static int
irix_dsp_get_freespace(struct sysdep_dsp_struct *dsp)
{
struct irix_dsp_priv_data *priv = dsp->_priv;
return alGetFillable(priv->devAudio);;
}
// return: how many bytes can be played without blocking
static int get_space(void) {
// printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_data.outburst);
// printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
return alGetFillable(ao_port) * bytes_per_frame;
}
void static AL_WaitAudio(_THIS)
{
Sint32 timeleft;
timeleft = this->spec.samples - alGetFillable(audio_port);
if ( timeleft > 0 ) {
timeleft /= (this->spec.freq/1000);
SDL_Delay((Uint32)timeleft);
}
}
void static
IRIXAUDIO_WaitDevice(_THIS)
{
Sint32 timeleft;
timeleft = this->spec.samples - alGetFillable(this->hidden->audio_port);
if (timeleft > 0) {
timeleft /= (this->spec.freq / 1000);
SDL_Delay((Uint32) timeleft);
}
}
int audriv_write(char *buff, int n)
/* audio に buff を n バイト分流し込みます．
* audriv_set_noblock_write() で非ブロック・モードが設定された
* 場合は，この関数の呼び出しは即座に処理が返ります．
* 返り値は実際に流し込まれたバイト数であり，非ブロック・モードが設定
* されている場合は，引数 n より少ない場合があります．
* 失敗すると -1 を返し，成功すると，実際に流し込まれたバイト数を返します．
*/
{
n /= play_frame_width;
if(audio_write_noblocking)
{
int size;
size = alGetFillable(out);
if(size < n)
n = size;
}
add_sample_counter(n);
if(play_encoding != AENC_G711_ULAW)
{
alWriteFrames(out, buff, n);
return n * play_frame_width;
}
else
{
/* AENC_G711_ULAW */
int i, m, ret;
short samps[BUFSIZ];
n *= play_frame_width;
ret = n;
while(n > 0)
{
m = n;
if(m > BUFSIZ)
m = BUFSIZ;
for(i = 0; i < m; i++)
samps[i] = AUDIO_U2S(buff[i]);
alWriteFrames(out, samps, m / play_frame_width);
buff += m;
n -= m;
}
return ret;
}
}
static int
irix_dsp_write(struct sysdep_dsp_struct *dsp, unsigned char *data, int count)
{
struct irix_dsp_priv_data *priv = dsp->_priv;
int playcnt;
int maxsize;
/*
* We write as many samples as possible (up to count) without blocking
*/
maxsize = alGetFillable(priv->devAudio);
playcnt = (count <= maxsize) ? count : maxsize;
alWriteFrames(priv->devAudio, data, playcnt);
return playcnt;
}
void SNDDMA_Submit(void)
{
int nFillable, nFilled, nPos;
int nFrames, nFramesLeft;
unsigned endtime;
if (!sgisnd_aport) return;
nFillable = alGetFillable( sgisnd_aport );
nFilled = QSND_BUFFER_FRAMES - nFillable;
nFrames = dma.samples >> (dma.channels - 1);
if (paintedtime - soundtime < nFrames)
nFrames = paintedtime - soundtime;
if (nFrames <= QSND_SKID) return;
nPos = UST_TO_BUFFPOS( sgisnd_startframe );
// dump re-written contents of the buffer
if (sgisnd_lastframewritten > sgisnd_startframe)
{
alDiscardFrames( sgisnd_aport, sgisnd_lastframewritten - sgisnd_startframe );
}
else if ((int)(sgisnd_startframe - sgisnd_lastframewritten) >= QSND_BUFFER_FRAMES)
{
// blow away everything if we've underflowed
alDiscardFrames( sgisnd_aport, QSND_BUFFER_FRAMES );
}
// don't block
if (nFrames > nFillable) nFrames = nFillable;
// account for stereo
nFramesLeft = nFrames;
if (nPos + nFrames * dma.channels > QSND_BUFFER_SIZE)
{
int nFramesAtEnd = (QSND_BUFFER_SIZE - nPos) >> (dma.channels - 1);
alWriteFrames( sgisnd_aport, &dma_buffer[nPos], nFramesAtEnd );
nPos = 0;
nFramesLeft -= nFramesAtEnd;
}
BOOL
wave_out_set_format(WAVEFORMATEX * pwfx)
{
int channels;
int frameSize, channelCount;
ALpv params;
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_set_format: init...\n");
#endif
g_swapaudio = False;
if (pwfx->wBitsPerSample == 8)
width = AL_SAMPLE_8;
else if (pwfx->wBitsPerSample == 16)
{
width = AL_SAMPLE_16;
/* Do we need to swap the 16bit values? (Are we BigEndian) */
#if (defined(B_ENDIAN))
g_swapaudio = 1;
#else
g_swapaudio = 0;
#endif
}
/* Limited support to configure an opened audio port in IRIX. The
number of channels is a static setting and can not be changed after
a port is opened. So if the number of channels remains the same, we
can configure other settings; otherwise we have to reopen the audio
port, using same config. */
channels = pwfx->nChannels;
g_snd_rate = pwfx->nSamplesPerSec;
alSetSampFmt(audioconfig, AL_SAMPFMT_TWOSCOMP);
alSetWidth(audioconfig, width);
if (channels != alGetChannels(audioconfig))
{
alClosePort(output_port);
alSetChannels(audioconfig, channels);
output_port = alOpenPort("rdpsnd", "w", audioconfig);
if (output_port == (ALport) 0)
{
fprintf(stderr, "wave_out_set_format: alOpenPort failed: %s\n",
alGetErrorString(oserror()));
return False;
}
}
resource = alGetResource(output_port);
maxFillable = alGetFillable(output_port);
channelCount = alGetChannels(audioconfig);
frameSize = alGetWidth(audioconfig);
if (frameSize == 0 || channelCount == 0)
{
fprintf(stderr, "wave_out_set_format: bad frameSize or channelCount\n");
return False;
}
combinedFrameSize = frameSize * channelCount;
params.param = AL_RATE;
params.value.ll = (long long) g_snd_rate << 32;
if (alSetParams(resource, ¶ms, 1) < 0)
{
fprintf(stderr, "wave_set_format: alSetParams failed: %s\n",
alGetErrorString(oserror()));
return False;
}
if (params.sizeOut < 0)
{
fprintf(stderr, "wave_set_format: invalid rate %d\n", g_snd_rate);
return False;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_set_format: returning...\n");
#endif
return True;
}
