rtpbin: ts adjustment needs filtering/smoothing

[charleslaub at sbcglobal.net]

>Perhaps you didn't notice the default slave-method, which is skew. Skew will remove or fill with silence the audio stream to fix the synchronization >between the audio card clock and the NTP clock. As it does that naively, you will ear ticks and pop. You'll also notice the resample method, which in >theory should work, but in practice the resampler there is buggy and the adjustment too aggressive. Some work is needed there to make it work.

As you say "buggy" is an understatement - it is TERRIBLE. It's essentially unlistenable because the "over aggressive" action introduces a warble to the music, like you are listening underwater.  

So, the only solution that comes to mind is custom slaving, which is not available to the general public? Hmmmm.

Perhaps in place of the buggy ALSA resampler, you could implement asynchronous resampling? This was brought up in a post a couple of days ago in this list under the subject "Adaptive resampling in gstaudiobasesink.c". I think it would be very worthwhile and would really improve the audio performance of Gstreamer in general.

Could you or someone follow up on that topic? 

-Charlie

-----Original Message-----
From: gstreamer-devel <gstreamer-devel-bounces at lists.freedesktop.org> On Behalf Of Nicolas Dufresne
Sent: Saturday, October 17, 2020 10:50 AM
To: Discussion of the development of and with GStreamer <gstreamer-devel at lists.freedesktop.org>
Subject: Re: rtpbin: ts adjustment needs filtering/smoothing

Perhaps you didn't notice the default slave-method, which is skew. Skew will remove or fill with silence the audio stream to fix the synchronization between the audio card clock and the NTP clock. As it does that naively, you will ear ticks and pop. You'll also notice the resample method, which in theory should work, but in practice the resampler there is buggy and the adjustment too aggressive. Some work is needed there to make it work.

Well known companies using GStreamer have introduced the "custom"
slave-method. Along with
gst_audio_base_sink_set_custom_slaving_callback(), you get notified per units of the drift. These vendor controlls the HW, so instead of resampling, they will adjust the audio clock PLL in order to lock the audio to the system/ntp/ptp/etc. clock in use.
>  
> I would be happy to provide my pipelines if that would be helpful, 
> provide measurement data, or any other info that might lead to a 
> solution.
>  
> -Charlie
>  
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> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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