[Pkg-alsa-devel] Bug#643936: libasound2-plugins: the a52 module, filter fails to convert pcm stream to ac3

Luis Listas lfl at soeiro.com.br
Wed Oct 12 19:50:31 UTC 2011


* riesebie at lxtec.de [111010 21:37 -+0200]:

> Adopted in to Debian svn. At the Moment we get:
>
> E: lib64asound2-plugins: shlib-without-PT_GNU_STACK-section usr/lib64/alsa-lib/libasound_module_ctl_arcam_av.so
> E: lib64asound2-plugins: shlib-without-PT_GNU_STACK-section usr/lib64/alsa-lib/libasound_module_ctl_oss.so
> E: lib64asound2-plugins: shlib-without-PT_GNU_STACK-section usr/lib64/alsa-lib/libasound_module_pcm_oss.so
> E: lib64asound2-plugins: shlib-without-PT_GNU_STACK-section usr/lib64/alsa-lib/libasound_module_pcm_upmix.so
> E: lib64asound2-plugins: shlib-without-PT_GNU_STACK-section usr/lib64/alsa-lib/libasound_module_pcm_usb_stream.so
> E: lib64asound2-plugins: shlib-without-PT_GNU_STACK-section usr/lib64/alsa-lib/libasound_module_pcm_vdownmix.so
> E: lib64asound2-plugins: shlib-without-PT_GNU_STACK-section usr/lib64/alsa-lib/libasound_module_rate_speexrate.so
>
> from lintian and i don't know how to solve this yet.
>
> @Jordi: Any idea?
>

It seems it works with those few changes. I've downloaded the source
deb file (apt-get source libasound2-plugins), then I manually patched
a few lines in a52/pcm_a52.c using:

http://git.alsa-project.org/?p=alsa-plugins.git;a=commitdiff;h=40c129a160f37fe9488b2828d6299f99c269703e;hp=06ca71522f1dc48b8503d945da81fdbcbab0aafa

It failed to compile because of missing files:

/usr/lib/pkgconfig/libpulse.pc
/usr/lib/pkgconfig/jack.pc

I've then changed debian/rules to point instead to:

/usr/lib/x86_64-linux-gnu/pkgconfig/libpulse.pc
/usr/lib/x86_64-linux-gnu/pkgconfig/dbus-1.pc
/usr/lib/x86_64-linux-gnu/pkgconfig/jack.pc

Finally, I compiled and packaged it with:

dpkg-buildpackage -rfakeroot -uc -b

And then installed with:

dpkg -i libasound2-plugins_1.0.24-2_amd64.deb

Those patches seem to solve the problem, because now a52 is working again:

$ speaker-test -Dhometheater -c6 -twav

speaker-test 1.0.24.2

Playback device is hometheater
Stream parameters are 48000Hz, S16_LE, 6 channels
WAV file(s)
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 3072 to 15360
Period size range from 1536 to 1536
Using max buffer size 15360
Periods = 4
was set period_size = 1536
was set buffer_size = 15360
  0 - Front Left
  4 - Center
  1 - Front Right
  3 - Rear Right
  2 - Rear Left
  5 - LFE
Time per period = 8,299085
  0 - Front Left

Since I could barely understand what I was doing (trial and error),
I hope you guys can fix this definitely for everybody else.

Thanks,
Luis






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