[Libav-user] Problem with compressing AAC files with sampling rate 16000

Claudio Freire klaussfreire at gmail.com
Thu Sep 18 15:41:08 CEST 2014

On Thu, Sep 18, 2014 at 9:43 AM, adev dev <androiddevmar11 at gmail.com> wrote:
> I am compressing movies from bitmaps and AAC files. Normally AAC files are
> recorded with following params: bit rate: 192000, sampling rate 44100. These
> audio params are set in output context (AVFormatContext) in compression. In
> happy day scenario it is working correctly. Movie and sound are correct.
> The problem occures if sound params are "strange". From time to time I have
> AAC files recorded with params: bitrate 96000, sampling rate 16000. In this
> case sound is very fast and ends before end of movie. If I set these params
> (96000, 16000) in output context there is no sound in the movie at all.
> Created movie file has audio stream with bitrate about 500 b/s what is
> really strange.
> What else has to be set in FFMPEG to correctly compress AAC files with bit
> rate 96000 and sampling rate 16000?
> I am using FFMPEG 2.1 and native AAC encoder.

If you can build from source, try head after applying the patch in
here: https://trac.ffmpeg.org/attachment/ticket/2686/aac-improvements-wip-v8g.patch

It fixes some issues with some sampling rates, although your issue
doesn't sound familiar to me, it's worth a try.

More information about the Libav-user mailing list