AudioTrack.h revision 4bae3649d504d590a546717a8e49f96a30d9a745
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() method is 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 or paused track. All previously buffered data is discarded immediately. 251 * This has the effect of draining the buffers without mixing or output. 252 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 253 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 254 */ 255 void flush(); 256 257 /* Pause a track. If set, the callback will cease being called and 258 * obtainBuffer returns STOPPED. Note that obtainBuffer() still works 259 * and will fill up buffers until the pool is exhausted. 260 */ 261 void pause(); 262 263 /* Set volume for this track, mostly used for games' sound effects 264 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 265 * This is the older API. New applications should use setVolume(float) when possible. 266 */ 267 status_t setVolume(float left, float right); 268 269 /* Set volume for all channels. This is the preferred API for new applications, 270 * especially for multi-channel content. 271 */ 272 status_t setVolume(float volume); 273 274 /* Set the send level for this track. An auxiliary effect should be attached 275 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 276 */ 277 status_t setAuxEffectSendLevel(float level); 278 void getAuxEffectSendLevel(float* level) const; 279 280 /* Set sample rate for this track in Hz, mostly used for games' sound effects 281 */ 282 status_t setSampleRate(uint32_t sampleRate); 283 284 /* Return current sample rate in Hz, or 0 if unknown */ 285 uint32_t getSampleRate() const; 286 287 /* Enables looping and sets the start and end points of looping. 288 * 289 * Parameters: 290 * 291 * loopStart: loop start expressed as the number of PCM frames played since AudioTrack start. 292 * loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start. 293 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 294 * pending or active loop. loopCount = -1 means infinite looping. 295 * 296 * For proper operation the following condition must be respected: 297 * (loopEnd-loopStart) <= framecount() 298 */ 299 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 300 301 /* Sets marker position. When playback reaches the number of frames specified, a callback with 302 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 303 * notification callback. 304 * If the AudioTrack has been opened with no callback function associated, the operation will 305 * fail. 306 * 307 * Parameters: 308 * 309 * marker: marker position expressed in frames. 310 * 311 * Returned status (from utils/Errors.h) can be: 312 * - NO_ERROR: successful operation 313 * - INVALID_OPERATION: the AudioTrack has no callback installed. 314 */ 315 status_t setMarkerPosition(uint32_t marker); 316 status_t getMarkerPosition(uint32_t *marker) const; 317 318 319 /* Sets position update period. Every time the number of frames specified has been played, 320 * a callback with event type EVENT_NEW_POS is called. 321 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 322 * callback. 323 * If the AudioTrack has been opened with no callback function associated, the operation will 324 * fail. 325 * 326 * Parameters: 327 * 328 * updatePeriod: position update notification period expressed in frames. 329 * 330 * Returned status (from utils/Errors.h) can be: 331 * - NO_ERROR: successful operation 332 * - INVALID_OPERATION: the AudioTrack has no callback installed. 333 */ 334 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 335 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 336 337 /* Sets playback head position within AudioTrack buffer. The new position is specified 338 * in number of frames. 339 * This method must be called with the AudioTrack in paused or stopped state. 340 * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames. 341 * Therefore using this method makes sense only when playing a "static" audio buffer 342 * as opposed to streaming. 343 * The getPosition() method on the other hand returns the total number of frames played since 344 * playback start. 345 * 346 * Parameters: 347 * 348 * position: New playback head position within AudioTrack buffer. 349 * 350 * Returned status (from utils/Errors.h) can be: 351 * - NO_ERROR: successful operation 352 * - INVALID_OPERATION: the AudioTrack is not stopped. 353 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 354 * buffer 355 */ 356 status_t setPosition(uint32_t position); 357 status_t getPosition(uint32_t *position); 358 359 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 360 * rewriting the buffer before restarting playback after a stop. 361 * This method must be called with the AudioTrack in paused or stopped state. 362 * 363 * Returned status (from utils/Errors.h) can be: 364 * - NO_ERROR: successful operation 365 * - INVALID_OPERATION: the AudioTrack is not stopped. 366 */ 367 status_t reload(); 368 369 /* Returns a handle on the audio output used by this AudioTrack. 370 * 371 * Parameters: 372 * none. 373 * 374 * Returned value: 375 * handle on audio hardware output 376 */ 377 audio_io_handle_t getOutput(); 378 379 /* Returns the unique session ID associated with this track. 380 * 381 * Parameters: 382 * none. 383 * 384 * Returned value: 385 * AudioTrack session ID. 386 */ 387 int getSessionId() const { return mSessionId; } 388 389 /* Attach track auxiliary output to specified effect. Use effectId = 0 390 * to detach track from effect. 391 * 392 * Parameters: 393 * 394 * effectId: effectId obtained from AudioEffect::id(). 395 * 396 * Returned status (from utils/Errors.h) can be: 397 * - NO_ERROR: successful operation 398 * - INVALID_OPERATION: the effect is not an auxiliary effect. 399 * - BAD_VALUE: The specified effect ID is invalid 400 */ 401 status_t attachAuxEffect(int effectId); 402 403 /* Obtains a buffer of "frameCount" frames. The buffer must be 404 * filled entirely, and then released with releaseBuffer(). 405 * If the track is stopped, obtainBuffer() returns 406 * STOPPED instead of NO_ERROR as long as there are buffers available, 407 * at which point NO_MORE_BUFFERS is returned. 408 * Buffers will be returned until the pool 409 * is exhausted, at which point obtainBuffer() will either block 410 * or return WOULD_BLOCK depending on the value of the "blocking" 411 * parameter. 412 * 413 * Interpretation of waitCount: 414 * +n limits wait time to n * WAIT_PERIOD_MS, 415 * -1 causes an (almost) infinite wait time, 416 * 0 non-blocking. 417 * 418 * Buffer fields 419 * On entry: 420 * frameCount number of frames requested 421 * After error return: 422 * frameCount 0 423 * size 0 424 * raw undefined 425 * After successful return: 426 * frameCount actual number of frames available, <= number requested 427 * size actual number of bytes available 428 * raw pointer to the buffer 429 */ 430 431 enum { 432 NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value 433 STOPPED = 1 434 }; 435 436 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); 437 438 /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */ 439 void releaseBuffer(Buffer* audioBuffer); 440 441 /* As a convenience we provide a write() interface to the audio buffer. 442 * This is implemented on top of obtainBuffer/releaseBuffer. For best 443 * performance use callbacks. Returns actual number of bytes written >= 0, 444 * or one of the following negative status codes: 445 * INVALID_OPERATION AudioTrack is configured for shared buffer mode 446 * BAD_VALUE size is invalid 447 * STOPPED AudioTrack was stopped during the write 448 * NO_MORE_BUFFERS when obtainBuffer() returns same 449 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 450 */ 451 ssize_t write(const void* buffer, size_t size); 452 453 /* 454 * Dumps the state of an audio track. 455 */ 456 status_t dump(int fd, const Vector<String16>& args) const; 457 458protected: 459 /* copying audio tracks is not allowed */ 460 AudioTrack(const AudioTrack& other); 461 AudioTrack& operator = (const AudioTrack& other); 462 463 /* a small internal class to handle the callback */ 464 class AudioTrackThread : public Thread 465 { 466 public: 467 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 468 469 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 470 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 471 virtual void requestExit(); 472 473 void pause(); // suspend thread from execution at next loop boundary 474 void resume(); // allow thread to execute, if not requested to exit 475 476 private: 477 friend class AudioTrack; 478 virtual bool threadLoop(); 479 AudioTrack& mReceiver; 480 ~AudioTrackThread(); 481 Mutex mMyLock; // Thread::mLock is private 482 Condition mMyCond; // Thread::mThreadExitedCondition is private 483 bool mPaused; // whether thread is currently paused 484 }; 485 486 // body of AudioTrackThread::threadLoop() 487 bool processAudioBuffer(const sp<AudioTrackThread>& thread); 488 489 // caller must hold lock on mLock for all _l methods 490 status_t createTrack_l(audio_stream_type_t streamType, 491 uint32_t sampleRate, 492 audio_format_t format, 493 size_t frameCount, 494 audio_output_flags_t flags, 495 const sp<IMemory>& sharedBuffer, 496 audio_io_handle_t output); 497 498 // can only be called when !mActive 499 void flush_l(); 500 501 status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 502 audio_io_handle_t getOutput_l(); 503 status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart); 504 bool stopped_l() const { return !mActive; } 505 506 sp<IAudioTrack> mAudioTrack; 507 sp<IMemory> mCblkMemory; 508 sp<AudioTrackThread> mAudioTrackThread; 509 510 float mVolume[2]; 511 float mSendLevel; 512 size_t mFrameCount; // corresponds to current IAudioTrack 513 size_t mReqFrameCount; // frame count to request the next time a new 514 // IAudioTrack is needed 515 516 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 517 518 // Starting address of buffers in shared memory. If there is a shared buffer, mBuffers 519 // is the value of pointer() for the shared buffer, otherwise mBuffers points 520 // immediately after the control block. This address is for the mapping within client 521 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 522 void* mBuffers; 523 524 audio_format_t mFormat; // as requested by client, not forced to 16-bit 525 audio_stream_type_t mStreamType; 526 uint32_t mChannelCount; 527 audio_channel_mask_t mChannelMask; 528 529 // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. 530 // For 8-bit PCM data, mFrameSizeAF is 531 // twice as large because data is expanded to 16-bit before being stored in buffer. 532 size_t mFrameSize; // app-level frame size 533 size_t mFrameSizeAF; // AudioFlinger frame size 534 535 status_t mStatus; 536 uint32_t mLatency; 537 538 bool mActive; // protected by mLock 539 540 callback_t mCbf; // callback handler for events, or NULL 541 void* mUserData; // for client callback handler 542 543 // for notification APIs 544 uint32_t mNotificationFramesReq; // requested number of frames between each 545 // notification callback 546 uint32_t mNotificationFramesAct; // actual number of frames between each 547 // notification callback 548 sp<IMemory> mSharedBuffer; 549 int mLoopCount; 550 uint32_t mRemainingFrames; 551 uint32_t mMarkerPosition; // in frames 552 bool mMarkerReached; 553 uint32_t mNewPosition; // in frames 554 uint32_t mUpdatePeriod; // in frames 555 556 bool mFlushed; // FIXME will be made obsolete by making flush() synchronous 557 audio_output_flags_t mFlags; 558 int mSessionId; 559 int mAuxEffectId; 560 561 // When locking both mLock and mCblk->lock, must lock in this order to avoid deadlock: 562 // 1. mLock 563 // 2. mCblk->lock 564 // It is OK to lock only mCblk->lock. 565 mutable Mutex mLock; 566 567 bool mIsTimed; 568 int mPreviousPriority; // before start() 569 SchedPolicy mPreviousSchedulingGroup; 570}; 571 572class TimedAudioTrack : public AudioTrack 573{ 574public: 575 TimedAudioTrack(); 576 577 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 578 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 579 580 /* queue a buffer obtained via allocateTimedBuffer for playback at the 581 given timestamp. PTS units are microseconds on the media time timeline. 582 The media time transform (set with setMediaTimeTransform) set by the 583 audio producer will handle converting from media time to local time 584 (perhaps going through the common time timeline in the case of 585 synchronized multiroom audio case) */ 586 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 587 588 /* define a transform between media time and either common time or 589 local time */ 590 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 591 status_t setMediaTimeTransform(const LinearTransform& xform, 592 TargetTimeline target); 593}; 594 595}; // namespace android 596 597#endif // ANDROID_AUDIOTRACK_H 598