[Libav-user] Converting audio sample buffer format

Brad O'Hearne brado at bighillsoftware.com
Tue Feb 26 02:54:04 CET 2013

On Feb 25, 2013, at 12:11 PM, Brad O'Hearne <brado at bighillsoftware.com> wrote:

> However, the little nugget of info this dished out on next run might be enough to give a foothold to find the problem. It appears that the offending line that is crashing is a call to
> swri_realloc_audio
> inside of 
> swr_convert.
> I've attached a small image that shows this part of the stack trace in Xcode. Does that spark any theories by the FFmpeg gurus out there?

In lieu of being at a bit of a standstill for getting this resampling to work, I thought I might tap the audio gurus out there to expound a little bit on what your best guess is on how a buffer would be laid out to accommodate a sample format as the one I am being handed from capture: 

- Linear PCM, 24 bit little-endian signed integer, 2 channels, 44100 Hz

I understand this is the QTSampleBuffer byte buffer, which is QTKit (and FWIW I have a question out on the QuickTime API mailing list about this too), but perhaps with the audio knowledge on this list you can speculate on a structure for that data, which I can verify in code. From there, perhaps with a little more guidance on how AV_SAMPLE_FMT_S16 samples are laid out, I can just write my own resampling algorithm. 

Any guidance is very welcome.



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