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 28 namespace android { 29 30 // ---------------------------------------------------------------------------- 31 32 struct audio_track_cblk_t; 33 class AudioTrackClientProxy; 34 class StaticAudioTrackClientProxy; 35 36 // ---------------------------------------------------------------------------- 37 38 class AudioTrack : public RefBase 39 { 40 public: 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 by server. 190 * Zero means to use a default value, which is typically: 191 * - fast tracks: HAL buffer size, even if track frameCount is larger 192 * - normal tracks: 1/2 of track frameCount 193 * A positive value means that many frames at initial source sample rate. 194 * A negative value for this parameter specifies the negative of the 195 * requested number of notifications (sub-buffers) in the entire buffer. 196 * For fast tracks, the FastMixer will process one sub-buffer at a time. 197 * The size of each sub-buffer is determined by the HAL. 198 * To get "double buffering", for example, one should pass -2. 199 * The minimum number of sub-buffers is 1 (expressed as -1), 200 * and the maximum number of sub-buffers is 8 (expressed as -8). 201 * Negative is only permitted for fast tracks, and if frameCount is zero. 202 * TODO It is ugly to overload a parameter in this way depending on 203 * whether it is positive, negative, or zero. Consider splitting apart. 204 * sessionId: Specific session ID, or zero to use default. 205 * transferType: How data is transferred to AudioTrack. 206 * offloadInfo: If not NULL, provides offload parameters for 207 * AudioSystem::getOutputForAttr(). 208 * uid: User ID of the app which initially requested this AudioTrack 209 * for power management tracking, or -1 for current user ID. 210 * pid: Process ID of the app which initially requested this AudioTrack 211 * for power management tracking, or -1 for current process ID. 212 * pAttributes: If not NULL, supersedes streamType for use case selection. 213 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 214 binder to AudioFlinger. 215 It will return an error instead. The application will recreate 216 the track based on offloading or different channel configuration, etc. 217 * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow 218 * maxRequiredSpeed playback. Values less than 1.0f and greater than 219 * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks 220 * and direct or offloaded tracks, this parameter is ignored. 221 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 222 */ 223 224 AudioTrack( audio_stream_type_t streamType, 225 uint32_t sampleRate, 226 audio_format_t format, 227 audio_channel_mask_t channelMask, 228 size_t frameCount = 0, 229 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 230 callback_t cbf = NULL, 231 void* user = NULL, 232 int32_t notificationFrames = 0, 233 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 234 transfer_type transferType = TRANSFER_DEFAULT, 235 const audio_offload_info_t *offloadInfo = NULL, 236 int uid = -1, 237 pid_t pid = -1, 238 const audio_attributes_t* pAttributes = NULL, 239 bool doNotReconnect = false, 240 float maxRequiredSpeed = 1.0f); 241 242 /* Creates an audio track and registers it with AudioFlinger. 243 * With this constructor, the track is configured for static buffer mode. 244 * Data to be rendered is passed in a shared memory buffer 245 * identified by the argument sharedBuffer, which should be non-0. 246 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 247 * but without the ability to specify a non-zero value for the frameCount parameter. 248 * The memory should be initialized to the desired data before calling start(). 249 * The write() method is not supported in this case. 250 * It is recommended to pass a callback function to be notified of playback end by an 251 * EVENT_UNDERRUN event. 252 */ 253 254 AudioTrack( audio_stream_type_t streamType, 255 uint32_t sampleRate, 256 audio_format_t format, 257 audio_channel_mask_t channelMask, 258 const sp<IMemory>& sharedBuffer, 259 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 260 callback_t cbf = NULL, 261 void* user = NULL, 262 int32_t notificationFrames = 0, 263 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 264 transfer_type transferType = TRANSFER_DEFAULT, 265 const audio_offload_info_t *offloadInfo = NULL, 266 int uid = -1, 267 pid_t pid = -1, 268 const audio_attributes_t* pAttributes = NULL, 269 bool doNotReconnect = false, 270 float maxRequiredSpeed = 1.0f); 271 272 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 273 * Also destroys all resources associated with the AudioTrack. 274 */ 275 protected: 276 virtual ~AudioTrack(); 277 public: 278 279 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 280 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 281 * set() is not multi-thread safe. 282 * Returned status (from utils/Errors.h) can be: 283 * - NO_ERROR: successful initialization 284 * - INVALID_OPERATION: AudioTrack is already initialized 285 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 286 * - NO_INIT: audio server or audio hardware not initialized 287 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 288 * If sharedBuffer is non-0, the frameCount parameter is ignored and 289 * replaced by the shared buffer's total allocated size in frame units. 290 * 291 * Parameters not listed in the AudioTrack constructors above: 292 * 293 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 294 * 295 * Internal state post condition: 296 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 297 */ 298 status_t set(audio_stream_type_t streamType, 299 uint32_t sampleRate, 300 audio_format_t format, 301 audio_channel_mask_t channelMask, 302 size_t frameCount = 0, 303 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 304 callback_t cbf = NULL, 305 void* user = NULL, 306 int32_t notificationFrames = 0, 307 const sp<IMemory>& sharedBuffer = 0, 308 bool threadCanCallJava = false, 309 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 310 transfer_type transferType = TRANSFER_DEFAULT, 311 const audio_offload_info_t *offloadInfo = NULL, 312 int uid = -1, 313 pid_t pid = -1, 314 const audio_attributes_t* pAttributes = NULL, 315 bool doNotReconnect = false, 316 float maxRequiredSpeed = 1.0f); 317 318 /* Result of constructing the AudioTrack. This must be checked for successful initialization 319 * before using any AudioTrack API (except for set()), because using 320 * an uninitialized AudioTrack produces undefined results. 321 * See set() method above for possible return codes. 322 */ 323 status_t initCheck() const { return mStatus; } 324 325 /* Returns this track's estimated latency in milliseconds. 326 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 327 * and audio hardware driver. 328 */ 329 uint32_t latency() const { return mLatency; } 330 331 /* Returns the number of application-level buffer underruns 332 * since the AudioTrack was created. 333 */ 334 uint32_t getUnderrunCount() const; 335 336 /* getters, see constructors and set() */ 337 338 audio_stream_type_t streamType() const; 339 audio_format_t format() const { return mFormat; } 340 341 /* Return frame size in bytes, which for linear PCM is 342 * channelCount * (bit depth per channel / 8). 343 * channelCount is determined from channelMask, and bit depth comes from format. 344 * For non-linear formats, the frame size is typically 1 byte. 345 */ 346 size_t frameSize() const { return mFrameSize; } 347 348 uint32_t channelCount() const { return mChannelCount; } 349 size_t frameCount() const { return mFrameCount; } 350 351 // TODO consider notificationFrames() if needed 352 353 /* Return effective size of audio buffer that an application writes to 354 * or a negative error if the track is uninitialized. 355 */ 356 ssize_t getBufferSizeInFrames(); 357 358 /* Returns the buffer duration in microseconds at current playback rate. 359 */ 360 status_t getBufferDurationInUs(int64_t *duration); 361 362 /* Set the effective size of audio buffer that an application writes to. 363 * This is used to determine the amount of available room in the buffer, 364 * which determines when a write will block. 365 * This allows an application to raise and lower the audio latency. 366 * The requested size may be adjusted so that it is 367 * greater or equal to the absolute minimum and 368 * less than or equal to the getBufferCapacityInFrames(). 369 * It may also be adjusted slightly for internal reasons. 370 * 371 * Return the final size or a negative error if the track is unitialized 372 * or does not support variable sizes. 373 */ 374 ssize_t setBufferSizeInFrames(size_t size); 375 376 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ 377 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 378 379 /* After it's created the track is not active. Call start() to 380 * make it active. If set, the callback will start being called. 381 * If the track was previously paused, volume is ramped up over the first mix buffer. 382 */ 383 status_t start(); 384 385 /* Stop a track. 386 * In static buffer mode, the track is stopped immediately. 387 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 388 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 389 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 390 * is first drained, mixed, and output, and only then is the track marked as stopped. 391 */ 392 void stop(); 393 bool stopped() const; 394 395 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 396 * This has the effect of draining the buffers without mixing or output. 397 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 398 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 399 */ 400 void flush(); 401 402 /* Pause a track. After pause, the callback will cease being called and 403 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 404 * and will fill up buffers until the pool is exhausted. 405 * Volume is ramped down over the next mix buffer following the pause request, 406 * and then the track is marked as paused. It can be resumed with ramp up by start(). 407 */ 408 void pause(); 409 410 /* Set volume for this track, mostly used for games' sound effects 411 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 412 * This is the older API. New applications should use setVolume(float) when possible. 413 */ 414 status_t setVolume(float left, float right); 415 416 /* Set volume for all channels. This is the preferred API for new applications, 417 * especially for multi-channel content. 418 */ 419 status_t setVolume(float volume); 420 421 /* Set the send level for this track. An auxiliary effect should be attached 422 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 423 */ 424 status_t setAuxEffectSendLevel(float level); 425 void getAuxEffectSendLevel(float* level) const; 426 427 /* Set source sample rate for this track in Hz, mostly used for games' sound effects. 428 * Zero is not permitted. 429 */ 430 status_t setSampleRate(uint32_t sampleRate); 431 432 /* Return current source sample rate in Hz. 433 * If specified as zero in constructor or set(), this will be the sink sample rate. 434 */ 435 uint32_t getSampleRate() const; 436 437 /* Return the original source sample rate in Hz. This corresponds to the sample rate 438 * if playback rate had normal speed and pitch. 439 */ 440 uint32_t getOriginalSampleRate() const; 441 442 /* Set source playback rate for timestretch 443 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 444 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 445 * 446 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 447 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 448 * 449 * Speed increases the playback rate of media, but does not alter pitch. 450 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 451 */ 452 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 453 454 /* Return current playback rate */ 455 const AudioPlaybackRate& getPlaybackRate() const; 456 457 /* Enables looping and sets the start and end points of looping. 458 * Only supported for static buffer mode. 459 * 460 * Parameters: 461 * 462 * loopStart: loop start in frames relative to start of buffer. 463 * loopEnd: loop end in frames relative to start of buffer. 464 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 465 * pending or active loop. loopCount == -1 means infinite looping. 466 * 467 * For proper operation the following condition must be respected: 468 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 469 * 470 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 471 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 472 * 473 */ 474 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 475 476 /* Sets marker position. When playback reaches the number of frames specified, a callback with 477 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 478 * notification callback. To set a marker at a position which would compute as 0, 479 * a workaround is to set the marker at a nearby position such as ~0 or 1. 480 * If the AudioTrack has been opened with no callback function associated, the operation will 481 * fail. 482 * 483 * Parameters: 484 * 485 * marker: marker position expressed in wrapping (overflow) frame units, 486 * like the return value of getPosition(). 487 * 488 * Returned status (from utils/Errors.h) can be: 489 * - NO_ERROR: successful operation 490 * - INVALID_OPERATION: the AudioTrack has no callback installed. 491 */ 492 status_t setMarkerPosition(uint32_t marker); 493 status_t getMarkerPosition(uint32_t *marker) const; 494 495 /* Sets position update period. Every time the number of frames specified has been played, 496 * a callback with event type EVENT_NEW_POS is called. 497 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 498 * callback. 499 * If the AudioTrack has been opened with no callback function associated, the operation will 500 * fail. 501 * Extremely small values may be rounded up to a value the implementation can support. 502 * 503 * Parameters: 504 * 505 * updatePeriod: position update notification period expressed in frames. 506 * 507 * Returned status (from utils/Errors.h) can be: 508 * - NO_ERROR: successful operation 509 * - INVALID_OPERATION: the AudioTrack has no callback installed. 510 */ 511 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 512 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 513 514 /* Sets playback head position. 515 * Only supported for static buffer mode. 516 * 517 * Parameters: 518 * 519 * position: New playback head position in frames relative to start of buffer. 520 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 521 * but will result in an immediate underrun if started. 522 * 523 * Returned status (from utils/Errors.h) can be: 524 * - NO_ERROR: successful operation 525 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 526 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 527 * buffer 528 */ 529 status_t setPosition(uint32_t position); 530 531 /* Return the total number of frames played since playback start. 532 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 533 * It is reset to zero by flush(), reload(), and stop(). 534 * 535 * Parameters: 536 * 537 * position: Address where to return play head position. 538 * 539 * Returned status (from utils/Errors.h) can be: 540 * - NO_ERROR: successful operation 541 * - BAD_VALUE: position is NULL 542 */ 543 status_t getPosition(uint32_t *position); 544 545 /* For static buffer mode only, this returns the current playback position in frames 546 * relative to start of buffer. It is analogous to the position units used by 547 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 548 */ 549 status_t getBufferPosition(uint32_t *position); 550 551 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 552 * rewriting the buffer before restarting playback after a stop. 553 * This method must be called with the AudioTrack in paused or stopped state. 554 * Not allowed in streaming mode. 555 * 556 * Returned status (from utils/Errors.h) can be: 557 * - NO_ERROR: successful operation 558 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 559 */ 560 status_t reload(); 561 562 /* Returns a handle on the audio output used by this AudioTrack. 563 * 564 * Parameters: 565 * none. 566 * 567 * Returned value: 568 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 569 * track needed to be re-created but that failed 570 */ 571 private: 572 audio_io_handle_t getOutput() const; 573 public: 574 575 /* Selects the audio device to use for output of this AudioTrack. A value of 576 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 577 * 578 * Parameters: 579 * The device ID of the selected device (as returned by the AudioDevicesManager API). 580 * 581 * Returned value: 582 * - NO_ERROR: successful operation 583 * TODO: what else can happen here? 584 */ 585 status_t setOutputDevice(audio_port_handle_t deviceId); 586 587 /* Returns the ID of the audio device selected for this AudioTrack. 588 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 589 * 590 * Parameters: 591 * none. 592 */ 593 audio_port_handle_t getOutputDevice(); 594 595 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 596 * attached. 597 * A value of AUDIO_PORT_HANDLE_NONE indicates the audio track is not attached to any output. 598 * 599 * Parameters: 600 * none. 601 */ 602 audio_port_handle_t getRoutedDeviceId(); 603 604 /* Returns the unique session ID associated with this track. 605 * 606 * Parameters: 607 * none. 608 * 609 * Returned value: 610 * AudioTrack session ID. 611 */ 612 audio_session_t getSessionId() const { return mSessionId; } 613 614 /* Attach track auxiliary output to specified effect. Use effectId = 0 615 * to detach track from effect. 616 * 617 * Parameters: 618 * 619 * effectId: effectId obtained from AudioEffect::id(). 620 * 621 * Returned status (from utils/Errors.h) can be: 622 * - NO_ERROR: successful operation 623 * - INVALID_OPERATION: the effect is not an auxiliary effect. 624 * - BAD_VALUE: The specified effect ID is invalid 625 */ 626 status_t attachAuxEffect(int effectId); 627 628 /* Public API for TRANSFER_OBTAIN mode. 629 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 630 * After filling these slots with data, the caller should release them with releaseBuffer(). 631 * If the track buffer is not full, obtainBuffer() returns as many contiguous 632 * [empty slots for] frames as are available immediately. 633 * 634 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 635 * additional non-contiguous frames that are predicted to be available immediately, 636 * if the client were to release the first frames and then call obtainBuffer() again. 637 * This value is only a prediction, and needs to be confirmed. 638 * It will be set to zero for an error return. 639 * 640 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 641 * regardless of the value of waitCount. 642 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 643 * maximum timeout based on waitCount; see chart below. 644 * Buffers will be returned until the pool 645 * is exhausted, at which point obtainBuffer() will either block 646 * or return WOULD_BLOCK depending on the value of the "waitCount" 647 * parameter. 648 * 649 * Interpretation of waitCount: 650 * +n limits wait time to n * WAIT_PERIOD_MS, 651 * -1 causes an (almost) infinite wait time, 652 * 0 non-blocking. 653 * 654 * Buffer fields 655 * On entry: 656 * frameCount number of [empty slots for] frames requested 657 * size ignored 658 * raw ignored 659 * After error return: 660 * frameCount 0 661 * size 0 662 * raw undefined 663 * After successful return: 664 * frameCount actual number of [empty slots for] frames available, <= number requested 665 * size actual number of bytes available 666 * raw pointer to the buffer 667 */ 668 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 669 size_t *nonContig = NULL); 670 671 private: 672 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 673 * additional non-contiguous frames that are predicted to be available immediately, 674 * if the client were to release the first frames and then call obtainBuffer() again. 675 * This value is only a prediction, and needs to be confirmed. 676 * It will be set to zero for an error return. 677 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 678 * in case the requested amount of frames is in two or more non-contiguous regions. 679 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 680 */ 681 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 682 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 683 public: 684 685 /* Public API for TRANSFER_OBTAIN mode. 686 * Release a filled buffer of frames for AudioFlinger to process. 687 * 688 * Buffer fields: 689 * frameCount currently ignored but recommend to set to actual number of frames filled 690 * size actual number of bytes filled, must be multiple of frameSize 691 * raw ignored 692 */ 693 void releaseBuffer(const Buffer* audioBuffer); 694 695 /* As a convenience we provide a write() interface to the audio buffer. 696 * Input parameter 'size' is in byte units. 697 * This is implemented on top of obtainBuffer/releaseBuffer. For best 698 * performance use callbacks. Returns actual number of bytes written >= 0, 699 * or one of the following negative status codes: 700 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 701 * BAD_VALUE size is invalid 702 * WOULD_BLOCK when obtainBuffer() returns same, or 703 * AudioTrack was stopped during the write 704 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 705 * the track cannot be automatically restored. 706 * The application needs to recreate the AudioTrack 707 * because the audio device changed or AudioFlinger died. 708 * This typically occurs for direct or offload tracks 709 * or if mDoNotReconnect is true. 710 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 711 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 712 * false for the method to return immediately without waiting to try multiple times to write 713 * the full content of the buffer. 714 */ 715 ssize_t write(const void* buffer, size_t size, bool blocking = true); 716 717 /* 718 * Dumps the state of an audio track. 719 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 720 */ 721 status_t dump(int fd, const Vector<String16>& args) const; 722 723 /* 724 * Return the total number of frames which AudioFlinger desired but were unavailable, 725 * and thus which resulted in an underrun. Reset to zero by stop(). 726 */ 727 uint32_t getUnderrunFrames() const; 728 729 /* Get the flags */ 730 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 731 732 /* Set parameters - only possible when using direct output */ 733 status_t setParameters(const String8& keyValuePairs); 734 735 /* Get parameters */ 736 String8 getParameters(const String8& keys); 737 738 /* Poll for a timestamp on demand. 739 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 740 * or if you need to get the most recent timestamp outside of the event callback handler. 741 * Caution: calling this method too often may be inefficient; 742 * if you need a high resolution mapping between frame position and presentation time, 743 * consider implementing that at application level, based on the low resolution timestamps. 744 * Returns NO_ERROR if timestamp is valid. 745 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 746 * start/ACTIVE, when the number of frames consumed is less than the 747 * overall hardware latency to physical output. In WOULD_BLOCK cases, 748 * one might poll again, or use getPosition(), or use 0 position and 749 * current time for the timestamp. 750 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 751 * the track cannot be automatically restored. 752 * The application needs to recreate the AudioTrack 753 * because the audio device changed or AudioFlinger died. 754 * This typically occurs for direct or offload tracks 755 * or if mDoNotReconnect is true. 756 * INVALID_OPERATION wrong state, or some other error. 757 * 758 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 759 */ 760 status_t getTimestamp(AudioTimestamp& timestamp); 761 762 /* Return the extended timestamp, with additional timebase info and improved drain behavior. 763 * 764 * This is similar to the AudioTrack.java API: 765 * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) 766 * 767 * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method 768 * 769 * 1. stop() by itself does not reset the frame position. 770 * A following start() resets the frame position to 0. 771 * 2. flush() by itself does not reset the frame position. 772 * The frame position advances by the number of frames flushed, 773 * when the first frame after flush reaches the audio sink. 774 * 3. BOOTTIME clock offsets are provided to help synchronize with 775 * non-audio streams, e.g. sensor data. 776 * 4. Position is returned with 64 bits of resolution. 777 * 778 * Parameters: 779 * timestamp: A pointer to the caller allocated ExtendedTimestamp. 780 * 781 * Returns NO_ERROR on success; timestamp is filled with valid data. 782 * BAD_VALUE if timestamp is NULL. 783 * WOULD_BLOCK if called immediately after start() when the number 784 * of frames consumed is less than the 785 * overall hardware latency to physical output. In WOULD_BLOCK cases, 786 * one might poll again, or use getPosition(), or use 0 position and 787 * current time for the timestamp. 788 * If WOULD_BLOCK is returned, the timestamp is still 789 * modified with the LOCATION_CLIENT portion filled. 790 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 791 * the track cannot be automatically restored. 792 * The application needs to recreate the AudioTrack 793 * because the audio device changed or AudioFlinger died. 794 * This typically occurs for direct or offloaded tracks 795 * or if mDoNotReconnect is true. 796 * INVALID_OPERATION if called on a offloaded or direct track. 797 * Use getTimestamp(AudioTimestamp& timestamp) instead. 798 */ 799 status_t getTimestamp(ExtendedTimestamp *timestamp); 800 private: 801 status_t getTimestamp_l(ExtendedTimestamp *timestamp); 802 public: 803 804 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 805 * AudioTrack is routed is updated. 806 * Replaces any previously installed callback. 807 * Parameters: 808 * callback: The callback interface 809 * Returns NO_ERROR if successful. 810 * INVALID_OPERATION if the same callback is already installed. 811 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 812 * BAD_VALUE if the callback is NULL 813 */ 814 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 815 816 /* remove an AudioDeviceCallback. 817 * Parameters: 818 * callback: The callback interface 819 * Returns NO_ERROR if successful. 820 * INVALID_OPERATION if the callback is not installed 821 * BAD_VALUE if the callback is NULL 822 */ 823 status_t removeAudioDeviceCallback( 824 const sp<AudioSystem::AudioDeviceCallback>& callback); 825 826 /* Obtain the pending duration in milliseconds for playback of pure PCM 827 * (mixable without embedded timing) data remaining in AudioTrack. 828 * 829 * This is used to estimate the drain time for the client-server buffer 830 * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. 831 * One may optionally request to find the duration to play through the HAL 832 * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, 833 * INVALID_OPERATION may be returned if the kernel location is unavailable. 834 * 835 * Returns NO_ERROR if successful. 836 * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained 837 * or the AudioTrack does not contain pure PCM data. 838 * BAD_VALUE if msec is nullptr or location is invalid. 839 */ 840 status_t pendingDuration(int32_t *msec, 841 ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); 842 843 protected: 844 /* copying audio tracks is not allowed */ 845 AudioTrack(const AudioTrack& other); 846 AudioTrack& operator = (const AudioTrack& other); 847 848 /* a small internal class to handle the callback */ 849 class AudioTrackThread : public Thread 850 { 851 public: 852 AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); 853 854 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 855 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 856 virtual void requestExit(); 857 858 void pause(); // suspend thread from execution at next loop boundary 859 void resume(); // allow thread to execute, if not requested to exit 860 void wake(); // wake to handle changed notification conditions. 861 862 private: 863 void pauseInternal(nsecs_t ns = 0LL); 864 // like pause(), but only used internally within thread 865 866 friend class AudioTrack; 867 virtual bool threadLoop(); 868 AudioTrack& mReceiver; 869 virtual ~AudioTrackThread(); 870 Mutex mMyLock; // Thread::mLock is private 871 Condition mMyCond; // Thread::mThreadExitedCondition is private 872 bool mPaused; // whether thread is requested to pause at next loop entry 873 bool mPausedInt; // whether thread internally requests pause 874 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 875 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 876 // to processAudioBuffer() as state may have changed 877 // since pause time calculated. 878 }; 879 880 // body of AudioTrackThread::threadLoop() 881 // returns the maximum amount of time before we would like to run again, where: 882 // 0 immediately 883 // > 0 no later than this many nanoseconds from now 884 // NS_WHENEVER still active but no particular deadline 885 // NS_INACTIVE inactive so don't run again until re-started 886 // NS_NEVER never again 887 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 888 nsecs_t processAudioBuffer(); 889 890 // caller must hold lock on mLock for all _l methods 891 892 status_t createTrack_l(); 893 894 // can only be called when mState != STATE_ACTIVE 895 void flush_l(); 896 897 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 898 899 // FIXME enum is faster than strcmp() for parameter 'from' 900 status_t restoreTrack_l(const char *from); 901 902 uint32_t getUnderrunCount_l() const; 903 904 bool isOffloaded() const; 905 bool isDirect() const; 906 bool isOffloadedOrDirect() const; 907 908 bool isOffloaded_l() const 909 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 910 911 bool isOffloadedOrDirect_l() const 912 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 913 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 914 915 bool isDirect_l() const 916 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 917 918 // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) 919 bool isPurePcmData_l() const 920 { return audio_is_linear_pcm(mFormat) 921 && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } 922 923 // increment mPosition by the delta of mServer, and return new value of mPosition 924 Modulo<uint32_t> updateAndGetPosition_l(); 925 926 // check sample rate and speed is compatible with AudioTrack 927 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const; 928 929 void restartIfDisabled(); 930 931 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 932 sp<IAudioTrack> mAudioTrack; 933 sp<IMemory> mCblkMemory; 934 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 935 audio_io_handle_t mOutput; // returned by AudioSystem::getOutput() 936 937 sp<AudioTrackThread> mAudioTrackThread; 938 bool mThreadCanCallJava; 939 940 float mVolume[2]; 941 float mSendLevel; 942 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 943 uint32_t mOriginalSampleRate; 944 AudioPlaybackRate mPlaybackRate; 945 float mMaxRequiredSpeed; // use PCM buffer size to allow this speed 946 947 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 948 // This allocated buffer size is maintained by the proxy. 949 size_t mFrameCount; // maximum size of buffer 950 951 size_t mReqFrameCount; // frame count to request the first or next time 952 // a new IAudioTrack is needed, non-decreasing 953 954 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 955 // These values can be used for informational purposes until the track is invalidated, 956 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 957 uint32_t mAfLatency; // AudioFlinger latency in ms 958 size_t mAfFrameCount; // AudioFlinger frame count 959 uint32_t mAfSampleRate; // AudioFlinger sample rate 960 961 // constant after constructor or set() 962 audio_format_t mFormat; // as requested by client, not forced to 16-bit 963 audio_stream_type_t mStreamType; // mStreamType == AUDIO_STREAM_DEFAULT implies 964 // this AudioTrack has valid attributes 965 uint32_t mChannelCount; 966 audio_channel_mask_t mChannelMask; 967 sp<IMemory> mSharedBuffer; 968 transfer_type mTransfer; 969 audio_offload_info_t mOffloadInfoCopy; 970 const audio_offload_info_t* mOffloadInfo; 971 audio_attributes_t mAttributes; 972 973 size_t mFrameSize; // frame size in bytes 974 975 status_t mStatus; 976 977 // can change dynamically when IAudioTrack invalidated 978 uint32_t mLatency; // in ms 979 980 // Indicates the current track state. Protected by mLock. 981 enum State { 982 STATE_ACTIVE, 983 STATE_STOPPED, 984 STATE_PAUSED, 985 STATE_PAUSED_STOPPING, 986 STATE_FLUSHED, 987 STATE_STOPPING, 988 } mState; 989 990 // for client callback handler 991 callback_t mCbf; // callback handler for events, or NULL 992 void* mUserData; 993 994 // for notification APIs 995 996 // next 2 fields are const after constructor or set() 997 uint32_t mNotificationFramesReq; // requested number of frames between each 998 // notification callback, 999 // at initial source sample rate 1000 uint32_t mNotificationsPerBufferReq; 1001 // requested number of notifications per buffer, 1002 // currently only used for fast tracks with 1003 // default track buffer size 1004 1005 uint32_t mNotificationFramesAct; // actual number of frames between each 1006 // notification callback, 1007 // at initial source sample rate 1008 bool mRefreshRemaining; // processAudioBuffer() should refresh 1009 // mRemainingFrames and mRetryOnPartialBuffer 1010 1011 // used for static track cbf and restoration 1012 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 1013 uint32_t mLoopStart; // last setLoop loopStart 1014 uint32_t mLoopEnd; // last setLoop loopEnd 1015 int32_t mLoopCountNotified; // the last loopCount notified by callback. 1016 // mLoopCountNotified counts down, matching 1017 // the remaining loop count for static track 1018 // playback. 1019 1020 // These are private to processAudioBuffer(), and are not protected by a lock 1021 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 1022 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 1023 uint32_t mObservedSequence; // last observed value of mSequence 1024 1025 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 1026 bool mMarkerReached; 1027 Modulo<uint32_t> mNewPosition; // in frames 1028 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 1029 1030 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 1031 // which is count of frames consumed by server, 1032 // reset by new IAudioTrack, 1033 // whether it is reset by stop() is TBD 1034 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 1035 // monotonically after new IAudioTrack, 1036 // and could be easily widened to uint64_t 1037 Modulo<uint32_t> mReleased; // count of frames released to server 1038 // but not necessarily consumed by server, 1039 // reset by stop() but continues monotonically 1040 // after new IAudioTrack to restore mPosition, 1041 // and could be easily widened to uint64_t 1042 int64_t mStartUs; // the start time after flush or stop. 1043 // only used for offloaded and direct tracks. 1044 1045 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 1046 bool mTimestampStartupGlitchReported; // reduce log spam 1047 bool mRetrogradeMotionReported; // reduce log spam 1048 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 1049 ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp 1050 1051 uint32_t mUnderrunCountOffset; // updated when restoring tracks 1052 1053 int64_t mFramesWritten; // total frames written. reset to zero after 1054 // the start() following stop(). It is not 1055 // changed after restoring the track or 1056 // after flush. 1057 int64_t mFramesWrittenServerOffset; // An offset to server frames due to 1058 // restoring AudioTrack, or stop/start. 1059 1060 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 1061 // be denied by client or server, such as 1062 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 1063 // held to read or write those bits reliably. 1064 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 1065 1066 bool mDoNotReconnect; 1067 1068 audio_session_t mSessionId; 1069 int mAuxEffectId; 1070 1071 mutable Mutex mLock; 1072 1073 int mPreviousPriority; // before start() 1074 SchedPolicy mPreviousSchedulingGroup; 1075 bool mAwaitBoost; // thread should wait for priority boost before running 1076 1077 // The proxy should only be referenced while a lock is held because the proxy isn't 1078 // multi-thread safe, especially the SingleStateQueue part of the proxy. 1079 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 1080 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 1081 // them around in case they are replaced during the obtainBuffer(). 1082 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 1083 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 1084 1085 bool mInUnderrun; // whether track is currently in underrun state 1086 uint32_t mPausedPosition; 1087 1088 // For Device Selection API 1089 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 1090 audio_port_handle_t mSelectedDeviceId; 1091 1092 private: 1093 class DeathNotifier : public IBinder::DeathRecipient { 1094 public: 1095 DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 1096 protected: 1097 virtual void binderDied(const wp<IBinder>& who); 1098 private: 1099 const wp<AudioTrack> mAudioTrack; 1100 }; 1101 1102 sp<DeathNotifier> mDeathNotifier; 1103 uint32_t mSequence; // incremented for each new IAudioTrack attempt 1104 int mClientUid; 1105 pid_t mClientPid; 1106 1107 sp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1108 }; 1109 1110 }; // namespace android 1111 1112 #endif // ANDROID_AUDIOTRACK_H 1113
