Re: [Libquicktime-devel] AAC audio

[Burkhard P. <bur...@ig...>]

Hi,

Am 07.10.2013 21:48, schrieb Erik Johansson:
> Hi,
[...]
>> AVCodecContext provides a field for audio sample format as follows:
>>      /**
>>       * audio sample format
>>       * - encoding: Set by user.
>>       * - decoding: Set by libavcodec.
>>       */
>>      enum AVSampleFormat sample_fmt;  ///< sample format
>> The audio decoder in libquicktimes ffmpeg plugin sets sample_fmt to 16bit signed, and
>> then relies on that being the case throughout the code. Only the latest ffmpeg resets
>> sample_fmt to float as per the documentation.

This was written at a time when all ffmpeg audio codecs used 16 bit samples. Nowadays it's
different. The behaviour of the core is to call the decode() function with output = NULL at
the beginning. During this initialization call, the member atrack->sample_format can be set
(according to what's in ffmpeg). In most codecs it's hardwired, but in audiocodec/pcm.c
you'll find a function init_decode_lpcm(), which does runtime detection of the sample format.

Making proper sample format detection and supporting planar audio in ffmpeg would be a very
patch to start with.

Burkhard