// license:BSD-3-Clause
// copyright-holders:Aaron Giles
/***************************************************************************
sound.c
Core sound functions and definitions.
***************************************************************************/
#include "emu.h"
#include "emuopts.h"
#include "osdepend.h"
#include "config.h"
#include "sound/wavwrite.h"
//**************************************************************************
// DEBUGGING
//**************************************************************************
#define VERBOSE (0)
#define VPRINTF(x) do { if (VERBOSE) osd_printf_debug x; } while (0)
//**************************************************************************
// CONSTANTS
//**************************************************************************
//**************************************************************************
// GLOBAL VARIABLES
//**************************************************************************
const attotime sound_manager::STREAMS_UPDATE_ATTOTIME = attotime::from_hz(STREAMS_UPDATE_FREQUENCY);
//**************************************************************************
// INITIALIZATION
//**************************************************************************
//-------------------------------------------------
// sound_stream - constructor
//-------------------------------------------------
sound_stream::sound_stream(device_t &device, int inputs, int outputs, int sample_rate, stream_update_delegate callback)
: m_device(device),
m_next(NULL),
m_sample_rate(sample_rate),
m_new_sample_rate(0),
m_attoseconds_per_sample(0),
m_max_samples_per_update(0),
m_input(inputs),
m_input_array(inputs),
m_resample_bufalloc(0),
m_output(outputs),
m_output_array(outputs),
m_output_bufalloc(0),
m_output_sampindex(0),
m_output_update_sampindex(0),
m_output_base_sampindex(0),
m_callback(callback)
{
// get the device's sound interface
device_sound_interface *sound;
if (!device.interface(sound))
throw emu_fatalerror("Attempted to create a sound_stream with a non-sound device");
if(m_callback.isnull())
m_callback = stream_update_delegate(FUNC(device_sound_interface::sound_stream_update),(device_sound_interface *)sound);
// create a unique tag for saving
astring state_tag;
state_tag.printf("%d", m_device.machine().sound().m_stream_list.count());
m_device.machine().save().save_item("stream", state_tag, 0, NAME(m_sample_rate));
m_device.machine().save().register_postload(save_prepost_delegate(FUNC(sound_stream::postload), this));
// save the gain of each input and output
for (int inputnum = 0; inputnum < m_input.count(); inputnum++)
{
m_device.machine().save().save_item("stream", state_tag, inputnum, NAME(m_input[inputnum].m_gain));
m_device.machine().save().save_item("stream", state_tag, inputnum, NAME(m_input[inputnum].m_user_gain));
}
for (int outputnum = 0; outputnum < m_output.count(); outputnum++)
{
m_output[outputnum].m_stream = this;
m_device.machine().save().save_item("stream", state_tag, outputnum, NAME(m_output[outputnum].m_gain));
}
// Mark synchronous streams as such
m_synchronous = m_sample_rate == STREAM_SYNC;
if (m_synchronous)
{
m_sample_rate = 0;
m_sync_timer = m_device.machine().scheduler().timer_alloc(timer_expired_delegate(FUNC(sound_stream::sync_update), this));
}
else
m_sync_timer = NULL;
// force an update to the sample rates; this will cause everything to be recomputed
// and will generate the initial resample buffers for our inputs
recompute_sample_rate_data();
// set up the initial output buffer positions now that we have data
m_output_base_sampindex = -m_max_samples_per_update;
}
//-------------------------------------------------
// sample_time - return the emulation time of the
// next sample to be generated on the stream
//-------------------------------------------------
attotime sound_stream::sample_time() const
{
return attotime(m_device.machine().sound().last_update().seconds, 0) + attotime(0, m_output_sampindex * m_attoseconds_per_sample);
}
//-------------------------------------------------
// user_gain - return the user-controllable gain
// on a given stream's input
//-------------------------------------------------
float sound_stream::user_gain(int inputnum) const
{
assert(inputnum >= 0 && inputnum < m_input.count());
return float(m_input[inputnum].m_user_gain) / 256.0f;
}
//-------------------------------------------------
// input_gain - return the input gain on a
// given stream's input
//-------------------------------------------------
float sound_stream::input_gain(int inputnum) const
{
assert(inputnum >= 0 && inputnum < m_input.count());
return float(m_input[inputnum].m_gain) / 256.0f;
}
//-------------------------------------------------
// output_gain - return the output gain on a
// given stream's output
//-------------------------------------------------
float sound_stream::output_gain(int outputnum) const
{
assert(outputnum >= 0 && outputnum < m_output.count());
return float(m_output[outputnum].m_gain) / 256.0f;
}
//-------------------------------------------------
// input_name - return the original input gain
// on a given stream's input
//-------------------------------------------------
const char *sound_stream::input_name(int inputnum, astring &string) const
{
// start with our device name and tag
assert(inputnum >= 0 && inputnum < m_input.count());
string.printf("%s '%s': ", m_device.name(), m_device.tag());
// if we have a source, indicate where the sound comes from by device name and tag
if (m_input[inputnum].m_source != NULL && m_input[inputnum].m_source->m_stream != NULL)
{
device_t &source = m_input[inputnum].m_source->m_stream->device();
string.catprintf("%s '%s'", source.name(), source.tag());
// get the sound interface; if there is more than 1 output we need to figure out which one
device_sound_interface *sound;
if (source.interface(sound) && sound->outputs() > 1)
{
// iterate over outputs until we find the stream that matches our source
// then look for a match on the output number
sound_stream *outstream;
int streamoutputnum;
for (int outputnum = 0; (outstream = sound->output_to_stream_output(outputnum, streamoutputnum)) != NULL; outputnum++)
if (outstream == m_input[inputnum].m_source->m_stream && m_input[inputnum].m_source == &outstream->m_output[streamoutputnum])
{
string.catprintf(" Ch.%d", outputnum);
break;
}
}
}
return string;
}
//-------------------------------------------------
// input_source_device - return the device
// attached as a given input's source
//-------------------------------------------------
device_t *sound_stream::input_source_device(int inputnum) const
{
assert(inputnum >= 0 && inputnum < m_input.count());
return (m_input[inputnum].m_source != NULL) ? &m_input[inputnum].m_source->m_stream->device() : NULL;
}
//-------------------------------------------------
// input_source_device - return the output number
// attached as a given input's source
//-------------------------------------------------
int sound_stream::input_source_outputnum(int inputnum) const
{
assert(inputnum >= 0 && inputnum < m_input.count());
return (m_input[inputnum].m_source != NULL) ? (m_input[inputnum].m_source - &m_input[inputnum].m_source->m_stream->m_output[0]) : -1;
}
//-------------------------------------------------
// set_input - configure a stream's input
//-------------------------------------------------
void sound_stream::set_input(int index, sound_stream *input_stream, int output_index, float gain)
{
VPRINTF(("stream_set_input(%p, '%s', %d, %p, %d, %f)\n", this, m_device.tag(), index, input_stream, output_index, gain));
// make sure it's a valid input
if (index >= m_input.count())
fatalerror("Fatal error: stream_set_input attempted to configure non-existant input %d (%d max)\n", index, m_input.count());
// make sure it's a valid output
if (input_stream != NULL && output_index >= input_stream->m_output.count())
fatalerror("Fatal error: stream_set_input attempted to use a non-existant output %d (%d max)\n", output_index, m_output.count());
// if this input is already wired, update the dependent info
stream_input &input = m_input[index];
if (input.m_source != NULL)
input.m_source->m_dependents--;
// wire it up
input.m_source = (input_stream != NULL) ? &input_stream->m_output[output_index] : NULL;
input.m_gain = int(0x100 * gain);
input.m_user_gain = 0x100;
// update the dependent info
if (input.m_source != NULL)
input.m_source->m_dependents++;
// update sample rates now that we know the input
recompute_sample_rate_data();
}
//-------------------------------------------------
// update - force a stream to update to
// the current emulated time
//-------------------------------------------------
void sound_stream::update()
{
// determine the number of samples since the start of this second
attotime time = m_device.machine().time();
INT32 update_sampindex = INT32(time.attoseconds / m_attoseconds_per_sample);
// if we're ahead of the last update, then adjust upwards
attotime last_update = m_device.machine().sound().last_update();
if (time.seconds > last_update.seconds)
{
assert(time.seconds == last_update.seconds + 1);
update_sampindex += m_sample_rate;
}
// if we're behind the last update, then adjust downwards
if (time.seconds < last_update.seconds)
{
assert(time.seconds == last_update.seconds - 1);
update_sampindex -= m_sample_rate;
}
// generate samples to get us up to the appropriate time
g_profiler.start(PROFILER_SOUND);
assert(m_output_sampindex - m_output_base_sampindex >= 0);
assert(update_sampindex - m_output_base_sampindex <= m_output_bufalloc);
generate_samples(update_sampindex - m_output_sampindex);
g_profiler.stop();
// remember this info for next time
m_output_sampindex = update_sampindex;
}
void sound_stream::sync_update(void *, INT32)
{
update();
attotime time = m_device.machine().time();
attoseconds_t next_edge = m_attoseconds_per_sample - (time.attoseconds % m_attoseconds_per_sample);
m_sync_timer->adjust(attotime(0, next_edge));
}
//-------------------------------------------------
// output_since_last_update - return a pointer to
// the output buffer and the number of samples
// since the last global update
//-------------------------------------------------
const stream_sample_t *sound_stream::output_since_last_update(int outputnum, int &numsamples)
{
// force an update on the stream
update();
// compute the number of samples and a pointer to the output buffer
numsamples = m_output_sampindex - m_output_update_sampindex;
return &m_output[outputnum].m_buffer[m_output_update_sampindex - m_output_base_sampindex];
}
//-------------------------------------------------
// set_sample_rate - set the sample rate on a
// given stream
//-------------------------------------------------
void sound_stream::set_sample_rate(int new_rate)
{
// we will update this on the next global update
if (new_rate != sample_rate())
m_new_sample_rate = new_rate;
}
//-------------------------------------------------
// set_user_gain - set the user-controllable gain
// on a given stream's input
//-------------------------------------------------
void sound_stream::set_user_gain(int inputnum, float gain)
{
update();
assert(inputnum >= 0 && inputnum < m_input.count());
m_input[inputnum].m_user_gain = int(0x100 * gain);
}
//-------------------------------------------------
// set_input_gain - set the input gain on a
// given stream's input
//-------------------------------------------------
void sound_stream::set_input_gain(int inputnum, float gain)
{
update();
assert(inputnum >= 0 && inputnum < m_input.count());
m_input[inputnum].m_gain = int(0x100 * gain);
}
//-------------------------------------------------
// set_output_gain - set the output gain on a
// given stream's output
//-------------------------------------------------
void sound_stream::set_output_gain(int outputnum, float gain)
{
update();
assert(outputnum >= 0 && outputnum < m_output.count());
m_output[outputnum].m_gain = int(0x100 * gain);
}
//-------------------------------------------------
// update_with_accounting - do a regular update,
// but also do periodic accounting
//-------------------------------------------------
void sound_stream::update_with_accounting(bool second_tick)
{
// do the normal update
update();
// if we've ticked over another second, adjust all the counters that are relative to
// the current second
INT32 output_bufindex = m_output_sampindex - m_output_base_sampindex;
if (second_tick)
{
m_output_sampindex -= m_sample_rate;
m_output_base_sampindex -= m_sample_rate;
}
// note our current output sample
m_output_update_sampindex = m_output_sampindex;
// if we don't have enough output buffer space to hold two updates' worth of samples,
// we need to shuffle things down
if (m_output_bufalloc - output_bufindex < 2 * m_max_samples_per_update)
{
INT32 samples_to_lose = output_bufindex - m_max_samples_per_update;
if (samples_to_lose > 0)
{
// if we have samples to move, do so for each output
if (output_bufindex > 0)
for (int outputnum = 0; outputnum < m_output.count(); outputnum++)
{
stream_output &output = m_output[outputnum];
memmove(&output.m_buffer[0], &output.m_buffer[samples_to_lose], sizeof(output.m_buffer[0]) * (output_bufindex - samples_to_lose));
}
// update the base position
m_output_base_sampindex += samples_to_lose;
}
}
}
//-------------------------------------------------
// apply_sample_rate_changes - if there is a
// pending sample rate change, apply it now
//-------------------------------------------------
void sound_stream::apply_sample_rate_changes()
{
// skip if nothing to do
if (m_new_sample_rate == 0)
return;
// update to the new rate and remember the old rate
UINT32 old_rate = m_sample_rate;
m_sample_rate = m_new_sample_rate;
m_new_sample_rate = 0;
// recompute all the data
recompute_sample_rate_data();
// reset our sample indexes to the current time
m_output_sampindex = (INT64)m_output_sampindex * (INT64)m_sample_rate / old_rate;
m_output_update_sampindex = (INT64)m_output_update_sampindex * (INT64)m_sample_rate / old_rate;
m_output_base_sampindex = m_output_sampindex - m_max_samples_per_update;
// clear out the buffer
for (int outputnum = 0; outputnum < m_output.count(); outputnum++)
memset(&m_output[outputnum].m_buffer[0], 0, m_max_samples_per_update * sizeof(m_output[outputnum].m_buffer[0]));
}
//-------------------------------------------------
// recompute_sample_rate_data - recompute sample
// rate data, and all streams that are affected
// by this stream
//-------------------------------------------------
void sound_stream::recompute_sample_rate_data()
{
if (m_synchronous)
{
m_sample_rate = 0;
// When synchronous, pick the sample rate for the inputs, if any
for (int inputnum = 0; inputnum < m_input.count(); inputnum++)
{
stream_input &input = m_input[inputnum];
if (input.m_source != NULL)
{
if (!m_sample_rate)
m_sample_rate = input.m_source->m_stream->m_sample_rate;
else if (m_sample_rate != input.m_source->m_stream->m_sample_rate)
throw emu_fatalerror("Incompatible sample rates as input of a synchronous stream: %d and %d\n", m_sample_rate, input.m_source->m_stream->m_sample_rate);
}
}
if (!m_sample_rate)
m_sample_rate = 1000;
}
// recompute the timing parameters
attoseconds_t update_attoseconds = m_device.machine().sound().update_attoseconds();
m_attoseconds_per_sample = ATTOSECONDS_PER_SECOND / m_sample_rate;
m_max_samples_per_update = (update_attoseconds + m_attoseconds_per_sample - 1) / m_attoseconds_per_sample;
// update resample and output buffer sizes
allocate_resample_buffers();
allocate_output_buffers();
// iterate over each input
for (int inputnum = 0; inputnum < m_input.count(); inputnum++)
{
// if we have a source, see if its sample rate changed
stream_input &input = m_input[inputnum];
if (input.m_source != NULL)
{
// okay, we have a new sample rate; recompute the latency to be the maximum
// sample period between us and our input
attoseconds_t new_attosecs_per_sample = ATTOSECONDS_PER_SECOND / input.m_source->m_stream->m_sample_rate;
attoseconds_t latency = MAX(new_attosecs_per_sample, m_attoseconds_per_sample);
// if the input stream's sample rate is lower, we will use linear interpolation
// this requires an extra sample from the source
if (input.m_source->m_stream->m_sample_rate < m_sample_rate)
latency += new_attosecs_per_sample;
// if our sample rates match exactly, we don't need any latency
else if (input.m_source->m_stream->m_sample_rate == m_sample_rate)
latency = 0;
// we generally don't want to tweak the latency, so we just keep the greatest
// one we've computed thus far
input.m_latency_attoseconds = MAX(input.m_latency_attoseconds, latency);
assert(input.m_latency_attoseconds < update_attoseconds);
}
}
// If synchronous, prime the timer
if (m_synchronous)
{
attotime time = m_device.machine().time();
attoseconds_t next_edge = m_attoseconds_per_sample - (time.attoseconds % m_attoseconds_per_sample);
m_sync_timer->adjust(attotime(0, next_edge));
}
}
//-------------------------------------------------
// allocate_resample_buffers - recompute the
// resample buffer sizes and expand if necessary
//-------------------------------------------------
void sound_stream::allocate_resample_buffers()
{
// compute the target number of samples
INT32 bufsize = 2 * m_max_samples_per_update;
// if we don't have enough room, allocate more
if (m_resample_bufalloc < bufsize)
{
// this becomes the new allocation size
m_resample_bufalloc = bufsize;
// iterate over outputs and realloc their buffers
for (int inputnum = 0; inputnum < m_input.count(); inputnum++)
m_input[inputnum].m_resample.resize_keep_and_clear_new(m_resample_bufalloc);
}
}
//-------------------------------------------------
// allocate_output_buffers - recompute the
// output buffer sizes and expand if necessary
//-------------------------------------------------
void sound_stream::allocate_output_buffers()
{
// if we don't have enough room, allocate more
INT32 bufsize = OUTPUT_BUFFER_UPDATES * m_max_samples_per_update;
if (m_output_bufalloc < bufsize)
{
// this becomes the new allocation size
m_output_bufalloc = bufsize;
// iterate over outputs and realloc their buffers
for (int outputnum = 0; outputnum < m_output.count(); outputnum++)
m_output[outputnum].m_buffer.resize_keep_and_clear_new(m_output_bufalloc);
}
}
//-------------------------------------------------
// postload - save/restore callback
//-------------------------------------------------
void sound_stream::postload()
{
// recompute the same rate information
recompute_sample_rate_data();
// make sure our output buffers are fully cleared
for (int outputnum = 0; outputnum < m_output.count(); outputnum++)
memset(m_output[outputnum].m_buffer, 0, m_output_bufalloc * sizeof(m_output[outputnum].m_buffer[0]));
// recompute the sample indexes to make sense
m_output_sampindex = m_device.machine().sound().last_update().attoseconds / m_attoseconds_per_sample;
m_output_update_sampindex = m_output_sampindex;
m_output_base_sampindex = m_output_sampindex - m_max_samples_per_update;
}
//-------------------------------------------------
// generate_samples - generate the requested
// number of samples for a stream, making sure
// all inputs have the appropriate number of
// samples generated
//-------------------------------------------------
void sound_stream::generate_samples(int samples)
{
// if we're already there, skip it
if (samples <= 0)
return;
VPRINTF(("generate_samples(%p, %d)\n", this, samples));
// ensure all inputs are up to date and generate resampled data
for (int inputnum = 0; inputnum < m_input.count(); inputnum++)
{
// update the stream to the current time
stream_input &input = m_input[inputnum];
if (input.m_source != NULL)
input.m_source->m_stream->update();
// generate the resampled data
m_input_array[inputnum] = generate_resampled_data(input, samples);
}
// loop over all outputs and compute the output pointer
for (int outputnum = 0; outputnum < m_output.count(); outputnum++)
{
stream_output &output = m_output[outputnum];
m_output_array[outputnum] = &output.m_buffer[m_output_sampindex - m_output_base_sampindex];
}
// run the callback
VPRINTF((" callback(%p, %d)\n", this, samples));
m_callback(*this, m_input_array, m_output_array, samples);
VPRINTF((" callback done\n"));
}
//-------------------------------------------------
// generate_resampled_data - generate the
// resample buffer for a given input
//-------------------------------------------------
stream_sample_t *sound_stream::generate_resampled_data(stream_input &input, UINT32 numsamples)
{
// if we don't have an output to pull data from, generate silence
stream_sample_t *dest = input.m_resample;
if (input.m_source == NULL)
{
memset(dest, 0, numsamples * sizeof(*dest));
return input.m_resample;
}
// grab data from the output
stream_output &output = *input.m_source;
sound_stream &input_stream = *output.m_stream;
int gain = (input.m_gain * input.m_user_gain * output.m_gain) >> 16;
// determine the time at which the current sample begins, accounting for the
// latency we calculated between the input and output streams
attoseconds_t basetime = m_output_sampindex * m_attoseconds_per_sample - input.m_latency_attoseconds;
// now convert that time into a sample in the input stream
INT32 basesample;
if (basetime >= 0)
basesample = basetime / input_stream.m_attoseconds_per_sample;
else
basesample = -(-basetime / input_stream.m_attoseconds_per_sample) - 1;
// compute a source pointer to the first sample
assert(basesample >= input_stream.m_output_base_sampindex);
stream_sample_t *source = &output.m_buffer[basesample - input_stream.m_output_base_sampindex];
// determine the current fraction of a sample, expressed as a fraction of FRAC_ONE
// (Note: this formula is valid as long as input_stream.m_attoseconds_per_sample signficantly exceeds FRAC_ONE > attoseconds = 4.2E-12 s)
UINT32 basefrac = (basetime - basesample * input_stream.m_attoseconds_per_sample) / ((input_stream.m_attoseconds_per_sample + FRAC_ONE - 1) >> FRAC_BITS);
assert(basefrac < FRAC_ONE);
// compute the stepping fraction
UINT32 step = (UINT64(input_stream.m_sample_rate) << FRAC_BITS) / m_sample_rate;
// if we have equal sample rates, we just need to copy
if (step == FRAC_ONE)
{
while (numsamples--)
{
// compute the sample
stream_sample_t sample = *source++;
*dest++ = (sample * gain) >> 8;
}
}
// input is undersampled: point sample except where our sample period covers a boundary
else if (step < FRAC_ONE)
{
while (numsamples != 0)
{
// fill in with point samples until we hit a boundary
int nextfrac;
while ((nextfrac = basefrac + step) < FRAC_ONE && numsamples--)
{
*dest++ = (source[0] * gain) >> 8;
basefrac = nextfrac;
}
// if we're done, we're done
if (INT32(numsamples--) < 0)
break;
// compute starting and ending fractional positions
int startfrac = basefrac >> (FRAC_BITS - 12);
int endfrac = nextfrac >> (FRAC_BITS - 12);
// blend between the two samples accordingly
stream_sample_t sample = (source[0] * (0x1000 - startfrac) + source[1] * (endfrac - 0x1000)) / (endfrac - startfrac);
*dest++ = (sample * gain) >> 8;
// advance
basefrac = nextfrac & FRAC_MASK;
source++;
}
}
// input is oversampled: sum the energy
else
{
// use 8 bits to allow some extra headroom
int smallstep = step >> (FRAC_BITS - 8);
while (numsamples--)
{
int remainder = smallstep;
int tpos = 0;
// compute the sample
int scale = (FRAC_ONE - basefrac) >> (FRAC_BITS - 8);
stream_sample_t sample = source[tpos++] * scale;
remainder -= scale;
while (remainder > 0x100)
{
sample += source[tpos++] * 0x100;
remainder -= 0x100;
}
sample += source[tpos] * remainder;
sample /= smallstep;
*dest++ = (sample * gain) >> 8;
// advance
basefrac += step;
source += basefrac >> FRAC_BITS;
basefrac &= FRAC_MASK;
}
}
return input.m_resample;
}
//**************************************************************************
// STREAM INPUT
//**************************************************************************
//-------------------------------------------------
// stream_input - constructor
//-------------------------------------------------
sound_stream::stream_input::stream_input()
: m_source(NULL),
m_latency_attoseconds(0),
m_gain(0x100),
m_user_gain(0x100)
{
}
//**************************************************************************
// STREAM OUTPUT
//**************************************************************************
//-------------------------------------------------
// stream_output - constructor
//-------------------------------------------------
sound_stream::stream_output::stream_output()
: m_dependents(0),
m_gain(0x100)
{
}
//**************************************************************************
// SOUND MANAGER
//**************************************************************************
//-------------------------------------------------
// sound_manager - constructor
//-------------------------------------------------
sound_manager::sound_manager(running_machine &machine)
: m_machine(machine),
m_update_timer(NULL),
m_finalmix_leftover(0),
m_finalmix(machine.sample_rate()),
m_leftmix(machine.sample_rate()),
m_rightmix(machine.sample_rate()),
m_muted(0),
m_attenuation(0),
m_nosound_mode(machine.osd().no_sound()),
m_wavfile(NULL),
m_update_attoseconds(STREAMS_UPDATE_ATTOTIME.attoseconds),
m_last_update(attotime::zero)
{
// get filename for WAV file or AVI file if specified
const char *wavfile = machine.options().wav_write();
const char *avifile = machine.options().avi_write();
// handle -nosound and lower sample rate if not recording WAV or AVI
if (m_nosound_mode && wavfile[0] == 0 && avifile[0] == 0)
machine.m_sample_rate = 11025;
// count the mixers
#if VERBOSE
mixer_interface_iterator iter(machine.root_device());
VPRINTF(("total mixers = %d\n", iter.count()));
#endif
// open the output WAV file if specified
if (wavfile[0] != 0)
m_wavfile = wav_open(wavfile, machine.sample_rate(), 2);
// register callbacks
config_register(machine, "mixer", config_saveload_delegate(FUNC(sound_manager::config_load), this), config_saveload_delegate(FUNC(sound_manager::config_save), this));
machine.add_notifier(MACHINE_NOTIFY_PAUSE, machine_notify_delegate(FUNC(sound_manager::pause), this));
machine.add_notifier(MACHINE_NOTIFY_RESUME, machine_notify_delegate(FUNC(sound_manager::resume), this));
machine.add_notifier(MACHINE_NOTIFY_RESET, machine_notify_delegate(FUNC(sound_manager::reset), this));
// register global states
machine.save().save_item(NAME(m_last_update));
// set the starting attenuation
set_attenuation(machine.options().volume());
// start the periodic update flushing timer
m_update_timer = machine.scheduler().timer_alloc(timer_expired_delegate(FUNC(sound_manager::update), this));
m_update_timer->adjust(STREAMS_UPDATE_ATTOTIME, 0, STREAMS_UPDATE_ATTOTIME);
}
//-------------------------------------------------
// sound_manager - destructor
//-------------------------------------------------
sound_manager::~sound_manager()
{
// close any open WAV file
if (m_wavfile != NULL)
wav_close(m_wavfile);
m_wavfile = NULL;
}
//-------------------------------------------------
// stream_alloc - allocate a new stream
//-------------------------------------------------
sound_stream *sound_manager::stream_alloc(device_t &device, int inputs, int outputs, int sample_rate, stream_update_delegate callback)
{
return &m_stream_list.append(*global_alloc(sound_stream(device, inputs, outputs, sample_rate, callback)));
}
//-------------------------------------------------
// set_attenuation - set the global volume
//-------------------------------------------------
void sound_manager::set_attenuation(int attenuation)
{
m_attenuation = attenuation;
machine().osd().set_mastervolume(m_muted ? -32 : m_attenuation);
}
//-------------------------------------------------
// indexed_mixer_input - return the mixer
// device and input index of the global mixer
// input
//-------------------------------------------------
bool sound_manager::indexed_mixer_input(int index, mixer_input &info) const
{
// scan through the mixers until we find the indexed input
mixer_interface_iterator iter(machine().root_device());
for (info.mixer = iter.first(); info.mixer != NULL; info.mixer = iter.next())
{
if (index < info.mixer->inputs())
{
info.stream = info.mixer->input_to_stream_input(index, info.inputnum);
assert(info.stream != NULL);
return true;
}
index -= info.mixer->inputs();
}
// didn't locate
return false;
}
//-------------------------------------------------
// mute - mute sound output
//-------------------------------------------------
void sound_manager::mute(bool mute, UINT8 reason)
{
if (mute)
m_muted |= reason;
else
m_muted &= ~reason;
set_attenuation(m_attenuation);
}
//-------------------------------------------------
// reset - reset all sound chips
//-------------------------------------------------
void sound_manager::reset()
{
// reset all the sound chips
sound_interface_iterator iter(machine().root_device());
for (device_sound_interface *sound = iter.first(); sound != NULL; sound = iter.next())
sound->device().reset();
}
//-------------------------------------------------
// pause - pause sound output
//-------------------------------------------------
void sound_manager::pause()
{
mute(true, MUTE_REASON_PAUSE);
}
//-------------------------------------------------
// resume - resume sound output
//-------------------------------------------------
void sound_manager::resume()
{
mute(false, MUTE_REASON_PAUSE);
}
//-------------------------------------------------
// config_load - read and apply data from the
// configuration file
//-------------------------------------------------
void sound_manager::config_load(int config_type, xml_data_node *parentnode)
{
// we only care about game files
if (config_type != CONFIG_TYPE_GAME)
return;
// might not have any data
if (parentnode == NULL)
return;
// iterate over channel nodes
for (xml_data_node *channelnode = xml_get_sibling(parentnode->child, "channel"); channelnode != NULL; channelnode = xml_get_sibling(channelnode->next, "channel"))
{
mixer_input info;
if (indexed_mixer_input(xml_get_attribute_int(channelnode, "index", -1), info))
{
float defvol = xml_get_attribute_float(channelnode, "defvol", 1.0);
float newvol = xml_get_attribute_float(channelnode, "newvol", -1000.0);
if (newvol != -1000.0)
info.stream->set_user_gain(info.inputnum, newvol / defvol);
}
}
}
//-------------------------------------------------
// config_save - save data to the configuration
// file
//-------------------------------------------------
void sound_manager::config_save(int config_type, xml_data_node *parentnode)
{
// we only care about game files
if (config_type != CONFIG_TYPE_GAME)
return;
// iterate over mixer channels
if (parentnode != NULL)
for (int mixernum = 0; ; mixernum++)
{
mixer_input info;
if (!indexed_mixer_input(mixernum, info))
break;
float newvol = info.stream->user_gain(info.inputnum);
if (newvol != 1.0f)
{
xml_data_node *channelnode = xml_add_child(parentnode, "channel", NULL);
if (channelnode != NULL)
{
xml_set_attribute_int(channelnode, "index", mixernum);
xml_set_attribute_float(channelnode, "newvol", newvol);
}
}
}
}
//-------------------------------------------------
// update - mix everything down to its final form
// and send it to the OSD layer
//-------------------------------------------------
void sound_manager::update(void *ptr, int param)
{
VPRINTF(("sound_update\n"));
g_profiler.start(PROFILER_SOUND);
// force all the speaker streams to generate the proper number of samples
int samples_this_update = 0;
speaker_device_iterator iter(machine().root_device());
for (speaker_device *speaker = iter.first(); speaker != NULL; speaker = iter.next())
speaker->mix(m_leftmix, m_rightmix, samples_this_update, (m_muted & MUTE_REASON_SYSTEM));
// now downmix the final result
UINT32 finalmix_step = machine().video().speed_factor();
UINT32 finalmix_offset = 0;
INT16 *finalmix = m_finalmix;
int sample;
for (sample = m_finalmix_leftover; sample < samples_this_update * 1000; sample += finalmix_step)
{
int sampindex = sample / 1000;
// clamp the left side
INT32 samp = m_leftmix[sampindex];
if (samp < -32768)
samp = -32768;
else if (samp > 32767)
samp = 32767;
finalmix[finalmix_offset++] = samp;
// clamp the right side
samp = m_rightmix[sampindex];
if (samp < -32768)
samp = -32768;
else if (samp > 32767)
samp = 32767;
finalmix[finalmix_offset++] = samp;
}
m_finalmix_leftover = sample - samples_this_update * 1000;
// play the result
if (finalmix_offset > 0)
{
if (!m_nosound_mode)
machine().osd().update_audio_stream(finalmix, finalmix_offset / 2);
machine().video().add_sound_to_recording(finalmix, finalmix_offset / 2);
if (m_wavfile != NULL)
wav_add_data_16(m_wavfile, finalmix, finalmix_offset);
}
// see if we ticked over to the next second
attotime curtime = machine().time();
bool second_tick = false;
if (curtime.seconds != m_last_update.seconds)
{
assert(curtime.seconds == m_last_update.seconds + 1);
second_tick = true;
}
// iterate over all the streams and update them
for (sound_stream *stream = m_stream_list.first(); stream != NULL; stream = stream->next())
stream->update_with_accounting(second_tick);
// remember the update time
m_last_update = curtime;
// update sample rates if they have changed
for (sound_stream *stream = m_stream_list.first(); stream != NULL; stream = stream->next())
stream->apply_sample_rate_changes();
g_profiler.stop();
}