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I'm attempting to convert an AAC audio stream for playback. I've discovered that I need to convert from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16 but when I do so the audio plays back at about half speed. 

swr = swr_alloc();
assert(av_opt_set_int(swr, "in_channel_layout", audioContext->channel_layout, 0) == 0);
assert(av_opt_set_int(swr, "out_channel_layout", audioContext->channel_layout, 0) == 0);
assert(av_opt_set_int(swr, "in_sample_rate", audioContext->sample_rate, 0) == 0);
assert(av_opt_set_int(swr, "out_sample_rate", 44100, 0) == 0);
assert(av_opt_set_int(swr, "in_sample_fmt", audioContext->sample_fmt, 0) == 0);
assert(av_opt_set_int(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0) == 0);
swr_init(swr);

There is my code to convert. The input sample rate is 44100 and the audio is stereo.

I call the code with 

swr_convert(swr, &output, aDecodedFrame->nb_samples, (const uint8_t**)aDecodedFrame->extended_data, aDecodedFrame->nb_samples) >= 0)
AnthonyM
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1 Answers1

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You didn't show the actual audio encoding code, so I'd speculate there's a chance you might not handle the resampling properly. Note that you read twice less data from the resampling operation (i.e. if you pass 80 bytes, you'll read 40 from the resampler).

You may take a look at my video writing code, and strip off the audio encoding part. It is here: http://sourceforge.net/p/karlyriceditor/code/HEAD/tree/src/ffmpegvideoencoder.cpp

George Y.
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