mirror of
https://github.com/libretro/scummvm.git
synced 2024-12-16 14:50:17 +00:00
77fc15ce82
My previous commit which tried to support this does not work correctly when using QueuingAudioStream; it then just leads to nasty crashes. Hence I am removing this again for now, until I get around to implement one of the better alternatives. svn-id: r48239
553 lines
14 KiB
C++
553 lines
14 KiB
C++
/* ScummVM - Graphic Adventure Engine
|
|
*
|
|
* ScummVM is the legal property of its developers, whose names
|
|
* are too numerous to list here. Please refer to the COPYRIGHT
|
|
* file distributed with this source distribution.
|
|
*
|
|
* This program is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU General Public License
|
|
* as published by the Free Software Foundation; either version 2
|
|
* of the License, or (at your option) any later version.
|
|
|
|
* This program is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
* GNU General Public License for more details.
|
|
|
|
* You should have received a copy of the GNU General Public License
|
|
* along with this program; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
|
|
*
|
|
* $URL$
|
|
* $Id$
|
|
*
|
|
*/
|
|
|
|
#include "common/util.h"
|
|
#include "common/system.h"
|
|
|
|
#include "sound/mixer_intern.h"
|
|
#include "sound/rate.h"
|
|
#include "sound/audiostream.h"
|
|
#include "sound/timestamp.h"
|
|
|
|
|
|
namespace Audio {
|
|
|
|
#pragma mark -
|
|
#pragma mark --- Channel classes ---
|
|
#pragma mark -
|
|
|
|
|
|
/**
|
|
* Channel used by the default Mixer implementation.
|
|
*/
|
|
class Channel {
|
|
public:
|
|
Channel(Mixer *mixer, Mixer::SoundType type, AudioStream *input, DisposeAfterUse::Flag autofreeStream, bool reverseStereo, int id, bool permanent);
|
|
~Channel();
|
|
|
|
/**
|
|
* Mixes the channel's samples into the given buffer.
|
|
*
|
|
* @param data buffer where to mix the data
|
|
* @param len number of sample *pairs*. So a value of
|
|
* 10 means that the buffer contains twice 10 sample, each
|
|
* 16 bits, for a total of 40 bytes.
|
|
*/
|
|
void mix(int16 *data, uint len);
|
|
|
|
/**
|
|
* Queries whether the channel is still playing or not.
|
|
*/
|
|
bool isFinished() const { return _input->endOfStream(); }
|
|
|
|
/**
|
|
* Queries whether the channel is a permanent channel.
|
|
* A permanent channel is not affected by a Mixer::stopAll
|
|
* call.
|
|
*/
|
|
bool isPermanent() const { return _permanent; }
|
|
|
|
/**
|
|
* Returns the id of the channel.
|
|
*/
|
|
int getId() const { return _id; }
|
|
|
|
/**
|
|
* Pauses or unpaused the channel in a recursive fashion.
|
|
*
|
|
* @param paused true, when the channel should be paused.
|
|
* false when it should be unpaused.
|
|
*/
|
|
void pause(bool paused);
|
|
|
|
/**
|
|
* Queries whether the channel is currently paused.
|
|
*/
|
|
bool isPaused() const { return (_pauseLevel != 0); }
|
|
|
|
/**
|
|
* Sets the channel's own volume.
|
|
*
|
|
* @param volume new volume
|
|
*/
|
|
void setVolume(const byte volume);
|
|
|
|
/**
|
|
* Sets the channel's balance setting.
|
|
*
|
|
* @param balance new balance
|
|
*/
|
|
void setBalance(const int8 balance);
|
|
|
|
/**
|
|
* Notifies the channel that the global sound type
|
|
* volume settings changed.
|
|
*/
|
|
void notifyGlobalVolChange() { updateChannelVolumes(); }
|
|
|
|
/**
|
|
* Queries how long the channel has been playing.
|
|
*/
|
|
Timestamp getElapsedTime();
|
|
|
|
/**
|
|
* Queries the channel's sound type.
|
|
*/
|
|
Mixer::SoundType getType() const { return _type; }
|
|
|
|
/**
|
|
* Sets the channel's sound handle.
|
|
*
|
|
* @param handle new handle
|
|
*/
|
|
void setHandle(const SoundHandle handle) { _handle = handle; }
|
|
|
|
/**
|
|
* Queries the channel's sound handle.
|
|
*/
|
|
SoundHandle getHandle() const { return _handle; }
|
|
|
|
private:
|
|
const Mixer::SoundType _type;
|
|
SoundHandle _handle;
|
|
bool _permanent;
|
|
int _pauseLevel;
|
|
int _id;
|
|
|
|
byte _volume;
|
|
int8 _balance;
|
|
|
|
void updateChannelVolumes();
|
|
st_volume_t _volL, _volR;
|
|
|
|
Mixer *_mixer;
|
|
|
|
uint32 _samplesConsumed;
|
|
uint32 _samplesDecoded;
|
|
uint32 _mixerTimeStamp;
|
|
uint32 _pauseStartTime;
|
|
uint32 _pauseTime;
|
|
|
|
DisposeAfterUse::Flag _autofreeStream;
|
|
RateConverter *_converter;
|
|
AudioStream *_input;
|
|
};
|
|
|
|
#pragma mark -
|
|
#pragma mark --- Mixer ---
|
|
#pragma mark -
|
|
|
|
|
|
MixerImpl::MixerImpl(OSystem *system, uint sampleRate)
|
|
: _syst(system), _sampleRate(sampleRate), _mixerReady(false), _handleSeed(0) {
|
|
|
|
assert(sampleRate > 0);
|
|
|
|
int i;
|
|
|
|
for (i = 0; i < ARRAYSIZE(_volumeForSoundType); i++)
|
|
_volumeForSoundType[i] = kMaxMixerVolume;
|
|
|
|
for (i = 0; i != NUM_CHANNELS; i++)
|
|
_channels[i] = 0;
|
|
}
|
|
|
|
MixerImpl::~MixerImpl() {
|
|
for (int i = 0; i != NUM_CHANNELS; i++)
|
|
delete _channels[i];
|
|
}
|
|
|
|
void MixerImpl::setReady(bool ready) {
|
|
_mixerReady = ready;
|
|
}
|
|
|
|
uint MixerImpl::getOutputRate() const {
|
|
return _sampleRate;
|
|
}
|
|
|
|
void MixerImpl::insertChannel(SoundHandle *handle, Channel *chan) {
|
|
int index = -1;
|
|
for (int i = 0; i != NUM_CHANNELS; i++) {
|
|
if (_channels[i] == 0) {
|
|
index = i;
|
|
break;
|
|
}
|
|
}
|
|
if (index == -1) {
|
|
warning("MixerImpl::out of mixer slots");
|
|
delete chan;
|
|
return;
|
|
}
|
|
|
|
_channels[index] = chan;
|
|
|
|
SoundHandle chanHandle;
|
|
chanHandle._val = index + (_handleSeed * NUM_CHANNELS);
|
|
|
|
chan->setHandle(chanHandle);
|
|
_handleSeed++;
|
|
if (handle)
|
|
*handle = chanHandle;
|
|
}
|
|
|
|
void MixerImpl::playInputStream(
|
|
SoundType type,
|
|
SoundHandle *handle,
|
|
AudioStream *input,
|
|
int id, byte volume, int8 balance,
|
|
DisposeAfterUse::Flag autofreeStream,
|
|
bool permanent,
|
|
bool reverseStereo) {
|
|
Common::StackLock lock(_mutex);
|
|
|
|
if (input == 0) {
|
|
warning("input stream is 0");
|
|
return;
|
|
}
|
|
|
|
|
|
assert(_mixerReady);
|
|
|
|
// Prevent duplicate sounds
|
|
if (id != -1) {
|
|
for (int i = 0; i != NUM_CHANNELS; i++)
|
|
if (_channels[i] != 0 && _channels[i]->getId() == id) {
|
|
// Delete the stream if were asked to auto-dispose it.
|
|
// Note: This could cause trouble if the client code does not
|
|
// yet expect the stream to be gone. The primary example to
|
|
// keep in mind here is QueuingAudioStream.
|
|
// Thus, as a quick rule of thumb, you should never, ever,
|
|
// try to play QueuingAudioStreams with a sound id.
|
|
if (autofreeStream == DisposeAfterUse::YES)
|
|
delete input;
|
|
return;
|
|
}
|
|
}
|
|
|
|
// Create the channel
|
|
Channel *chan = new Channel(this, type, input, autofreeStream, reverseStereo, id, permanent);
|
|
chan->setVolume(volume);
|
|
chan->setBalance(balance);
|
|
insertChannel(handle, chan);
|
|
}
|
|
|
|
void MixerImpl::mixCallback(byte *samples, uint len) {
|
|
assert(samples);
|
|
|
|
Common::StackLock lock(_mutex);
|
|
|
|
int16 *buf = (int16 *)samples;
|
|
len >>= 2;
|
|
|
|
// Since the mixer callback has been called, the mixer must be ready...
|
|
_mixerReady = true;
|
|
|
|
// zero the buf
|
|
memset(buf, 0, 2 * len * sizeof(int16));
|
|
|
|
// mix all channels
|
|
for (int i = 0; i != NUM_CHANNELS; i++)
|
|
if (_channels[i]) {
|
|
if (_channels[i]->isFinished()) {
|
|
delete _channels[i];
|
|
_channels[i] = 0;
|
|
} else if (!_channels[i]->isPaused())
|
|
_channels[i]->mix(buf, len);
|
|
}
|
|
}
|
|
|
|
void MixerImpl::stopAll() {
|
|
Common::StackLock lock(_mutex);
|
|
for (int i = 0; i != NUM_CHANNELS; i++) {
|
|
if (_channels[i] != 0 && !_channels[i]->isPermanent()) {
|
|
delete _channels[i];
|
|
_channels[i] = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
void MixerImpl::stopID(int id) {
|
|
Common::StackLock lock(_mutex);
|
|
for (int i = 0; i != NUM_CHANNELS; i++) {
|
|
if (_channels[i] != 0 && _channels[i]->getId() == id) {
|
|
delete _channels[i];
|
|
_channels[i] = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
void MixerImpl::stopHandle(SoundHandle handle) {
|
|
Common::StackLock lock(_mutex);
|
|
|
|
// Simply ignore stop requests for handles of sounds that already terminated
|
|
const int index = handle._val % NUM_CHANNELS;
|
|
if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
|
|
return;
|
|
|
|
delete _channels[index];
|
|
_channels[index] = 0;
|
|
}
|
|
|
|
void MixerImpl::setChannelVolume(SoundHandle handle, byte volume) {
|
|
Common::StackLock lock(_mutex);
|
|
|
|
const int index = handle._val % NUM_CHANNELS;
|
|
if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
|
|
return;
|
|
|
|
_channels[index]->setVolume(volume);
|
|
}
|
|
|
|
void MixerImpl::setChannelBalance(SoundHandle handle, int8 balance) {
|
|
Common::StackLock lock(_mutex);
|
|
|
|
const int index = handle._val % NUM_CHANNELS;
|
|
if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
|
|
return;
|
|
|
|
_channels[index]->setBalance(balance);
|
|
}
|
|
|
|
uint32 MixerImpl::getSoundElapsedTime(SoundHandle handle) {
|
|
return getElapsedTime(handle).msecs();
|
|
}
|
|
|
|
Timestamp MixerImpl::getElapsedTime(SoundHandle handle) {
|
|
Common::StackLock lock(_mutex);
|
|
|
|
const int index = handle._val % NUM_CHANNELS;
|
|
if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
|
|
return Timestamp(0, _sampleRate);
|
|
|
|
return _channels[index]->getElapsedTime();
|
|
}
|
|
|
|
void MixerImpl::pauseAll(bool paused) {
|
|
Common::StackLock lock(_mutex);
|
|
for (int i = 0; i != NUM_CHANNELS; i++) {
|
|
if (_channels[i] != 0) {
|
|
_channels[i]->pause(paused);
|
|
}
|
|
}
|
|
}
|
|
|
|
void MixerImpl::pauseID(int id, bool paused) {
|
|
Common::StackLock lock(_mutex);
|
|
for (int i = 0; i != NUM_CHANNELS; i++) {
|
|
if (_channels[i] != 0 && _channels[i]->getId() == id) {
|
|
_channels[i]->pause(paused);
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void MixerImpl::pauseHandle(SoundHandle handle, bool paused) {
|
|
Common::StackLock lock(_mutex);
|
|
|
|
// Simply ignore (un)pause requests for sounds that already terminated
|
|
const int index = handle._val % NUM_CHANNELS;
|
|
if (!_channels[index] || _channels[index]->getHandle()._val != handle._val)
|
|
return;
|
|
|
|
_channels[index]->pause(paused);
|
|
}
|
|
|
|
bool MixerImpl::isSoundIDActive(int id) {
|
|
Common::StackLock lock(_mutex);
|
|
for (int i = 0; i != NUM_CHANNELS; i++)
|
|
if (_channels[i] && _channels[i]->getId() == id)
|
|
return true;
|
|
return false;
|
|
}
|
|
|
|
int MixerImpl::getSoundID(SoundHandle handle) {
|
|
Common::StackLock lock(_mutex);
|
|
const int index = handle._val % NUM_CHANNELS;
|
|
if (_channels[index] && _channels[index]->getHandle()._val == handle._val)
|
|
return _channels[index]->getId();
|
|
return 0;
|
|
}
|
|
|
|
bool MixerImpl::isSoundHandleActive(SoundHandle handle) {
|
|
Common::StackLock lock(_mutex);
|
|
const int index = handle._val % NUM_CHANNELS;
|
|
return _channels[index] && _channels[index]->getHandle()._val == handle._val;
|
|
}
|
|
|
|
bool MixerImpl::hasActiveChannelOfType(SoundType type) {
|
|
Common::StackLock lock(_mutex);
|
|
for (int i = 0; i != NUM_CHANNELS; i++)
|
|
if (_channels[i] && _channels[i]->getType() == type)
|
|
return true;
|
|
return false;
|
|
}
|
|
|
|
void MixerImpl::setVolumeForSoundType(SoundType type, int volume) {
|
|
assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType));
|
|
|
|
// Check range
|
|
if (volume > kMaxMixerVolume)
|
|
volume = kMaxMixerVolume;
|
|
else if (volume < 0)
|
|
volume = 0;
|
|
|
|
// TODO: Maybe we should do logarithmic (not linear) volume
|
|
// scaling? See also Player_V2::setMasterVolume
|
|
|
|
Common::StackLock lock(_mutex);
|
|
_volumeForSoundType[type] = volume;
|
|
|
|
for (int i = 0; i != NUM_CHANNELS; ++i) {
|
|
if (_channels[i] && _channels[i]->getType() == type)
|
|
_channels[i]->notifyGlobalVolChange();
|
|
}
|
|
}
|
|
|
|
int MixerImpl::getVolumeForSoundType(SoundType type) const {
|
|
assert(0 <= type && type < ARRAYSIZE(_volumeForSoundType));
|
|
|
|
return _volumeForSoundType[type];
|
|
}
|
|
|
|
|
|
#pragma mark -
|
|
#pragma mark --- Channel implementations ---
|
|
#pragma mark -
|
|
|
|
Channel::Channel(Mixer *mixer, Mixer::SoundType type, AudioStream *input,
|
|
DisposeAfterUse::Flag autofreeStream, bool reverseStereo, int id, bool permanent)
|
|
: _type(type), _mixer(mixer), _id(id), _permanent(permanent), _volume(Mixer::kMaxChannelVolume),
|
|
_balance(0), _pauseLevel(0), _samplesConsumed(0), _samplesDecoded(0), _mixerTimeStamp(0),
|
|
_pauseStartTime(0), _pauseTime(0), _autofreeStream(autofreeStream), _converter(0),
|
|
_input(input) {
|
|
assert(mixer);
|
|
assert(input);
|
|
|
|
// Get a rate converter instance
|
|
_converter = makeRateConverter(_input->getRate(), mixer->getOutputRate(), _input->isStereo(), reverseStereo);
|
|
}
|
|
|
|
Channel::~Channel() {
|
|
delete _converter;
|
|
if (_autofreeStream == DisposeAfterUse::YES)
|
|
delete _input;
|
|
}
|
|
|
|
void Channel::setVolume(const byte volume) {
|
|
_volume = volume;
|
|
updateChannelVolumes();
|
|
}
|
|
|
|
void Channel::setBalance(const int8 balance) {
|
|
_balance = balance;
|
|
updateChannelVolumes();
|
|
}
|
|
|
|
void Channel::updateChannelVolumes() {
|
|
// From the channel balance/volume and the global volume, we compute
|
|
// the effective volume for the left and right channel. Note the
|
|
// slightly odd divisor: the 255 reflects the fact that the maximal
|
|
// value for _volume is 255, while the 127 is there because the
|
|
// balance value ranges from -127 to 127. The mixer (music/sound)
|
|
// volume is in the range 0 - kMaxMixerVolume.
|
|
// Hence, the vol_l/vol_r values will be in that range, too
|
|
|
|
int vol = _mixer->getVolumeForSoundType(_type) * _volume;
|
|
|
|
if (_balance == 0) {
|
|
_volL = vol / Mixer::kMaxChannelVolume;
|
|
_volR = vol / Mixer::kMaxChannelVolume;
|
|
} else if (_balance < 0) {
|
|
_volL = vol / Mixer::kMaxChannelVolume;
|
|
_volR = ((127 + _balance) * vol) / (Mixer::kMaxChannelVolume * 127);
|
|
} else {
|
|
_volL = ((127 - _balance) * vol) / (Mixer::kMaxChannelVolume * 127);
|
|
_volR = vol / Mixer::kMaxChannelVolume;
|
|
}
|
|
}
|
|
|
|
void Channel::pause(bool paused) {
|
|
//assert((paused && _pauseLevel >= 0) || (!paused && _pauseLevel));
|
|
|
|
if (paused) {
|
|
_pauseLevel++;
|
|
|
|
if (_pauseLevel == 1)
|
|
_pauseStartTime = g_system->getMillis();
|
|
} else if (_pauseLevel > 0) {
|
|
_pauseLevel--;
|
|
|
|
if (!_pauseLevel) {
|
|
_pauseTime = (g_system->getMillis() - _pauseStartTime);
|
|
_pauseStartTime = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
Timestamp Channel::getElapsedTime() {
|
|
const uint32 rate = _mixer->getOutputRate();
|
|
uint32 delta = 0;
|
|
|
|
Audio::Timestamp ts(0, rate);
|
|
|
|
if (_mixerTimeStamp == 0)
|
|
return ts;
|
|
|
|
if (isPaused())
|
|
delta = _pauseStartTime - _mixerTimeStamp;
|
|
else
|
|
delta = g_system->getMillis() - _mixerTimeStamp - _pauseTime;
|
|
|
|
// Convert the number of samples into a time duration.
|
|
|
|
ts = ts.addFrames(_samplesConsumed);
|
|
ts = ts.addMsecs(delta);
|
|
|
|
// In theory it would seem like a good idea to limit the approximation
|
|
// so that it never exceeds the theoretical upper bound set by
|
|
// _samplesDecoded. Meanwhile, back in the real world, doing so makes
|
|
// the Broken Sword cutscenes noticeably jerkier. I guess the mixer
|
|
// isn't invoked at the regular intervals that I first imagined.
|
|
|
|
return ts;
|
|
}
|
|
|
|
void Channel::mix(int16 *data, uint len) {
|
|
assert(_input);
|
|
|
|
if (_input->endOfData()) {
|
|
// TODO: call drain method
|
|
} else {
|
|
assert(_converter);
|
|
|
|
_samplesConsumed = _samplesDecoded;
|
|
_mixerTimeStamp = g_system->getMillis();
|
|
_pauseTime = 0;
|
|
_samplesDecoded += _converter->flow(*_input, data, len, _volL, _volR);
|
|
}
|
|
}
|
|
|
|
} // End of namespace Audio
|