Sam Lantinga wrote:
> I'm attaching the latest versions of the SDL audio files, and I'd
> really appreciate it if you take a look and sanity check our code.
> device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
ALSA's devices look at certain environment variables; you are supposed
to use names like "default" or "surround51" to get that default
configuration.
> if (channels == 6) device = "surround51";
> else if (channels == 4) device = "surround40";
The devices do not have automatic sample format/rate conversion; you
might want to use "plug:surround.." instead.
> /* This function waits until it is possible to write a full sound buffer */
> static void ALSA_WaitAudio(_THIS)
This function doesn't actually wait ...
> if ( status == -EAGAIN ) {
> SDL_Delay(1);
> continue;
> }
Not necessary in blocking mode.
> rate = spec->freq;
> status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL);
> spec->freq = rate;
The returned rate could be wildly different, but I guess SDL correctly
handles the new value in spec->freq.
> /* Set the buffer size, in samples */
> frames = spec->samples;
> status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL);
This does _not_ set the buffer size.
> periods = 2;
> SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL);
A bigger number of periods would make the writing of audio data less
bursty.
> status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 0);
Zero doesn't really make sense; the default value 1 would be OK.
> status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, frames);
The default value is the period size anyway, so you can remove this.
(If you change the buffer size code above to use _set_buffer_size_near,
this call would be wrong.)
Best regards,
Clemens
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