static void discrete_stream_update_mono(int ch,INT16 *buffer, int length)
{
/* Now we must do length iterations of the node list, one output for each step */
int loop,loop2,loop3;
struct node_description *node;
for(loop=0;loop<length;loop++)
{
for(loop2=0;loop2<node_count;loop2++)
{
/* Pick the first node to process */
node=running_order[loop2];
/* Work out what nodes/inputs are required, dont process NO CONNECT nodes */
/* these are ones that are connected to NODE_LIST[0] */
for(loop3=0;loop3<node->active_inputs;loop3++)
{
if(node->input_node[loop3] && (node->input_node[loop3])->node!=NODE_NC) node->input[loop3]=(node->input_node[loop3])->output;
}
/* Now step the node */
if(module_list[node->module].step) (*module_list[node->module].step)(node);
}
/* Now put the output into the buffer */
buffer[loop]=(((struct dso_output_context*)(output_node->context))->left+((struct dso_output_context*)(output_node->context))->right)/2;
}
#ifdef DISCRETE_WAVELOG
wav_add_data_16(wav_file, buffer, length);
#endif
}
int osd_update_audio_stream(INT16 *buffer)
{
// if nothing to do, don't do it
if (stream_buffer)
{
int original_bytes = bytes_in_stream_buffer();
int input_bytes = samples_this_frame * stream_format.nBlockAlign;
int final_bytes;
// update the sample adjustment
update_sample_adjustment(original_bytes);
// copy data into the sound buffer
copy_sample_data(buffer, input_bytes);
// check for overflows
final_bytes = bytes_in_stream_buffer();
if (final_bytes < original_bytes)
buffer_overflows++;
}
if( wavptr != NULL )
wav_add_data_16( wavptr, buffer, samples_this_frame * 2 );
// reset underflow/overflow tracking
if (++total_frames == INGORE_UNDERFLOW_FRAMES)
buffer_overflows = buffer_underflows = 0;
// update underflow/overflow logging
#if (DISPLAY_UNDEROVERFLOW || LOG_SOUND)
{
static int prev_overflows, prev_underflows;
if (total_frames > INGORE_UNDERFLOW_FRAMES && (buffer_overflows != prev_overflows || buffer_underflows != prev_underflows))
{
prev_overflows = buffer_overflows;
prev_underflows = buffer_underflows;
#if DISPLAY_UNDEROVERFLOW
popmessage("overflows=%d underflows=%d", buffer_overflows, buffer_underflows);
#endif
#if LOG_SOUND
fprintf(sound_log, "************************ overflows=%d underflows=%d\n", buffer_overflows, buffer_underflows);
#endif
}
}
#endif
// determine the number of samples per frame
samples_per_frame = (double)Machine->sample_rate / (double)Machine->screen[0].refresh;
// compute how many samples to generate next frame
samples_left_over += samples_per_frame;
samples_this_frame = (UINT32)samples_left_over;
samples_left_over -= (double)samples_this_frame;
samples_this_frame += current_adjustment;
// return the samples to play this next frame
return samples_this_frame;
}
int osd_update_audio_stream(INT16 *buffer)
{
if (wavptr && (Machine->sample_rate != 0))
wav_add_data_16(wavptr, buffer, samples_this_frame * 2);
return samples_this_frame;
}
static TIMER_CALLBACK( sound_update )
{
UINT32 finalmix_step, finalmix_offset;
int samples_this_update = 0;
int sample, spknum;
VPRINTF(("sound_update\n"));
profiler_mark(PROFILER_SOUND);
/* force all the speaker streams to generate the proper number of samples */
for (spknum = 0; spknum < totalspeakers; spknum++)
{
speaker_info *spk = &speaker[spknum];
const stream_sample_t *stream_buf;
/* get the output buffer */
if (spk->mixer_stream != NULL)
{
int numsamples;
/* update the stream, getting the start/end pointers around the operation */
stream_buf = stream_get_output_since_last_update(spk->mixer_stream, 0, &numsamples);
/* set or assert that all streams have the same count */
if (samples_this_update == 0)
{
samples_this_update = numsamples;
/* reset the mixing streams */
memset(leftmix, 0, samples_this_update * sizeof(*leftmix));
memset(rightmix, 0, samples_this_update * sizeof(*rightmix));
}
assert(samples_this_update == numsamples);
#ifdef MAME_DEBUG
/* debug version: keep track of the maximum sample */
for (sample = 0; sample < samples_this_update; sample++)
{
if (stream_buf[sample] > spk->max_sample)
spk->max_sample = stream_buf[sample];
else if (-stream_buf[sample] > spk->max_sample)
spk->max_sample = -stream_buf[sample];
if (stream_buf[sample] > 32767 || stream_buf[sample] < -32768)
spk->clipped_samples++;
spk->total_samples++;
}
#endif
/* mix if sound is enabled */
if (global_sound_enabled && !nosound_mode)
{
/* if the speaker is centered, send to both left and right */
if (spk->speaker->x == 0)
for (sample = 0; sample < samples_this_update; sample++)
{
leftmix[sample] += stream_buf[sample];
rightmix[sample] += stream_buf[sample];
}
/* if the speaker is to the left, send only to the left */
else if (spk->speaker->x < 0)
for (sample = 0; sample < samples_this_update; sample++)
leftmix[sample] += stream_buf[sample];
/* if the speaker is to the right, send only to the right */
else
for (sample = 0; sample < samples_this_update; sample++)
rightmix[sample] += stream_buf[sample];
}
}
}
/* now downmix the final result */
finalmix_step = video_get_speed_factor();
finalmix_offset = 0;
for (sample = finalmix_leftover; sample < samples_this_update * 100; sample += finalmix_step)
{
int sampindex = sample / 100;
INT32 samp;
/* clamp the left side */
samp = leftmix[sampindex];
if (samp < -32768)
samp = -32768;
else if (samp > 32767)
samp = 32767;
finalmix[finalmix_offset++] = samp;
/* clamp the right side */
samp = rightmix[sampindex];
if (samp < -32768)
samp = -32768;
else if (samp > 32767)
samp = 32767;
finalmix[finalmix_offset++] = samp;
}
finalmix_leftover = sample - samples_this_update * 100;
/* play the result */
if (finalmix_offset > 0)
{
osd_update_audio_stream(finalmix, finalmix_offset / 2);
if (wavfile != NULL)
wav_add_data_16(wavfile, finalmix, finalmix_offset);
}
/* update the streamer */
streams_update(machine);
profiler_mark(PROFILER_END);
}
void osd_update_audio_stream(running_machine &machine, INT16 *buffer, int samples_this_frame)
{
if (wavptr && (machine.sample_rate() != 0))
wav_add_data_16((wav_file*)wavptr, buffer, samples_this_frame * 2);
}
void sound_frame_update(void)
{
int sample, spknum;
VPRINTF(("sound_frame_update\n"));
profiler_mark(PROFILER_SOUND);
/* reset the mixing streams */
memset(leftmix, 0, samples_this_frame * sizeof(*leftmix));
memset(rightmix, 0, samples_this_frame * sizeof(*rightmix));
/* if we're not paused, keep the sounds going */
if (!mame_is_paused(Machine))
{
/* force all the speaker streams to generate the proper number of samples */
for (spknum = 0; spknum < totalspeakers; spknum++)
{
speaker_info *spk = &speaker[spknum];
stream_sample_t *stream_buf;
/* get the output buffer */
if (spk->mixer_stream)
{
stream_buf = stream_consume_output(spk->mixer_stream, 0, samples_this_frame);
#ifdef MAME_DEBUG
/* debug version: keep track of the maximum sample */
for (sample = 0; sample < samples_this_frame; sample++)
{
if (stream_buf[sample] > spk->max_sample)
spk->max_sample = stream_buf[sample];
else if (-stream_buf[sample] > spk->max_sample)
spk->max_sample = -stream_buf[sample];
if (stream_buf[sample] > 32767 || stream_buf[sample] < -32768)
spk->clipped_samples++;
spk->total_samples++;
}
#endif
/* mix if sound is enabled */
if (global_sound_enabled && !nosound_mode)
{
/* if the speaker is centered, send to both left and right */
if (spk->speaker->x == 0)
for (sample = 0; sample < samples_this_frame; sample++)
{
leftmix[sample] += stream_buf[sample];
rightmix[sample] += stream_buf[sample];
}
/* if the speaker is to the left, send only to the left */
else if (spk->speaker->x < 0)
for (sample = 0; sample < samples_this_frame; sample++)
leftmix[sample] += stream_buf[sample];
/* if the speaker is to the right, send only to the right */
else
for (sample = 0; sample < samples_this_frame; sample++)
rightmix[sample] += stream_buf[sample];
}
}
}
}
/* now downmix the final result */
for (sample = 0; sample < samples_this_frame; sample++)
{
INT32 samp;
/* clamp the left side */
samp = leftmix[sample];
if (samp < -32768)
samp = -32768;
else if (samp > 32767)
samp = 32767;
finalmix[sample*2+0] = samp;
/* clamp the right side */
samp = rightmix[sample];
if (samp < -32768)
samp = -32768;
else if (samp > 32767)
samp = 32767;
finalmix[sample*2+1] = samp;
}
if (wavfile && !mame_is_paused(Machine))
wav_add_data_16(wavfile, finalmix, samples_this_frame * 2);
/* play the result */
samples_this_frame = osd_update_audio_stream(finalmix);
/* update the streamer */
streams_frame_update();
/* reset the timer to resync for this frame */
mame_timer_adjust(sound_update_timer, time_never, 0, time_never);
profiler_mark(PROFILER_END);
}
static TIMER_CALLBACK( sound_update )
{
UINT32 finalmix_step, finalmix_offset;
int samples_this_update = 0;
int sample;
sound_private *global = machine->sound_data;
INT16 *finalmix;
INT32 *leftmix, *rightmix;
VPRINTF(("sound_update\n"));
profiler_mark_start(PROFILER_SOUND);
leftmix = global->leftmix;
rightmix = global->rightmix;
finalmix = global->finalmix;
/* force all the speaker streams to generate the proper number of samples */
for (speaker_device *speaker = speaker_first(*machine); speaker != NULL; speaker = speaker_next(speaker))
speaker->mix(leftmix, rightmix, samples_this_update, !global->enabled || global->nosound_mode);
/* now downmix the final result */
finalmix_step = video_get_speed_factor();
finalmix_offset = 0;
for (sample = global->finalmix_leftover; sample < samples_this_update * 100; sample += finalmix_step)
{
int sampindex = sample / 100;
INT32 samp;
/* clamp the left side */
samp = leftmix[sampindex];
if (samp < -32768)
samp = -32768;
else if (samp > 32767)
samp = 32767;
finalmix[finalmix_offset++] = samp;
/* clamp the right side */
samp = rightmix[sampindex];
if (samp < -32768)
samp = -32768;
else if (samp > 32767)
samp = 32767;
finalmix[finalmix_offset++] = samp;
}
global->finalmix_leftover = sample - samples_this_update * 100;
/* play the result */
if (finalmix_offset > 0)
{
osd_update_audio_stream(machine, finalmix, finalmix_offset / 2);
video_avi_add_sound(machine, finalmix, finalmix_offset / 2);
if (global->wavfile != NULL)
wav_add_data_16(global->wavfile, finalmix, finalmix_offset);
}
/* update the streamer */
streams_update(machine);
profiler_mark_end();
}
