O_NONBLOCK)  = 0
fcntl(4, F_GETFD)   = 0
fcntl(4, F_SETFD, FD_CLOEXEC)   = 0
fcntl(5, F_GETFD)   = 0
fcntl(5, F_SETFD, FD_CLOEXEC)   = 0
write(2, AL lib: pulseaudio.c:369: startin..., 64AL lib: 
pulseaudio.c:369: starting pa_threaded_mainloop_start()

) = 64
mmap(NULL, 8392704, PROT_READ|PROT_WRITE|PROT_EXEC, 
MAP_PRIVATE|MAP_ANONYMOUS|MAP_STACK, -1, 0) = 0x7f07afb57000

mprotect(0x7f07afb57000, 4096, PROT_NONE) = 0
clone(child_stack=0x7f07b03571f0, 
flags=CLONE_VM|CLONE_FS|CLONE_FILES|CLONE_SIGHAND|CLONE_THREAD|CLONE_SYSVSEM|CLONE_SETTLS|CLONE_PARENT_SETTID|CLONE_CHILD_CLEARTID, 
parent_tidptr=0x7f07b03579e0, tls=0x7f07b0357910, 
child_tidptr=0x7f07b03579e0) = 5393
write(2, AL lib: pulseaudio.c:376: startin..., 63AL lib: 
pulseaudio.c:376: starting pa_threaded_mainloop_lock()

) = 63
write(2, AL lib: pulseaudio.c:380: startin..., 52AL lib: 
pulseaudio.c:380: starting pa_context_new()

) = 52
open(/home/kelly/.pulse/client.conf, O_RDONLY) = -1 ENOENT (No such 
file or directory)

open(/etc/pulse/client.conf, O_RDONLY) = 6
fstat(6, {st_mode=S_IFREG|0644, st_size=1184, ...}) = 0
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) 
= 0x7f07b833e000

read(6, # This file is part of PulseAudio..., 4096) = 1184
read(6, ..., 4096)= 0
open(/home/kelly/.pulse-cookie, O_RDWR|O_CREAT|O_NOCTTY, 0600) = 7
fcntl(7, F_SETLKW, {type=F_WRLCK, whence=SEEK_SET, start=0, len=0}) = 0
read(7, 
d\321\303\334\277\346B\370\217!7\367v.H\261\273\212\222\330))=k\360q\24g\3...@e..., 
256) = 256

fcntl(7, F_SETLKW, {type=F_UNLCK, whence=SEEK_SET, start=0, len=0}) = 0
close(7)= 0
close(6)= 0
munmap(0x7f07b833e000, 4096)= 0
socket(PF_FILE, SOCK_STREAM, 0) = 6
connect(6, {sa_family=AF_FILE, path=@/tmp/.X11-unix/X0...}, 20) = 0
getpeername(6, {sa_family=AF_FILE, path=@/tmp/.X11-unix/X0\22...}, 
[139668041498644]) = 0

uname({sys=Linux, node=bs, ...})= 0
access(/home/kelly/.Xauthority, R_OK) = 0
open(/home/kelly/.Xauthority, O_RDONLY) = 7
fstat(7, {st_mode=S_IFREG|0600, st_size=581, ...}) = 0
mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) 
= 0x7f07b833e000
read(7, \0\0\0\4\300\250\1\7\0\0010\0\22MIT-MAGIC-COOKIE-1\0\20..., 
4096) = 581

read(7, ..., 4096)= 0
close(7)= 0
munmap(0x7f07b833e000, 4096)= 0
getsockname(6, {sa_family=AF_FILE, path=@/tmp/.X11-unix/X0\22...}, 
[139668041498626]) = 0

fcntl(6, F_GETFL)   = 0x2 (flags O_RDWR)
fcntl(6, F_SETFL, O_RDWR|O_NONBLOCK)= 0
fcntl(6, F_SETFD, FD_CLOEXEC)   = 0
poll([{fd=6, events=POLLIN|POLLOUT}], 1, -1) = 1 ([{fd=6, revents=POLLOUT}])
writev(6, [{l\0\v\0\0\0\22\0\20\0\0\0..., 12}, {..., 0}, 
{MIT-MAGIC-COOKIE-1..., 18}, {\0\0..., 2}, 
{lfn\236\303N\331\370\32,\271y\213b;..., 16}, {..., 0}], 6) = 48

read(6, \1\0\v\0\0\0c\0..., 8)= 8
read(6, 
\230P\243\0\0\0\340\3\377\377\37\0\0\1\0\0\24\0\377\377\1\7\0\0  
\10\377\0\0\0\0T..., 396) = 396

poll([{fd=6, events=POLLIN|POLLOUT}], 1, -1) = 1 ([{fd=6, revents=POLLOUT}])
writev(6, [{b\0\5\0\f\0\0\0BIG-REQUESTS..., 20}], 1) = 20
poll([{fd=6, events=POLLIN}], 1, -1)= 1 ([{fd=6, revents=POLLIN}])
read(6, 
\1\0\1\0\0\0\0\0\1\216\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0..., 
4096) = 32

poll([{fd=6, events=POLLIN|POLLOUT}], 1, -1) = 1 ([{fd=6, revents=POLLOUT}])
writev(6, [{\216\0\1\0..., 4}], 1)= 4
poll([{fd=6, events=POLLIN}], 1, -1)= 1 ([{fd=6, revents=POLLIN}])
read(6,

Re: [pulseaudio-discuss] PulseAudio 0.9.21 deadlocking when accessed through OpenAl 1.10.622 (AlienArena)

2009-12-23 Thread Kelly Anderson 

On 12/23/2009 01:56 AM, Arun Raghavan wrote:

On Wed, 2009-12-23 at 01:21 -0700, Kelly Anderson wrote:
   

AlienArena was deadlocking trying to go through OpenAl which was set to
use PulseAudio.  After quite a bit of debugging (entering prints in
OpenAl pulse code), I finally found a solution with the help of a SUSE
bug report.

http://lists.opensuse.org/opensuse-bugs/2009-10/msg10497.html

which points at this commit.

http://git.0pointer.de/?p=pulseaudio.git;a=commitdiff;h=4f5e2b745ea357e2b5c815ff33a556505a7d1f18

 

Are you using openal-soft? If yes, did you try the fix that was
mentioned on the bug you referred to? It could be this one (but the
dates do not match):

http://repo.or.cz/w/openal-soft.git/commit/0e1e8503e0f6fb73d06ba0dff6e5b2771b1df856
   
If it is that commit, they've restructured their code in 
context_state_callback and stream_state_callback.  That patch will not 
apply to OpenAl 1.10.622.  I'm not too excited about using repository 
code (although I use it when necessary).


IMHO, the pulseaudio commit looks like a suspicious/dangerous optimization.


-- Arun

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Re: [pulseaudio-discuss] PulseAudio 0.9.21 deadlocking when accessed through OpenAl 1.10.622 (AlienArena)

2009-12-23 Thread Kelly Anderson 

On 12/23/2009 03:04 AM, Colin Guthrie wrote:

'Twas brillig, and Kelly Anderson at 23/12/09 08:21 did gyre and gimble:
   

AlienArena was deadlocking trying to go through OpenAl which was set to
use PulseAudio.  After quite a bit of debugging (entering prints in
OpenAl pulse code), I finally found a solution with the help of a SUSE
bug report.

http://lists.opensuse.org/opensuse-bugs/2009-10/msg10497.html

which points at this commit.

http://git.0pointer.de/?p=pulseaudio.git;a=commitdiff;h=4f5e2b745ea357e2b5c815ff33a556505a7d1f18


Here's an strace (with prints sprinkled in the OpenAl pulse code) which
demonstrates the deadlock.  A patch to undo the commit is at the end.
 

Does
http://git.0pointer.de/?p=pulseaudio.git;a=commit;h=978d33b609969c3b9bbbd759e0f11aaf856c80cf

Fix the problem?

   
volatile does not fix the problem.  It still deadlocks unless I revert 
the original commit.

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