On 2017-08-15 04:40 PM, Paul B Mahol wrote:
On 8/15/17, Philippe Gorley <philippe.gor...@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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Thanks,
--
Philippe Gorley
Free Software Consultant | Montréal, Qc
Savoir-faire Linux
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