Yes, you need to buffer sufficient audio frames to feed to the encoder.
Calculate the number of in samples:
/* compute src number of samples */
in_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) +
out_nb_samples,
in_sample_rate, c->sample_rate, AV_ROUND_DOWN);
then allocate buffers to concatenate the in samples until you have enough
to pass to swr_ctx.
Gabriel.
http://lives-video.com
https://www.openhub.net/accounts/salsaman
On Tue, Jul 25, 2017 at 10:37 PM, Andy Shaules <[email protected]> wrote:
>
>
> On Jul 24, 2017 5:44 AM, "Kerry Loux" <[email protected]> wrote:
>
>
>
> On Fri, Jul 21, 2017 at 11:38 PM, Andy Shaules <[email protected]>
> wrote:
>
>>
>>
>> On Jul 19, 2017 10:59 AM, "Kerry Loux" <[email protected]> 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
>>
>> _______________________________________________
>> Libav-user mailing list
>> [email protected]
>> http://ffmpeg.org/mailman/listinfo/libav-user
>>
>>
>>
>> 2 things perhaps. what sample rate does your sdp advertize? what scale
>> are your audio rtp timstamps?
>>
>> _______________________________________________
>> Libav-user mailing list
>> [email protected]
>> http://ffmpeg.org/mailman/listinfo/libav-user
>>
>> SDP correctly advertises 16000 Hz and my timestamps are scaled to
> microseconds.
>
> Thanks,
>
> Kerry
>
> _______________________________________________
> Libav-user mailing list
> [email protected]
> http://ffmpeg.org/mailman/listinfo/libav-user
>
> then yes, most likely needing to concatenate frames to the currect
> duration for the sample rate
>
> _______________________________________________
> Libav-user mailing list
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> http://ffmpeg.org/mailman/listinfo/libav-user
>
>
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