[Libav-user] Extracting samples from audio file

Philippe Gorley philippe.gorley at savoirfairelinux.com
Wed Aug 16 00:06:44 EEST 2017


On 2017-08-15 04:40 PM, Paul B Mahol wrote:
> On 8/15/17, Philippe Gorley <philippe.gorley at savoirfairelinux.com> wrote:
>> 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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>>
>> I'm still struggling with getting this to work. My code is here:
>> https://pastebin.com/arzUw2za
>>
>> 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.
>>
>> Notes:
>> 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?
> 
> needResamping is triggered only when same sample rate is both ways.
> Are you sure that swr resample code works that way, you can not
> guarantee it will
> give output frame for each input frame. There are nice swr examples in
> repo, Have you looked at them?

Even when commenting out the else block and forcing the resampling, I 
get the problem. But yes, that might be a future problem. Thanks for 
flagging it.

I have looked at docs/examples/resampling_audio.c (but does not use 
AVFrame), the code in libswresample, and have read the docs (at least, 
whatever I could find).

Maybe I should use swr_convert instead of swr_convert_frame and directly 
append the out samples to my vector?

> 
>>
>> Thanks,
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> Libav-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/libav-user
> 

Thanks,

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

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