[Libav-user] Extracting samples from audio file

Philippe Gorley philippe.gorley at savoirfairelinux.com
Tue Aug 15 22:58:23 EEST 2017

On 2017-08-10 01:17 PM, salsaman wrote:
> Correct, you would first create the swr_context then use it to convert 
> the data. 
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> Libav-user mailing list
> Libav-user at ffmpeg.org
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I'm still struggling with getting this to work. My code is here: 

The commented out code at the bottom is the old Sndfile code, which I'd 
like to replace using FFmpeg. What it does is read all the audio samples 
into an array of int16_t. NB: Sndfile calls them frames, while FFmpeg 
calls them samples.

Right now, my code gives me low volume static.

I'm setting up the decoding pipeline, it (seems to, at least) works. I'm 
getting the same number of samples as Sndfile does. I'm guessing the 
problem lies with the resampling code.

I'm calling swr_config_frame and swr_convert_frame for each decoded 
frame, loop through the output frame's extended_data[0] and append those 
samples to an std::vector.

AudioSample is an alias for int16_t.
AudioBuffer is a container with an std::vector<std::vector<AudioSample>> 
(one for each channel).
AudioFormat is a POD struct with the sample rate and number of channels.

Can anyone look at this and tell me what I'm doing wrong?


Philippe Gorley
Free Software Consultant | Montréal, Qc
Savoir-faire Linux

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