AudioTrack.h revision 679e56914e83f05780003c71b84cc5b340e8fc0a
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 <cutils/sched_policy.h> 21#include <media/AudioSystem.h> 22#include <media/AudioTimestamp.h> 23#include <media/IAudioTrack.h> 24#include <media/AudioResamplerPublic.h> 25#include <utils/threads.h> 26 27namespace android { 28 29// ---------------------------------------------------------------------------- 30 31struct audio_track_cblk_t; 32class AudioTrackClientProxy; 33class StaticAudioTrackClientProxy; 34 35// ---------------------------------------------------------------------------- 36 37class AudioTrack : public RefBase 38{ 39public: 40 41 /* Events used by AudioTrack callback function (callback_t). 42 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 43 */ 44 enum event_type { 45 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 46 // If this event is delivered but the callback handler 47 // does not want to write more data, the handler must explicitly 48 // ignore the event by setting frameCount to zero. 49 EVENT_UNDERRUN = 1, // Buffer underrun occurred. 50 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 51 // loop start if loop count was not 0. 52 EVENT_MARKER = 3, // Playback head is at the specified marker position 53 // (See setMarkerPosition()). 54 EVENT_NEW_POS = 4, // Playback head is at a new position 55 // (See setPositionUpdatePeriod()). 56 EVENT_BUFFER_END = 5, // Playback head is at the end of the buffer. 57 // Not currently used by android.media.AudioTrack. 58 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 59 // voluntary invalidation by mediaserver, or mediaserver crash. 60 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 61 // back (after stop is called) 62#if 0 // FIXME not yet implemented 63 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 64 // in the mapping from frame position to presentation time. 65 // See AudioTimestamp for the information included with event. 66#endif 67 }; 68 69 /* Client should declare a Buffer and pass the address to obtainBuffer() 70 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 71 */ 72 73 class Buffer 74 { 75 public: 76 // FIXME use m prefix 77 size_t frameCount; // number of sample frames corresponding to size; 78 // on input to obtainBuffer() it is the number of frames desired, 79 // on output from obtainBuffer() it is the number of available 80 // [empty slots for] frames to be filled 81 // on input to releaseBuffer() it is currently ignored 82 83 size_t size; // input/output in bytes == frameCount * frameSize 84 // on input to obtainBuffer() it is ignored 85 // on output from obtainBuffer() it is the number of available 86 // [empty slots for] bytes to be filled, 87 // which is frameCount * frameSize 88 // on input to releaseBuffer() it is the number of bytes to 89 // release 90 // FIXME This is redundant with respect to frameCount. Consider 91 // removing size and making frameCount the primary field. 92 93 union { 94 void* raw; 95 short* i16; // signed 16-bit 96 int8_t* i8; // unsigned 8-bit, offset by 0x80 97 }; // input to obtainBuffer(): unused, output: pointer to buffer 98 }; 99 100 /* As a convenience, if a callback is supplied, a handler thread 101 * is automatically created with the appropriate priority. This thread 102 * invokes the callback when a new buffer becomes available or various conditions occur. 103 * Parameters: 104 * 105 * event: type of event notified (see enum AudioTrack::event_type). 106 * user: Pointer to context for use by the callback receiver. 107 * info: Pointer to optional parameter according to event type: 108 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 109 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 110 * written. 111 * - EVENT_UNDERRUN: unused. 112 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 113 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 114 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 115 * - EVENT_BUFFER_END: unused. 116 * - EVENT_NEW_IAUDIOTRACK: unused. 117 * - EVENT_STREAM_END: unused. 118 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 119 */ 120 121 typedef void (*callback_t)(int event, void* user, void *info); 122 123 /* Returns the minimum frame count required for the successful creation of 124 * an AudioTrack object. 125 * Returned status (from utils/Errors.h) can be: 126 * - NO_ERROR: successful operation 127 * - NO_INIT: audio server or audio hardware not initialized 128 * - BAD_VALUE: unsupported configuration 129 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 130 * and is undefined otherwise. 131 * FIXME This API assumes a route, and so should be deprecated. 132 */ 133 134 static status_t getMinFrameCount(size_t* frameCount, 135 audio_stream_type_t streamType, 136 uint32_t sampleRate); 137 138 /* How data is transferred to AudioTrack 139 */ 140 enum transfer_type { 141 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 142 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 143 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 144 TRANSFER_SYNC, // synchronous write() 145 TRANSFER_SHARED, // shared memory 146 }; 147 148 /* Constructs an uninitialized AudioTrack. No connection with 149 * AudioFlinger takes place. Use set() after this. 150 */ 151 AudioTrack(); 152 153 /* Creates an AudioTrack object and registers it with AudioFlinger. 154 * Once created, the track needs to be started before it can be used. 155 * Unspecified values are set to appropriate default values. 156 * 157 * Parameters: 158 * 159 * streamType: Select the type of audio stream this track is attached to 160 * (e.g. AUDIO_STREAM_MUSIC). 161 * sampleRate: Data source sampling rate in Hz. 162 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 163 * For direct and offloaded tracks, the possible format(s) depends on the 164 * output sink. 165 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 166 * frameCount: Minimum size of track PCM buffer in frames. This defines the 167 * application's contribution to the 168 * latency of the track. The actual size selected by the AudioTrack could be 169 * larger if the requested size is not compatible with current audio HAL 170 * configuration. Zero means to use a default value. 171 * flags: See comments on audio_output_flags_t in <system/audio.h>. 172 * cbf: Callback function. If not null, this function is called periodically 173 * to provide new data in TRANSFER_CALLBACK mode 174 * and inform of marker, position updates, etc. 175 * user: Context for use by the callback receiver. 176 * notificationFrames: The callback function is called each time notificationFrames PCM 177 * frames have been consumed from track input buffer. 178 * This is expressed in units of frames at the initial source sample rate. 179 * sessionId: Specific session ID, or zero to use default. 180 * transferType: How data is transferred to AudioTrack. 181 * offloadInfo: If not NULL, provides offload parameters for 182 * AudioSystem::getOutputForAttr(). 183 * uid: User ID of the app which initially requested this AudioTrack 184 * for power management tracking, or -1 for current user ID. 185 * pid: Process ID of the app which initially requested this AudioTrack 186 * for power management tracking, or -1 for current process ID. 187 * pAttributes: If not NULL, supersedes streamType for use case selection. 188 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 189 binder to AudioFlinger. 190 It will return an error instead. The application will recreate 191 the track based on offloading or different channel configuration, etc. 192 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 193 */ 194 195 AudioTrack( audio_stream_type_t streamType, 196 uint32_t sampleRate, 197 audio_format_t format, 198 audio_channel_mask_t channelMask, 199 size_t frameCount = 0, 200 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 201 callback_t cbf = NULL, 202 void* user = NULL, 203 uint32_t notificationFrames = 0, 204 int sessionId = AUDIO_SESSION_ALLOCATE, 205 transfer_type transferType = TRANSFER_DEFAULT, 206 const audio_offload_info_t *offloadInfo = NULL, 207 int uid = -1, 208 pid_t pid = -1, 209 const audio_attributes_t* pAttributes = NULL, 210 bool doNotReconnect = false); 211 212 /* Creates an audio track and registers it with AudioFlinger. 213 * With this constructor, the track is configured for static buffer mode. 214 * Data to be rendered is passed in a shared memory buffer 215 * identified by the argument sharedBuffer, which should be non-0. 216 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 217 * but without the ability to specify a non-zero value for the frameCount parameter. 218 * The memory should be initialized to the desired data before calling start(). 219 * The write() method is not supported in this case. 220 * It is recommended to pass a callback function to be notified of playback end by an 221 * EVENT_UNDERRUN event. 222 */ 223 224 AudioTrack( audio_stream_type_t streamType, 225 uint32_t sampleRate, 226 audio_format_t format, 227 audio_channel_mask_t channelMask, 228 const sp<IMemory>& sharedBuffer, 229 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 230 callback_t cbf = NULL, 231 void* user = NULL, 232 uint32_t notificationFrames = 0, 233 int sessionId = AUDIO_SESSION_ALLOCATE, 234 transfer_type transferType = TRANSFER_DEFAULT, 235 const audio_offload_info_t *offloadInfo = NULL, 236 int uid = -1, 237 pid_t pid = -1, 238 const audio_attributes_t* pAttributes = NULL, 239 bool doNotReconnect = false); 240 241 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 242 * Also destroys all resources associated with the AudioTrack. 243 */ 244protected: 245 virtual ~AudioTrack(); 246public: 247 248 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 249 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 250 * set() is not multi-thread safe. 251 * Returned status (from utils/Errors.h) can be: 252 * - NO_ERROR: successful initialization 253 * - INVALID_OPERATION: AudioTrack is already initialized 254 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 255 * - NO_INIT: audio server or audio hardware not initialized 256 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 257 * If sharedBuffer is non-0, the frameCount parameter is ignored and 258 * replaced by the shared buffer's total allocated size in frame units. 259 * 260 * Parameters not listed in the AudioTrack constructors above: 261 * 262 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 263 * 264 * Internal state post condition: 265 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 266 */ 267 status_t set(audio_stream_type_t streamType, 268 uint32_t sampleRate, 269 audio_format_t format, 270 audio_channel_mask_t channelMask, 271 size_t frameCount = 0, 272 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 273 callback_t cbf = NULL, 274 void* user = NULL, 275 uint32_t notificationFrames = 0, 276 const sp<IMemory>& sharedBuffer = 0, 277 bool threadCanCallJava = false, 278 int sessionId = AUDIO_SESSION_ALLOCATE, 279 transfer_type transferType = TRANSFER_DEFAULT, 280 const audio_offload_info_t *offloadInfo = NULL, 281 int uid = -1, 282 pid_t pid = -1, 283 const audio_attributes_t* pAttributes = NULL, 284 bool doNotReconnect = false); 285 286 /* Result of constructing the AudioTrack. This must be checked for successful initialization 287 * before using any AudioTrack API (except for set()), because using 288 * an uninitialized AudioTrack produces undefined results. 289 * See set() method above for possible return codes. 290 */ 291 status_t initCheck() const { return mStatus; } 292 293 /* Returns this track's estimated latency in milliseconds. 294 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 295 * and audio hardware driver. 296 */ 297 uint32_t latency() const { return mLatency; } 298 299 /* getters, see constructors and set() */ 300 301 audio_stream_type_t streamType() const; 302 audio_format_t format() const { return mFormat; } 303 304 /* Return frame size in bytes, which for linear PCM is 305 * channelCount * (bit depth per channel / 8). 306 * channelCount is determined from channelMask, and bit depth comes from format. 307 * For non-linear formats, the frame size is typically 1 byte. 308 */ 309 size_t frameSize() const { return mFrameSize; } 310 311 uint32_t channelCount() const { return mChannelCount; } 312 size_t frameCount() const { return mFrameCount; } 313 314 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 315 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 316 317 /* After it's created the track is not active. Call start() to 318 * make it active. If set, the callback will start being called. 319 * If the track was previously paused, volume is ramped up over the first mix buffer. 320 */ 321 status_t start(); 322 323 /* Stop a track. 324 * In static buffer mode, the track is stopped immediately. 325 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 326 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 327 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 328 * is first drained, mixed, and output, and only then is the track marked as stopped. 329 */ 330 void stop(); 331 bool stopped() const; 332 333 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 334 * This has the effect of draining the buffers without mixing or output. 335 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 336 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 337 */ 338 void flush(); 339 340 /* Pause a track. After pause, the callback will cease being called and 341 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 342 * and will fill up buffers until the pool is exhausted. 343 * Volume is ramped down over the next mix buffer following the pause request, 344 * and then the track is marked as paused. It can be resumed with ramp up by start(). 345 */ 346 void pause(); 347 348 /* Set volume for this track, mostly used for games' sound effects 349 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 350 * This is the older API. New applications should use setVolume(float) when possible. 351 */ 352 status_t setVolume(float left, float right); 353 354 /* Set volume for all channels. This is the preferred API for new applications, 355 * especially for multi-channel content. 356 */ 357 status_t setVolume(float volume); 358 359 /* Set the send level for this track. An auxiliary effect should be attached 360 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 361 */ 362 status_t setAuxEffectSendLevel(float level); 363 void getAuxEffectSendLevel(float* level) const; 364 365 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 366 */ 367 status_t setSampleRate(uint32_t sampleRate); 368 369 /* Return current source sample rate in Hz */ 370 uint32_t getSampleRate() const; 371 372 /* Return the original source sample rate in Hz. This corresponds to the sample rate 373 * if playback rate had normal speed and pitch. 374 */ 375 uint32_t getOriginalSampleRate() const; 376 377 /* Set source playback rate for timestretch 378 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 379 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 380 * 381 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 382 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 383 * 384 * Speed increases the playback rate of media, but does not alter pitch. 385 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 386 */ 387 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 388 389 /* Return current playback rate */ 390 const AudioPlaybackRate& getPlaybackRate() const; 391 392 /* Enables looping and sets the start and end points of looping. 393 * Only supported for static buffer mode. 394 * 395 * Parameters: 396 * 397 * loopStart: loop start in frames relative to start of buffer. 398 * loopEnd: loop end in frames relative to start of buffer. 399 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 400 * pending or active loop. loopCount == -1 means infinite looping. 401 * 402 * For proper operation the following condition must be respected: 403 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 404 * 405 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 406 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 407 * 408 */ 409 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 410 411 /* Sets marker position. When playback reaches the number of frames specified, a callback with 412 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 413 * notification callback. To set a marker at a position which would compute as 0, 414 * a workaround is to set the marker at a nearby position such as ~0 or 1. 415 * If the AudioTrack has been opened with no callback function associated, the operation will 416 * fail. 417 * 418 * Parameters: 419 * 420 * marker: marker position expressed in wrapping (overflow) frame units, 421 * like the return value of getPosition(). 422 * 423 * Returned status (from utils/Errors.h) can be: 424 * - NO_ERROR: successful operation 425 * - INVALID_OPERATION: the AudioTrack has no callback installed. 426 */ 427 status_t setMarkerPosition(uint32_t marker); 428 status_t getMarkerPosition(uint32_t *marker) const; 429 430 /* Sets position update period. Every time the number of frames specified has been played, 431 * a callback with event type EVENT_NEW_POS is called. 432 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 433 * callback. 434 * If the AudioTrack has been opened with no callback function associated, the operation will 435 * fail. 436 * Extremely small values may be rounded up to a value the implementation can support. 437 * 438 * Parameters: 439 * 440 * updatePeriod: position update notification period expressed in frames. 441 * 442 * Returned status (from utils/Errors.h) can be: 443 * - NO_ERROR: successful operation 444 * - INVALID_OPERATION: the AudioTrack has no callback installed. 445 */ 446 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 447 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 448 449 /* Sets playback head position. 450 * Only supported for static buffer mode. 451 * 452 * Parameters: 453 * 454 * position: New playback head position in frames relative to start of buffer. 455 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 456 * but will result in an immediate underrun if started. 457 * 458 * Returned status (from utils/Errors.h) can be: 459 * - NO_ERROR: successful operation 460 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 461 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 462 * buffer 463 */ 464 status_t setPosition(uint32_t position); 465 466 /* Return the total number of frames played since playback start. 467 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 468 * It is reset to zero by flush(), reload(), and stop(). 469 * 470 * Parameters: 471 * 472 * position: Address where to return play head position. 473 * 474 * Returned status (from utils/Errors.h) can be: 475 * - NO_ERROR: successful operation 476 * - BAD_VALUE: position is NULL 477 */ 478 status_t getPosition(uint32_t *position); 479 480 /* For static buffer mode only, this returns the current playback position in frames 481 * relative to start of buffer. It is analogous to the position units used by 482 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 483 */ 484 status_t getBufferPosition(uint32_t *position); 485 486 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 487 * rewriting the buffer before restarting playback after a stop. 488 * This method must be called with the AudioTrack in paused or stopped state. 489 * Not allowed in streaming mode. 490 * 491 * Returned status (from utils/Errors.h) can be: 492 * - NO_ERROR: successful operation 493 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 494 */ 495 status_t reload(); 496 497 /* Returns a handle on the audio output used by this AudioTrack. 498 * 499 * Parameters: 500 * none. 501 * 502 * Returned value: 503 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 504 * track needed to be re-created but that failed 505 */ 506private: 507 audio_io_handle_t getOutput() const; 508public: 509 510 /* Selects the audio device to use for output of this AudioTrack. A value of 511 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 512 * 513 * Parameters: 514 * The device ID of the selected device (as returned by the AudioDevicesManager API). 515 * 516 * Returned value: 517 * - NO_ERROR: successful operation 518 * TODO: what else can happen here? 519 */ 520 status_t setOutputDevice(audio_port_handle_t deviceId); 521 522 /* Returns the ID of the audio device selected for this AudioTrack. 523 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 524 * 525 * Parameters: 526 * none. 527 */ 528 audio_port_handle_t getOutputDevice(); 529 530 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 531 * attached. 532 * A value of AUDIO_PORT_HANDLE_NONE indicates the audio track is not attached to any output. 533 * 534 * Parameters: 535 * none. 536 */ 537 audio_port_handle_t getRoutedDeviceId(); 538 539 /* Returns the unique session ID associated with this track. 540 * 541 * Parameters: 542 * none. 543 * 544 * Returned value: 545 * AudioTrack session ID. 546 */ 547 int getSessionId() const { return mSessionId; } 548 549 /* Attach track auxiliary output to specified effect. Use effectId = 0 550 * to detach track from effect. 551 * 552 * Parameters: 553 * 554 * effectId: effectId obtained from AudioEffect::id(). 555 * 556 * Returned status (from utils/Errors.h) can be: 557 * - NO_ERROR: successful operation 558 * - INVALID_OPERATION: the effect is not an auxiliary effect. 559 * - BAD_VALUE: The specified effect ID is invalid 560 */ 561 status_t attachAuxEffect(int effectId); 562 563 /* Public API for TRANSFER_OBTAIN mode. 564 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 565 * After filling these slots with data, the caller should release them with releaseBuffer(). 566 * If the track buffer is not full, obtainBuffer() returns as many contiguous 567 * [empty slots for] frames as are available immediately. 568 * 569 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 570 * additional non-contiguous frames that are predicted to be available immediately, 571 * if the client were to release the first frames and then call obtainBuffer() again. 572 * This value is only a prediction, and needs to be confirmed. 573 * It will be set to zero for an error return. 574 * 575 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 576 * regardless of the value of waitCount. 577 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 578 * maximum timeout based on waitCount; see chart below. 579 * Buffers will be returned until the pool 580 * is exhausted, at which point obtainBuffer() will either block 581 * or return WOULD_BLOCK depending on the value of the "waitCount" 582 * parameter. 583 * 584 * Interpretation of waitCount: 585 * +n limits wait time to n * WAIT_PERIOD_MS, 586 * -1 causes an (almost) infinite wait time, 587 * 0 non-blocking. 588 * 589 * Buffer fields 590 * On entry: 591 * frameCount number of [empty slots for] frames requested 592 * size ignored 593 * raw ignored 594 * After error return: 595 * frameCount 0 596 * size 0 597 * raw undefined 598 * After successful return: 599 * frameCount actual number of [empty slots for] frames available, <= number requested 600 * size actual number of bytes available 601 * raw pointer to the buffer 602 */ 603 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 604 size_t *nonContig = NULL); 605 606private: 607 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 608 * additional non-contiguous frames that are predicted to be available immediately, 609 * if the client were to release the first frames and then call obtainBuffer() again. 610 * This value is only a prediction, and needs to be confirmed. 611 * It will be set to zero for an error return. 612 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 613 * in case the requested amount of frames is in two or more non-contiguous regions. 614 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 615 */ 616 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 617 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 618public: 619 620 /* Public API for TRANSFER_OBTAIN mode. 621 * Release a filled buffer of frames for AudioFlinger to process. 622 * 623 * Buffer fields: 624 * frameCount currently ignored but recommend to set to actual number of frames filled 625 * size actual number of bytes filled, must be multiple of frameSize 626 * raw ignored 627 */ 628 void releaseBuffer(const Buffer* audioBuffer); 629 630 /* As a convenience we provide a write() interface to the audio buffer. 631 * Input parameter 'size' is in byte units. 632 * This is implemented on top of obtainBuffer/releaseBuffer. For best 633 * performance use callbacks. Returns actual number of bytes written >= 0, 634 * or one of the following negative status codes: 635 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 636 * BAD_VALUE size is invalid 637 * WOULD_BLOCK when obtainBuffer() returns same, or 638 * AudioTrack was stopped during the write 639 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 640 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 641 * false for the method to return immediately without waiting to try multiple times to write 642 * the full content of the buffer. 643 */ 644 ssize_t write(const void* buffer, size_t size, bool blocking = true); 645 646 /* 647 * Dumps the state of an audio track. 648 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 649 */ 650 status_t dump(int fd, const Vector<String16>& args) const; 651 652 /* 653 * Return the total number of frames which AudioFlinger desired but were unavailable, 654 * and thus which resulted in an underrun. Reset to zero by stop(). 655 */ 656 uint32_t getUnderrunFrames() const; 657 658 /* Get the flags */ 659 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 660 661 /* Set parameters - only possible when using direct output */ 662 status_t setParameters(const String8& keyValuePairs); 663 664 /* Get parameters */ 665 String8 getParameters(const String8& keys); 666 667 /* Poll for a timestamp on demand. 668 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 669 * or if you need to get the most recent timestamp outside of the event callback handler. 670 * Caution: calling this method too often may be inefficient; 671 * if you need a high resolution mapping between frame position and presentation time, 672 * consider implementing that at application level, based on the low resolution timestamps. 673 * Returns NO_ERROR if timestamp is valid. 674 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 675 * start/ACTIVE, when the number of frames consumed is less than the 676 * overall hardware latency to physical output. In WOULD_BLOCK cases, 677 * one might poll again, or use getPosition(), or use 0 position and 678 * current time for the timestamp. 679 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 680 * 681 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 682 */ 683 status_t getTimestamp(AudioTimestamp& timestamp); 684 685 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 686 * AudioTrack is routed is updated. 687 * Replaces any previously installed callback. 688 * Parameters: 689 * callback: The callback interface 690 * Returns NO_ERROR if successful. 691 * INVALID_OPERATION if the same callback is already installed. 692 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 693 * BAD_VALUE if the callback is NULL 694 */ 695 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 696 697 /* remove an AudioDeviceCallback. 698 * Parameters: 699 * callback: The callback interface 700 * Returns NO_ERROR if successful. 701 * INVALID_OPERATION if the callback is not installed 702 * BAD_VALUE if the callback is NULL 703 */ 704 status_t removeAudioDeviceCallback( 705 const sp<AudioSystem::AudioDeviceCallback>& callback); 706 707protected: 708 /* copying audio tracks is not allowed */ 709 AudioTrack(const AudioTrack& other); 710 AudioTrack& operator = (const AudioTrack& other); 711 712 /* a small internal class to handle the callback */ 713 class AudioTrackThread : public Thread 714 { 715 public: 716 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 717 718 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 719 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 720 virtual void requestExit(); 721 722 void pause(); // suspend thread from execution at next loop boundary 723 void resume(); // allow thread to execute, if not requested to exit 724 void wake(); // wake to handle changed notification conditions. 725 726 private: 727 void pauseInternal(nsecs_t ns = 0LL); 728 // like pause(), but only used internally within thread 729 730 friend class AudioTrack; 731 virtual bool threadLoop(); 732 AudioTrack& mReceiver; 733 virtual ~AudioTrackThread(); 734 Mutex mMyLock; // Thread::mLock is private 735 Condition mMyCond; // Thread::mThreadExitedCondition is private 736 bool mPaused; // whether thread is requested to pause at next loop entry 737 bool mPausedInt; // whether thread internally requests pause 738 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 739 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 740 // to processAudioBuffer() as state may have changed 741 // since pause time calculated. 742 }; 743 744 // body of AudioTrackThread::threadLoop() 745 // returns the maximum amount of time before we would like to run again, where: 746 // 0 immediately 747 // > 0 no later than this many nanoseconds from now 748 // NS_WHENEVER still active but no particular deadline 749 // NS_INACTIVE inactive so don't run again until re-started 750 // NS_NEVER never again 751 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 752 nsecs_t processAudioBuffer(); 753 754 // caller must hold lock on mLock for all _l methods 755 756 status_t createTrack_l(); 757 758 // can only be called when mState != STATE_ACTIVE 759 void flush_l(); 760 761 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 762 763 // FIXME enum is faster than strcmp() for parameter 'from' 764 status_t restoreTrack_l(const char *from); 765 766 bool isOffloaded() const; 767 bool isDirect() const; 768 bool isOffloadedOrDirect() const; 769 770 bool isOffloaded_l() const 771 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 772 773 bool isOffloadedOrDirect_l() const 774 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 775 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 776 777 bool isDirect_l() const 778 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 779 780 // increment mPosition by the delta of mServer, and return new value of mPosition 781 uint32_t updateAndGetPosition_l(); 782 783 // check sample rate and speed is compatible with AudioTrack 784 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; 785 786 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 787 sp<IAudioTrack> mAudioTrack; 788 sp<IMemory> mCblkMemory; 789 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 790 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 791 792 sp<AudioTrackThread> mAudioTrackThread; 793 794 float mVolume[2]; 795 float mSendLevel; 796 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 797 uint32_t mOriginalSampleRate; 798 AudioPlaybackRate mPlaybackRate; 799 size_t mFrameCount; // corresponds to current IAudioTrack, value is 800 // reported back by AudioFlinger to the client 801 size_t mReqFrameCount; // frame count to request the first or next time 802 // a new IAudioTrack is needed, non-decreasing 803 804 // constant after constructor or set() 805 audio_format_t mFormat; // as requested by client, not forced to 16-bit 806 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 807 // this AudioTrack has valid attributes 808 uint32_t mChannelCount; 809 audio_channel_mask_t mChannelMask; 810 sp<IMemory> mSharedBuffer; 811 transfer_type mTransfer; 812 audio_offload_info_t mOffloadInfoCopy; 813 const audio_offload_info_t* mOffloadInfo; 814 audio_attributes_t mAttributes; 815 816 size_t mFrameSize; // frame size in bytes 817 818 status_t mStatus; 819 820 // can change dynamically when IAudioTrack invalidated 821 uint32_t mLatency; // in ms 822 823 // Indicates the current track state. Protected by mLock. 824 enum State { 825 STATE_ACTIVE, 826 STATE_STOPPED, 827 STATE_PAUSED, 828 STATE_PAUSED_STOPPING, 829 STATE_FLUSHED, 830 STATE_STOPPING, 831 } mState; 832 833 // for client callback handler 834 callback_t mCbf; // callback handler for events, or NULL 835 void* mUserData; 836 837 // for notification APIs 838 uint32_t mNotificationFramesReq; // requested number of frames between each 839 // notification callback, 840 // at initial source sample rate 841 uint32_t mNotificationFramesAct; // actual number of frames between each 842 // notification callback, 843 // at initial source sample rate 844 bool mRefreshRemaining; // processAudioBuffer() should refresh 845 // mRemainingFrames and mRetryOnPartialBuffer 846 847 // used for static track cbf and restoration 848 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 849 uint32_t mLoopStart; // last setLoop loopStart 850 uint32_t mLoopEnd; // last setLoop loopEnd 851 int32_t mLoopCountNotified; // the last loopCount notified by callback. 852 // mLoopCountNotified counts down, matching 853 // the remaining loop count for static track 854 // playback. 855 856 // These are private to processAudioBuffer(), and are not protected by a lock 857 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 858 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 859 uint32_t mObservedSequence; // last observed value of mSequence 860 861 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 862 bool mMarkerReached; 863 uint32_t mNewPosition; // in frames 864 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 865 866 uint32_t mServer; // in frames, last known mProxy->getPosition() 867 // which is count of frames consumed by server, 868 // reset by new IAudioTrack, 869 // whether it is reset by stop() is TBD 870 uint32_t mPosition; // in frames, like mServer except continues 871 // monotonically after new IAudioTrack, 872 // and could be easily widened to uint64_t 873 uint32_t mReleased; // in frames, count of frames released to server 874 // but not necessarily consumed by server, 875 // reset by stop() but continues monotonically 876 // after new IAudioTrack to restore mPosition, 877 // and could be easily widened to uint64_t 878 int64_t mStartUs; // the start time after flush or stop. 879 // only used for offloaded and direct tracks. 880 881 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 882 bool mRetrogradeMotionReported; // reduce log spam 883 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 884 885 audio_output_flags_t mFlags; 886 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 887 // mLock must be held to read or write those bits reliably. 888 889 bool mDoNotReconnect; 890 891 int mSessionId; 892 int mAuxEffectId; 893 894 mutable Mutex mLock; 895 896 bool mIsTimed; 897 int mPreviousPriority; // before start() 898 SchedPolicy mPreviousSchedulingGroup; 899 bool mAwaitBoost; // thread should wait for priority boost before running 900 901 // The proxy should only be referenced while a lock is held because the proxy isn't 902 // multi-thread safe, especially the SingleStateQueue part of the proxy. 903 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 904 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 905 // them around in case they are replaced during the obtainBuffer(). 906 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 907 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 908 909 bool mInUnderrun; // whether track is currently in underrun state 910 uint32_t mPausedPosition; 911 912 // For Device Selection API 913 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 914 audio_port_handle_t mSelectedDeviceId; 915 916private: 917 class DeathNotifier : public IBinder::DeathRecipient { 918 public: 919 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 920 protected: 921 virtual void binderDied(const wp<IBinder>& who); 922 private: 923 const wp<AudioTrack> mAudioTrack; 924 }; 925 926 sp<DeathNotifier> mDeathNotifier; 927 uint32_t mSequence; // incremented for each new IAudioTrack attempt 928 int mClientUid; 929 pid_t mClientPid; 930 931 sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 932}; 933 934class TimedAudioTrack : public AudioTrack 935{ 936public: 937 TimedAudioTrack(); 938 939 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 940 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 941 942 /* queue a buffer obtained via allocateTimedBuffer for playback at the 943 given timestamp. PTS units are microseconds on the media time timeline. 944 The media time transform (set with setMediaTimeTransform) set by the 945 audio producer will handle converting from media time to local time 946 (perhaps going through the common time timeline in the case of 947 synchronized multiroom audio case) */ 948 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 949 950 /* define a transform between media time and either common time or 951 local time */ 952 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 953 status_t setMediaTimeTransform(const LinearTransform& xform, 954 TargetTimeline target); 955}; 956 957}; // namespace android 958 959#endif // ANDROID_AUDIOTRACK_H 960