[Libav-user] Resample frame to specified number of samples

[Kerry Loux]

On Fri, Jul 21, 2017 at 11:38 PM, Andy Shaules <bowljoman at gmail.com> wrote:

>
>
> On Jul 19, 2017 10:59 AM, "Kerry Loux" <louxkr at gmail.com> wrote:
>
> Hello all,
>
> I have an application where I am opening an audio file that was sampled at
> 44100 Hz, decoding it, resampling to 16000 Hz, encoding it again (AAC) then
> broadcasting it on an RTSP stream.  On the receiving end, I decode the
> incoming AAC packets and render them.
>
> The rendered audio is very slow.
>
> It appears to me that the problem is related to the AVFrame.nb_samples
> field.  When I read a packet from file (using av_read_frame()), the packet
> size is 1024 samples (at 44100 Hz).  After I resample to 16000 Hz, I have
> ~1/3 the samples that I had in the original frame (as expected).  Then, the
> frame gets encoded, streamed and decoded.  After decoding, the
> AVFrame.nb_samples is 1024 when I expect it to be 372 or so.  The
> AVCodecContext passed to avcodec_receive_frame() has frame_size = 1024, so
> I assume that the decoder is setting the number of samples of the decoded
> frame to 1024 regardless of the number of samples actually contained in the
> input packet?  Or maybe it's my job to ensure that the input packets always
> contain 1024 samples?
>
> I'm not entirely sure what's going on.  My thoughts include:
> - Try buffering 3x number of input frames prior to resampling so the
> resulting frame will be ~1024 samples
> - Calculate the number of samples manually (how to do this is unclear) and
> override the number of samples assigned by the decoder (this seems wrong...)
>
> Any recommendations?  Can I just stick multiple frames together in a
> larger buffer prior to resampling (i.e. calling swr_convert())?
>
> Thanks,
>
> Kerry
>
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>
>
> 2 things perhaps. what sample rate does your sdp advertize? what scale are
> your audio rtp timstamps?
>
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> SDP correctly advertises 16000 Hz and my timestamps are scaled to
microseconds.

Thanks,

Kerry
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