AudioTrack.h revision faeb0f291330134dc4468359a36e099aae508449
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 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 63 // in the mapping from frame position to presentation time. 64 // See AudioTimestamp for the information included with event. 65 }; 66 67 /* Client should declare a Buffer and pass the address to obtainBuffer() 68 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 69 */ 70 71 class Buffer 72 { 73 public: 74 // FIXME use m prefix 75 size_t frameCount; // number of sample frames corresponding to size; 76 // on input to obtainBuffer() it is the number of frames desired, 77 // on output from obtainBuffer() it is the number of available 78 // [empty slots for] frames to be filled 79 // on input to releaseBuffer() it is currently ignored 80 81 size_t size; // input/output in bytes == frameCount * frameSize 82 // on input to obtainBuffer() it is ignored 83 // on output from obtainBuffer() it is the number of available 84 // [empty slots for] bytes to be filled, 85 // which is frameCount * frameSize 86 // on input to releaseBuffer() it is the number of bytes to 87 // release 88 // FIXME This is redundant with respect to frameCount. Consider 89 // removing size and making frameCount the primary field. 90 91 union { 92 void* raw; 93 short* i16; // signed 16-bit 94 int8_t* i8; // unsigned 8-bit, offset by 0x80 95 }; // input to obtainBuffer(): unused, output: pointer to buffer 96 }; 97 98 /* As a convenience, if a callback is supplied, a handler thread 99 * is automatically created with the appropriate priority. This thread 100 * invokes the callback when a new buffer becomes available or various conditions occur. 101 * Parameters: 102 * 103 * event: type of event notified (see enum AudioTrack::event_type). 104 * user: Pointer to context for use by the callback receiver. 105 * info: Pointer to optional parameter according to event type: 106 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 107 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 108 * written. 109 * - EVENT_UNDERRUN: unused. 110 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 111 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 112 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 113 * - EVENT_BUFFER_END: unused. 114 * - EVENT_NEW_IAUDIOTRACK: unused. 115 * - EVENT_STREAM_END: unused. 116 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 117 */ 118 119 typedef void (*callback_t)(int event, void* user, void *info); 120 121 /* Returns the minimum frame count required for the successful creation of 122 * an AudioTrack object. 123 * Returned status (from utils/Errors.h) can be: 124 * - NO_ERROR: successful operation 125 * - NO_INIT: audio server or audio hardware not initialized 126 * - BAD_VALUE: unsupported configuration 127 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 128 * and is undefined otherwise. 129 * FIXME This API assumes a route, and so should be deprecated. 130 */ 131 132 static status_t getMinFrameCount(size_t* frameCount, 133 audio_stream_type_t streamType, 134 uint32_t sampleRate); 135 136 /* How data is transferred to AudioTrack 137 */ 138 enum transfer_type { 139 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 140 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 141 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 142 TRANSFER_SYNC, // synchronous write() 143 TRANSFER_SHARED, // shared memory 144 }; 145 146 /* Constructs an uninitialized AudioTrack. No connection with 147 * AudioFlinger takes place. Use set() after this. 148 */ 149 AudioTrack(); 150 151 /* Creates an AudioTrack object and registers it with AudioFlinger. 152 * Once created, the track needs to be started before it can be used. 153 * Unspecified values are set to appropriate default values. 154 * 155 * Parameters: 156 * 157 * streamType: Select the type of audio stream this track is attached to 158 * (e.g. AUDIO_STREAM_MUSIC). 159 * sampleRate: Data source sampling rate in Hz. 160 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 161 * For direct and offloaded tracks, the possible format(s) depends on the 162 * output sink. 163 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 164 * frameCount: Minimum size of track PCM buffer in frames. This defines the 165 * application's contribution to the 166 * latency of the track. The actual size selected by the AudioTrack could be 167 * larger if the requested size is not compatible with current audio HAL 168 * configuration. Zero means to use a default value. 169 * flags: See comments on audio_output_flags_t in <system/audio.h>. 170 * cbf: Callback function. If not null, this function is called periodically 171 * to provide new data in TRANSFER_CALLBACK mode 172 * and inform of marker, position updates, etc. 173 * user: Context for use by the callback receiver. 174 * notificationFrames: The callback function is called each time notificationFrames PCM 175 * frames have been consumed from track input buffer. 176 * This is expressed in units of frames at the initial source sample rate. 177 * sessionId: Specific session ID, or zero to use default. 178 * transferType: How data is transferred to AudioTrack. 179 * offloadInfo: If not NULL, provides offload parameters for 180 * AudioSystem::getOutputForAttr(). 181 * uid: User ID of the app which initially requested this AudioTrack 182 * for power management tracking, or -1 for current user ID. 183 * pid: Process ID of the app which initially requested this AudioTrack 184 * for power management tracking, or -1 for current process ID. 185 * pAttributes: If not NULL, supersedes streamType for use case selection. 186 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 187 binder to AudioFlinger. 188 It will return an error instead. The application will recreate 189 the track based on offloading or different channel configuration, etc. 190 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 191 */ 192 193 AudioTrack( audio_stream_type_t streamType, 194 uint32_t sampleRate, 195 audio_format_t format, 196 audio_channel_mask_t channelMask, 197 size_t frameCount = 0, 198 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 199 callback_t cbf = NULL, 200 void* user = NULL, 201 uint32_t notificationFrames = 0, 202 int sessionId = AUDIO_SESSION_ALLOCATE, 203 transfer_type transferType = TRANSFER_DEFAULT, 204 const audio_offload_info_t *offloadInfo = NULL, 205 int uid = -1, 206 pid_t pid = -1, 207 const audio_attributes_t* pAttributes = NULL, 208 bool doNotReconnect = false); 209 210 /* Creates an audio track and registers it with AudioFlinger. 211 * With this constructor, the track is configured for static buffer mode. 212 * Data to be rendered is passed in a shared memory buffer 213 * identified by the argument sharedBuffer, which should be non-0. 214 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 215 * but without the ability to specify a non-zero value for the frameCount parameter. 216 * The memory should be initialized to the desired data before calling start(). 217 * The write() method is not supported in this case. 218 * It is recommended to pass a callback function to be notified of playback end by an 219 * EVENT_UNDERRUN event. 220 */ 221 222 AudioTrack( audio_stream_type_t streamType, 223 uint32_t sampleRate, 224 audio_format_t format, 225 audio_channel_mask_t channelMask, 226 const sp<IMemory>& sharedBuffer, 227 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 228 callback_t cbf = NULL, 229 void* user = NULL, 230 uint32_t notificationFrames = 0, 231 int sessionId = AUDIO_SESSION_ALLOCATE, 232 transfer_type transferType = TRANSFER_DEFAULT, 233 const audio_offload_info_t *offloadInfo = NULL, 234 int uid = -1, 235 pid_t pid = -1, 236 const audio_attributes_t* pAttributes = NULL, 237 bool doNotReconnect = false); 238 239 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 240 * Also destroys all resources associated with the AudioTrack. 241 */ 242protected: 243 virtual ~AudioTrack(); 244public: 245 246 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 247 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 248 * set() is not multi-thread safe. 249 * Returned status (from utils/Errors.h) can be: 250 * - NO_ERROR: successful initialization 251 * - INVALID_OPERATION: AudioTrack is already initialized 252 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 253 * - NO_INIT: audio server or audio hardware not initialized 254 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 255 * If sharedBuffer is non-0, the frameCount parameter is ignored and 256 * replaced by the shared buffer's total allocated size in frame units. 257 * 258 * Parameters not listed in the AudioTrack constructors above: 259 * 260 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 261 * 262 * Internal state post condition: 263 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 264 */ 265 status_t set(audio_stream_type_t streamType, 266 uint32_t sampleRate, 267 audio_format_t format, 268 audio_channel_mask_t channelMask, 269 size_t frameCount = 0, 270 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 271 callback_t cbf = NULL, 272 void* user = NULL, 273 uint32_t notificationFrames = 0, 274 const sp<IMemory>& sharedBuffer = 0, 275 bool threadCanCallJava = false, 276 int sessionId = AUDIO_SESSION_ALLOCATE, 277 transfer_type transferType = TRANSFER_DEFAULT, 278 const audio_offload_info_t *offloadInfo = NULL, 279 int uid = -1, 280 pid_t pid = -1, 281 const audio_attributes_t* pAttributes = NULL, 282 bool doNotReconnect = false); 283 284 /* Result of constructing the AudioTrack. This must be checked for successful initialization 285 * before using any AudioTrack API (except for set()), because using 286 * an uninitialized AudioTrack produces undefined results. 287 * See set() method above for possible return codes. 288 */ 289 status_t initCheck() const { return mStatus; } 290 291 /* Returns this track's estimated latency in milliseconds. 292 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 293 * and audio hardware driver. 294 */ 295 uint32_t latency() const { return mLatency; } 296 297 /* getters, see constructors and set() */ 298 299 audio_stream_type_t streamType() const; 300 audio_format_t format() const { return mFormat; } 301 302 /* Return frame size in bytes, which for linear PCM is 303 * channelCount * (bit depth per channel / 8). 304 * channelCount is determined from channelMask, and bit depth comes from format. 305 * For non-linear formats, the frame size is typically 1 byte. 306 */ 307 size_t frameSize() const { return mFrameSize; } 308 309 uint32_t channelCount() const { return mChannelCount; } 310 size_t frameCount() const { return mFrameCount; } 311 312 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 313 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 314 315 /* After it's created the track is not active. Call start() to 316 * make it active. If set, the callback will start being called. 317 * If the track was previously paused, volume is ramped up over the first mix buffer. 318 */ 319 status_t start(); 320 321 /* Stop a track. 322 * In static buffer mode, the track is stopped immediately. 323 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 324 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 325 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 326 * is first drained, mixed, and output, and only then is the track marked as stopped. 327 */ 328 void stop(); 329 bool stopped() const; 330 331 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 332 * This has the effect of draining the buffers without mixing or output. 333 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 334 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 335 */ 336 void flush(); 337 338 /* Pause a track. After pause, the callback will cease being called and 339 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 340 * and will fill up buffers until the pool is exhausted. 341 * Volume is ramped down over the next mix buffer following the pause request, 342 * and then the track is marked as paused. It can be resumed with ramp up by start(). 343 */ 344 void pause(); 345 346 /* Set volume for this track, mostly used for games' sound effects 347 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 348 * This is the older API. New applications should use setVolume(float) when possible. 349 */ 350 status_t setVolume(float left, float right); 351 352 /* Set volume for all channels. This is the preferred API for new applications, 353 * especially for multi-channel content. 354 */ 355 status_t setVolume(float volume); 356 357 /* Set the send level for this track. An auxiliary effect should be attached 358 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 359 */ 360 status_t setAuxEffectSendLevel(float level); 361 void getAuxEffectSendLevel(float* level) const; 362 363 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 364 */ 365 status_t setSampleRate(uint32_t sampleRate); 366 367 /* Return current source sample rate in Hz */ 368 uint32_t getSampleRate() const; 369 370 /* Return the original source sample rate in Hz. This corresponds to the sample rate 371 * if playback rate had normal speed and pitch. 372 */ 373 uint32_t getOriginalSampleRate() const; 374 375 /* Set source playback rate for timestretch 376 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 377 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 378 * 379 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 380 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 381 * 382 * Speed increases the playback rate of media, but does not alter pitch. 383 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 384 */ 385 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 386 387 /* Return current playback rate */ 388 const AudioPlaybackRate& getPlaybackRate() const; 389 390 /* Enables looping and sets the start and end points of looping. 391 * Only supported for static buffer mode. 392 * 393 * Parameters: 394 * 395 * loopStart: loop start in frames relative to start of buffer. 396 * loopEnd: loop end in frames relative to start of buffer. 397 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 398 * pending or active loop. loopCount == -1 means infinite looping. 399 * 400 * For proper operation the following condition must be respected: 401 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 402 * 403 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 404 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 405 * 406 */ 407 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 408 409 /* Sets marker position. When playback reaches the number of frames specified, a callback with 410 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 411 * notification callback. To set a marker at a position which would compute as 0, 412 * a workaround is to set the marker at a nearby position such as ~0 or 1. 413 * If the AudioTrack has been opened with no callback function associated, the operation will 414 * fail. 415 * 416 * Parameters: 417 * 418 * marker: marker position expressed in wrapping (overflow) frame units, 419 * like the return value of getPosition(). 420 * 421 * Returned status (from utils/Errors.h) can be: 422 * - NO_ERROR: successful operation 423 * - INVALID_OPERATION: the AudioTrack has no callback installed. 424 */ 425 status_t setMarkerPosition(uint32_t marker); 426 status_t getMarkerPosition(uint32_t *marker) const; 427 428 /* Sets position update period. Every time the number of frames specified has been played, 429 * a callback with event type EVENT_NEW_POS is called. 430 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 431 * callback. 432 * If the AudioTrack has been opened with no callback function associated, the operation will 433 * fail. 434 * Extremely small values may be rounded up to a value the implementation can support. 435 * 436 * Parameters: 437 * 438 * updatePeriod: position update notification period expressed in frames. 439 * 440 * Returned status (from utils/Errors.h) can be: 441 * - NO_ERROR: successful operation 442 * - INVALID_OPERATION: the AudioTrack has no callback installed. 443 */ 444 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 445 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 446 447 /* Sets playback head position. 448 * Only supported for static buffer mode. 449 * 450 * Parameters: 451 * 452 * position: New playback head position in frames relative to start of buffer. 453 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 454 * but will result in an immediate underrun if started. 455 * 456 * Returned status (from utils/Errors.h) can be: 457 * - NO_ERROR: successful operation 458 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 459 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 460 * buffer 461 */ 462 status_t setPosition(uint32_t position); 463 464 /* Return the total number of frames played since playback start. 465 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 466 * It is reset to zero by flush(), reload(), and stop(). 467 * 468 * Parameters: 469 * 470 * position: Address where to return play head position. 471 * 472 * Returned status (from utils/Errors.h) can be: 473 * - NO_ERROR: successful operation 474 * - BAD_VALUE: position is NULL 475 */ 476 status_t getPosition(uint32_t *position); 477 478 /* For static buffer mode only, this returns the current playback position in frames 479 * relative to start of buffer. It is analogous to the position units used by 480 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 481 */ 482 status_t getBufferPosition(uint32_t *position); 483 484 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 485 * rewriting the buffer before restarting playback after a stop. 486 * This method must be called with the AudioTrack in paused or stopped state. 487 * Not allowed in streaming mode. 488 * 489 * Returned status (from utils/Errors.h) can be: 490 * - NO_ERROR: successful operation 491 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 492 */ 493 status_t reload(); 494 495 /* Returns a handle on the audio output used by this AudioTrack. 496 * 497 * Parameters: 498 * none. 499 * 500 * Returned value: 501 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 502 * track needed to be re-created but that failed 503 */ 504private: 505 audio_io_handle_t getOutput() const; 506public: 507 508 /* Selects the audio device to use for output of this AudioTrack. A value of 509 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 510 * 511 * Parameters: 512 * The device ID of the selected device (as returned by the AudioDevicesManager API). 513 * 514 * Returned value: 515 * - NO_ERROR: successful operation 516 * TODO: what else can happen here? 517 */ 518 status_t setOutputDevice(audio_port_handle_t deviceId); 519 520 /* Returns the ID of the audio device selected for this AudioTrack. 521 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 522 * 523 * Parameters: 524 * none. 525 */ 526 audio_port_handle_t getOutputDevice(); 527 528 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 529 * attached. 530 * A value of AUDIO_PORT_HANDLE_NONE indicates the audio track is not attached to any output. 531 * 532 * Parameters: 533 * none. 534 */ 535 audio_port_handle_t getRoutedDeviceId(); 536 537 /* Returns the unique session ID associated with this track. 538 * 539 * Parameters: 540 * none. 541 * 542 * Returned value: 543 * AudioTrack session ID. 544 */ 545 int getSessionId() const { return mSessionId; } 546 547 /* Attach track auxiliary output to specified effect. Use effectId = 0 548 * to detach track from effect. 549 * 550 * Parameters: 551 * 552 * effectId: effectId obtained from AudioEffect::id(). 553 * 554 * Returned status (from utils/Errors.h) can be: 555 * - NO_ERROR: successful operation 556 * - INVALID_OPERATION: the effect is not an auxiliary effect. 557 * - BAD_VALUE: The specified effect ID is invalid 558 */ 559 status_t attachAuxEffect(int effectId); 560 561 /* Public API for TRANSFER_OBTAIN mode. 562 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 563 * After filling these slots with data, the caller should release them with releaseBuffer(). 564 * If the track buffer is not full, obtainBuffer() returns as many contiguous 565 * [empty slots for] frames as are available immediately. 566 * 567 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 568 * additional non-contiguous frames that are predicted to be available immediately, 569 * if the client were to release the first frames and then call obtainBuffer() again. 570 * This value is only a prediction, and needs to be confirmed. 571 * It will be set to zero for an error return. 572 * 573 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 574 * regardless of the value of waitCount. 575 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 576 * maximum timeout based on waitCount; see chart below. 577 * Buffers will be returned until the pool 578 * is exhausted, at which point obtainBuffer() will either block 579 * or return WOULD_BLOCK depending on the value of the "waitCount" 580 * parameter. 581 * 582 * Interpretation of waitCount: 583 * +n limits wait time to n * WAIT_PERIOD_MS, 584 * -1 causes an (almost) infinite wait time, 585 * 0 non-blocking. 586 * 587 * Buffer fields 588 * On entry: 589 * frameCount number of [empty slots for] frames requested 590 * size ignored 591 * raw ignored 592 * After error return: 593 * frameCount 0 594 * size 0 595 * raw undefined 596 * After successful return: 597 * frameCount actual number of [empty slots for] frames available, <= number requested 598 * size actual number of bytes available 599 * raw pointer to the buffer 600 */ 601 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 602 size_t *nonContig = NULL); 603 604private: 605 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 606 * additional non-contiguous frames that are predicted to be available immediately, 607 * if the client were to release the first frames and then call obtainBuffer() again. 608 * This value is only a prediction, and needs to be confirmed. 609 * It will be set to zero for an error return. 610 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 611 * in case the requested amount of frames is in two or more non-contiguous regions. 612 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 613 */ 614 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 615 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 616public: 617 618 /* Public API for TRANSFER_OBTAIN mode. 619 * Release a filled buffer of frames for AudioFlinger to process. 620 * 621 * Buffer fields: 622 * frameCount currently ignored but recommend to set to actual number of frames filled 623 * size actual number of bytes filled, must be multiple of frameSize 624 * raw ignored 625 */ 626 void releaseBuffer(const Buffer* audioBuffer); 627 628 /* As a convenience we provide a write() interface to the audio buffer. 629 * Input parameter 'size' is in byte units. 630 * This is implemented on top of obtainBuffer/releaseBuffer. For best 631 * performance use callbacks. Returns actual number of bytes written >= 0, 632 * or one of the following negative status codes: 633 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 634 * BAD_VALUE size is invalid 635 * WOULD_BLOCK when obtainBuffer() returns same, or 636 * AudioTrack was stopped during the write 637 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 638 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 639 * false for the method to return immediately without waiting to try multiple times to write 640 * the full content of the buffer. 641 */ 642 ssize_t write(const void* buffer, size_t size, bool blocking = true); 643 644 /* 645 * Dumps the state of an audio track. 646 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 647 */ 648 status_t dump(int fd, const Vector<String16>& args) const; 649 650 /* 651 * Return the total number of frames which AudioFlinger desired but were unavailable, 652 * and thus which resulted in an underrun. Reset to zero by stop(). 653 */ 654 uint32_t getUnderrunFrames() const; 655 656 /* Get the flags */ 657 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 658 659 /* Set parameters - only possible when using direct output */ 660 status_t setParameters(const String8& keyValuePairs); 661 662 /* Get parameters */ 663 String8 getParameters(const String8& keys); 664 665 /* Poll for a timestamp on demand. 666 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 667 * or if you need to get the most recent timestamp outside of the event callback handler. 668 * Caution: calling this method too often may be inefficient; 669 * if you need a high resolution mapping between frame position and presentation time, 670 * consider implementing that at application level, based on the low resolution timestamps. 671 * Returns NO_ERROR if timestamp is valid. 672 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 673 * start/ACTIVE, when the number of frames consumed is less than the 674 * overall hardware latency to physical output. In WOULD_BLOCK cases, 675 * one might poll again, or use getPosition(), or use 0 position and 676 * current time for the timestamp. 677 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 678 * 679 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 680 */ 681 status_t getTimestamp(AudioTimestamp& timestamp); 682 683 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 684 * AudioTrack is routed is updated. 685 * Replaces any previously installed callback. 686 * Parameters: 687 * callback: The callback interface 688 * Returns NO_ERROR if successful. 689 * INVALID_OPERATION if the same callback is already installed. 690 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 691 * BAD_VALUE if the callback is NULL 692 */ 693 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 694 695 /* remove an AudioDeviceCallback. 696 * Parameters: 697 * callback: The callback interface 698 * Returns NO_ERROR if successful. 699 * INVALID_OPERATION if the callback is not installed 700 * BAD_VALUE if the callback is NULL 701 */ 702 status_t removeAudioDeviceCallback( 703 const sp<AudioSystem::AudioDeviceCallback>& callback); 704 705protected: 706 /* copying audio tracks is not allowed */ 707 AudioTrack(const AudioTrack& other); 708 AudioTrack& operator = (const AudioTrack& other); 709 710 /* a small internal class to handle the callback */ 711 class AudioTrackThread : public Thread 712 { 713 public: 714 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 715 716 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 717 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 718 virtual void requestExit(); 719 720 void pause(); // suspend thread from execution at next loop boundary 721 void resume(); // allow thread to execute, if not requested to exit 722 void wake(); // wake to handle changed notification conditions. 723 724 private: 725 void pauseInternal(nsecs_t ns = 0LL); 726 // like pause(), but only used internally within thread 727 728 friend class AudioTrack; 729 virtual bool threadLoop(); 730 AudioTrack& mReceiver; 731 virtual ~AudioTrackThread(); 732 Mutex mMyLock; // Thread::mLock is private 733 Condition mMyCond; // Thread::mThreadExitedCondition is private 734 bool mPaused; // whether thread is requested to pause at next loop entry 735 bool mPausedInt; // whether thread internally requests pause 736 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 737 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 738 // to processAudioBuffer() as state may have changed 739 // since pause time calculated. 740 }; 741 742 // body of AudioTrackThread::threadLoop() 743 // returns the maximum amount of time before we would like to run again, where: 744 // 0 immediately 745 // > 0 no later than this many nanoseconds from now 746 // NS_WHENEVER still active but no particular deadline 747 // NS_INACTIVE inactive so don't run again until re-started 748 // NS_NEVER never again 749 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 750 nsecs_t processAudioBuffer(); 751 752 // caller must hold lock on mLock for all _l methods 753 754 status_t createTrack_l(); 755 756 // can only be called when mState != STATE_ACTIVE 757 void flush_l(); 758 759 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 760 761 // FIXME enum is faster than strcmp() for parameter 'from' 762 status_t restoreTrack_l(const char *from); 763 764 bool isOffloaded() const; 765 bool isDirect() const; 766 bool isOffloadedOrDirect() const; 767 768 bool isOffloaded_l() const 769 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 770 771 bool isOffloadedOrDirect_l() const 772 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 773 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 774 775 bool isDirect_l() const 776 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 777 778 // increment mPosition by the delta of mServer, and return new value of mPosition 779 uint32_t updateAndGetPosition_l(); 780 781 // check sample rate and speed is compatible with AudioTrack 782 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; 783 784 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 785 sp<IAudioTrack> mAudioTrack; 786 sp<IMemory> mCblkMemory; 787 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 788 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 789 790 sp<AudioTrackThread> mAudioTrackThread; 791 792 float mVolume[2]; 793 float mSendLevel; 794 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 795 uint32_t mOriginalSampleRate; 796 AudioPlaybackRate mPlaybackRate; 797 size_t mFrameCount; // corresponds to current IAudioTrack, value is 798 // reported back by AudioFlinger to the client 799 size_t mReqFrameCount; // frame count to request the first or next time 800 // a new IAudioTrack is needed, non-decreasing 801 802 // constant after constructor or set() 803 audio_format_t mFormat; // as requested by client, not forced to 16-bit 804 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 805 // this AudioTrack has valid attributes 806 uint32_t mChannelCount; 807 audio_channel_mask_t mChannelMask; 808 sp<IMemory> mSharedBuffer; 809 transfer_type mTransfer; 810 audio_offload_info_t mOffloadInfoCopy; 811 const audio_offload_info_t* mOffloadInfo; 812 audio_attributes_t mAttributes; 813 814 size_t mFrameSize; // frame size in bytes 815 816 status_t mStatus; 817 818 // can change dynamically when IAudioTrack invalidated 819 uint32_t mLatency; // in ms 820 821 // Indicates the current track state. Protected by mLock. 822 enum State { 823 STATE_ACTIVE, 824 STATE_STOPPED, 825 STATE_PAUSED, 826 STATE_PAUSED_STOPPING, 827 STATE_FLUSHED, 828 STATE_STOPPING, 829 } mState; 830 831 // for client callback handler 832 callback_t mCbf; // callback handler for events, or NULL 833 void* mUserData; 834 835 // for notification APIs 836 uint32_t mNotificationFramesReq; // requested number of frames between each 837 // notification callback, 838 // at initial source sample rate 839 uint32_t mNotificationFramesAct; // actual number of frames between each 840 // notification callback, 841 // at initial source sample rate 842 bool mRefreshRemaining; // processAudioBuffer() should refresh 843 // mRemainingFrames and mRetryOnPartialBuffer 844 845 // used for static track cbf and restoration 846 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 847 uint32_t mLoopStart; // last setLoop loopStart 848 uint32_t mLoopEnd; // last setLoop loopEnd 849 int32_t mLoopCountNotified; // the last loopCount notified by callback. 850 // mLoopCountNotified counts down, matching 851 // the remaining loop count for static track 852 // playback. 853 854 // These are private to processAudioBuffer(), and are not protected by a lock 855 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 856 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 857 uint32_t mObservedSequence; // last observed value of mSequence 858 859 uint32_t mMarkerPosition; // in wrapping (overflow) frame units 860 bool mMarkerReached; 861 uint32_t mNewPosition; // in frames 862 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 863 864 uint32_t mServer; // in frames, last known mProxy->getPosition() 865 // which is count of frames consumed by server, 866 // reset by new IAudioTrack, 867 // whether it is reset by stop() is TBD 868 uint32_t mPosition; // in frames, like mServer except continues 869 // monotonically after new IAudioTrack, 870 // and could be easily widened to uint64_t 871 uint32_t mReleased; // in frames, count of frames released to server 872 // but not necessarily consumed by server, 873 // reset by stop() but continues monotonically 874 // after new IAudioTrack to restore mPosition, 875 // and could be easily widened to uint64_t 876 int64_t mStartUs; // the start time after flush or stop. 877 // only used for offloaded and direct tracks. 878 879 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 880 bool mRetrogradeMotionReported; // reduce log spam 881 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 882 883 audio_output_flags_t mFlags; 884 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 885 // mLock must be held to read or write those bits reliably. 886 887 bool mDoNotReconnect; 888 889 int mSessionId; 890 int mAuxEffectId; 891 892 mutable Mutex mLock; 893 894 bool mIsTimed; 895 int mPreviousPriority; // before start() 896 SchedPolicy mPreviousSchedulingGroup; 897 bool mAwaitBoost; // thread should wait for priority boost before running 898 899 // The proxy should only be referenced while a lock is held because the proxy isn't 900 // multi-thread safe, especially the SingleStateQueue part of the proxy. 901 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 902 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 903 // them around in case they are replaced during the obtainBuffer(). 904 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 905 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 906 907 bool mInUnderrun; // whether track is currently in underrun state 908 uint32_t mPausedPosition; 909 910 // For Device Selection API 911 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 912 audio_port_handle_t mSelectedDeviceId; 913 914private: 915 class DeathNotifier : public IBinder::DeathRecipient { 916 public: 917 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 918 protected: 919 virtual void binderDied(const wp<IBinder>& who); 920 private: 921 const wp<AudioTrack> mAudioTrack; 922 }; 923 924 sp<DeathNotifier> mDeathNotifier; 925 uint32_t mSequence; // incremented for each new IAudioTrack attempt 926 int mClientUid; 927 pid_t mClientPid; 928 929 sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 930}; 931 932class TimedAudioTrack : public AudioTrack 933{ 934public: 935 TimedAudioTrack(); 936 937 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 938 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 939 940 /* queue a buffer obtained via allocateTimedBuffer for playback at the 941 given timestamp. PTS units are microseconds on the media time timeline. 942 The media time transform (set with setMediaTimeTransform) set by the 943 audio producer will handle converting from media time to local time 944 (perhaps going through the common time timeline in the case of 945 synchronized multiroom audio case) */ 946 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 947 948 /* define a transform between media time and either common time or 949 local time */ 950 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 951 status_t setMediaTimeTransform(const LinearTransform& xform, 952 TargetTimeline target); 953}; 954 955}; // namespace android 956 957#endif // ANDROID_AUDIOTRACK_H 958