On Feb 18, 2013, at 3:50 PM, Carl Eugen Hoyos <[email protected]> wrote:
>> Note that the flv codec
>
> While I have _no_ idea what the "flv audio codec" could
> be, please use either the aconvert filter or libswresample
> directly to convert from one audio format to another.
Perhaps this was a misuse of terms, but my intent in specifying that was to
indicate the audio codec which is either tethered to or influenced by the flv
video codec or stream. When creating an output context to address flv format as
such:
const char *cFileNameExt = [@"flv" UTF8String];
const char *cMimeType = [@"video/x-flv" UTF8String];
_avOutputFormat = av_guess_format(cStreamName, cFileNameExt, cMimeType);
If I then create an audio stream on the associated output format (context), any
attempt to open an audio codec with a sample format other than
AV_SAMPLE_FMT_S16 fails. That would lead me to believe that something about
having an FLV output format is restricting the audio codec to that sample
format -- otherwise, I would think that any sample format could be used. That
may or may not be true, but that is what my meaning was when saying "flv audio
codec"...I should have probably said the "sample format compatible with flv".
That said, thank you for your recommendation. I have found the
resampling_audio.c in the FFmpeg source examples which appear to address
libswresample, and I have found the filter_audio.c -- it would appear to be an
aconvert example -- but if there's more doc somewhere on aconvert, I'd be
interested in knowing where....that would be great. One final question -- what
essentially is the difference in approach between aconvert filter and
libswresample, and why would I want to use one over the other? Does it matter?
Thanks,
Brad
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