void sound_play() {
#if defined(D_SOUND_OSS) || defined(D_SOUND_ALSA) || defined(D_SOUND_PULSE)
static int retval,retval2;
#endif
ordenador.current_buffer=sound[0];
//remove_dc(ordenador.current_buffer,ordenador.buffer_len);
switch(sound_type) {
case SOUND_NO: // no sound
usleep(75000); // wait 1/20 second
return;
break;
case SOUND_SDL: // SDL
if (!started_sound_sdl) {
SDL_PauseAudio(0);
ordenador.current_buffer = sound[0];
buffer0_occupied=1;
started_sound_sdl = 1;
buffer_reading=0;
}
//Double buffer
while ((buffer0_occupied)&&(buffer1_occupied)){usleep(1000);}; //Wait for one buffer to be free
if (!buffer0_occupied) //Buffer 0 is now free
{buffer0_occupied=1;
ordenador.current_buffer = sound[0]; }
else //Buffer 1 is now free
{buffer1_occupied=1;
ordenador.current_buffer = sound[1]; }
return;
break;
#ifdef D_SOUND_OSS
case SOUND_OSS: // OSS
retval=write(audio_fd,ordenador.current_buffer,ordenador.buffer_len*ordenador.increment);
return;
break;
#endif
#ifdef D_SOUND_ALSA
case SOUND_ALSA: // ALSA
if (started_sound==0) {
snd_pcm_prepare (_soundDevice);
started_sound=1;
}
retval=snd_pcm_writei(_soundDevice,sound[0],ordenador.buffer_len);
if (retval<0) {
retval=snd_pcm_prepare(_soundDevice);
}
return;
break;
#endif
#ifdef D_SOUND_PULSE
case SOUND_PULSEAUDIO: // PulseAudio
// Remove the DC component to avoid losing the sound when multiplexing with other sources
sound_remove_dc(ordenador.current_buffer,ordenador.buffer_len);
retval=pa_simple_write(pulse_s,sound[0],ordenador.buffer_len,&retval2);
return;
break;
#endif
#ifdef GEKKO
case SOUND_ASND: // ASND
if (!started_sound_asnd) {
ASND_SetVoice(1,VOICE_STEREO_16BIT_BE,ordenador.freq,0,sound[0],ordenador.buffer_len*ordenador.increment,
255, 255, callback);
started_sound_asnd = 1;
}
//Double buffer
while (!ASND_TestVoiceBufferReady(1)){}; //Wait for one buffer to be free
if (!ASND_TestPointer (1, sound[0]))
{ASND_AddVoice(1,sound[0],ordenador.buffer_len*ordenador.increment);
ordenador.current_buffer = sound[0]; }
else
{ASND_AddVoice(1,sound[1],ordenador.buffer_len*ordenador.increment);
ordenador.current_buffer = sound[1]; }
return;
break;
#endif
default:
break;
}
}
static void * ogg_player_thread(private_data_ogg * priv)
{
int first_time = 1;
long ret;
//init
LWP_InitQueue(&oggplayer_queue);
priv[0].vi = ov_info(&priv[0].vf, -1);
ASND_Pause(0);
priv[0].pcm_indx = 0;
priv[0].pcmout_pos = 0;
priv[0].eof = 0;
priv[0].flag = 0;
priv[0].current_section = 0;
ogg_thread_running = 1;
while (!priv[0].eof && ogg_thread_running)
{
if (priv[0].flag)
LWP_ThreadSleep(oggplayer_queue); // wait only when i have samples to send
if (priv[0].flag == 0) // wait to all samples are sent
{
if (ASND_TestPointer(0, priv[0].pcmout[priv[0].pcmout_pos])
&& ASND_StatusVoice(0) != SND_UNUSED)
{
priv[0].flag |= 64;
continue;
}
if (priv[0].pcm_indx < READ_SAMPLES)
{
priv[0].flag = 3;
if (priv[0].seek_time >= 0)
{
ov_time_seek(&priv[0].vf, priv[0].seek_time);
priv[0].seek_time = -1;
}
ret
= ov_read(
&priv[0].vf,
(void *) &priv[0].pcmout[priv[0].pcmout_pos][priv[0].pcm_indx],
MAX_PCMOUT,/*0,2,1,*/&priv[0].current_section);
priv[0].flag &= 192;
if (ret == 0)
{
/* EOF */
if (priv[0].mode & 1)
ov_time_seek(&priv[0].vf, 0); // repeat
else
priv[0].eof = 1; // stops
}
else if (ret < 0)
{
/* error in the stream. Not a problem, just reporting it in
case we (the app) cares. In this case, we don't. */
if (ret != OV_HOLE)
{
if (priv[0].mode & 1)
ov_time_seek(&priv[0].vf, 0); // repeat
else
priv[0].eof = 1; // stops
}
}
else
{
/* we don't bother dealing with sample rate changes, etc, but
you'll have to*/
priv[0].pcm_indx += ret >> 1; //get 16 bits samples
}
}
else
priv[0].flag = 1;
}
