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 <binder/IMemory.h> 21 #include <cutils/sched_policy.h> 22 #include <media/AudioSystem.h> 23 #include <media/AudioTimestamp.h> 24 #include <media/AudioResamplerPublic.h> 25 #include <media/MediaMetricsItem.h> 26 #include <media/Modulo.h> 27 #include <media/VolumeShaper.h> 28 #include <utils/threads.h> 29 #include <android/content/AttributionSourceState.h> 30 31 #include <string> 32 33 #include "android/media/BnAudioTrackCallback.h" 34 #include "android/media/IAudioTrack.h" 35 #include "android/media/IAudioTrackCallback.h" 36 37 namespace android { 38 39 using content::AttributionSourceState; 40 41 // ---------------------------------------------------------------------------- 42 43 struct audio_track_cblk_t; 44 class AudioTrackClientProxy; 45 class StaticAudioTrackClientProxy; 46 47 // ---------------------------------------------------------------------------- 48 49 class AudioTrack : public AudioSystem::AudioDeviceCallback 50 { 51 public: 52 53 /* Events used by AudioTrack callback function (callback_t). 54 * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*. 55 */ 56 enum event_type { 57 EVENT_MORE_DATA = 0, // Request to write more data to buffer. 58 // This event only occurs for TRANSFER_CALLBACK. 59 // If this event is delivered but the callback handler 60 // does not want to write more data, the handler must 61 // ignore the event by setting frameCount to zero. 62 // This might occur, for example, if the application is 63 // waiting for source data or is at the end of stream. 64 // 65 // For data filling, it is preferred that the callback 66 // does not block and instead returns a short count on 67 // the amount of data actually delivered 68 // (or 0, if no data is currently available). 69 EVENT_UNDERRUN = 1, // Buffer underrun occurred. This will not occur for 70 // static tracks. 71 EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from 72 // loop start if loop count was not 0 for a static track. 73 EVENT_MARKER = 3, // Playback head is at the specified marker position 74 // (See setMarkerPosition()). 75 EVENT_NEW_POS = 4, // Playback head is at a new position 76 // (See setPositionUpdatePeriod()). 77 EVENT_BUFFER_END = 5, // Playback has completed for a static track. 78 EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and 79 // voluntary invalidation by mediaserver, or mediaserver crash. 80 EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played 81 // back (after stop is called) for an offloaded track. 82 #if 0 // FIXME not yet implemented 83 EVENT_NEW_TIMESTAMP = 8, // Delivered periodically and when there's a significant change 84 // in the mapping from frame position to presentation time. 85 // See AudioTimestamp for the information included with event. 86 #endif 87 EVENT_CAN_WRITE_MORE_DATA = 9,// Notification that more data can be given by write() 88 // This event only occurs for TRANSFER_SYNC_NOTIF_CALLBACK. 89 }; 90 91 /* Client should declare a Buffer and pass the address to obtainBuffer() 92 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 93 */ 94 95 class Buffer 96 { 97 public: 98 // FIXME use m prefix 99 size_t frameCount; // number of sample frames corresponding to size; 100 // on input to obtainBuffer() it is the number of frames desired, 101 // on output from obtainBuffer() it is the number of available 102 // [empty slots for] frames to be filled 103 // on input to releaseBuffer() it is currently ignored 104 105 size_t size; // input/output in bytes == frameCount * frameSize 106 // on input to obtainBuffer() it is ignored 107 // on output from obtainBuffer() it is the number of available 108 // [empty slots for] bytes to be filled, 109 // which is frameCount * frameSize 110 // on input to releaseBuffer() it is the number of bytes to 111 // release 112 // FIXME This is redundant with respect to frameCount. Consider 113 // removing size and making frameCount the primary field. 114 115 union { 116 void* raw; 117 int16_t* i16; // signed 16-bit 118 int8_t* i8; // unsigned 8-bit, offset by 0x80 119 }; // input to obtainBuffer(): unused, output: pointer to buffer 120 121 uint32_t sequence; // IAudioTrack instance sequence number, as of obtainBuffer(). 122 // It is set by obtainBuffer() and confirmed by releaseBuffer(). 123 // Not "user-serviceable". 124 // TODO Consider sp<IMemory> instead, or in addition to this. 125 }; 126 127 /* As a convenience, if a callback is supplied, a handler thread 128 * is automatically created with the appropriate priority. This thread 129 * invokes the callback when a new buffer becomes available or various conditions occur. 130 * Parameters: 131 * 132 * event: type of event notified (see enum AudioTrack::event_type). 133 * user: Pointer to context for use by the callback receiver. 134 * info: Pointer to optional parameter according to event type: 135 * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write 136 * more bytes than indicated by 'size' field and update 'size' if fewer bytes are 137 * written. 138 * - EVENT_UNDERRUN: unused. 139 * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. 140 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 141 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 142 * - EVENT_BUFFER_END: unused. 143 * - EVENT_NEW_IAUDIOTRACK: unused. 144 * - EVENT_STREAM_END: unused. 145 * - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp. 146 */ 147 148 typedef void (*callback_t)(int event, void* user, void *info); 149 150 /* Returns the minimum frame count required for the successful creation of 151 * an AudioTrack object. 152 * Returned status (from utils/Errors.h) can be: 153 * - NO_ERROR: successful operation 154 * - NO_INIT: audio server or audio hardware not initialized 155 * - BAD_VALUE: unsupported configuration 156 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 157 * and is undefined otherwise. 158 * FIXME This API assumes a route, and so should be deprecated. 159 */ 160 161 static status_t getMinFrameCount(size_t* frameCount, 162 audio_stream_type_t streamType, 163 uint32_t sampleRate); 164 165 /* Check if direct playback is possible for the given audio configuration and attributes. 166 * Return true if output is possible for the given parameters. Otherwise returns false. 167 */ 168 static bool isDirectOutputSupported(const audio_config_base_t& config, 169 const audio_attributes_t& attributes); 170 171 /* How data is transferred to AudioTrack 172 */ 173 enum transfer_type { 174 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 175 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 176 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 177 TRANSFER_SYNC, // synchronous write() 178 TRANSFER_SHARED, // shared memory 179 TRANSFER_SYNC_NOTIF_CALLBACK, // synchronous write(), notif EVENT_CAN_WRITE_MORE_DATA 180 }; 181 182 /* Constructs an uninitialized AudioTrack. No connection with 183 * AudioFlinger takes place. Use set() after this. 184 */ 185 AudioTrack(); 186 187 AudioTrack(const AttributionSourceState& attributionSourceState); 188 189 /* Creates an AudioTrack object and registers it with AudioFlinger. 190 * Once created, the track needs to be started before it can be used. 191 * Unspecified values are set to appropriate default values. 192 * 193 * Parameters: 194 * 195 * streamType: Select the type of audio stream this track is attached to 196 * (e.g. AUDIO_STREAM_MUSIC). 197 * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate. 198 * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set. 199 * 0 will not work with current policy implementation for direct output 200 * selection where an exact match is needed for sampling rate. 201 * format: Audio format. For mixed tracks, any PCM format supported by server is OK. 202 * For direct and offloaded tracks, the possible format(s) depends on the 203 * output sink. 204 * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. 205 * frameCount: Minimum size of track PCM buffer in frames. This defines the 206 * application's contribution to the 207 * latency of the track. The actual size selected by the AudioTrack could be 208 * larger if the requested size is not compatible with current audio HAL 209 * configuration. Zero means to use a default value. 210 * flags: See comments on audio_output_flags_t in <system/audio.h>. 211 * cbf: Callback function. If not null, this function is called periodically 212 * to provide new data in TRANSFER_CALLBACK mode 213 * and inform of marker, position updates, etc. 214 * user: Context for use by the callback receiver. 215 * notificationFrames: The callback function is called each time notificationFrames PCM 216 * frames have been consumed from track input buffer by server. 217 * Zero means to use a default value, which is typically: 218 * - fast tracks: HAL buffer size, even if track frameCount is larger 219 * - normal tracks: 1/2 of track frameCount 220 * A positive value means that many frames at initial source sample rate. 221 * A negative value for this parameter specifies the negative of the 222 * requested number of notifications (sub-buffers) in the entire buffer. 223 * For fast tracks, the FastMixer will process one sub-buffer at a time. 224 * The size of each sub-buffer is determined by the HAL. 225 * To get "double buffering", for example, one should pass -2. 226 * The minimum number of sub-buffers is 1 (expressed as -1), 227 * and the maximum number of sub-buffers is 8 (expressed as -8). 228 * Negative is only permitted for fast tracks, and if frameCount is zero. 229 * TODO It is ugly to overload a parameter in this way depending on 230 * whether it is positive, negative, or zero. Consider splitting apart. 231 * sessionId: Specific session ID, or zero to use default. 232 * transferType: How data is transferred to AudioTrack. 233 * offloadInfo: If not NULL, provides offload parameters for 234 * AudioSystem::getOutputForAttr(). 235 * attributionSource: The attribution source of the app which initially requested this 236 * AudioTrack. 237 * Includes the UID and PID for power management tracking, or -1 for 238 * current user/process ID, plus the package name. 239 * pAttributes: If not NULL, supersedes streamType for use case selection. 240 * doNotReconnect: If set to true, AudioTrack won't automatically recreate the IAudioTrack 241 binder to AudioFlinger. 242 It will return an error instead. The application will recreate 243 the track based on offloading or different channel configuration, etc. 244 * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow 245 * maxRequiredSpeed playback. Values less than 1.0f and greater than 246 * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks 247 * and direct or offloaded tracks, this parameter is ignored. 248 * selectedDeviceId: Selected device id of the app which initially requested the AudioTrack 249 * to open with a specific device. 250 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 251 */ 252 253 AudioTrack( audio_stream_type_t streamType, 254 uint32_t sampleRate, 255 audio_format_t format, 256 audio_channel_mask_t channelMask, 257 size_t frameCount = 0, 258 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 259 callback_t cbf = NULL, 260 void* user = NULL, 261 int32_t notificationFrames = 0, 262 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 263 transfer_type transferType = TRANSFER_DEFAULT, 264 const audio_offload_info_t *offloadInfo = NULL, 265 const AttributionSourceState& attributionSource = 266 AttributionSourceState(), 267 const audio_attributes_t* pAttributes = NULL, 268 bool doNotReconnect = false, 269 float maxRequiredSpeed = 1.0f, 270 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 271 272 /* Creates an audio track and registers it with AudioFlinger. 273 * With this constructor, the track is configured for static buffer mode. 274 * Data to be rendered is passed in a shared memory buffer 275 * identified by the argument sharedBuffer, which should be non-0. 276 * If sharedBuffer is zero, this constructor is equivalent to the previous constructor 277 * but without the ability to specify a non-zero value for the frameCount parameter. 278 * The memory should be initialized to the desired data before calling start(). 279 * The write() method is not supported in this case. 280 * It is recommended to pass a callback function to be notified of playback end by an 281 * EVENT_UNDERRUN event. 282 */ 283 284 AudioTrack( audio_stream_type_t streamType, 285 uint32_t sampleRate, 286 audio_format_t format, 287 audio_channel_mask_t channelMask, 288 const sp<IMemory>& sharedBuffer, 289 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 290 callback_t cbf = NULL, 291 void* user = NULL, 292 int32_t notificationFrames = 0, 293 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 294 transfer_type transferType = TRANSFER_DEFAULT, 295 const audio_offload_info_t *offloadInfo = NULL, 296 const AttributionSourceState& attributionSource = 297 AttributionSourceState(), 298 const audio_attributes_t* pAttributes = NULL, 299 bool doNotReconnect = false, 300 float maxRequiredSpeed = 1.0f); 301 302 /* Terminates the AudioTrack and unregisters it from AudioFlinger. 303 * Also destroys all resources associated with the AudioTrack. 304 */ 305 protected: 306 virtual ~AudioTrack(); 307 public: 308 309 /* Initialize an AudioTrack that was created using the AudioTrack() constructor. 310 * Don't call set() more than once, or after the AudioTrack() constructors that take parameters. 311 * set() is not multi-thread safe. 312 * Returned status (from utils/Errors.h) can be: 313 * - NO_ERROR: successful initialization 314 * - INVALID_OPERATION: AudioTrack is already initialized 315 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 316 * - NO_INIT: audio server or audio hardware not initialized 317 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack. 318 * If sharedBuffer is non-0, the frameCount parameter is ignored and 319 * replaced by the shared buffer's total allocated size in frame units. 320 * 321 * Parameters not listed in the AudioTrack constructors above: 322 * 323 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 324 * Only set to true when AudioTrack object is used for a java android.media.AudioTrack 325 * in its JNI code. 326 * 327 * Internal state post condition: 328 * (mStreamType == AUDIO_STREAM_DEFAULT) implies this AudioTrack has valid attributes 329 */ 330 status_t set(audio_stream_type_t streamType, 331 uint32_t sampleRate, 332 audio_format_t format, 333 audio_channel_mask_t channelMask, 334 size_t frameCount = 0, 335 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 336 callback_t cbf = NULL, 337 void* user = NULL, 338 int32_t notificationFrames = 0, 339 const sp<IMemory>& sharedBuffer = 0, 340 bool threadCanCallJava = false, 341 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 342 transfer_type transferType = TRANSFER_DEFAULT, 343 const audio_offload_info_t *offloadInfo = NULL, 344 const AttributionSourceState& attributionSource = 345 AttributionSourceState(), 346 const audio_attributes_t* pAttributes = NULL, 347 bool doNotReconnect = false, 348 float maxRequiredSpeed = 1.0f, 349 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 350 // FIXME(b/169889714): Vendor code depends on the old method signature at link time 351 status_t set(audio_stream_type_t streamType, 352 uint32_t sampleRate, 353 audio_format_t format, 354 uint32_t channelMask, 355 size_t frameCount = 0, 356 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 357 callback_t cbf = NULL, 358 void* user = NULL, 359 int32_t notificationFrames = 0, 360 const sp<IMemory>& sharedBuffer = 0, 361 bool threadCanCallJava = false, 362 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 363 transfer_type transferType = TRANSFER_DEFAULT, 364 const audio_offload_info_t *offloadInfo = NULL, 365 uid_t uid = AUDIO_UID_INVALID, 366 pid_t pid = -1, 367 const audio_attributes_t* pAttributes = NULL, 368 bool doNotReconnect = false, 369 float maxRequiredSpeed = 1.0f, 370 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 371 372 /* Result of constructing the AudioTrack. This must be checked for successful initialization 373 * before using any AudioTrack API (except for set()), because using 374 * an uninitialized AudioTrack produces undefined results. 375 * See set() method above for possible return codes. 376 */ initCheck()377 status_t initCheck() const { return mStatus; } 378 379 /* Returns this track's estimated latency in milliseconds. 380 * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) 381 * and audio hardware driver. 382 */ 383 uint32_t latency(); 384 385 /* Returns the number of application-level buffer underruns 386 * since the AudioTrack was created. 387 */ 388 uint32_t getUnderrunCount() const; 389 390 /* getters, see constructors and set() */ 391 392 audio_stream_type_t streamType() const; format()393 audio_format_t format() const { return mFormat; } 394 395 /* Return frame size in bytes, which for linear PCM is 396 * channelCount * (bit depth per channel / 8). 397 * channelCount is determined from channelMask, and bit depth comes from format. 398 * For non-linear formats, the frame size is typically 1 byte. 399 */ frameSize()400 size_t frameSize() const { return mFrameSize; } 401 channelCount()402 uint32_t channelCount() const { return mChannelCount; } frameCount()403 size_t frameCount() const { return mFrameCount; } 404 405 /* 406 * Return the period of the notification callback in frames. 407 * This value is set when the AudioTrack is constructed. 408 * It can be modified if the AudioTrack is rerouted. 409 */ getNotificationPeriodInFrames()410 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 411 412 /* Return effective size of audio buffer that an application writes to 413 * or a negative error if the track is uninitialized. 414 */ 415 ssize_t getBufferSizeInFrames(); 416 417 /* Returns the buffer duration in microseconds at current playback rate. 418 */ 419 status_t getBufferDurationInUs(int64_t *duration); 420 421 /* Set the effective size of audio buffer that an application writes to. 422 * This is used to determine the amount of available room in the buffer, 423 * which determines when a write will block. 424 * This allows an application to raise and lower the audio latency. 425 * The requested size may be adjusted so that it is 426 * greater or equal to the absolute minimum and 427 * less than or equal to the getBufferCapacityInFrames(). 428 * It may also be adjusted slightly for internal reasons. 429 * 430 * Return the final size or a negative error if the track is unitialized 431 * or does not support variable sizes. 432 */ 433 ssize_t setBufferSizeInFrames(size_t size); 434 435 /* Returns the start threshold on the buffer for audio streaming 436 * or a negative value if the AudioTrack is not initialized. 437 */ 438 ssize_t getStartThresholdInFrames() const; 439 440 /* Sets the start threshold in frames on the buffer for audio streaming. 441 * 442 * May be clamped internally. Returns the actual value set, or a negative 443 * value if the AudioTrack is not initialized or if the input 444 * is zero or greater than INT_MAX. 445 */ 446 ssize_t setStartThresholdInFrames(size_t startThresholdInFrames); 447 448 /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */ sharedBuffer()449 sp<IMemory> sharedBuffer() const { return mSharedBuffer; } 450 451 /* 452 * return metrics information for the current track. 453 */ 454 status_t getMetrics(mediametrics::Item * &item); 455 456 /* 457 * Set name of API that is using this object. 458 * For example "aaudio" or "opensles". 459 * This may be logged or reported as part of MediaMetrics. 460 */ setCallerName(const std::string & name)461 void setCallerName(const std::string &name) { 462 mCallerName = name; 463 } 464 getCallerName()465 std::string getCallerName() const { 466 return mCallerName; 467 }; 468 469 /* After it's created the track is not active. Call start() to 470 * make it active. If set, the callback will start being called. 471 * If the track was previously paused, volume is ramped up over the first mix buffer. 472 */ 473 status_t start(); 474 475 /* Stop a track. 476 * In static buffer mode, the track is stopped immediately. 477 * In streaming mode, the callback will cease being called. Note that obtainBuffer() still 478 * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK. 479 * In streaming mode the stop does not occur immediately: any data remaining in the buffer 480 * is first drained, mixed, and output, and only then is the track marked as stopped. 481 */ 482 void stop(); 483 bool stopped() const; 484 485 /* Call stop() and then wait for all of the callbacks to return. 486 * It is safe to call this if stop() or pause() has already been called. 487 * 488 * This function is called from the destructor. But since AudioTrack 489 * is ref counted, the destructor may be called later than desired. 490 * This can be called explicitly as part of closing an AudioTrack 491 * if you want to be certain that callbacks have completely finished. 492 * 493 * This is not thread safe and should only be called from one thread, 494 * ideally as the AudioTrack is being closed. 495 */ 496 void stopAndJoinCallbacks(); 497 498 /* Flush a stopped or paused track. All previously buffered data is discarded immediately. 499 * This has the effect of draining the buffers without mixing or output. 500 * Flush is intended for streaming mode, for example before switching to non-contiguous content. 501 * This function is a no-op if the track is not stopped or paused, or uses a static buffer. 502 */ 503 void flush(); 504 505 /* Pause a track. After pause, the callback will cease being called and 506 * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works 507 * and will fill up buffers until the pool is exhausted. 508 * Volume is ramped down over the next mix buffer following the pause request, 509 * and then the track is marked as paused. It can be resumed with ramp up by start(). 510 */ 511 void pause(); 512 513 /* Set volume for this track, mostly used for games' sound effects 514 * left and right volumes. Levels must be >= 0.0 and <= 1.0. 515 * This is the older API. New applications should use setVolume(float) when possible. 516 */ 517 status_t setVolume(float left, float right); 518 519 /* Set volume for all channels. This is the preferred API for new applications, 520 * especially for multi-channel content. 521 */ 522 status_t setVolume(float volume); 523 524 /* Set the send level for this track. An auxiliary effect should be attached 525 * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0. 526 */ 527 status_t setAuxEffectSendLevel(float level); 528 void getAuxEffectSendLevel(float* level) const; 529 530 /* Set source sample rate for this track in Hz, mostly used for games' sound effects. 531 * Zero is not permitted. 532 */ 533 status_t setSampleRate(uint32_t sampleRate); 534 535 /* Return current source sample rate in Hz. 536 * If specified as zero in constructor or set(), this will be the sink sample rate. 537 */ 538 uint32_t getSampleRate() const; 539 540 /* Return the original source sample rate in Hz. This corresponds to the sample rate 541 * if playback rate had normal speed and pitch. 542 */ 543 uint32_t getOriginalSampleRate() const; 544 545 /* Sets the Dual Mono mode presentation on the output device. */ 546 status_t setDualMonoMode(audio_dual_mono_mode_t mode); 547 548 /* Returns the Dual Mono mode presentation setting. */ 549 status_t getDualMonoMode(audio_dual_mono_mode_t* mode) const; 550 551 /* Sets the Audio Description Mix level in dB. */ 552 status_t setAudioDescriptionMixLevel(float leveldB); 553 554 /* Returns the Audio Description Mix level in dB. */ 555 status_t getAudioDescriptionMixLevel(float* leveldB) const; 556 557 /* Set source playback rate for timestretch 558 * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster 559 * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch 560 * 561 * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX 562 * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX 563 * 564 * Speed increases the playback rate of media, but does not alter pitch. 565 * Pitch increases the "tonal frequency" of media, but does not affect the playback rate. 566 */ 567 status_t setPlaybackRate(const AudioPlaybackRate &playbackRate); 568 569 /* Return current playback rate */ 570 const AudioPlaybackRate& getPlaybackRate(); 571 572 /* Enables looping and sets the start and end points of looping. 573 * Only supported for static buffer mode. 574 * 575 * Parameters: 576 * 577 * loopStart: loop start in frames relative to start of buffer. 578 * loopEnd: loop end in frames relative to start of buffer. 579 * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any 580 * pending or active loop. loopCount == -1 means infinite looping. 581 * 582 * For proper operation the following condition must be respected: 583 * loopCount != 0 implies 0 <= loopStart < loopEnd <= frameCount(). 584 * 585 * If the loop period (loopEnd - loopStart) is too small for the implementation to support, 586 * setLoop() will return BAD_VALUE. loopCount must be >= -1. 587 * 588 */ 589 status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); 590 591 /* Sets marker position. When playback reaches the number of frames specified, a callback with 592 * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker 593 * notification callback. To set a marker at a position which would compute as 0, 594 * a workaround is to set the marker at a nearby position such as ~0 or 1. 595 * If the AudioTrack has been opened with no callback function associated, the operation will 596 * fail. 597 * 598 * Parameters: 599 * 600 * marker: marker position expressed in wrapping (overflow) frame units, 601 * like the return value of getPosition(). 602 * 603 * Returned status (from utils/Errors.h) can be: 604 * - NO_ERROR: successful operation 605 * - INVALID_OPERATION: the AudioTrack has no callback installed. 606 */ 607 status_t setMarkerPosition(uint32_t marker); 608 status_t getMarkerPosition(uint32_t *marker) const; 609 610 /* Sets position update period. Every time the number of frames specified has been played, 611 * a callback with event type EVENT_NEW_POS is called. 612 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 613 * callback. 614 * If the AudioTrack has been opened with no callback function associated, the operation will 615 * fail. 616 * Extremely small values may be rounded up to a value the implementation can support. 617 * 618 * Parameters: 619 * 620 * updatePeriod: position update notification period expressed in frames. 621 * 622 * Returned status (from utils/Errors.h) can be: 623 * - NO_ERROR: successful operation 624 * - INVALID_OPERATION: the AudioTrack has no callback installed. 625 */ 626 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 627 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 628 629 /* Sets playback head position. 630 * Only supported for static buffer mode. 631 * 632 * Parameters: 633 * 634 * position: New playback head position in frames relative to start of buffer. 635 * 0 <= position <= frameCount(). Note that end of buffer is permitted, 636 * but will result in an immediate underrun if started. 637 * 638 * Returned status (from utils/Errors.h) can be: 639 * - NO_ERROR: successful operation 640 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 641 * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack 642 * buffer 643 */ 644 status_t setPosition(uint32_t position); 645 646 /* Return the total number of frames played since playback start. 647 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 648 * It is reset to zero by flush(), reload(), and stop(). 649 * 650 * Parameters: 651 * 652 * position: Address where to return play head position. 653 * 654 * Returned status (from utils/Errors.h) can be: 655 * - NO_ERROR: successful operation 656 * - BAD_VALUE: position is NULL 657 */ 658 status_t getPosition(uint32_t *position); 659 660 /* For static buffer mode only, this returns the current playback position in frames 661 * relative to start of buffer. It is analogous to the position units used by 662 * setLoop() and setPosition(). After underrun, the position will be at end of buffer. 663 */ 664 status_t getBufferPosition(uint32_t *position); 665 666 /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids 667 * rewriting the buffer before restarting playback after a stop. 668 * This method must be called with the AudioTrack in paused or stopped state. 669 * Not allowed in streaming mode. 670 * 671 * Returned status (from utils/Errors.h) can be: 672 * - NO_ERROR: successful operation 673 * - INVALID_OPERATION: the AudioTrack is not stopped or paused, or is streaming mode. 674 */ 675 status_t reload(); 676 677 /** 678 * @param transferType 679 * @return text string that matches the enum name 680 */ 681 static const char * convertTransferToText(transfer_type transferType); 682 683 /* Returns a handle on the audio output used by this AudioTrack. 684 * 685 * Parameters: 686 * none. 687 * 688 * Returned value: 689 * handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the 690 * track needed to be re-created but that failed 691 */ 692 audio_io_handle_t getOutput() const; 693 694 /* Selects the audio device to use for output of this AudioTrack. A value of 695 * AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 696 * 697 * Parameters: 698 * The device ID of the selected device (as returned by the AudioDevicesManager API). 699 * 700 * Returned value: 701 * - NO_ERROR: successful operation 702 * TODO: what else can happen here? 703 */ 704 status_t setOutputDevice(audio_port_handle_t deviceId); 705 706 /* Returns the ID of the audio device selected for this AudioTrack. 707 * A value of AUDIO_PORT_HANDLE_NONE indicates default (AudioPolicyManager) routing. 708 * 709 * Parameters: 710 * none. 711 */ 712 audio_port_handle_t getOutputDevice(); 713 714 /* Returns the ID of the audio device actually used by the output to which this AudioTrack is 715 * attached. 716 * When the AudioTrack is inactive, the device ID returned can be either: 717 * - AUDIO_PORT_HANDLE_NONE if the AudioTrack is not attached to any output. 718 * - The device ID used before paused or stopped. 719 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioTrack 720 * has not been started yet. 721 * 722 * Parameters: 723 * none. 724 */ 725 audio_port_handle_t getRoutedDeviceId(); 726 727 /* Returns the unique session ID associated with this track. 728 * 729 * Parameters: 730 * none. 731 * 732 * Returned value: 733 * AudioTrack session ID. 734 */ getSessionId()735 audio_session_t getSessionId() const { return mSessionId; } 736 737 /* Attach track auxiliary output to specified effect. Use effectId = 0 738 * to detach track from effect. 739 * 740 * Parameters: 741 * 742 * effectId: effectId obtained from AudioEffect::id(). 743 * 744 * Returned status (from utils/Errors.h) can be: 745 * - NO_ERROR: successful operation 746 * - INVALID_OPERATION: the effect is not an auxiliary effect. 747 * - BAD_VALUE: The specified effect ID is invalid 748 */ 749 status_t attachAuxEffect(int effectId); 750 751 /* Public API for TRANSFER_OBTAIN mode. 752 * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames. 753 * After filling these slots with data, the caller should release them with releaseBuffer(). 754 * If the track buffer is not full, obtainBuffer() returns as many contiguous 755 * [empty slots for] frames as are available immediately. 756 * 757 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 758 * additional non-contiguous frames that are predicted to be available immediately, 759 * if the client were to release the first frames and then call obtainBuffer() again. 760 * This value is only a prediction, and needs to be confirmed. 761 * It will be set to zero for an error return. 762 * 763 * If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK 764 * regardless of the value of waitCount. 765 * If the track buffer is full and track is not stopped, obtainBuffer() blocks with a 766 * maximum timeout based on waitCount; see chart below. 767 * Buffers will be returned until the pool 768 * is exhausted, at which point obtainBuffer() will either block 769 * or return WOULD_BLOCK depending on the value of the "waitCount" 770 * parameter. 771 * 772 * Interpretation of waitCount: 773 * +n limits wait time to n * WAIT_PERIOD_MS, 774 * -1 causes an (almost) infinite wait time, 775 * 0 non-blocking. 776 * 777 * Buffer fields 778 * On entry: 779 * frameCount number of [empty slots for] frames requested 780 * size ignored 781 * raw ignored 782 * sequence ignored 783 * After error return: 784 * frameCount 0 785 * size 0 786 * raw undefined 787 * sequence undefined 788 * After successful return: 789 * frameCount actual number of [empty slots for] frames available, <= number requested 790 * size actual number of bytes available 791 * raw pointer to the buffer 792 * sequence IAudioTrack instance sequence number, as of obtainBuffer() 793 */ 794 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 795 size_t *nonContig = NULL); 796 797 private: 798 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 799 * additional non-contiguous frames that are predicted to be available immediately, 800 * if the client were to release the first frames and then call obtainBuffer() again. 801 * This value is only a prediction, and needs to be confirmed. 802 * It will be set to zero for an error return. 803 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 804 * in case the requested amount of frames is in two or more non-contiguous regions. 805 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 806 */ 807 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 808 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 809 public: 810 811 /* Public API for TRANSFER_OBTAIN mode. 812 * Release a filled buffer of frames for AudioFlinger to process. 813 * 814 * Buffer fields: 815 * frameCount currently ignored but recommend to set to actual number of frames filled 816 * size actual number of bytes filled, must be multiple of frameSize 817 * raw ignored 818 */ 819 void releaseBuffer(const Buffer* audioBuffer); 820 821 /* As a convenience we provide a write() interface to the audio buffer. 822 * Input parameter 'size' is in byte units. 823 * This is implemented on top of obtainBuffer/releaseBuffer. For best 824 * performance use callbacks. Returns actual number of bytes written >= 0, 825 * or one of the following negative status codes: 826 * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode 827 * BAD_VALUE size is invalid 828 * WOULD_BLOCK when obtainBuffer() returns same, or 829 * AudioTrack was stopped during the write 830 * DEAD_OBJECT when AudioFlinger dies or the output device changes and 831 * the track cannot be automatically restored. 832 * The application needs to recreate the AudioTrack 833 * because the audio device changed or AudioFlinger died. 834 * This typically occurs for direct or offload tracks 835 * or if mDoNotReconnect is true. 836 * or any other error code returned by IAudioTrack::start() or restoreTrack_l(). 837 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 838 * false for the method to return immediately without waiting to try multiple times to write 839 * the full content of the buffer. 840 */ 841 ssize_t write(const void* buffer, size_t size, bool blocking = true); 842 843 /* 844 * Dumps the state of an audio track. 845 * Not a general-purpose API; intended only for use by media player service to dump its tracks. 846 */ 847 status_t dump(int fd, const Vector<String16>& args) const; 848 849 /* 850 * Return the total number of frames which AudioFlinger desired but were unavailable, 851 * and thus which resulted in an underrun. Reset to zero by stop(). 852 */ 853 uint32_t getUnderrunFrames() const; 854 855 /* Get the flags */ getFlags()856 audio_output_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 857 858 /* Set parameters - only possible when using direct output */ 859 status_t setParameters(const String8& keyValuePairs); 860 861 /* Sets the volume shaper object */ 862 media::VolumeShaper::Status applyVolumeShaper( 863 const sp<media::VolumeShaper::Configuration>& configuration, 864 const sp<media::VolumeShaper::Operation>& operation); 865 866 /* Gets the volume shaper state */ 867 sp<media::VolumeShaper::State> getVolumeShaperState(int id); 868 869 /* Selects the presentation (if available) */ 870 status_t selectPresentation(int presentationId, int programId); 871 872 /* Get parameters */ 873 String8 getParameters(const String8& keys); 874 875 /* Poll for a timestamp on demand. 876 * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs, 877 * or if you need to get the most recent timestamp outside of the event callback handler. 878 * Caution: calling this method too often may be inefficient; 879 * if you need a high resolution mapping between frame position and presentation time, 880 * consider implementing that at application level, based on the low resolution timestamps. 881 * Returns NO_ERROR if timestamp is valid. 882 * WOULD_BLOCK if called in STOPPED or FLUSHED state, or if called immediately after 883 * start/ACTIVE, when the number of frames consumed is less than the 884 * overall hardware latency to physical output. In WOULD_BLOCK cases, 885 * one might poll again, or use getPosition(), or use 0 position and 886 * current time for the timestamp. 887 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 888 * the track cannot be automatically restored. 889 * The application needs to recreate the AudioTrack 890 * because the audio device changed or AudioFlinger died. 891 * This typically occurs for direct or offload tracks 892 * or if mDoNotReconnect is true. 893 * INVALID_OPERATION wrong state, or some other error. 894 * 895 * The timestamp parameter is undefined on return, if status is not NO_ERROR. 896 */ 897 status_t getTimestamp(AudioTimestamp& timestamp); 898 private: 899 status_t getTimestamp_l(AudioTimestamp& timestamp); 900 public: 901 902 /* Return the extended timestamp, with additional timebase info and improved drain behavior. 903 * 904 * This is similar to the AudioTrack.java API: 905 * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase) 906 * 907 * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method 908 * 909 * 1. stop() by itself does not reset the frame position. 910 * A following start() resets the frame position to 0. 911 * 2. flush() by itself does not reset the frame position. 912 * The frame position advances by the number of frames flushed, 913 * when the first frame after flush reaches the audio sink. 914 * 3. BOOTTIME clock offsets are provided to help synchronize with 915 * non-audio streams, e.g. sensor data. 916 * 4. Position is returned with 64 bits of resolution. 917 * 918 * Parameters: 919 * timestamp: A pointer to the caller allocated ExtendedTimestamp. 920 * 921 * Returns NO_ERROR on success; timestamp is filled with valid data. 922 * BAD_VALUE if timestamp is NULL. 923 * WOULD_BLOCK if called immediately after start() when the number 924 * of frames consumed is less than the 925 * overall hardware latency to physical output. In WOULD_BLOCK cases, 926 * one might poll again, or use getPosition(), or use 0 position and 927 * current time for the timestamp. 928 * If WOULD_BLOCK is returned, the timestamp is still 929 * modified with the LOCATION_CLIENT portion filled. 930 * DEAD_OBJECT if AudioFlinger dies or the output device changes and 931 * the track cannot be automatically restored. 932 * The application needs to recreate the AudioTrack 933 * because the audio device changed or AudioFlinger died. 934 * This typically occurs for direct or offloaded tracks 935 * or if mDoNotReconnect is true. 936 * INVALID_OPERATION if called on a offloaded or direct track. 937 * Use getTimestamp(AudioTimestamp& timestamp) instead. 938 */ 939 status_t getTimestamp(ExtendedTimestamp *timestamp); 940 private: 941 status_t getTimestamp_l(ExtendedTimestamp *timestamp); 942 public: 943 944 /* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this 945 * AudioTrack is routed is updated. 946 * Replaces any previously installed callback. 947 * Parameters: 948 * callback: The callback interface 949 * Returns NO_ERROR if successful. 950 * INVALID_OPERATION if the same callback is already installed. 951 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 952 * BAD_VALUE if the callback is NULL 953 */ 954 status_t addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback); 955 956 /* remove an AudioDeviceCallback. 957 * Parameters: 958 * callback: The callback interface 959 * Returns NO_ERROR if successful. 960 * INVALID_OPERATION if the callback is not installed 961 * BAD_VALUE if the callback is NULL 962 */ 963 status_t removeAudioDeviceCallback( 964 const sp<AudioSystem::AudioDeviceCallback>& callback); 965 966 // AudioSystem::AudioDeviceCallback> virtuals 967 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 968 audio_port_handle_t deviceId); 969 970 /* Obtain the pending duration in milliseconds for playback of pure PCM 971 * (mixable without embedded timing) data remaining in AudioTrack. 972 * 973 * This is used to estimate the drain time for the client-server buffer 974 * so the choice of ExtendedTimestamp::LOCATION_SERVER is default. 975 * One may optionally request to find the duration to play through the HAL 976 * by specifying a location ExtendedTimestamp::LOCATION_KERNEL; however, 977 * INVALID_OPERATION may be returned if the kernel location is unavailable. 978 * 979 * Returns NO_ERROR if successful. 980 * INVALID_OPERATION if ExtendedTimestamp::LOCATION_KERNEL cannot be obtained 981 * or the AudioTrack does not contain pure PCM data. 982 * BAD_VALUE if msec is nullptr or location is invalid. 983 */ 984 status_t pendingDuration(int32_t *msec, 985 ExtendedTimestamp::Location location = ExtendedTimestamp::LOCATION_SERVER); 986 987 /* hasStarted() is used to determine if audio is now audible at the device after 988 * a start() command. The underlying implementation checks a nonzero timestamp position 989 * or increment for the audible assumption. 990 * 991 * hasStarted() returns true if the track has been started() and audio is audible 992 * and no subsequent pause() or flush() has been called. Immediately after pause() or 993 * flush() hasStarted() will return false. 994 * 995 * If stop() has been called, hasStarted() will return true if audio is still being 996 * delivered or has finished delivery (even if no audio was written) for both offloaded 997 * and normal tracks. This property removes a race condition in checking hasStarted() 998 * for very short clips, where stop() must be called to finish drain. 999 * 1000 * In all cases, hasStarted() may turn false briefly after a subsequent start() is called 1001 * until audio becomes audible again. 1002 */ 1003 bool hasStarted(); // not const 1004 isPlaying()1005 bool isPlaying() { 1006 AutoMutex lock(mLock); 1007 return mState == STATE_ACTIVE || mState == STATE_STOPPING; 1008 } 1009 1010 /* Get the unique port ID assigned to this AudioTrack instance by audio policy manager. 1011 * The ID is unique across all audioserver clients and can change during the life cycle 1012 * of a given AudioTrack instance if the connection to audioserver is restored. 1013 */ getPortId()1014 audio_port_handle_t getPortId() const { return mPortId; }; 1015 1016 /* Sets the LogSessionId field which is used for metrics association of 1017 * this object with other objects. A nullptr or empty string clears 1018 * the logSessionId. 1019 */ 1020 void setLogSessionId(const char *logSessionId); 1021 1022 /* Sets the playerIId field to associate the AudioTrack with an interface managed by 1023 * AudioService. 1024 * 1025 * If this value is not set, then the playerIId is reported as -1 1026 * (not associated with an AudioService player interface). 1027 * 1028 * For metrics purposes, we keep the playerIId association in the native 1029 * client AudioTrack to improve the robustness under track restoration. 1030 */ 1031 void setPlayerIId(int playerIId); 1032 setAudioTrackCallback(const sp<media::IAudioTrackCallback> & callback)1033 void setAudioTrackCallback(const sp<media::IAudioTrackCallback>& callback) { 1034 mAudioTrackCallback->setAudioTrackCallback(callback); 1035 } 1036 1037 protected: 1038 /* copying audio tracks is not allowed */ 1039 AudioTrack(const AudioTrack& other); 1040 AudioTrack& operator = (const AudioTrack& other); 1041 1042 /* a small internal class to handle the callback */ 1043 class AudioTrackThread : public Thread 1044 { 1045 public: 1046 explicit AudioTrackThread(AudioTrack& receiver); 1047 1048 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 1049 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 1050 virtual void requestExit(); 1051 1052 void pause(); // suspend thread from execution at next loop boundary 1053 void resume(); // allow thread to execute, if not requested to exit 1054 void wake(); // wake to handle changed notification conditions. 1055 1056 private: 1057 void pauseInternal(nsecs_t ns = 0LL); 1058 // like pause(), but only used internally within thread 1059 1060 friend class AudioTrack; 1061 virtual bool threadLoop(); 1062 AudioTrack& mReceiver; 1063 virtual ~AudioTrackThread(); 1064 Mutex mMyLock; // Thread::mLock is private 1065 Condition mMyCond; // Thread::mThreadExitedCondition is private 1066 bool mPaused; // whether thread is requested to pause at next loop entry 1067 bool mPausedInt; // whether thread internally requests pause 1068 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 1069 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 1070 // to processAudioBuffer() as state may have changed 1071 // since pause time calculated. 1072 }; 1073 1074 // body of AudioTrackThread::threadLoop() 1075 // returns the maximum amount of time before we would like to run again, where: 1076 // 0 immediately 1077 // > 0 no later than this many nanoseconds from now 1078 // NS_WHENEVER still active but no particular deadline 1079 // NS_INACTIVE inactive so don't run again until re-started 1080 // NS_NEVER never again 1081 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 1082 nsecs_t processAudioBuffer(); 1083 1084 // caller must hold lock on mLock for all _l methods 1085 1086 void updateLatency_l(); // updates mAfLatency and mLatency from AudioSystem cache 1087 1088 status_t createTrack_l(); 1089 1090 // can only be called when mState != STATE_ACTIVE 1091 void flush_l(); 1092 1093 void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount); 1094 1095 // FIXME enum is faster than strcmp() for parameter 'from' 1096 status_t restoreTrack_l(const char *from); 1097 1098 uint32_t getUnderrunCount_l() const; 1099 1100 bool isOffloaded() const; 1101 bool isDirect() const; 1102 bool isOffloadedOrDirect() const; 1103 isOffloaded_l()1104 bool isOffloaded_l() const 1105 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } 1106 isOffloadedOrDirect_l()1107 bool isOffloadedOrDirect_l() const 1108 { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD| 1109 AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } 1110 isDirect_l()1111 bool isDirect_l() const 1112 { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } 1113 1114 // pure pcm data is mixable (which excludes HW_AV_SYNC, with embedded timing) isPurePcmData_l()1115 bool isPurePcmData_l() const 1116 { return audio_is_linear_pcm(mFormat) 1117 && (mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) == 0; } 1118 1119 // increment mPosition by the delta of mServer, and return new value of mPosition 1120 Modulo<uint32_t> updateAndGetPosition_l(); 1121 1122 // check sample rate and speed is compatible with AudioTrack 1123 bool isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed); 1124 1125 void restartIfDisabled(); 1126 1127 void updateRoutedDeviceId_l(); 1128 1129 /* Sets the Dual Mono mode presentation on the output device. */ 1130 status_t setDualMonoMode_l(audio_dual_mono_mode_t mode); 1131 1132 /* Sets the Audio Description Mix level in dB. */ 1133 status_t setAudioDescriptionMixLevel_l(float leveldB); 1134 1135 // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0 1136 sp<media::IAudioTrack> mAudioTrack; 1137 sp<IMemory> mCblkMemory; 1138 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 1139 audio_io_handle_t mOutput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getOutputForAttr() 1140 1141 sp<AudioTrackThread> mAudioTrackThread; 1142 bool mThreadCanCallJava; 1143 1144 float mVolume[2]; 1145 float mSendLevel; 1146 mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it 1147 uint32_t mOriginalSampleRate; 1148 AudioPlaybackRate mPlaybackRate; 1149 float mMaxRequiredSpeed; // use PCM buffer size to allow this speed 1150 1151 // Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client. 1152 // This allocated buffer size is maintained by the proxy. 1153 size_t mFrameCount; // maximum size of buffer 1154 1155 size_t mReqFrameCount; // frame count to request the first or next time 1156 // a new IAudioTrack is needed, non-decreasing 1157 1158 // The following AudioFlinger server-side values are cached in createAudioTrack_l(). 1159 // These values can be used for informational purposes until the track is invalidated, 1160 // whereupon restoreTrack_l() calls createTrack_l() to update the values. 1161 uint32_t mAfLatency; // AudioFlinger latency in ms 1162 size_t mAfFrameCount; // AudioFlinger frame count 1163 uint32_t mAfSampleRate; // AudioFlinger sample rate 1164 1165 // constant after constructor or set() 1166 audio_format_t mFormat; // as requested by client, not forced to 16-bit 1167 // mOriginalStreamType == AUDIO_STREAM_DEFAULT implies this AudioTrack has valid attributes 1168 audio_stream_type_t mOriginalStreamType = AUDIO_STREAM_DEFAULT; 1169 audio_stream_type_t mStreamType = AUDIO_STREAM_DEFAULT; 1170 uint32_t mChannelCount; 1171 audio_channel_mask_t mChannelMask; 1172 sp<IMemory> mSharedBuffer; 1173 transfer_type mTransfer; 1174 audio_offload_info_t mOffloadInfoCopy; 1175 const audio_offload_info_t* mOffloadInfo; 1176 audio_attributes_t mAttributes; 1177 1178 size_t mFrameSize; // frame size in bytes 1179 1180 status_t mStatus; 1181 1182 // can change dynamically when IAudioTrack invalidated 1183 uint32_t mLatency; // in ms 1184 1185 // Indicates the current track state. Protected by mLock. 1186 enum State { 1187 STATE_ACTIVE, 1188 STATE_STOPPED, 1189 STATE_PAUSED, 1190 STATE_PAUSED_STOPPING, 1191 STATE_FLUSHED, 1192 STATE_STOPPING, 1193 } mState; 1194 stateToString(State state)1195 static constexpr const char *stateToString(State state) 1196 { 1197 switch (state) { 1198 case STATE_ACTIVE: return "STATE_ACTIVE"; 1199 case STATE_STOPPED: return "STATE_STOPPED"; 1200 case STATE_PAUSED: return "STATE_PAUSED"; 1201 case STATE_PAUSED_STOPPING: return "STATE_PAUSED_STOPPING"; 1202 case STATE_FLUSHED: return "STATE_FLUSHED"; 1203 case STATE_STOPPING: return "STATE_STOPPING"; 1204 default: return "UNKNOWN"; 1205 } 1206 } 1207 1208 // for client callback handler 1209 callback_t mCbf; // callback handler for events, or NULL 1210 void* mUserData; 1211 1212 // for notification APIs 1213 1214 // next 2 fields are const after constructor or set() 1215 uint32_t mNotificationFramesReq; // requested number of frames between each 1216 // notification callback, 1217 // at initial source sample rate 1218 uint32_t mNotificationsPerBufferReq; 1219 // requested number of notifications per buffer, 1220 // currently only used for fast tracks with 1221 // default track buffer size 1222 1223 uint32_t mNotificationFramesAct; // actual number of frames between each 1224 // notification callback, 1225 // at initial source sample rate 1226 bool mRefreshRemaining; // processAudioBuffer() should refresh 1227 // mRemainingFrames and mRetryOnPartialBuffer 1228 1229 // used for static track cbf and restoration 1230 int32_t mLoopCount; // last setLoop loopCount; zero means disabled 1231 uint32_t mLoopStart; // last setLoop loopStart 1232 uint32_t mLoopEnd; // last setLoop loopEnd 1233 int32_t mLoopCountNotified; // the last loopCount notified by callback. 1234 // mLoopCountNotified counts down, matching 1235 // the remaining loop count for static track 1236 // playback. 1237 1238 // These are private to processAudioBuffer(), and are not protected by a lock 1239 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 1240 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 1241 uint32_t mObservedSequence; // last observed value of mSequence 1242 1243 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 1244 bool mMarkerReached; 1245 Modulo<uint32_t> mNewPosition; // in frames 1246 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 1247 1248 Modulo<uint32_t> mServer; // in frames, last known mProxy->getPosition() 1249 // which is count of frames consumed by server, 1250 // reset by new IAudioTrack, 1251 // whether it is reset by stop() is TBD 1252 Modulo<uint32_t> mPosition; // in frames, like mServer except continues 1253 // monotonically after new IAudioTrack, 1254 // and could be easily widened to uint64_t 1255 Modulo<uint32_t> mReleased; // count of frames released to server 1256 // but not necessarily consumed by server, 1257 // reset by stop() but continues monotonically 1258 // after new IAudioTrack to restore mPosition, 1259 // and could be easily widened to uint64_t 1260 int64_t mStartFromZeroUs; // the start time after flush or stop, 1261 // when position should be 0. 1262 // only used for offloaded and direct tracks. 1263 int64_t mStartNs; // the time when start() is called. 1264 ExtendedTimestamp mStartEts; // Extended timestamp at start for normal 1265 // AudioTracks. 1266 AudioTimestamp mStartTs; // Timestamp at start for offloaded or direct 1267 // AudioTracks. 1268 1269 bool mPreviousTimestampValid;// true if mPreviousTimestamp is valid 1270 bool mTimestampStartupGlitchReported; // reduce log spam 1271 bool mTimestampRetrogradePositionReported; // reduce log spam 1272 bool mTimestampRetrogradeTimeReported; // reduce log spam 1273 bool mTimestampStallReported; // reduce log spam 1274 bool mTimestampStaleTimeReported; // reduce log spam 1275 AudioTimestamp mPreviousTimestamp; // used to detect retrograde motion 1276 ExtendedTimestamp::Location mPreviousLocation; // location used for previous timestamp 1277 1278 uint32_t mUnderrunCountOffset; // updated when restoring tracks 1279 1280 int64_t mFramesWritten; // total frames written. reset to zero after 1281 // the start() following stop(). It is not 1282 // changed after restoring the track or 1283 // after flush. 1284 int64_t mFramesWrittenServerOffset; // An offset to server frames due to 1285 // restoring AudioTrack, or stop/start. 1286 // This offset is also used for static tracks. 1287 int64_t mFramesWrittenAtRestore; // Frames written at restore point (or frames 1288 // delivered for static tracks). 1289 // -1 indicates no previous restore point. 1290 1291 audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may 1292 // be denied by client or server, such as 1293 // AUDIO_OUTPUT_FLAG_FAST. mLock must be 1294 // held to read or write those bits reliably. 1295 audio_output_flags_t mOrigFlags; // as specified in constructor or set(), const 1296 1297 bool mDoNotReconnect; 1298 1299 audio_session_t mSessionId; 1300 int mAuxEffectId; 1301 audio_port_handle_t mPortId; // Id from Audio Policy Manager 1302 1303 /** 1304 * mPlayerIId is the player id of the AudioTrack used by AudioManager. 1305 * For an AudioTrack created by the Java interface, this is generally set once. 1306 */ 1307 int mPlayerIId = -1; // AudioManager.h PLAYER_PIID_INVALID 1308 1309 /** 1310 * mLogSessionId is a string identifying this AudioTrack for the metrics service. 1311 * It may be unique or shared with other objects. An empty string means the 1312 * logSessionId is not set. 1313 */ 1314 std::string mLogSessionId{}; 1315 1316 mutable Mutex mLock; 1317 1318 int mPreviousPriority; // before start() 1319 SchedPolicy mPreviousSchedulingGroup; 1320 bool mAwaitBoost; // thread should wait for priority boost before running 1321 1322 // The proxy should only be referenced while a lock is held because the proxy isn't 1323 // multi-thread safe, especially the SingleStateQueue part of the proxy. 1324 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 1325 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 1326 // them around in case they are replaced during the obtainBuffer(). 1327 sp<StaticAudioTrackClientProxy> mStaticProxy; // for type safety only 1328 sp<AudioTrackClientProxy> mProxy; // primary owner of the memory 1329 1330 bool mInUnderrun; // whether track is currently in underrun state 1331 uint32_t mPausedPosition; 1332 1333 // For Device Selection API 1334 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 1335 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 1336 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 1337 // May not match the app selection depending on other 1338 // activity and connected devices. 1339 1340 sp<media::VolumeHandler> mVolumeHandler; 1341 1342 private: 1343 class DeathNotifier : public IBinder::DeathRecipient { 1344 public: DeathNotifier(AudioTrack * audioTrack)1345 explicit DeathNotifier(AudioTrack* audioTrack) : mAudioTrack(audioTrack) { } 1346 protected: 1347 virtual void binderDied(const wp<IBinder>& who); 1348 private: 1349 const wp<AudioTrack> mAudioTrack; 1350 }; 1351 1352 sp<DeathNotifier> mDeathNotifier; 1353 uint32_t mSequence; // incremented for each new IAudioTrack attempt 1354 AttributionSourceState mClientAttributionSource; 1355 1356 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 1357 1358 // Cached values to restore along with the AudioTrack. 1359 audio_dual_mono_mode_t mDualMonoMode = AUDIO_DUAL_MONO_MODE_OFF; 1360 float mAudioDescriptionMixLeveldB = -std::numeric_limits<float>::infinity(); 1361 1362 private: 1363 class MediaMetrics { 1364 public: MediaMetrics()1365 MediaMetrics() : mMetricsItem(mediametrics::Item::create("audiotrack")) { 1366 } ~MediaMetrics()1367 ~MediaMetrics() { 1368 // mMetricsItem alloc failure will be flagged in the constructor 1369 // don't log empty records 1370 if (mMetricsItem->count() > 0) { 1371 mMetricsItem->selfrecord(); 1372 } 1373 } 1374 void gather(const AudioTrack *track); dup()1375 mediametrics::Item *dup() { return mMetricsItem->dup(); } 1376 private: 1377 std::unique_ptr<mediametrics::Item> mMetricsItem; 1378 }; 1379 MediaMetrics mMediaMetrics; 1380 std::string mMetricsId; // GUARDED_BY(mLock), could change in createTrack_l(). 1381 std::string mCallerName; // for example "aaudio" 1382 1383 private: 1384 class AudioTrackCallback : public media::BnAudioTrackCallback { 1385 public: 1386 binder::Status onCodecFormatChanged(const std::vector<uint8_t>& audioMetadata) override; 1387 1388 void setAudioTrackCallback(const sp<media::IAudioTrackCallback>& callback); 1389 private: 1390 Mutex mAudioTrackCbLock; 1391 wp<media::IAudioTrackCallback> mCallback; 1392 }; 1393 sp<AudioTrackCallback> mAudioTrackCallback; 1394 }; 1395 1396 }; // namespace android 1397 1398 #endif // ANDROID_AUDIOTRACK_H 1399