The Android Open Source Project | 9066cfe | 2009-03-03 19:31:44 -0800 | [diff] [blame] | 1 | /* //device/extlibs/pv/android/AudioTrack.cpp |
| 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 | |
| 19 | //#define LOG_NDEBUG 0 |
| 20 | #define LOG_TAG "AudioTrack" |
| 21 | |
| 22 | #include <stdint.h> |
| 23 | #include <sys/types.h> |
| 24 | #include <limits.h> |
| 25 | |
| 26 | #include <sched.h> |
| 27 | #include <sys/resource.h> |
| 28 | |
| 29 | #include <private/media/AudioTrackShared.h> |
| 30 | |
| 31 | #include <media/AudioSystem.h> |
| 32 | #include <media/AudioTrack.h> |
| 33 | |
| 34 | #include <utils/Log.h> |
| 35 | #include <utils/MemoryDealer.h> |
| 36 | #include <utils/Parcel.h> |
| 37 | #include <utils/IPCThreadState.h> |
| 38 | #include <utils/Timers.h> |
| 39 | #include <cutils/atomic.h> |
| 40 | |
| 41 | #define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) |
| 42 | #define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) |
| 43 | |
| 44 | namespace android { |
| 45 | |
| 46 | // --------------------------------------------------------------------------- |
| 47 | |
| 48 | AudioTrack::AudioTrack() |
| 49 | : mStatus(NO_INIT) |
| 50 | { |
| 51 | } |
| 52 | |
| 53 | AudioTrack::AudioTrack( |
| 54 | int streamType, |
| 55 | uint32_t sampleRate, |
| 56 | int format, |
| 57 | int channelCount, |
| 58 | int frameCount, |
| 59 | uint32_t flags, |
| 60 | callback_t cbf, |
| 61 | void* user, |
| 62 | int notificationFrames) |
| 63 | : mStatus(NO_INIT) |
| 64 | { |
| 65 | mStatus = set(streamType, sampleRate, format, channelCount, |
| 66 | frameCount, flags, cbf, user, notificationFrames, 0); |
| 67 | } |
| 68 | |
| 69 | AudioTrack::AudioTrack( |
| 70 | int streamType, |
| 71 | uint32_t sampleRate, |
| 72 | int format, |
| 73 | int channelCount, |
| 74 | const sp<IMemory>& sharedBuffer, |
| 75 | uint32_t flags, |
| 76 | callback_t cbf, |
| 77 | void* user, |
| 78 | int notificationFrames) |
| 79 | : mStatus(NO_INIT) |
| 80 | { |
| 81 | mStatus = set(streamType, sampleRate, format, channelCount, |
| 82 | 0, flags, cbf, user, notificationFrames, sharedBuffer); |
| 83 | } |
| 84 | |
| 85 | AudioTrack::~AudioTrack() |
| 86 | { |
| 87 | LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); |
| 88 | |
| 89 | if (mStatus == NO_ERROR) { |
| 90 | // Make sure that callback function exits in the case where |
| 91 | // it is looping on buffer full condition in obtainBuffer(). |
| 92 | // Otherwise the callback thread will never exit. |
| 93 | stop(); |
| 94 | if (mAudioTrackThread != 0) { |
| 95 | mCblk->cv.signal(); |
| 96 | mAudioTrackThread->requestExitAndWait(); |
| 97 | mAudioTrackThread.clear(); |
| 98 | } |
| 99 | mAudioTrack.clear(); |
| 100 | IPCThreadState::self()->flushCommands(); |
| 101 | } |
| 102 | } |
| 103 | |
| 104 | status_t AudioTrack::set( |
| 105 | int streamType, |
| 106 | uint32_t sampleRate, |
| 107 | int format, |
| 108 | int channelCount, |
| 109 | int frameCount, |
| 110 | uint32_t flags, |
| 111 | callback_t cbf, |
| 112 | void* user, |
| 113 | int notificationFrames, |
| 114 | const sp<IMemory>& sharedBuffer, |
| 115 | bool threadCanCallJava) |
| 116 | { |
| 117 | |
| 118 | LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); |
| 119 | |
| 120 | if (mAudioFlinger != 0) { |
| 121 | LOGE("Track already in use"); |
| 122 | return INVALID_OPERATION; |
| 123 | } |
| 124 | |
| 125 | const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); |
| 126 | if (audioFlinger == 0) { |
| 127 | LOGE("Could not get audioflinger"); |
| 128 | return NO_INIT; |
| 129 | } |
| 130 | int afSampleRate; |
| 131 | if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { |
| 132 | return NO_INIT; |
| 133 | } |
| 134 | int afFrameCount; |
| 135 | if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { |
| 136 | return NO_INIT; |
| 137 | } |
| 138 | uint32_t afLatency; |
| 139 | if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { |
| 140 | return NO_INIT; |
| 141 | } |
| 142 | |
| 143 | // handle default values first. |
| 144 | if (streamType == AudioSystem::DEFAULT) { |
| 145 | streamType = AudioSystem::MUSIC; |
| 146 | } |
| 147 | if (sampleRate == 0) { |
| 148 | sampleRate = afSampleRate; |
| 149 | } |
| 150 | // these below should probably come from the audioFlinger too... |
| 151 | if (format == 0) { |
| 152 | format = AudioSystem::PCM_16_BIT; |
| 153 | } |
| 154 | if (channelCount == 0) { |
| 155 | channelCount = 2; |
| 156 | } |
| 157 | |
| 158 | // validate parameters |
| 159 | if (((format != AudioSystem::PCM_8_BIT) || sharedBuffer != 0) && |
| 160 | (format != AudioSystem::PCM_16_BIT)) { |
| 161 | LOGE("Invalid format"); |
| 162 | return BAD_VALUE; |
| 163 | } |
| 164 | if (channelCount != 1 && channelCount != 2) { |
| 165 | LOGE("Invalid channel number"); |
| 166 | return BAD_VALUE; |
| 167 | } |
| 168 | |
| 169 | // Ensure that buffer depth covers at least audio hardware latency |
| 170 | uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); |
| 171 | if (minBufCount < 2) minBufCount = 2; |
| 172 | |
| 173 | // When playing from shared buffer, playback will start even if last audioflinger |
| 174 | // block is partly filled. |
| 175 | if (sharedBuffer != 0 && minBufCount > 1) { |
| 176 | minBufCount--; |
| 177 | } |
| 178 | |
| 179 | int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; |
| 180 | |
| 181 | if (sharedBuffer == 0) { |
| 182 | if (frameCount == 0) { |
| 183 | frameCount = minFrameCount; |
| 184 | } |
| 185 | if (notificationFrames == 0) { |
| 186 | notificationFrames = frameCount/2; |
| 187 | } |
| 188 | // Make sure that application is notified with sufficient margin |
| 189 | // before underrun |
| 190 | if (notificationFrames > frameCount/2) { |
| 191 | notificationFrames = frameCount/2; |
| 192 | } |
| 193 | } else { |
| 194 | // Ensure that buffer alignment matches channelcount |
| 195 | if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) { |
| 196 | LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount); |
| 197 | return BAD_VALUE; |
| 198 | } |
| 199 | frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); |
| 200 | } |
| 201 | |
| 202 | if (frameCount < minFrameCount) { |
| 203 | LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount); |
| 204 | return BAD_VALUE; |
| 205 | } |
| 206 | |
| 207 | // create the track |
| 208 | status_t status; |
| 209 | sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), |
| 210 | streamType, sampleRate, format, channelCount, frameCount, flags, sharedBuffer, &status); |
| 211 | |
| 212 | if (track == 0) { |
| 213 | LOGE("AudioFlinger could not create track, status: %d", status); |
| 214 | return status; |
| 215 | } |
| 216 | sp<IMemory> cblk = track->getCblk(); |
| 217 | if (cblk == 0) { |
| 218 | LOGE("Could not get control block"); |
| 219 | return NO_INIT; |
| 220 | } |
| 221 | if (cbf != 0) { |
| 222 | mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); |
| 223 | if (mAudioTrackThread == 0) { |
| 224 | LOGE("Could not create callback thread"); |
| 225 | return NO_INIT; |
| 226 | } |
| 227 | } |
| 228 | |
| 229 | mStatus = NO_ERROR; |
| 230 | |
| 231 | mAudioFlinger = audioFlinger; |
| 232 | mAudioTrack = track; |
| 233 | mCblkMemory = cblk; |
| 234 | mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); |
| 235 | mCblk->out = 1; |
| 236 | // Update buffer size in case it has been limited by AudioFlinger during track creation |
| 237 | mFrameCount = mCblk->frameCount; |
| 238 | if (sharedBuffer == 0) { |
| 239 | mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); |
| 240 | } else { |
| 241 | mCblk->buffers = sharedBuffer->pointer(); |
| 242 | // Force buffer full condition as data is already present in shared memory |
| 243 | mCblk->stepUser(mFrameCount); |
| 244 | } |
| 245 | mCblk->volume[0] = mCblk->volume[1] = 0x1000; |
| 246 | mVolume[LEFT] = 1.0f; |
| 247 | mVolume[RIGHT] = 1.0f; |
| 248 | mSampleRate = sampleRate; |
| 249 | mStreamType = streamType; |
| 250 | mFormat = format; |
| 251 | mChannelCount = channelCount; |
| 252 | mSharedBuffer = sharedBuffer; |
| 253 | mMuted = false; |
| 254 | mActive = 0; |
| 255 | mCbf = cbf; |
| 256 | mNotificationFrames = notificationFrames; |
| 257 | mRemainingFrames = notificationFrames; |
| 258 | mUserData = user; |
| 259 | mLatency = afLatency + (1000*mFrameCount) / mSampleRate; |
| 260 | mLoopCount = 0; |
| 261 | mMarkerPosition = 0; |
| 262 | mNewPosition = 0; |
| 263 | mUpdatePeriod = 0; |
| 264 | |
| 265 | return NO_ERROR; |
| 266 | } |
| 267 | |
| 268 | status_t AudioTrack::initCheck() const |
| 269 | { |
| 270 | return mStatus; |
| 271 | } |
| 272 | |
| 273 | // ------------------------------------------------------------------------- |
| 274 | |
| 275 | uint32_t AudioTrack::latency() const |
| 276 | { |
| 277 | return mLatency; |
| 278 | } |
| 279 | |
| 280 | int AudioTrack::streamType() const |
| 281 | { |
| 282 | return mStreamType; |
| 283 | } |
| 284 | |
| 285 | uint32_t AudioTrack::sampleRate() const |
| 286 | { |
| 287 | return mSampleRate; |
| 288 | } |
| 289 | |
| 290 | int AudioTrack::format() const |
| 291 | { |
| 292 | return mFormat; |
| 293 | } |
| 294 | |
| 295 | int AudioTrack::channelCount() const |
| 296 | { |
| 297 | return mChannelCount; |
| 298 | } |
| 299 | |
| 300 | uint32_t AudioTrack::frameCount() const |
| 301 | { |
| 302 | return mFrameCount; |
| 303 | } |
| 304 | |
| 305 | int AudioTrack::frameSize() const |
| 306 | { |
| 307 | return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t)); |
| 308 | } |
| 309 | |
| 310 | sp<IMemory>& AudioTrack::sharedBuffer() |
| 311 | { |
| 312 | return mSharedBuffer; |
| 313 | } |
| 314 | |
| 315 | // ------------------------------------------------------------------------- |
| 316 | |
| 317 | void AudioTrack::start() |
| 318 | { |
| 319 | sp<AudioTrackThread> t = mAudioTrackThread; |
| 320 | |
| 321 | LOGV("start %p", this); |
| 322 | if (t != 0) { |
| 323 | if (t->exitPending()) { |
| 324 | if (t->requestExitAndWait() == WOULD_BLOCK) { |
| 325 | LOGE("AudioTrack::start called from thread"); |
| 326 | return; |
| 327 | } |
| 328 | } |
| 329 | t->mLock.lock(); |
| 330 | } |
| 331 | |
| 332 | if (android_atomic_or(1, &mActive) == 0) { |
| 333 | mNewPosition = mCblk->server + mUpdatePeriod; |
| 334 | mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; |
| 335 | mCblk->waitTimeMs = 0; |
| 336 | if (t != 0) { |
| 337 | t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT); |
| 338 | } else { |
| 339 | setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT); |
| 340 | } |
| 341 | mAudioTrack->start(); |
| 342 | } |
| 343 | |
| 344 | if (t != 0) { |
| 345 | t->mLock.unlock(); |
| 346 | } |
| 347 | } |
| 348 | |
| 349 | void AudioTrack::stop() |
| 350 | { |
| 351 | sp<AudioTrackThread> t = mAudioTrackThread; |
| 352 | |
| 353 | LOGV("stop %p", this); |
| 354 | if (t != 0) { |
| 355 | t->mLock.lock(); |
| 356 | } |
| 357 | |
| 358 | if (android_atomic_and(~1, &mActive) == 1) { |
| 359 | mAudioTrack->stop(); |
| 360 | // Cancel loops (If we are in the middle of a loop, playback |
| 361 | // would not stop until loopCount reaches 0). |
| 362 | setLoop(0, 0, 0); |
| 363 | // Force flush if a shared buffer is used otherwise audioflinger |
| 364 | // will not stop before end of buffer is reached. |
| 365 | if (mSharedBuffer != 0) { |
| 366 | flush(); |
| 367 | } |
| 368 | if (t != 0) { |
| 369 | t->requestExit(); |
| 370 | } else { |
| 371 | setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); |
| 372 | } |
| 373 | } |
| 374 | |
| 375 | if (t != 0) { |
| 376 | t->mLock.unlock(); |
| 377 | } |
| 378 | } |
| 379 | |
| 380 | bool AudioTrack::stopped() const |
| 381 | { |
| 382 | return !mActive; |
| 383 | } |
| 384 | |
| 385 | void AudioTrack::flush() |
| 386 | { |
| 387 | LOGV("flush"); |
| 388 | |
| 389 | if (!mActive) { |
| 390 | mCblk->lock.lock(); |
| 391 | mAudioTrack->flush(); |
| 392 | // Release AudioTrack callback thread in case it was waiting for new buffers |
| 393 | // in AudioTrack::obtainBuffer() |
| 394 | mCblk->cv.signal(); |
| 395 | mCblk->lock.unlock(); |
| 396 | } |
| 397 | } |
| 398 | |
| 399 | void AudioTrack::pause() |
| 400 | { |
| 401 | LOGV("pause"); |
| 402 | if (android_atomic_and(~1, &mActive) == 1) { |
| 403 | mActive = 0; |
| 404 | mAudioTrack->pause(); |
| 405 | } |
| 406 | } |
| 407 | |
| 408 | void AudioTrack::mute(bool e) |
| 409 | { |
| 410 | mAudioTrack->mute(e); |
| 411 | mMuted = e; |
| 412 | } |
| 413 | |
| 414 | bool AudioTrack::muted() const |
| 415 | { |
| 416 | return mMuted; |
| 417 | } |
| 418 | |
| 419 | void AudioTrack::setVolume(float left, float right) |
| 420 | { |
| 421 | mVolume[LEFT] = left; |
| 422 | mVolume[RIGHT] = right; |
| 423 | |
| 424 | // write must be atomic |
| 425 | mCblk->volumeLR = (int32_t(int16_t(left * 0x1000)) << 16) | int16_t(right * 0x1000); |
| 426 | } |
| 427 | |
| 428 | void AudioTrack::getVolume(float* left, float* right) |
| 429 | { |
| 430 | *left = mVolume[LEFT]; |
| 431 | *right = mVolume[RIGHT]; |
| 432 | } |
| 433 | |
| 434 | void AudioTrack::setSampleRate(int rate) |
| 435 | { |
| 436 | int afSamplingRate; |
| 437 | |
| 438 | if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { |
| 439 | return; |
| 440 | } |
| 441 | // Resampler implementation limits input sampling rate to 2 x output sampling rate. |
| 442 | if (rate <= 0) rate = 1; |
| 443 | if (rate > afSamplingRate*2) rate = afSamplingRate*2; |
| 444 | if (rate > MAX_SAMPLE_RATE) rate = MAX_SAMPLE_RATE; |
| 445 | |
| 446 | mCblk->sampleRate = rate; |
| 447 | } |
| 448 | |
| 449 | uint32_t AudioTrack::getSampleRate() |
| 450 | { |
| 451 | return uint32_t(mCblk->sampleRate); |
| 452 | } |
| 453 | |
| 454 | status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) |
| 455 | { |
| 456 | audio_track_cblk_t* cblk = mCblk; |
| 457 | |
| 458 | |
| 459 | Mutex::Autolock _l(cblk->lock); |
| 460 | |
| 461 | if (loopCount == 0) { |
| 462 | cblk->loopStart = UINT_MAX; |
| 463 | cblk->loopEnd = UINT_MAX; |
| 464 | cblk->loopCount = 0; |
| 465 | mLoopCount = 0; |
| 466 | return NO_ERROR; |
| 467 | } |
| 468 | |
| 469 | if (loopStart >= loopEnd || |
| 470 | loopEnd - loopStart > mFrameCount) { |
| 471 | LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user); |
| 472 | return BAD_VALUE; |
| 473 | } |
| 474 | |
| 475 | if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) { |
| 476 | LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", |
| 477 | loopStart, loopEnd, mFrameCount); |
| 478 | return BAD_VALUE; |
| 479 | } |
| 480 | |
| 481 | cblk->loopStart = loopStart; |
| 482 | cblk->loopEnd = loopEnd; |
| 483 | cblk->loopCount = loopCount; |
| 484 | mLoopCount = loopCount; |
| 485 | |
| 486 | return NO_ERROR; |
| 487 | } |
| 488 | |
| 489 | status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) |
| 490 | { |
| 491 | if (loopStart != 0) { |
| 492 | *loopStart = mCblk->loopStart; |
| 493 | } |
| 494 | if (loopEnd != 0) { |
| 495 | *loopEnd = mCblk->loopEnd; |
| 496 | } |
| 497 | if (loopCount != 0) { |
| 498 | if (mCblk->loopCount < 0) { |
| 499 | *loopCount = -1; |
| 500 | } else { |
| 501 | *loopCount = mCblk->loopCount; |
| 502 | } |
| 503 | } |
| 504 | |
| 505 | return NO_ERROR; |
| 506 | } |
| 507 | |
| 508 | status_t AudioTrack::setMarkerPosition(uint32_t marker) |
| 509 | { |
| 510 | if (mCbf == 0) return INVALID_OPERATION; |
| 511 | |
| 512 | mMarkerPosition = marker; |
| 513 | |
| 514 | return NO_ERROR; |
| 515 | } |
| 516 | |
| 517 | status_t AudioTrack::getMarkerPosition(uint32_t *marker) |
| 518 | { |
| 519 | if (marker == 0) return BAD_VALUE; |
| 520 | |
| 521 | *marker = mMarkerPosition; |
| 522 | |
| 523 | return NO_ERROR; |
| 524 | } |
| 525 | |
| 526 | status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) |
| 527 | { |
| 528 | if (mCbf == 0) return INVALID_OPERATION; |
| 529 | |
| 530 | uint32_t curPosition; |
| 531 | getPosition(&curPosition); |
| 532 | mNewPosition = curPosition + updatePeriod; |
| 533 | mUpdatePeriod = updatePeriod; |
| 534 | |
| 535 | return NO_ERROR; |
| 536 | } |
| 537 | |
| 538 | status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) |
| 539 | { |
| 540 | if (updatePeriod == 0) return BAD_VALUE; |
| 541 | |
| 542 | *updatePeriod = mUpdatePeriod; |
| 543 | |
| 544 | return NO_ERROR; |
| 545 | } |
| 546 | |
| 547 | status_t AudioTrack::setPosition(uint32_t position) |
| 548 | { |
| 549 | Mutex::Autolock _l(mCblk->lock); |
| 550 | |
| 551 | if (!stopped()) return INVALID_OPERATION; |
| 552 | |
| 553 | if (position > mCblk->user) return BAD_VALUE; |
| 554 | |
| 555 | mCblk->server = position; |
| 556 | mCblk->forceReady = 1; |
| 557 | |
| 558 | return NO_ERROR; |
| 559 | } |
| 560 | |
| 561 | status_t AudioTrack::getPosition(uint32_t *position) |
| 562 | { |
| 563 | if (position == 0) return BAD_VALUE; |
| 564 | |
| 565 | *position = mCblk->server; |
| 566 | |
| 567 | return NO_ERROR; |
| 568 | } |
| 569 | |
| 570 | status_t AudioTrack::reload() |
| 571 | { |
| 572 | if (!stopped()) return INVALID_OPERATION; |
| 573 | |
| 574 | flush(); |
| 575 | |
| 576 | mCblk->stepUser(mFrameCount); |
| 577 | |
| 578 | return NO_ERROR; |
| 579 | } |
| 580 | |
| 581 | // ------------------------------------------------------------------------- |
| 582 | |
| 583 | status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) |
| 584 | { |
| 585 | int active; |
| 586 | int timeout = 0; |
| 587 | status_t result; |
| 588 | audio_track_cblk_t* cblk = mCblk; |
| 589 | uint32_t framesReq = audioBuffer->frameCount; |
| 590 | |
| 591 | audioBuffer->frameCount = 0; |
| 592 | audioBuffer->size = 0; |
| 593 | |
| 594 | uint32_t framesAvail = cblk->framesAvailable(); |
| 595 | |
| 596 | if (framesAvail == 0) { |
| 597 | Mutex::Autolock _l(cblk->lock); |
| 598 | goto start_loop_here; |
| 599 | while (framesAvail == 0) { |
| 600 | active = mActive; |
| 601 | if (UNLIKELY(!active)) { |
| 602 | LOGV("Not active and NO_MORE_BUFFERS"); |
| 603 | return NO_MORE_BUFFERS; |
| 604 | } |
| 605 | if (UNLIKELY(!waitCount)) |
| 606 | return WOULD_BLOCK; |
| 607 | timeout = 0; |
| 608 | result = cblk->cv.waitRelative(cblk->lock, milliseconds(WAIT_PERIOD_MS)); |
| 609 | if (__builtin_expect(result!=NO_ERROR, false)) { |
| 610 | cblk->waitTimeMs += WAIT_PERIOD_MS; |
| 611 | if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { |
| 612 | // timing out when a loop has been set and we have already written upto loop end |
| 613 | // is a normal condition: no need to wake AudioFlinger up. |
| 614 | if (cblk->user < cblk->loopEnd) { |
| 615 | LOGW( "obtainBuffer timed out (is the CPU pegged?) %p " |
| 616 | "user=%08x, server=%08x", this, cblk->user, cblk->server); |
| 617 | //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) |
| 618 | cblk->lock.unlock(); |
| 619 | mAudioTrack->start(); |
| 620 | cblk->lock.lock(); |
| 621 | timeout = 1; |
| 622 | } |
| 623 | cblk->waitTimeMs = 0; |
| 624 | } |
| 625 | |
| 626 | if (--waitCount == 0) { |
| 627 | return TIMED_OUT; |
| 628 | } |
| 629 | } |
| 630 | // read the server count again |
| 631 | start_loop_here: |
| 632 | framesAvail = cblk->framesAvailable_l(); |
| 633 | } |
| 634 | } |
| 635 | |
| 636 | cblk->waitTimeMs = 0; |
| 637 | |
| 638 | if (framesReq > framesAvail) { |
| 639 | framesReq = framesAvail; |
| 640 | } |
| 641 | |
| 642 | uint32_t u = cblk->user; |
| 643 | uint32_t bufferEnd = cblk->userBase + cblk->frameCount; |
| 644 | |
| 645 | if (u + framesReq > bufferEnd) { |
| 646 | framesReq = bufferEnd - u; |
| 647 | } |
| 648 | |
| 649 | LOGW_IF(timeout, |
| 650 | "*** SERIOUS WARNING *** obtainBuffer() timed out " |
| 651 | "but didn't need to be locked. We recovered, but " |
| 652 | "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server); |
| 653 | |
| 654 | audioBuffer->flags = mMuted ? Buffer::MUTE : 0; |
| 655 | audioBuffer->channelCount= mChannelCount; |
| 656 | audioBuffer->format = AudioSystem::PCM_16_BIT; |
| 657 | audioBuffer->frameCount = framesReq; |
| 658 | audioBuffer->size = framesReq*mChannelCount*sizeof(int16_t); |
| 659 | audioBuffer->raw = (int8_t *)cblk->buffer(u); |
| 660 | active = mActive; |
| 661 | return active ? status_t(NO_ERROR) : status_t(STOPPED); |
| 662 | } |
| 663 | |
| 664 | void AudioTrack::releaseBuffer(Buffer* audioBuffer) |
| 665 | { |
| 666 | audio_track_cblk_t* cblk = mCblk; |
| 667 | cblk->stepUser(audioBuffer->frameCount); |
| 668 | } |
| 669 | |
| 670 | // ------------------------------------------------------------------------- |
| 671 | |
| 672 | ssize_t AudioTrack::write(const void* buffer, size_t userSize) |
| 673 | { |
| 674 | |
| 675 | if (mSharedBuffer != 0) return INVALID_OPERATION; |
| 676 | |
| 677 | if (ssize_t(userSize) < 0) { |
| 678 | // sanity-check. user is most-likely passing an error code. |
| 679 | LOGE("AudioTrack::write(buffer=%p, size=%u (%d)", |
| 680 | buffer, userSize, userSize); |
| 681 | return BAD_VALUE; |
| 682 | } |
| 683 | |
| 684 | LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); |
| 685 | |
| 686 | ssize_t written = 0; |
| 687 | const int8_t *src = (const int8_t *)buffer; |
| 688 | Buffer audioBuffer; |
| 689 | |
| 690 | do { |
| 691 | audioBuffer.frameCount = userSize/mChannelCount; |
| 692 | if (mFormat == AudioSystem::PCM_16_BIT) { |
| 693 | audioBuffer.frameCount >>= 1; |
| 694 | } |
| 695 | // Calling obtainBuffer() with a negative wait count causes |
| 696 | // an (almost) infinite wait time. |
| 697 | status_t err = obtainBuffer(&audioBuffer, -1); |
| 698 | if (err < 0) { |
| 699 | // out of buffers, return #bytes written |
| 700 | if (err == status_t(NO_MORE_BUFFERS)) |
| 701 | break; |
| 702 | return ssize_t(err); |
| 703 | } |
| 704 | |
| 705 | size_t toWrite; |
| 706 | if (mFormat == AudioSystem::PCM_8_BIT) { |
| 707 | // Divide capacity by 2 to take expansion into account |
| 708 | toWrite = audioBuffer.size>>1; |
| 709 | // 8 to 16 bit conversion |
| 710 | int count = toWrite; |
| 711 | int16_t *dst = (int16_t *)(audioBuffer.i8); |
| 712 | while(count--) { |
| 713 | *dst++ = (int16_t)(*src++^0x80) << 8; |
| 714 | } |
| 715 | }else { |
| 716 | toWrite = audioBuffer.size; |
| 717 | memcpy(audioBuffer.i8, src, toWrite); |
| 718 | src += toWrite; |
| 719 | } |
| 720 | userSize -= toWrite; |
| 721 | written += toWrite; |
| 722 | |
| 723 | releaseBuffer(&audioBuffer); |
| 724 | } while (userSize); |
| 725 | |
| 726 | return written; |
| 727 | } |
| 728 | |
| 729 | // ------------------------------------------------------------------------- |
| 730 | |
| 731 | bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) |
| 732 | { |
| 733 | Buffer audioBuffer; |
| 734 | uint32_t frames; |
| 735 | size_t writtenSize; |
| 736 | |
| 737 | // Manage underrun callback |
| 738 | if (mActive && (mCblk->framesReady() == 0)) { |
| 739 | LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag); |
| 740 | if (mCblk->flowControlFlag == 0) { |
| 741 | mCbf(EVENT_UNDERRUN, mUserData, 0); |
| 742 | if (mCblk->server == mCblk->frameCount) { |
| 743 | mCbf(EVENT_BUFFER_END, mUserData, 0); |
| 744 | } |
| 745 | mCblk->flowControlFlag = 1; |
| 746 | if (mSharedBuffer != 0) return false; |
| 747 | } |
| 748 | } |
| 749 | |
| 750 | // Manage loop end callback |
| 751 | while (mLoopCount > mCblk->loopCount) { |
| 752 | int loopCount = -1; |
| 753 | mLoopCount--; |
| 754 | if (mLoopCount >= 0) loopCount = mLoopCount; |
| 755 | |
| 756 | mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); |
| 757 | } |
| 758 | |
| 759 | // Manage marker callback |
| 760 | if(mMarkerPosition > 0) { |
| 761 | if (mCblk->server >= mMarkerPosition) { |
| 762 | mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); |
| 763 | mMarkerPosition = 0; |
| 764 | } |
| 765 | } |
| 766 | |
| 767 | // Manage new position callback |
| 768 | if(mUpdatePeriod > 0) { |
| 769 | while (mCblk->server >= mNewPosition) { |
| 770 | mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); |
| 771 | mNewPosition += mUpdatePeriod; |
| 772 | } |
| 773 | } |
| 774 | |
| 775 | // If Shared buffer is used, no data is requested from client. |
| 776 | if (mSharedBuffer != 0) { |
| 777 | frames = 0; |
| 778 | } else { |
| 779 | frames = mRemainingFrames; |
| 780 | } |
| 781 | |
| 782 | do { |
| 783 | |
| 784 | audioBuffer.frameCount = frames; |
| 785 | |
| 786 | // Calling obtainBuffer() with a wait count of 1 |
| 787 | // limits wait time to WAIT_PERIOD_MS. This prevents from being |
| 788 | // stuck here not being able to handle timed events (position, markers, loops). |
| 789 | status_t err = obtainBuffer(&audioBuffer, 1); |
| 790 | if (err < NO_ERROR) { |
| 791 | if (err != TIMED_OUT) { |
| 792 | LOGE("Error obtaining an audio buffer, giving up."); |
| 793 | return false; |
| 794 | } |
| 795 | break; |
| 796 | } |
| 797 | if (err == status_t(STOPPED)) return false; |
| 798 | |
| 799 | // Divide buffer size by 2 to take into account the expansion |
| 800 | // due to 8 to 16 bit conversion: the callback must fill only half |
| 801 | // of the destination buffer |
| 802 | if (mFormat == AudioSystem::PCM_8_BIT) { |
| 803 | audioBuffer.size >>= 1; |
| 804 | } |
| 805 | |
| 806 | size_t reqSize = audioBuffer.size; |
| 807 | mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); |
| 808 | writtenSize = audioBuffer.size; |
| 809 | |
| 810 | // Sanity check on returned size |
| 811 | if (ssize_t(writtenSize) <= 0) break; |
| 812 | if (writtenSize > reqSize) writtenSize = reqSize; |
| 813 | |
| 814 | if (mFormat == AudioSystem::PCM_8_BIT) { |
| 815 | // 8 to 16 bit conversion |
| 816 | const int8_t *src = audioBuffer.i8 + writtenSize-1; |
| 817 | int count = writtenSize; |
| 818 | int16_t *dst = audioBuffer.i16 + writtenSize-1; |
| 819 | while(count--) { |
| 820 | *dst-- = (int16_t)(*src--^0x80) << 8; |
| 821 | } |
| 822 | writtenSize <<= 1; |
| 823 | } |
| 824 | |
| 825 | audioBuffer.size = writtenSize; |
| 826 | audioBuffer.frameCount = writtenSize/mChannelCount/sizeof(int16_t); |
| 827 | frames -= audioBuffer.frameCount; |
| 828 | |
| 829 | releaseBuffer(&audioBuffer); |
| 830 | } |
| 831 | while (frames); |
| 832 | |
| 833 | if (frames == 0) { |
| 834 | mRemainingFrames = mNotificationFrames; |
| 835 | } else { |
| 836 | mRemainingFrames = frames; |
| 837 | } |
| 838 | return true; |
| 839 | } |
| 840 | |
| 841 | status_t AudioTrack::dump(int fd, const Vector<String16>& args) const |
| 842 | { |
| 843 | |
| 844 | const size_t SIZE = 256; |
| 845 | char buffer[SIZE]; |
| 846 | String8 result; |
| 847 | |
| 848 | result.append(" AudioTrack::dump\n"); |
| 849 | snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); |
| 850 | result.append(buffer); |
| 851 | snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount); |
| 852 | result.append(buffer); |
| 853 | snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", mSampleRate, mStatus, mMuted); |
| 854 | result.append(buffer); |
| 855 | snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); |
| 856 | result.append(buffer); |
| 857 | ::write(fd, result.string(), result.size()); |
| 858 | return NO_ERROR; |
| 859 | } |
| 860 | |
| 861 | // ========================================================================= |
| 862 | |
| 863 | AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) |
| 864 | : Thread(bCanCallJava), mReceiver(receiver) |
| 865 | { |
| 866 | } |
| 867 | |
| 868 | bool AudioTrack::AudioTrackThread::threadLoop() |
| 869 | { |
| 870 | return mReceiver.processAudioBuffer(this); |
| 871 | } |
| 872 | |
| 873 | status_t AudioTrack::AudioTrackThread::readyToRun() |
| 874 | { |
| 875 | return NO_ERROR; |
| 876 | } |
| 877 | |
| 878 | void AudioTrack::AudioTrackThread::onFirstRef() |
| 879 | { |
| 880 | } |
| 881 | |
| 882 | // ========================================================================= |
| 883 | |
| 884 | audio_track_cblk_t::audio_track_cblk_t() |
| 885 | : user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0), |
| 886 | loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), flowControlFlag(1), forceReady(0) |
| 887 | { |
| 888 | } |
| 889 | |
| 890 | uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) |
| 891 | { |
| 892 | uint32_t u = this->user; |
| 893 | |
| 894 | u += frameCount; |
| 895 | // Ensure that user is never ahead of server for AudioRecord |
| 896 | if (out) { |
| 897 | // If stepServer() has been called once, switch to normal obtainBuffer() timeout period |
| 898 | if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { |
| 899 | bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; |
| 900 | } |
| 901 | } else if (u > this->server) { |
| 902 | LOGW("stepServer occured after track reset"); |
| 903 | u = this->server; |
| 904 | } |
| 905 | |
| 906 | if (u >= userBase + this->frameCount) { |
| 907 | userBase += this->frameCount; |
| 908 | } |
| 909 | |
| 910 | this->user = u; |
| 911 | |
| 912 | // Clear flow control error condition as new data has been written/read to/from buffer. |
| 913 | flowControlFlag = 0; |
| 914 | |
| 915 | return u; |
| 916 | } |
| 917 | |
| 918 | bool audio_track_cblk_t::stepServer(uint32_t frameCount) |
| 919 | { |
| 920 | // the code below simulates lock-with-timeout |
| 921 | // we MUST do this to protect the AudioFlinger server |
| 922 | // as this lock is shared with the client. |
| 923 | status_t err; |
| 924 | |
| 925 | err = lock.tryLock(); |
| 926 | if (err == -EBUSY) { // just wait a bit |
| 927 | usleep(1000); |
| 928 | err = lock.tryLock(); |
| 929 | } |
| 930 | if (err != NO_ERROR) { |
| 931 | // probably, the client just died. |
| 932 | return false; |
| 933 | } |
| 934 | |
| 935 | uint32_t s = this->server; |
| 936 | |
| 937 | s += frameCount; |
| 938 | if (out) { |
| 939 | // Mark that we have read the first buffer so that next time stepUser() is called |
| 940 | // we switch to normal obtainBuffer() timeout period |
| 941 | if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { |
| 942 | bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1; |
| 943 | } |
| 944 | // It is possible that we receive a flush() |
| 945 | // while the mixer is processing a block: in this case, |
| 946 | // stepServer() is called After the flush() has reset u & s and |
| 947 | // we have s > u |
| 948 | if (s > this->user) { |
| 949 | LOGW("stepServer occured after track reset"); |
| 950 | s = this->user; |
| 951 | } |
| 952 | } |
| 953 | |
| 954 | if (s >= loopEnd) { |
| 955 | LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); |
| 956 | s = loopStart; |
| 957 | if (--loopCount == 0) { |
| 958 | loopEnd = UINT_MAX; |
| 959 | loopStart = UINT_MAX; |
| 960 | } |
| 961 | } |
| 962 | if (s >= serverBase + this->frameCount) { |
| 963 | serverBase += this->frameCount; |
| 964 | } |
| 965 | |
| 966 | this->server = s; |
| 967 | |
| 968 | cv.signal(); |
| 969 | lock.unlock(); |
| 970 | return true; |
| 971 | } |
| 972 | |
| 973 | void* audio_track_cblk_t::buffer(uint32_t offset) const |
| 974 | { |
| 975 | return (int16_t *)this->buffers + (offset-userBase)*this->channels; |
| 976 | } |
| 977 | |
| 978 | uint32_t audio_track_cblk_t::framesAvailable() |
| 979 | { |
| 980 | Mutex::Autolock _l(lock); |
| 981 | return framesAvailable_l(); |
| 982 | } |
| 983 | |
| 984 | uint32_t audio_track_cblk_t::framesAvailable_l() |
| 985 | { |
| 986 | uint32_t u = this->user; |
| 987 | uint32_t s = this->server; |
| 988 | |
| 989 | if (out) { |
| 990 | uint32_t limit = (s < loopStart) ? s : loopStart; |
| 991 | return limit + frameCount - u; |
| 992 | } else { |
| 993 | return frameCount + u - s; |
| 994 | } |
| 995 | } |
| 996 | |
| 997 | uint32_t audio_track_cblk_t::framesReady() |
| 998 | { |
| 999 | uint32_t u = this->user; |
| 1000 | uint32_t s = this->server; |
| 1001 | |
| 1002 | if (out) { |
| 1003 | if (u < loopEnd) { |
| 1004 | return u - s; |
| 1005 | } else { |
| 1006 | Mutex::Autolock _l(lock); |
| 1007 | if (loopCount >= 0) { |
| 1008 | return (loopEnd - loopStart)*loopCount + u - s; |
| 1009 | } else { |
| 1010 | return UINT_MAX; |
| 1011 | } |
| 1012 | } |
| 1013 | } else { |
| 1014 | return s - u; |
| 1015 | } |
| 1016 | } |
| 1017 | |
| 1018 | // ------------------------------------------------------------------------- |
| 1019 | |
| 1020 | }; // namespace android |
| 1021 | |