AudioTrack.h revision 01437b7cdaecf53acb46b50ff8b5d86b9d36eb20
1/* 2 * Copyright (C) 2007 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17#ifndef ANDROID_AUDIOTRACK_H 18#define ANDROID_AUDIOTRACK_H 19 20#include <stdint.h> 21#include <sys/types.h> 22 23#include <media/IAudioFlinger.h> 24#include <media/IAudioTrack.h> 25#include <media/AudioSystem.h> 26 27#include <utils/RefBase.h> 28#include <utils/Errors.h> 29#include <binder/IInterface.h> 30#include <binder/IMemory.h> 31#include <cutils/sched_policy.h> 32#include <utils/threads.h> 33 34namespace android { 35 36// ---------------------------------------------------------------------------- 37 38class audio_track_cblk_t; 39 40// ---------------------------------------------------------------------------- 41 42class AudioTrack : virtual public RefBase 43{ 44public: 45 enum channel_index { 46 MONO = 0, 47 LEFT = 0, 48 RIGHT = 1 49 }; 50 51 /* Events used by AudioTrack callback function (audio_track_cblk_t). 52 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 53 */ 54 enum event_type { 55 EVENT_MORE_DATA = 0, // Request to write more data to PCM buffer. 56 EVENT_UNDERRUN = 1, // PCM buffer underrun occurred. 57 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 58 // loop start if loop count was not 0. 59 EVENT_MARKER = 3, // Playback head is at the specified marker position 60 // (See setMarkerPosition()). 61 EVENT_NEW_POS = 4, // Playback head is at a new position 62 // (See setPositionUpdatePeriod()). 63 EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer. 64 }; 65 66 /* Client should declare Buffer on the stack and pass address to obtainBuffer() 67 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 68 */ 69 70 class Buffer 71 { 72 public: 73 size_t frameCount; // number of sample frames corresponding to size; 74 // on input it is the number of frames desired, 75 // on output is the number of frames actually filled 76 77 size_t size; // input/output in byte units 78 union { 79 void* raw; 80 short* i16; // signed 16-bit 81 int8_t* i8; // unsigned 8-bit, offset by 0x80 82 }; 83 }; 84 85 86 /* As a convenience, if a callback is supplied, a handler thread 87 * is automatically created with the appropriate priority. This thread 88 * invokes the callback when a new buffer becomes available or various conditions occur. 89 * Parameters: 90 * 91 * event: type of event notified (see enum AudioTrack::event_type). 92 * user: Pointer to context for use by the callback receiver. 93 * info: Pointer to optional parameter according to event type: 94 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 95 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 96 * written. 97 * - EVENT_UNDERRUN: unused. 98 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 99 * - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames. 100 * - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames. 101 * - EVENT_BUFFER_END: unused. 102 */ 103 104 typedef void (*callback_t)(int event, void* user, void *info); 105 106 /* Returns the minimum frame count required for the successful creation of 107 * an AudioTrack object. 108 * Returned status (from utils/Errors.h) can be: 109 * - NO_ERROR: successful operation 110 * - NO_INIT: audio server or audio hardware not initialized 111 */ 112 113 static status_t getMinFrameCount(size_t* frameCount, 114 audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, 115 uint32_t sampleRate = 0); 116 117 /* Constructs an uninitialized AudioTrack. No connection with 118 * AudioFlinger takes place. 119 */ 120 AudioTrack(); 121 122 /* Creates an AudioTrack object and registers it with AudioFlinger. 123 * Once created, the track needs to be started before it can be used. 124 * Unspecified values are set to the audio hardware's current 125 * values. 126 * 127 * Parameters: 128 * 129 * streamType: Select the type of audio stream this track is attached to 130 * (e.g. AUDIO_STREAM_MUSIC). 131 * sampleRate: Track sampling rate in Hz. 132 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 133 * 16 bits per sample). 134 * channelMask: Channel mask. 135 * frameCount: Minimum size of track PCM buffer in frames. This defines the 136 * application's contribution to the 137 * latency of the track. The actual size selected by the AudioTrack could be 138 * larger if the requested size is not compatible with current audio HAL 139 * latency. Zero means to use a default value. 140 * flags: See comments on audio_output_flags_t in <system/audio.h>. 141 * cbf: Callback function. If not null, this function is called periodically 142 * to provide new PCM data. 143 * user: Context for use by the callback receiver. 144 * notificationFrames: The callback function is called each time notificationFrames PCM 145 * frames have been consumed from track input buffer. 146 * sessionId: Specific session ID, or zero to use default. 147 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 148 * If not present in parameter list, then fixed at false. 149 */ 150 151 AudioTrack( audio_stream_type_t streamType, 152 uint32_t sampleRate = 0, 153 audio_format_t format = AUDIO_FORMAT_DEFAULT, 154 audio_channel_mask_t channelMask = 0, 155 int frameCount = 0, 156 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 157 callback_t cbf = NULL, 158 void* user = NULL, 159 int notificationFrames = 0, 160 int sessionId = 0); 161 162 /* Creates an audio track and registers it with AudioFlinger. With this constructor, 163 * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer 164 * identified by the argument sharedBuffer. This prototype is for static buffer playback. 165 * PCM data must be present in memory before the AudioTrack is started. 166 * The write() and flush() methods are not supported in this case. 167 * It is recommended to pass a callback function to be notified of playback end by an 168 * EVENT_UNDERRUN event. 169 */ 170 171 AudioTrack( audio_stream_type_t streamType, 172 uint32_t sampleRate = 0, 173 audio_format_t format = AUDIO_FORMAT_DEFAULT, 174 audio_channel_mask_t channelMask = 0, 175 const sp<IMemory>& sharedBuffer = 0, 176 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 177 callback_t cbf = NULL, 178 void* user = NULL, 179 int notificationFrames = 0, 180 int sessionId = 0); 181 182 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 183 * Also destroys all resources associated with the AudioTrack. 184 */ 185 ~AudioTrack(); 186 187 188 /* Initialize an uninitialized AudioTrack. 189 * Returned status (from utils/Errors.h) can be: 190 * - NO_ERROR: successful initialization 191 * - INVALID_OPERATION: AudioTrack is already initialized 192 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 193 * - NO_INIT: audio server or audio hardware not initialized 194 */ 195 status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT, 196 uint32_t sampleRate = 0, 197 audio_format_t format = AUDIO_FORMAT_DEFAULT, 198 audio_channel_mask_t channelMask = 0, 199 int frameCount = 0, 200 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 201 callback_t cbf = NULL, 202 void* user = NULL, 203 int notificationFrames = 0, 204 const sp<IMemory>& sharedBuffer = 0, 205 bool threadCanCallJava = false, 206 int sessionId = 0); 207 208 209 /* Result of constructing the AudioTrack. This must be checked 210 * before using any AudioTrack API (except for set()), because using 211 * an uninitialized AudioTrack produces undefined results. 212 * See set() method above for possible return codes. 213 */ 214 status_t initCheck() const { return mStatus; } 215 216 /* Returns this track's estimated latency in milliseconds. 217 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 218 * and audio hardware driver. 219 */ 220 uint32_t latency() const { return mLatency; } 221 222 /* getters, see constructors and set() */ 223 224 audio_stream_type_t streamType() const { return mStreamType; } 225 audio_format_t format() const { return mFormat; } 226 227 /* Return channelCount * (bit depth per channel / 8). 228 * channelCount is determined from channelMask, and bit depth comes from format. 229 */ 230 uint32_t channelCount() const { return mChannelCount; } 231 232 uint32_t frameCount() const { return mFrameCount; } 233 size_t frameSize() const { return mFrameSize; } 234 235 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 236 237 238 /* After it's created the track is not active. Call start() to 239 * make it active. If set, the callback will start being called. 240 */ 241 void start(); 242 243 /* Stop a track. If set, the callback will cease being called and 244 * obtainBuffer returns STOPPED. Note that obtainBuffer() still works 245 * and will fill up buffers until the pool is exhausted. 246 */ 247 void stop(); 248 bool stopped() const; 249 250 /* Flush a stopped track. All pending buffers are discarded. 251 * This function has no effect if the track is not stopped. 252 */ 253 void flush(); 254 255 /* Pause a track. If set, the callback will cease being called and 256 * obtainBuffer returns STOPPED. Note that obtainBuffer() still works 257 * and will fill up buffers until the pool is exhausted. 258 */ 259 void pause(); 260 261 /* Mute or unmute this track. 262 * While muted, the callback, if set, is still called. 263 */ 264 void mute(bool); 265 bool muted() const { return mMuted; } 266 267 /* Set volume for this track, mostly used for games' sound effects 268 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 269 * This is the older API. New applications should use setVolume(float) when possible. 270 */ 271 status_t setVolume(float left, float right); 272 273 /* Set volume for all channels. This is the preferred API for new applications, 274 * especially for multi-channel content. 275 */ 276 status_t setVolume(float volume); 277 278 /* Set the send level for this track. An auxiliary effect should be attached 279 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 280 */ 281 status_t setAuxEffectSendLevel(float level); 282 void getAuxEffectSendLevel(float* level) const; 283 284 /* Set sample rate for this track in Hz, mostly used for games' sound effects 285 */ 286 status_t setSampleRate(uint32_t sampleRate); 287 288 /* Return current sample rate in Hz, or 0 if unknown */ 289 uint32_t getSampleRate() const; 290 291 /* Enables looping and sets the start and end points of looping. 292 * 293 * Parameters: 294 * 295 * loopStart: loop start expressed as the number of PCM frames played since AudioTrack start. 296 * loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start. 297 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 298 * pending or active loop. loopCount = -1 means infinite looping. 299 * 300 * For proper operation the following condition must be respected: 301 * (loopEnd-loopStart) <= framecount() 302 */ 303 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 304 305 /* Sets marker position. When playback reaches the number of frames specified, a callback with 306 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 307 * notification callback. 308 * If the AudioTrack has been opened with no callback function associated, the operation will 309 * fail. 310 * 311 * Parameters: 312 * 313 * marker: marker position expressed in frames. 314 * 315 * Returned status (from utils/Errors.h) can be: 316 * - NO_ERROR: successful operation 317 * - INVALID_OPERATION: the AudioTrack has no callback installed. 318 */ 319 status_t setMarkerPosition(uint32_t marker); 320 status_t getMarkerPosition(uint32_t *marker) const; 321 322 323 /* Sets position update period. Every time the number of frames specified has been played, 324 * a callback with event type EVENT_NEW_POS is called. 325 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 326 * callback. 327 * If the AudioTrack has been opened with no callback function associated, the operation will 328 * fail. 329 * 330 * Parameters: 331 * 332 * updatePeriod: position update notification period expressed in frames. 333 * 334 * Returned status (from utils/Errors.h) can be: 335 * - NO_ERROR: successful operation 336 * - INVALID_OPERATION: the AudioTrack has no callback installed. 337 */ 338 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 339 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 340 341 /* Sets playback head position within AudioTrack buffer. The new position is specified 342 * in number of frames. 343 * This method must be called with the AudioTrack in paused or stopped state. 344 * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames. 345 * Therefore using this method makes sense only when playing a "static" audio buffer 346 * as opposed to streaming. 347 * The getPosition() method on the other hand returns the total number of frames played since 348 * playback start. 349 * 350 * Parameters: 351 * 352 * position: New playback head position within AudioTrack buffer. 353 * 354 * Returned status (from utils/Errors.h) can be: 355 * - NO_ERROR: successful operation 356 * - INVALID_OPERATION: the AudioTrack is not stopped. 357 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 358 * buffer 359 */ 360 status_t setPosition(uint32_t position); 361 status_t getPosition(uint32_t *position); 362 363 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 364 * rewriting the buffer before restarting playback after a stop. 365 * This method must be called with the AudioTrack in paused or stopped state. 366 * 367 * Returned status (from utils/Errors.h) can be: 368 * - NO_ERROR: successful operation 369 * - INVALID_OPERATION: the AudioTrack is not stopped. 370 */ 371 status_t reload(); 372 373 /* Returns a handle on the audio output used by this AudioTrack. 374 * 375 * Parameters: 376 * none. 377 * 378 * Returned value: 379 * handle on audio hardware output 380 */ 381 audio_io_handle_t getOutput(); 382 383 /* Returns the unique session ID associated with this track. 384 * 385 * Parameters: 386 * none. 387 * 388 * Returned value: 389 * AudioTrack session ID. 390 */ 391 int getSessionId() const { return mSessionId; } 392 393 /* Attach track auxiliary output to specified effect. Use effectId = 0 394 * to detach track from effect. 395 * 396 * Parameters: 397 * 398 * effectId: effectId obtained from AudioEffect::id(). 399 * 400 * Returned status (from utils/Errors.h) can be: 401 * - NO_ERROR: successful operation 402 * - INVALID_OPERATION: the effect is not an auxiliary effect. 403 * - BAD_VALUE: The specified effect ID is invalid 404 */ 405 status_t attachAuxEffect(int effectId); 406 407 /* Obtains a buffer of "frameCount" frames. The buffer must be 408 * filled entirely, and then released with releaseBuffer(). 409 * If the track is stopped, obtainBuffer() returns 410 * STOPPED instead of NO_ERROR as long as there are buffers available, 411 * at which point NO_MORE_BUFFERS is returned. 412 * Buffers will be returned until the pool 413 * is exhausted, at which point obtainBuffer() will either block 414 * or return WOULD_BLOCK depending on the value of the "blocking" 415 * parameter. 416 * 417 * Interpretation of waitCount: 418 * +n limits wait time to n * WAIT_PERIOD_MS, 419 * -1 causes an (almost) infinite wait time, 420 * 0 non-blocking. 421 * 422 * Buffer fields 423 * On entry: 424 * frameCount number of frames requested 425 * After error return: 426 * frameCount 0 427 * size 0 428 * raw undefined 429 * After successful return: 430 * frameCount actual number of frames available, <= number requested 431 * size actual number of bytes available 432 * raw pointer to the buffer 433 */ 434 435 enum { 436 NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 437 STOPPED = 1 438 }; 439 440 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); 441 442 /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */ 443 void releaseBuffer(Buffer* audioBuffer); 444 445 /* As a convenience we provide a write() interface to the audio buffer. 446 * This is implemented on top of obtainBuffer/releaseBuffer. For best 447 * performance use callbacks. Returns actual number of bytes written >= 0, 448 * or one of the following negative status codes: 449 * INVALID_OPERATION AudioTrack is configured for shared buffer mode 450 * BAD_VALUE size is invalid 451 * STOPPED AudioTrack was stopped during the write 452 * NO_MORE_BUFFERS when obtainBuffer() returns same 453 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 454 */ 455 ssize_t write(const void* buffer, size_t size); 456 457 /* 458 * Dumps the state of an audio track. 459 */ 460 status_t dump(int fd, const Vector<String16>& args) const; 461 462protected: 463 /* copying audio tracks is not allowed */ 464 AudioTrack(const AudioTrack& other); 465 AudioTrack& operator = (const AudioTrack& other); 466 467 /* a small internal class to handle the callback */ 468 class AudioTrackThread : public Thread 469 { 470 public: 471 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 472 473 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 474 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 475 virtual void requestExit(); 476 477 void pause(); // suspend thread from execution at next loop boundary 478 void resume(); // allow thread to execute, if not requested to exit 479 480 private: 481 friend class AudioTrack; 482 virtual bool threadLoop(); 483 AudioTrack& mReceiver; 484 ~AudioTrackThread(); 485 Mutex mMyLock; // Thread::mLock is private 486 Condition mMyCond; // Thread::mThreadExitedCondition is private 487 bool mPaused; // whether thread is currently paused 488 }; 489 490 // body of AudioTrackThread::threadLoop() 491 bool processAudioBuffer(const sp<AudioTrackThread>& thread); 492 493 // caller must hold lock on mLock for all _l methods 494 status_t createTrack_l(audio_stream_type_t streamType, 495 uint32_t sampleRate, 496 audio_format_t format, 497 size_t frameCount, 498 audio_output_flags_t flags, 499 const sp<IMemory>& sharedBuffer, 500 audio_io_handle_t output); 501 void flush_l(); 502 status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 503 audio_io_handle_t getOutput_l(); 504 status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart); 505 bool stopped_l() const { return !mActive; } 506 507 sp<IAudioTrack> mAudioTrack; 508 sp<IMemory> mCblkMemory; 509 sp<AudioTrackThread> mAudioTrackThread; 510 511 float mVolume[2]; 512 float mSendLevel; 513 size_t mFrameCount; // corresponds to current IAudioTrack 514 size_t mReqFrameCount; // frame count to request the next time a new 515 // IAudioTrack is needed 516 517 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 518 519 // Starting address of buffers in shared memory. If there is a shared buffer, mBuffers 520 // is the value of pointer() for the shared buffer, otherwise mBuffers points 521 // immediately after the control block. This address is for the mapping within client 522 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 523 void* mBuffers; 524 525 audio_format_t mFormat; // as requested by client, not forced to 16-bit 526 audio_stream_type_t mStreamType; 527 uint8_t mChannelCount; 528 uint8_t mMuted; 529 uint8_t mReserved; 530 audio_channel_mask_t mChannelMask; 531 532 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. 533 // For 8-bit PCM data, mFrameSizeAF is 534 // twice as large because data is expanded to 16-bit before being stored in buffer. 535 size_t mFrameSize; // app-level frame size 536 size_t mFrameSizeAF; // AudioFlinger frame size 537 538 status_t mStatus; 539 uint32_t mLatency; 540 541 bool mActive; // protected by mLock 542 543 callback_t mCbf; // callback handler for events, or NULL 544 void* mUserData; // for client callback handler 545 546 // for notification APIs 547 uint32_t mNotificationFramesReq; // requested number of frames between each 548 // notification callback 549 uint32_t mNotificationFramesAct; // actual number of frames between each 550 // notification callback 551 sp<IMemory> mSharedBuffer; 552 int mLoopCount; 553 uint32_t mRemainingFrames; 554 uint32_t mMarkerPosition; // in frames 555 bool mMarkerReached; 556 uint32_t mNewPosition; // in frames 557 uint32_t mUpdatePeriod; // in frames 558 559 bool mFlushed; // FIXME will be made obsolete by making flush() synchronous 560 audio_output_flags_t mFlags; 561 int mSessionId; 562 int mAuxEffectId; 563 564 // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock: 565 // 1. mLock 566 // 2. mCblk->lock 567 // It is OK to lock only mCblk->lock. 568 mutable Mutex mLock; 569 570 bool mIsTimed; 571 int mPreviousPriority; // before start() 572 SchedPolicy mPreviousSchedulingGroup; 573}; 574 575class TimedAudioTrack : public AudioTrack 576{ 577public: 578 TimedAudioTrack(); 579 580 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 581 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 582 583 /* queue a buffer obtained via allocateTimedBuffer for playback at the 584 given timestamp. PTS units are microseconds on the media time timeline. 585 The media time transform (set with setMediaTimeTransform) set by the 586 audio producer will handle converting from media time to local time 587 (perhaps going through the common time timeline in the case of 588 synchronized multiroom audio case) */ 589 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 590 591 /* define a transform between media time and either common time or 592 local time */ 593 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 594 status_t setMediaTimeTransform(const LinearTransform& xform, 595 TargetTimeline target); 596}; 597 598}; // namespace android 599 600#endif // ANDROID_AUDIOTRACK_H 601