[gst-devel] Audio Playback

[Wim Taymans]

On Sat, 2005-06-25 at 20:13 +0100, Matt Morten wrote:
> Try this:
> 
> Link a "filesrc" plugin with a regular CBR MP3 loaded, to a "mad"
> plugin, to an "alsasink". Then, experiment with playing and pausing
> the elements in the pipeline.
> 
> Does anyone else notice that after pausing, playing the pipeline
> causes slight skips in the audio - usually ahead by a beat or so?

Samples sent to the device are not kept in the element anymore and on
PAUSE the device buffer is flushed...

> 
> I am trying to build a DJ application, and I'm a little stuck because
> of this. It is very difficult to DJ without being certain that the
> audio stream can play at *exactly* the same point it was paused at.
> 
> I'm using Gstreamer 0.8 (I would love to try 0.9, but it seems the 3
> plugins I need arent ported yet), with the stock 2.6.12 kernel.

The 3 plugins are ported allright. Current 0.9 alsasink does not flush
the audiobuffer on PAUSED (it should) so it currently never skips a
sample. The only problem is that audio continues to play after PAUSE for
the amount of queued samples in the device. With a sufficiently small
segment size this time can be reduced to a few milliseconds. 

Then there are also the alsa functions to pause/resume the device which
have not been tested yet in GStreamer AFAIK...

> 
> I've also been playing around with some of the buffer settings in the
> "alsasink" - none of which seem to make a difference.

making the period-size and buffer-size smaller should give you less
samples dropped in 0.8.

hope this helps,

Wim 

> 
> Can anyone help?
> Thanks in advance
> Matt
> 
> 
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-- 
Wim Taymans <wim at fluendo.com>