1 /* 2 ** 3 ** Copyright 2007, The Android Open Source Project 4 ** 5 ** Licensed under the Apache License, Version 2.0 (the "License"); 6 ** you may not use this file except in compliance with the License. 7 ** You may obtain a copy of the License at 8 ** 9 ** http://www.apache.org/licenses/LICENSE-2.0 10 ** 11 ** Unless required by applicable law or agreed to in writing, software 12 ** distributed under the License is distributed on an "AS IS" BASIS, 13 ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14 ** See the License for the specific language governing permissions and 15 ** limitations under the License. 16 */ 17 18 #define LOG_TAG "AudioMixer" 19 //#define LOG_NDEBUG 0 20 21 #include <stdint.h> 22 #include <string.h> 23 #include <stdlib.h> 24 #include <math.h> 25 #include <sys/types.h> 26 27 #include <utils/Errors.h> 28 #include <utils/Log.h> 29 30 #include <cutils/compiler.h> 31 #include <utils/Debug.h> 32 33 #include <system/audio.h> 34 35 #include <audio_utils/primitives.h> 36 #include <audio_utils/format.h> 37 #include <media/AudioMixer.h> 38 39 #include "AudioMixerOps.h" 40 41 // The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer. 42 #ifndef FCC_2 43 #define FCC_2 2 44 #endif 45 46 // Look for MONO_HACK for any Mono hack involving legacy mono channel to 47 // stereo channel conversion. 48 49 /* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is 50 * being used. This is a considerable amount of log spam, so don't enable unless you 51 * are verifying the hook based code. 52 */ 53 //#define VERY_VERY_VERBOSE_LOGGING 54 #ifdef VERY_VERY_VERBOSE_LOGGING 55 #define ALOGVV ALOGV 56 //define ALOGVV printf // for test-mixer.cpp 57 #else 58 #define ALOGVV(a...) do { } while (0) 59 #endif 60 61 #ifndef ARRAY_SIZE 62 #define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0])) 63 #endif 64 65 // TODO: Move these macro/inlines to a header file. 66 template <typename T> 67 static inline 68 T max(const T& x, const T& y) { 69 return x > y ? x : y; 70 } 71 72 // Set kUseNewMixer to true to use the new mixer engine always. Otherwise the 73 // original code will be used for stereo sinks, the new mixer for multichannel. 74 static const bool kUseNewMixer = true; 75 76 // Set kUseFloat to true to allow floating input into the mixer engine. 77 // If kUseNewMixer is false, this is ignored or may be overridden internally 78 // because of downmix/upmix support. 79 static const bool kUseFloat = true; 80 81 // Set to default copy buffer size in frames for input processing. 82 static const size_t kCopyBufferFrameCount = 256; 83 84 namespace android { 85 86 // ---------------------------------------------------------------------------- 87 88 template <typename T> 89 T min(const T& a, const T& b) 90 { 91 return a < b ? a : b; 92 } 93 94 // ---------------------------------------------------------------------------- 95 96 // Ensure mConfiguredNames bitmask is initialized properly on all architectures. 97 // The value of 1 << x is undefined in C when x >= 32. 98 99 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) 100 : mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1), 101 mSampleRate(sampleRate) 102 { 103 ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u", 104 maxNumTracks, MAX_NUM_TRACKS); 105 106 // AudioMixer is not yet capable of more than 32 active track inputs 107 ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS); 108 109 pthread_once(&sOnceControl, &sInitRoutine); 110 111 mState.enabledTracks= 0; 112 mState.needsChanged = 0; 113 mState.frameCount = frameCount; 114 mState.hook = process__nop; 115 mState.outputTemp = NULL; 116 mState.resampleTemp = NULL; 117 mState.mNBLogWriter = &mDummyLogWriter; 118 // mState.reserved 119 120 // FIXME Most of the following initialization is probably redundant since 121 // tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0 122 // and mTrackNames is initially 0. However, leave it here until that's verified. 123 track_t* t = mState.tracks; 124 for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { 125 t->resampler = NULL; 126 t->downmixerBufferProvider = NULL; 127 t->mReformatBufferProvider = NULL; 128 t->mTimestretchBufferProvider = NULL; 129 t++; 130 } 131 132 } 133 134 AudioMixer::~AudioMixer() 135 { 136 track_t* t = mState.tracks; 137 for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) { 138 delete t->resampler; 139 delete t->downmixerBufferProvider; 140 delete t->mReformatBufferProvider; 141 delete t->mTimestretchBufferProvider; 142 t++; 143 } 144 delete [] mState.outputTemp; 145 delete [] mState.resampleTemp; 146 } 147 148 void AudioMixer::setNBLogWriter(NBLog::Writer *logWriter) 149 { 150 mState.mNBLogWriter = logWriter; 151 } 152 153 static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) { 154 return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; 155 } 156 157 int AudioMixer::getTrackName(audio_channel_mask_t channelMask, 158 audio_format_t format, int sessionId) 159 { 160 if (!isValidPcmTrackFormat(format)) { 161 ALOGE("AudioMixer::getTrackName invalid format (%#x)", format); 162 return -1; 163 } 164 uint32_t names = (~mTrackNames) & mConfiguredNames; 165 if (names != 0) { 166 int n = __builtin_ctz(names); 167 ALOGV("add track (%d)", n); 168 // assume default parameters for the track, except where noted below 169 track_t* t = &mState.tracks[n]; 170 t->needs = 0; 171 172 // Integer volume. 173 // Currently integer volume is kept for the legacy integer mixer. 174 // Will be removed when the legacy mixer path is removed. 175 t->volume[0] = UNITY_GAIN_INT; 176 t->volume[1] = UNITY_GAIN_INT; 177 t->prevVolume[0] = UNITY_GAIN_INT << 16; 178 t->prevVolume[1] = UNITY_GAIN_INT << 16; 179 t->volumeInc[0] = 0; 180 t->volumeInc[1] = 0; 181 t->auxLevel = 0; 182 t->auxInc = 0; 183 t->prevAuxLevel = 0; 184 185 // Floating point volume. 186 t->mVolume[0] = UNITY_GAIN_FLOAT; 187 t->mVolume[1] = UNITY_GAIN_FLOAT; 188 t->mPrevVolume[0] = UNITY_GAIN_FLOAT; 189 t->mPrevVolume[1] = UNITY_GAIN_FLOAT; 190 t->mVolumeInc[0] = 0.; 191 t->mVolumeInc[1] = 0.; 192 t->mAuxLevel = 0.; 193 t->mAuxInc = 0.; 194 t->mPrevAuxLevel = 0.; 195 196 // no initialization needed 197 // t->frameCount 198 t->channelCount = audio_channel_count_from_out_mask(channelMask); 199 t->enabled = false; 200 ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO, 201 "Non-stereo channel mask: %d\n", channelMask); 202 t->channelMask = channelMask; 203 t->sessionId = sessionId; 204 // setBufferProvider(name, AudioBufferProvider *) is required before enable(name) 205 t->bufferProvider = NULL; 206 t->buffer.raw = NULL; 207 // no initialization needed 208 // t->buffer.frameCount 209 t->hook = NULL; 210 t->in = NULL; 211 t->resampler = NULL; 212 t->sampleRate = mSampleRate; 213 // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name) 214 t->mainBuffer = NULL; 215 t->auxBuffer = NULL; 216 t->mInputBufferProvider = NULL; 217 t->mReformatBufferProvider = NULL; 218 t->downmixerBufferProvider = NULL; 219 t->mPostDownmixReformatBufferProvider = NULL; 220 t->mTimestretchBufferProvider = NULL; 221 t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 222 t->mFormat = format; 223 t->mMixerInFormat = selectMixerInFormat(format); 224 t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required 225 t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits( 226 AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO); 227 t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask); 228 t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT; 229 // Check the downmixing (or upmixing) requirements. 230 status_t status = t->prepareForDownmix(); 231 if (status != OK) { 232 ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask); 233 return -1; 234 } 235 // prepareForDownmix() may change mDownmixRequiresFormat 236 ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat); 237 t->prepareForReformat(); 238 mTrackNames |= 1 << n; 239 return TRACK0 + n; 240 } 241 ALOGE("AudioMixer::getTrackName out of available tracks"); 242 return -1; 243 } 244 245 void AudioMixer::invalidateState(uint32_t mask) 246 { 247 if (mask != 0) { 248 mState.needsChanged |= mask; 249 mState.hook = process__validate; 250 } 251 } 252 253 // Called when channel masks have changed for a track name 254 // TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format, 255 // which will simplify this logic. 256 bool AudioMixer::setChannelMasks(int name, 257 audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) { 258 track_t &track = mState.tracks[name]; 259 260 if (trackChannelMask == track.channelMask 261 && mixerChannelMask == track.mMixerChannelMask) { 262 return false; // no need to change 263 } 264 // always recompute for both channel masks even if only one has changed. 265 const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask); 266 const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask); 267 const bool mixerChannelCountChanged = track.mMixerChannelCount != mixerChannelCount; 268 269 ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) 270 && trackChannelCount 271 && mixerChannelCount); 272 track.channelMask = trackChannelMask; 273 track.channelCount = trackChannelCount; 274 track.mMixerChannelMask = mixerChannelMask; 275 track.mMixerChannelCount = mixerChannelCount; 276 277 // channel masks have changed, does this track need a downmixer? 278 // update to try using our desired format (if we aren't already using it) 279 const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat; 280 const status_t status = mState.tracks[name].prepareForDownmix(); 281 ALOGE_IF(status != OK, 282 "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x", 283 status, track.channelMask, track.mMixerChannelMask); 284 285 if (prevDownmixerFormat != track.mDownmixRequiresFormat) { 286 track.prepareForReformat(); // because of downmixer, track format may change! 287 } 288 289 if (track.resampler && mixerChannelCountChanged) { 290 // resampler channels may have changed. 291 const uint32_t resetToSampleRate = track.sampleRate; 292 delete track.resampler; 293 track.resampler = NULL; 294 track.sampleRate = mSampleRate; // without resampler, track rate is device sample rate. 295 // recreate the resampler with updated format, channels, saved sampleRate. 296 track.setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/); 297 } 298 return true; 299 } 300 301 void AudioMixer::track_t::unprepareForDownmix() { 302 ALOGV("AudioMixer::unprepareForDownmix(%p)", this); 303 304 if (mPostDownmixReformatBufferProvider != nullptr) { 305 // release any buffers held by the mPostDownmixReformatBufferProvider 306 // before deallocating the downmixerBufferProvider. 307 mPostDownmixReformatBufferProvider->reset(); 308 } 309 310 mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; 311 if (downmixerBufferProvider != NULL) { 312 // this track had previously been configured with a downmixer, delete it 313 ALOGV(" deleting old downmixer"); 314 delete downmixerBufferProvider; 315 downmixerBufferProvider = NULL; 316 reconfigureBufferProviders(); 317 } else { 318 ALOGV(" nothing to do, no downmixer to delete"); 319 } 320 } 321 322 status_t AudioMixer::track_t::prepareForDownmix() 323 { 324 ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x", 325 this, channelMask); 326 327 // discard the previous downmixer if there was one 328 unprepareForDownmix(); 329 // MONO_HACK Only remix (upmix or downmix) if the track and mixer/device channel masks 330 // are not the same and not handled internally, as mono -> stereo currently is. 331 if (channelMask == mMixerChannelMask 332 || (channelMask == AUDIO_CHANNEL_OUT_MONO 333 && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) { 334 return NO_ERROR; 335 } 336 // DownmixerBufferProvider is only used for position masks. 337 if (audio_channel_mask_get_representation(channelMask) 338 == AUDIO_CHANNEL_REPRESENTATION_POSITION 339 && DownmixerBufferProvider::isMultichannelCapable()) { 340 DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask, 341 mMixerChannelMask, 342 AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */, 343 sampleRate, sessionId, kCopyBufferFrameCount); 344 345 if (pDbp->isValid()) { // if constructor completed properly 346 mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix 347 downmixerBufferProvider = pDbp; 348 reconfigureBufferProviders(); 349 return NO_ERROR; 350 } 351 delete pDbp; 352 } 353 354 // Effect downmixer does not accept the channel conversion. Let's use our remixer. 355 RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask, 356 mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount); 357 // Remix always finds a conversion whereas Downmixer effect above may fail. 358 downmixerBufferProvider = pRbp; 359 reconfigureBufferProviders(); 360 return NO_ERROR; 361 } 362 363 void AudioMixer::track_t::unprepareForReformat() { 364 ALOGV("AudioMixer::unprepareForReformat(%p)", this); 365 bool requiresReconfigure = false; 366 if (mReformatBufferProvider != NULL) { 367 delete mReformatBufferProvider; 368 mReformatBufferProvider = NULL; 369 requiresReconfigure = true; 370 } 371 if (mPostDownmixReformatBufferProvider != NULL) { 372 delete mPostDownmixReformatBufferProvider; 373 mPostDownmixReformatBufferProvider = NULL; 374 requiresReconfigure = true; 375 } 376 if (requiresReconfigure) { 377 reconfigureBufferProviders(); 378 } 379 } 380 381 status_t AudioMixer::track_t::prepareForReformat() 382 { 383 ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat); 384 // discard previous reformatters 385 unprepareForReformat(); 386 // only configure reformatters as needed 387 const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID 388 ? mDownmixRequiresFormat : mMixerInFormat; 389 bool requiresReconfigure = false; 390 if (mFormat != targetFormat) { 391 mReformatBufferProvider = new ReformatBufferProvider( 392 audio_channel_count_from_out_mask(channelMask), 393 mFormat, 394 targetFormat, 395 kCopyBufferFrameCount); 396 requiresReconfigure = true; 397 } 398 if (targetFormat != mMixerInFormat) { 399 mPostDownmixReformatBufferProvider = new ReformatBufferProvider( 400 audio_channel_count_from_out_mask(mMixerChannelMask), 401 targetFormat, 402 mMixerInFormat, 403 kCopyBufferFrameCount); 404 requiresReconfigure = true; 405 } 406 if (requiresReconfigure) { 407 reconfigureBufferProviders(); 408 } 409 return NO_ERROR; 410 } 411 412 void AudioMixer::track_t::reconfigureBufferProviders() 413 { 414 bufferProvider = mInputBufferProvider; 415 if (mReformatBufferProvider) { 416 mReformatBufferProvider->setBufferProvider(bufferProvider); 417 bufferProvider = mReformatBufferProvider; 418 } 419 if (downmixerBufferProvider) { 420 downmixerBufferProvider->setBufferProvider(bufferProvider); 421 bufferProvider = downmixerBufferProvider; 422 } 423 if (mPostDownmixReformatBufferProvider) { 424 mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider); 425 bufferProvider = mPostDownmixReformatBufferProvider; 426 } 427 if (mTimestretchBufferProvider) { 428 mTimestretchBufferProvider->setBufferProvider(bufferProvider); 429 bufferProvider = mTimestretchBufferProvider; 430 } 431 } 432 433 void AudioMixer::deleteTrackName(int name) 434 { 435 ALOGV("AudioMixer::deleteTrackName(%d)", name); 436 name -= TRACK0; 437 LOG_ALWAYS_FATAL_IF(name < 0 || name >= (int)MAX_NUM_TRACKS, "bad track name %d", name); 438 ALOGV("deleteTrackName(%d)", name); 439 track_t& track(mState.tracks[ name ]); 440 if (track.enabled) { 441 track.enabled = false; 442 invalidateState(1<<name); 443 } 444 // delete the resampler 445 delete track.resampler; 446 track.resampler = NULL; 447 // delete the downmixer 448 mState.tracks[name].unprepareForDownmix(); 449 // delete the reformatter 450 mState.tracks[name].unprepareForReformat(); 451 // delete the timestretch provider 452 delete track.mTimestretchBufferProvider; 453 track.mTimestretchBufferProvider = NULL; 454 mTrackNames &= ~(1<<name); 455 } 456 457 void AudioMixer::enable(int name) 458 { 459 name -= TRACK0; 460 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 461 track_t& track = mState.tracks[name]; 462 463 if (!track.enabled) { 464 track.enabled = true; 465 ALOGV("enable(%d)", name); 466 invalidateState(1 << name); 467 } 468 } 469 470 void AudioMixer::disable(int name) 471 { 472 name -= TRACK0; 473 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 474 track_t& track = mState.tracks[name]; 475 476 if (track.enabled) { 477 track.enabled = false; 478 ALOGV("disable(%d)", name); 479 invalidateState(1 << name); 480 } 481 } 482 483 /* Sets the volume ramp variables for the AudioMixer. 484 * 485 * The volume ramp variables are used to transition from the previous 486 * volume to the set volume. ramp controls the duration of the transition. 487 * Its value is typically one state framecount period, but may also be 0, 488 * meaning "immediate." 489 * 490 * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment 491 * even if there is a nonzero floating point increment (in that case, the volume 492 * change is immediate). This restriction should be changed when the legacy mixer 493 * is removed (see #2). 494 * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed 495 * when no longer needed. 496 * 497 * @param newVolume set volume target in floating point [0.0, 1.0]. 498 * @param ramp number of frames to increment over. if ramp is 0, the volume 499 * should be set immediately. Currently ramp should not exceed 65535 (frames). 500 * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return. 501 * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return. 502 * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return. 503 * @param pSetVolume pointer to the float target volume, set on return. 504 * @param pPrevVolume pointer to the float previous volume, set on return. 505 * @param pVolumeInc pointer to the float increment per output audio frame, set on return. 506 * @return true if the volume has changed, false if volume is same. 507 */ 508 static inline bool setVolumeRampVariables(float newVolume, int32_t ramp, 509 int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc, 510 float *pSetVolume, float *pPrevVolume, float *pVolumeInc) { 511 // check floating point volume to see if it is identical to the previously 512 // set volume. 513 // We do not use a tolerance here (and reject changes too small) 514 // as it may be confusing to use a different value than the one set. 515 // If the resulting volume is too small to ramp, it is a direct set of the volume. 516 if (newVolume == *pSetVolume) { 517 return false; 518 } 519 if (newVolume < 0) { 520 newVolume = 0; // should not have negative volumes 521 } else { 522 switch (fpclassify(newVolume)) { 523 case FP_SUBNORMAL: 524 case FP_NAN: 525 newVolume = 0; 526 break; 527 case FP_ZERO: 528 break; // zero volume is fine 529 case FP_INFINITE: 530 // Infinite volume could be handled consistently since 531 // floating point math saturates at infinities, 532 // but we limit volume to unity gain float. 533 // ramp = 0; break; 534 // 535 newVolume = AudioMixer::UNITY_GAIN_FLOAT; 536 break; 537 case FP_NORMAL: 538 default: 539 // Floating point does not have problems with overflow wrap 540 // that integer has. However, we limit the volume to 541 // unity gain here. 542 // TODO: Revisit the volume limitation and perhaps parameterize. 543 if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) { 544 newVolume = AudioMixer::UNITY_GAIN_FLOAT; 545 } 546 break; 547 } 548 } 549 550 // set floating point volume ramp 551 if (ramp != 0) { 552 // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there 553 // is no computational mismatch; hence equality is checked here. 554 ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished," 555 " prev:%f set_to:%f", *pPrevVolume, *pSetVolume); 556 const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal 557 const float maxv = max(newVolume, *pPrevVolume); // could be inf, cannot be nan, subnormal 558 559 if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan) 560 && maxv + inc != maxv) { // inc must make forward progress 561 *pVolumeInc = inc; 562 // ramp is set now. 563 // Note: if newVolume is 0, then near the end of the ramp, 564 // it may be possible that the ramped volume may be subnormal or 565 // temporarily negative by a small amount or subnormal due to floating 566 // point inaccuracies. 567 } else { 568 ramp = 0; // ramp not allowed 569 } 570 } 571 572 // compute and check integer volume, no need to check negative values 573 // The integer volume is limited to "unity_gain" to avoid wrapping and other 574 // audio artifacts, so it never reaches the range limit of U4.28. 575 // We safely use signed 16 and 32 bit integers here. 576 const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan 577 const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ? 578 AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume; 579 580 // set integer volume ramp 581 if (ramp != 0) { 582 // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28. 583 // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there 584 // is no computational mismatch; hence equality is checked here. 585 ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished," 586 " prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16); 587 const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp; 588 589 if (inc != 0) { // inc must make forward progress 590 *pIntVolumeInc = inc; 591 } else { 592 ramp = 0; // ramp not allowed 593 } 594 } 595 596 // if no ramp, or ramp not allowed, then clear float and integer increments 597 if (ramp == 0) { 598 *pVolumeInc = 0; 599 *pPrevVolume = newVolume; 600 *pIntVolumeInc = 0; 601 *pIntPrevVolume = intVolume << 16; 602 } 603 *pSetVolume = newVolume; 604 *pIntSetVolume = intVolume; 605 return true; 606 } 607 608 void AudioMixer::setParameter(int name, int target, int param, void *value) 609 { 610 name -= TRACK0; 611 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 612 track_t& track = mState.tracks[name]; 613 614 int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value)); 615 int32_t *valueBuf = reinterpret_cast<int32_t*>(value); 616 617 switch (target) { 618 619 case TRACK: 620 switch (param) { 621 case CHANNEL_MASK: { 622 const audio_channel_mask_t trackChannelMask = 623 static_cast<audio_channel_mask_t>(valueInt); 624 if (setChannelMasks(name, trackChannelMask, track.mMixerChannelMask)) { 625 ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask); 626 invalidateState(1 << name); 627 } 628 } break; 629 case MAIN_BUFFER: 630 if (track.mainBuffer != valueBuf) { 631 track.mainBuffer = valueBuf; 632 ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf); 633 invalidateState(1 << name); 634 } 635 break; 636 case AUX_BUFFER: 637 if (track.auxBuffer != valueBuf) { 638 track.auxBuffer = valueBuf; 639 ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf); 640 invalidateState(1 << name); 641 } 642 break; 643 case FORMAT: { 644 audio_format_t format = static_cast<audio_format_t>(valueInt); 645 if (track.mFormat != format) { 646 ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format); 647 track.mFormat = format; 648 ALOGV("setParameter(TRACK, FORMAT, %#x)", format); 649 track.prepareForReformat(); 650 invalidateState(1 << name); 651 } 652 } break; 653 // FIXME do we want to support setting the downmix type from AudioFlinger? 654 // for a specific track? or per mixer? 655 /* case DOWNMIX_TYPE: 656 break */ 657 case MIXER_FORMAT: { 658 audio_format_t format = static_cast<audio_format_t>(valueInt); 659 if (track.mMixerFormat != format) { 660 track.mMixerFormat = format; 661 ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format); 662 } 663 } break; 664 case MIXER_CHANNEL_MASK: { 665 const audio_channel_mask_t mixerChannelMask = 666 static_cast<audio_channel_mask_t>(valueInt); 667 if (setChannelMasks(name, track.channelMask, mixerChannelMask)) { 668 ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask); 669 invalidateState(1 << name); 670 } 671 } break; 672 default: 673 LOG_ALWAYS_FATAL("setParameter track: bad param %d", param); 674 } 675 break; 676 677 case RESAMPLE: 678 switch (param) { 679 case SAMPLE_RATE: 680 ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt); 681 if (track.setResampler(uint32_t(valueInt), mSampleRate)) { 682 ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)", 683 uint32_t(valueInt)); 684 invalidateState(1 << name); 685 } 686 break; 687 case RESET: 688 track.resetResampler(); 689 invalidateState(1 << name); 690 break; 691 case REMOVE: 692 delete track.resampler; 693 track.resampler = NULL; 694 track.sampleRate = mSampleRate; 695 invalidateState(1 << name); 696 break; 697 default: 698 LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param); 699 } 700 break; 701 702 case RAMP_VOLUME: 703 case VOLUME: 704 switch (param) { 705 case AUXLEVEL: 706 if (setVolumeRampVariables(*reinterpret_cast<float*>(value), 707 target == RAMP_VOLUME ? mState.frameCount : 0, 708 &track.auxLevel, &track.prevAuxLevel, &track.auxInc, 709 &track.mAuxLevel, &track.mPrevAuxLevel, &track.mAuxInc)) { 710 ALOGV("setParameter(%s, AUXLEVEL: %04x)", 711 target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track.auxLevel); 712 invalidateState(1 << name); 713 } 714 break; 715 default: 716 if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) { 717 if (setVolumeRampVariables(*reinterpret_cast<float*>(value), 718 target == RAMP_VOLUME ? mState.frameCount : 0, 719 &track.volume[param - VOLUME0], &track.prevVolume[param - VOLUME0], 720 &track.volumeInc[param - VOLUME0], 721 &track.mVolume[param - VOLUME0], &track.mPrevVolume[param - VOLUME0], 722 &track.mVolumeInc[param - VOLUME0])) { 723 ALOGV("setParameter(%s, VOLUME%d: %04x)", 724 target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0, 725 track.volume[param - VOLUME0]); 726 invalidateState(1 << name); 727 } 728 } else { 729 LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param); 730 } 731 } 732 break; 733 case TIMESTRETCH: 734 switch (param) { 735 case PLAYBACK_RATE: { 736 const AudioPlaybackRate *playbackRate = 737 reinterpret_cast<AudioPlaybackRate*>(value); 738 ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate), 739 "bad parameters speed %f, pitch %f",playbackRate->mSpeed, 740 playbackRate->mPitch); 741 if (track.setPlaybackRate(*playbackRate)) { 742 ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE " 743 "%f %f %d %d", 744 playbackRate->mSpeed, 745 playbackRate->mPitch, 746 playbackRate->mStretchMode, 747 playbackRate->mFallbackMode); 748 // invalidateState(1 << name); 749 } 750 } break; 751 default: 752 LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param); 753 } 754 break; 755 756 default: 757 LOG_ALWAYS_FATAL("setParameter: bad target %d", target); 758 } 759 } 760 761 bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate) 762 { 763 if (trackSampleRate != devSampleRate || resampler != NULL) { 764 if (sampleRate != trackSampleRate) { 765 sampleRate = trackSampleRate; 766 if (resampler == NULL) { 767 ALOGV("Creating resampler from track %d Hz to device %d Hz", 768 trackSampleRate, devSampleRate); 769 AudioResampler::src_quality quality; 770 // force lowest quality level resampler if use case isn't music or video 771 // FIXME this is flawed for dynamic sample rates, as we choose the resampler 772 // quality level based on the initial ratio, but that could change later. 773 // Should have a way to distinguish tracks with static ratios vs. dynamic ratios. 774 if (isMusicRate(trackSampleRate)) { 775 quality = AudioResampler::DEFAULT_QUALITY; 776 } else { 777 quality = AudioResampler::DYN_LOW_QUALITY; 778 } 779 780 // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer 781 // but if none exists, it is the channel count (1 for mono). 782 const int resamplerChannelCount = downmixerBufferProvider != NULL 783 ? mMixerChannelCount : channelCount; 784 ALOGVV("Creating resampler:" 785 " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n", 786 mMixerInFormat, resamplerChannelCount, devSampleRate, quality); 787 resampler = AudioResampler::create( 788 mMixerInFormat, 789 resamplerChannelCount, 790 devSampleRate, quality); 791 } 792 return true; 793 } 794 } 795 return false; 796 } 797 798 bool AudioMixer::track_t::setPlaybackRate(const AudioPlaybackRate &playbackRate) 799 { 800 if ((mTimestretchBufferProvider == NULL && 801 fabs(playbackRate.mSpeed - mPlaybackRate.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA && 802 fabs(playbackRate.mPitch - mPlaybackRate.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) || 803 isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) { 804 return false; 805 } 806 mPlaybackRate = playbackRate; 807 if (mTimestretchBufferProvider == NULL) { 808 // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer 809 // but if none exists, it is the channel count (1 for mono). 810 const int timestretchChannelCount = downmixerBufferProvider != NULL 811 ? mMixerChannelCount : channelCount; 812 mTimestretchBufferProvider = new TimestretchBufferProvider(timestretchChannelCount, 813 mMixerInFormat, sampleRate, playbackRate); 814 reconfigureBufferProviders(); 815 } else { 816 reinterpret_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider) 817 ->setPlaybackRate(playbackRate); 818 } 819 return true; 820 } 821 822 /* Checks to see if the volume ramp has completed and clears the increment 823 * variables appropriately. 824 * 825 * FIXME: There is code to handle int/float ramp variable switchover should it not 826 * complete within a mixer buffer processing call, but it is preferred to avoid switchover 827 * due to precision issues. The switchover code is included for legacy code purposes 828 * and can be removed once the integer volume is removed. 829 * 830 * It is not sufficient to clear only the volumeInc integer variable because 831 * if one channel requires ramping, all channels are ramped. 832 * 833 * There is a bit of duplicated code here, but it keeps backward compatibility. 834 */ 835 inline void AudioMixer::track_t::adjustVolumeRamp(bool aux, bool useFloat) 836 { 837 if (useFloat) { 838 for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) { 839 if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) || 840 (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) { 841 volumeInc[i] = 0; 842 prevVolume[i] = volume[i] << 16; 843 mVolumeInc[i] = 0.; 844 mPrevVolume[i] = mVolume[i]; 845 } else { 846 //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]); 847 prevVolume[i] = u4_28_from_float(mPrevVolume[i]); 848 } 849 } 850 } else { 851 for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) { 852 if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) || 853 ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) { 854 volumeInc[i] = 0; 855 prevVolume[i] = volume[i] << 16; 856 mVolumeInc[i] = 0.; 857 mPrevVolume[i] = mVolume[i]; 858 } else { 859 //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]); 860 mPrevVolume[i] = float_from_u4_28(prevVolume[i]); 861 } 862 } 863 } 864 /* TODO: aux is always integer regardless of output buffer type */ 865 if (aux) { 866 if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) || 867 ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) { 868 auxInc = 0; 869 prevAuxLevel = auxLevel << 16; 870 mAuxInc = 0.; 871 mPrevAuxLevel = mAuxLevel; 872 } else { 873 //ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc); 874 } 875 } 876 } 877 878 size_t AudioMixer::getUnreleasedFrames(int name) const 879 { 880 name -= TRACK0; 881 if (uint32_t(name) < MAX_NUM_TRACKS) { 882 return mState.tracks[name].getUnreleasedFrames(); 883 } 884 return 0; 885 } 886 887 void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider) 888 { 889 name -= TRACK0; 890 ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); 891 892 if (mState.tracks[name].mInputBufferProvider == bufferProvider) { 893 return; // don't reset any buffer providers if identical. 894 } 895 if (mState.tracks[name].mReformatBufferProvider != NULL) { 896 mState.tracks[name].mReformatBufferProvider->reset(); 897 } else if (mState.tracks[name].downmixerBufferProvider != NULL) { 898 mState.tracks[name].downmixerBufferProvider->reset(); 899 } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) { 900 mState.tracks[name].mPostDownmixReformatBufferProvider->reset(); 901 } else if (mState.tracks[name].mTimestretchBufferProvider != NULL) { 902 mState.tracks[name].mTimestretchBufferProvider->reset(); 903 } 904 905 mState.tracks[name].mInputBufferProvider = bufferProvider; 906 mState.tracks[name].reconfigureBufferProviders(); 907 } 908 909 910 void AudioMixer::process() 911 { 912 mState.hook(&mState); 913 } 914 915 916 void AudioMixer::process__validate(state_t* state) 917 { 918 ALOGW_IF(!state->needsChanged, 919 "in process__validate() but nothing's invalid"); 920 921 uint32_t changed = state->needsChanged; 922 state->needsChanged = 0; // clear the validation flag 923 924 // recompute which tracks are enabled / disabled 925 uint32_t enabled = 0; 926 uint32_t disabled = 0; 927 while (changed) { 928 const int i = 31 - __builtin_clz(changed); 929 const uint32_t mask = 1<<i; 930 changed &= ~mask; 931 track_t& t = state->tracks[i]; 932 (t.enabled ? enabled : disabled) |= mask; 933 } 934 state->enabledTracks &= ~disabled; 935 state->enabledTracks |= enabled; 936 937 // compute everything we need... 938 int countActiveTracks = 0; 939 // TODO: fix all16BitsStereNoResample logic to 940 // either properly handle muted tracks (it should ignore them) 941 // or remove altogether as an obsolete optimization. 942 bool all16BitsStereoNoResample = true; 943 bool resampling = false; 944 bool volumeRamp = false; 945 uint32_t en = state->enabledTracks; 946 while (en) { 947 const int i = 31 - __builtin_clz(en); 948 en &= ~(1<<i); 949 950 countActiveTracks++; 951 track_t& t = state->tracks[i]; 952 uint32_t n = 0; 953 // FIXME can overflow (mask is only 3 bits) 954 n |= NEEDS_CHANNEL_1 + t.channelCount - 1; 955 if (t.doesResample()) { 956 n |= NEEDS_RESAMPLE; 957 } 958 if (t.auxLevel != 0 && t.auxBuffer != NULL) { 959 n |= NEEDS_AUX; 960 } 961 962 if (t.volumeInc[0]|t.volumeInc[1]) { 963 volumeRamp = true; 964 } else if (!t.doesResample() && t.volumeRL == 0) { 965 n |= NEEDS_MUTE; 966 } 967 t.needs = n; 968 969 if (n & NEEDS_MUTE) { 970 t.hook = track__nop; 971 } else { 972 if (n & NEEDS_AUX) { 973 all16BitsStereoNoResample = false; 974 } 975 if (n & NEEDS_RESAMPLE) { 976 all16BitsStereoNoResample = false; 977 resampling = true; 978 t.hook = getTrackHook(TRACKTYPE_RESAMPLE, t.mMixerChannelCount, 979 t.mMixerInFormat, t.mMixerFormat); 980 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2, 981 "Track %d needs downmix + resample", i); 982 } else { 983 if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){ 984 t.hook = getTrackHook( 985 (t.mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO // TODO: MONO_HACK 986 && t.channelMask == AUDIO_CHANNEL_OUT_MONO) 987 ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE, 988 t.mMixerChannelCount, 989 t.mMixerInFormat, t.mMixerFormat); 990 all16BitsStereoNoResample = false; 991 } 992 if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){ 993 t.hook = getTrackHook(TRACKTYPE_NORESAMPLE, t.mMixerChannelCount, 994 t.mMixerInFormat, t.mMixerFormat); 995 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2, 996 "Track %d needs downmix", i); 997 } 998 } 999 } 1000 } 1001 1002 // select the processing hooks 1003 state->hook = process__nop; 1004 if (countActiveTracks > 0) { 1005 if (resampling) { 1006 if (!state->outputTemp) { 1007 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 1008 } 1009 if (!state->resampleTemp) { 1010 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 1011 } 1012 state->hook = process__genericResampling; 1013 } else { 1014 if (state->outputTemp) { 1015 delete [] state->outputTemp; 1016 state->outputTemp = NULL; 1017 } 1018 if (state->resampleTemp) { 1019 delete [] state->resampleTemp; 1020 state->resampleTemp = NULL; 1021 } 1022 state->hook = process__genericNoResampling; 1023 if (all16BitsStereoNoResample && !volumeRamp) { 1024 if (countActiveTracks == 1) { 1025 const int i = 31 - __builtin_clz(state->enabledTracks); 1026 track_t& t = state->tracks[i]; 1027 if ((t.needs & NEEDS_MUTE) == 0) { 1028 // The check prevents a muted track from acquiring a process hook. 1029 // 1030 // This is dangerous if the track is MONO as that requires 1031 // special case handling due to implicit channel duplication. 1032 // Stereo or Multichannel should actually be fine here. 1033 state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK, 1034 t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat); 1035 } 1036 } 1037 } 1038 } 1039 } 1040 1041 ALOGV("mixer configuration change: %d activeTracks (%08x) " 1042 "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d", 1043 countActiveTracks, state->enabledTracks, 1044 all16BitsStereoNoResample, resampling, volumeRamp); 1045 1046 state->hook(state); 1047 1048 // Now that the volume ramp has been done, set optimal state and 1049 // track hooks for subsequent mixer process 1050 if (countActiveTracks > 0) { 1051 bool allMuted = true; 1052 uint32_t en = state->enabledTracks; 1053 while (en) { 1054 const int i = 31 - __builtin_clz(en); 1055 en &= ~(1<<i); 1056 track_t& t = state->tracks[i]; 1057 if (!t.doesResample() && t.volumeRL == 0) { 1058 t.needs |= NEEDS_MUTE; 1059 t.hook = track__nop; 1060 } else { 1061 allMuted = false; 1062 } 1063 } 1064 if (allMuted) { 1065 state->hook = process__nop; 1066 } else if (all16BitsStereoNoResample) { 1067 if (countActiveTracks == 1) { 1068 const int i = 31 - __builtin_clz(state->enabledTracks); 1069 track_t& t = state->tracks[i]; 1070 // Muted single tracks handled by allMuted above. 1071 state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK, 1072 t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat); 1073 } 1074 } 1075 } 1076 } 1077 1078 1079 void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, 1080 int32_t* temp, int32_t* aux) 1081 { 1082 ALOGVV("track__genericResample\n"); 1083 t->resampler->setSampleRate(t->sampleRate); 1084 1085 // ramp gain - resample to temp buffer and scale/mix in 2nd step 1086 if (aux != NULL) { 1087 // always resample with unity gain when sending to auxiliary buffer to be able 1088 // to apply send level after resampling 1089 t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT); 1090 memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(int32_t)); 1091 t->resampler->resample(temp, outFrameCount, t->bufferProvider); 1092 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 1093 volumeRampStereo(t, out, outFrameCount, temp, aux); 1094 } else { 1095 volumeStereo(t, out, outFrameCount, temp, aux); 1096 } 1097 } else { 1098 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 1099 t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT); 1100 memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t)); 1101 t->resampler->resample(temp, outFrameCount, t->bufferProvider); 1102 volumeRampStereo(t, out, outFrameCount, temp, aux); 1103 } 1104 1105 // constant gain 1106 else { 1107 t->resampler->setVolume(t->mVolume[0], t->mVolume[1]); 1108 t->resampler->resample(out, outFrameCount, t->bufferProvider); 1109 } 1110 } 1111 } 1112 1113 void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused, 1114 size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused) 1115 { 1116 } 1117 1118 void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 1119 int32_t* aux) 1120 { 1121 int32_t vl = t->prevVolume[0]; 1122 int32_t vr = t->prevVolume[1]; 1123 const int32_t vlInc = t->volumeInc[0]; 1124 const int32_t vrInc = t->volumeInc[1]; 1125 1126 //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 1127 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 1128 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1129 1130 // ramp volume 1131 if (CC_UNLIKELY(aux != NULL)) { 1132 int32_t va = t->prevAuxLevel; 1133 const int32_t vaInc = t->auxInc; 1134 int32_t l; 1135 int32_t r; 1136 1137 do { 1138 l = (*temp++ >> 12); 1139 r = (*temp++ >> 12); 1140 *out++ += (vl >> 16) * l; 1141 *out++ += (vr >> 16) * r; 1142 *aux++ += (va >> 17) * (l + r); 1143 vl += vlInc; 1144 vr += vrInc; 1145 va += vaInc; 1146 } while (--frameCount); 1147 t->prevAuxLevel = va; 1148 } else { 1149 do { 1150 *out++ += (vl >> 16) * (*temp++ >> 12); 1151 *out++ += (vr >> 16) * (*temp++ >> 12); 1152 vl += vlInc; 1153 vr += vrInc; 1154 } while (--frameCount); 1155 } 1156 t->prevVolume[0] = vl; 1157 t->prevVolume[1] = vr; 1158 t->adjustVolumeRamp(aux != NULL); 1159 } 1160 1161 void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 1162 int32_t* aux) 1163 { 1164 const int16_t vl = t->volume[0]; 1165 const int16_t vr = t->volume[1]; 1166 1167 if (CC_UNLIKELY(aux != NULL)) { 1168 const int16_t va = t->auxLevel; 1169 do { 1170 int16_t l = (int16_t)(*temp++ >> 12); 1171 int16_t r = (int16_t)(*temp++ >> 12); 1172 out[0] = mulAdd(l, vl, out[0]); 1173 int16_t a = (int16_t)(((int32_t)l + r) >> 1); 1174 out[1] = mulAdd(r, vr, out[1]); 1175 out += 2; 1176 aux[0] = mulAdd(a, va, aux[0]); 1177 aux++; 1178 } while (--frameCount); 1179 } else { 1180 do { 1181 int16_t l = (int16_t)(*temp++ >> 12); 1182 int16_t r = (int16_t)(*temp++ >> 12); 1183 out[0] = mulAdd(l, vl, out[0]); 1184 out[1] = mulAdd(r, vr, out[1]); 1185 out += 2; 1186 } while (--frameCount); 1187 } 1188 } 1189 1190 void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, 1191 int32_t* temp __unused, int32_t* aux) 1192 { 1193 ALOGVV("track__16BitsStereo\n"); 1194 const int16_t *in = static_cast<const int16_t *>(t->in); 1195 1196 if (CC_UNLIKELY(aux != NULL)) { 1197 int32_t l; 1198 int32_t r; 1199 // ramp gain 1200 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 1201 int32_t vl = t->prevVolume[0]; 1202 int32_t vr = t->prevVolume[1]; 1203 int32_t va = t->prevAuxLevel; 1204 const int32_t vlInc = t->volumeInc[0]; 1205 const int32_t vrInc = t->volumeInc[1]; 1206 const int32_t vaInc = t->auxInc; 1207 // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 1208 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 1209 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1210 1211 do { 1212 l = (int32_t)*in++; 1213 r = (int32_t)*in++; 1214 *out++ += (vl >> 16) * l; 1215 *out++ += (vr >> 16) * r; 1216 *aux++ += (va >> 17) * (l + r); 1217 vl += vlInc; 1218 vr += vrInc; 1219 va += vaInc; 1220 } while (--frameCount); 1221 1222 t->prevVolume[0] = vl; 1223 t->prevVolume[1] = vr; 1224 t->prevAuxLevel = va; 1225 t->adjustVolumeRamp(true); 1226 } 1227 1228 // constant gain 1229 else { 1230 const uint32_t vrl = t->volumeRL; 1231 const int16_t va = (int16_t)t->auxLevel; 1232 do { 1233 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1234 int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1); 1235 in += 2; 1236 out[0] = mulAddRL(1, rl, vrl, out[0]); 1237 out[1] = mulAddRL(0, rl, vrl, out[1]); 1238 out += 2; 1239 aux[0] = mulAdd(a, va, aux[0]); 1240 aux++; 1241 } while (--frameCount); 1242 } 1243 } else { 1244 // ramp gain 1245 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 1246 int32_t vl = t->prevVolume[0]; 1247 int32_t vr = t->prevVolume[1]; 1248 const int32_t vlInc = t->volumeInc[0]; 1249 const int32_t vrInc = t->volumeInc[1]; 1250 1251 // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 1252 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 1253 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1254 1255 do { 1256 *out++ += (vl >> 16) * (int32_t) *in++; 1257 *out++ += (vr >> 16) * (int32_t) *in++; 1258 vl += vlInc; 1259 vr += vrInc; 1260 } while (--frameCount); 1261 1262 t->prevVolume[0] = vl; 1263 t->prevVolume[1] = vr; 1264 t->adjustVolumeRamp(false); 1265 } 1266 1267 // constant gain 1268 else { 1269 const uint32_t vrl = t->volumeRL; 1270 do { 1271 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1272 in += 2; 1273 out[0] = mulAddRL(1, rl, vrl, out[0]); 1274 out[1] = mulAddRL(0, rl, vrl, out[1]); 1275 out += 2; 1276 } while (--frameCount); 1277 } 1278 } 1279 t->in = in; 1280 } 1281 1282 void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, 1283 int32_t* temp __unused, int32_t* aux) 1284 { 1285 ALOGVV("track__16BitsMono\n"); 1286 const int16_t *in = static_cast<int16_t const *>(t->in); 1287 1288 if (CC_UNLIKELY(aux != NULL)) { 1289 // ramp gain 1290 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) { 1291 int32_t vl = t->prevVolume[0]; 1292 int32_t vr = t->prevVolume[1]; 1293 int32_t va = t->prevAuxLevel; 1294 const int32_t vlInc = t->volumeInc[0]; 1295 const int32_t vrInc = t->volumeInc[1]; 1296 const int32_t vaInc = t->auxInc; 1297 1298 // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 1299 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 1300 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1301 1302 do { 1303 int32_t l = *in++; 1304 *out++ += (vl >> 16) * l; 1305 *out++ += (vr >> 16) * l; 1306 *aux++ += (va >> 16) * l; 1307 vl += vlInc; 1308 vr += vrInc; 1309 va += vaInc; 1310 } while (--frameCount); 1311 1312 t->prevVolume[0] = vl; 1313 t->prevVolume[1] = vr; 1314 t->prevAuxLevel = va; 1315 t->adjustVolumeRamp(true); 1316 } 1317 // constant gain 1318 else { 1319 const int16_t vl = t->volume[0]; 1320 const int16_t vr = t->volume[1]; 1321 const int16_t va = (int16_t)t->auxLevel; 1322 do { 1323 int16_t l = *in++; 1324 out[0] = mulAdd(l, vl, out[0]); 1325 out[1] = mulAdd(l, vr, out[1]); 1326 out += 2; 1327 aux[0] = mulAdd(l, va, aux[0]); 1328 aux++; 1329 } while (--frameCount); 1330 } 1331 } else { 1332 // ramp gain 1333 if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) { 1334 int32_t vl = t->prevVolume[0]; 1335 int32_t vr = t->prevVolume[1]; 1336 const int32_t vlInc = t->volumeInc[0]; 1337 const int32_t vrInc = t->volumeInc[1]; 1338 1339 // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d", 1340 // t, vlInc/65536.0f, vl/65536.0f, t->volume[0], 1341 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1342 1343 do { 1344 int32_t l = *in++; 1345 *out++ += (vl >> 16) * l; 1346 *out++ += (vr >> 16) * l; 1347 vl += vlInc; 1348 vr += vrInc; 1349 } while (--frameCount); 1350 1351 t->prevVolume[0] = vl; 1352 t->prevVolume[1] = vr; 1353 t->adjustVolumeRamp(false); 1354 } 1355 // constant gain 1356 else { 1357 const int16_t vl = t->volume[0]; 1358 const int16_t vr = t->volume[1]; 1359 do { 1360 int16_t l = *in++; 1361 out[0] = mulAdd(l, vl, out[0]); 1362 out[1] = mulAdd(l, vr, out[1]); 1363 out += 2; 1364 } while (--frameCount); 1365 } 1366 } 1367 t->in = in; 1368 } 1369 1370 // no-op case 1371 void AudioMixer::process__nop(state_t* state) 1372 { 1373 ALOGVV("process__nop\n"); 1374 uint32_t e0 = state->enabledTracks; 1375 while (e0) { 1376 // process by group of tracks with same output buffer to 1377 // avoid multiple memset() on same buffer 1378 uint32_t e1 = e0, e2 = e0; 1379 int i = 31 - __builtin_clz(e1); 1380 { 1381 track_t& t1 = state->tracks[i]; 1382 e2 &= ~(1<<i); 1383 while (e2) { 1384 i = 31 - __builtin_clz(e2); 1385 e2 &= ~(1<<i); 1386 track_t& t2 = state->tracks[i]; 1387 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1388 e1 &= ~(1<<i); 1389 } 1390 } 1391 e0 &= ~(e1); 1392 1393 memset(t1.mainBuffer, 0, state->frameCount * t1.mMixerChannelCount 1394 * audio_bytes_per_sample(t1.mMixerFormat)); 1395 } 1396 1397 while (e1) { 1398 i = 31 - __builtin_clz(e1); 1399 e1 &= ~(1<<i); 1400 { 1401 track_t& t3 = state->tracks[i]; 1402 size_t outFrames = state->frameCount; 1403 while (outFrames) { 1404 t3.buffer.frameCount = outFrames; 1405 t3.bufferProvider->getNextBuffer(&t3.buffer); 1406 if (t3.buffer.raw == NULL) break; 1407 outFrames -= t3.buffer.frameCount; 1408 t3.bufferProvider->releaseBuffer(&t3.buffer); 1409 } 1410 } 1411 } 1412 } 1413 } 1414 1415 // generic code without resampling 1416 void AudioMixer::process__genericNoResampling(state_t* state) 1417 { 1418 ALOGVV("process__genericNoResampling\n"); 1419 int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32))); 1420 1421 // acquire each track's buffer 1422 uint32_t enabledTracks = state->enabledTracks; 1423 uint32_t e0 = enabledTracks; 1424 while (e0) { 1425 const int i = 31 - __builtin_clz(e0); 1426 e0 &= ~(1<<i); 1427 track_t& t = state->tracks[i]; 1428 t.buffer.frameCount = state->frameCount; 1429 t.bufferProvider->getNextBuffer(&t.buffer); 1430 t.frameCount = t.buffer.frameCount; 1431 t.in = t.buffer.raw; 1432 } 1433 1434 e0 = enabledTracks; 1435 while (e0) { 1436 // process by group of tracks with same output buffer to 1437 // optimize cache use 1438 uint32_t e1 = e0, e2 = e0; 1439 int j = 31 - __builtin_clz(e1); 1440 track_t& t1 = state->tracks[j]; 1441 e2 &= ~(1<<j); 1442 while (e2) { 1443 j = 31 - __builtin_clz(e2); 1444 e2 &= ~(1<<j); 1445 track_t& t2 = state->tracks[j]; 1446 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1447 e1 &= ~(1<<j); 1448 } 1449 } 1450 e0 &= ~(e1); 1451 // this assumes output 16 bits stereo, no resampling 1452 int32_t *out = t1.mainBuffer; 1453 size_t numFrames = 0; 1454 do { 1455 memset(outTemp, 0, sizeof(outTemp)); 1456 e2 = e1; 1457 while (e2) { 1458 const int i = 31 - __builtin_clz(e2); 1459 e2 &= ~(1<<i); 1460 track_t& t = state->tracks[i]; 1461 size_t outFrames = BLOCKSIZE; 1462 int32_t *aux = NULL; 1463 if (CC_UNLIKELY(t.needs & NEEDS_AUX)) { 1464 aux = t.auxBuffer + numFrames; 1465 } 1466 while (outFrames) { 1467 // t.in == NULL can happen if the track was flushed just after having 1468 // been enabled for mixing. 1469 if (t.in == NULL) { 1470 enabledTracks &= ~(1<<i); 1471 e1 &= ~(1<<i); 1472 break; 1473 } 1474 size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; 1475 if (inFrames > 0) { 1476 t.hook(&t, outTemp + (BLOCKSIZE - outFrames) * t.mMixerChannelCount, 1477 inFrames, state->resampleTemp, aux); 1478 t.frameCount -= inFrames; 1479 outFrames -= inFrames; 1480 if (CC_UNLIKELY(aux != NULL)) { 1481 aux += inFrames; 1482 } 1483 } 1484 if (t.frameCount == 0 && outFrames) { 1485 t.bufferProvider->releaseBuffer(&t.buffer); 1486 t.buffer.frameCount = (state->frameCount - numFrames) - 1487 (BLOCKSIZE - outFrames); 1488 t.bufferProvider->getNextBuffer(&t.buffer); 1489 t.in = t.buffer.raw; 1490 if (t.in == NULL) { 1491 enabledTracks &= ~(1<<i); 1492 e1 &= ~(1<<i); 1493 break; 1494 } 1495 t.frameCount = t.buffer.frameCount; 1496 } 1497 } 1498 } 1499 1500 convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat, 1501 BLOCKSIZE * t1.mMixerChannelCount); 1502 // TODO: fix ugly casting due to choice of out pointer type 1503 out = reinterpret_cast<int32_t*>((uint8_t*)out 1504 + BLOCKSIZE * t1.mMixerChannelCount 1505 * audio_bytes_per_sample(t1.mMixerFormat)); 1506 numFrames += BLOCKSIZE; 1507 } while (numFrames < state->frameCount); 1508 } 1509 1510 // release each track's buffer 1511 e0 = enabledTracks; 1512 while (e0) { 1513 const int i = 31 - __builtin_clz(e0); 1514 e0 &= ~(1<<i); 1515 track_t& t = state->tracks[i]; 1516 t.bufferProvider->releaseBuffer(&t.buffer); 1517 } 1518 } 1519 1520 1521 // generic code with resampling 1522 void AudioMixer::process__genericResampling(state_t* state) 1523 { 1524 ALOGVV("process__genericResampling\n"); 1525 // this const just means that local variable outTemp doesn't change 1526 int32_t* const outTemp = state->outputTemp; 1527 size_t numFrames = state->frameCount; 1528 1529 uint32_t e0 = state->enabledTracks; 1530 while (e0) { 1531 // process by group of tracks with same output buffer 1532 // to optimize cache use 1533 uint32_t e1 = e0, e2 = e0; 1534 int j = 31 - __builtin_clz(e1); 1535 track_t& t1 = state->tracks[j]; 1536 e2 &= ~(1<<j); 1537 while (e2) { 1538 j = 31 - __builtin_clz(e2); 1539 e2 &= ~(1<<j); 1540 track_t& t2 = state->tracks[j]; 1541 if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) { 1542 e1 &= ~(1<<j); 1543 } 1544 } 1545 e0 &= ~(e1); 1546 int32_t *out = t1.mainBuffer; 1547 memset(outTemp, 0, sizeof(*outTemp) * t1.mMixerChannelCount * state->frameCount); 1548 while (e1) { 1549 const int i = 31 - __builtin_clz(e1); 1550 e1 &= ~(1<<i); 1551 track_t& t = state->tracks[i]; 1552 int32_t *aux = NULL; 1553 if (CC_UNLIKELY(t.needs & NEEDS_AUX)) { 1554 aux = t.auxBuffer; 1555 } 1556 1557 // this is a little goofy, on the resampling case we don't 1558 // acquire/release the buffers because it's done by 1559 // the resampler. 1560 if (t.needs & NEEDS_RESAMPLE) { 1561 t.hook(&t, outTemp, numFrames, state->resampleTemp, aux); 1562 } else { 1563 1564 size_t outFrames = 0; 1565 1566 while (outFrames < numFrames) { 1567 t.buffer.frameCount = numFrames - outFrames; 1568 t.bufferProvider->getNextBuffer(&t.buffer); 1569 t.in = t.buffer.raw; 1570 // t.in == NULL can happen if the track was flushed just after having 1571 // been enabled for mixing. 1572 if (t.in == NULL) break; 1573 1574 if (CC_UNLIKELY(aux != NULL)) { 1575 aux += outFrames; 1576 } 1577 t.hook(&t, outTemp + outFrames * t.mMixerChannelCount, t.buffer.frameCount, 1578 state->resampleTemp, aux); 1579 outFrames += t.buffer.frameCount; 1580 t.bufferProvider->releaseBuffer(&t.buffer); 1581 } 1582 } 1583 } 1584 convertMixerFormat(out, t1.mMixerFormat, 1585 outTemp, t1.mMixerInFormat, numFrames * t1.mMixerChannelCount); 1586 } 1587 } 1588 1589 // one track, 16 bits stereo without resampling is the most common case 1590 void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state) 1591 { 1592 ALOGVV("process__OneTrack16BitsStereoNoResampling\n"); 1593 // This method is only called when state->enabledTracks has exactly 1594 // one bit set. The asserts below would verify this, but are commented out 1595 // since the whole point of this method is to optimize performance. 1596 //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled"); 1597 const int i = 31 - __builtin_clz(state->enabledTracks); 1598 //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled"); 1599 const track_t& t = state->tracks[i]; 1600 1601 AudioBufferProvider::Buffer& b(t.buffer); 1602 1603 int32_t* out = t.mainBuffer; 1604 float *fout = reinterpret_cast<float*>(out); 1605 size_t numFrames = state->frameCount; 1606 1607 const int16_t vl = t.volume[0]; 1608 const int16_t vr = t.volume[1]; 1609 const uint32_t vrl = t.volumeRL; 1610 while (numFrames) { 1611 b.frameCount = numFrames; 1612 t.bufferProvider->getNextBuffer(&b); 1613 const int16_t *in = b.i16; 1614 1615 // in == NULL can happen if the track was flushed just after having 1616 // been enabled for mixing. 1617 if (in == NULL || (((uintptr_t)in) & 3)) { 1618 if ( AUDIO_FORMAT_PCM_FLOAT == t.mMixerFormat ) { 1619 memset((char*)fout, 0, numFrames 1620 * t.mMixerChannelCount * audio_bytes_per_sample(t.mMixerFormat)); 1621 } else { 1622 memset((char*)out, 0, numFrames 1623 * t.mMixerChannelCount * audio_bytes_per_sample(t.mMixerFormat)); 1624 } 1625 ALOGE_IF((((uintptr_t)in) & 3), 1626 "process__OneTrack16BitsStereoNoResampling: misaligned buffer" 1627 " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f", 1628 in, i, t.channelCount, t.needs, vrl, t.mVolume[0], t.mVolume[1]); 1629 return; 1630 } 1631 size_t outFrames = b.frameCount; 1632 1633 switch (t.mMixerFormat) { 1634 case AUDIO_FORMAT_PCM_FLOAT: 1635 do { 1636 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1637 in += 2; 1638 int32_t l = mulRL(1, rl, vrl); 1639 int32_t r = mulRL(0, rl, vrl); 1640 *fout++ = float_from_q4_27(l); 1641 *fout++ = float_from_q4_27(r); 1642 // Note: In case of later int16_t sink output, 1643 // conversion and clamping is done by memcpy_to_i16_from_float(). 1644 } while (--outFrames); 1645 break; 1646 case AUDIO_FORMAT_PCM_16_BIT: 1647 if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) { 1648 // volume is boosted, so we might need to clamp even though 1649 // we process only one track. 1650 do { 1651 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1652 in += 2; 1653 int32_t l = mulRL(1, rl, vrl) >> 12; 1654 int32_t r = mulRL(0, rl, vrl) >> 12; 1655 // clamping... 1656 l = clamp16(l); 1657 r = clamp16(r); 1658 *out++ = (r<<16) | (l & 0xFFFF); 1659 } while (--outFrames); 1660 } else { 1661 do { 1662 uint32_t rl = *reinterpret_cast<const uint32_t *>(in); 1663 in += 2; 1664 int32_t l = mulRL(1, rl, vrl) >> 12; 1665 int32_t r = mulRL(0, rl, vrl) >> 12; 1666 *out++ = (r<<16) | (l & 0xFFFF); 1667 } while (--outFrames); 1668 } 1669 break; 1670 default: 1671 LOG_ALWAYS_FATAL("bad mixer format: %d", t.mMixerFormat); 1672 } 1673 numFrames -= b.frameCount; 1674 t.bufferProvider->releaseBuffer(&b); 1675 } 1676 } 1677 1678 /*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT; 1679 1680 /*static*/ void AudioMixer::sInitRoutine() 1681 { 1682 DownmixerBufferProvider::init(); // for the downmixer 1683 } 1684 1685 /* TODO: consider whether this level of optimization is necessary. 1686 * Perhaps just stick with a single for loop. 1687 */ 1688 1689 // Needs to derive a compile time constant (constexpr). Could be targeted to go 1690 // to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication. 1691 #define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \ 1692 (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype)) 1693 1694 /* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) 1695 * TO: int32_t (Q4.27) or float 1696 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float 1697 * TA: int32_t (Q4.27) 1698 */ 1699 template <int MIXTYPE, 1700 typename TO, typename TI, typename TV, typename TA, typename TAV> 1701 static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, 1702 const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) 1703 { 1704 switch (channels) { 1705 case 1: 1706 volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc); 1707 break; 1708 case 2: 1709 volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc); 1710 break; 1711 case 3: 1712 volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, 1713 frameCount, in, aux, vol, volinc, vola, volainc); 1714 break; 1715 case 4: 1716 volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, 1717 frameCount, in, aux, vol, volinc, vola, volainc); 1718 break; 1719 case 5: 1720 volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, 1721 frameCount, in, aux, vol, volinc, vola, volainc); 1722 break; 1723 case 6: 1724 volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, 1725 frameCount, in, aux, vol, volinc, vola, volainc); 1726 break; 1727 case 7: 1728 volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, 1729 frameCount, in, aux, vol, volinc, vola, volainc); 1730 break; 1731 case 8: 1732 volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, 1733 frameCount, in, aux, vol, volinc, vola, volainc); 1734 break; 1735 } 1736 } 1737 1738 /* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) 1739 * TO: int32_t (Q4.27) or float 1740 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float 1741 * TA: int32_t (Q4.27) 1742 */ 1743 template <int MIXTYPE, 1744 typename TO, typename TI, typename TV, typename TA, typename TAV> 1745 static void volumeMulti(uint32_t channels, TO* out, size_t frameCount, 1746 const TI* in, TA* aux, const TV *vol, TAV vola) 1747 { 1748 switch (channels) { 1749 case 1: 1750 volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola); 1751 break; 1752 case 2: 1753 volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola); 1754 break; 1755 case 3: 1756 volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola); 1757 break; 1758 case 4: 1759 volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola); 1760 break; 1761 case 5: 1762 volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola); 1763 break; 1764 case 6: 1765 volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola); 1766 break; 1767 case 7: 1768 volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola); 1769 break; 1770 case 8: 1771 volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola); 1772 break; 1773 } 1774 } 1775 1776 /* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) 1777 * USEFLOATVOL (set to true if float volume is used) 1778 * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards) 1779 * TO: int32_t (Q4.27) or float 1780 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float 1781 * TA: int32_t (Q4.27) 1782 */ 1783 template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL, 1784 typename TO, typename TI, typename TA> 1785 void AudioMixer::volumeMix(TO *out, size_t outFrames, 1786 const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t) 1787 { 1788 if (USEFLOATVOL) { 1789 if (ramp) { 1790 volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux, 1791 t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc); 1792 if (ADJUSTVOL) { 1793 t->adjustVolumeRamp(aux != NULL, true); 1794 } 1795 } else { 1796 volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux, 1797 t->mVolume, t->auxLevel); 1798 } 1799 } else { 1800 if (ramp) { 1801 volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux, 1802 t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc); 1803 if (ADJUSTVOL) { 1804 t->adjustVolumeRamp(aux != NULL); 1805 } 1806 } else { 1807 volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux, 1808 t->volume, t->auxLevel); 1809 } 1810 } 1811 } 1812 1813 /* This process hook is called when there is a single track without 1814 * aux buffer, volume ramp, or resampling. 1815 * TODO: Update the hook selection: this can properly handle aux and ramp. 1816 * 1817 * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) 1818 * TO: int32_t (Q4.27) or float 1819 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float 1820 * TA: int32_t (Q4.27) 1821 */ 1822 template <int MIXTYPE, typename TO, typename TI, typename TA> 1823 void AudioMixer::process_NoResampleOneTrack(state_t* state) 1824 { 1825 ALOGVV("process_NoResampleOneTrack\n"); 1826 // CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz. 1827 const int i = 31 - __builtin_clz(state->enabledTracks); 1828 ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled"); 1829 track_t *t = &state->tracks[i]; 1830 const uint32_t channels = t->mMixerChannelCount; 1831 TO* out = reinterpret_cast<TO*>(t->mainBuffer); 1832 TA* aux = reinterpret_cast<TA*>(t->auxBuffer); 1833 const bool ramp = t->needsRamp(); 1834 1835 for (size_t numFrames = state->frameCount; numFrames; ) { 1836 AudioBufferProvider::Buffer& b(t->buffer); 1837 // get input buffer 1838 b.frameCount = numFrames; 1839 t->bufferProvider->getNextBuffer(&b); 1840 const TI *in = reinterpret_cast<TI*>(b.raw); 1841 1842 // in == NULL can happen if the track was flushed just after having 1843 // been enabled for mixing. 1844 if (in == NULL || (((uintptr_t)in) & 3)) { 1845 memset(out, 0, numFrames 1846 * channels * audio_bytes_per_sample(t->mMixerFormat)); 1847 ALOGE_IF((((uintptr_t)in) & 3), "process_NoResampleOneTrack: bus error: " 1848 "buffer %p track %p, channels %d, needs %#x", 1849 in, t, t->channelCount, t->needs); 1850 return; 1851 } 1852 1853 const size_t outFrames = b.frameCount; 1854 volumeMix<MIXTYPE, is_same<TI, float>::value, false> ( 1855 out, outFrames, in, aux, ramp, t); 1856 1857 out += outFrames * channels; 1858 if (aux != NULL) { 1859 aux += channels; 1860 } 1861 numFrames -= b.frameCount; 1862 1863 // release buffer 1864 t->bufferProvider->releaseBuffer(&b); 1865 } 1866 if (ramp) { 1867 t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value); 1868 } 1869 } 1870 1871 /* This track hook is called to do resampling then mixing, 1872 * pulling from the track's upstream AudioBufferProvider. 1873 * 1874 * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) 1875 * TO: int32_t (Q4.27) or float 1876 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float 1877 * TA: int32_t (Q4.27) 1878 */ 1879 template <int MIXTYPE, typename TO, typename TI, typename TA> 1880 void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux) 1881 { 1882 ALOGVV("track__Resample\n"); 1883 t->resampler->setSampleRate(t->sampleRate); 1884 const bool ramp = t->needsRamp(); 1885 if (ramp || aux != NULL) { 1886 // if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step. 1887 // if aux != NULL: resample with unity gain to temp buffer then apply send level. 1888 1889 t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT); 1890 memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO)); 1891 t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider); 1892 1893 volumeMix<MIXTYPE, is_same<TI, float>::value, true>( 1894 out, outFrameCount, temp, aux, ramp, t); 1895 1896 } else { // constant volume gain 1897 t->resampler->setVolume(t->mVolume[0], t->mVolume[1]); 1898 t->resampler->resample((int32_t*)out, outFrameCount, t->bufferProvider); 1899 } 1900 } 1901 1902 /* This track hook is called to mix a track, when no resampling is required. 1903 * The input buffer should be present in t->in. 1904 * 1905 * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration) 1906 * TO: int32_t (Q4.27) or float 1907 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float 1908 * TA: int32_t (Q4.27) 1909 */ 1910 template <int MIXTYPE, typename TO, typename TI, typename TA> 1911 void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount, 1912 TO* temp __unused, TA* aux) 1913 { 1914 ALOGVV("track__NoResample\n"); 1915 const TI *in = static_cast<const TI *>(t->in); 1916 1917 volumeMix<MIXTYPE, is_same<TI, float>::value, true>( 1918 out, frameCount, in, aux, t->needsRamp(), t); 1919 1920 // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels. 1921 // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels. 1922 in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * t->mMixerChannelCount; 1923 t->in = in; 1924 } 1925 1926 /* The Mixer engine generates either int32_t (Q4_27) or float data. 1927 * We use this function to convert the engine buffers 1928 * to the desired mixer output format, either int16_t (Q.15) or float. 1929 */ 1930 void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat, 1931 void *in, audio_format_t mixerInFormat, size_t sampleCount) 1932 { 1933 switch (mixerInFormat) { 1934 case AUDIO_FORMAT_PCM_FLOAT: 1935 switch (mixerOutFormat) { 1936 case AUDIO_FORMAT_PCM_FLOAT: 1937 memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out 1938 break; 1939 case AUDIO_FORMAT_PCM_16_BIT: 1940 memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount); 1941 break; 1942 default: 1943 LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat); 1944 break; 1945 } 1946 break; 1947 case AUDIO_FORMAT_PCM_16_BIT: 1948 switch (mixerOutFormat) { 1949 case AUDIO_FORMAT_PCM_FLOAT: 1950 memcpy_to_float_from_q4_27((float*)out, (int32_t*)in, sampleCount); 1951 break; 1952 case AUDIO_FORMAT_PCM_16_BIT: 1953 // two int16_t are produced per iteration 1954 ditherAndClamp((int32_t*)out, (int32_t*)in, sampleCount >> 1); 1955 break; 1956 default: 1957 LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat); 1958 break; 1959 } 1960 break; 1961 default: 1962 LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat); 1963 break; 1964 } 1965 } 1966 1967 /* Returns the proper track hook to use for mixing the track into the output buffer. 1968 */ 1969 AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount, 1970 audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused) 1971 { 1972 if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) { 1973 switch (trackType) { 1974 case TRACKTYPE_NOP: 1975 return track__nop; 1976 case TRACKTYPE_RESAMPLE: 1977 return track__genericResample; 1978 case TRACKTYPE_NORESAMPLEMONO: 1979 return track__16BitsMono; 1980 case TRACKTYPE_NORESAMPLE: 1981 return track__16BitsStereo; 1982 default: 1983 LOG_ALWAYS_FATAL("bad trackType: %d", trackType); 1984 break; 1985 } 1986 } 1987 LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS); 1988 switch (trackType) { 1989 case TRACKTYPE_NOP: 1990 return track__nop; 1991 case TRACKTYPE_RESAMPLE: 1992 switch (mixerInFormat) { 1993 case AUDIO_FORMAT_PCM_FLOAT: 1994 return (AudioMixer::hook_t) 1995 track__Resample<MIXTYPE_MULTI, float /*TO*/, float /*TI*/, int32_t /*TA*/>; 1996 case AUDIO_FORMAT_PCM_16_BIT: 1997 return (AudioMixer::hook_t)\ 1998 track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>; 1999 default: 2000 LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat); 2001 break; 2002 } 2003 break; 2004 case TRACKTYPE_NORESAMPLEMONO: 2005 switch (mixerInFormat) { 2006 case AUDIO_FORMAT_PCM_FLOAT: 2007 return (AudioMixer::hook_t) 2008 track__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>; 2009 case AUDIO_FORMAT_PCM_16_BIT: 2010 return (AudioMixer::hook_t) 2011 track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>; 2012 default: 2013 LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat); 2014 break; 2015 } 2016 break; 2017 case TRACKTYPE_NORESAMPLE: 2018 switch (mixerInFormat) { 2019 case AUDIO_FORMAT_PCM_FLOAT: 2020 return (AudioMixer::hook_t) 2021 track__NoResample<MIXTYPE_MULTI, float, float, int32_t>; 2022 case AUDIO_FORMAT_PCM_16_BIT: 2023 return (AudioMixer::hook_t) 2024 track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>; 2025 default: 2026 LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat); 2027 break; 2028 } 2029 break; 2030 default: 2031 LOG_ALWAYS_FATAL("bad trackType: %d", trackType); 2032 break; 2033 } 2034 return NULL; 2035 } 2036 2037 /* Returns the proper process hook for mixing tracks. Currently works only for 2038 * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling. 2039 * 2040 * TODO: Due to the special mixing considerations of duplicating to 2041 * a stereo output track, the input track cannot be MONO. This should be 2042 * prevented by the caller. 2043 */ 2044 AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t channelCount, 2045 audio_format_t mixerInFormat, audio_format_t mixerOutFormat) 2046 { 2047 if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK 2048 LOG_ALWAYS_FATAL("bad processType: %d", processType); 2049 return NULL; 2050 } 2051 if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) { 2052 return process__OneTrack16BitsStereoNoResampling; 2053 } 2054 LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS); 2055 switch (mixerInFormat) { 2056 case AUDIO_FORMAT_PCM_FLOAT: 2057 switch (mixerOutFormat) { 2058 case AUDIO_FORMAT_PCM_FLOAT: 2059 return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2060 float /*TO*/, float /*TI*/, int32_t /*TA*/>; 2061 case AUDIO_FORMAT_PCM_16_BIT: 2062 return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2063 int16_t, float, int32_t>; 2064 default: 2065 LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat); 2066 break; 2067 } 2068 break; 2069 case AUDIO_FORMAT_PCM_16_BIT: 2070 switch (mixerOutFormat) { 2071 case AUDIO_FORMAT_PCM_FLOAT: 2072 return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2073 float, int16_t, int32_t>; 2074 case AUDIO_FORMAT_PCM_16_BIT: 2075 return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2076 int16_t, int16_t, int32_t>; 2077 default: 2078 LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat); 2079 break; 2080 } 2081 break; 2082 default: 2083 LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat); 2084 break; 2085 } 2086 return NULL; 2087 } 2088 2089 // ---------------------------------------------------------------------------- 2090 } // namespace android 2091
