AudioTrack.h revision d848eb48c121c119e8ba7583efc75415fe102570
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. Zero means to use the sink sample rate. 171 * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set. 172 * 0 will not work with current policy implementation for direct output 173 * selection where an exact match is needed for sampling rate. 174 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 175 * For direct and offloaded tracks, the possible format(s) depends on the 176 * output sink. 177 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 178 * frameCount: Minimum size of track PCM buffer in frames. This defines the 179 * application's contribution to the 180 * latency of the track. The actual size selected by the AudioTrack could be 181 * larger if the requested size is not compatible with current audio HAL 182 * configuration. Zero means to use a default value. 183 * flags: See comments on audio_output_flags_t in <system/audio.h>. 184 * cbf: Callback function. If not null, this function is called periodically 185 * to provide new data in TRANSFER_CALLBACK mode 186 * and inform of marker, position updates, etc. 187 * user: Context for use by the callback receiver. 188 * notificationFrames: The callback function is called each time notificationFrames PCM 189 * frames have been consumed from track input buffer. 190 * This is expressed in units of frames at the initial source sample rate. 191 * sessionId: Specific session ID, or zero to use default. 192 * transferType: How data is transferred to AudioTrack. 193 * offloadInfo: If not NULL, provides offload parameters for 194 * AudioSystem::getOutputForAttr(). 195 * uid: User ID of the app which initially requested this AudioTrack 196 * for power management tracking, or -1 for current user ID. 197 * pid: Process ID of the app which initially requested this AudioTrack 198 * for power management tracking, or -1 for current process ID. 199 * pAttributes: If not NULL, supersedes streamType for use case selection. 200 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 201 binder to AudioFlinger. 202 It will return an error instead. The application will recreate 203 the track based on offloading or different channel configuration, etc. 204 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 205 */ 206 207 AudioTrack( audio_stream_type_t streamType, 208 uint32_t sampleRate, 209 audio_format_t format, 210 audio_channel_mask_t channelMask, 211 size_t frameCount = 0, 212 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 213 callback_t cbf = NULL, 214 void* user = NULL, 215 uint32_t notificationFrames = 0, 216 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 217 transfer_type transferType = TRANSFER_DEFAULT, 218 const audio_offload_info_t *offloadInfo = NULL, 219 int uid = -1, 220 pid_t pid = -1, 221 const audio_attributes_t* pAttributes = NULL, 222 bool doNotReconnect = false); 223 224 /* Creates an audio track and registers it with AudioFlinger. 225 * With this constructor, the track is configured for static buffer mode. 226 * Data to be rendered is passed in a shared memory buffer 227 * identified by the argument sharedBuffer, which should be non-0. 228 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 229 * but without the ability to specify a non-zero value for the frameCount parameter. 230 * The memory should be initialized to the desired data before calling start(). 231 * The write() method is not supported in this case. 232 * It is recommended to pass a callback function to be notified of playback end by an 233 * EVENT_UNDERRUN event. 234 */ 235 236 AudioTrack( audio_stream_type_t streamType, 237 uint32_t sampleRate, 238 audio_format_t format, 239 audio_channel_mask_t channelMask, 240 const sp<IMemory>& sharedBuffer, 241 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 242 callback_t cbf = NULL, 243 void* user = NULL, 244 uint32_t notificationFrames = 0, 245 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 246 transfer_type transferType = TRANSFER_DEFAULT, 247 const audio_offload_info_t *offloadInfo = NULL, 248 int uid = -1, 249 pid_t pid = -1, 250 const audio_attributes_t* pAttributes = NULL, 251 bool doNotReconnect = false); 252 253 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 254 * Also destroys all resources associated with the AudioTrack. 255 */ 256protected: 257 virtual ~AudioTrack(); 258public: 259 260 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 261 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 262 * set() is not multi-thread safe. 263 * Returned status (from utils/Errors.h) can be: 264 * - NO_ERROR: successful initialization 265 * - INVALID_OPERATION: AudioTrack is already initialized 266 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 267 * - NO_INIT: audio server or audio hardware not initialized 268 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 269 * If sharedBuffer is non-0, the frameCount parameter is ignored and 270 * replaced by the shared buffer's total allocated size in frame units. 271 * 272 * Parameters not listed in the AudioTrack constructors above: 273 * 274 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 275 * 276 * Internal state post condition: 277 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 278 */ 279 status_t set(audio_stream_type_t streamType, 280 uint32_t sampleRate, 281 audio_format_t format, 282 audio_channel_mask_t channelMask, 283 size_t frameCount = 0, 284 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 285 callback_t cbf = NULL, 286 void* user = NULL, 287 uint32_t notificationFrames = 0, 288 const sp<IMemory>& sharedBuffer = 0, 289 bool threadCanCallJava = false, 290 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 291 transfer_type transferType = TRANSFER_DEFAULT, 292 const audio_offload_info_t *offloadInfo = NULL, 293 int uid = -1, 294 pid_t pid = -1, 295 const audio_attributes_t* pAttributes = NULL, 296 bool doNotReconnect = false); 297 298 /* Result of constructing the AudioTrack. This must be checked for successful initialization 299 * before using any AudioTrack API (except for set()), because using 300 * an uninitialized AudioTrack produces undefined results. 301 * See set() method above for possible return codes. 302 */ 303 status_t initCheck() const { return mStatus; } 304 305 /* Returns this track's estimated latency in milliseconds. 306 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 307 * and audio hardware driver. 308 */ 309 uint32_t latency() const { return mLatency; } 310 311 /* Returns the number of application-level buffer underruns 312 * since the AudioTrack was created. 313 */ 314 uint32_t getUnderrunCount() const; 315 316 /* getters, see constructors and set() */ 317 318 audio_stream_type_t streamType() const; 319 audio_format_t format() const { return mFormat; } 320 321 /* Return frame size in bytes, which for linear PCM is 322 * channelCount * (bit depth per channel / 8). 323 * channelCount is determined from channelMask, and bit depth comes from format. 324 * For non-linear formats, the frame size is typically 1 byte. 325 */ 326 size_t frameSize() const { return mFrameSize; } 327 328 uint32_t channelCount() const { return mChannelCount; } 329 size_t frameCount() const { return mFrameCount; } 330 331 /* Return effective size of audio buffer that an application writes to 332 * or a negative error if the track is uninitialized. 333 */ 334 ssize_t getBufferSizeInFrames(); 335 336 /* Set the effective size of audio buffer that an application writes to. 337 * This is used to determine the amount of available room in the buffer, 338 * which determines when a write will block. 339 * This allows an application to raise and lower the audio latency. 340 * The requested size may be adjusted so that it is 341 * greater or equal to the absolute minimum and 342 * less than or equal to the getBufferCapacityInFrames(). 343 * It may also be adjusted slightly for internal reasons. 344 * 345 * Return the final size or a negative error if the track is unitialized 346 * or does not support variable sizes. 347 */ 348 ssize_t setBufferSizeInFrames(size_t size); 349 350 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 351 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 352 353 /* After it's created the track is not active. Call start() to 354 * make it active. If set, the callback will start being called. 355 * If the track was previously paused, volume is ramped up over the first mix buffer. 356 */ 357 status_t start(); 358 359 /* Stop a track. 360 * In static buffer mode, the track is stopped immediately. 361 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 362 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 363 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 364 * is first drained, mixed, and output, and only then is the track marked as stopped. 365 */ 366 void stop(); 367 bool stopped() const; 368 369 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 370 * This has the effect of draining the buffers without mixing or output. 371 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 372 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 373 */ 374 void flush(); 375 376 /* Pause a track. After pause, the callback will cease being called and 377 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 378 * and will fill up buffers until the pool is exhausted. 379 * Volume is ramped down over the next mix buffer following the pause request, 380 * and then the track is marked as paused. It can be resumed with ramp up by start(). 381 */ 382 void pause(); 383 384 /* Set volume for this track, mostly used for games' sound effects 385 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 386 * This is the older API. New applications should use setVolume(float) when possible. 387 */ 388 status_t setVolume(float left, float right); 389 390 /* Set volume for all channels. This is the preferred API for new applications, 391 * especially for multi-channel content. 392 */ 393 status_t setVolume(float volume); 394 395 /* Set the send level for this track. An auxiliary effect should be attached 396 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 397 */ 398 status_t setAuxEffectSendLevel(float level); 399 void getAuxEffectSendLevel(float* level) const; 400 401 /* Set source sample rate for this track in Hz, mostly used for games' sound effects. 402 * Zero is not permitted. 403 */ 404 status_t setSampleRate(uint32_t sampleRate); 405 406 /* Return current source sample rate in Hz. 407 * If specified as zero in constructor or set(), this will be the sink sample rate. 408 */ 409 uint32_t getSampleRate() const; 410 411 /* Return the original source sample rate in Hz. This corresponds to the sample rate 412 * if playback rate had normal speed and pitch. 413 */ 414 uint32_t getOriginalSampleRate() const; 415 416 /* Set source playback rate for timestretch 417 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 418 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 419 * 420 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 421 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 422 * 423 * Speed increases the playback rate of media, but does not alter pitch. 424 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 425 */ 426 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 427 428 /* Return current playback rate */ 429 const AudioPlaybackRate& getPlaybackRate() const; 430 431 /* Enables looping and sets the start and end points of looping. 432 * Only supported for static buffer mode. 433 * 434 * Parameters: 435 * 436 * loopStart: loop start in frames relative to start of buffer. 437 * loopEnd: loop end in frames relative to start of buffer. 438 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 439 * pending or active loop. loopCount == -1 means infinite looping. 440 * 441 * For proper operation the following condition must be respected: 442 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 443 * 444 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 445 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 446 * 447 */ 448 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 449 450 /* Sets marker position. When playback reaches the number of frames specified, a callback with 451 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 452 * notification callback. To set a marker at a position which would compute as 0, 453 * a workaround is to set the marker at a nearby position such as ~0 or 1. 454 * If the AudioTrack has been opened with no callback function associated, the operation will 455 * fail. 456 * 457 * Parameters: 458 * 459 * marker: marker position expressed in wrapping (overflow) frame units, 460 * like the return value of getPosition(). 461 * 462 * Returned status (from utils/Errors.h) can be: 463 * - NO_ERROR: successful operation 464 * - INVALID_OPERATION: the AudioTrack has no callback installed. 465 */ 466 status_t setMarkerPosition(uint32_t marker); 467 status_t getMarkerPosition(uint32_t *marker) const; 468 469 /* Sets position update period. Every time the number of frames specified has been played, 470 * a callback with event type EVENT_NEW_POS is called. 471 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 472 * callback. 473 * If the AudioTrack has been opened with no callback function associated, the operation will 474 * fail. 475 * Extremely small values may be rounded up to a value the implementation can support. 476 * 477 * Parameters: 478 * 479 * updatePeriod: position update notification period expressed in frames. 480 * 481 * Returned status (from utils/Errors.h) can be: 482 * - NO_ERROR: successful operation 483 * - INVALID_OPERATION: the AudioTrack has no callback installed. 484 */ 485 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 486 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 487 488 /* Sets playback head position. 489 * Only supported for static buffer mode. 490 * 491 * Parameters: 492 * 493 * position: New playback head position in frames relative to start of buffer. 494 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 495 * but will result in an immediate underrun if started. 496 * 497 * Returned status (from utils/Errors.h) can be: 498 * - NO_ERROR: successful operation 499 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 500 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 501 * buffer 502 */ 503 status_t setPosition(uint32_t position); 504 505 /* Return the total number of frames played since playback start. 506 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 507 * It is reset to zero by flush(), reload(), and stop(). 508 * 509 * Parameters: 510 * 511 * position: Address where to return play head position. 512 * 513 * Returned status (from utils/Errors.h) can be: 514 * - NO_ERROR: successful operation 515 * - BAD_VALUE: position is NULL 516 */ 517 status_t getPosition(uint32_t *position); 518 519 /* For static buffer mode only, this returns the current playback position in frames 520 * relative to start of buffer. It is analogous to the position units used by 521 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 522 */ 523 status_t getBufferPosition(uint32_t *position); 524 525 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 526 * rewriting the buffer before restarting playback after a stop. 527 * This method must be called with the AudioTrack in paused or stopped state. 528 * Not allowed in streaming mode. 529 * 530 * Returned status (from utils/Errors.h) can be: 531 * - NO_ERROR: successful operation 532 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 533 */ 534 status_t reload(); 535 536 /* Returns a handle on the audio output used by this AudioTrack. 537 * 538 * Parameters: 539 * none. 540 * 541 * Returned value: 542 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 543 * track needed to be re-created but that failed 544 */ 545private: 546 audio_io_handle_t getOutput() const; 547public: 548 549 /* Selects the audio device to use for output of this AudioTrack. A value of 550 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 551 * 552 * Parameters: 553 * The device ID of the selected device (as returned by the AudioDevicesManager API). 554 * 555 * Returned value: 556 * - NO_ERROR: successful operation 557 * TODO: what else can happen here? 558 */ 559 status_t setOutputDevice(audio_port_handle_t deviceId); 560 561 /* Returns the ID of the audio device selected for this AudioTrack. 562 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 563 * 564 * Parameters: 565 * none. 566 */ 567 audio_port_handle_t getOutputDevice(); 568 569 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 570 * attached. 571 * A value of AUDIO_PORT_HANDLE_NONE indicates the audio track is not attached to any output. 572 * 573 * Parameters: 574 * none. 575 */ 576 audio_port_handle_t getRoutedDeviceId(); 577 578 /* Returns the unique session ID associated with this track. 579 * 580 * Parameters: 581 * none. 582 * 583 * Returned value: 584 * AudioTrack session ID. 585 */ 586 audio_session_t getSessionId() const { return mSessionId; } 587 588 /* Attach track auxiliary output to specified effect. Use effectId = 0 589 * to detach track from effect. 590 * 591 * Parameters: 592 * 593 * effectId: effectId obtained from AudioEffect::id(). 594 * 595 * Returned status (from utils/Errors.h) can be: 596 * - NO_ERROR: successful operation 597 * - INVALID_OPERATION: the effect is not an auxiliary effect. 598 * - BAD_VALUE: The specified effect ID is invalid 599 */ 600 status_t attachAuxEffect(int effectId); 601 602 /* Public API for TRANSFER_OBTAIN mode. 603 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 604 * After filling these slots with data, the caller should release them with releaseBuffer(). 605 * If the track buffer is not full, obtainBuffer() returns as many contiguous 606 * [empty slots for] frames as are available immediately. 607 * 608 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 609 * additional non-contiguous frames that are predicted to be available immediately, 610 * if the client were to release the first frames and then call obtainBuffer() again. 611 * This value is only a prediction, and needs to be confirmed. 612 * It will be set to zero for an error return. 613 * 614 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 615 * regardless of the value of waitCount. 616 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 617 * maximum timeout based on waitCount; see chart below. 618 * Buffers will be returned until the pool 619 * is exhausted, at which point obtainBuffer() will either block 620 * or return WOULD_BLOCK depending on the value of the "waitCount" 621 * parameter. 622 * 623 * Interpretation of waitCount: 624 * +n limits wait time to n * WAIT_PERIOD_MS, 625 * -1 causes an (almost) infinite wait time, 626 * 0 non-blocking. 627 * 628 * Buffer fields 629 * On entry: 630 * frameCount number of [empty slots for] frames requested 631 * size ignored 632 * raw ignored 633 * After error return: 634 * frameCount 0 635 * size 0 636 * raw undefined 637 * After successful return: 638 * frameCount actual number of [empty slots for] frames available, <= number requested 639 * size actual number of bytes available 640 * raw pointer to the buffer 641 */ 642 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 643 size_t *nonContig = NULL); 644 645private: 646 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 647 * additional non-contiguous frames that are predicted to be available immediately, 648 * if the client were to release the first frames and then call obtainBuffer() again. 649 * This value is only a prediction, and needs to be confirmed. 650 * It will be set to zero for an error return. 651 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 652 * in case the requested amount of frames is in two or more non-contiguous regions. 653 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 654 */ 655 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 656 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 657public: 658 659 /* Public API for TRANSFER_OBTAIN mode. 660 * Release a filled buffer of frames for AudioFlinger to process. 661 * 662 * Buffer fields: 663 * frameCount currently ignored but recommend to set to actual number of frames filled 664 * size actual number of bytes filled, must be multiple of frameSize 665 * raw ignored 666 */ 667 void releaseBuffer(const Buffer* audioBuffer); 668 669 /* As a convenience we provide a write() interface to the audio buffer. 670 * Input parameter 'size' is in byte units. 671 * This is implemented on top of obtainBuffer/releaseBuffer. For best 672 * performance use callbacks. Returns actual number of bytes written >= 0, 673 * or one of the following negative status codes: 674 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 675 * BAD_VALUE size is invalid 676 * WOULD_BLOCK when obtainBuffer() returns same, or 677 * AudioTrack was stopped during the write 678 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 679 * the track cannot be automatically restored. 680 * The application needs to recreate the AudioTrack 681 * because the audio device changed or AudioFlinger died. 682 * This typically occurs for direct or offload tracks 683 * or if mDoNotReconnect is true. 684 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 685 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 686 * false for the method to return immediately without waiting to try multiple times to write 687 * the full content of the buffer. 688 */ 689 ssize_t write(const void* buffer, size_t size, bool blocking = true); 690 691 /* 692 * Dumps the state of an audio track. 693 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 694 */ 695 status_t dump(int fd, const Vector<String16>& args) const; 696 697 /* 698 * Return the total number of frames which AudioFlinger desired but were unavailable, 699 * and thus which resulted in an underrun. Reset to zero by stop(). 700 */ 701 uint32_t getUnderrunFrames() const; 702 703 /* Get the flags */ 704 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 705 706 /* Set parameters - only possible when using direct output */ 707 status_t setParameters(const String8& keyValuePairs); 708 709 /* Get parameters */ 710 String8 getParameters(const String8& keys); 711 712 /* Poll for a timestamp on demand. 713 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 714 * or if you need to get the most recent timestamp outside of the event callback handler. 715 * Caution: calling this method too often may be inefficient; 716 * if you need a high resolution mapping between frame position and presentation time, 717 * consider implementing that at application level, based on the low resolution timestamps. 718 * Returns NO_ERROR if timestamp is valid. 719 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 720 * start/ACTIVE, when the number of frames consumed is less than the 721 * overall hardware latency to physical output. In WOULD_BLOCK cases, 722 * one might poll again, or use getPosition(), or use 0 position and 723 * current time for the timestamp. 724 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 725 * the track cannot be automatically restored. 726 * The application needs to recreate the AudioTrack 727 * because the audio device changed or AudioFlinger died. 728 * This typically occurs for direct or offload tracks 729 * or if mDoNotReconnect is true. 730 * INVALID_OPERATION if called on a FastTrack, wrong state, or some other error. 731 * 732 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 733 */ 734 status_t getTimestamp(AudioTimestamp& timestamp); 735 736 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 737 * AudioTrack is routed is updated. 738 * Replaces any previously installed callback. 739 * Parameters: 740 * callback: The callback interface 741 * Returns NO_ERROR if successful. 742 * INVALID_OPERATION if the same callback is already installed. 743 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 744 * BAD_VALUE if the callback is NULL 745 */ 746 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 747 748 /* remove an AudioDeviceCallback. 749 * Parameters: 750 * callback: The callback interface 751 * Returns NO_ERROR if successful. 752 * INVALID_OPERATION if the callback is not installed 753 * BAD_VALUE if the callback is NULL 754 */ 755 status_t removeAudioDeviceCallback( 756 const sp<AudioSystem::AudioDeviceCallback>& callback); 757 758protected: 759 /* copying audio tracks is not allowed */ 760 AudioTrack(const AudioTrack& other); 761 AudioTrack& operator = (const AudioTrack& other); 762 763 /* a small internal class to handle the callback */ 764 class AudioTrackThread : public Thread 765 { 766 public: 767 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 768 769 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 770 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 771 virtual void requestExit(); 772 773 void pause(); // suspend thread from execution at next loop boundary 774 void resume(); // allow thread to execute, if not requested to exit 775 void wake(); // wake to handle changed notification conditions. 776 777 private: 778 void pauseInternal(nsecs_t ns = 0LL); 779 // like pause(), but only used internally within thread 780 781 friend class AudioTrack; 782 virtual bool threadLoop(); 783 AudioTrack& mReceiver; 784 virtual ~AudioTrackThread(); 785 Mutex mMyLock; // Thread::mLock is private 786 Condition mMyCond; // Thread::mThreadExitedCondition is private 787 bool mPaused; // whether thread is requested to pause at next loop entry 788 bool mPausedInt; // whether thread internally requests pause 789 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 790 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 791 // to processAudioBuffer() as state may have changed 792 // since pause time calculated. 793 }; 794 795 // body of AudioTrackThread::threadLoop() 796 // returns the maximum amount of time before we would like to run again, where: 797 // 0 immediately 798 // > 0 no later than this many nanoseconds from now 799 // NS_WHENEVER still active but no particular deadline 800 // NS_INACTIVE inactive so don't run again until re-started 801 // NS_NEVER never again 802 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 803 nsecs_t processAudioBuffer(); 804 805 // caller must hold lock on mLock for all _l methods 806 807 status_t createTrack_l(); 808 809 // can only be called when mState != STATE_ACTIVE 810 void flush_l(); 811 812 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 813 814 // FIXME enum is faster than strcmp() for parameter 'from' 815 status_t restoreTrack_l(const char *from); 816 817 uint32_t getUnderrunCount_l() const; 818 819 bool isOffloaded() const; 820 bool isDirect() const; 821 bool isOffloadedOrDirect() const; 822 823 bool isOffloaded_l() const 824 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 825 826 bool isOffloadedOrDirect_l() const 827 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 828 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 829 830 bool isDirect_l() const 831 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 832 833 // increment mPosition by the delta of mServer, and return new value of mPosition 834 Modulo<uint32_t> updateAndGetPosition_l(); 835 836 // check sample rate and speed is compatible with AudioTrack 837 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; 838 839 void restartIfDisabled(); 840 841 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 842 sp<IAudioTrack> mAudioTrack; 843 sp<IMemory> mCblkMemory; 844 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 845 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 846 847 sp<AudioTrackThread> mAudioTrackThread; 848 bool mThreadCanCallJava; 849 850 float mVolume[2]; 851 float mSendLevel; 852 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 853 uint32_t mOriginalSampleRate; 854 AudioPlaybackRate mPlaybackRate; 855 856 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 857 // This allocated buffer size is maintained by the proxy. 858 size_t mFrameCount; // maximum size of buffer 859 860 size_t mReqFrameCount; // frame count to request the first or next time 861 // a new IAudioTrack is needed, non-decreasing 862 863 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 864 // These values can be used for informational purposes until the track is invalidated, 865 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 866 uint32_t mAfLatency; // AudioFlinger latency in ms 867 size_t mAfFrameCount; // AudioFlinger frame count 868 uint32_t mAfSampleRate; // AudioFlinger sample rate 869 870 // constant after constructor or set() 871 audio_format_t mFormat; // as requested by client, not forced to 16-bit 872 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 873 // this AudioTrack has valid attributes 874 uint32_t mChannelCount; 875 audio_channel_mask_t mChannelMask; 876 sp<IMemory> mSharedBuffer; 877 transfer_type mTransfer; 878 audio_offload_info_t mOffloadInfoCopy; 879 const audio_offload_info_t* mOffloadInfo; 880 audio_attributes_t mAttributes; 881 882 size_t mFrameSize; // frame size in bytes 883 884 status_t mStatus; 885 886 // can change dynamically when IAudioTrack invalidated 887 uint32_t mLatency; // in ms 888 889 // Indicates the current track state. Protected by mLock. 890 enum State { 891 STATE_ACTIVE, 892 STATE_STOPPED, 893 STATE_PAUSED, 894 STATE_PAUSED_STOPPING, 895 STATE_FLUSHED, 896 STATE_STOPPING, 897 } mState; 898 899 // for client callback handler 900 callback_t mCbf; // callback handler for events, or NULL 901 void* mUserData; 902 903 // for notification APIs 904 uint32_t mNotificationFramesReq; // requested number of frames between each 905 // notification callback, 906 // at initial source sample rate 907 uint32_t mNotificationFramesAct; // actual number of frames between each 908 // notification callback, 909 // at initial source sample rate 910 bool mRefreshRemaining; // processAudioBuffer() should refresh 911 // mRemainingFrames and mRetryOnPartialBuffer 912 913 // used for static track cbf and restoration 914 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 915 uint32_t mLoopStart; // last setLoop loopStart 916 uint32_t mLoopEnd; // last setLoop loopEnd 917 int32_t mLoopCountNotified; // the last loopCount notified by callback. 918 // mLoopCountNotified counts down, matching 919 // the remaining loop count for static track 920 // playback. 921 922 // These are private to processAudioBuffer(), and are not protected by a lock 923 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 924 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 925 uint32_t mObservedSequence; // last observed value of mSequence 926 927 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 928 bool mMarkerReached; 929 Modulo<uint32_t> mNewPosition; // in frames 930 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 931 932 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 933 // which is count of frames consumed by server, 934 // reset by new IAudioTrack, 935 // whether it is reset by stop() is TBD 936 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 937 // monotonically after new IAudioTrack, 938 // and could be easily widened to uint64_t 939 Modulo<uint32_t> mReleased; // count of frames released to server 940 // but not necessarily consumed by server, 941 // reset by stop() but continues monotonically 942 // after new IAudioTrack to restore mPosition, 943 // and could be easily widened to uint64_t 944 int64_t mStartUs; // the start time after flush or stop. 945 // only used for offloaded and direct tracks. 946 947 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 948 bool mTimestampStartupGlitchReported; // reduce log spam 949 bool mRetrogradeMotionReported; // reduce log spam 950 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 951 952 uint32_t mUnderrunCountOffset; // updated when restoring tracks 953 954 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 955 // be denied by client or server, such as 956 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 957 // held to read or write those bits reliably. 958 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 959 960 bool mDoNotReconnect; 961 962 audio_session_t mSessionId; 963 int mAuxEffectId; 964 965 mutable Mutex mLock; 966 967 int mPreviousPriority; // before start() 968 SchedPolicy mPreviousSchedulingGroup; 969 bool mAwaitBoost; // thread should wait for priority boost before running 970 971 // The proxy should only be referenced while a lock is held because the proxy isn't 972 // multi-thread safe, especially the SingleStateQueue part of the proxy. 973 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 974 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 975 // them around in case they are replaced during the obtainBuffer(). 976 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 977 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 978 979 bool mInUnderrun; // whether track is currently in underrun state 980 uint32_t mPausedPosition; 981 982 // For Device Selection API 983 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 984 audio_port_handle_t mSelectedDeviceId; 985 986private: 987 class DeathNotifier : public IBinder::DeathRecipient { 988 public: 989 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 990 protected: 991 virtual void binderDied(const wp<IBinder>& who); 992 private: 993 const wp<AudioTrack> mAudioTrack; 994 }; 995 996 sp<DeathNotifier> mDeathNotifier; 997 uint32_t mSequence; // incremented for each new IAudioTrack attempt 998 int mClientUid; 999 pid_t mClientPid; 1000 1001 sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1002}; 1003 1004}; // namespace android 1005 1006#endif // ANDROID_AUDIOTRACK_H 1007