[Libav-user] How to get raw audio frames from an audio file

[Finalspace]

Hi,

the example you are following just writes the frame out in raw format, 
without doing any kind of sample rate or format conversion.

To convert a sample you need to check against the sample format 
(dec_ctx->sample_fmt) and implement the right conversion.
For example if your audio stream uses a sample format of S16 (Signed 
16-bit integer), converting that to float is really easy (Just divide by 
max int16):

     int sampleStride = av_get_bytes_per_sample(dec_ctx->sample_fmt);
     for (int sampleIndex = 0; sampleIndex < frame->nb_samples; 
++sampleIndex) {
         for (int channel = 0; channel < dec_ctx->channels; ++channel) {
           // For signed 16-bit integer to float conversion
             if (dec_ctx->sample_fmt == AV_SAMPLE_FMT_S16) {
                 int16_t *inputPtr = (int16_t *)(frame->data[ch] + 
sampleStride * sampleIndex);
                 int16_t inputSample = *inputPtr;
                 float outputSample;
                 if (inputSample < 0) {
                     outputSample = inputSample / (float)(INT16_MAX - 1);
                 } else {
                     outputSample = inputSample / (float)INT16_MAX;
                 }
             }
         }
     }

Hope this helps.

Greetings,
Final

Am 20.10.2018 um 11:30 schrieb Matthieu Regnauld:
> Hello,
>
> I try to extract raw audio frames from an audio file, using the 
> following example: 
> http://ffmpeg.org/doxygen/3.4/decode_audio_8c-example.html
>
> As far as I understand, I get these frames in the decode() function, 
> in the frame->data array.
>
> That said, I need to convert the frames into floats between -1 and 1. 
> Here is what I tried (in my test, dataSize == 4):
>
> // here is where I want to get my frame:
> float myFrame;
>
> // first try:
> memcpy(&myFrame, &frame->data[ch][i], dataSize * sizeof(uint8_t));
>
> // second try:
> myFrame = (frame->data[ch][i]<<0) | (frame->data[ch][i + 1]<<8) | 
> (frame->data[ch][i + 2]<<16) | (frame->data[ch][i + 3]<<24);
>
> // third try:
> myFrame = (frame->data[ch][i + 3]<<0) | (frame->data[ch][i + 2]<<8) | 
> (frame->data[ch][i + 1]<<16) | (frame->data[ch][i]<<24);
>
> But the problem is; it doesn't work, all I got so far is some kind of 
> white noice.
>
> So what is the proper way to extract audio frames and convert them to 
> float amplitudes?
>
> Thanks for your help.
>
>
>
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> Libav-user at ffmpeg.org
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