1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
|
/* 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)
: _syst(system), _sampleRate(0), _mixerReady(false), _handleSeed(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;
// If the mixer is set to ready, then we better have a positive sample rate!
assert(!_mixerReady || _sampleRate > 0);
}
uint MixerImpl::getOutputRate() const {
return _sampleRate;
}
void MixerImpl::setOutputRate(uint sampleRate) {
if (_sampleRate != 0 && _sampleRate != sampleRate)
error("Changing the Audio::Mixer output sample rate is not supported");
_sampleRate = 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;
}
// Prevent duplicate sounds
if (id != -1) {
for (int i = 0; i != NUM_CHANNELS; i++)
if (_channels[i] != 0 && _channels[i]->getId() == id) {
if (autofreeStream == DisposeAfterUse::YES)
delete input;
return;
}
}
// Check if the mixer is not ready. This most probably means that
// no Audio output is possible according to the backend (we should
// clarify this in the Mixer creation API, though).
//
// For now we deal with this by simply ignore the sound here, never
// scheduling it for playback, never giving it a valid sound
// handle. For a game engine, this is indistinguishable from a
// sound which finishes playback before playInputStream returns.
//
// This is certainly not ideal for many engine; e.g. if a game has
// scripts which sync by waiting for certain sounds to play, then
// this syncing is broken if we just remove the sound.
//
// We could try to be more graceful, by using TimerManager and
// emulating (or rather: faking) actual audio playback; essentially
// we run the mixer callback from a timer instead of an audio
// callback.
// However, this may very well lead to new problems of its own,
// plus it would complicate the Mixer code. It seems that a better
// solution would be to adapt backends to setup a fake mixer thread
// which calls the mixer callback. We'd then still need a way to
// signal the mixer / the client code that no actual audio playback
// takes places... Anyway, either way, we first would have to
// investigate ramifications of any such approach.
//
// And also, by far the best solution is to adapt engines to be
// properly aware of the possibility of missing audio, and how to
// deal with it; be it by refusing to launch (e.g. when audio is an
// integral part of a game), by switching to alternate script
// syncing means, etc. It also seems important to test every game
// individually in a system without audio, if we really want
// to support such systems.
if (!_mixerReady) {
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;
assert(_sampleRate > 0);
// 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 ? _sampleRate : 1);
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
|