AudioTrack.h revision 33ff89ba94a527e4293ee5349da01483252d5c83
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 <media/Modulo.h> 26#include <utils/threads.h> 27 28namespace android { 29 30// ---------------------------------------------------------------------------- 31 32struct audio_track_cblk_t; 33class AudioTrackClientProxy; 34class StaticAudioTrackClientProxy; 35 36// ---------------------------------------------------------------------------- 37 38class AudioTrack : public RefBase 39{ 40public: 41 42 /* Events used by AudioTrack callback function (callback_t). 43 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 44 */ 45 enum event_type { 46 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 47 // This event only occurs for TRANSFER_CALLBACK. 48 // If this event is delivered but the callback handler 49 // does not want to write more data, the handler must 50 // ignore the event by setting frameCount to zero. 51 // This might occur, for example, if the application is 52 // waiting for source data or is at the end of stream. 53 // 54 // For data filling, it is preferred that the callback 55 // does not block and instead returns a short count on 56 // the amount of data actually delivered 57 // (or 0, if no data is currently available). 58 EVENT_UNDERRUN = 1, // Buffer underrun occurred. This will not occur for 59 // static tracks. 60 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 61 // loop start if loop count was not 0 for a static track. 62 EVENT_MARKER = 3, // Playback head is at the specified marker position 63 // (See setMarkerPosition()). 64 EVENT_NEW_POS = 4, // Playback head is at a new position 65 // (See setPositionUpdatePeriod()). 66 EVENT_BUFFER_END = 5, // Playback has completed for a static track. 67 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 68 // voluntary invalidation by mediaserver, or mediaserver crash. 69 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 70 // back (after stop is called) for an offloaded track. 71#if 0 // FIXME not yet implemented 72 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 73 // in the mapping from frame position to presentation time. 74 // See AudioTimestamp for the information included with event. 75#endif 76 }; 77 78 /* Client should declare a Buffer and pass the address to obtainBuffer() 79 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 80 */ 81 82 class Buffer 83 { 84 public: 85 // FIXME use m prefix 86 size_t frameCount; // number of sample frames corresponding to size; 87 // on input to obtainBuffer() it is the number of frames desired, 88 // on output from obtainBuffer() it is the number of available 89 // [empty slots for] frames to be filled 90 // on input to releaseBuffer() it is currently ignored 91 92 size_t size; // input/output in bytes == frameCount * frameSize 93 // on input to obtainBuffer() it is ignored 94 // on output from obtainBuffer() it is the number of available 95 // [empty slots for] bytes to be filled, 96 // which is frameCount * frameSize 97 // on input to releaseBuffer() it is the number of bytes to 98 // release 99 // FIXME This is redundant with respect to frameCount. Consider 100 // removing size and making frameCount the primary field. 101 102 union { 103 void* raw; 104 short* i16; // signed 16-bit 105 int8_t* i8; // unsigned 8-bit, offset by 0x80 106 }; // input to obtainBuffer(): unused, output: pointer to buffer 107 }; 108 109 /* As a convenience, if a callback is supplied, a handler thread 110 * is automatically created with the appropriate priority. This thread 111 * invokes the callback when a new buffer becomes available or various conditions occur. 112 * Parameters: 113 * 114 * event: type of event notified (see enum AudioTrack::event_type). 115 * user: Pointer to context for use by the callback receiver. 116 * info: Pointer to optional parameter according to event type: 117 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 118 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 119 * written. 120 * - EVENT_UNDERRUN: unused. 121 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 122 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 123 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 124 * - EVENT_BUFFER_END: unused. 125 * - EVENT_NEW_IAUDIOTRACK: unused. 126 * - EVENT_STREAM_END: unused. 127 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 128 */ 129 130 typedef void (*callback_t)(int event, void* user, void *info); 131 132 /* Returns the minimum frame count required for the successful creation of 133 * an AudioTrack object. 134 * Returned status (from utils/Errors.h) can be: 135 * - NO_ERROR: successful operation 136 * - NO_INIT: audio server or audio hardware not initialized 137 * - BAD_VALUE: unsupported configuration 138 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 139 * and is undefined otherwise. 140 * FIXME This API assumes a route, and so should be deprecated. 141 */ 142 143 static status_t getMinFrameCount(size_t* frameCount, 144 audio_stream_type_t streamType, 145 uint32_t sampleRate); 146 147 /* How data is transferred to AudioTrack 148 */ 149 enum transfer_type { 150 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 151 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 152 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 153 TRANSFER_SYNC, // synchronous write() 154 TRANSFER_SHARED, // shared memory 155 }; 156 157 /* Constructs an uninitialized AudioTrack. No connection with 158 * AudioFlinger takes place. Use set() after this. 159 */ 160 AudioTrack(); 161 162 /* Creates an AudioTrack object and registers it with AudioFlinger. 163 * Once created, the track needs to be started before it can be used. 164 * Unspecified values are set to appropriate default values. 165 * 166 * Parameters: 167 * 168 * streamType: Select the type of audio stream this track is attached to 169 * (e.g. AUDIO_STREAM_MUSIC). 170 * sampleRate: Data source sampling rate in Hz. 171 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 172 * For direct and offloaded tracks, the possible format(s) depends on the 173 * output sink. 174 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 175 * frameCount: Minimum size of track PCM buffer in frames. This defines the 176 * application's contribution to the 177 * latency of the track. The actual size selected by the AudioTrack could be 178 * larger if the requested size is not compatible with current audio HAL 179 * configuration. Zero means to use a default value. 180 * flags: See comments on audio_output_flags_t in <system/audio.h>. 181 * cbf: Callback function. If not null, this function is called periodically 182 * to provide new data in TRANSFER_CALLBACK mode 183 * and inform of marker, position updates, etc. 184 * user: Context for use by the callback receiver. 185 * notificationFrames: The callback function is called each time notificationFrames PCM 186 * frames have been consumed from track input buffer. 187 * This is expressed in units of frames at the initial source sample rate. 188 * sessionId: Specific session ID, or zero to use default. 189 * transferType: How data is transferred to AudioTrack. 190 * offloadInfo: If not NULL, provides offload parameters for 191 * AudioSystem::getOutputForAttr(). 192 * uid: User ID of the app which initially requested this AudioTrack 193 * for power management tracking, or -1 for current user ID. 194 * pid: Process ID of the app which initially requested this AudioTrack 195 * for power management tracking, or -1 for current process ID. 196 * pAttributes: If not NULL, supersedes streamType for use case selection. 197 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 198 binder to AudioFlinger. 199 It will return an error instead. The application will recreate 200 the track based on offloading or different channel configuration, etc. 201 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 202 */ 203 204 AudioTrack( audio_stream_type_t streamType, 205 uint32_t sampleRate, 206 audio_format_t format, 207 audio_channel_mask_t channelMask, 208 size_t frameCount = 0, 209 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 210 callback_t cbf = NULL, 211 void* user = NULL, 212 uint32_t notificationFrames = 0, 213 int sessionId = AUDIO_SESSION_ALLOCATE, 214 transfer_type transferType = TRANSFER_DEFAULT, 215 const audio_offload_info_t *offloadInfo = NULL, 216 int uid = -1, 217 pid_t pid = -1, 218 const audio_attributes_t* pAttributes = NULL, 219 bool doNotReconnect = false); 220 221 /* Creates an audio track and registers it with AudioFlinger. 222 * With this constructor, the track is configured for static buffer mode. 223 * Data to be rendered is passed in a shared memory buffer 224 * identified by the argument sharedBuffer, which should be non-0. 225 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 226 * but without the ability to specify a non-zero value for the frameCount parameter. 227 * The memory should be initialized to the desired data before calling start(). 228 * The write() method is not supported in this case. 229 * It is recommended to pass a callback function to be notified of playback end by an 230 * EVENT_UNDERRUN event. 231 */ 232 233 AudioTrack( audio_stream_type_t streamType, 234 uint32_t sampleRate, 235 audio_format_t format, 236 audio_channel_mask_t channelMask, 237 const sp<IMemory>& sharedBuffer, 238 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 239 callback_t cbf = NULL, 240 void* user = NULL, 241 uint32_t notificationFrames = 0, 242 int sessionId = AUDIO_SESSION_ALLOCATE, 243 transfer_type transferType = TRANSFER_DEFAULT, 244 const audio_offload_info_t *offloadInfo = NULL, 245 int uid = -1, 246 pid_t pid = -1, 247 const audio_attributes_t* pAttributes = NULL, 248 bool doNotReconnect = false); 249 250 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 251 * Also destroys all resources associated with the AudioTrack. 252 */ 253protected: 254 virtual ~AudioTrack(); 255public: 256 257 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 258 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 259 * set() is not multi-thread safe. 260 * Returned status (from utils/Errors.h) can be: 261 * - NO_ERROR: successful initialization 262 * - INVALID_OPERATION: AudioTrack is already initialized 263 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 264 * - NO_INIT: audio server or audio hardware not initialized 265 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 266 * If sharedBuffer is non-0, the frameCount parameter is ignored and 267 * replaced by the shared buffer's total allocated size in frame units. 268 * 269 * Parameters not listed in the AudioTrack constructors above: 270 * 271 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 272 * 273 * Internal state post condition: 274 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 275 */ 276 status_t set(audio_stream_type_t streamType, 277 uint32_t sampleRate, 278 audio_format_t format, 279 audio_channel_mask_t channelMask, 280 size_t frameCount = 0, 281 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 282 callback_t cbf = NULL, 283 void* user = NULL, 284 uint32_t notificationFrames = 0, 285 const sp<IMemory>& sharedBuffer = 0, 286 bool threadCanCallJava = false, 287 int sessionId = AUDIO_SESSION_ALLOCATE, 288 transfer_type transferType = TRANSFER_DEFAULT, 289 const audio_offload_info_t *offloadInfo = NULL, 290 int uid = -1, 291 pid_t pid = -1, 292 const audio_attributes_t* pAttributes = NULL, 293 bool doNotReconnect = false); 294 295 /* Result of constructing the AudioTrack. This must be checked for successful initialization 296 * before using any AudioTrack API (except for set()), because using 297 * an uninitialized AudioTrack produces undefined results. 298 * See set() method above for possible return codes. 299 */ 300 status_t initCheck() const { return mStatus; } 301 302 /* Returns this track's estimated latency in milliseconds. 303 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 304 * and audio hardware driver. 305 */ 306 uint32_t latency() const { return mLatency; } 307 308 /* getters, see constructors and set() */ 309 310 audio_stream_type_t streamType() const; 311 audio_format_t format() const { return mFormat; } 312 313 /* Return frame size in bytes, which for linear PCM is 314 * channelCount * (bit depth per channel / 8). 315 * channelCount is determined from channelMask, and bit depth comes from format. 316 * For non-linear formats, the frame size is typically 1 byte. 317 */ 318 size_t frameSize() const { return mFrameSize; } 319 320 uint32_t channelCount() const { return mChannelCount; } 321 size_t frameCount() const { return mFrameCount; } 322 323 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 324 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 325 326 /* After it's created the track is not active. Call start() to 327 * make it active. If set, the callback will start being called. 328 * If the track was previously paused, volume is ramped up over the first mix buffer. 329 */ 330 status_t start(); 331 332 /* Stop a track. 333 * In static buffer mode, the track is stopped immediately. 334 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 335 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 336 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 337 * is first drained, mixed, and output, and only then is the track marked as stopped. 338 */ 339 void stop(); 340 bool stopped() const; 341 342 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 343 * This has the effect of draining the buffers without mixing or output. 344 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 345 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 346 */ 347 void flush(); 348 349 /* Pause a track. After pause, the callback will cease being called and 350 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 351 * and will fill up buffers until the pool is exhausted. 352 * Volume is ramped down over the next mix buffer following the pause request, 353 * and then the track is marked as paused. It can be resumed with ramp up by start(). 354 */ 355 void pause(); 356 357 /* Set volume for this track, mostly used for games' sound effects 358 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 359 * This is the older API. New applications should use setVolume(float) when possible. 360 */ 361 status_t setVolume(float left, float right); 362 363 /* Set volume for all channels. This is the preferred API for new applications, 364 * especially for multi-channel content. 365 */ 366 status_t setVolume(float volume); 367 368 /* Set the send level for this track. An auxiliary effect should be attached 369 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 370 */ 371 status_t setAuxEffectSendLevel(float level); 372 void getAuxEffectSendLevel(float* level) const; 373 374 /* Set source sample rate for this track in Hz, mostly used for games' sound effects 375 */ 376 status_t setSampleRate(uint32_t sampleRate); 377 378 /* Return current source sample rate in Hz */ 379 uint32_t getSampleRate() const; 380 381 /* Return the original source sample rate in Hz. This corresponds to the sample rate 382 * if playback rate had normal speed and pitch. 383 */ 384 uint32_t getOriginalSampleRate() const; 385 386 /* Set source playback rate for timestretch 387 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 388 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 389 * 390 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 391 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 392 * 393 * Speed increases the playback rate of media, but does not alter pitch. 394 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 395 */ 396 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 397 398 /* Return current playback rate */ 399 const AudioPlaybackRate& getPlaybackRate() const; 400 401 /* Enables looping and sets the start and end points of looping. 402 * Only supported for static buffer mode. 403 * 404 * Parameters: 405 * 406 * loopStart: loop start in frames relative to start of buffer. 407 * loopEnd: loop end in frames relative to start of buffer. 408 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 409 * pending or active loop. loopCount == -1 means infinite looping. 410 * 411 * For proper operation the following condition must be respected: 412 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 413 * 414 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 415 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 416 * 417 */ 418 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 419 420 /* Sets marker position. When playback reaches the number of frames specified, a callback with 421 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 422 * notification callback. To set a marker at a position which would compute as 0, 423 * a workaround is to set the marker at a nearby position such as ~0 or 1. 424 * If the AudioTrack has been opened with no callback function associated, the operation will 425 * fail. 426 * 427 * Parameters: 428 * 429 * marker: marker position expressed in wrapping (overflow) frame units, 430 * like the return value of getPosition(). 431 * 432 * Returned status (from utils/Errors.h) can be: 433 * - NO_ERROR: successful operation 434 * - INVALID_OPERATION: the AudioTrack has no callback installed. 435 */ 436 status_t setMarkerPosition(uint32_t marker); 437 status_t getMarkerPosition(uint32_t *marker) const; 438 439 /* Sets position update period. Every time the number of frames specified has been played, 440 * a callback with event type EVENT_NEW_POS is called. 441 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 442 * callback. 443 * If the AudioTrack has been opened with no callback function associated, the operation will 444 * fail. 445 * Extremely small values may be rounded up to a value the implementation can support. 446 * 447 * Parameters: 448 * 449 * updatePeriod: position update notification period expressed in frames. 450 * 451 * Returned status (from utils/Errors.h) can be: 452 * - NO_ERROR: successful operation 453 * - INVALID_OPERATION: the AudioTrack has no callback installed. 454 */ 455 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 456 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 457 458 /* Sets playback head position. 459 * Only supported for static buffer mode. 460 * 461 * Parameters: 462 * 463 * position: New playback head position in frames relative to start of buffer. 464 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 465 * but will result in an immediate underrun if started. 466 * 467 * Returned status (from utils/Errors.h) can be: 468 * - NO_ERROR: successful operation 469 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 470 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 471 * buffer 472 */ 473 status_t setPosition(uint32_t position); 474 475 /* Return the total number of frames played since playback start. 476 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 477 * It is reset to zero by flush(), reload(), and stop(). 478 * 479 * Parameters: 480 * 481 * position: Address where to return play head position. 482 * 483 * Returned status (from utils/Errors.h) can be: 484 * - NO_ERROR: successful operation 485 * - BAD_VALUE: position is NULL 486 */ 487 status_t getPosition(uint32_t *position); 488 489 /* For static buffer mode only, this returns the current playback position in frames 490 * relative to start of buffer. It is analogous to the position units used by 491 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 492 */ 493 status_t getBufferPosition(uint32_t *position); 494 495 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 496 * rewriting the buffer before restarting playback after a stop. 497 * This method must be called with the AudioTrack in paused or stopped state. 498 * Not allowed in streaming mode. 499 * 500 * Returned status (from utils/Errors.h) can be: 501 * - NO_ERROR: successful operation 502 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 503 */ 504 status_t reload(); 505 506 /* Returns a handle on the audio output used by this AudioTrack. 507 * 508 * Parameters: 509 * none. 510 * 511 * Returned value: 512 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 513 * track needed to be re-created but that failed 514 */ 515private: 516 audio_io_handle_t getOutput() const; 517public: 518 519 /* Selects the audio device to use for output of this AudioTrack. A value of 520 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 521 * 522 * Parameters: 523 * The device ID of the selected device (as returned by the AudioDevicesManager API). 524 * 525 * Returned value: 526 * - NO_ERROR: successful operation 527 * TODO: what else can happen here? 528 */ 529 status_t setOutputDevice(audio_port_handle_t deviceId); 530 531 /* Returns the ID of the audio device selected for this AudioTrack. 532 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 533 * 534 * Parameters: 535 * none. 536 */ 537 audio_port_handle_t getOutputDevice(); 538 539 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 540 * attached. 541 * A value of AUDIO_PORT_HANDLE_NONE indicates the audio track is not attached to any output. 542 * 543 * Parameters: 544 * none. 545 */ 546 audio_port_handle_t getRoutedDeviceId(); 547 548 /* Returns the unique session ID associated with this track. 549 * 550 * Parameters: 551 * none. 552 * 553 * Returned value: 554 * AudioTrack session ID. 555 */ 556 int getSessionId() const { return mSessionId; } 557 558 /* Attach track auxiliary output to specified effect. Use effectId = 0 559 * to detach track from effect. 560 * 561 * Parameters: 562 * 563 * effectId: effectId obtained from AudioEffect::id(). 564 * 565 * Returned status (from utils/Errors.h) can be: 566 * - NO_ERROR: successful operation 567 * - INVALID_OPERATION: the effect is not an auxiliary effect. 568 * - BAD_VALUE: The specified effect ID is invalid 569 */ 570 status_t attachAuxEffect(int effectId); 571 572 /* Public API for TRANSFER_OBTAIN mode. 573 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 574 * After filling these slots with data, the caller should release them with releaseBuffer(). 575 * If the track buffer is not full, obtainBuffer() returns as many contiguous 576 * [empty slots for] frames as are available immediately. 577 * 578 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 579 * additional non-contiguous frames that are predicted to be available immediately, 580 * if the client were to release the first frames and then call obtainBuffer() again. 581 * This value is only a prediction, and needs to be confirmed. 582 * It will be set to zero for an error return. 583 * 584 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 585 * regardless of the value of waitCount. 586 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 587 * maximum timeout based on waitCount; see chart below. 588 * Buffers will be returned until the pool 589 * is exhausted, at which point obtainBuffer() will either block 590 * or return WOULD_BLOCK depending on the value of the "waitCount" 591 * parameter. 592 * 593 * Interpretation of waitCount: 594 * +n limits wait time to n * WAIT_PERIOD_MS, 595 * -1 causes an (almost) infinite wait time, 596 * 0 non-blocking. 597 * 598 * Buffer fields 599 * On entry: 600 * frameCount number of [empty slots for] frames requested 601 * size ignored 602 * raw ignored 603 * After error return: 604 * frameCount 0 605 * size 0 606 * raw undefined 607 * After successful return: 608 * frameCount actual number of [empty slots for] frames available, <= number requested 609 * size actual number of bytes available 610 * raw pointer to the buffer 611 */ 612 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 613 size_t *nonContig = NULL); 614 615private: 616 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 617 * additional non-contiguous frames that are predicted to be available immediately, 618 * if the client were to release the first frames and then call obtainBuffer() again. 619 * This value is only a prediction, and needs to be confirmed. 620 * It will be set to zero for an error return. 621 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 622 * in case the requested amount of frames is in two or more non-contiguous regions. 623 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 624 */ 625 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 626 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 627public: 628 629 /* Public API for TRANSFER_OBTAIN mode. 630 * Release a filled buffer of frames for AudioFlinger to process. 631 * 632 * Buffer fields: 633 * frameCount currently ignored but recommend to set to actual number of frames filled 634 * size actual number of bytes filled, must be multiple of frameSize 635 * raw ignored 636 */ 637 void releaseBuffer(const Buffer* audioBuffer); 638 639 /* As a convenience we provide a write() interface to the audio buffer. 640 * Input parameter 'size' is in byte units. 641 * This is implemented on top of obtainBuffer/releaseBuffer. For best 642 * performance use callbacks. Returns actual number of bytes written >= 0, 643 * or one of the following negative status codes: 644 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 645 * BAD_VALUE size is invalid 646 * WOULD_BLOCK when obtainBuffer() returns same, or 647 * AudioTrack was stopped during the write 648 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 649 * the track cannot be automatically restored. 650 * The application needs to recreate the AudioTrack 651 * because the audio device changed or AudioFlinger died. 652 * This typically occurs for direct or offload tracks 653 * or if mDoNotReconnect is true. 654 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 655 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 656 * false for the method to return immediately without waiting to try multiple times to write 657 * the full content of the buffer. 658 */ 659 ssize_t write(const void* buffer, size_t size, bool blocking = true); 660 661 /* 662 * Dumps the state of an audio track. 663 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 664 */ 665 status_t dump(int fd, const Vector<String16>& args) const; 666 667 /* 668 * Return the total number of frames which AudioFlinger desired but were unavailable, 669 * and thus which resulted in an underrun. Reset to zero by stop(). 670 */ 671 uint32_t getUnderrunFrames() const; 672 673 /* Get the flags */ 674 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 675 676 /* Set parameters - only possible when using direct output */ 677 status_t setParameters(const String8& keyValuePairs); 678 679 /* Get parameters */ 680 String8 getParameters(const String8& keys); 681 682 /* Poll for a timestamp on demand. 683 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 684 * or if you need to get the most recent timestamp outside of the event callback handler. 685 * Caution: calling this method too often may be inefficient; 686 * if you need a high resolution mapping between frame position and presentation time, 687 * consider implementing that at application level, based on the low resolution timestamps. 688 * Returns NO_ERROR if timestamp is valid. 689 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 690 * start/ACTIVE, when the number of frames consumed is less than the 691 * overall hardware latency to physical output. In WOULD_BLOCK cases, 692 * one might poll again, or use getPosition(), or use 0 position and 693 * current time for the timestamp. 694 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 695 * the track cannot be automatically restored. 696 * The application needs to recreate the AudioTrack 697 * because the audio device changed or AudioFlinger died. 698 * This typically occurs for direct or offload tracks 699 * or if mDoNotReconnect is true. 700 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 701 * 702 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 703 */ 704 status_t getTimestamp(AudioTimestamp& timestamp); 705 706 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 707 * AudioTrack is routed is updated. 708 * Replaces any previously installed callback. 709 * Parameters: 710 * callback: The callback interface 711 * Returns NO_ERROR if successful. 712 * INVALID_OPERATION if the same callback is already installed. 713 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 714 * BAD_VALUE if the callback is NULL 715 */ 716 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 717 718 /* remove an AudioDeviceCallback. 719 * Parameters: 720 * callback: The callback interface 721 * Returns NO_ERROR if successful. 722 * INVALID_OPERATION if the callback is not installed 723 * BAD_VALUE if the callback is NULL 724 */ 725 status_t removeAudioDeviceCallback( 726 const sp<AudioSystem::AudioDeviceCallback>& callback); 727 728protected: 729 /* copying audio tracks is not allowed */ 730 AudioTrack(const AudioTrack& other); 731 AudioTrack& operator = (const AudioTrack& other); 732 733 /* a small internal class to handle the callback */ 734 class AudioTrackThread : public Thread 735 { 736 public: 737 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 738 739 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 740 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 741 virtual void requestExit(); 742 743 void pause(); // suspend thread from execution at next loop boundary 744 void resume(); // allow thread to execute, if not requested to exit 745 void wake(); // wake to handle changed notification conditions. 746 747 private: 748 void pauseInternal(nsecs_t ns = 0LL); 749 // like pause(), but only used internally within thread 750 751 friend class AudioTrack; 752 virtual bool threadLoop(); 753 AudioTrack& mReceiver; 754 virtual ~AudioTrackThread(); 755 Mutex mMyLock; // Thread::mLock is private 756 Condition mMyCond; // Thread::mThreadExitedCondition is private 757 bool mPaused; // whether thread is requested to pause at next loop entry 758 bool mPausedInt; // whether thread internally requests pause 759 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 760 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 761 // to processAudioBuffer() as state may have changed 762 // since pause time calculated. 763 }; 764 765 // body of AudioTrackThread::threadLoop() 766 // returns the maximum amount of time before we would like to run again, where: 767 // 0 immediately 768 // > 0 no later than this many nanoseconds from now 769 // NS_WHENEVER still active but no particular deadline 770 // NS_INACTIVE inactive so don't run again until re-started 771 // NS_NEVER never again 772 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 773 nsecs_t processAudioBuffer(); 774 775 // caller must hold lock on mLock for all _l methods 776 777 status_t createTrack_l(); 778 779 // can only be called when mState != STATE_ACTIVE 780 void flush_l(); 781 782 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 783 784 // FIXME enum is faster than strcmp() for parameter 'from' 785 status_t restoreTrack_l(const char *from); 786 787 bool isOffloaded() const; 788 bool isDirect() const; 789 bool isOffloadedOrDirect() const; 790 791 bool isOffloaded_l() const 792 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 793 794 bool isOffloadedOrDirect_l() const 795 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 796 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 797 798 bool isDirect_l() const 799 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 800 801 // increment mPosition by the delta of mServer, and return new value of mPosition 802 Modulo<uint32_t> updateAndGetPosition_l(); 803 804 // check sample rate and speed is compatible with AudioTrack 805 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; 806 807 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 808 sp<IAudioTrack> mAudioTrack; 809 sp<IMemory> mCblkMemory; 810 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 811 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 812 813 sp<AudioTrackThread> mAudioTrackThread; 814 bool mThreadCanCallJava; 815 816 float mVolume[2]; 817 float mSendLevel; 818 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 819 uint32_t mOriginalSampleRate; 820 AudioPlaybackRate mPlaybackRate; 821 size_t mFrameCount; // corresponds to current IAudioTrack, value is 822 // reported back by AudioFlinger to the client 823 size_t mReqFrameCount; // frame count to request the first or next time 824 // a new IAudioTrack is needed, non-decreasing 825 826 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 827 // These values can be used for informational purposes until the track is invalidated, 828 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 829 uint32_t mAfLatency; // AudioFlinger latency in ms 830 size_t mAfFrameCount; // AudioFlinger frame count 831 uint32_t mAfSampleRate; // AudioFlinger sample rate 832 833 // constant after constructor or set() 834 audio_format_t mFormat; // as requested by client, not forced to 16-bit 835 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 836 // this AudioTrack has valid attributes 837 uint32_t mChannelCount; 838 audio_channel_mask_t mChannelMask; 839 sp<IMemory> mSharedBuffer; 840 transfer_type mTransfer; 841 audio_offload_info_t mOffloadInfoCopy; 842 const audio_offload_info_t* mOffloadInfo; 843 audio_attributes_t mAttributes; 844 845 size_t mFrameSize; // frame size in bytes 846 847 status_t mStatus; 848 849 // can change dynamically when IAudioTrack invalidated 850 uint32_t mLatency; // in ms 851 852 // Indicates the current track state. Protected by mLock. 853 enum State { 854 STATE_ACTIVE, 855 STATE_STOPPED, 856 STATE_PAUSED, 857 STATE_PAUSED_STOPPING, 858 STATE_FLUSHED, 859 STATE_STOPPING, 860 } mState; 861 862 // for client callback handler 863 callback_t mCbf; // callback handler for events, or NULL 864 void* mUserData; 865 866 // for notification APIs 867 uint32_t mNotificationFramesReq; // requested number of frames between each 868 // notification callback, 869 // at initial source sample rate 870 uint32_t mNotificationFramesAct; // actual number of frames between each 871 // notification callback, 872 // at initial source sample rate 873 bool mRefreshRemaining; // processAudioBuffer() should refresh 874 // mRemainingFrames and mRetryOnPartialBuffer 875 876 // used for static track cbf and restoration 877 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 878 uint32_t mLoopStart; // last setLoop loopStart 879 uint32_t mLoopEnd; // last setLoop loopEnd 880 int32_t mLoopCountNotified; // the last loopCount notified by callback. 881 // mLoopCountNotified counts down, matching 882 // the remaining loop count for static track 883 // playback. 884 885 // These are private to processAudioBuffer(), and are not protected by a lock 886 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 887 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 888 uint32_t mObservedSequence; // last observed value of mSequence 889 890 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 891 bool mMarkerReached; 892 Modulo<uint32_t> mNewPosition; // in frames 893 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 894 895 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 896 // which is count of frames consumed by server, 897 // reset by new IAudioTrack, 898 // whether it is reset by stop() is TBD 899 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 900 // monotonically after new IAudioTrack, 901 // and could be easily widened to uint64_t 902 Modulo<uint32_t> mReleased; // count of frames released to server 903 // but not necessarily consumed by server, 904 // reset by stop() but continues monotonically 905 // after new IAudioTrack to restore mPosition, 906 // and could be easily widened to uint64_t 907 int64_t mStartUs; // the start time after flush or stop. 908 // only used for offloaded and direct tracks. 909 910 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 911 bool mTimestampStartupGlitchReported; // reduce log spam 912 bool mRetrogradeMotionReported; // reduce log spam 913 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 914 915 audio_output_flags_t mFlags; 916 // const after set(), except for bits AUDIO_OUTPUT_FLAG_FAST and AUDIO_OUTPUT_FLAG_OFFLOAD. 917 // mLock must be held to read or write those bits reliably. 918 919 bool mDoNotReconnect; 920 921 int mSessionId; 922 int mAuxEffectId; 923 924 mutable Mutex mLock; 925 926 bool mIsTimed; 927 int mPreviousPriority; // before start() 928 SchedPolicy mPreviousSchedulingGroup; 929 bool mAwaitBoost; // thread should wait for priority boost before running 930 931 // The proxy should only be referenced while a lock is held because the proxy isn't 932 // multi-thread safe, especially the SingleStateQueue part of the proxy. 933 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 934 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 935 // them around in case they are replaced during the obtainBuffer(). 936 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 937 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 938 939 bool mInUnderrun; // whether track is currently in underrun state 940 uint32_t mPausedPosition; 941 942 // For Device Selection API 943 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 944 audio_port_handle_t mSelectedDeviceId; 945 946private: 947 class DeathNotifier : public IBinder::DeathRecipient { 948 public: 949 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 950 protected: 951 virtual void binderDied(const wp<IBinder>& who); 952 private: 953 const wp<AudioTrack> mAudioTrack; 954 }; 955 956 sp<DeathNotifier> mDeathNotifier; 957 uint32_t mSequence; // incremented for each new IAudioTrack attempt 958 int mClientUid; 959 pid_t mClientPid; 960 961 sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 962}; 963 964class TimedAudioTrack : public AudioTrack 965{ 966public: 967 TimedAudioTrack(); 968 969 /* allocate a shared memory buffer that can be passed to queueTimedBuffer */ 970 status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer); 971 972 /* queue a buffer obtained via allocateTimedBuffer for playback at the 973 given timestamp. PTS units are microseconds on the media time timeline. 974 The media time transform (set with setMediaTimeTransform) set by the 975 audio producer will handle converting from media time to local time 976 (perhaps going through the common time timeline in the case of 977 synchronized multiroom audio case) */ 978 status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts); 979 980 /* define a transform between media time and either common time or 981 local time */ 982 enum TargetTimeline {LOCAL_TIME, COMMON_TIME}; 983 status_t setMediaTimeTransform(const LinearTransform& xform, 984 TargetTimeline target); 985}; 986 987}; // namespace android 988 989#endif // ANDROID_AUDIOTRACK_H 990