[SDL] Using SDL_AudioCVT inside audio callback

[Gabriel Gambetta]

Yeah, replying to my own question. I moved the resampling code out of the
callback and into the audio decoder and everything works fine now.

--Gabriel

On Wed, Apr 23, 2008 at 9:02 PM, Gabriel Gambetta <mystml at adinet.com.uy>
wrote:

> I'm integrating the dranger example FFMPEG player with SDL_Mixer and other
> things. I got everything working, except one thing. I open the mixer at
> 22KHz and the AVI provides audio at 44KHz. To compensate for this, I added
> an SDL_AudioCVT which is supposed to do the right thing. The old callback
> looked like this (simplified) :
>
> void audio_callback (void *userdata, Uint8 * stream, int len)
> {
>     while (len > 0)
>     {
>         // Read audio data from the AVI and synchronize to video
>         readAudioData();
>
>         len1 = is->audio_buf_size - is->audio_buf_index;
>         if (len1 > len)
>             len1 = len;
>
>         memcpy (stream, (uint8_t *) is->audio_buf + is->audio_buf_index,
> nLen1);
>
>         len -= len1;
>         stream += nLen1;
>         is->audio_buf_index += len1;
>     }
> }
>
> The 44KHz audio played at 22KHz sounded slow, as expected. So I did this :
>
> void audio_callback (void *userdata, Uint8 * stream, int len)
> {
>     while (len > 0)
>     {
>         // Read audio data from the AVI and synchronize to video
>         readAudioData();
>
>         len1 = is->audio_buf_size - is->audio_buf_index;
>         if (len1 > len)
>             len1 = len;
>
>         int nOut = len1;
>         if (is->bConvertAudio)
>         {
>             is->pAudioCVT.buf = is->audio_buf + is->audio_buf_index;
>             is->pAudioCVT.len = len1;
>             SDL_ConvertAudio(&is->pAudioCVT);
>             nOut = len1/2;
>         }
>
>         memcpy (stream, (uint8_t *) is->audio_buf + is->audio_buf_index,
> nOut);
>
>         len -= len1;
>         stream += nOut;
>         is->audio_buf_index += len1;
>     }
> }
>
> I did nOut = len1/2 because of the sampling rate difference. But with that
> or without it, I get sound at the correct speed, but sounding skippy, as if
> I was fulling only half of the buffer each time (as far as I can tell, I'm
> filling the whole buffer)
>
> Am I doing something obviously wrong here?
>
> Thanks,
> --Gabriel
>
>
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