[alsa-devel] What is the required accuracy of audio sampling rate?
yiliangb at gmail.com
Wed Feb 17 18:14:46 CET 2010
Thanks for all the replies. To reply Pierre's question:
> Can you be more specific on your sync issues? Generally the audio clock is
used as reference, and video adjusts to it.
I played the audio using gstreamer alsasrc and alsasink. The audio plays
well at the beginning, and gradually the sound becomes broken, and
eventually unlistenable. If I play the audio along with video, there is a
similar issue. In addition to the problem in the audio channel, the video
will become intermittent as well after some time. I think the gstreamer is
dropping video frames trying to synchronize the video with audio.
Similar to what is suggested here, I did some experiment on the audio driver
so that its output is synchronized to the kernel clock, and the problems are
gone (at least for the testing period of 2 hours). This confirms that the
issues were indeed caused by inaccuracy in audio sampling rate and it
appears gstreamer is not adaptive enough to accommodate such sampling error.
On Wed, Feb 17, 2010 at 5:03 AM, Jon Smirl <jonsmirl at gmail.com> wrote:
> On Wed, Feb 17, 2010 at 7:52 AM, James Courtier-Dutton
> <james.dutton at gmail.com> wrote:
> > On 16 February 2010 23:58, Yiliang Bao <yiliangb at gmail.com> wrote:
> >> Hi,
> >> I have a PCI-e audio card that does not give very accurate sampling rate
> >> (about -0.2% error, i.e., if I set the sampling rate to 8K, I got only
> >> samples per second on average). It caused various synchronization
> >> in applications like gstreamer.
> >> Since there will be some difference from one clock domain to another, I
> >> wondering if there is any defined tolerance on the sampling rate, and
> >> the right place is to correct this kind of error. I would really
> >> if someone can point me to the related documents/links.
> > There is no defined tolerance on sampling rate.
> > Audio apps should be able to deal with a certain amount of error.
> > xine uses the system clock for its synchronization, and resamples the
> > audio output if the audio hardware is running at a different rate.
> > Other apps might use the audio clock and sync to that. In that case,
> > it might get the problems you describe.
> > Try playing your media in xine and see if the problem disappears.
> The -0.2% error is cumulative, not random. After a three hour movie
> your audio will be off by 20 seconds.
> Best solution is the get the audio clock synched as closely as
> possible to the video one. Any remaining error has to be dealt with in
> software like xine does.
> > Kind Regards
> > James
> > _______________________________________________
> > Alsa-devel mailing list
> > Alsa-devel at alsa-project.org
> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
> Jon Smirl
> jonsmirl at gmail.com
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