[mythtv] "Extra Audio Buffering" setting

Mark Spieth mark at digivation.com.au
Fri Nov 10 00:22:03 UTC 2017

On 10/11/17 08:26, John P Poet wrote:
> On Thu, Nov 9, 2017 at 2:20 PM Peter Bennett <pb.mythtv at gmail.com 
> <mailto:pb.mythtv at gmail.com>> wrote:
>     Hi Devs
>     There is a Frontend General Playback setting called "Extra audio
>     buffering" which mentions certain types of tuners that cause crackly
>     audio. Actually it enables setting DecodeExtraAudio which then maybe
>     sets a flag called lowbuffers in the demuxer (class AvFormatDecoder).
>     There is lots of complicated logic around whether or not that flag is
>     set and in some cases its value is saved and reset.
>     The lowbuffers flag is only tested once in the system and if set the
>     program buffers 100ms of video packets in
>     AvFormatDecoder::GetFrame, so
>     that audio is not left behind.
>     Bug #13172 describes how playback becomes extremely jerky if you
>     adjust
>     audio sync negatively more than a small amount. The solution is very
>     simple, just buffer more than 100ms in AvFormatDecoder::GetFrame. I
>     tested with 1000ms and then you can successfully set audio sync to
>     somewhere around -1200 before it becomes jerky. In the patch for
>     the fix
>     I settled on a figure of 250 as a good default and added for a setting
>     called AudioReadAhead for the value.
>     I would also like to get rid of the complex logic and flag surrounding
>     the DecodeExtraAudio flag and replace that setting in frontend setup
>     with AudioReadAhead with a default value of 250. With a value of
>     250 or
>     1000 it works on everything I have tried. There is no adverse impact
>     that I can see, although it will take some extra memory for the
>     additional buffering. I do not know why there is all the extra logic
>     around whether or not to set the flag, it is safe and easy to set
>     it all
>     the time.
>     I propose a setting that specifies number of milliseconds of audio
>     read-ahead with an explanation that says to increase this value if the
>     audio is jerky especially when adjusting audio sync negative.
>     Any comments?
>     Peter
> I have not looked at any of this code, but my guess is that it was to 
> handle frame-grabber cards, where Myth had to mux the audio and video 
> together itself.  We pretty much don't support those anymore (using 
> one in this age is just asking for pain).
> If you want to play it safe, create a branch and ask as many people as 
> possible to try it.  I will.
> However, if it is working well for you, even with your RPi units, then 
> I expect it to be fine.
> John
Beware of and test with mythmusic and audio only streams when you change 

Note that audio is rendered as it is seen in the decoded stream whereas 
video is not.
Audio only for mythmusic has as special shortcut that limits the buffer 
to 500ms or less so that when you seek or change streams the response is 

You are correct in that the audio buffer size directly relates to the 
amount of -ve sync achievable. This is a design constraint.

Also some OTA streams have largely varying A to V time gaps on PTS and 
the buffer is also required to absorb this. mkv's can also be quite 
un-aligned Ive found.

If you enable some of the more fancy features in audio land such as 
ch->6ch upsampling and AC3 reencoding, you will also chew up the buffer. 
The upsampling is especially a buffer hog partially due to the LPF for 
the subwoofer channel. I have noticed this is quite long when I 
start/skip for audio to catch up (600+ms).

Changing the extra buffering to a number in settings is ok but you need 
to add some help text.
That said, I have patches to change mine to 1.5 secs as default audio 
buffer size because the existing is inadequate for some of my media.

As an aside, there is also a bug in passthru mode in that if the rate 
does not match the standard AC3 rate, then you get no sound out of your 
digital amp as the AC3/DTS stream is not framed correctly due to 
differing bitrates on the SPIDF line. The workaround is to enable 
timestretch as this reencodes correctly and then all sound is good. This 
happens on most of our (AU) HD channels.

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