| 1 | /* |
| 2 | * Copyright (C) 2010 Google Inc. All rights reserved. |
| 3 | * |
| 4 | * Redistribution and use in source and binary forms, with or without |
| 5 | * modification, are permitted provided that the following conditions |
| 6 | * are met: |
| 7 | * |
| 8 | * 1. Redistributions of source code must retain the above copyright |
| 9 | * notice, this list of conditions and the following disclaimer. |
| 10 | * 2. Redistributions in binary form must reproduce the above copyright |
| 11 | * notice, this list of conditions and the following disclaimer in the |
| 12 | * documentation and/or other materials provided with the distribution. |
| 13 | * 3. Neither the name of Apple Inc. ("Apple") nor the names of |
| 14 | * its contributors may be used to endorse or promote products derived |
| 15 | * from this software without specific prior written permission. |
| 16 | * |
| 17 | * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY |
| 18 | * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
| 19 | * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
| 20 | * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
| 21 | * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
| 22 | * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
| 23 | * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND |
| 24 | * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
| 25 | * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF |
| 26 | * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| 27 | */ |
| 28 | |
| 29 | #include "config.h" |
| 30 | |
| 31 | #if ENABLE(WEB_AUDIO) |
| 32 | |
| 33 | #include "AudioBus.h" |
| 34 | |
| 35 | #include "DenormalDisabler.h" |
| 36 | |
| 37 | #include "SincResampler.h" |
| 38 | #include "VectorMath.h" |
| 39 | #include <algorithm> |
| 40 | #include <assert.h> |
| 41 | #include <math.h> |
| 42 | |
| 43 | namespace WebCore { |
| 44 | |
| 45 | using namespace VectorMath; |
| 46 | |
| 47 | const unsigned MaxBusChannels = 32; |
| 48 | |
| 49 | RefPtr<AudioBus> AudioBus::create(unsigned numberOfChannels, size_t length, bool allocate) |
| 50 | { |
| 51 | ASSERT(numberOfChannels <= MaxBusChannels); |
| 52 | if (numberOfChannels > MaxBusChannels) |
| 53 | return nullptr; |
| 54 | |
| 55 | return adoptRef(*new AudioBus(numberOfChannels, length, allocate)); |
| 56 | } |
| 57 | |
| 58 | AudioBus::AudioBus(unsigned numberOfChannels, size_t length, bool allocate) |
| 59 | : m_length(length) |
| 60 | , m_busGain(1) |
| 61 | , m_isFirstTime(true) |
| 62 | , m_sampleRate(0) |
| 63 | { |
| 64 | m_channels.reserveInitialCapacity(numberOfChannels); |
| 65 | |
| 66 | for (unsigned i = 0; i < numberOfChannels; ++i) { |
| 67 | auto channel = allocate ? std::make_unique<AudioChannel>(length) : std::make_unique<AudioChannel>(nullptr, length); |
| 68 | m_channels.uncheckedAppend(WTFMove(channel)); |
| 69 | } |
| 70 | |
| 71 | m_layout = LayoutCanonical; // for now this is the only layout we define |
| 72 | } |
| 73 | |
| 74 | void AudioBus::setChannelMemory(unsigned channelIndex, float* storage, size_t length) |
| 75 | { |
| 76 | if (channelIndex < m_channels.size()) { |
| 77 | channel(channelIndex)->set(storage, length); |
| 78 | m_length = length; // FIXME: verify that this length matches all the other channel lengths |
| 79 | } |
| 80 | } |
| 81 | |
| 82 | void AudioBus::resizeSmaller(size_t newLength) |
| 83 | { |
| 84 | ASSERT(newLength <= m_length); |
| 85 | if (newLength <= m_length) |
| 86 | m_length = newLength; |
| 87 | |
| 88 | for (unsigned i = 0; i < m_channels.size(); ++i) |
| 89 | m_channels[i]->resizeSmaller(newLength); |
| 90 | } |
| 91 | |
| 92 | void AudioBus::zero() |
| 93 | { |
| 94 | for (unsigned i = 0; i < m_channels.size(); ++i) |
| 95 | m_channels[i]->zero(); |
| 96 | } |
| 97 | |
| 98 | AudioChannel* AudioBus::channelByType(unsigned channelType) |
| 99 | { |
| 100 | // For now we only support canonical channel layouts... |
| 101 | if (m_layout != LayoutCanonical) |
| 102 | return 0; |
| 103 | |
| 104 | switch (numberOfChannels()) { |
| 105 | case 1: // mono |
| 106 | if (channelType == ChannelMono || channelType == ChannelLeft) |
| 107 | return channel(0); |
| 108 | return 0; |
| 109 | |
| 110 | case 2: // stereo |
| 111 | switch (channelType) { |
| 112 | case ChannelLeft: return channel(0); |
| 113 | case ChannelRight: return channel(1); |
| 114 | default: return 0; |
| 115 | } |
| 116 | |
| 117 | case 4: // quad |
| 118 | switch (channelType) { |
| 119 | case ChannelLeft: return channel(0); |
| 120 | case ChannelRight: return channel(1); |
| 121 | case ChannelSurroundLeft: return channel(2); |
| 122 | case ChannelSurroundRight: return channel(3); |
| 123 | default: return 0; |
| 124 | } |
| 125 | |
| 126 | case 5: // 5.0 |
| 127 | switch (channelType) { |
| 128 | case ChannelLeft: return channel(0); |
| 129 | case ChannelRight: return channel(1); |
| 130 | case ChannelCenter: return channel(2); |
| 131 | case ChannelSurroundLeft: return channel(3); |
| 132 | case ChannelSurroundRight: return channel(4); |
| 133 | default: return 0; |
| 134 | } |
| 135 | |
| 136 | case 6: // 5.1 |
| 137 | switch (channelType) { |
| 138 | case ChannelLeft: return channel(0); |
| 139 | case ChannelRight: return channel(1); |
| 140 | case ChannelCenter: return channel(2); |
| 141 | case ChannelLFE: return channel(3); |
| 142 | case ChannelSurroundLeft: return channel(4); |
| 143 | case ChannelSurroundRight: return channel(5); |
| 144 | default: return 0; |
| 145 | } |
| 146 | } |
| 147 | |
| 148 | ASSERT_NOT_REACHED(); |
| 149 | return 0; |
| 150 | } |
| 151 | |
| 152 | const AudioChannel* AudioBus::channelByType(unsigned type) const |
| 153 | { |
| 154 | return const_cast<AudioBus*>(this)->channelByType(type); |
| 155 | } |
| 156 | |
| 157 | // Returns true if the channel count and frame-size match. |
| 158 | bool AudioBus::topologyMatches(const AudioBus& bus) const |
| 159 | { |
| 160 | if (numberOfChannels() != bus.numberOfChannels()) |
| 161 | return false; // channel mismatch |
| 162 | |
| 163 | // Make sure source bus has enough frames. |
| 164 | if (length() > bus.length()) |
| 165 | return false; // frame-size mismatch |
| 166 | |
| 167 | return true; |
| 168 | } |
| 169 | |
| 170 | RefPtr<AudioBus> AudioBus::createBufferFromRange(const AudioBus* sourceBuffer, unsigned startFrame, unsigned endFrame) |
| 171 | { |
| 172 | size_t numberOfSourceFrames = sourceBuffer->length(); |
| 173 | unsigned numberOfChannels = sourceBuffer->numberOfChannels(); |
| 174 | |
| 175 | // Sanity checking |
| 176 | bool isRangeSafe = startFrame < endFrame && endFrame <= numberOfSourceFrames; |
| 177 | ASSERT(isRangeSafe); |
| 178 | if (!isRangeSafe) |
| 179 | return nullptr; |
| 180 | |
| 181 | size_t rangeLength = endFrame - startFrame; |
| 182 | |
| 183 | RefPtr<AudioBus> audioBus = create(numberOfChannels, rangeLength); |
| 184 | audioBus->setSampleRate(sourceBuffer->sampleRate()); |
| 185 | |
| 186 | for (unsigned i = 0; i < numberOfChannels; ++i) |
| 187 | audioBus->channel(i)->copyFromRange(sourceBuffer->channel(i), startFrame, endFrame); |
| 188 | |
| 189 | return audioBus; |
| 190 | } |
| 191 | |
| 192 | float AudioBus::maxAbsValue() const |
| 193 | { |
| 194 | float max = 0.0f; |
| 195 | for (unsigned i = 0; i < numberOfChannels(); ++i) { |
| 196 | const AudioChannel* channel = this->channel(i); |
| 197 | max = std::max(max, channel->maxAbsValue()); |
| 198 | } |
| 199 | |
| 200 | return max; |
| 201 | } |
| 202 | |
| 203 | void AudioBus::normalize() |
| 204 | { |
| 205 | float max = maxAbsValue(); |
| 206 | if (max) |
| 207 | scale(1.0f / max); |
| 208 | } |
| 209 | |
| 210 | void AudioBus::scale(float scale) |
| 211 | { |
| 212 | for (unsigned i = 0; i < numberOfChannels(); ++i) |
| 213 | channel(i)->scale(scale); |
| 214 | } |
| 215 | |
| 216 | void AudioBus::copyFromRange(const AudioBus& sourceBus, unsigned startFrame, unsigned endFrame) |
| 217 | { |
| 218 | if (!topologyMatches(sourceBus)) { |
| 219 | ASSERT_NOT_REACHED(); |
| 220 | zero(); |
| 221 | return; |
| 222 | } |
| 223 | |
| 224 | size_t numberOfSourceFrames = sourceBus.length(); |
| 225 | bool isRangeSafe = startFrame < endFrame && endFrame <= numberOfSourceFrames; |
| 226 | ASSERT(isRangeSafe); |
| 227 | if (!isRangeSafe) { |
| 228 | zero(); |
| 229 | return; |
| 230 | } |
| 231 | |
| 232 | unsigned numberOfChannels = this->numberOfChannels(); |
| 233 | ASSERT(numberOfChannels <= MaxBusChannels); |
| 234 | if (numberOfChannels > MaxBusChannels) { |
| 235 | zero(); |
| 236 | return; |
| 237 | } |
| 238 | |
| 239 | for (unsigned i = 0; i < numberOfChannels; ++i) |
| 240 | channel(i)->copyFromRange(sourceBus.channel(i), startFrame, endFrame); |
| 241 | } |
| 242 | |
| 243 | void AudioBus::copyFrom(const AudioBus& sourceBus, ChannelInterpretation channelInterpretation) |
| 244 | { |
| 245 | if (&sourceBus == this) |
| 246 | return; |
| 247 | |
| 248 | unsigned numberOfSourceChannels = sourceBus.numberOfChannels(); |
| 249 | unsigned numberOfDestinationChannels = numberOfChannels(); |
| 250 | |
| 251 | if (numberOfDestinationChannels == numberOfSourceChannels) { |
| 252 | for (unsigned i = 0; i < numberOfSourceChannels; ++i) |
| 253 | channel(i)->copyFrom(sourceBus.channel(i)); |
| 254 | } else { |
| 255 | switch (channelInterpretation) { |
| 256 | case Speakers: |
| 257 | speakersCopyFrom(sourceBus); |
| 258 | break; |
| 259 | case Discrete: |
| 260 | discreteCopyFrom(sourceBus); |
| 261 | break; |
| 262 | default: |
| 263 | ASSERT_NOT_REACHED(); |
| 264 | } |
| 265 | } |
| 266 | } |
| 267 | |
| 268 | void AudioBus::sumFrom(const AudioBus& sourceBus, ChannelInterpretation channelInterpretation) |
| 269 | { |
| 270 | if (&sourceBus == this) |
| 271 | return; |
| 272 | |
| 273 | unsigned numberOfSourceChannels = sourceBus.numberOfChannels(); |
| 274 | unsigned numberOfDestinationChannels = numberOfChannels(); |
| 275 | |
| 276 | if (numberOfDestinationChannels == numberOfSourceChannels) { |
| 277 | for (unsigned i = 0; i < numberOfSourceChannels; ++i) |
| 278 | channel(i)->sumFrom(sourceBus.channel(i)); |
| 279 | } else { |
| 280 | switch (channelInterpretation) { |
| 281 | case Speakers: |
| 282 | speakersSumFrom(sourceBus); |
| 283 | break; |
| 284 | case Discrete: |
| 285 | discreteSumFrom(sourceBus); |
| 286 | break; |
| 287 | default: |
| 288 | ASSERT_NOT_REACHED(); |
| 289 | } |
| 290 | } |
| 291 | } |
| 292 | |
| 293 | void AudioBus::speakersCopyFrom(const AudioBus& sourceBus) |
| 294 | { |
| 295 | // FIXME: Implement down mixing 5.1 to stereo. |
| 296 | // https://bugs.webkit.org/show_bug.cgi?id=79192 |
| 297 | |
| 298 | unsigned numberOfSourceChannels = sourceBus.numberOfChannels(); |
| 299 | unsigned numberOfDestinationChannels = numberOfChannels(); |
| 300 | |
| 301 | if (numberOfDestinationChannels == 2 && numberOfSourceChannels == 1) { |
| 302 | // Handle mono -> stereo case (for now simply copy mono channel into both left and right) |
| 303 | // FIXME: Really we should apply an equal-power scaling factor here, since we're effectively panning center... |
| 304 | const AudioChannel* sourceChannel = sourceBus.channel(0); |
| 305 | channel(0)->copyFrom(sourceChannel); |
| 306 | channel(1)->copyFrom(sourceChannel); |
| 307 | } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 2) { |
| 308 | // Handle stereo -> mono case. output = 0.5 * (input.L + input.R). |
| 309 | AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus); |
| 310 | |
| 311 | const float* sourceL = sourceBusSafe.channelByType(ChannelLeft)->data(); |
| 312 | const float* sourceR = sourceBusSafe.channelByType(ChannelRight)->data(); |
| 313 | |
| 314 | float* destination = channelByType(ChannelLeft)->mutableData(); |
| 315 | vadd(sourceL, 1, sourceR, 1, destination, 1, length()); |
| 316 | float scale = 0.5; |
| 317 | vsmul(destination, 1, &scale, destination, 1, length()); |
| 318 | } else if (numberOfDestinationChannels == 6 && numberOfSourceChannels == 1) { |
| 319 | // Handle mono -> 5.1 case, copy mono channel to center. |
| 320 | channel(2)->copyFrom(sourceBus.channel(0)); |
| 321 | channel(0)->zero(); |
| 322 | channel(1)->zero(); |
| 323 | channel(3)->zero(); |
| 324 | channel(4)->zero(); |
| 325 | channel(5)->zero(); |
| 326 | } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 6) { |
| 327 | // Handle 5.1 -> mono case. |
| 328 | zero(); |
| 329 | speakersSumFrom5_1_ToMono(sourceBus); |
| 330 | } else { |
| 331 | // Fallback for unknown combinations. |
| 332 | discreteCopyFrom(sourceBus); |
| 333 | } |
| 334 | } |
| 335 | |
| 336 | void AudioBus::(const AudioBus& sourceBus) |
| 337 | { |
| 338 | // FIXME: Implement down mixing 5.1 to stereo. |
| 339 | // https://bugs.webkit.org/show_bug.cgi?id=79192 |
| 340 | |
| 341 | unsigned numberOfSourceChannels = sourceBus.numberOfChannels(); |
| 342 | unsigned numberOfDestinationChannels = numberOfChannels(); |
| 343 | |
| 344 | if (numberOfDestinationChannels == 2 && numberOfSourceChannels == 1) { |
| 345 | // Handle mono -> stereo case (summing mono channel into both left and right). |
| 346 | const AudioChannel* sourceChannel = sourceBus.channel(0); |
| 347 | channel(0)->sumFrom(sourceChannel); |
| 348 | channel(1)->sumFrom(sourceChannel); |
| 349 | } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 2) { |
| 350 | // Handle stereo -> mono case. output += 0.5 * (input.L + input.R). |
| 351 | AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus); |
| 352 | |
| 353 | const float* sourceL = sourceBusSafe.channelByType(ChannelLeft)->data(); |
| 354 | const float* sourceR = sourceBusSafe.channelByType(ChannelRight)->data(); |
| 355 | |
| 356 | float* destination = channelByType(ChannelLeft)->mutableData(); |
| 357 | float scale = 0.5; |
| 358 | vsma(sourceL, 1, &scale, destination, 1, length()); |
| 359 | vsma(sourceR, 1, &scale, destination, 1, length()); |
| 360 | } else if (numberOfDestinationChannels == 6 && numberOfSourceChannels == 1) { |
| 361 | // Handle mono -> 5.1 case, sum mono channel into center. |
| 362 | channel(2)->sumFrom(sourceBus.channel(0)); |
| 363 | } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 6) { |
| 364 | // Handle 5.1 -> mono case. |
| 365 | speakersSumFrom5_1_ToMono(sourceBus); |
| 366 | } else { |
| 367 | // Fallback for unknown combinations. |
| 368 | discreteSumFrom(sourceBus); |
| 369 | } |
| 370 | } |
| 371 | |
| 372 | void AudioBus::(const AudioBus& sourceBus) |
| 373 | { |
| 374 | AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus); |
| 375 | |
| 376 | const float* sourceL = sourceBusSafe.channelByType(ChannelLeft)->data(); |
| 377 | const float* sourceR = sourceBusSafe.channelByType(ChannelRight)->data(); |
| 378 | const float* sourceC = sourceBusSafe.channelByType(ChannelCenter)->data(); |
| 379 | const float* sourceSL = sourceBusSafe.channelByType(ChannelSurroundLeft)->data(); |
| 380 | const float* sourceSR = sourceBusSafe.channelByType(ChannelSurroundRight)->data(); |
| 381 | |
| 382 | float* destination = channelByType(ChannelLeft)->mutableData(); |
| 383 | |
| 384 | AudioFloatArray temp(length()); |
| 385 | float* tempData = temp.data(); |
| 386 | |
| 387 | // Sum in L and R. |
| 388 | vadd(sourceL, 1, sourceR, 1, tempData, 1, length()); |
| 389 | float scale = 0.7071; |
| 390 | vsmul(tempData, 1, &scale, tempData, 1, length()); |
| 391 | vadd(tempData, 1, destination, 1, destination, 1, length()); |
| 392 | |
| 393 | // Sum in SL and SR. |
| 394 | vadd(sourceSL, 1, sourceSR, 1, tempData, 1, length()); |
| 395 | scale = 0.5; |
| 396 | vsmul(tempData, 1, &scale, tempData, 1, length()); |
| 397 | vadd(tempData, 1, destination, 1, destination, 1, length()); |
| 398 | |
| 399 | // Sum in center. |
| 400 | vadd(sourceC, 1, destination, 1, destination, 1, length()); |
| 401 | } |
| 402 | |
| 403 | void AudioBus::discreteCopyFrom(const AudioBus& sourceBus) |
| 404 | { |
| 405 | unsigned numberOfSourceChannels = sourceBus.numberOfChannels(); |
| 406 | unsigned numberOfDestinationChannels = numberOfChannels(); |
| 407 | |
| 408 | if (numberOfDestinationChannels < numberOfSourceChannels) { |
| 409 | // Down-mix by copying channels and dropping the remaining. |
| 410 | for (unsigned i = 0; i < numberOfDestinationChannels; ++i) |
| 411 | channel(i)->copyFrom(sourceBus.channel(i)); |
| 412 | } else if (numberOfDestinationChannels > numberOfSourceChannels) { |
| 413 | // Up-mix by copying as many channels as we have, then zeroing remaining channels. |
| 414 | for (unsigned i = 0; i < numberOfSourceChannels; ++i) |
| 415 | channel(i)->copyFrom(sourceBus.channel(i)); |
| 416 | for (unsigned i = numberOfSourceChannels; i < numberOfDestinationChannels; ++i) |
| 417 | channel(i)->zero(); |
| 418 | } |
| 419 | } |
| 420 | |
| 421 | void AudioBus::discreteSumFrom(const AudioBus& sourceBus) |
| 422 | { |
| 423 | unsigned numberOfSourceChannels = sourceBus.numberOfChannels(); |
| 424 | unsigned numberOfDestinationChannels = numberOfChannels(); |
| 425 | |
| 426 | if (numberOfDestinationChannels < numberOfSourceChannels) { |
| 427 | // Down-mix by summing channels and dropping the remaining. |
| 428 | for (unsigned i = 0; i < numberOfDestinationChannels; ++i) |
| 429 | channel(i)->sumFrom(sourceBus.channel(i)); |
| 430 | } else if (numberOfDestinationChannels > numberOfSourceChannels) { |
| 431 | // Up-mix by summing as many channels as we have. |
| 432 | for (unsigned i = 0; i < numberOfSourceChannels; ++i) |
| 433 | channel(i)->sumFrom(sourceBus.channel(i)); |
| 434 | } |
| 435 | } |
| 436 | |
| 437 | void AudioBus::copyWithGainFrom(const AudioBus &sourceBus, float* lastMixGain, float targetGain) |
| 438 | { |
| 439 | if (!topologyMatches(sourceBus)) { |
| 440 | ASSERT_NOT_REACHED(); |
| 441 | zero(); |
| 442 | return; |
| 443 | } |
| 444 | |
| 445 | if (sourceBus.isSilent()) { |
| 446 | zero(); |
| 447 | return; |
| 448 | } |
| 449 | |
| 450 | unsigned numberOfChannels = this->numberOfChannels(); |
| 451 | ASSERT(numberOfChannels <= MaxBusChannels); |
| 452 | if (numberOfChannels > MaxBusChannels) |
| 453 | return; |
| 454 | |
| 455 | // If it is copying from the same bus and no need to change gain, just return. |
| 456 | if (this == &sourceBus && *lastMixGain == targetGain && targetGain == 1) |
| 457 | return; |
| 458 | |
| 459 | AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus); |
| 460 | const float* sources[MaxBusChannels]; |
| 461 | float* destinations[MaxBusChannels]; |
| 462 | |
| 463 | for (unsigned i = 0; i < numberOfChannels; ++i) { |
| 464 | sources[i] = sourceBusSafe.channel(i)->data(); |
| 465 | destinations[i] = channel(i)->mutableData(); |
| 466 | } |
| 467 | |
| 468 | // We don't want to suddenly change the gain from mixing one time slice to the next, |
| 469 | // so we "de-zipper" by slowly changing the gain each sample-frame until we've achieved the target gain. |
| 470 | |
| 471 | // Take master bus gain into account as well as the targetGain. |
| 472 | float totalDesiredGain = static_cast<float>(m_busGain * targetGain); |
| 473 | |
| 474 | // First time, snap directly to totalDesiredGain. |
| 475 | float gain = static_cast<float>(m_isFirstTime ? totalDesiredGain : *lastMixGain); |
| 476 | m_isFirstTime = false; |
| 477 | |
| 478 | const float DezipperRate = 0.005f; |
| 479 | unsigned framesToProcess = length(); |
| 480 | |
| 481 | // If the gain is within epsilon of totalDesiredGain, we can skip dezippering. |
| 482 | // FIXME: this value may need tweaking. |
| 483 | const float epsilon = 0.001f; |
| 484 | float gainDiff = fabs(totalDesiredGain - gain); |
| 485 | |
| 486 | // Number of frames to de-zipper before we are close enough to the target gain. |
| 487 | // FIXME: framesToDezipper could be smaller when target gain is close enough within this process loop. |
| 488 | unsigned framesToDezipper = (gainDiff < epsilon) ? 0 : framesToProcess; |
| 489 | |
| 490 | if (framesToDezipper) { |
| 491 | if (!m_dezipperGainValues.get() || m_dezipperGainValues->size() < framesToDezipper) |
| 492 | m_dezipperGainValues = std::make_unique<AudioFloatArray>(framesToDezipper); |
| 493 | |
| 494 | float* gainValues = m_dezipperGainValues->data(); |
| 495 | for (unsigned i = 0; i < framesToDezipper; ++i) { |
| 496 | gain += (totalDesiredGain - gain) * DezipperRate; |
| 497 | |
| 498 | // FIXME: If we are clever enough in calculating the framesToDezipper value, we can probably get |
| 499 | // rid of this DenormalDisabler::flushDenormalFloatToZero() call. |
| 500 | gain = DenormalDisabler::flushDenormalFloatToZero(gain); |
| 501 | *gainValues++ = gain; |
| 502 | } |
| 503 | |
| 504 | for (unsigned channelIndex = 0; channelIndex < numberOfChannels; ++channelIndex) { |
| 505 | vmul(sources[channelIndex], 1, m_dezipperGainValues->data(), 1, destinations[channelIndex], 1, framesToDezipper); |
| 506 | sources[channelIndex] += framesToDezipper; |
| 507 | destinations[channelIndex] += framesToDezipper; |
| 508 | } |
| 509 | } else |
| 510 | gain = totalDesiredGain; |
| 511 | |
| 512 | // Apply constant gain after de-zippering has converged on target gain. |
| 513 | if (framesToDezipper < framesToProcess) { |
| 514 | // Handle gains of 0 and 1 (exactly) specially. |
| 515 | if (gain == 1) { |
| 516 | for (unsigned channelIndex = 0; channelIndex < numberOfChannels; ++channelIndex) |
| 517 | memcpy(destinations[channelIndex], sources[channelIndex], (framesToProcess - framesToDezipper) * sizeof(*destinations[channelIndex])); |
| 518 | } else if (!gain) { |
| 519 | for (unsigned channelIndex = 0; channelIndex < numberOfChannels; ++channelIndex) |
| 520 | memset(destinations[channelIndex], 0, (framesToProcess - framesToDezipper) * sizeof(*destinations[channelIndex])); |
| 521 | } else { |
| 522 | for (unsigned channelIndex = 0; channelIndex < numberOfChannels; ++channelIndex) |
| 523 | vsmul(sources[channelIndex], 1, &gain, destinations[channelIndex], 1, framesToProcess - framesToDezipper); |
| 524 | } |
| 525 | } |
| 526 | |
| 527 | // Save the target gain as the starting point for next time around. |
| 528 | *lastMixGain = gain; |
| 529 | } |
| 530 | |
| 531 | void AudioBus::copyWithSampleAccurateGainValuesFrom(const AudioBus &sourceBus, float* gainValues, unsigned numberOfGainValues) |
| 532 | { |
| 533 | // Make sure we're processing from the same type of bus. |
| 534 | // We *are* able to process from mono -> stereo |
| 535 | if (sourceBus.numberOfChannels() != 1 && !topologyMatches(sourceBus)) { |
| 536 | ASSERT_NOT_REACHED(); |
| 537 | return; |
| 538 | } |
| 539 | |
| 540 | if (!gainValues || numberOfGainValues > sourceBus.length()) { |
| 541 | ASSERT_NOT_REACHED(); |
| 542 | return; |
| 543 | } |
| 544 | |
| 545 | if (sourceBus.length() == numberOfGainValues && sourceBus.length() == length() && sourceBus.isSilent()) { |
| 546 | zero(); |
| 547 | return; |
| 548 | } |
| 549 | |
| 550 | // We handle both the 1 -> N and N -> N case here. |
| 551 | const float* source = sourceBus.channel(0)->data(); |
| 552 | for (unsigned channelIndex = 0; channelIndex < numberOfChannels(); ++channelIndex) { |
| 553 | if (sourceBus.numberOfChannels() == numberOfChannels()) |
| 554 | source = sourceBus.channel(channelIndex)->data(); |
| 555 | float* destination = channel(channelIndex)->mutableData(); |
| 556 | vmul(source, 1, gainValues, 1, destination, 1, numberOfGainValues); |
| 557 | } |
| 558 | } |
| 559 | |
| 560 | RefPtr<AudioBus> AudioBus::createBySampleRateConverting(const AudioBus* sourceBus, bool mixToMono, double newSampleRate) |
| 561 | { |
| 562 | // sourceBus's sample-rate must be known. |
| 563 | ASSERT(sourceBus && sourceBus->sampleRate()); |
| 564 | if (!sourceBus || !sourceBus->sampleRate()) |
| 565 | return nullptr; |
| 566 | |
| 567 | double sourceSampleRate = sourceBus->sampleRate(); |
| 568 | double destinationSampleRate = newSampleRate; |
| 569 | double sampleRateRatio = sourceSampleRate / destinationSampleRate; |
| 570 | unsigned numberOfSourceChannels = sourceBus->numberOfChannels(); |
| 571 | |
| 572 | if (numberOfSourceChannels == 1) |
| 573 | mixToMono = false; // already mono |
| 574 | |
| 575 | if (sourceSampleRate == destinationSampleRate) { |
| 576 | // No sample-rate conversion is necessary. |
| 577 | if (mixToMono) |
| 578 | return AudioBus::createByMixingToMono(sourceBus); |
| 579 | |
| 580 | // Return exact copy. |
| 581 | return AudioBus::createBufferFromRange(sourceBus, 0, sourceBus->length()); |
| 582 | } |
| 583 | |
| 584 | if (sourceBus->isSilent()) { |
| 585 | RefPtr<AudioBus> silentBus = create(numberOfSourceChannels, sourceBus->length() / sampleRateRatio); |
| 586 | silentBus->setSampleRate(newSampleRate); |
| 587 | return silentBus; |
| 588 | } |
| 589 | |
| 590 | // First, mix to mono (if necessary) then sample-rate convert. |
| 591 | const AudioBus* resamplerSourceBus; |
| 592 | RefPtr<AudioBus> mixedMonoBus; |
| 593 | if (mixToMono) { |
| 594 | mixedMonoBus = AudioBus::createByMixingToMono(sourceBus); |
| 595 | resamplerSourceBus = mixedMonoBus.get(); |
| 596 | } else { |
| 597 | // Directly resample without down-mixing. |
| 598 | resamplerSourceBus = sourceBus; |
| 599 | } |
| 600 | |
| 601 | // Calculate destination length based on the sample-rates. |
| 602 | int sourceLength = resamplerSourceBus->length(); |
| 603 | int destinationLength = sourceLength / sampleRateRatio; |
| 604 | |
| 605 | // Create destination bus with same number of channels. |
| 606 | unsigned numberOfDestinationChannels = resamplerSourceBus->numberOfChannels(); |
| 607 | RefPtr<AudioBus> destinationBus = create(numberOfDestinationChannels, destinationLength); |
| 608 | |
| 609 | // Sample-rate convert each channel. |
| 610 | for (unsigned i = 0; i < numberOfDestinationChannels; ++i) { |
| 611 | const float* source = resamplerSourceBus->channel(i)->data(); |
| 612 | float* destination = destinationBus->channel(i)->mutableData(); |
| 613 | |
| 614 | SincResampler resampler(sampleRateRatio); |
| 615 | resampler.process(source, destination, sourceLength); |
| 616 | } |
| 617 | |
| 618 | destinationBus->clearSilentFlag(); |
| 619 | destinationBus->setSampleRate(newSampleRate); |
| 620 | return destinationBus; |
| 621 | } |
| 622 | |
| 623 | RefPtr<AudioBus> AudioBus::createByMixingToMono(const AudioBus* sourceBus) |
| 624 | { |
| 625 | if (sourceBus->isSilent()) |
| 626 | return create(1, sourceBus->length()); |
| 627 | |
| 628 | switch (sourceBus->numberOfChannels()) { |
| 629 | case 1: |
| 630 | // Simply create an exact copy. |
| 631 | return AudioBus::createBufferFromRange(sourceBus, 0, sourceBus->length()); |
| 632 | case 2: |
| 633 | { |
| 634 | unsigned n = sourceBus->length(); |
| 635 | RefPtr<AudioBus> destinationBus = create(1, n); |
| 636 | |
| 637 | const float* sourceL = sourceBus->channel(0)->data(); |
| 638 | const float* sourceR = sourceBus->channel(1)->data(); |
| 639 | float* destination = destinationBus->channel(0)->mutableData(); |
| 640 | |
| 641 | // Do the mono mixdown. |
| 642 | for (unsigned i = 0; i < n; ++i) |
| 643 | destination[i] = (sourceL[i] + sourceR[i]) / 2; |
| 644 | |
| 645 | destinationBus->clearSilentFlag(); |
| 646 | destinationBus->setSampleRate(sourceBus->sampleRate()); |
| 647 | return destinationBus; |
| 648 | } |
| 649 | } |
| 650 | |
| 651 | ASSERT_NOT_REACHED(); |
| 652 | return nullptr; |
| 653 | } |
| 654 | |
| 655 | bool AudioBus::isSilent() const |
| 656 | { |
| 657 | for (size_t i = 0; i < m_channels.size(); ++i) { |
| 658 | if (!m_channels[i]->isSilent()) |
| 659 | return false; |
| 660 | } |
| 661 | return true; |
| 662 | } |
| 663 | |
| 664 | void AudioBus::clearSilentFlag() |
| 665 | { |
| 666 | for (size_t i = 0; i < m_channels.size(); ++i) |
| 667 | m_channels[i]->clearSilentFlag(); |
| 668 | } |
| 669 | |
| 670 | } // WebCore |
| 671 | |
| 672 | #endif // ENABLE(WEB_AUDIO) |
| 673 | |