1 /* 2 * Copyright (C) 2009 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 #define LOG_TAG "APM_AudioPolicyManager" 18 //#define LOG_NDEBUG 0 19 20 //#define VERY_VERBOSE_LOGGING 21 #ifdef VERY_VERBOSE_LOGGING 22 #define ALOGVV ALOGV 23 #else 24 #define ALOGVV(a...) do { } while(0) 25 #endif 26 27 #define AUDIO_POLICY_XML_CONFIG_FILE_PATH_MAX_LENGTH 128 28 #define AUDIO_POLICY_XML_CONFIG_FILE_NAME "audio_policy_configuration.xml" 29 30 #include <inttypes.h> 31 #include <math.h> 32 33 #include <AudioPolicyManagerInterface.h> 34 #include <AudioPolicyEngineInstance.h> 35 #include <cutils/atomic.h> 36 #include <cutils/properties.h> 37 #include <utils/Log.h> 38 #include <media/AudioParameter.h> 39 #include <media/AudioPolicyHelper.h> 40 #include <soundtrigger/SoundTrigger.h> 41 #include <system/audio.h> 42 #include <audio_policy_conf.h> 43 #include "AudioPolicyManager.h" 44 #ifndef USE_XML_AUDIO_POLICY_CONF 45 #include <ConfigParsingUtils.h> 46 #include <StreamDescriptor.h> 47 #endif 48 #include <Serializer.h> 49 #include "TypeConverter.h" 50 #include <policy.h> 51 52 namespace android { 53 54 //FIXME: workaround for truncated touch sounds 55 // to be removed when the problem is handled by system UI 56 #define TOUCH_SOUND_FIXED_DELAY_MS 100 57 58 // Largest difference in dB on earpiece in call between the voice volume and another 59 // media / notification / system volume. 60 constexpr float IN_CALL_EARPIECE_HEADROOM_DB = 3.f; 61 62 // ---------------------------------------------------------------------------- 63 // AudioPolicyInterface implementation 64 // ---------------------------------------------------------------------------- 65 66 status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, 67 audio_policy_dev_state_t state, 68 const char *device_address, 69 const char *device_name) 70 { 71 return setDeviceConnectionStateInt(device, state, device_address, device_name); 72 } 73 74 void AudioPolicyManager::broadcastDeviceConnectionState(audio_devices_t device, 75 audio_policy_dev_state_t state, 76 const String8 &device_address) 77 { 78 AudioParameter param(device_address); 79 const String8 key(state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE ? 80 AudioParameter::keyStreamConnect : AudioParameter::keyStreamDisconnect); 81 param.addInt(key, device); 82 mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); 83 } 84 85 status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, 86 audio_policy_dev_state_t state, 87 const char *device_address, 88 const char *device_name) 89 { 90 ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s", 91 - device, state, device_address, device_name); 92 93 // connect/disconnect only 1 device at a time 94 if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; 95 96 sp<DeviceDescriptor> devDesc = 97 mHwModules.getDeviceDescriptor(device, device_address, device_name); 98 99 // handle output devices 100 if (audio_is_output_device(device)) { 101 SortedVector <audio_io_handle_t> outputs; 102 103 ssize_t index = mAvailableOutputDevices.indexOf(devDesc); 104 105 // save a copy of the opened output descriptors before any output is opened or closed 106 // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies() 107 mPreviousOutputs = mOutputs; 108 switch (state) 109 { 110 // handle output device connection 111 case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { 112 if (index >= 0) { 113 ALOGW("setDeviceConnectionState() device already connected: %x", device); 114 return INVALID_OPERATION; 115 } 116 ALOGV("setDeviceConnectionState() connecting device %x", device); 117 118 // register new device as available 119 index = mAvailableOutputDevices.add(devDesc); 120 if (index >= 0) { 121 sp<HwModule> module = mHwModules.getModuleForDevice(device); 122 if (module == 0) { 123 ALOGD("setDeviceConnectionState() could not find HW module for device %08x", 124 device); 125 mAvailableOutputDevices.remove(devDesc); 126 return INVALID_OPERATION; 127 } 128 mAvailableOutputDevices[index]->attach(module); 129 } else { 130 return NO_MEMORY; 131 } 132 133 // Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic 134 // parameters on newly connected devices (instead of opening the outputs...) 135 broadcastDeviceConnectionState(device, state, devDesc->mAddress); 136 137 if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) { 138 mAvailableOutputDevices.remove(devDesc); 139 140 broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, 141 devDesc->mAddress); 142 return INVALID_OPERATION; 143 } 144 // Propagate device availability to Engine 145 mEngine->setDeviceConnectionState(devDesc, state); 146 147 // outputs should never be empty here 148 ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():" 149 "checkOutputsForDevice() returned no outputs but status OK"); 150 ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs", 151 outputs.size()); 152 153 } break; 154 // handle output device disconnection 155 case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { 156 if (index < 0) { 157 ALOGW("setDeviceConnectionState() device not connected: %x", device); 158 return INVALID_OPERATION; 159 } 160 161 ALOGV("setDeviceConnectionState() disconnecting output device %x", device); 162 163 // Send Disconnect to HALs 164 broadcastDeviceConnectionState(device, state, devDesc->mAddress); 165 166 // remove device from available output devices 167 mAvailableOutputDevices.remove(devDesc); 168 169 checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress); 170 171 // Propagate device availability to Engine 172 mEngine->setDeviceConnectionState(devDesc, state); 173 } break; 174 175 default: 176 ALOGE("setDeviceConnectionState() invalid state: %x", state); 177 return BAD_VALUE; 178 } 179 180 // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP 181 // output is suspended before any tracks are moved to it 182 checkA2dpSuspend(); 183 checkOutputForAllStrategies(); 184 // outputs must be closed after checkOutputForAllStrategies() is executed 185 if (!outputs.isEmpty()) { 186 for (size_t i = 0; i < outputs.size(); i++) { 187 sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); 188 // close unused outputs after device disconnection or direct outputs that have been 189 // opened by checkOutputsForDevice() to query dynamic parameters 190 if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) || 191 (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && 192 (desc->mDirectOpenCount == 0))) { 193 closeOutput(outputs[i]); 194 } 195 } 196 // check again after closing A2DP output to reset mA2dpSuspended if needed 197 checkA2dpSuspend(); 198 } 199 200 updateDevicesAndOutputs(); 201 if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) { 202 audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); 203 updateCallRouting(newDevice); 204 } 205 for (size_t i = 0; i < mOutputs.size(); i++) { 206 sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); 207 if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) { 208 audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/); 209 // do not force device change on duplicated output because if device is 0, it will 210 // also force a device 0 for the two outputs it is duplicated to which may override 211 // a valid device selection on those outputs. 212 bool force = !desc->isDuplicated() 213 && (!device_distinguishes_on_address(device) 214 // always force when disconnecting (a non-duplicated device) 215 || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); 216 setOutputDevice(desc, newDevice, force, 0); 217 } 218 } 219 220 if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { 221 cleanUpForDevice(devDesc); 222 } 223 224 mpClientInterface->onAudioPortListUpdate(); 225 return NO_ERROR; 226 } // end if is output device 227 228 // handle input devices 229 if (audio_is_input_device(device)) { 230 SortedVector <audio_io_handle_t> inputs; 231 232 ssize_t index = mAvailableInputDevices.indexOf(devDesc); 233 switch (state) 234 { 235 // handle input device connection 236 case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { 237 if (index >= 0) { 238 ALOGW("setDeviceConnectionState() device already connected: %d", device); 239 return INVALID_OPERATION; 240 } 241 sp<HwModule> module = mHwModules.getModuleForDevice(device); 242 if (module == NULL) { 243 ALOGW("setDeviceConnectionState(): could not find HW module for device %08x", 244 device); 245 return INVALID_OPERATION; 246 } 247 248 // Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic 249 // parameters on newly connected devices (instead of opening the inputs...) 250 broadcastDeviceConnectionState(device, state, devDesc->mAddress); 251 252 if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) { 253 broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, 254 devDesc->mAddress); 255 return INVALID_OPERATION; 256 } 257 258 index = mAvailableInputDevices.add(devDesc); 259 if (index >= 0) { 260 mAvailableInputDevices[index]->attach(module); 261 } else { 262 return NO_MEMORY; 263 } 264 265 // Propagate device availability to Engine 266 mEngine->setDeviceConnectionState(devDesc, state); 267 } break; 268 269 // handle input device disconnection 270 case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { 271 if (index < 0) { 272 ALOGW("setDeviceConnectionState() device not connected: %d", device); 273 return INVALID_OPERATION; 274 } 275 276 ALOGV("setDeviceConnectionState() disconnecting input device %x", device); 277 278 // Set Disconnect to HALs 279 broadcastDeviceConnectionState(device, state, devDesc->mAddress); 280 281 checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress); 282 mAvailableInputDevices.remove(devDesc); 283 284 // Propagate device availability to Engine 285 mEngine->setDeviceConnectionState(devDesc, state); 286 } break; 287 288 default: 289 ALOGE("setDeviceConnectionState() invalid state: %x", state); 290 return BAD_VALUE; 291 } 292 293 closeAllInputs(); 294 // As the input device list can impact the output device selection, update 295 // getDeviceForStrategy() cache 296 updateDevicesAndOutputs(); 297 298 if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) { 299 audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); 300 updateCallRouting(newDevice); 301 } 302 303 if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { 304 cleanUpForDevice(devDesc); 305 } 306 307 mpClientInterface->onAudioPortListUpdate(); 308 return NO_ERROR; 309 } // end if is input device 310 311 ALOGW("setDeviceConnectionState() invalid device: %x", device); 312 return BAD_VALUE; 313 } 314 315 audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device, 316 const char *device_address) 317 { 318 sp<DeviceDescriptor> devDesc = 319 mHwModules.getDeviceDescriptor(device, device_address, "", 320 (strlen(device_address) != 0)/*matchAddress*/); 321 322 if (devDesc == 0) { 323 ALOGW("getDeviceConnectionState() undeclared device, type %08x, address: %s", 324 device, device_address); 325 return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; 326 } 327 328 DeviceVector *deviceVector; 329 330 if (audio_is_output_device(device)) { 331 deviceVector = &mAvailableOutputDevices; 332 } else if (audio_is_input_device(device)) { 333 deviceVector = &mAvailableInputDevices; 334 } else { 335 ALOGW("getDeviceConnectionState() invalid device type %08x", device); 336 return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; 337 } 338 339 return (deviceVector->getDevice(device, String8(device_address)) != 0) ? 340 AUDIO_POLICY_DEVICE_STATE_AVAILABLE : AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; 341 } 342 343 status_t AudioPolicyManager::handleDeviceConfigChange(audio_devices_t device, 344 const char *device_address, 345 const char *device_name) 346 { 347 status_t status; 348 349 ALOGV("handleDeviceConfigChange(() device: 0x%X, address %s name %s", 350 device, device_address, device_name); 351 352 // connect/disconnect only 1 device at a time 353 if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; 354 355 // Check if the device is currently connected 356 sp<DeviceDescriptor> devDesc = 357 mHwModules.getDeviceDescriptor(device, device_address, device_name); 358 ssize_t index = mAvailableOutputDevices.indexOf(devDesc); 359 if (index < 0) { 360 // Nothing to do: device is not connected 361 return NO_ERROR; 362 } 363 364 // Toggle the device state: UNAVAILABLE -> AVAILABLE 365 // This will force reading again the device configuration 366 status = setDeviceConnectionState(device, 367 AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, 368 device_address, device_name); 369 if (status != NO_ERROR) { 370 ALOGW("handleDeviceConfigChange() error disabling connection state: %d", 371 status); 372 return status; 373 } 374 375 status = setDeviceConnectionState(device, 376 AUDIO_POLICY_DEVICE_STATE_AVAILABLE, 377 device_address, device_name); 378 if (status != NO_ERROR) { 379 ALOGW("handleDeviceConfigChange() error enabling connection state: %d", 380 status); 381 return status; 382 } 383 384 return NO_ERROR; 385 } 386 387 uint32_t AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, uint32_t delayMs) 388 { 389 bool createTxPatch = false; 390 status_t status; 391 audio_patch_handle_t afPatchHandle; 392 DeviceVector deviceList; 393 uint32_t muteWaitMs = 0; 394 395 if(!hasPrimaryOutput() || mPrimaryOutput->device() == AUDIO_DEVICE_OUT_STUB) { 396 return muteWaitMs; 397 } 398 audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); 399 ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice); 400 401 // release existing RX patch if any 402 if (mCallRxPatch != 0) { 403 mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); 404 mCallRxPatch.clear(); 405 } 406 // release TX patch if any 407 if (mCallTxPatch != 0) { 408 mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); 409 mCallTxPatch.clear(); 410 } 411 412 // If the RX device is on the primary HW module, then use legacy routing method for voice calls 413 // via setOutputDevice() on primary output. 414 // Otherwise, create two audio patches for TX and RX path. 415 if (availablePrimaryOutputDevices() & rxDevice) { 416 muteWaitMs = setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs); 417 // If the TX device is also on the primary HW module, setOutputDevice() will take care 418 // of it due to legacy implementation. If not, create a patch. 419 if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN) 420 == AUDIO_DEVICE_NONE) { 421 createTxPatch = true; 422 } 423 } else { // create RX path audio patch 424 struct audio_patch patch; 425 426 patch.num_sources = 1; 427 patch.num_sinks = 1; 428 deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice); 429 ALOG_ASSERT(!deviceList.isEmpty(), 430 "updateCallRouting() selected device not in output device list"); 431 sp<DeviceDescriptor> rxSinkDeviceDesc = deviceList.itemAt(0); 432 deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX); 433 ALOG_ASSERT(!deviceList.isEmpty(), 434 "updateCallRouting() no telephony RX device"); 435 sp<DeviceDescriptor> rxSourceDeviceDesc = deviceList.itemAt(0); 436 437 rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); 438 rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); 439 440 // request to reuse existing output stream if one is already opened to reach the RX device 441 SortedVector<audio_io_handle_t> outputs = 442 getOutputsForDevice(rxDevice, mOutputs); 443 audio_io_handle_t output = selectOutput(outputs, 444 AUDIO_OUTPUT_FLAG_NONE, 445 AUDIO_FORMAT_INVALID); 446 if (output != AUDIO_IO_HANDLE_NONE) { 447 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); 448 ALOG_ASSERT(!outputDesc->isDuplicated(), 449 "updateCallRouting() RX device output is duplicated"); 450 outputDesc->toAudioPortConfig(&patch.sources[1]); 451 patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; 452 patch.num_sources = 2; 453 } 454 455 afPatchHandle = AUDIO_PATCH_HANDLE_NONE; 456 status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, delayMs); 457 ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch", 458 status); 459 if (status == NO_ERROR) { 460 mCallRxPatch = new AudioPatch(&patch, mUidCached); 461 mCallRxPatch->mAfPatchHandle = afPatchHandle; 462 mCallRxPatch->mUid = mUidCached; 463 } 464 createTxPatch = true; 465 } 466 if (createTxPatch) { // create TX path audio patch 467 struct audio_patch patch; 468 469 patch.num_sources = 1; 470 patch.num_sinks = 1; 471 deviceList = mAvailableInputDevices.getDevicesFromType(txDevice); 472 ALOG_ASSERT(!deviceList.isEmpty(), 473 "updateCallRouting() selected device not in input device list"); 474 sp<DeviceDescriptor> txSourceDeviceDesc = deviceList.itemAt(0); 475 txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); 476 deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX); 477 ALOG_ASSERT(!deviceList.isEmpty(), 478 "updateCallRouting() no telephony TX device"); 479 sp<DeviceDescriptor> txSinkDeviceDesc = deviceList.itemAt(0); 480 txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); 481 482 SortedVector<audio_io_handle_t> outputs = 483 getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs); 484 audio_io_handle_t output = selectOutput(outputs, 485 AUDIO_OUTPUT_FLAG_NONE, 486 AUDIO_FORMAT_INVALID); 487 // request to reuse existing output stream if one is already opened to reach the TX 488 // path output device 489 if (output != AUDIO_IO_HANDLE_NONE) { 490 sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); 491 ALOG_ASSERT(!outputDesc->isDuplicated(), 492 "updateCallRouting() RX device output is duplicated"); 493 outputDesc->toAudioPortConfig(&patch.sources[1]); 494 patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; 495 patch.num_sources = 2; 496 } 497 498 // terminate active capture if on the same HW module as the call TX source device 499 // FIXME: would be better to refine to only inputs whose profile connects to the 500 // call TX device but this information is not in the audio patch and logic here must be 501 // symmetric to the one in startInput() 502 Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs(); 503 for (size_t i = 0; i < activeInputs.size(); i++) { 504 sp<AudioInputDescriptor> activeDesc = activeInputs[i]; 505 if (activeDesc->hasSameHwModuleAs(txSourceDeviceDesc)) { 506 AudioSessionCollection activeSessions = 507 activeDesc->getAudioSessions(true /*activeOnly*/); 508 for (size_t j = 0; j < activeSessions.size(); j++) { 509 audio_session_t activeSession = activeSessions.keyAt(j); 510 stopInput(activeDesc->mIoHandle, activeSession); 511 releaseInput(activeDesc->mIoHandle, activeSession); 512 } 513 } 514 } 515 516 afPatchHandle = AUDIO_PATCH_HANDLE_NONE; 517 status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, delayMs); 518 ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch", 519 status); 520 if (status == NO_ERROR) { 521 mCallTxPatch = new AudioPatch(&patch, mUidCached); 522 mCallTxPatch->mAfPatchHandle = afPatchHandle; 523 mCallTxPatch->mUid = mUidCached; 524 } 525 } 526 527 return muteWaitMs; 528 } 529 530 void AudioPolicyManager::setPhoneState(audio_mode_t state) 531 { 532 ALOGV("setPhoneState() state %d", state); 533 // store previous phone state for management of sonification strategy below 534 int oldState = mEngine->getPhoneState(); 535 536 if (mEngine->setPhoneState(state) != NO_ERROR) { 537 ALOGW("setPhoneState() invalid or same state %d", state); 538 return; 539 } 540 /// Opens: can these line be executed after the switch of volume curves??? 541 // if leaving call state, handle special case of active streams 542 // pertaining to sonification strategy see handleIncallSonification() 543 if (isStateInCall(oldState)) { 544 ALOGV("setPhoneState() in call state management: new state is %d", state); 545 for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) { 546 handleIncallSonification((audio_stream_type_t)stream, false, true); 547 } 548 549 // force reevaluating accessibility routing when call stops 550 mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); 551 } 552 553 /** 554 * Switching to or from incall state or switching between telephony and VoIP lead to force 555 * routing command. 556 */ 557 bool force = ((is_state_in_call(oldState) != is_state_in_call(state)) 558 || (is_state_in_call(state) && (state != oldState))); 559 560 // check for device and output changes triggered by new phone state 561 checkA2dpSuspend(); 562 checkOutputForAllStrategies(); 563 updateDevicesAndOutputs(); 564 565 int delayMs = 0; 566 if (isStateInCall(state)) { 567 nsecs_t sysTime = systemTime(); 568 for (size_t i = 0; i < mOutputs.size(); i++) { 569 sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); 570 // mute media and sonification strategies and delay device switch by the largest 571 // latency of any output where either strategy is active. 572 // This avoid sending the ring tone or music tail into the earpiece or headset. 573 if ((isStrategyActive(desc, STRATEGY_MEDIA, 574 SONIFICATION_HEADSET_MUSIC_DELAY, 575 sysTime) || 576 isStrategyActive(desc, STRATEGY_SONIFICATION, 577 SONIFICATION_HEADSET_MUSIC_DELAY, 578 sysTime)) && 579 (delayMs < (int)desc->latency()*2)) { 580 delayMs = desc->latency()*2; 581 } 582 setStrategyMute(STRATEGY_MEDIA, true, desc); 583 setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS, 584 getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); 585 setStrategyMute(STRATEGY_SONIFICATION, true, desc); 586 setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS, 587 getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); 588 } 589 } 590 591 if (hasPrimaryOutput()) { 592 // Note that despite the fact that getNewOutputDevice() is called on the primary output, 593 // the device returned is not necessarily reachable via this output 594 audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); 595 // force routing command to audio hardware when ending call 596 // even if no device change is needed 597 if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) { 598 rxDevice = mPrimaryOutput->device(); 599 } 600 601 if (state == AUDIO_MODE_IN_CALL) { 602 updateCallRouting(rxDevice, delayMs); 603 } else if (oldState == AUDIO_MODE_IN_CALL) { 604 if (mCallRxPatch != 0) { 605 mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); 606 mCallRxPatch.clear(); 607 } 608 if (mCallTxPatch != 0) { 609 mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); 610 mCallTxPatch.clear(); 611 } 612 setOutputDevice(mPrimaryOutput, rxDevice, force, 0); 613 } else { 614 setOutputDevice(mPrimaryOutput, rxDevice, force, 0); 615 } 616 } 617 // if entering in call state, handle special case of active streams 618 // pertaining to sonification strategy see handleIncallSonification() 619 if (isStateInCall(state)) { 620 ALOGV("setPhoneState() in call state management: new state is %d", state); 621 for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) { 622 handleIncallSonification((audio_stream_type_t)stream, true, true); 623 } 624 625 // force reevaluating accessibility routing when call starts 626 mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); 627 } 628 629 // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE 630 if (state == AUDIO_MODE_RINGTONE && 631 isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { 632 mLimitRingtoneVolume = true; 633 } else { 634 mLimitRingtoneVolume = false; 635 } 636 } 637 638 audio_mode_t AudioPolicyManager::getPhoneState() { 639 return mEngine->getPhoneState(); 640 } 641 642 void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage, 643 audio_policy_forced_cfg_t config) 644 { 645 ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState()); 646 if (config == mEngine->getForceUse(usage)) { 647 return; 648 } 649 650 if (mEngine->setForceUse(usage, config) != NO_ERROR) { 651 ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage); 652 return; 653 } 654 bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) || 655 (usage == AUDIO_POLICY_FORCE_FOR_DOCK) || 656 (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM); 657 658 // check for device and output changes triggered by new force usage 659 checkA2dpSuspend(); 660 checkOutputForAllStrategies(); 661 updateDevicesAndOutputs(); 662 663 //FIXME: workaround for truncated touch sounds 664 // to be removed when the problem is handled by system UI 665 uint32_t delayMs = 0; 666 uint32_t waitMs = 0; 667 if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) { 668 delayMs = TOUCH_SOUND_FIXED_DELAY_MS; 669 } 670 if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) { 671 audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/); 672 waitMs = updateCallRouting(newDevice, delayMs); 673 } 674 for (size_t i = 0; i < mOutputs.size(); i++) { 675 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); 676 audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/); 677 if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) { 678 waitMs = setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE), 679 delayMs); 680 } 681 if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { 682 applyStreamVolumes(outputDesc, newDevice, waitMs, true); 683 } 684 } 685 686 Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs(); 687 for (size_t i = 0; i < activeInputs.size(); i++) { 688 sp<AudioInputDescriptor> activeDesc = activeInputs[i]; 689 audio_devices_t newDevice = getNewInputDevice(activeDesc); 690 // Force new input selection if the new device can not be reached via current input 691 if (activeDesc->mProfile->getSupportedDevices().types() & 692 (newDevice & ~AUDIO_DEVICE_BIT_IN)) { 693 setInputDevice(activeDesc->mIoHandle, newDevice); 694 } else { 695 closeInput(activeDesc->mIoHandle); 696 } 697 } 698 } 699 700 void AudioPolicyManager::setSystemProperty(const char* property, const char* value) 701 { 702 ALOGV("setSystemProperty() property %s, value %s", property, value); 703 } 704 705 // Find a direct output profile compatible with the parameters passed, even if the input flags do 706 // not explicitly request a direct output 707 sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput( 708 audio_devices_t device, 709 uint32_t samplingRate, 710 audio_format_t format, 711 audio_channel_mask_t channelMask, 712 audio_output_flags_t flags) 713 { 714 // only retain flags that will drive the direct output profile selection 715 // if explicitly requested 716 static const uint32_t kRelevantFlags = 717 (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | 718 AUDIO_OUTPUT_FLAG_VOIP_RX); 719 flags = 720 (audio_output_flags_t)((flags & kRelevantFlags) | AUDIO_OUTPUT_FLAG_DIRECT); 721 722 sp<IOProfile> profile; 723 724 for (size_t i = 0; i < mHwModules.size(); i++) { 725 if (mHwModules[i]->mHandle == 0) { 726 continue; 727 } 728 for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { 729 sp<IOProfile> curProfile = mHwModules[i]->mOutputProfiles[j]; 730 if (!curProfile->isCompatibleProfile(device, String8(""), 731 samplingRate, NULL /*updatedSamplingRate*/, 732 format, NULL /*updatedFormat*/, 733 channelMask, NULL /*updatedChannelMask*/, 734 flags)) { 735 continue; 736 } 737 // reject profiles not corresponding to a device currently available 738 if ((mAvailableOutputDevices.types() & curProfile->getSupportedDevicesType()) == 0) { 739 continue; 740 } 741 // if several profiles are compatible, give priority to one with offload capability 742 if (profile != 0 && ((curProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0)) { 743 continue; 744 } 745 profile = curProfile; 746 if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { 747 break; 748 } 749 } 750 } 751 return profile; 752 } 753 754 audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream, 755 uint32_t samplingRate, 756 audio_format_t format, 757 audio_channel_mask_t channelMask, 758 audio_output_flags_t flags, 759 const audio_offload_info_t *offloadInfo) 760 { 761 routing_strategy strategy = getStrategy(stream); 762 audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); 763 ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x", 764 device, stream, samplingRate, format, channelMask, flags); 765 766 return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE, stream, samplingRate, format, 767 channelMask, flags, offloadInfo); 768 } 769 770 status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, 771 audio_io_handle_t *output, 772 audio_session_t session, 773 audio_stream_type_t *stream, 774 uid_t uid, 775 const audio_config_t *config, 776 audio_output_flags_t flags, 777 audio_port_handle_t *selectedDeviceId, 778 audio_port_handle_t *portId) 779 { 780 audio_attributes_t attributes; 781 if (attr != NULL) { 782 if (!isValidAttributes(attr)) { 783 ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", 784 attr->usage, attr->content_type, attr->flags, 785 attr->tags); 786 return BAD_VALUE; 787 } 788 attributes = *attr; 789 } else { 790 if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) { 791 ALOGE("getOutputForAttr(): invalid stream type"); 792 return BAD_VALUE; 793 } 794 stream_type_to_audio_attributes(*stream, &attributes); 795 } 796 797 // TODO: check for existing client for this port ID 798 if (*portId == AUDIO_PORT_HANDLE_NONE) { 799 *portId = AudioPort::getNextUniqueId(); 800 } 801 802 sp<SwAudioOutputDescriptor> desc; 803 if (mPolicyMixes.getOutputForAttr(attributes, uid, desc) == NO_ERROR) { 804 ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr"); 805 if (!audio_has_proportional_frames(config->format)) { 806 return BAD_VALUE; 807 } 808 *stream = streamTypefromAttributesInt(&attributes); 809 *output = desc->mIoHandle; 810 ALOGV("getOutputForAttr() returns output %d", *output); 811 return NO_ERROR; 812 } 813 if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) { 814 ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE"); 815 return BAD_VALUE; 816 } 817 818 ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x" 819 " session %d selectedDeviceId %d", 820 attributes.usage, attributes.content_type, attributes.tags, attributes.flags, 821 session, *selectedDeviceId); 822 823 *stream = streamTypefromAttributesInt(&attributes); 824 825 // Explicit routing? 826 sp<DeviceDescriptor> deviceDesc; 827 if (*selectedDeviceId != AUDIO_PORT_HANDLE_NONE) { 828 for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { 829 if (mAvailableOutputDevices[i]->getId() == *selectedDeviceId) { 830 deviceDesc = mAvailableOutputDevices[i]; 831 break; 832 } 833 } 834 } 835 mOutputRoutes.addRoute(session, *stream, SessionRoute::SOURCE_TYPE_NA, deviceDesc, uid); 836 837 routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes); 838 audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); 839 840 if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { 841 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); 842 } 843 844 ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x", 845 device, config->sample_rate, config->format, config->channel_mask, flags); 846 847 *output = getOutputForDevice(device, session, *stream, 848 config->sample_rate, config->format, config->channel_mask, 849 flags, &config->offload_info); 850 if (*output == AUDIO_IO_HANDLE_NONE) { 851 mOutputRoutes.removeRoute(session); 852 return INVALID_OPERATION; 853 } 854 855 DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromType(device); 856 *selectedDeviceId = outputDevices.size() > 0 ? outputDevices.itemAt(0)->getId() 857 : AUDIO_PORT_HANDLE_NONE; 858 859 ALOGV(" getOutputForAttr() returns output %d selectedDeviceId %d", *output, *selectedDeviceId); 860 861 return NO_ERROR; 862 } 863 864 audio_io_handle_t AudioPolicyManager::getOutputForDevice( 865 audio_devices_t device, 866 audio_session_t session, 867 audio_stream_type_t stream, 868 uint32_t samplingRate, 869 audio_format_t format, 870 audio_channel_mask_t channelMask, 871 audio_output_flags_t flags, 872 const audio_offload_info_t *offloadInfo) 873 { 874 audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; 875 status_t status; 876 877 #ifdef AUDIO_POLICY_TEST 878 if (mCurOutput != 0) { 879 ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d", 880 mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); 881 882 if (mTestOutputs[mCurOutput] == 0) { 883 ALOGV("getOutput() opening test output"); 884 sp<AudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(NULL, 885 mpClientInterface); 886 outputDesc->mDevice = mTestDevice; 887 outputDesc->mLatency = mTestLatencyMs; 888 outputDesc->mFlags = 889 (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0); 890 outputDesc->mRefCount[stream] = 0; 891 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 892 config.sample_rate = mTestSamplingRate; 893 config.channel_mask = mTestChannels; 894 config.format = mTestFormat; 895 if (offloadInfo != NULL) { 896 config.offload_info = *offloadInfo; 897 } 898 status = mpClientInterface->openOutput(0, 899 &mTestOutputs[mCurOutput], 900 &config, 901 &outputDesc->mDevice, 902 String8(""), 903 &outputDesc->mLatency, 904 outputDesc->mFlags); 905 if (status == NO_ERROR) { 906 outputDesc->mSamplingRate = config.sample_rate; 907 outputDesc->mFormat = config.format; 908 outputDesc->mChannelMask = config.channel_mask; 909 AudioParameter outputCmd = AudioParameter(); 910 outputCmd.addInt(String8("set_id"),mCurOutput); 911 mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString()); 912 addOutput(mTestOutputs[mCurOutput], outputDesc); 913 } 914 } 915 return mTestOutputs[mCurOutput]; 916 } 917 #endif //AUDIO_POLICY_TEST 918 919 // open a direct output if required by specified parameters 920 //force direct flag if offload flag is set: offloading implies a direct output stream 921 // and all common behaviors are driven by checking only the direct flag 922 // this should normally be set appropriately in the policy configuration file 923 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { 924 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); 925 } 926 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { 927 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); 928 } 929 // only allow deep buffering for music stream type 930 if (stream != AUDIO_STREAM_MUSIC) { 931 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 932 } else if (/* stream == AUDIO_STREAM_MUSIC && */ 933 flags == AUDIO_OUTPUT_FLAG_NONE && 934 property_get_bool("audio.deep_buffer.media", false /* default_value */)) { 935 // use DEEP_BUFFER as default output for music stream type 936 flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_DEEP_BUFFER; 937 } 938 if (stream == AUDIO_STREAM_TTS) { 939 flags = AUDIO_OUTPUT_FLAG_TTS; 940 } else if (stream == AUDIO_STREAM_VOICE_CALL && 941 audio_is_linear_pcm(format)) { 942 flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX | 943 AUDIO_OUTPUT_FLAG_DIRECT); 944 ALOGV("Set VoIP and Direct output flags for PCM format"); 945 } 946 947 sp<IOProfile> profile; 948 949 // skip direct output selection if the request can obviously be attached to a mixed output 950 // and not explicitly requested 951 if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) && 952 audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX && 953 audio_channel_count_from_out_mask(channelMask) <= 2) { 954 goto non_direct_output; 955 } 956 957 // Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled. 958 // This prevents creating an offloaded track and tearing it down immediately after start 959 // when audioflinger detects there is an active non offloadable effect. 960 // FIXME: We should check the audio session here but we do not have it in this context. 961 // This may prevent offloading in rare situations where effects are left active by apps 962 // in the background. 963 964 if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || 965 !(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) { 966 profile = getProfileForDirectOutput(device, 967 samplingRate, 968 format, 969 channelMask, 970 (audio_output_flags_t)flags); 971 } 972 973 if (profile != 0) { 974 sp<SwAudioOutputDescriptor> outputDesc = NULL; 975 976 for (size_t i = 0; i < mOutputs.size(); i++) { 977 sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); 978 if (!desc->isDuplicated() && (profile == desc->mProfile)) { 979 outputDesc = desc; 980 // reuse direct output if currently open by the same client 981 // and configured with same parameters 982 if ((samplingRate == outputDesc->mSamplingRate) && 983 audio_formats_match(format, outputDesc->mFormat) && 984 (channelMask == outputDesc->mChannelMask)) { 985 if (session == outputDesc->mDirectClientSession) { 986 outputDesc->mDirectOpenCount++; 987 ALOGV("getOutput() reusing direct output %d for session %d", 988 mOutputs.keyAt(i), session); 989 return mOutputs.keyAt(i); 990 } else { 991 ALOGV("getOutput() do not reuse direct output because current client (%d) " 992 "is not the same as requesting client (%d)", 993 outputDesc->mDirectClientSession, session); 994 goto non_direct_output; 995 } 996 } 997 } 998 } 999 // close direct output if currently open and configured with different parameters 1000 if (outputDesc != NULL) { 1001 closeOutput(outputDesc->mIoHandle); 1002 } 1003 1004 // if the selected profile is offloaded and no offload info was specified, 1005 // create a default one 1006 audio_offload_info_t defaultOffloadInfo = AUDIO_INFO_INITIALIZER; 1007 if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) { 1008 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1009 defaultOffloadInfo.sample_rate = samplingRate; 1010 defaultOffloadInfo.channel_mask = channelMask; 1011 defaultOffloadInfo.format = format; 1012 defaultOffloadInfo.stream_type = stream; 1013 defaultOffloadInfo.bit_rate = 0; 1014 defaultOffloadInfo.duration_us = -1; 1015 defaultOffloadInfo.has_video = true; // conservative 1016 defaultOffloadInfo.is_streaming = true; // likely 1017 offloadInfo = &defaultOffloadInfo; 1018 } 1019 1020 outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface); 1021 outputDesc->mDevice = device; 1022 outputDesc->mLatency = 0; 1023 outputDesc->mFlags = (audio_output_flags_t)(outputDesc->mFlags | flags); 1024 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 1025 config.sample_rate = samplingRate; 1026 config.channel_mask = channelMask; 1027 config.format = format; 1028 if (offloadInfo != NULL) { 1029 config.offload_info = *offloadInfo; 1030 } 1031 DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromType(device); 1032 String8 address = outputDevices.size() > 0 ? outputDevices.itemAt(0)->mAddress 1033 : String8(""); 1034 status = mpClientInterface->openOutput(profile->getModuleHandle(), 1035 &output, 1036 &config, 1037 &outputDesc->mDevice, 1038 address, 1039 &outputDesc->mLatency, 1040 outputDesc->mFlags); 1041 1042 // only accept an output with the requested parameters 1043 if (status != NO_ERROR || 1044 (samplingRate != 0 && samplingRate != config.sample_rate) || 1045 (format != AUDIO_FORMAT_DEFAULT && !audio_formats_match(format, config.format)) || 1046 (channelMask != 0 && channelMask != config.channel_mask)) { 1047 ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d," 1048 "format %d %d, channelMask %04x %04x", output, samplingRate, 1049 outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask, 1050 outputDesc->mChannelMask); 1051 if (output != AUDIO_IO_HANDLE_NONE) { 1052 mpClientInterface->closeOutput(output); 1053 } 1054 // fall back to mixer output if possible when the direct output could not be open 1055 if (audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX) { 1056 goto non_direct_output; 1057 } 1058 return AUDIO_IO_HANDLE_NONE; 1059 } 1060 outputDesc->mSamplingRate = config.sample_rate; 1061 outputDesc->mChannelMask = config.channel_mask; 1062 outputDesc->mFormat = config.format; 1063 outputDesc->mRefCount[stream] = 0; 1064 outputDesc->mStopTime[stream] = 0; 1065 outputDesc->mDirectOpenCount = 1; 1066 outputDesc->mDirectClientSession = session; 1067 1068 addOutput(output, outputDesc); 1069 mPreviousOutputs = mOutputs; 1070 ALOGV("getOutput() returns new direct output %d", output); 1071 mpClientInterface->onAudioPortListUpdate(); 1072 return output; 1073 } 1074 1075 non_direct_output: 1076 1077 // A request for HW A/V sync cannot fallback to a mixed output because time 1078 // stamps are embedded in audio data 1079 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { 1080 return AUDIO_IO_HANDLE_NONE; 1081 } 1082 1083 // ignoring channel mask due to downmix capability in mixer 1084 1085 // open a non direct output 1086 1087 // for non direct outputs, only PCM is supported 1088 if (audio_is_linear_pcm(format)) { 1089 // get which output is suitable for the specified stream. The actual 1090 // routing change will happen when startOutput() will be called 1091 SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs); 1092 1093 // at this stage we should ignore the DIRECT flag as no direct output could be found earlier 1094 flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1095 output = selectOutput(outputs, flags, format); 1096 } 1097 ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," 1098 "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); 1099 1100 return output; 1101 } 1102 1103 audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs, 1104 audio_output_flags_t flags, 1105 audio_format_t format) 1106 { 1107 // select one output among several that provide a path to a particular device or set of 1108 // devices (the list was previously build by getOutputsForDevice()). 1109 // The priority is as follows: 1110 // 1: the output with the highest number of requested policy flags 1111 // 2: the output with the bit depth the closest to the requested one 1112 // 3: the primary output 1113 // 4: the first output in the list 1114 1115 if (outputs.size() == 0) { 1116 return 0; 1117 } 1118 if (outputs.size() == 1) { 1119 return outputs[0]; 1120 } 1121 1122 int maxCommonFlags = 0; 1123 audio_io_handle_t outputForFlags = 0; 1124 audio_io_handle_t outputForPrimary = 0; 1125 audio_io_handle_t outputForFormat = 0; 1126 audio_format_t bestFormat = AUDIO_FORMAT_INVALID; 1127 audio_format_t bestFormatForFlags = AUDIO_FORMAT_INVALID; 1128 1129 for (size_t i = 0; i < outputs.size(); i++) { 1130 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]); 1131 if (!outputDesc->isDuplicated()) { 1132 // if a valid format is specified, skip output if not compatible 1133 if (format != AUDIO_FORMAT_INVALID) { 1134 if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1135 if (!audio_formats_match(format, outputDesc->mFormat)) { 1136 continue; 1137 } 1138 } else if (!audio_is_linear_pcm(format)) { 1139 continue; 1140 } 1141 if (AudioPort::isBetterFormatMatch( 1142 outputDesc->mFormat, bestFormat, format)) { 1143 outputForFormat = outputs[i]; 1144 bestFormat = outputDesc->mFormat; 1145 } 1146 } 1147 1148 int commonFlags = popcount(outputDesc->mProfile->getFlags() & flags); 1149 if (commonFlags >= maxCommonFlags) { 1150 if (commonFlags == maxCommonFlags) { 1151 if (AudioPort::isBetterFormatMatch( 1152 outputDesc->mFormat, bestFormatForFlags, format)) { 1153 outputForFlags = outputs[i]; 1154 bestFormatForFlags = outputDesc->mFormat; 1155 } 1156 } else { 1157 outputForFlags = outputs[i]; 1158 maxCommonFlags = commonFlags; 1159 bestFormatForFlags = outputDesc->mFormat; 1160 } 1161 ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags); 1162 } 1163 if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) { 1164 outputForPrimary = outputs[i]; 1165 } 1166 } 1167 } 1168 1169 if (outputForFlags != 0) { 1170 return outputForFlags; 1171 } 1172 if (outputForFormat != 0) { 1173 return outputForFormat; 1174 } 1175 if (outputForPrimary != 0) { 1176 return outputForPrimary; 1177 } 1178 1179 return outputs[0]; 1180 } 1181 1182 status_t AudioPolicyManager::startOutput(audio_io_handle_t output, 1183 audio_stream_type_t stream, 1184 audio_session_t session) 1185 { 1186 ALOGV("startOutput() output %d, stream %d, session %d", 1187 output, stream, session); 1188 ssize_t index = mOutputs.indexOfKey(output); 1189 if (index < 0) { 1190 ALOGW("startOutput() unknown output %d", output); 1191 return BAD_VALUE; 1192 } 1193 1194 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); 1195 1196 // Routing? 1197 mOutputRoutes.incRouteActivity(session); 1198 1199 audio_devices_t newDevice; 1200 AudioMix *policyMix = NULL; 1201 const char *address = NULL; 1202 if (outputDesc->mPolicyMix != NULL) { 1203 policyMix = outputDesc->mPolicyMix; 1204 address = policyMix->mDeviceAddress.string(); 1205 if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) { 1206 newDevice = policyMix->mDeviceType; 1207 } else { 1208 newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; 1209 } 1210 } else if (mOutputRoutes.hasRouteChanged(session)) { 1211 newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/); 1212 checkStrategyRoute(getStrategy(stream), output); 1213 } else { 1214 newDevice = AUDIO_DEVICE_NONE; 1215 } 1216 1217 uint32_t delayMs = 0; 1218 1219 status_t status = startSource(outputDesc, stream, newDevice, address, &delayMs); 1220 1221 if (status != NO_ERROR) { 1222 mOutputRoutes.decRouteActivity(session); 1223 return status; 1224 } 1225 // Automatically enable the remote submix input when output is started on a re routing mix 1226 // of type MIX_TYPE_RECORDERS 1227 if (audio_is_remote_submix_device(newDevice) && policyMix != NULL && 1228 policyMix->mMixType == MIX_TYPE_RECORDERS) { 1229 setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, 1230 AUDIO_POLICY_DEVICE_STATE_AVAILABLE, 1231 address, 1232 "remote-submix"); 1233 } 1234 1235 if (delayMs != 0) { 1236 usleep(delayMs * 1000); 1237 } 1238 1239 return status; 1240 } 1241 1242 status_t AudioPolicyManager::startSource(const sp<AudioOutputDescriptor>& outputDesc, 1243 audio_stream_type_t stream, 1244 audio_devices_t device, 1245 const char *address, 1246 uint32_t *delayMs) 1247 { 1248 // cannot start playback of STREAM_TTS if any other output is being used 1249 uint32_t beaconMuteLatency = 0; 1250 1251 *delayMs = 0; 1252 if (stream == AUDIO_STREAM_TTS) { 1253 ALOGV("\t found BEACON stream"); 1254 if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) { 1255 return INVALID_OPERATION; 1256 } else { 1257 beaconMuteLatency = handleEventForBeacon(STARTING_BEACON); 1258 } 1259 } else { 1260 // some playback other than beacon starts 1261 beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); 1262 } 1263 1264 // force device change if the output is inactive and no audio patch is already present. 1265 // check active before incrementing usage count 1266 bool force = !outputDesc->isActive() && 1267 (outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE); 1268 1269 // increment usage count for this stream on the requested output: 1270 // NOTE that the usage count is the same for duplicated output and hardware output which is 1271 // necessary for a correct control of hardware output routing by startOutput() and stopOutput() 1272 outputDesc->changeRefCount(stream, 1); 1273 1274 if (stream == AUDIO_STREAM_MUSIC) { 1275 selectOutputForMusicEffects(); 1276 } 1277 1278 if (outputDesc->mRefCount[stream] == 1 || device != AUDIO_DEVICE_NONE) { 1279 // starting an output being rerouted? 1280 if (device == AUDIO_DEVICE_NONE) { 1281 device = getNewOutputDevice(outputDesc, false /*fromCache*/); 1282 } 1283 1284 routing_strategy strategy = getStrategy(stream); 1285 bool shouldWait = (strategy == STRATEGY_SONIFICATION) || 1286 (strategy == STRATEGY_SONIFICATION_RESPECTFUL) || 1287 (beaconMuteLatency > 0); 1288 uint32_t waitMs = beaconMuteLatency; 1289 for (size_t i = 0; i < mOutputs.size(); i++) { 1290 sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); 1291 if (desc != outputDesc) { 1292 // force a device change if any other output is: 1293 // - managed by the same hw module 1294 // - has a current device selection that differs from selected device. 1295 // - supports currently selected device 1296 // - has an active audio patch 1297 // In this case, the audio HAL must receive the new device selection so that it can 1298 // change the device currently selected by the other active output. 1299 if (outputDesc->sharesHwModuleWith(desc) && 1300 desc->device() != device && 1301 desc->supportedDevices() & device && 1302 desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) { 1303 force = true; 1304 } 1305 // wait for audio on other active outputs to be presented when starting 1306 // a notification so that audio focus effect can propagate, or that a mute/unmute 1307 // event occurred for beacon 1308 uint32_t latency = desc->latency(); 1309 if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) { 1310 waitMs = latency; 1311 } 1312 } 1313 } 1314 uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force, 0, NULL, address); 1315 1316 // handle special case for sonification while in call 1317 if (isInCall()) { 1318 handleIncallSonification(stream, true, false); 1319 } 1320 1321 // apply volume rules for current stream and device if necessary 1322 checkAndSetVolume(stream, 1323 mVolumeCurves->getVolumeIndex(stream, outputDesc->device()), 1324 outputDesc, 1325 outputDesc->device()); 1326 1327 // update the outputs if starting an output with a stream that can affect notification 1328 // routing 1329 handleNotificationRoutingForStream(stream); 1330 1331 // force reevaluating accessibility routing when ringtone or alarm starts 1332 if (strategy == STRATEGY_SONIFICATION) { 1333 mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); 1334 } 1335 1336 if (waitMs > muteWaitMs) { 1337 *delayMs = waitMs - muteWaitMs; 1338 } 1339 } 1340 1341 return NO_ERROR; 1342 } 1343 1344 1345 status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, 1346 audio_stream_type_t stream, 1347 audio_session_t session) 1348 { 1349 ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); 1350 ssize_t index = mOutputs.indexOfKey(output); 1351 if (index < 0) { 1352 ALOGW("stopOutput() unknown output %d", output); 1353 return BAD_VALUE; 1354 } 1355 1356 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); 1357 1358 if (outputDesc->mRefCount[stream] == 1) { 1359 // Automatically disable the remote submix input when output is stopped on a 1360 // re routing mix of type MIX_TYPE_RECORDERS 1361 if (audio_is_remote_submix_device(outputDesc->mDevice) && 1362 outputDesc->mPolicyMix != NULL && 1363 outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { 1364 setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, 1365 AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, 1366 outputDesc->mPolicyMix->mDeviceAddress, 1367 "remote-submix"); 1368 } 1369 } 1370 1371 // Routing? 1372 bool forceDeviceUpdate = false; 1373 if (outputDesc->mRefCount[stream] > 0) { 1374 int activityCount = mOutputRoutes.decRouteActivity(session); 1375 forceDeviceUpdate = (mOutputRoutes.hasRoute(session) && (activityCount == 0)); 1376 1377 if (forceDeviceUpdate) { 1378 checkStrategyRoute(getStrategy(stream), AUDIO_IO_HANDLE_NONE); 1379 } 1380 } 1381 1382 return stopSource(outputDesc, stream, forceDeviceUpdate); 1383 } 1384 1385 status_t AudioPolicyManager::stopSource(const sp<AudioOutputDescriptor>& outputDesc, 1386 audio_stream_type_t stream, 1387 bool forceDeviceUpdate) 1388 { 1389 // always handle stream stop, check which stream type is stopping 1390 handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); 1391 1392 // handle special case for sonification while in call 1393 if (isInCall()) { 1394 handleIncallSonification(stream, false, false); 1395 } 1396 1397 if (outputDesc->mRefCount[stream] > 0) { 1398 // decrement usage count of this stream on the output 1399 outputDesc->changeRefCount(stream, -1); 1400 1401 // store time at which the stream was stopped - see isStreamActive() 1402 if (outputDesc->mRefCount[stream] == 0 || forceDeviceUpdate) { 1403 outputDesc->mStopTime[stream] = systemTime(); 1404 audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/); 1405 // delay the device switch by twice the latency because stopOutput() is executed when 1406 // the track stop() command is received and at that time the audio track buffer can 1407 // still contain data that needs to be drained. The latency only covers the audio HAL 1408 // and kernel buffers. Also the latency does not always include additional delay in the 1409 // audio path (audio DSP, CODEC ...) 1410 setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2); 1411 1412 // force restoring the device selection on other active outputs if it differs from the 1413 // one being selected for this output 1414 uint32_t delayMs = outputDesc->latency()*2; 1415 for (size_t i = 0; i < mOutputs.size(); i++) { 1416 sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); 1417 if (desc != outputDesc && 1418 desc->isActive() && 1419 outputDesc->sharesHwModuleWith(desc) && 1420 (newDevice != desc->device())) { 1421 audio_devices_t newDevice2 = getNewOutputDevice(desc, false /*fromCache*/); 1422 bool force = desc->device() != newDevice2; 1423 setOutputDevice(desc, 1424 newDevice2, 1425 force, 1426 delayMs); 1427 // re-apply device specific volume if not done by setOutputDevice() 1428 if (!force) { 1429 applyStreamVolumes(desc, newDevice2, delayMs); 1430 } 1431 } 1432 } 1433 // update the outputs if stopping one with a stream that can affect notification routing 1434 handleNotificationRoutingForStream(stream); 1435 } 1436 if (stream == AUDIO_STREAM_MUSIC) { 1437 selectOutputForMusicEffects(); 1438 } 1439 return NO_ERROR; 1440 } else { 1441 ALOGW("stopOutput() refcount is already 0"); 1442 return INVALID_OPERATION; 1443 } 1444 } 1445 1446 void AudioPolicyManager::releaseOutput(audio_io_handle_t output, 1447 audio_stream_type_t stream __unused, 1448 audio_session_t session __unused) 1449 { 1450 ALOGV("releaseOutput() %d", output); 1451 ssize_t index = mOutputs.indexOfKey(output); 1452 if (index < 0) { 1453 ALOGW("releaseOutput() releasing unknown output %d", output); 1454 return; 1455 } 1456 1457 #ifdef AUDIO_POLICY_TEST 1458 int testIndex = testOutputIndex(output); 1459 if (testIndex != 0) { 1460 sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); 1461 if (outputDesc->isActive()) { 1462 mpClientInterface->closeOutput(output); 1463 removeOutput(output); 1464 mTestOutputs[testIndex] = 0; 1465 } 1466 return; 1467 } 1468 #endif //AUDIO_POLICY_TEST 1469 1470 // Routing 1471 mOutputRoutes.removeRoute(session); 1472 1473 sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(index); 1474 if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1475 if (desc->mDirectOpenCount <= 0) { 1476 ALOGW("releaseOutput() invalid open count %d for output %d", 1477 desc->mDirectOpenCount, output); 1478 return; 1479 } 1480 if (--desc->mDirectOpenCount == 0) { 1481 closeOutput(output); 1482 mpClientInterface->onAudioPortListUpdate(); 1483 } 1484 } 1485 } 1486 1487 1488 status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, 1489 audio_io_handle_t *input, 1490 audio_session_t session, 1491 uid_t uid, 1492 const audio_config_base_t *config, 1493 audio_input_flags_t flags, 1494 audio_port_handle_t *selectedDeviceId, 1495 input_type_t *inputType, 1496 audio_port_handle_t *portId) 1497 { 1498 ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x," 1499 "session %d, flags %#x", 1500 attr->source, config->sample_rate, config->format, config->channel_mask, session, flags); 1501 1502 status_t status = NO_ERROR; 1503 // handle legacy remote submix case where the address was not always specified 1504 String8 address = String8(""); 1505 audio_source_t halInputSource; 1506 audio_source_t inputSource = attr->source; 1507 AudioMix *policyMix = NULL; 1508 DeviceVector inputDevices; 1509 1510 // Explicit routing? 1511 sp<DeviceDescriptor> deviceDesc; 1512 if (*selectedDeviceId != AUDIO_PORT_HANDLE_NONE) { 1513 for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { 1514 if (mAvailableInputDevices[i]->getId() == *selectedDeviceId) { 1515 deviceDesc = mAvailableInputDevices[i]; 1516 break; 1517 } 1518 } 1519 } 1520 mInputRoutes.addRoute(session, SessionRoute::STREAM_TYPE_NA, inputSource, deviceDesc, uid); 1521 1522 // special case for mmap capture: if an input IO handle is specified, we reuse this input if 1523 // possible 1524 if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) == AUDIO_INPUT_FLAG_MMAP_NOIRQ && 1525 *input != AUDIO_IO_HANDLE_NONE) { 1526 ssize_t index = mInputs.indexOfKey(*input); 1527 if (index < 0) { 1528 ALOGW("getInputForAttr() unknown MMAP input %d", *input); 1529 status = BAD_VALUE; 1530 goto error; 1531 } 1532 sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index); 1533 sp<AudioSession> audioSession = inputDesc->getAudioSession(session); 1534 if (audioSession == 0) { 1535 ALOGW("getInputForAttr() unknown session %d on input %d", session, *input); 1536 status = BAD_VALUE; 1537 goto error; 1538 } 1539 // For MMAP mode, the first call to getInputForAttr() is made on behalf of audioflinger. 1540 // The second call is for the first active client and sets the UID. Any further call 1541 // corresponds to a new client and is only permitted from the same UId. 1542 if (audioSession->openCount() == 1) { 1543 audioSession->setUid(uid); 1544 } else if (audioSession->uid() != uid) { 1545 ALOGW("getInputForAttr() bad uid %d for session %d uid %d", 1546 uid, session, audioSession->uid()); 1547 status = INVALID_OPERATION; 1548 goto error; 1549 } 1550 audioSession->changeOpenCount(1); 1551 *inputType = API_INPUT_LEGACY; 1552 if (*portId == AUDIO_PORT_HANDLE_NONE) { 1553 *portId = AudioPort::getNextUniqueId(); 1554 } 1555 inputDevices = mAvailableInputDevices.getDevicesFromType(inputDesc->mDevice); 1556 *selectedDeviceId = inputDevices.size() > 0 ? inputDevices.itemAt(0)->getId() 1557 : AUDIO_PORT_HANDLE_NONE; 1558 ALOGI("%s reusing MMAP input %d for session %d", __FUNCTION__, *input, session); 1559 1560 return NO_ERROR; 1561 } 1562 1563 *input = AUDIO_IO_HANDLE_NONE; 1564 *inputType = API_INPUT_INVALID; 1565 1566 if (inputSource == AUDIO_SOURCE_DEFAULT) { 1567 inputSource = AUDIO_SOURCE_MIC; 1568 } 1569 halInputSource = inputSource; 1570 1571 // TODO: check for existing client for this port ID 1572 if (*portId == AUDIO_PORT_HANDLE_NONE) { 1573 *portId = AudioPort::getNextUniqueId(); 1574 } 1575 1576 audio_devices_t device; 1577 1578 if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX && 1579 strncmp(attr->tags, "addr=", strlen("addr=")) == 0) { 1580 status = mPolicyMixes.getInputMixForAttr(*attr, &policyMix); 1581 if (status != NO_ERROR) { 1582 goto error; 1583 } 1584 *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; 1585 device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; 1586 address = String8(attr->tags + strlen("addr=")); 1587 } else { 1588 device = getDeviceAndMixForInputSource(inputSource, &policyMix); 1589 if (device == AUDIO_DEVICE_NONE) { 1590 ALOGW("getInputForAttr() could not find device for source %d", inputSource); 1591 status = BAD_VALUE; 1592 goto error; 1593 } 1594 if (policyMix != NULL) { 1595 address = policyMix->mDeviceAddress; 1596 if (policyMix->mMixType == MIX_TYPE_RECORDERS) { 1597 // there is an external policy, but this input is attached to a mix of recorders, 1598 // meaning it receives audio injected into the framework, so the recorder doesn't 1599 // know about it and is therefore considered "legacy" 1600 *inputType = API_INPUT_LEGACY; 1601 } else { 1602 // recording a mix of players defined by an external policy, we're rerouting for 1603 // an external policy 1604 *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; 1605 } 1606 } else if (audio_is_remote_submix_device(device)) { 1607 address = String8("0"); 1608 *inputType = API_INPUT_MIX_CAPTURE; 1609 } else if (device == AUDIO_DEVICE_IN_TELEPHONY_RX) { 1610 *inputType = API_INPUT_TELEPHONY_RX; 1611 } else { 1612 *inputType = API_INPUT_LEGACY; 1613 } 1614 1615 } 1616 1617 *input = getInputForDevice(device, address, session, uid, inputSource, 1618 config->sample_rate, config->format, config->channel_mask, flags, 1619 policyMix); 1620 if (*input == AUDIO_IO_HANDLE_NONE) { 1621 status = INVALID_OPERATION; 1622 goto error; 1623 } 1624 1625 inputDevices = mAvailableInputDevices.getDevicesFromType(device); 1626 *selectedDeviceId = inputDevices.size() > 0 ? inputDevices.itemAt(0)->getId() 1627 : AUDIO_PORT_HANDLE_NONE; 1628 1629 ALOGV("getInputForAttr() returns input %d type %d selectedDeviceId %d", 1630 *input, *inputType, *selectedDeviceId); 1631 1632 return NO_ERROR; 1633 1634 error: 1635 mInputRoutes.removeRoute(session); 1636 return status; 1637 } 1638 1639 1640 audio_io_handle_t AudioPolicyManager::getInputForDevice(audio_devices_t device, 1641 String8 address, 1642 audio_session_t session, 1643 uid_t uid, 1644 audio_source_t inputSource, 1645 uint32_t samplingRate, 1646 audio_format_t format, 1647 audio_channel_mask_t channelMask, 1648 audio_input_flags_t flags, 1649 AudioMix *policyMix) 1650 { 1651 audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; 1652 audio_source_t halInputSource = inputSource; 1653 bool isSoundTrigger = false; 1654 1655 if (inputSource == AUDIO_SOURCE_HOTWORD) { 1656 ssize_t index = mSoundTriggerSessions.indexOfKey(session); 1657 if (index >= 0) { 1658 input = mSoundTriggerSessions.valueFor(session); 1659 isSoundTrigger = true; 1660 flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD); 1661 ALOGV("SoundTrigger capture on session %d input %d", session, input); 1662 } else { 1663 halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION; 1664 } 1665 } else if (inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION && 1666 audio_is_linear_pcm(format)) { 1667 flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_VOIP_TX); 1668 } 1669 1670 // find a compatible input profile (not necessarily identical in parameters) 1671 sp<IOProfile> profile; 1672 // samplingRate and flags may be updated by getInputProfile 1673 uint32_t profileSamplingRate = (samplingRate == 0) ? SAMPLE_RATE_HZ_DEFAULT : samplingRate; 1674 audio_format_t profileFormat = format; 1675 audio_channel_mask_t profileChannelMask = channelMask; 1676 audio_input_flags_t profileFlags = flags; 1677 for (;;) { 1678 profile = getInputProfile(device, address, 1679 profileSamplingRate, profileFormat, profileChannelMask, 1680 profileFlags); 1681 if (profile != 0) { 1682 break; // success 1683 } else if (profileFlags & AUDIO_INPUT_FLAG_RAW) { 1684 profileFlags = (audio_input_flags_t) (profileFlags & ~AUDIO_INPUT_FLAG_RAW); // retry 1685 } else if (profileFlags != AUDIO_INPUT_FLAG_NONE) { 1686 profileFlags = AUDIO_INPUT_FLAG_NONE; // retry 1687 } else { // fail 1688 ALOGW("getInputForDevice() could not find profile for device 0x%X," 1689 "samplingRate %u, format %#x, channelMask 0x%X, flags %#x", 1690 device, samplingRate, format, channelMask, flags); 1691 return input; 1692 } 1693 } 1694 // Pick input sampling rate if not specified by client 1695 if (samplingRate == 0) { 1696 samplingRate = profileSamplingRate; 1697 } 1698 1699 if (profile->getModuleHandle() == 0) { 1700 ALOGE("getInputForAttr(): HW module %s not opened", profile->getModuleName()); 1701 return input; 1702 } 1703 1704 sp<AudioSession> audioSession = new AudioSession(session, 1705 inputSource, 1706 format, 1707 samplingRate, 1708 channelMask, 1709 flags, 1710 uid, 1711 isSoundTrigger, 1712 policyMix, mpClientInterface); 1713 1714 // FIXME: disable concurrent capture until UI is ready 1715 #if 0 1716 // reuse an open input if possible 1717 sp<AudioInputDescriptor> reusedInputDesc; 1718 for (size_t i = 0; i < mInputs.size(); i++) { 1719 sp<AudioInputDescriptor> desc = mInputs.valueAt(i); 1720 // reuse input if: 1721 // - it shares the same profile 1722 // AND 1723 // - it is not a reroute submix input 1724 // AND 1725 // - it is: not used for sound trigger 1726 // OR 1727 // used for sound trigger and all clients use the same session ID 1728 // 1729 if ((profile == desc->mProfile) && 1730 (isSoundTrigger == desc->isSoundTrigger()) && 1731 !is_virtual_input_device(device)) { 1732 1733 sp<AudioSession> as = desc->getAudioSession(session); 1734 if (as != 0) { 1735 // do not allow unmatching properties on same session 1736 if (as->matches(audioSession)) { 1737 as->changeOpenCount(1); 1738 } else { 1739 ALOGW("getInputForDevice() record with different attributes" 1740 " exists for session %d", session); 1741 continue; 1742 } 1743 } else if (isSoundTrigger) { 1744 continue; 1745 } 1746 1747 // Reuse the already opened input stream on this profile if: 1748 // - the new capture source is background OR 1749 // - the path requested configurations match OR 1750 // - the new source priority is less than the highest source priority on this input 1751 // If the input stream cannot be reused, close it before opening a new stream 1752 // on the same profile for the new client so that the requested path configuration 1753 // can be selected. 1754 if (!isConcurrentSource(inputSource) && 1755 ((desc->mSamplingRate != samplingRate || 1756 desc->mChannelMask != channelMask || 1757 !audio_formats_match(desc->mFormat, format)) && 1758 (source_priority(desc->getHighestPrioritySource(false /*activeOnly*/)) < 1759 source_priority(inputSource)))) { 1760 reusedInputDesc = desc; 1761 continue; 1762 } else { 1763 desc->addAudioSession(session, audioSession); 1764 ALOGV("%s: reusing input %d", __FUNCTION__, mInputs.keyAt(i)); 1765 return mInputs.keyAt(i); 1766 } 1767 } 1768 } 1769 1770 if (reusedInputDesc != 0) { 1771 AudioSessionCollection sessions = reusedInputDesc->getAudioSessions(false /*activeOnly*/); 1772 for (size_t j = 0; j < sessions.size(); j++) { 1773 audio_session_t currentSession = sessions.keyAt(j); 1774 stopInput(reusedInputDesc->mIoHandle, currentSession); 1775 releaseInput(reusedInputDesc->mIoHandle, currentSession); 1776 } 1777 } 1778 #endif 1779 1780 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 1781 config.sample_rate = profileSamplingRate; 1782 config.channel_mask = profileChannelMask; 1783 config.format = profileFormat; 1784 1785 if (address == "") { 1786 DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(device); 1787 // the inputs vector must be of size 1, but we don't want to crash here 1788 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress : String8(""); 1789 } 1790 1791 status_t status = mpClientInterface->openInput(profile->getModuleHandle(), 1792 &input, 1793 &config, 1794 &device, 1795 address, 1796 halInputSource, 1797 profileFlags); 1798 1799 // only accept input with the exact requested set of parameters 1800 if (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE || 1801 (profileSamplingRate != config.sample_rate) || 1802 !audio_formats_match(profileFormat, config.format) || 1803 (profileChannelMask != config.channel_mask)) { 1804 ALOGW("getInputForAttr() failed opening input: samplingRate %d" 1805 ", format %d, channelMask %x", 1806 samplingRate, format, channelMask); 1807 if (input != AUDIO_IO_HANDLE_NONE) { 1808 mpClientInterface->closeInput(input); 1809 } 1810 return AUDIO_IO_HANDLE_NONE; 1811 } 1812 1813 sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile); 1814 inputDesc->mSamplingRate = profileSamplingRate; 1815 inputDesc->mFormat = profileFormat; 1816 inputDesc->mChannelMask = profileChannelMask; 1817 inputDesc->mDevice = device; 1818 inputDesc->mPolicyMix = policyMix; 1819 inputDesc->addAudioSession(session, audioSession); 1820 1821 addInput(input, inputDesc); 1822 mpClientInterface->onAudioPortListUpdate(); 1823 1824 return input; 1825 } 1826 1827 //static 1828 bool AudioPolicyManager::isConcurrentSource(audio_source_t source) 1829 { 1830 return (source == AUDIO_SOURCE_HOTWORD) || 1831 (source == AUDIO_SOURCE_VOICE_RECOGNITION) || 1832 (source == AUDIO_SOURCE_FM_TUNER); 1833 } 1834 1835 bool AudioPolicyManager::isConcurentCaptureAllowed(const sp<AudioInputDescriptor>& inputDesc, 1836 const sp<AudioSession>& audioSession) 1837 { 1838 // Do not allow capture if an active voice call is using a software patch and 1839 // the call TX source device is on the same HW module. 1840 // FIXME: would be better to refine to only inputs whose profile connects to the 1841 // call TX device but this information is not in the audio patch 1842 if (mCallTxPatch != 0 && 1843 inputDesc->getModuleHandle() == mCallTxPatch->mPatch.sources[0].ext.device.hw_module) { 1844 return false; 1845 } 1846 1847 // starting concurrent capture is enabled if: 1848 // 1) capturing for re-routing 1849 // 2) capturing for HOTWORD source 1850 // 3) capturing for FM TUNER source 1851 // 3) All other active captures are either for re-routing or HOTWORD 1852 1853 if (is_virtual_input_device(inputDesc->mDevice) || 1854 isConcurrentSource(audioSession->inputSource())) { 1855 return true; 1856 } 1857 1858 Vector< sp<AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs(); 1859 for (size_t i = 0; i < activeInputs.size(); i++) { 1860 sp<AudioInputDescriptor> activeInput = activeInputs[i]; 1861 if (!isConcurrentSource(activeInput->inputSource(true)) && 1862 !is_virtual_input_device(activeInput->mDevice)) { 1863 return false; 1864 } 1865 } 1866 1867 return true; 1868 } 1869 1870 // FIXME: remove when concurrent capture is ready. This is a hack to work around bug b/63083537. 1871 bool AudioPolicyManager::soundTriggerSupportsConcurrentCapture() { 1872 if (!mHasComputedSoundTriggerSupportsConcurrentCapture) { 1873 bool soundTriggerSupportsConcurrentCapture = false; 1874 unsigned int numModules = 0; 1875 struct sound_trigger_module_descriptor* nModules = NULL; 1876 1877 status_t status = SoundTrigger::listModules(nModules, &numModules); 1878 if (status == NO_ERROR && numModules != 0) { 1879 nModules = (struct sound_trigger_module_descriptor*) calloc( 1880 numModules, sizeof(struct sound_trigger_module_descriptor)); 1881 if (nModules == NULL) { 1882 // We failed to malloc the buffer, so just say no for now, and hope that we have more 1883 // ram the next time this function is called. 1884 ALOGE("Failed to allocate buffer for module descriptors"); 1885 return false; 1886 } 1887 1888 status = SoundTrigger::listModules(nModules, &numModules); 1889 if (status == NO_ERROR) { 1890 soundTriggerSupportsConcurrentCapture = true; 1891 for (size_t i = 0; i < numModules; ++i) { 1892 soundTriggerSupportsConcurrentCapture &= 1893 nModules[i].properties.concurrent_capture; 1894 } 1895 } 1896 free(nModules); 1897 } 1898 mSoundTriggerSupportsConcurrentCapture = soundTriggerSupportsConcurrentCapture; 1899 mHasComputedSoundTriggerSupportsConcurrentCapture = true; 1900 } 1901 return mSoundTriggerSupportsConcurrentCapture; 1902 } 1903 1904 1905 status_t AudioPolicyManager::startInput(audio_io_handle_t input, 1906 audio_session_t session, 1907 concurrency_type__mask_t *concurrency) 1908 { 1909 ALOGV("startInput() input %d", input); 1910 *concurrency = API_INPUT_CONCURRENCY_NONE; 1911 ssize_t index = mInputs.indexOfKey(input); 1912 if (index < 0) { 1913 ALOGW("startInput() unknown input %d", input); 1914 return BAD_VALUE; 1915 } 1916 sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index); 1917 1918 sp<AudioSession> audioSession = inputDesc->getAudioSession(session); 1919 if (audioSession == 0) { 1920 ALOGW("startInput() unknown session %d on input %d", session, input); 1921 return BAD_VALUE; 1922 } 1923 1924 // FIXME: disable concurrent capture until UI is ready 1925 #if 0 1926 if (!isConcurentCaptureAllowed(inputDesc, audioSession)) { 1927 ALOGW("startInput(%d) failed: other input already started", input); 1928 return INVALID_OPERATION; 1929 } 1930 1931 if (isInCall()) { 1932 *concurrency |= API_INPUT_CONCURRENCY_CALL; 1933 } 1934 if (mInputs.activeInputsCountOnDevices() != 0) { 1935 *concurrency |= API_INPUT_CONCURRENCY_CAPTURE; 1936 } 1937 #else 1938 if (!is_virtual_input_device(inputDesc->mDevice)) { 1939 if (mCallTxPatch != 0 && 1940 inputDesc->getModuleHandle() == mCallTxPatch->mPatch.sources[0].ext.device.hw_module) { 1941 ALOGW("startInput(%d) failed: call in progress", input); 1942 return INVALID_OPERATION; 1943 } 1944 1945 Vector< sp<AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs(); 1946 for (size_t i = 0; i < activeInputs.size(); i++) { 1947 sp<AudioInputDescriptor> activeDesc = activeInputs[i]; 1948 1949 if (is_virtual_input_device(activeDesc->mDevice)) { 1950 continue; 1951 } 1952 1953 if ((audioSession->flags() & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0 && 1954 activeDesc->getId() == inputDesc->getId()) { 1955 continue; 1956 } 1957 1958 audio_source_t activeSource = activeDesc->inputSource(true); 1959 if (audioSession->inputSource() == AUDIO_SOURCE_HOTWORD) { 1960 if (activeSource == AUDIO_SOURCE_HOTWORD) { 1961 if (activeDesc->hasPreemptedSession(session)) { 1962 ALOGW("startInput(%d) failed for HOTWORD: " 1963 "other input %d already started for HOTWORD", 1964 input, activeDesc->mIoHandle); 1965 return INVALID_OPERATION; 1966 } 1967 } else { 1968 ALOGV("startInput(%d) failed for HOTWORD: other input %d already started", 1969 input, activeDesc->mIoHandle); 1970 return INVALID_OPERATION; 1971 } 1972 } else { 1973 if (activeSource != AUDIO_SOURCE_HOTWORD) { 1974 ALOGW("startInput(%d) failed: other input %d already started", 1975 input, activeDesc->mIoHandle); 1976 return INVALID_OPERATION; 1977 } 1978 } 1979 } 1980 1981 // We only need to check if the sound trigger session supports concurrent capture if the 1982 // input is also a sound trigger input. Otherwise, we should preempt any hotword stream 1983 // that's running. 1984 const bool allowConcurrentWithSoundTrigger = 1985 inputDesc->isSoundTrigger() ? soundTriggerSupportsConcurrentCapture() : false; 1986 1987 // if capture is allowed, preempt currently active HOTWORD captures 1988 for (size_t i = 0; i < activeInputs.size(); i++) { 1989 sp<AudioInputDescriptor> activeDesc = activeInputs[i]; 1990 1991 if (is_virtual_input_device(activeDesc->mDevice)) { 1992 continue; 1993 } 1994 1995 if (allowConcurrentWithSoundTrigger && activeDesc->isSoundTrigger()) { 1996 continue; 1997 } 1998 1999 audio_source_t activeSource = activeDesc->inputSource(true); 2000 if (activeSource == AUDIO_SOURCE_HOTWORD) { 2001 AudioSessionCollection activeSessions = 2002 activeDesc->getAudioSessions(true /*activeOnly*/); 2003 audio_session_t activeSession = activeSessions.keyAt(0); 2004 audio_io_handle_t activeHandle = activeDesc->mIoHandle; 2005 SortedVector<audio_session_t> sessions = activeDesc->getPreemptedSessions(); 2006 sessions.add(activeSession); 2007 inputDesc->setPreemptedSessions(sessions); 2008 stopInput(activeHandle, activeSession); 2009 releaseInput(activeHandle, activeSession); 2010 ALOGV("startInput(%d) for HOTWORD preempting HOTWORD input %d", 2011 input, activeDesc->mIoHandle); 2012 } 2013 } 2014 } 2015 #endif 2016 2017 // increment activity count before calling getNewInputDevice() below as only active sessions 2018 // are considered for device selection 2019 audioSession->changeActiveCount(1); 2020 2021 // Routing? 2022 mInputRoutes.incRouteActivity(session); 2023 2024 if (audioSession->activeCount() == 1 || mInputRoutes.hasRouteChanged(session)) { 2025 // indicate active capture to sound trigger service if starting capture from a mic on 2026 // primary HW module 2027 audio_devices_t device = getNewInputDevice(inputDesc); 2028 setInputDevice(input, device, true /* force */); 2029 2030 if (inputDesc->getAudioSessionCount(true/*activeOnly*/) == 1) { 2031 // if input maps to a dynamic policy with an activity listener, notify of state change 2032 if ((inputDesc->mPolicyMix != NULL) 2033 && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) { 2034 mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mDeviceAddress, 2035 MIX_STATE_MIXING); 2036 } 2037 2038 audio_devices_t primaryInputDevices = availablePrimaryInputDevices(); 2039 if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) && 2040 mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1) { 2041 SoundTrigger::setCaptureState(true); 2042 } 2043 2044 // automatically enable the remote submix output when input is started if not 2045 // used by a policy mix of type MIX_TYPE_RECORDERS 2046 // For remote submix (a virtual device), we open only one input per capture request. 2047 if (audio_is_remote_submix_device(inputDesc->mDevice)) { 2048 String8 address = String8(""); 2049 if (inputDesc->mPolicyMix == NULL) { 2050 address = String8("0"); 2051 } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { 2052 address = inputDesc->mPolicyMix->mDeviceAddress; 2053 } 2054 if (address != "") { 2055 setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, 2056 AUDIO_POLICY_DEVICE_STATE_AVAILABLE, 2057 address, "remote-submix"); 2058 } 2059 } 2060 } 2061 } 2062 2063 ALOGV("AudioPolicyManager::startInput() input source = %d", audioSession->inputSource()); 2064 2065 return NO_ERROR; 2066 } 2067 2068 status_t AudioPolicyManager::stopInput(audio_io_handle_t input, 2069 audio_session_t session) 2070 { 2071 ALOGV("stopInput() input %d", input); 2072 ssize_t index = mInputs.indexOfKey(input); 2073 if (index < 0) { 2074 ALOGW("stopInput() unknown input %d", input); 2075 return BAD_VALUE; 2076 } 2077 sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index); 2078 2079 sp<AudioSession> audioSession = inputDesc->getAudioSession(session); 2080 if (index < 0) { 2081 ALOGW("stopInput() unknown session %d on input %d", session, input); 2082 return BAD_VALUE; 2083 } 2084 2085 if (audioSession->activeCount() == 0) { 2086 ALOGW("stopInput() input %d already stopped", input); 2087 return INVALID_OPERATION; 2088 } 2089 2090 audioSession->changeActiveCount(-1); 2091 2092 // Routing? 2093 mInputRoutes.decRouteActivity(session); 2094 2095 if (audioSession->activeCount() == 0) { 2096 2097 if (inputDesc->isActive()) { 2098 setInputDevice(input, getNewInputDevice(inputDesc), false /* force */); 2099 } else { 2100 // if input maps to a dynamic policy with an activity listener, notify of state change 2101 if ((inputDesc->mPolicyMix != NULL) 2102 && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) { 2103 mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mDeviceAddress, 2104 MIX_STATE_IDLE); 2105 } 2106 2107 // automatically disable the remote submix output when input is stopped if not 2108 // used by a policy mix of type MIX_TYPE_RECORDERS 2109 if (audio_is_remote_submix_device(inputDesc->mDevice)) { 2110 String8 address = String8(""); 2111 if (inputDesc->mPolicyMix == NULL) { 2112 address = String8("0"); 2113 } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { 2114 address = inputDesc->mPolicyMix->mDeviceAddress; 2115 } 2116 if (address != "") { 2117 setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, 2118 AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, 2119 address, "remote-submix"); 2120 } 2121 } 2122 2123 audio_devices_t device = inputDesc->mDevice; 2124 resetInputDevice(input); 2125 2126 // indicate inactive capture to sound trigger service if stopping capture from a mic on 2127 // primary HW module 2128 audio_devices_t primaryInputDevices = availablePrimaryInputDevices(); 2129 if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) && 2130 mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) { 2131 SoundTrigger::setCaptureState(false); 2132 } 2133 inputDesc->clearPreemptedSessions(); 2134 } 2135 } 2136 return NO_ERROR; 2137 } 2138 2139 void AudioPolicyManager::releaseInput(audio_io_handle_t input, 2140 audio_session_t session) 2141 { 2142 2143 ALOGV("releaseInput() %d", input); 2144 ssize_t index = mInputs.indexOfKey(input); 2145 if (index < 0) { 2146 ALOGW("releaseInput() releasing unknown input %d", input); 2147 return; 2148 } 2149 2150 // Routing 2151 mInputRoutes.removeRoute(session); 2152 2153 sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index); 2154 ALOG_ASSERT(inputDesc != 0); 2155 2156 sp<AudioSession> audioSession = inputDesc->getAudioSession(session); 2157 if (audioSession == 0) { 2158 ALOGW("releaseInput() unknown session %d on input %d", session, input); 2159 return; 2160 } 2161 2162 if (audioSession->openCount() == 0) { 2163 ALOGW("releaseInput() invalid open count %d on session %d", 2164 audioSession->openCount(), session); 2165 return; 2166 } 2167 2168 if (audioSession->changeOpenCount(-1) == 0) { 2169 inputDesc->removeAudioSession(session); 2170 } 2171 2172 if (inputDesc->getOpenRefCount() > 0) { 2173 ALOGV("releaseInput() exit > 0"); 2174 return; 2175 } 2176 2177 closeInput(input); 2178 mpClientInterface->onAudioPortListUpdate(); 2179 ALOGV("releaseInput() exit"); 2180 } 2181 2182 void AudioPolicyManager::closeAllInputs() { 2183 bool patchRemoved = false; 2184 2185 for(size_t input_index = 0; input_index < mInputs.size(); input_index++) { 2186 sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index); 2187 ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); 2188 if (patch_index >= 0) { 2189 sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index); 2190 (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); 2191 mAudioPatches.removeItemsAt(patch_index); 2192 patchRemoved = true; 2193 } 2194 mpClientInterface->closeInput(mInputs.keyAt(input_index)); 2195 } 2196 mInputs.clear(); 2197 SoundTrigger::setCaptureState(false); 2198 nextAudioPortGeneration(); 2199 2200 if (patchRemoved) { 2201 mpClientInterface->onAudioPatchListUpdate(); 2202 } 2203 } 2204 2205 void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream, 2206 int indexMin, 2207 int indexMax) 2208 { 2209 ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); 2210 mVolumeCurves->initStreamVolume(stream, indexMin, indexMax); 2211 2212 // initialize other private stream volumes which follow this one 2213 for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) { 2214 if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) { 2215 continue; 2216 } 2217 mVolumeCurves->initStreamVolume((audio_stream_type_t)curStream, indexMin, indexMax); 2218 } 2219 } 2220 2221 status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, 2222 int index, 2223 audio_devices_t device) 2224 { 2225 2226 if ((index < mVolumeCurves->getVolumeIndexMin(stream)) || 2227 (index > mVolumeCurves->getVolumeIndexMax(stream))) { 2228 return BAD_VALUE; 2229 } 2230 if (!audio_is_output_device(device)) { 2231 return BAD_VALUE; 2232 } 2233 2234 // Force max volume if stream cannot be muted 2235 if (!mVolumeCurves->canBeMuted(stream)) index = mVolumeCurves->getVolumeIndexMax(stream); 2236 2237 ALOGV("setStreamVolumeIndex() stream %d, device %08x, index %d", 2238 stream, device, index); 2239 2240 // update other private stream volumes which follow this one 2241 for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) { 2242 if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) { 2243 continue; 2244 } 2245 mVolumeCurves->addCurrentVolumeIndex((audio_stream_type_t)curStream, device, index); 2246 } 2247 2248 // update volume on all outputs and streams matching the following: 2249 // - The requested stream (or a stream matching for volume control) is active on the output 2250 // - The device (or devices) selected by the strategy corresponding to this stream includes 2251 // the requested device 2252 // - For non default requested device, currently selected device on the output is either the 2253 // requested device or one of the devices selected by the strategy 2254 // - For default requested device (AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME), apply volume only if 2255 // no specific device volume value exists for currently selected device. 2256 status_t status = NO_ERROR; 2257 for (size_t i = 0; i < mOutputs.size(); i++) { 2258 sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); 2259 audio_devices_t curDevice = Volume::getDeviceForVolume(desc->device()); 2260 for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) { 2261 if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) { 2262 continue; 2263 } 2264 if (!(desc->isStreamActive((audio_stream_type_t)curStream) || 2265 (isInCall() && (curStream == AUDIO_STREAM_VOICE_CALL)))) { 2266 continue; 2267 } 2268 routing_strategy curStrategy = getStrategy((audio_stream_type_t)curStream); 2269 audio_devices_t curStreamDevice = Volume::getDeviceForVolume(getDeviceForStrategy( 2270 curStrategy, false /*fromCache*/)); 2271 if ((device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) && 2272 ((curStreamDevice & device) == 0)) { 2273 continue; 2274 } 2275 bool applyVolume; 2276 if (device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) { 2277 curStreamDevice |= device; 2278 applyVolume = (curDevice & curStreamDevice) != 0; 2279 } else { 2280 applyVolume = !mVolumeCurves->hasVolumeIndexForDevice( 2281 stream, curStreamDevice); 2282 } 2283 2284 if (applyVolume) { 2285 //FIXME: workaround for truncated touch sounds 2286 // delayed volume change for system stream to be removed when the problem is 2287 // handled by system UI 2288 status_t volStatus = 2289 checkAndSetVolume((audio_stream_type_t)curStream, index, desc, curDevice, 2290 (stream == AUDIO_STREAM_SYSTEM) ? TOUCH_SOUND_FIXED_DELAY_MS : 0); 2291 if (volStatus != NO_ERROR) { 2292 status = volStatus; 2293 } 2294 } 2295 } 2296 } 2297 return status; 2298 } 2299 2300 status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream, 2301 int *index, 2302 audio_devices_t device) 2303 { 2304 if (index == NULL) { 2305 return BAD_VALUE; 2306 } 2307 if (!audio_is_output_device(device)) { 2308 return BAD_VALUE; 2309 } 2310 // if device is AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME, return volume for device corresponding to 2311 // the strategy the stream belongs to. 2312 if (device == AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) { 2313 device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); 2314 } 2315 device = Volume::getDeviceForVolume(device); 2316 2317 *index = mVolumeCurves->getVolumeIndex(stream, device); 2318 ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); 2319 return NO_ERROR; 2320 } 2321 2322 audio_io_handle_t AudioPolicyManager::selectOutputForMusicEffects() 2323 { 2324 // select one output among several suitable for global effects. 2325 // The priority is as follows: 2326 // 1: An offloaded output. If the effect ends up not being offloadable, 2327 // AudioFlinger will invalidate the track and the offloaded output 2328 // will be closed causing the effect to be moved to a PCM output. 2329 // 2: A deep buffer output 2330 // 3: The primary output 2331 // 4: the first output in the list 2332 2333 routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC); 2334 audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); 2335 SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs); 2336 2337 if (outputs.size() == 0) { 2338 return AUDIO_IO_HANDLE_NONE; 2339 } 2340 2341 audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; 2342 bool activeOnly = true; 2343 2344 while (output == AUDIO_IO_HANDLE_NONE) { 2345 audio_io_handle_t outputOffloaded = AUDIO_IO_HANDLE_NONE; 2346 audio_io_handle_t outputDeepBuffer = AUDIO_IO_HANDLE_NONE; 2347 audio_io_handle_t outputPrimary = AUDIO_IO_HANDLE_NONE; 2348 2349 for (size_t i = 0; i < outputs.size(); i++) { 2350 sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); 2351 if (activeOnly && !desc->isStreamActive(AUDIO_STREAM_MUSIC)) { 2352 continue; 2353 } 2354 ALOGV("selectOutputForMusicEffects activeOnly %d outputs[%zu] flags 0x%08x", 2355 activeOnly, i, desc->mFlags); 2356 if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { 2357 outputOffloaded = outputs[i]; 2358 } 2359 if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { 2360 outputDeepBuffer = outputs[i]; 2361 } 2362 if ((desc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) != 0) { 2363 outputPrimary = outputs[i]; 2364 } 2365 } 2366 if (outputOffloaded != AUDIO_IO_HANDLE_NONE) { 2367 output = outputOffloaded; 2368 } else if (outputDeepBuffer != AUDIO_IO_HANDLE_NONE) { 2369 output = outputDeepBuffer; 2370 } else if (outputPrimary != AUDIO_IO_HANDLE_NONE) { 2371 output = outputPrimary; 2372 } else { 2373 output = outputs[0]; 2374 } 2375 activeOnly = false; 2376 } 2377 2378 if (output != mMusicEffectOutput) { 2379 mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mMusicEffectOutput, output); 2380 mMusicEffectOutput = output; 2381 } 2382 2383 ALOGV("selectOutputForMusicEffects selected output %d", output); 2384 return output; 2385 } 2386 2387 audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc __unused) 2388 { 2389 return selectOutputForMusicEffects(); 2390 } 2391 2392 status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc, 2393 audio_io_handle_t io, 2394 uint32_t strategy, 2395 int session, 2396 int id) 2397 { 2398 ssize_t index = mOutputs.indexOfKey(io); 2399 if (index < 0) { 2400 index = mInputs.indexOfKey(io); 2401 if (index < 0) { 2402 ALOGW("registerEffect() unknown io %d", io); 2403 return INVALID_OPERATION; 2404 } 2405 } 2406 return mEffects.registerEffect(desc, io, strategy, session, id); 2407 } 2408 2409 bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const 2410 { 2411 bool active = false; 2412 for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT && !active; curStream++) { 2413 if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) { 2414 continue; 2415 } 2416 active = mOutputs.isStreamActive((audio_stream_type_t)curStream, inPastMs); 2417 } 2418 return active; 2419 } 2420 2421 bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const 2422 { 2423 return mOutputs.isStreamActiveRemotely(stream, inPastMs); 2424 } 2425 2426 bool AudioPolicyManager::isSourceActive(audio_source_t source) const 2427 { 2428 for (size_t i = 0; i < mInputs.size(); i++) { 2429 const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i); 2430 if (inputDescriptor->isSourceActive(source)) { 2431 return true; 2432 } 2433 } 2434 return false; 2435 } 2436 2437 // Register a list of custom mixes with their attributes and format. 2438 // When a mix is registered, corresponding input and output profiles are 2439 // added to the remote submix hw module. The profile contains only the 2440 // parameters (sampling rate, format...) specified by the mix. 2441 // The corresponding input remote submix device is also connected. 2442 // 2443 // When a remote submix device is connected, the address is checked to select the 2444 // appropriate profile and the corresponding input or output stream is opened. 2445 // 2446 // When capture starts, getInputForAttr() will: 2447 // - 1 look for a mix matching the address passed in attribtutes tags if any 2448 // - 2 if none found, getDeviceForInputSource() will: 2449 // - 2.1 look for a mix matching the attributes source 2450 // - 2.2 if none found, default to device selection by policy rules 2451 // At this time, the corresponding output remote submix device is also connected 2452 // and active playback use cases can be transferred to this mix if needed when reconnecting 2453 // after AudioTracks are invalidated 2454 // 2455 // When playback starts, getOutputForAttr() will: 2456 // - 1 look for a mix matching the address passed in attribtutes tags if any 2457 // - 2 if none found, look for a mix matching the attributes usage 2458 // - 3 if none found, default to device and output selection by policy rules. 2459 2460 status_t AudioPolicyManager::registerPolicyMixes(const Vector<AudioMix>& mixes) 2461 { 2462 ALOGV("registerPolicyMixes() %zu mix(es)", mixes.size()); 2463 status_t res = NO_ERROR; 2464 2465 sp<HwModule> rSubmixModule; 2466 // examine each mix's route type 2467 for (size_t i = 0; i < mixes.size(); i++) { 2468 // we only support MIX_ROUTE_FLAG_LOOP_BACK or MIX_ROUTE_FLAG_RENDER, not the combination 2469 if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_ALL) == MIX_ROUTE_FLAG_ALL) { 2470 res = INVALID_OPERATION; 2471 break; 2472 } 2473 if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) { 2474 // Loop back through "remote submix" 2475 if (rSubmixModule == 0) { 2476 for (size_t j = 0; i < mHwModules.size(); j++) { 2477 if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[j]->mName) == 0 2478 && mHwModules[j]->mHandle != 0) { 2479 rSubmixModule = mHwModules[j]; 2480 break; 2481 } 2482 } 2483 } 2484 2485 ALOGV("registerPolicyMixes() mix %zu of %zu is LOOP_BACK", i, mixes.size()); 2486 2487 if (rSubmixModule == 0) { 2488 ALOGE(" Unable to find audio module for submix, aborting mix %zu registration", i); 2489 res = INVALID_OPERATION; 2490 break; 2491 } 2492 2493 String8 address = mixes[i].mDeviceAddress; 2494 2495 if (mPolicyMixes.registerMix(address, mixes[i], 0 /*output desc*/) != NO_ERROR) { 2496 ALOGE(" Error registering mix %zu for address %s", i, address.string()); 2497 res = INVALID_OPERATION; 2498 break; 2499 } 2500 audio_config_t outputConfig = mixes[i].mFormat; 2501 audio_config_t inputConfig = mixes[i].mFormat; 2502 // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in 2503 // stereo and let audio flinger do the channel conversion if needed. 2504 outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO; 2505 inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO; 2506 rSubmixModule->addOutputProfile(address, &outputConfig, 2507 AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address); 2508 rSubmixModule->addInputProfile(address, &inputConfig, 2509 AUDIO_DEVICE_IN_REMOTE_SUBMIX, address); 2510 2511 if (mixes[i].mMixType == MIX_TYPE_PLAYERS) { 2512 setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, 2513 AUDIO_POLICY_DEVICE_STATE_AVAILABLE, 2514 address.string(), "remote-submix"); 2515 } else { 2516 setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, 2517 AUDIO_POLICY_DEVICE_STATE_AVAILABLE, 2518 address.string(), "remote-submix"); 2519 } 2520 } else if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) { 2521 String8 address = mixes[i].mDeviceAddress; 2522 audio_devices_t device = mixes[i].mDeviceType; 2523 ALOGV(" registerPolicyMixes() mix %zu of %zu is RENDER, dev=0x%X addr=%s", 2524 i, mixes.size(), device, address.string()); 2525 2526 bool foundOutput = false; 2527 for (size_t j = 0 ; j < mOutputs.size() ; j++) { 2528 sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(j); 2529 sp<AudioPatch> patch = mAudioPatches.valueFor(desc->getPatchHandle()); 2530 if ((patch != 0) && (patch->mPatch.num_sinks != 0) 2531 && (patch->mPatch.sinks[0].type == AUDIO_PORT_TYPE_DEVICE) 2532 && (patch->mPatch.sinks[0].ext.device.type == device) 2533 && (strncmp(patch->mPatch.sinks[0].ext.device.address, address.string(), 2534 AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) { 2535 if (mPolicyMixes.registerMix(address, mixes[i], desc) != NO_ERROR) { 2536 res = INVALID_OPERATION; 2537 } else { 2538 foundOutput = true; 2539 } 2540 break; 2541 } 2542 } 2543 2544 if (res != NO_ERROR) { 2545 ALOGE(" Error registering mix %zu for device 0x%X addr %s", 2546 i, device, address.string()); 2547 res = INVALID_OPERATION; 2548 break; 2549 } else if (!foundOutput) { 2550 ALOGE(" Output not found for mix %zu for device 0x%X addr %s", 2551 i, device, address.string()); 2552 res = INVALID_OPERATION; 2553 break; 2554 } 2555 } 2556 } 2557 if (res != NO_ERROR) { 2558 unregisterPolicyMixes(mixes); 2559 } 2560 return res; 2561 } 2562 2563 status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes) 2564 { 2565 ALOGV("unregisterPolicyMixes() num mixes %zu", mixes.size()); 2566 status_t res = NO_ERROR; 2567 sp<HwModule> rSubmixModule; 2568 // examine each mix's route type 2569 for (size_t i = 0; i < mixes.size(); i++) { 2570 if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) { 2571 2572 if (rSubmixModule == 0) { 2573 for (size_t j = 0; i < mHwModules.size(); j++) { 2574 if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[j]->mName) == 0 2575 && mHwModules[j]->mHandle != 0) { 2576 rSubmixModule = mHwModules[j]; 2577 break; 2578 } 2579 } 2580 } 2581 if (rSubmixModule == 0) { 2582 res = INVALID_OPERATION; 2583 continue; 2584 } 2585 2586 String8 address = mixes[i].mDeviceAddress; 2587 2588 if (mPolicyMixes.unregisterMix(address) != NO_ERROR) { 2589 res = INVALID_OPERATION; 2590 continue; 2591 } 2592 2593 if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) == 2594 AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { 2595 setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, 2596 AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, 2597 address.string(), "remote-submix"); 2598 } 2599 if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) == 2600 AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { 2601 setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, 2602 AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, 2603 address.string(), "remote-submix"); 2604 } 2605 rSubmixModule->removeOutputProfile(address); 2606 rSubmixModule->removeInputProfile(address); 2607 2608 } if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) { 2609 if (mPolicyMixes.unregisterMix(mixes[i].mDeviceAddress) != NO_ERROR) { 2610 res = INVALID_OPERATION; 2611 continue; 2612 } 2613 } 2614 } 2615 return res; 2616 } 2617 2618 2619 status_t AudioPolicyManager::dump(int fd) 2620 { 2621 const size_t SIZE = 256; 2622 char buffer[SIZE]; 2623 String8 result; 2624 2625 snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this); 2626 result.append(buffer); 2627 2628 snprintf(buffer, SIZE, " Primary Output: %d\n", 2629 hasPrimaryOutput() ? mPrimaryOutput->mIoHandle : AUDIO_IO_HANDLE_NONE); 2630 result.append(buffer); 2631 std::string stateLiteral; 2632 AudioModeConverter::toString(mEngine->getPhoneState(), stateLiteral); 2633 snprintf(buffer, SIZE, " Phone state: %s\n", stateLiteral.c_str()); 2634 result.append(buffer); 2635 snprintf(buffer, SIZE, " Force use for communications %d\n", 2636 mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)); 2637 result.append(buffer); 2638 snprintf(buffer, SIZE, " Force use for media %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA)); 2639 result.append(buffer); 2640 snprintf(buffer, SIZE, " Force use for record %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD)); 2641 result.append(buffer); 2642 snprintf(buffer, SIZE, " Force use for dock %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK)); 2643 result.append(buffer); 2644 snprintf(buffer, SIZE, " Force use for system %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM)); 2645 result.append(buffer); 2646 snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n", 2647 mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO)); 2648 result.append(buffer); 2649 snprintf(buffer, SIZE, " Force use for encoded surround output %d\n", 2650 mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND)); 2651 result.append(buffer); 2652 snprintf(buffer, SIZE, " TTS output %s\n", mTtsOutputAvailable ? "available" : "not available"); 2653 result.append(buffer); 2654 snprintf(buffer, SIZE, " Master mono: %s\n", mMasterMono ? "on" : "off"); 2655 result.append(buffer); 2656 2657 write(fd, result.string(), result.size()); 2658 2659 mAvailableOutputDevices.dump(fd, String8("Available output")); 2660 mAvailableInputDevices.dump(fd, String8("Available input")); 2661 mHwModules.dump(fd); 2662 mOutputs.dump(fd); 2663 mInputs.dump(fd); 2664 mVolumeCurves->dump(fd); 2665 mEffects.dump(fd); 2666 mAudioPatches.dump(fd); 2667 mPolicyMixes.dump(fd); 2668 2669 return NO_ERROR; 2670 } 2671 2672 // This function checks for the parameters which can be offloaded. 2673 // This can be enhanced depending on the capability of the DSP and policy 2674 // of the system. 2675 bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo) 2676 { 2677 ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," 2678 " BitRate=%u, duration=%" PRId64 " us, has_video=%d", 2679 offloadInfo.sample_rate, offloadInfo.channel_mask, 2680 offloadInfo.format, 2681 offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, 2682 offloadInfo.has_video); 2683 2684 if (mMasterMono) { 2685 return false; // no offloading if mono is set. 2686 } 2687 2688 // Check if offload has been disabled 2689 char propValue[PROPERTY_VALUE_MAX]; 2690 if (property_get("audio.offload.disable", propValue, "0")) { 2691 if (atoi(propValue) != 0) { 2692 ALOGV("offload disabled by audio.offload.disable=%s", propValue ); 2693 return false; 2694 } 2695 } 2696 2697 // Check if stream type is music, then only allow offload as of now. 2698 if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) 2699 { 2700 ALOGV("isOffloadSupported: stream_type != MUSIC, returning false"); 2701 return false; 2702 } 2703 2704 //TODO: enable audio offloading with video when ready 2705 const bool allowOffloadWithVideo = 2706 property_get_bool("audio.offload.video", false /* default_value */); 2707 if (offloadInfo.has_video && !allowOffloadWithVideo) { 2708 ALOGV("isOffloadSupported: has_video == true, returning false"); 2709 return false; 2710 } 2711 2712 //If duration is less than minimum value defined in property, return false 2713 if (property_get("audio.offload.min.duration.secs", propValue, NULL)) { 2714 if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) { 2715 ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); 2716 return false; 2717 } 2718 } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { 2719 ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS); 2720 return false; 2721 } 2722 2723 // Do not allow offloading if one non offloadable effect is enabled. This prevents from 2724 // creating an offloaded track and tearing it down immediately after start when audioflinger 2725 // detects there is an active non offloadable effect. 2726 // FIXME: We should check the audio session here but we do not have it in this context. 2727 // This may prevent offloading in rare situations where effects are left active by apps 2728 // in the background. 2729 if (mEffects.isNonOffloadableEffectEnabled()) { 2730 return false; 2731 } 2732 2733 // See if there is a profile to support this. 2734 // AUDIO_DEVICE_NONE 2735 sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */, 2736 offloadInfo.sample_rate, 2737 offloadInfo.format, 2738 offloadInfo.channel_mask, 2739 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 2740 ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT "); 2741 return (profile != 0); 2742 } 2743 2744 status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role, 2745 audio_port_type_t type, 2746 unsigned int *num_ports, 2747 struct audio_port *ports, 2748 unsigned int *generation) 2749 { 2750 if (num_ports == NULL || (*num_ports != 0 && ports == NULL) || 2751 generation == NULL) { 2752 return BAD_VALUE; 2753 } 2754 ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports); 2755 if (ports == NULL) { 2756 *num_ports = 0; 2757 } 2758 2759 size_t portsWritten = 0; 2760 size_t portsMax = *num_ports; 2761 *num_ports = 0; 2762 if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) { 2763 // do not report devices with type AUDIO_DEVICE_IN_STUB or AUDIO_DEVICE_OUT_STUB 2764 // as they are used by stub HALs by convention 2765 if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { 2766 for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { 2767 if (mAvailableOutputDevices[i]->type() == AUDIO_DEVICE_OUT_STUB) { 2768 continue; 2769 } 2770 if (portsWritten < portsMax) { 2771 mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]); 2772 } 2773 (*num_ports)++; 2774 } 2775 } 2776 if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { 2777 for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { 2778 if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_STUB) { 2779 continue; 2780 } 2781 if (portsWritten < portsMax) { 2782 mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]); 2783 } 2784 (*num_ports)++; 2785 } 2786 } 2787 } 2788 if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) { 2789 if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { 2790 for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) { 2791 mInputs[i]->toAudioPort(&ports[portsWritten++]); 2792 } 2793 *num_ports += mInputs.size(); 2794 } 2795 if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { 2796 size_t numOutputs = 0; 2797 for (size_t i = 0; i < mOutputs.size(); i++) { 2798 if (!mOutputs[i]->isDuplicated()) { 2799 numOutputs++; 2800 if (portsWritten < portsMax) { 2801 mOutputs[i]->toAudioPort(&ports[portsWritten++]); 2802 } 2803 } 2804 } 2805 *num_ports += numOutputs; 2806 } 2807 } 2808 *generation = curAudioPortGeneration(); 2809 ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports); 2810 return NO_ERROR; 2811 } 2812 2813 status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused) 2814 { 2815 return NO_ERROR; 2816 } 2817 2818 status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, 2819 audio_patch_handle_t *handle, 2820 uid_t uid) 2821 { 2822 ALOGV("createAudioPatch()"); 2823 2824 if (handle == NULL || patch == NULL) { 2825 return BAD_VALUE; 2826 } 2827 ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks); 2828 2829 if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX || 2830 patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) { 2831 return BAD_VALUE; 2832 } 2833 // only one source per audio patch supported for now 2834 if (patch->num_sources > 1) { 2835 return INVALID_OPERATION; 2836 } 2837 2838 if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) { 2839 return INVALID_OPERATION; 2840 } 2841 for (size_t i = 0; i < patch->num_sinks; i++) { 2842 if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) { 2843 return INVALID_OPERATION; 2844 } 2845 } 2846 2847 sp<AudioPatch> patchDesc; 2848 ssize_t index = mAudioPatches.indexOfKey(*handle); 2849 2850 ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id, 2851 patch->sources[0].role, 2852 patch->sources[0].type); 2853 #if LOG_NDEBUG == 0 2854 for (size_t i = 0; i < patch->num_sinks; i++) { 2855 ALOGV("createAudioPatch sink %zu: id %d role %d type %d", i, patch->sinks[i].id, 2856 patch->sinks[i].role, 2857 patch->sinks[i].type); 2858 } 2859 #endif 2860 2861 if (index >= 0) { 2862 patchDesc = mAudioPatches.valueAt(index); 2863 ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", 2864 mUidCached, patchDesc->mUid, uid); 2865 if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { 2866 return INVALID_OPERATION; 2867 } 2868 } else { 2869 *handle = AUDIO_PATCH_HANDLE_NONE; 2870 } 2871 2872 if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { 2873 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); 2874 if (outputDesc == NULL) { 2875 ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id); 2876 return BAD_VALUE; 2877 } 2878 ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports", 2879 outputDesc->mIoHandle); 2880 if (patchDesc != 0) { 2881 if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { 2882 ALOGV("createAudioPatch() source id differs for patch current id %d new id %d", 2883 patchDesc->mPatch.sources[0].id, patch->sources[0].id); 2884 return BAD_VALUE; 2885 } 2886 } 2887 DeviceVector devices; 2888 for (size_t i = 0; i < patch->num_sinks; i++) { 2889 // Only support mix to devices connection 2890 // TODO add support for mix to mix connection 2891 if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { 2892 ALOGV("createAudioPatch() source mix but sink is not a device"); 2893 return INVALID_OPERATION; 2894 } 2895 sp<DeviceDescriptor> devDesc = 2896 mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); 2897 if (devDesc == 0) { 2898 ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id); 2899 return BAD_VALUE; 2900 } 2901 2902 if (!outputDesc->mProfile->isCompatibleProfile(devDesc->type(), 2903 devDesc->mAddress, 2904 patch->sources[0].sample_rate, 2905 NULL, // updatedSamplingRate 2906 patch->sources[0].format, 2907 NULL, // updatedFormat 2908 patch->sources[0].channel_mask, 2909 NULL, // updatedChannelMask 2910 AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) { 2911 ALOGV("createAudioPatch() profile not supported for device %08x", 2912 devDesc->type()); 2913 return INVALID_OPERATION; 2914 } 2915 devices.add(devDesc); 2916 } 2917 if (devices.size() == 0) { 2918 return INVALID_OPERATION; 2919 } 2920 2921 // TODO: reconfigure output format and channels here 2922 ALOGV("createAudioPatch() setting device %08x on output %d", 2923 devices.types(), outputDesc->mIoHandle); 2924 setOutputDevice(outputDesc, devices.types(), true, 0, handle); 2925 index = mAudioPatches.indexOfKey(*handle); 2926 if (index >= 0) { 2927 if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { 2928 ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided"); 2929 } 2930 patchDesc = mAudioPatches.valueAt(index); 2931 patchDesc->mUid = uid; 2932 ALOGV("createAudioPatch() success"); 2933 } else { 2934 ALOGW("createAudioPatch() setOutputDevice() failed to create a patch"); 2935 return INVALID_OPERATION; 2936 } 2937 } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { 2938 if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { 2939 // input device to input mix connection 2940 // only one sink supported when connecting an input device to a mix 2941 if (patch->num_sinks > 1) { 2942 return INVALID_OPERATION; 2943 } 2944 sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id); 2945 if (inputDesc == NULL) { 2946 return BAD_VALUE; 2947 } 2948 if (patchDesc != 0) { 2949 if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) { 2950 return BAD_VALUE; 2951 } 2952 } 2953 sp<DeviceDescriptor> devDesc = 2954 mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); 2955 if (devDesc == 0) { 2956 return BAD_VALUE; 2957 } 2958 2959 if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(), 2960 devDesc->mAddress, 2961 patch->sinks[0].sample_rate, 2962 NULL, /*updatedSampleRate*/ 2963 patch->sinks[0].format, 2964 NULL, /*updatedFormat*/ 2965 patch->sinks[0].channel_mask, 2966 NULL, /*updatedChannelMask*/ 2967 // FIXME for the parameter type, 2968 // and the NONE 2969 (audio_output_flags_t) 2970 AUDIO_INPUT_FLAG_NONE)) { 2971 return INVALID_OPERATION; 2972 } 2973 // TODO: reconfigure output format and channels here 2974 ALOGV("createAudioPatch() setting device %08x on output %d", 2975 devDesc->type(), inputDesc->mIoHandle); 2976 setInputDevice(inputDesc->mIoHandle, devDesc->type(), true, handle); 2977 index = mAudioPatches.indexOfKey(*handle); 2978 if (index >= 0) { 2979 if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { 2980 ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided"); 2981 } 2982 patchDesc = mAudioPatches.valueAt(index); 2983 patchDesc->mUid = uid; 2984 ALOGV("createAudioPatch() success"); 2985 } else { 2986 ALOGW("createAudioPatch() setInputDevice() failed to create a patch"); 2987 return INVALID_OPERATION; 2988 } 2989 } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { 2990 // device to device connection 2991 if (patchDesc != 0) { 2992 if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { 2993 return BAD_VALUE; 2994 } 2995 } 2996 sp<DeviceDescriptor> srcDeviceDesc = 2997 mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); 2998 if (srcDeviceDesc == 0) { 2999 return BAD_VALUE; 3000 } 3001 3002 //update source and sink with our own data as the data passed in the patch may 3003 // be incomplete. 3004 struct audio_patch newPatch = *patch; 3005 srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]); 3006 3007 for (size_t i = 0; i < patch->num_sinks; i++) { 3008 if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { 3009 ALOGV("createAudioPatch() source device but one sink is not a device"); 3010 return INVALID_OPERATION; 3011 } 3012 3013 sp<DeviceDescriptor> sinkDeviceDesc = 3014 mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); 3015 if (sinkDeviceDesc == 0) { 3016 return BAD_VALUE; 3017 } 3018 sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]); 3019 3020 // create a software bridge in PatchPanel if: 3021 // - source and sink devices are on differnt HW modules OR 3022 // - audio HAL version is < 3.0 3023 if (!srcDeviceDesc->hasSameHwModuleAs(sinkDeviceDesc) || 3024 (srcDeviceDesc->mModule->getHalVersionMajor() < 3)) { 3025 // support only one sink device for now to simplify output selection logic 3026 if (patch->num_sinks > 1) { 3027 return INVALID_OPERATION; 3028 } 3029 SortedVector<audio_io_handle_t> outputs = 3030 getOutputsForDevice(sinkDeviceDesc->type(), mOutputs); 3031 // if the sink device is reachable via an opened output stream, request to go via 3032 // this output stream by adding a second source to the patch description 3033 audio_io_handle_t output = selectOutput(outputs, 3034 AUDIO_OUTPUT_FLAG_NONE, 3035 AUDIO_FORMAT_INVALID); 3036 if (output != AUDIO_IO_HANDLE_NONE) { 3037 sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); 3038 if (outputDesc->isDuplicated()) { 3039 return INVALID_OPERATION; 3040 } 3041 outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]); 3042 newPatch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; 3043 newPatch.num_sources = 2; 3044 } 3045 } 3046 } 3047 // TODO: check from routing capabilities in config file and other conflicting patches 3048 3049 audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; 3050 if (index >= 0) { 3051 afPatchHandle = patchDesc->mAfPatchHandle; 3052 } 3053 3054 status_t status = mpClientInterface->createAudioPatch(&newPatch, 3055 &afPatchHandle, 3056 0); 3057 ALOGV("createAudioPatch() patch panel returned %d patchHandle %d", 3058 status, afPatchHandle); 3059 if (status == NO_ERROR) { 3060 if (index < 0) { 3061 patchDesc = new AudioPatch(&newPatch, uid); 3062 addAudioPatch(patchDesc->mHandle, patchDesc); 3063 } else { 3064 patchDesc->mPatch = newPatch; 3065 } 3066 patchDesc->mAfPatchHandle = afPatchHandle; 3067 *handle = patchDesc->mHandle; 3068 nextAudioPortGeneration(); 3069 mpClientInterface->onAudioPatchListUpdate(); 3070 } else { 3071 ALOGW("createAudioPatch() patch panel could not connect device patch, error %d", 3072 status); 3073 return INVALID_OPERATION; 3074 } 3075 } else { 3076 return BAD_VALUE; 3077 } 3078 } else { 3079 return BAD_VALUE; 3080 } 3081 return NO_ERROR; 3082 } 3083 3084 status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle, 3085 uid_t uid) 3086 { 3087 ALOGV("releaseAudioPatch() patch %d", handle); 3088 3089 ssize_t index = mAudioPatches.indexOfKey(handle); 3090 3091 if (index < 0) { 3092 return BAD_VALUE; 3093 } 3094 sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index); 3095 ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", 3096 mUidCached, patchDesc->mUid, uid); 3097 if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { 3098 return INVALID_OPERATION; 3099 } 3100 3101 struct audio_patch *patch = &patchDesc->mPatch; 3102 patchDesc->mUid = mUidCached; 3103 if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { 3104 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); 3105 if (outputDesc == NULL) { 3106 ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id); 3107 return BAD_VALUE; 3108 } 3109 3110 setOutputDevice(outputDesc, 3111 getNewOutputDevice(outputDesc, true /*fromCache*/), 3112 true, 3113 0, 3114 NULL); 3115 } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { 3116 if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { 3117 sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id); 3118 if (inputDesc == NULL) { 3119 ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id); 3120 return BAD_VALUE; 3121 } 3122 setInputDevice(inputDesc->mIoHandle, 3123 getNewInputDevice(inputDesc), 3124 true, 3125 NULL); 3126 } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { 3127 status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); 3128 ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d", 3129 status, patchDesc->mAfPatchHandle); 3130 removeAudioPatch(patchDesc->mHandle); 3131 nextAudioPortGeneration(); 3132 mpClientInterface->onAudioPatchListUpdate(); 3133 } else { 3134 return BAD_VALUE; 3135 } 3136 } else { 3137 return BAD_VALUE; 3138 } 3139 return NO_ERROR; 3140 } 3141 3142 status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches, 3143 struct audio_patch *patches, 3144 unsigned int *generation) 3145 { 3146 if (generation == NULL) { 3147 return BAD_VALUE; 3148 } 3149 *generation = curAudioPortGeneration(); 3150 return mAudioPatches.listAudioPatches(num_patches, patches); 3151 } 3152 3153 status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config) 3154 { 3155 ALOGV("setAudioPortConfig()"); 3156 3157 if (config == NULL) { 3158 return BAD_VALUE; 3159 } 3160 ALOGV("setAudioPortConfig() on port handle %d", config->id); 3161 // Only support gain configuration for now 3162 if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) { 3163 return INVALID_OPERATION; 3164 } 3165 3166 sp<AudioPortConfig> audioPortConfig; 3167 if (config->type == AUDIO_PORT_TYPE_MIX) { 3168 if (config->role == AUDIO_PORT_ROLE_SOURCE) { 3169 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id); 3170 if (outputDesc == NULL) { 3171 return BAD_VALUE; 3172 } 3173 ALOG_ASSERT(!outputDesc->isDuplicated(), 3174 "setAudioPortConfig() called on duplicated output %d", 3175 outputDesc->mIoHandle); 3176 audioPortConfig = outputDesc; 3177 } else if (config->role == AUDIO_PORT_ROLE_SINK) { 3178 sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(config->id); 3179 if (inputDesc == NULL) { 3180 return BAD_VALUE; 3181 } 3182 audioPortConfig = inputDesc; 3183 } else { 3184 return BAD_VALUE; 3185 } 3186 } else if (config->type == AUDIO_PORT_TYPE_DEVICE) { 3187 sp<DeviceDescriptor> deviceDesc; 3188 if (config->role == AUDIO_PORT_ROLE_SOURCE) { 3189 deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id); 3190 } else if (config->role == AUDIO_PORT_ROLE_SINK) { 3191 deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id); 3192 } else { 3193 return BAD_VALUE; 3194 } 3195 if (deviceDesc == NULL) { 3196 return BAD_VALUE; 3197 } 3198 audioPortConfig = deviceDesc; 3199 } else { 3200 return BAD_VALUE; 3201 } 3202 3203 struct audio_port_config backupConfig; 3204 status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig); 3205 if (status == NO_ERROR) { 3206 struct audio_port_config newConfig; 3207 audioPortConfig->toAudioPortConfig(&newConfig, config); 3208 status = mpClientInterface->setAudioPortConfig(&newConfig, 0); 3209 } 3210 if (status != NO_ERROR) { 3211 audioPortConfig->applyAudioPortConfig(&backupConfig); 3212 } 3213 3214 return status; 3215 } 3216 3217 void AudioPolicyManager::releaseResourcesForUid(uid_t uid) 3218 { 3219 clearAudioSources(uid); 3220 clearAudioPatches(uid); 3221 clearSessionRoutes(uid); 3222 } 3223 3224 void AudioPolicyManager::clearAudioPatches(uid_t uid) 3225 { 3226 for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) { 3227 sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i); 3228 if (patchDesc->mUid == uid) { 3229 releaseAudioPatch(mAudioPatches.keyAt(i), uid); 3230 } 3231 } 3232 } 3233 3234 void AudioPolicyManager::checkStrategyRoute(routing_strategy strategy, 3235 audio_io_handle_t ouptutToSkip) 3236 { 3237 audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); 3238 SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs); 3239 for (size_t j = 0; j < mOutputs.size(); j++) { 3240 if (mOutputs.keyAt(j) == ouptutToSkip) { 3241 continue; 3242 } 3243 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(j); 3244 if (!isStrategyActive(outputDesc, (routing_strategy)strategy)) { 3245 continue; 3246 } 3247 // If the default device for this strategy is on another output mix, 3248 // invalidate all tracks in this strategy to force re connection. 3249 // Otherwise select new device on the output mix. 3250 if (outputs.indexOf(mOutputs.keyAt(j)) < 0) { 3251 for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) { 3252 if (getStrategy((audio_stream_type_t)stream) == strategy) { 3253 mpClientInterface->invalidateStream((audio_stream_type_t)stream); 3254 } 3255 } 3256 } else { 3257 audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/); 3258 setOutputDevice(outputDesc, newDevice, false); 3259 } 3260 } 3261 } 3262 3263 void AudioPolicyManager::clearSessionRoutes(uid_t uid) 3264 { 3265 // remove output routes associated with this uid 3266 SortedVector<routing_strategy> affectedStrategies; 3267 for (ssize_t i = (ssize_t)mOutputRoutes.size() - 1; i >= 0; i--) { 3268 sp<SessionRoute> route = mOutputRoutes.valueAt(i); 3269 if (route->mUid == uid) { 3270 mOutputRoutes.removeItemsAt(i); 3271 if (route->mDeviceDescriptor != 0) { 3272 affectedStrategies.add(getStrategy(route->mStreamType)); 3273 } 3274 } 3275 } 3276 // reroute outputs if necessary 3277 for (size_t i = 0; i < affectedStrategies.size(); i++) { 3278 checkStrategyRoute(affectedStrategies[i], AUDIO_IO_HANDLE_NONE); 3279 } 3280 3281 // remove input routes associated with this uid 3282 SortedVector<audio_source_t> affectedSources; 3283 for (ssize_t i = (ssize_t)mInputRoutes.size() - 1; i >= 0; i--) { 3284 sp<SessionRoute> route = mInputRoutes.valueAt(i); 3285 if (route->mUid == uid) { 3286 mInputRoutes.removeItemsAt(i); 3287 if (route->mDeviceDescriptor != 0) { 3288 affectedSources.add(route->mSource); 3289 } 3290 } 3291 } 3292 // reroute inputs if necessary 3293 SortedVector<audio_io_handle_t> inputsToClose; 3294 for (size_t i = 0; i < mInputs.size(); i++) { 3295 sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(i); 3296 if (affectedSources.indexOf(inputDesc->inputSource()) >= 0) { 3297 inputsToClose.add(inputDesc->mIoHandle); 3298 } 3299 } 3300 for (size_t i = 0; i < inputsToClose.size(); i++) { 3301 closeInput(inputsToClose[i]); 3302 } 3303 } 3304 3305 void AudioPolicyManager::clearAudioSources(uid_t uid) 3306 { 3307 for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--) { 3308 sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i); 3309 if (sourceDesc->mUid == uid) { 3310 stopAudioSource(mAudioSources.keyAt(i)); 3311 } 3312 } 3313 } 3314 3315 status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session, 3316 audio_io_handle_t *ioHandle, 3317 audio_devices_t *device) 3318 { 3319 *session = (audio_session_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 3320 *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 3321 *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD); 3322 3323 return mSoundTriggerSessions.acquireSession(*session, *ioHandle); 3324 } 3325 3326 status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source, 3327 const audio_attributes_t *attributes, 3328 audio_patch_handle_t *handle, 3329 uid_t uid) 3330 { 3331 ALOGV("%s source %p attributes %p handle %p", __FUNCTION__, source, attributes, handle); 3332 if (source == NULL || attributes == NULL || handle == NULL) { 3333 return BAD_VALUE; 3334 } 3335 3336 *handle = AUDIO_PATCH_HANDLE_NONE; 3337 3338 if (source->role != AUDIO_PORT_ROLE_SOURCE || 3339 source->type != AUDIO_PORT_TYPE_DEVICE) { 3340 ALOGV("%s INVALID_OPERATION source->role %d source->type %d", __FUNCTION__, source->role, source->type); 3341 return INVALID_OPERATION; 3342 } 3343 3344 sp<DeviceDescriptor> srcDeviceDesc = 3345 mAvailableInputDevices.getDevice(source->ext.device.type, 3346 String8(source->ext.device.address)); 3347 if (srcDeviceDesc == 0) { 3348 ALOGV("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type); 3349 return BAD_VALUE; 3350 } 3351 sp<AudioSourceDescriptor> sourceDesc = 3352 new AudioSourceDescriptor(srcDeviceDesc, attributes, uid); 3353 3354 struct audio_patch dummyPatch; 3355 sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid); 3356 sourceDesc->mPatchDesc = patchDesc; 3357 3358 status_t status = connectAudioSource(sourceDesc); 3359 if (status == NO_ERROR) { 3360 mAudioSources.add(sourceDesc->getHandle(), sourceDesc); 3361 *handle = sourceDesc->getHandle(); 3362 } 3363 return status; 3364 } 3365 3366 status_t AudioPolicyManager::connectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc) 3367 { 3368 ALOGV("%s handle %d", __FUNCTION__, sourceDesc->getHandle()); 3369 3370 // make sure we only have one patch per source. 3371 disconnectAudioSource(sourceDesc); 3372 3373 routing_strategy strategy = (routing_strategy) getStrategyForAttr(&sourceDesc->mAttributes); 3374 audio_stream_type_t stream = streamTypefromAttributesInt(&sourceDesc->mAttributes); 3375 sp<DeviceDescriptor> srcDeviceDesc = sourceDesc->mDevice; 3376 3377 audio_devices_t sinkDevice = getDeviceForStrategy(strategy, true); 3378 sp<DeviceDescriptor> sinkDeviceDesc = 3379 mAvailableOutputDevices.getDevice(sinkDevice, String8("")); 3380 3381 audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; 3382 struct audio_patch *patch = &sourceDesc->mPatchDesc->mPatch; 3383 3384 if (srcDeviceDesc->getAudioPort()->mModule->getHandle() == 3385 sinkDeviceDesc->getAudioPort()->mModule->getHandle() && 3386 srcDeviceDesc->getAudioPort()->mModule->getHalVersionMajor() >= 3 && 3387 srcDeviceDesc->getAudioPort()->mGains.size() > 0) { 3388 ALOGV("%s AUDIO_DEVICE_API_VERSION_3_0", __FUNCTION__); 3389 // create patch between src device and output device 3390 // create Hwoutput and add to mHwOutputs 3391 } else { 3392 SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(sinkDevice, mOutputs); 3393 audio_io_handle_t output = 3394 selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); 3395 if (output == AUDIO_IO_HANDLE_NONE) { 3396 ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevice); 3397 return INVALID_OPERATION; 3398 } 3399 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); 3400 if (outputDesc->isDuplicated()) { 3401 ALOGV("%s output for device %08x is duplicated", __FUNCTION__, sinkDevice); 3402 return INVALID_OPERATION; 3403 } 3404 // create a special patch with no sink and two sources: 3405 // - the second source indicates to PatchPanel through which output mix this patch should 3406 // be connected as well as the stream type for volume control 3407 // - the sink is defined by whatever output device is currently selected for the output 3408 // though which this patch is routed. 3409 patch->num_sinks = 0; 3410 patch->num_sources = 2; 3411 srcDeviceDesc->toAudioPortConfig(&patch->sources[0], NULL); 3412 outputDesc->toAudioPortConfig(&patch->sources[1], NULL); 3413 patch->sources[1].ext.mix.usecase.stream = stream; 3414 status_t status = mpClientInterface->createAudioPatch(patch, 3415 &afPatchHandle, 3416 0); 3417 ALOGV("%s patch panel returned %d patchHandle %d", __FUNCTION__, 3418 status, afPatchHandle); 3419 if (status != NO_ERROR) { 3420 ALOGW("%s patch panel could not connect device patch, error %d", 3421 __FUNCTION__, status); 3422 return INVALID_OPERATION; 3423 } 3424 uint32_t delayMs = 0; 3425 status = startSource(outputDesc, stream, sinkDevice, NULL, &delayMs); 3426 3427 if (status != NO_ERROR) { 3428 mpClientInterface->releaseAudioPatch(sourceDesc->mPatchDesc->mAfPatchHandle, 0); 3429 return status; 3430 } 3431 sourceDesc->mSwOutput = outputDesc; 3432 if (delayMs != 0) { 3433 usleep(delayMs * 1000); 3434 } 3435 } 3436 3437 sourceDesc->mPatchDesc->mAfPatchHandle = afPatchHandle; 3438 addAudioPatch(sourceDesc->mPatchDesc->mHandle, sourceDesc->mPatchDesc); 3439 3440 return NO_ERROR; 3441 } 3442 3443 status_t AudioPolicyManager::stopAudioSource(audio_patch_handle_t handle __unused) 3444 { 3445 sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueFor(handle); 3446 ALOGV("%s handle %d", __FUNCTION__, handle); 3447 if (sourceDesc == 0) { 3448 ALOGW("%s unknown source for handle %d", __FUNCTION__, handle); 3449 return BAD_VALUE; 3450 } 3451 status_t status = disconnectAudioSource(sourceDesc); 3452 3453 mAudioSources.removeItem(handle); 3454 return status; 3455 } 3456 3457 status_t AudioPolicyManager::setMasterMono(bool mono) 3458 { 3459 if (mMasterMono == mono) { 3460 return NO_ERROR; 3461 } 3462 mMasterMono = mono; 3463 // if enabling mono we close all offloaded devices, which will invalidate the 3464 // corresponding AudioTrack. The AudioTrack client/MediaPlayer is responsible 3465 // for recreating the new AudioTrack as non-offloaded PCM. 3466 // 3467 // If disabling mono, we leave all tracks as is: we don't know which clients 3468 // and tracks are able to be recreated as offloaded. The next "song" should 3469 // play back offloaded. 3470 if (mMasterMono) { 3471 Vector<audio_io_handle_t> offloaded; 3472 for (size_t i = 0; i < mOutputs.size(); ++i) { 3473 sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); 3474 if (desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 3475 offloaded.push(desc->mIoHandle); 3476 } 3477 } 3478 for (size_t i = 0; i < offloaded.size(); ++i) { 3479 closeOutput(offloaded[i]); 3480 } 3481 } 3482 // update master mono for all remaining outputs 3483 for (size_t i = 0; i < mOutputs.size(); ++i) { 3484 updateMono(mOutputs.keyAt(i)); 3485 } 3486 return NO_ERROR; 3487 } 3488 3489 status_t AudioPolicyManager::getMasterMono(bool *mono) 3490 { 3491 *mono = mMasterMono; 3492 return NO_ERROR; 3493 } 3494 3495 float AudioPolicyManager::getStreamVolumeDB( 3496 audio_stream_type_t stream, int index, audio_devices_t device) 3497 { 3498 return computeVolume(stream, index, device); 3499 } 3500 3501 status_t AudioPolicyManager::disconnectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc) 3502 { 3503 ALOGV("%s handle %d", __FUNCTION__, sourceDesc->getHandle()); 3504 3505 sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->mPatchDesc->mHandle); 3506 if (patchDesc == 0) { 3507 ALOGW("%s source has no patch with handle %d", __FUNCTION__, 3508 sourceDesc->mPatchDesc->mHandle); 3509 return BAD_VALUE; 3510 } 3511 removeAudioPatch(sourceDesc->mPatchDesc->mHandle); 3512 3513 audio_stream_type_t stream = streamTypefromAttributesInt(&sourceDesc->mAttributes); 3514 sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->mSwOutput.promote(); 3515 if (swOutputDesc != 0) { 3516 stopSource(swOutputDesc, stream, false); 3517 mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); 3518 } else { 3519 sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->mHwOutput.promote(); 3520 if (hwOutputDesc != 0) { 3521 // release patch between src device and output device 3522 // close Hwoutput and remove from mHwOutputs 3523 } else { 3524 ALOGW("%s source has neither SW nor HW output", __FUNCTION__); 3525 } 3526 } 3527 return NO_ERROR; 3528 } 3529 3530 sp<AudioSourceDescriptor> AudioPolicyManager::getSourceForStrategyOnOutput( 3531 audio_io_handle_t output, routing_strategy strategy) 3532 { 3533 sp<AudioSourceDescriptor> source; 3534 for (size_t i = 0; i < mAudioSources.size(); i++) { 3535 sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i); 3536 routing_strategy sourceStrategy = 3537 (routing_strategy) getStrategyForAttr(&sourceDesc->mAttributes); 3538 sp<SwAudioOutputDescriptor> outputDesc = sourceDesc->mSwOutput.promote(); 3539 if (sourceStrategy == strategy && outputDesc != 0 && outputDesc->mIoHandle == output) { 3540 source = sourceDesc; 3541 break; 3542 } 3543 } 3544 return source; 3545 } 3546 3547 // ---------------------------------------------------------------------------- 3548 // AudioPolicyManager 3549 // ---------------------------------------------------------------------------- 3550 uint32_t AudioPolicyManager::nextAudioPortGeneration() 3551 { 3552 return android_atomic_inc(&mAudioPortGeneration); 3553 } 3554 3555 #ifdef USE_XML_AUDIO_POLICY_CONF 3556 // Treblized audio policy xml config will be located in /odm/etc or /vendor/etc. 3557 static const char *kConfigLocationList[] = 3558 {"/odm/etc", "/vendor/etc", "/system/etc"}; 3559 static const int kConfigLocationListSize = 3560 (sizeof(kConfigLocationList) / sizeof(kConfigLocationList[0])); 3561 3562 static status_t deserializeAudioPolicyXmlConfig(AudioPolicyConfig &config) { 3563 char audioPolicyXmlConfigFile[AUDIO_POLICY_XML_CONFIG_FILE_PATH_MAX_LENGTH]; 3564 status_t ret; 3565 3566 for (int i = 0; i < kConfigLocationListSize; i++) { 3567 PolicySerializer serializer; 3568 snprintf(audioPolicyXmlConfigFile, 3569 sizeof(audioPolicyXmlConfigFile), 3570 "%s/%s", 3571 kConfigLocationList[i], 3572 AUDIO_POLICY_XML_CONFIG_FILE_NAME); 3573 ret = serializer.deserialize(audioPolicyXmlConfigFile, config); 3574 if (ret == NO_ERROR) { 3575 break; 3576 } 3577 } 3578 return ret; 3579 } 3580 #endif 3581 3582 AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface) 3583 : 3584 #ifdef AUDIO_POLICY_TEST 3585 Thread(false), 3586 #endif //AUDIO_POLICY_TEST 3587 mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), 3588 mA2dpSuspended(false), 3589 mAudioPortGeneration(1), 3590 mBeaconMuteRefCount(0), 3591 mBeaconPlayingRefCount(0), 3592 mBeaconMuted(false), 3593 mTtsOutputAvailable(false), 3594 mMasterMono(false), 3595 mMusicEffectOutput(AUDIO_IO_HANDLE_NONE), 3596 mHasComputedSoundTriggerSupportsConcurrentCapture(false) 3597 { 3598 mUidCached = getuid(); 3599 mpClientInterface = clientInterface; 3600 3601 // TODO: remove when legacy conf file is removed. true on devices that use DRC on the 3602 // DEVICE_CATEGORY_SPEAKER path to boost soft sounds, used to adjust volume curves accordingly. 3603 // Note: remove also speaker_drc_enabled from global configuration of XML config file. 3604 bool speakerDrcEnabled = false; 3605 3606 #ifdef USE_XML_AUDIO_POLICY_CONF 3607 mVolumeCurves = new VolumeCurvesCollection(); 3608 AudioPolicyConfig config(mHwModules, mAvailableOutputDevices, mAvailableInputDevices, 3609 mDefaultOutputDevice, speakerDrcEnabled, 3610 static_cast<VolumeCurvesCollection *>(mVolumeCurves)); 3611 if (deserializeAudioPolicyXmlConfig(config) != NO_ERROR) { 3612 #else 3613 mVolumeCurves = new StreamDescriptorCollection(); 3614 AudioPolicyConfig config(mHwModules, mAvailableOutputDevices, mAvailableInputDevices, 3615 mDefaultOutputDevice, speakerDrcEnabled); 3616 if ((ConfigParsingUtils::loadConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE, config) != NO_ERROR) && 3617 (ConfigParsingUtils::loadConfig(AUDIO_POLICY_CONFIG_FILE, config) != NO_ERROR)) { 3618 #endif 3619 ALOGE("could not load audio policy configuration file, setting defaults"); 3620 config.setDefault(); 3621 } 3622 // must be done after reading the policy (since conditionned by Speaker Drc Enabling) 3623 mVolumeCurves->initializeVolumeCurves(speakerDrcEnabled); 3624 3625 // Once policy config has been parsed, retrieve an instance of the engine and initialize it. 3626 audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance(); 3627 if (!engineInstance) { 3628 ALOGE("%s: Could not get an instance of policy engine", __FUNCTION__); 3629 return; 3630 } 3631 // Retrieve the Policy Manager Interface 3632 mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>(); 3633 if (mEngine == NULL) { 3634 ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__); 3635 return; 3636 } 3637 mEngine->setObserver(this); 3638 status_t status = mEngine->initCheck(); 3639 (void) status; 3640 ALOG_ASSERT(status == NO_ERROR, "Policy engine not initialized(err=%d)", status); 3641 3642 // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices 3643 // open all output streams needed to access attached devices 3644 audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types(); 3645 audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; 3646 for (size_t i = 0; i < mHwModules.size(); i++) { 3647 mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->getName()); 3648 if (mHwModules[i]->mHandle == 0) { 3649 ALOGW("could not open HW module %s", mHwModules[i]->getName()); 3650 continue; 3651 } 3652 // open all output streams needed to access attached devices 3653 // except for direct output streams that are only opened when they are actually 3654 // required by an app. 3655 // This also validates mAvailableOutputDevices list 3656 for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) 3657 { 3658 const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j]; 3659 3660 if (!outProfile->hasSupportedDevices()) { 3661 ALOGW("Output profile contains no device on module %s", mHwModules[i]->getName()); 3662 continue; 3663 } 3664 if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_TTS) != 0) { 3665 mTtsOutputAvailable = true; 3666 } 3667 3668 if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { 3669 continue; 3670 } 3671 audio_devices_t profileType = outProfile->getSupportedDevicesType(); 3672 if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) { 3673 profileType = mDefaultOutputDevice->type(); 3674 } else { 3675 // chose first device present in profile's SupportedDevices also part of 3676 // outputDeviceTypes 3677 profileType = outProfile->getSupportedDeviceForType(outputDeviceTypes); 3678 } 3679 if ((profileType & outputDeviceTypes) == 0) { 3680 continue; 3681 } 3682 sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile, 3683 mpClientInterface); 3684 const DeviceVector &supportedDevices = outProfile->getSupportedDevices(); 3685 const DeviceVector &devicesForType = supportedDevices.getDevicesFromType(profileType); 3686 String8 address = devicesForType.size() > 0 ? devicesForType.itemAt(0)->mAddress 3687 : String8(""); 3688 3689 outputDesc->mDevice = profileType; 3690 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 3691 config.sample_rate = outputDesc->mSamplingRate; 3692 config.channel_mask = outputDesc->mChannelMask; 3693 config.format = outputDesc->mFormat; 3694 audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; 3695 status_t status = mpClientInterface->openOutput(outProfile->getModuleHandle(), 3696 &output, 3697 &config, 3698 &outputDesc->mDevice, 3699 address, 3700 &outputDesc->mLatency, 3701 outputDesc->mFlags); 3702 3703 if (status != NO_ERROR) { 3704 ALOGW("Cannot open output stream for device %08x on hw module %s", 3705 outputDesc->mDevice, 3706 mHwModules[i]->getName()); 3707 } else { 3708 outputDesc->mSamplingRate = config.sample_rate; 3709 outputDesc->mChannelMask = config.channel_mask; 3710 outputDesc->mFormat = config.format; 3711 3712 for (size_t k = 0; k < supportedDevices.size(); k++) { 3713 ssize_t index = mAvailableOutputDevices.indexOf(supportedDevices[k]); 3714 // give a valid ID to an attached device once confirmed it is reachable 3715 if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) { 3716 mAvailableOutputDevices[index]->attach(mHwModules[i]); 3717 } 3718 } 3719 if (mPrimaryOutput == 0 && 3720 outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) { 3721 mPrimaryOutput = outputDesc; 3722 } 3723 addOutput(output, outputDesc); 3724 setOutputDevice(outputDesc, 3725 outputDesc->mDevice, 3726 true, 3727 0, 3728 NULL, 3729 address.string()); 3730 } 3731 } 3732 // open input streams needed to access attached devices to validate 3733 // mAvailableInputDevices list 3734 for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) 3735 { 3736 const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j]; 3737 3738 if (!inProfile->hasSupportedDevices()) { 3739 ALOGW("Input profile contains no device on module %s", mHwModules[i]->getName()); 3740 continue; 3741 } 3742 // chose first device present in profile's SupportedDevices also part of 3743 // inputDeviceTypes 3744 audio_devices_t profileType = inProfile->getSupportedDeviceForType(inputDeviceTypes); 3745 3746 if ((profileType & inputDeviceTypes) == 0) { 3747 continue; 3748 } 3749 sp<AudioInputDescriptor> inputDesc = 3750 new AudioInputDescriptor(inProfile); 3751 3752 inputDesc->mDevice = profileType; 3753 3754 // find the address 3755 DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType); 3756 // the inputs vector must be of size 1, but we don't want to crash here 3757 String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress 3758 : String8(""); 3759 ALOGV(" for input device 0x%x using address %s", profileType, address.string()); 3760 ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!"); 3761 3762 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 3763 config.sample_rate = inputDesc->mSamplingRate; 3764 config.channel_mask = inputDesc->mChannelMask; 3765 config.format = inputDesc->mFormat; 3766 audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; 3767 status_t status = mpClientInterface->openInput(inProfile->getModuleHandle(), 3768 &input, 3769 &config, 3770 &inputDesc->mDevice, 3771 address, 3772 AUDIO_SOURCE_MIC, 3773 AUDIO_INPUT_FLAG_NONE); 3774 3775 if (status == NO_ERROR) { 3776 const DeviceVector &supportedDevices = inProfile->getSupportedDevices(); 3777 for (size_t k = 0; k < supportedDevices.size(); k++) { 3778 ssize_t index = mAvailableInputDevices.indexOf(supportedDevices[k]); 3779 // give a valid ID to an attached device once confirmed it is reachable 3780 if (index >= 0) { 3781 sp<DeviceDescriptor> devDesc = mAvailableInputDevices[index]; 3782 if (!devDesc->isAttached()) { 3783 devDesc->attach(mHwModules[i]); 3784 devDesc->importAudioPort(inProfile, true); 3785 } 3786 } 3787 } 3788 mpClientInterface->closeInput(input); 3789 } else { 3790 ALOGW("Cannot open input stream for device %08x on hw module %s", 3791 inputDesc->mDevice, 3792 mHwModules[i]->getName()); 3793 } 3794 } 3795 } 3796 // make sure all attached devices have been allocated a unique ID 3797 for (size_t i = 0; i < mAvailableOutputDevices.size();) { 3798 if (!mAvailableOutputDevices[i]->isAttached()) { 3799 ALOGW("Output device %08x unreachable", mAvailableOutputDevices[i]->type()); 3800 mAvailableOutputDevices.remove(mAvailableOutputDevices[i]); 3801 continue; 3802 } 3803 // The device is now validated and can be appended to the available devices of the engine 3804 mEngine->setDeviceConnectionState(mAvailableOutputDevices[i], 3805 AUDIO_POLICY_DEVICE_STATE_AVAILABLE); 3806 i++; 3807 } 3808 for (size_t i = 0; i < mAvailableInputDevices.size();) { 3809 if (!mAvailableInputDevices[i]->isAttached()) { 3810 ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type()); 3811 mAvailableInputDevices.remove(mAvailableInputDevices[i]); 3812 continue; 3813 } 3814 // The device is now validated and can be appended to the available devices of the engine 3815 mEngine->setDeviceConnectionState(mAvailableInputDevices[i], 3816 AUDIO_POLICY_DEVICE_STATE_AVAILABLE); 3817 i++; 3818 } 3819 // make sure default device is reachable 3820 if (mDefaultOutputDevice == 0 || mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) { 3821 ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type()); 3822 } 3823 3824 ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output"); 3825 3826 updateDevicesAndOutputs(); 3827 3828 #ifdef AUDIO_POLICY_TEST 3829 if (mPrimaryOutput != 0) { 3830 AudioParameter outputCmd = AudioParameter(); 3831 outputCmd.addInt(String8("set_id"), 0); 3832 mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, outputCmd.toString()); 3833 3834 mTestDevice = AUDIO_DEVICE_OUT_SPEAKER; 3835 mTestSamplingRate = 44100; 3836 mTestFormat = AUDIO_FORMAT_PCM_16_BIT; 3837 mTestChannels = AUDIO_CHANNEL_OUT_STEREO; 3838 mTestLatencyMs = 0; 3839 mCurOutput = 0; 3840 mDirectOutput = false; 3841 for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { 3842 mTestOutputs[i] = 0; 3843 } 3844 3845 const size_t SIZE = 256; 3846 char buffer[SIZE]; 3847 snprintf(buffer, SIZE, "AudioPolicyManagerTest"); 3848 run(buffer, ANDROID_PRIORITY_AUDIO); 3849 } 3850 #endif //AUDIO_POLICY_TEST 3851 } 3852 3853 AudioPolicyManager::~AudioPolicyManager() 3854 { 3855 #ifdef AUDIO_POLICY_TEST 3856 exit(); 3857 #endif //AUDIO_POLICY_TEST 3858 for (size_t i = 0; i < mOutputs.size(); i++) { 3859 mpClientInterface->closeOutput(mOutputs.keyAt(i)); 3860 } 3861 for (size_t i = 0; i < mInputs.size(); i++) { 3862 mpClientInterface->closeInput(mInputs.keyAt(i)); 3863 } 3864 mAvailableOutputDevices.clear(); 3865 mAvailableInputDevices.clear(); 3866 mOutputs.clear(); 3867 mInputs.clear(); 3868 mHwModules.clear(); 3869 } 3870 3871 status_t AudioPolicyManager::initCheck() 3872 { 3873 return hasPrimaryOutput() ? NO_ERROR : NO_INIT; 3874 } 3875 3876 #ifdef AUDIO_POLICY_TEST 3877 bool AudioPolicyManager::threadLoop() 3878 { 3879 ALOGV("entering threadLoop()"); 3880 while (!exitPending()) 3881 { 3882 String8 command; 3883 int valueInt; 3884 String8 value; 3885 3886 Mutex::Autolock _l(mLock); 3887 mWaitWorkCV.waitRelative(mLock, milliseconds(50)); 3888 3889 command = mpClientInterface->getParameters(0, String8("test_cmd_policy")); 3890 AudioParameter param = AudioParameter(command); 3891 3892 if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR && 3893 valueInt != 0) { 3894 ALOGV("Test command %s received", command.string()); 3895 String8 target; 3896 if (param.get(String8("target"), target) != NO_ERROR) { 3897 target = "Manager"; 3898 } 3899 if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) { 3900 param.remove(String8("test_cmd_policy_output")); 3901 mCurOutput = valueInt; 3902 } 3903 if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) { 3904 param.remove(String8("test_cmd_policy_direct")); 3905 if (value == "false") { 3906 mDirectOutput = false; 3907 } else if (value == "true") { 3908 mDirectOutput = true; 3909 } 3910 } 3911 if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) { 3912 param.remove(String8("test_cmd_policy_input")); 3913 mTestInput = valueInt; 3914 } 3915 3916 if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) { 3917 param.remove(String8("test_cmd_policy_format")); 3918 int format = AUDIO_FORMAT_INVALID; 3919 if (value == "PCM 16 bits") { 3920 format = AUDIO_FORMAT_PCM_16_BIT; 3921 } else if (value == "PCM 8 bits") { 3922 format = AUDIO_FORMAT_PCM_8_BIT; 3923 } else if (value == "Compressed MP3") { 3924 format = AUDIO_FORMAT_MP3; 3925 } 3926 if (format != AUDIO_FORMAT_INVALID) { 3927 if (target == "Manager") { 3928 mTestFormat = format; 3929 } else if (mTestOutputs[mCurOutput] != 0) { 3930 AudioParameter outputParam = AudioParameter(); 3931 outputParam.addInt(String8(AudioParameter::keyStreamSupportedFormats), format); 3932 mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); 3933 } 3934 } 3935 } 3936 if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) { 3937 param.remove(String8("test_cmd_policy_channels")); 3938 int channels = 0; 3939 3940 if (value == "Channels Stereo") { 3941 channels = AUDIO_CHANNEL_OUT_STEREO; 3942 } else if (value == "Channels Mono") { 3943 channels = AUDIO_CHANNEL_OUT_MONO; 3944 } 3945 if (channels != 0) { 3946 if (target == "Manager") { 3947 mTestChannels = channels; 3948 } else if (mTestOutputs[mCurOutput] != 0) { 3949 AudioParameter outputParam = AudioParameter(); 3950 outputParam.addInt(String8(AudioParameter::keyStreamSupportedChannels), channels); 3951 mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); 3952 } 3953 } 3954 } 3955 if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) { 3956 param.remove(String8("test_cmd_policy_sampleRate")); 3957 if (valueInt >= 0 && valueInt <= 96000) { 3958 int samplingRate = valueInt; 3959 if (target == "Manager") { 3960 mTestSamplingRate = samplingRate; 3961 } else if (mTestOutputs[mCurOutput] != 0) { 3962 AudioParameter outputParam = AudioParameter(); 3963 outputParam.addInt(String8(AudioParameter::keyStreamSupportedSamplingRates), samplingRate); 3964 mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); 3965 } 3966 } 3967 } 3968 3969 if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) { 3970 param.remove(String8("test_cmd_policy_reopen")); 3971 3972 mpClientInterface->closeOutput(mpClientInterface->closeOutput(mPrimaryOutput);); 3973 3974 audio_module_handle_t moduleHandle = mPrimaryOutput->getModuleHandle(); 3975 3976 removeOutput(mPrimaryOutput->mIoHandle); 3977 sp<SwAudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL, 3978 mpClientInterface); 3979 outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; 3980 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 3981 config.sample_rate = outputDesc->mSamplingRate; 3982 config.channel_mask = outputDesc->mChannelMask; 3983 config.format = outputDesc->mFormat; 3984 audio_io_handle_t handle; 3985 status_t status = mpClientInterface->openOutput(moduleHandle, 3986 &handle, 3987 &config, 3988 &outputDesc->mDevice, 3989 String8(""), 3990 &outputDesc->mLatency, 3991 outputDesc->mFlags); 3992 if (status != NO_ERROR) { 3993 ALOGE("Failed to reopen hardware output stream, " 3994 "samplingRate: %d, format %d, channels %d", 3995 outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask); 3996 } else { 3997 outputDesc->mSamplingRate = config.sample_rate; 3998 outputDesc->mChannelMask = config.channel_mask; 3999 outputDesc->mFormat = config.format; 4000 mPrimaryOutput = outputDesc; 4001 AudioParameter outputCmd = AudioParameter(); 4002 outputCmd.addInt(String8("set_id"), 0); 4003 mpClientInterface->setParameters(handle, outputCmd.toString()); 4004 addOutput(handle, outputDesc); 4005 } 4006 } 4007 4008 4009 mpClientInterface->setParameters(0, String8("test_cmd_policy=")); 4010 } 4011 } 4012 return false; 4013 } 4014 4015 void AudioPolicyManager::exit() 4016 { 4017 { 4018 AutoMutex _l(mLock); 4019 requestExit(); 4020 mWaitWorkCV.signal(); 4021 } 4022 requestExitAndWait(); 4023 } 4024 4025 int AudioPolicyManager::testOutputIndex(audio_io_handle_t output) 4026 { 4027 for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { 4028 if (output == mTestOutputs[i]) return i; 4029 } 4030 return 0; 4031 } 4032 #endif //AUDIO_POLICY_TEST 4033 4034 // --- 4035 4036 void AudioPolicyManager::addOutput(audio_io_handle_t output, const sp<SwAudioOutputDescriptor>& outputDesc) 4037 { 4038 outputDesc->setIoHandle(output); 4039 mOutputs.add(output, outputDesc); 4040 updateMono(output); // update mono status when adding to output list 4041 selectOutputForMusicEffects(); 4042 nextAudioPortGeneration(); 4043 } 4044 4045 void AudioPolicyManager::removeOutput(audio_io_handle_t output) 4046 { 4047 mOutputs.removeItem(output); 4048 selectOutputForMusicEffects(); 4049 } 4050 4051 void AudioPolicyManager::addInput(audio_io_handle_t input, const sp<AudioInputDescriptor>& inputDesc) 4052 { 4053 inputDesc->setIoHandle(input); 4054 mInputs.add(input, inputDesc); 4055 nextAudioPortGeneration(); 4056 } 4057 4058 void AudioPolicyManager::findIoHandlesByAddress(const sp<SwAudioOutputDescriptor>& desc /*in*/, 4059 const audio_devices_t device /*in*/, 4060 const String8& address /*in*/, 4061 SortedVector<audio_io_handle_t>& outputs /*out*/) { 4062 sp<DeviceDescriptor> devDesc = 4063 desc->mProfile->getSupportedDeviceByAddress(device, address); 4064 if (devDesc != 0) { 4065 ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s", 4066 desc->mIoHandle, address.string()); 4067 outputs.add(desc->mIoHandle); 4068 } 4069 } 4070 4071 status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor>& devDesc, 4072 audio_policy_dev_state_t state, 4073 SortedVector<audio_io_handle_t>& outputs, 4074 const String8& address) 4075 { 4076 audio_devices_t device = devDesc->type(); 4077 sp<SwAudioOutputDescriptor> desc; 4078 4079 if (audio_device_is_digital(device)) { 4080 // erase all current sample rates, formats and channel masks 4081 devDesc->clearAudioProfiles(); 4082 } 4083 4084 if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { 4085 // first list already open outputs that can be routed to this device 4086 for (size_t i = 0; i < mOutputs.size(); i++) { 4087 desc = mOutputs.valueAt(i); 4088 if (!desc->isDuplicated() && (desc->supportedDevices() & device)) { 4089 if (!device_distinguishes_on_address(device)) { 4090 ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); 4091 outputs.add(mOutputs.keyAt(i)); 4092 } else { 4093 ALOGV(" checking address match due to device 0x%x", device); 4094 findIoHandlesByAddress(desc, device, address, outputs); 4095 } 4096 } 4097 } 4098 // then look for output profiles that can be routed to this device 4099 SortedVector< sp<IOProfile> > profiles; 4100 for (size_t i = 0; i < mHwModules.size(); i++) 4101 { 4102 if (mHwModules[i]->mHandle == 0) { 4103 continue; 4104 } 4105 for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) 4106 { 4107 sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j]; 4108 if (profile->supportDevice(device)) { 4109 if (!device_distinguishes_on_address(device) || 4110 profile->supportDeviceAddress(address)) { 4111 profiles.add(profile); 4112 ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i); 4113 } 4114 } 4115 } 4116 } 4117 4118 ALOGV(" found %zu profiles, %zu outputs", profiles.size(), outputs.size()); 4119 4120 if (profiles.isEmpty() && outputs.isEmpty()) { 4121 ALOGW("checkOutputsForDevice(): No output available for device %04x", device); 4122 return BAD_VALUE; 4123 } 4124 4125 // open outputs for matching profiles if needed. Direct outputs are also opened to 4126 // query for dynamic parameters and will be closed later by setDeviceConnectionState() 4127 for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { 4128 sp<IOProfile> profile = profiles[profile_index]; 4129 4130 // nothing to do if one output is already opened for this profile 4131 size_t j; 4132 for (j = 0; j < outputs.size(); j++) { 4133 desc = mOutputs.valueFor(outputs.itemAt(j)); 4134 if (!desc->isDuplicated() && desc->mProfile == profile) { 4135 // matching profile: save the sample rates, format and channel masks supported 4136 // by the profile in our device descriptor 4137 if (audio_device_is_digital(device)) { 4138 devDesc->importAudioPort(profile); 4139 } 4140 break; 4141 } 4142 } 4143 if (j != outputs.size()) { 4144 continue; 4145 } 4146 4147 ALOGV("opening output for device %08x with params %s profile %p name %s", 4148 device, address.string(), profile.get(), profile->getName().string()); 4149 desc = new SwAudioOutputDescriptor(profile, mpClientInterface); 4150 desc->mDevice = device; 4151 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 4152 config.sample_rate = desc->mSamplingRate; 4153 config.channel_mask = desc->mChannelMask; 4154 config.format = desc->mFormat; 4155 config.offload_info.sample_rate = desc->mSamplingRate; 4156 config.offload_info.channel_mask = desc->mChannelMask; 4157 config.offload_info.format = desc->mFormat; 4158 audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; 4159 status_t status = mpClientInterface->openOutput(profile->getModuleHandle(), 4160 &output, 4161 &config, 4162 &desc->mDevice, 4163 address, 4164 &desc->mLatency, 4165 desc->mFlags); 4166 if (status == NO_ERROR) { 4167 desc->mSamplingRate = config.sample_rate; 4168 desc->mChannelMask = config.channel_mask; 4169 desc->mFormat = config.format; 4170 4171 // Here is where the out_set_parameters() for card & device gets called 4172 if (!address.isEmpty()) { 4173 char *param = audio_device_address_to_parameter(device, address); 4174 mpClientInterface->setParameters(output, String8(param)); 4175 free(param); 4176 } 4177 updateAudioProfiles(device, output, profile->getAudioProfiles()); 4178 if (!profile->hasValidAudioProfile()) { 4179 ALOGW("checkOutputsForDevice() missing param"); 4180 mpClientInterface->closeOutput(output); 4181 output = AUDIO_IO_HANDLE_NONE; 4182 } else if (profile->hasDynamicAudioProfile()) { 4183 mpClientInterface->closeOutput(output); 4184 output = AUDIO_IO_HANDLE_NONE; 4185 profile->pickAudioProfile(config.sample_rate, config.channel_mask, config.format); 4186 config.offload_info.sample_rate = config.sample_rate; 4187 config.offload_info.channel_mask = config.channel_mask; 4188 config.offload_info.format = config.format; 4189 status = mpClientInterface->openOutput(profile->getModuleHandle(), 4190 &output, 4191 &config, 4192 &desc->mDevice, 4193 address, 4194 &desc->mLatency, 4195 desc->mFlags); 4196 if (status == NO_ERROR) { 4197 desc->mSamplingRate = config.sample_rate; 4198 desc->mChannelMask = config.channel_mask; 4199 desc->mFormat = config.format; 4200 } else { 4201 output = AUDIO_IO_HANDLE_NONE; 4202 } 4203 } 4204 4205 if (output != AUDIO_IO_HANDLE_NONE) { 4206 addOutput(output, desc); 4207 if (device_distinguishes_on_address(device) && address != "0") { 4208 sp<AudioPolicyMix> policyMix; 4209 if (mPolicyMixes.getAudioPolicyMix(address, policyMix) != NO_ERROR) { 4210 ALOGE("checkOutputsForDevice() cannot find policy for address %s", 4211 address.string()); 4212 } 4213 policyMix->setOutput(desc); 4214 desc->mPolicyMix = policyMix->getMix(); 4215 4216 } else if (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) && 4217 hasPrimaryOutput()) { 4218 // no duplicated output for direct outputs and 4219 // outputs used by dynamic policy mixes 4220 audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE; 4221 4222 // set initial stream volume for device 4223 applyStreamVolumes(desc, device, 0, true); 4224 4225 //TODO: configure audio effect output stage here 4226 4227 // open a duplicating output thread for the new output and the primary output 4228 duplicatedOutput = 4229 mpClientInterface->openDuplicateOutput(output, 4230 mPrimaryOutput->mIoHandle); 4231 if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) { 4232 // add duplicated output descriptor 4233 sp<SwAudioOutputDescriptor> dupOutputDesc = 4234 new SwAudioOutputDescriptor(NULL, mpClientInterface); 4235 dupOutputDesc->mOutput1 = mPrimaryOutput; 4236 dupOutputDesc->mOutput2 = desc; 4237 dupOutputDesc->mSamplingRate = desc->mSamplingRate; 4238 dupOutputDesc->mFormat = desc->mFormat; 4239 dupOutputDesc->mChannelMask = desc->mChannelMask; 4240 dupOutputDesc->mLatency = desc->mLatency; 4241 addOutput(duplicatedOutput, dupOutputDesc); 4242 applyStreamVolumes(dupOutputDesc, device, 0, true); 4243 } else { 4244 ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", 4245 mPrimaryOutput->mIoHandle, output); 4246 mpClientInterface->closeOutput(output); 4247 removeOutput(output); 4248 nextAudioPortGeneration(); 4249 output = AUDIO_IO_HANDLE_NONE; 4250 } 4251 } 4252 } 4253 } else { 4254 output = AUDIO_IO_HANDLE_NONE; 4255 } 4256 if (output == AUDIO_IO_HANDLE_NONE) { 4257 ALOGW("checkOutputsForDevice() could not open output for device %x", device); 4258 profiles.removeAt(profile_index); 4259 profile_index--; 4260 } else { 4261 outputs.add(output); 4262 // Load digital format info only for digital devices 4263 if (audio_device_is_digital(device)) { 4264 devDesc->importAudioPort(profile); 4265 } 4266 4267 if (device_distinguishes_on_address(device)) { 4268 ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)", 4269 device, address.string()); 4270 setOutputDevice(desc, device, true/*force*/, 0/*delay*/, 4271 NULL/*patch handle*/, address.string()); 4272 } 4273 ALOGV("checkOutputsForDevice(): adding output %d", output); 4274 } 4275 } 4276 4277 if (profiles.isEmpty()) { 4278 ALOGW("checkOutputsForDevice(): No output available for device %04x", device); 4279 return BAD_VALUE; 4280 } 4281 } else { // Disconnect 4282 // check if one opened output is not needed any more after disconnecting one device 4283 for (size_t i = 0; i < mOutputs.size(); i++) { 4284 desc = mOutputs.valueAt(i); 4285 if (!desc->isDuplicated()) { 4286 // exact match on device 4287 if (device_distinguishes_on_address(device) && 4288 (desc->supportedDevices() == device)) { 4289 findIoHandlesByAddress(desc, device, address, outputs); 4290 } else if (!(desc->supportedDevices() & mAvailableOutputDevices.types())) { 4291 ALOGV("checkOutputsForDevice(): disconnecting adding output %d", 4292 mOutputs.keyAt(i)); 4293 outputs.add(mOutputs.keyAt(i)); 4294 } 4295 } 4296 } 4297 // Clear any profiles associated with the disconnected device. 4298 for (size_t i = 0; i < mHwModules.size(); i++) 4299 { 4300 if (mHwModules[i]->mHandle == 0) { 4301 continue; 4302 } 4303 for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) 4304 { 4305 sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j]; 4306 if (profile->supportDevice(device)) { 4307 ALOGV("checkOutputsForDevice(): " 4308 "clearing direct output profile %zu on module %zu", j, i); 4309 profile->clearAudioProfiles(); 4310 } 4311 } 4312 } 4313 } 4314 return NO_ERROR; 4315 } 4316 4317 status_t AudioPolicyManager::checkInputsForDevice(const sp<DeviceDescriptor>& devDesc, 4318 audio_policy_dev_state_t state, 4319 SortedVector<audio_io_handle_t>& inputs, 4320 const String8& address) 4321 { 4322 audio_devices_t device = devDesc->type(); 4323 sp<AudioInputDescriptor> desc; 4324 4325 if (audio_device_is_digital(device)) { 4326 // erase all current sample rates, formats and channel masks 4327 devDesc->clearAudioProfiles(); 4328 } 4329 4330 if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { 4331 // first list already open inputs that can be routed to this device 4332 for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { 4333 desc = mInputs.valueAt(input_index); 4334 if (desc->mProfile->supportDevice(device)) { 4335 ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index)); 4336 inputs.add(mInputs.keyAt(input_index)); 4337 } 4338 } 4339 4340 // then look for input profiles that can be routed to this device 4341 SortedVector< sp<IOProfile> > profiles; 4342 for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++) 4343 { 4344 if (mHwModules[module_idx]->mHandle == 0) { 4345 continue; 4346 } 4347 for (size_t profile_index = 0; 4348 profile_index < mHwModules[module_idx]->mInputProfiles.size(); 4349 profile_index++) 4350 { 4351 sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index]; 4352 4353 if (profile->supportDevice(device)) { 4354 if (!device_distinguishes_on_address(device) || 4355 profile->supportDeviceAddress(address)) { 4356 profiles.add(profile); 4357 ALOGV("checkInputsForDevice(): adding profile %zu from module %zu", 4358 profile_index, module_idx); 4359 } 4360 } 4361 } 4362 } 4363 4364 if (profiles.isEmpty() && inputs.isEmpty()) { 4365 ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); 4366 return BAD_VALUE; 4367 } 4368 4369 // open inputs for matching profiles if needed. Direct inputs are also opened to 4370 // query for dynamic parameters and will be closed later by setDeviceConnectionState() 4371 for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { 4372 4373 sp<IOProfile> profile = profiles[profile_index]; 4374 // nothing to do if one input is already opened for this profile 4375 size_t input_index; 4376 for (input_index = 0; input_index < mInputs.size(); input_index++) { 4377 desc = mInputs.valueAt(input_index); 4378 if (desc->mProfile == profile) { 4379 if (audio_device_is_digital(device)) { 4380 devDesc->importAudioPort(profile); 4381 } 4382 break; 4383 } 4384 } 4385 if (input_index != mInputs.size()) { 4386 continue; 4387 } 4388 4389 ALOGV("opening input for device 0x%X with params %s", device, address.string()); 4390 desc = new AudioInputDescriptor(profile); 4391 desc->mDevice = device; 4392 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 4393 config.sample_rate = desc->mSamplingRate; 4394 config.channel_mask = desc->mChannelMask; 4395 config.format = desc->mFormat; 4396 audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; 4397 4398 ALOGV("opening inputput for device %08x with params %s profile %p name %s", 4399 desc->mDevice, address.string(), profile.get(), profile->getName().string()); 4400 4401 status_t status = mpClientInterface->openInput(profile->getModuleHandle(), 4402 &input, 4403 &config, 4404 &desc->mDevice, 4405 address, 4406 AUDIO_SOURCE_MIC, 4407 AUDIO_INPUT_FLAG_NONE /*FIXME*/); 4408 4409 if (status == NO_ERROR) { 4410 desc->mSamplingRate = config.sample_rate; 4411 desc->mChannelMask = config.channel_mask; 4412 desc->mFormat = config.format; 4413 4414 if (!address.isEmpty()) { 4415 char *param = audio_device_address_to_parameter(device, address); 4416 mpClientInterface->setParameters(input, String8(param)); 4417 free(param); 4418 } 4419 updateAudioProfiles(device, input, profile->getAudioProfiles()); 4420 if (!profile->hasValidAudioProfile()) { 4421 ALOGW("checkInputsForDevice() direct input missing param"); 4422 mpClientInterface->closeInput(input); 4423 input = AUDIO_IO_HANDLE_NONE; 4424 } 4425 4426 if (input != 0) { 4427 addInput(input, desc); 4428 } 4429 } // endif input != 0 4430 4431 if (input == AUDIO_IO_HANDLE_NONE) { 4432 ALOGW("checkInputsForDevice() could not open input for device 0x%X", device); 4433 profiles.removeAt(profile_index); 4434 profile_index--; 4435 } else { 4436 inputs.add(input); 4437 if (audio_device_is_digital(device)) { 4438 devDesc->importAudioPort(profile); 4439 } 4440 ALOGV("checkInputsForDevice(): adding input %d", input); 4441 } 4442 } // end scan profiles 4443 4444 if (profiles.isEmpty()) { 4445 ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); 4446 return BAD_VALUE; 4447 } 4448 } else { 4449 // Disconnect 4450 // check if one opened input is not needed any more after disconnecting one device 4451 for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { 4452 desc = mInputs.valueAt(input_index); 4453 if (!(desc->mProfile->supportDevice(mAvailableInputDevices.types()))) { 4454 ALOGV("checkInputsForDevice(): disconnecting adding input %d", 4455 mInputs.keyAt(input_index)); 4456 inputs.add(mInputs.keyAt(input_index)); 4457 } 4458 } 4459 // Clear any profiles associated with the disconnected device. 4460 for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) { 4461 if (mHwModules[module_index]->mHandle == 0) { 4462 continue; 4463 } 4464 for (size_t profile_index = 0; 4465 profile_index < mHwModules[module_index]->mInputProfiles.size(); 4466 profile_index++) { 4467 sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index]; 4468 if (profile->supportDevice(device)) { 4469 ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu", 4470 profile_index, module_index); 4471 profile->clearAudioProfiles(); 4472 } 4473 } 4474 } 4475 } // end disconnect 4476 4477 return NO_ERROR; 4478 } 4479 4480 4481 void AudioPolicyManager::closeOutput(audio_io_handle_t output) 4482 { 4483 ALOGV("closeOutput(%d)", output); 4484 4485 sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); 4486 if (outputDesc == NULL) { 4487 ALOGW("closeOutput() unknown output %d", output); 4488 return; 4489 } 4490 mPolicyMixes.closeOutput(outputDesc); 4491 4492 // look for duplicated outputs connected to the output being removed. 4493 for (size_t i = 0; i < mOutputs.size(); i++) { 4494 sp<SwAudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i); 4495 if (dupOutputDesc->isDuplicated() && 4496 (dupOutputDesc->mOutput1 == outputDesc || 4497 dupOutputDesc->mOutput2 == outputDesc)) { 4498 sp<AudioOutputDescriptor> outputDesc2; 4499 if (dupOutputDesc->mOutput1 == outputDesc) { 4500 outputDesc2 = dupOutputDesc->mOutput2; 4501 } else { 4502 outputDesc2 = dupOutputDesc->mOutput1; 4503 } 4504 // As all active tracks on duplicated output will be deleted, 4505 // and as they were also referenced on the other output, the reference 4506 // count for their stream type must be adjusted accordingly on 4507 // the other output. 4508 for (int j = 0; j < AUDIO_STREAM_CNT; j++) { 4509 int refCount = dupOutputDesc->mRefCount[j]; 4510 outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount); 4511 } 4512 audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); 4513 ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); 4514 4515 mpClientInterface->closeOutput(duplicatedOutput); 4516 removeOutput(duplicatedOutput); 4517 } 4518 } 4519 4520 nextAudioPortGeneration(); 4521 4522 ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle()); 4523 if (index >= 0) { 4524 sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index); 4525 (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); 4526 mAudioPatches.removeItemsAt(index); 4527 mpClientInterface->onAudioPatchListUpdate(); 4528 } 4529 4530 AudioParameter param; 4531 param.add(String8("closing"), String8("true")); 4532 mpClientInterface->setParameters(output, param.toString()); 4533 4534 mpClientInterface->closeOutput(output); 4535 removeOutput(output); 4536 mPreviousOutputs = mOutputs; 4537 } 4538 4539 void AudioPolicyManager::closeInput(audio_io_handle_t input) 4540 { 4541 ALOGV("closeInput(%d)", input); 4542 4543 sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input); 4544 if (inputDesc == NULL) { 4545 ALOGW("closeInput() unknown input %d", input); 4546 return; 4547 } 4548 4549 nextAudioPortGeneration(); 4550 4551 ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); 4552 if (index >= 0) { 4553 sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index); 4554 (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); 4555 mAudioPatches.removeItemsAt(index); 4556 mpClientInterface->onAudioPatchListUpdate(); 4557 } 4558 4559 mpClientInterface->closeInput(input); 4560 mInputs.removeItem(input); 4561 } 4562 4563 SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice( 4564 audio_devices_t device, 4565 const SwAudioOutputCollection& openOutputs) 4566 { 4567 SortedVector<audio_io_handle_t> outputs; 4568 4569 ALOGVV("getOutputsForDevice() device %04x", device); 4570 for (size_t i = 0; i < openOutputs.size(); i++) { 4571 ALOGVV("output %zu isDuplicated=%d device=%04x", 4572 i, openOutputs.valueAt(i)->isDuplicated(), 4573 openOutputs.valueAt(i)->supportedDevices()); 4574 if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) { 4575 ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i)); 4576 outputs.add(openOutputs.keyAt(i)); 4577 } 4578 } 4579 return outputs; 4580 } 4581 4582 bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, 4583 SortedVector<audio_io_handle_t>& outputs2) 4584 { 4585 if (outputs1.size() != outputs2.size()) { 4586 return false; 4587 } 4588 for (size_t i = 0; i < outputs1.size(); i++) { 4589 if (outputs1[i] != outputs2[i]) { 4590 return false; 4591 } 4592 } 4593 return true; 4594 } 4595 4596 void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) 4597 { 4598 audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/); 4599 audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/); 4600 SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); 4601 SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs); 4602 4603 // also take into account external policy-related changes: add all outputs which are 4604 // associated with policies in the "before" and "after" output vectors 4605 ALOGVV("checkOutputForStrategy(): policy related outputs"); 4606 for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) { 4607 const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i); 4608 if (desc != 0 && desc->mPolicyMix != NULL) { 4609 srcOutputs.add(desc->mIoHandle); 4610 ALOGVV(" previous outputs: adding %d", desc->mIoHandle); 4611 } 4612 } 4613 for (size_t i = 0 ; i < mOutputs.size() ; i++) { 4614 const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); 4615 if (desc != 0 && desc->mPolicyMix != NULL) { 4616 dstOutputs.add(desc->mIoHandle); 4617 ALOGVV(" new outputs: adding %d", desc->mIoHandle); 4618 } 4619 } 4620 4621 if (!vectorsEqual(srcOutputs,dstOutputs)) { 4622 ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d", 4623 strategy, srcOutputs[0], dstOutputs[0]); 4624 // mute strategy while moving tracks from one output to another 4625 for (size_t i = 0; i < srcOutputs.size(); i++) { 4626 sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]); 4627 if (isStrategyActive(desc, strategy)) { 4628 setStrategyMute(strategy, true, desc); 4629 setStrategyMute(strategy, false, desc, MUTE_TIME_MS, newDevice); 4630 } 4631 sp<AudioSourceDescriptor> source = 4632 getSourceForStrategyOnOutput(srcOutputs[i], strategy); 4633 if (source != 0){ 4634 connectAudioSource(source); 4635 } 4636 } 4637 4638 // Move effects associated to this strategy from previous output to new output 4639 if (strategy == STRATEGY_MEDIA) { 4640 selectOutputForMusicEffects(); 4641 } 4642 // Move tracks associated to this strategy from previous output to new output 4643 for (int i = 0; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) { 4644 if (getStrategy((audio_stream_type_t)i) == strategy) { 4645 mpClientInterface->invalidateStream((audio_stream_type_t)i); 4646 } 4647 } 4648 } 4649 } 4650 4651 void AudioPolicyManager::checkOutputForAllStrategies() 4652 { 4653 if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) 4654 checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); 4655 checkOutputForStrategy(STRATEGY_PHONE); 4656 if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) 4657 checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); 4658 checkOutputForStrategy(STRATEGY_SONIFICATION); 4659 checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); 4660 checkOutputForStrategy(STRATEGY_ACCESSIBILITY); 4661 checkOutputForStrategy(STRATEGY_MEDIA); 4662 checkOutputForStrategy(STRATEGY_DTMF); 4663 checkOutputForStrategy(STRATEGY_REROUTING); 4664 } 4665 4666 void AudioPolicyManager::checkA2dpSuspend() 4667 { 4668 audio_io_handle_t a2dpOutput = mOutputs.getA2dpOutput(); 4669 if (a2dpOutput == 0) { 4670 mA2dpSuspended = false; 4671 return; 4672 } 4673 4674 bool isScoConnected = 4675 ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET & 4676 ~AUDIO_DEVICE_BIT_IN) != 0) || 4677 ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0); 4678 4679 // if suspended, restore A2DP output if: 4680 // ((SCO device is NOT connected) || 4681 // ((forced usage communication is NOT SCO) && (forced usage for record is NOT SCO) && 4682 // (phone state is NOT in call) && (phone state is NOT ringing))) 4683 // 4684 // if not suspended, suspend A2DP output if: 4685 // (SCO device is connected) && 4686 // ((forced usage for communication is SCO) || (forced usage for record is SCO) || 4687 // ((phone state is in call) || (phone state is ringing))) 4688 // 4689 if (mA2dpSuspended) { 4690 if (!isScoConnected || 4691 ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) != 4692 AUDIO_POLICY_FORCE_BT_SCO) && 4693 (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) != 4694 AUDIO_POLICY_FORCE_BT_SCO) && 4695 (mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) && 4696 (mEngine->getPhoneState() != AUDIO_MODE_RINGTONE))) { 4697 4698 mpClientInterface->restoreOutput(a2dpOutput); 4699 mA2dpSuspended = false; 4700 } 4701 } else { 4702 if (isScoConnected && 4703 ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == 4704 AUDIO_POLICY_FORCE_BT_SCO) || 4705 (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == 4706 AUDIO_POLICY_FORCE_BT_SCO) || 4707 (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) || 4708 (mEngine->getPhoneState() == AUDIO_MODE_RINGTONE))) { 4709 4710 mpClientInterface->suspendOutput(a2dpOutput); 4711 mA2dpSuspended = true; 4712 } 4713 } 4714 } 4715 4716 audio_devices_t AudioPolicyManager::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, 4717 bool fromCache) 4718 { 4719 audio_devices_t device = AUDIO_DEVICE_NONE; 4720 4721 ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle()); 4722 if (index >= 0) { 4723 sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index); 4724 if (patchDesc->mUid != mUidCached) { 4725 ALOGV("getNewOutputDevice() device %08x forced by patch %d", 4726 outputDesc->device(), outputDesc->getPatchHandle()); 4727 return outputDesc->device(); 4728 } 4729 } 4730 4731 // check the following by order of priority to request a routing change if necessary: 4732 // 1: the strategy enforced audible is active and enforced on the output: 4733 // use device for strategy enforced audible 4734 // 2: we are in call or the strategy phone is active on the output: 4735 // use device for strategy phone 4736 // 3: the strategy sonification is active on the output: 4737 // use device for strategy sonification 4738 // 4: the strategy for enforced audible is active but not enforced on the output: 4739 // use the device for strategy enforced audible 4740 // 5: the strategy accessibility is active on the output: 4741 // use device for strategy accessibility 4742 // 6: the strategy "respectful" sonification is active on the output: 4743 // use device for strategy "respectful" sonification 4744 // 7: the strategy media is active on the output: 4745 // use device for strategy media 4746 // 8: the strategy DTMF is active on the output: 4747 // use device for strategy DTMF 4748 // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output: 4749 // use device for strategy t-t-s 4750 if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) && 4751 mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { 4752 device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); 4753 } else if (isInCall() || 4754 isStrategyActive(outputDesc, STRATEGY_PHONE)) { 4755 device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); 4756 } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)) { 4757 device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); 4758 } else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) { 4759 device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); 4760 } else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) { 4761 device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache); 4762 } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)) { 4763 device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); 4764 } else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) { 4765 device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); 4766 } else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) { 4767 device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); 4768 } else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) { 4769 device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache); 4770 } else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) { 4771 device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache); 4772 } 4773 4774 ALOGV("getNewOutputDevice() selected device %x", device); 4775 return device; 4776 } 4777 4778 audio_devices_t AudioPolicyManager::getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc) 4779 { 4780 audio_devices_t device = AUDIO_DEVICE_NONE; 4781 4782 ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); 4783 if (index >= 0) { 4784 sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index); 4785 if (patchDesc->mUid != mUidCached) { 4786 ALOGV("getNewInputDevice() device %08x forced by patch %d", 4787 inputDesc->mDevice, inputDesc->getPatchHandle()); 4788 return inputDesc->mDevice; 4789 } 4790 } 4791 4792 audio_source_t source = inputDesc->getHighestPrioritySource(true /*activeOnly*/); 4793 if (isInCall()) { 4794 device = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); 4795 } else if (source != AUDIO_SOURCE_DEFAULT) { 4796 device = getDeviceAndMixForInputSource(source); 4797 } 4798 4799 return device; 4800 } 4801 4802 bool AudioPolicyManager::streamsMatchForvolume(audio_stream_type_t stream1, 4803 audio_stream_type_t stream2) { 4804 return (stream1 == stream2); 4805 } 4806 4807 uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) { 4808 return (uint32_t)getStrategy(stream); 4809 } 4810 4811 audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) { 4812 // By checking the range of stream before calling getStrategy, we avoid 4813 // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE 4814 // and then return STRATEGY_MEDIA, but we want to return the empty set. 4815 if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) { 4816 return AUDIO_DEVICE_NONE; 4817 } 4818 audio_devices_t devices = AUDIO_DEVICE_NONE; 4819 for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) { 4820 if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) { 4821 continue; 4822 } 4823 routing_strategy curStrategy = getStrategy((audio_stream_type_t)curStream); 4824 audio_devices_t curDevices = 4825 getDeviceForStrategy((routing_strategy)curStrategy, false /*fromCache*/); 4826 SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(curDevices, mOutputs); 4827 for (size_t i = 0; i < outputs.size(); i++) { 4828 sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]); 4829 if (outputDesc->isStreamActive((audio_stream_type_t)curStream)) { 4830 curDevices |= outputDesc->device(); 4831 } 4832 } 4833 devices |= curDevices; 4834 } 4835 4836 /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it 4837 and doesn't really need to.*/ 4838 if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) { 4839 devices |= AUDIO_DEVICE_OUT_SPEAKER; 4840 devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE; 4841 } 4842 return devices; 4843 } 4844 4845 routing_strategy AudioPolicyManager::getStrategy(audio_stream_type_t stream) const 4846 { 4847 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH"); 4848 return mEngine->getStrategyForStream(stream); 4849 } 4850 4851 uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) { 4852 // flags to strategy mapping 4853 if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { 4854 return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER; 4855 } 4856 if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { 4857 return (uint32_t) STRATEGY_ENFORCED_AUDIBLE; 4858 } 4859 // usage to strategy mapping 4860 return static_cast<uint32_t>(mEngine->getStrategyForUsage(attr->usage)); 4861 } 4862 4863 void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) { 4864 switch(stream) { 4865 case AUDIO_STREAM_MUSIC: 4866 checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); 4867 updateDevicesAndOutputs(); 4868 break; 4869 default: 4870 break; 4871 } 4872 } 4873 4874 uint32_t AudioPolicyManager::handleEventForBeacon(int event) { 4875 4876 // skip beacon mute management if a dedicated TTS output is available 4877 if (mTtsOutputAvailable) { 4878 return 0; 4879 } 4880 4881 switch(event) { 4882 case STARTING_OUTPUT: 4883 mBeaconMuteRefCount++; 4884 break; 4885 case STOPPING_OUTPUT: 4886 if (mBeaconMuteRefCount > 0) { 4887 mBeaconMuteRefCount--; 4888 } 4889 break; 4890 case STARTING_BEACON: 4891 mBeaconPlayingRefCount++; 4892 break; 4893 case STOPPING_BEACON: 4894 if (mBeaconPlayingRefCount > 0) { 4895 mBeaconPlayingRefCount--; 4896 } 4897 break; 4898 } 4899 4900 if (mBeaconMuteRefCount > 0) { 4901 // any playback causes beacon to be muted 4902 return setBeaconMute(true); 4903 } else { 4904 // no other playback: unmute when beacon starts playing, mute when it stops 4905 return setBeaconMute(mBeaconPlayingRefCount == 0); 4906 } 4907 } 4908 4909 uint32_t AudioPolicyManager::setBeaconMute(bool mute) { 4910 ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d", 4911 mute, mBeaconMuteRefCount, mBeaconPlayingRefCount); 4912 // keep track of muted state to avoid repeating mute/unmute operations 4913 if (mBeaconMuted != mute) { 4914 // mute/unmute AUDIO_STREAM_TTS on all outputs 4915 ALOGV("\t muting %d", mute); 4916 uint32_t maxLatency = 0; 4917 for (size_t i = 0; i < mOutputs.size(); i++) { 4918 sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); 4919 setStreamMute(AUDIO_STREAM_TTS, mute/*on*/, 4920 desc, 4921 0 /*delay*/, AUDIO_DEVICE_NONE); 4922 const uint32_t latency = desc->latency() * 2; 4923 if (latency > maxLatency) { 4924 maxLatency = latency; 4925 } 4926 } 4927 mBeaconMuted = mute; 4928 return maxLatency; 4929 } 4930 return 0; 4931 } 4932 4933 audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy, 4934 bool fromCache) 4935 { 4936 // Routing 4937 // see if we have an explicit route 4938 // scan the whole RouteMap, for each entry, convert the stream type to a strategy 4939 // (getStrategy(stream)). 4940 // if the strategy from the stream type in the RouteMap is the same as the argument above, 4941 // and activity count is non-zero and the device in the route descriptor is available 4942 // then select this device. 4943 for (size_t routeIndex = 0; routeIndex < mOutputRoutes.size(); routeIndex++) { 4944 sp<SessionRoute> route = mOutputRoutes.valueAt(routeIndex); 4945 routing_strategy routeStrategy = getStrategy(route->mStreamType); 4946 if ((routeStrategy == strategy) && route->isActive() && 4947 (mAvailableOutputDevices.indexOf(route->mDeviceDescriptor) >= 0)) { 4948 return route->mDeviceDescriptor->type(); 4949 } 4950 } 4951 4952 if (fromCache) { 4953 ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", 4954 strategy, mDeviceForStrategy[strategy]); 4955 return mDeviceForStrategy[strategy]; 4956 } 4957 return mEngine->getDeviceForStrategy(strategy); 4958 } 4959 4960 void AudioPolicyManager::updateDevicesAndOutputs() 4961 { 4962 for (int i = 0; i < NUM_STRATEGIES; i++) { 4963 mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); 4964 } 4965 mPreviousOutputs = mOutputs; 4966 } 4967 4968 uint32_t AudioPolicyManager::checkDeviceMuteStrategies(const sp<AudioOutputDescriptor>& outputDesc, 4969 audio_devices_t prevDevice, 4970 uint32_t delayMs) 4971 { 4972 // mute/unmute strategies using an incompatible device combination 4973 // if muting, wait for the audio in pcm buffer to be drained before proceeding 4974 // if unmuting, unmute only after the specified delay 4975 if (outputDesc->isDuplicated()) { 4976 return 0; 4977 } 4978 4979 uint32_t muteWaitMs = 0; 4980 audio_devices_t device = outputDesc->device(); 4981 bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2); 4982 4983 for (size_t i = 0; i < NUM_STRATEGIES; i++) { 4984 audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); 4985 curDevice = curDevice & outputDesc->supportedDevices(); 4986 bool mute = shouldMute && (curDevice & device) && (curDevice != device); 4987 bool doMute = false; 4988 4989 if (mute && !outputDesc->mStrategyMutedByDevice[i]) { 4990 doMute = true; 4991 outputDesc->mStrategyMutedByDevice[i] = true; 4992 } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){ 4993 doMute = true; 4994 outputDesc->mStrategyMutedByDevice[i] = false; 4995 } 4996 if (doMute) { 4997 for (size_t j = 0; j < mOutputs.size(); j++) { 4998 sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j); 4999 // skip output if it does not share any device with current output 5000 if ((desc->supportedDevices() & outputDesc->supportedDevices()) 5001 == AUDIO_DEVICE_NONE) { 5002 continue; 5003 } 5004 ALOGVV("checkDeviceMuteStrategies() %s strategy %zu (curDevice %04x)", 5005 mute ? "muting" : "unmuting", i, curDevice); 5006 setStrategyMute((routing_strategy)i, mute, desc, mute ? 0 : delayMs); 5007 if (isStrategyActive(desc, (routing_strategy)i)) { 5008 if (mute) { 5009 // FIXME: should not need to double latency if volume could be applied 5010 // immediately by the audioflinger mixer. We must account for the delay 5011 // between now and the next time the audioflinger thread for this output 5012 // will process a buffer (which corresponds to one buffer size, 5013 // usually 1/2 or 1/4 of the latency). 5014 if (muteWaitMs < desc->latency() * 2) { 5015 muteWaitMs = desc->latency() * 2; 5016 } 5017 } 5018 } 5019 } 5020 } 5021 } 5022 5023 // temporary mute output if device selection changes to avoid volume bursts due to 5024 // different per device volumes 5025 if (outputDesc->isActive() && (device != prevDevice)) { 5026 uint32_t tempMuteWaitMs = outputDesc->latency() * 2; 5027 // temporary mute duration is conservatively set to 4 times the reported latency 5028 uint32_t tempMuteDurationMs = outputDesc->latency() * 4; 5029 if (muteWaitMs < tempMuteWaitMs) { 5030 muteWaitMs = tempMuteWaitMs; 5031 } 5032 5033 for (size_t i = 0; i < NUM_STRATEGIES; i++) { 5034 if (isStrategyActive(outputDesc, (routing_strategy)i)) { 5035 // make sure that we do not start the temporary mute period too early in case of 5036 // delayed device change 5037 setStrategyMute((routing_strategy)i, true, outputDesc, delayMs); 5038 setStrategyMute((routing_strategy)i, false, outputDesc, 5039 delayMs + tempMuteDurationMs, device); 5040 } 5041 } 5042 } 5043 5044 // wait for the PCM output buffers to empty before proceeding with the rest of the command 5045 if (muteWaitMs > delayMs) { 5046 muteWaitMs -= delayMs; 5047 usleep(muteWaitMs * 1000); 5048 return muteWaitMs; 5049 } 5050 return 0; 5051 } 5052 5053 uint32_t AudioPolicyManager::setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, 5054 audio_devices_t device, 5055 bool force, 5056 int delayMs, 5057 audio_patch_handle_t *patchHandle, 5058 const char* address) 5059 { 5060 ALOGV("setOutputDevice() device %04x delayMs %d", device, delayMs); 5061 AudioParameter param; 5062 uint32_t muteWaitMs; 5063 5064 if (outputDesc->isDuplicated()) { 5065 muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs); 5066 muteWaitMs += setOutputDevice(outputDesc->subOutput2(), device, force, delayMs); 5067 return muteWaitMs; 5068 } 5069 // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current 5070 // output profile 5071 if ((device != AUDIO_DEVICE_NONE) && 5072 ((device & outputDesc->supportedDevices()) == 0)) { 5073 return 0; 5074 } 5075 5076 // filter devices according to output selected 5077 device = (audio_devices_t)(device & outputDesc->supportedDevices()); 5078 5079 audio_devices_t prevDevice = outputDesc->mDevice; 5080 5081 ALOGV("setOutputDevice() prevDevice 0x%04x", prevDevice); 5082 5083 if (device != AUDIO_DEVICE_NONE) { 5084 outputDesc->mDevice = device; 5085 } 5086 muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs); 5087 5088 // Do not change the routing if: 5089 // the requested device is AUDIO_DEVICE_NONE 5090 // OR the requested device is the same as current device 5091 // AND force is not specified 5092 // AND the output is connected by a valid audio patch. 5093 // Doing this check here allows the caller to call setOutputDevice() without conditions 5094 if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && 5095 !force && 5096 outputDesc->getPatchHandle() != 0) { 5097 ALOGV("setOutputDevice() setting same device 0x%04x or null device", device); 5098 return muteWaitMs; 5099 } 5100 5101 ALOGV("setOutputDevice() changing device"); 5102 5103 // do the routing 5104 if (device == AUDIO_DEVICE_NONE) { 5105 resetOutputDevice(outputDesc, delayMs, NULL); 5106 } else { 5107 DeviceVector deviceList; 5108 if ((address == NULL) || (strlen(address) == 0)) { 5109 deviceList = mAvailableOutputDevices.getDevicesFromType(device); 5110 } else { 5111 deviceList = mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address)); 5112 } 5113 5114 if (!deviceList.isEmpty()) { 5115 struct audio_patch patch; 5116 outputDesc->toAudioPortConfig(&patch.sources[0]); 5117 patch.num_sources = 1; 5118 patch.num_sinks = 0; 5119 for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) { 5120 deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]); 5121 patch.num_sinks++; 5122 } 5123 ssize_t index; 5124 if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { 5125 index = mAudioPatches.indexOfKey(*patchHandle); 5126 } else { 5127 index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle()); 5128 } 5129 sp< AudioPatch> patchDesc; 5130 audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; 5131 if (index >= 0) { 5132 patchDesc = mAudioPatches.valueAt(index); 5133 afPatchHandle = patchDesc->mAfPatchHandle; 5134 } 5135 5136 status_t status = mpClientInterface->createAudioPatch(&patch, 5137 &afPatchHandle, 5138 delayMs); 5139 ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d" 5140 "num_sources %d num_sinks %d", 5141 status, afPatchHandle, patch.num_sources, patch.num_sinks); 5142 if (status == NO_ERROR) { 5143 if (index < 0) { 5144 patchDesc = new AudioPatch(&patch, mUidCached); 5145 addAudioPatch(patchDesc->mHandle, patchDesc); 5146 } else { 5147 patchDesc->mPatch = patch; 5148 } 5149 patchDesc->mAfPatchHandle = afPatchHandle; 5150 if (patchHandle) { 5151 *patchHandle = patchDesc->mHandle; 5152 } 5153 outputDesc->setPatchHandle(patchDesc->mHandle); 5154 nextAudioPortGeneration(); 5155 mpClientInterface->onAudioPatchListUpdate(); 5156 } 5157 } 5158 5159 // inform all input as well 5160 for (size_t i = 0; i < mInputs.size(); i++) { 5161 const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i); 5162 if (!is_virtual_input_device(inputDescriptor->mDevice)) { 5163 AudioParameter inputCmd = AudioParameter(); 5164 ALOGV("%s: inform input %d of device:%d", __func__, 5165 inputDescriptor->mIoHandle, device); 5166 inputCmd.addInt(String8(AudioParameter::keyRouting),device); 5167 mpClientInterface->setParameters(inputDescriptor->mIoHandle, 5168 inputCmd.toString(), 5169 delayMs); 5170 } 5171 } 5172 } 5173 5174 // update stream volumes according to new device 5175 applyStreamVolumes(outputDesc, device, delayMs); 5176 5177 return muteWaitMs; 5178 } 5179 5180 status_t AudioPolicyManager::resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, 5181 int delayMs, 5182 audio_patch_handle_t *patchHandle) 5183 { 5184 ssize_t index; 5185 if (patchHandle) { 5186 index = mAudioPatches.indexOfKey(*patchHandle); 5187 } else { 5188 index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle()); 5189 } 5190 if (index < 0) { 5191 return INVALID_OPERATION; 5192 } 5193 sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); 5194 status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs); 5195 ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status); 5196 outputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE); 5197 removeAudioPatch(patchDesc->mHandle); 5198 nextAudioPortGeneration(); 5199 mpClientInterface->onAudioPatchListUpdate(); 5200 return status; 5201 } 5202 5203 status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input, 5204 audio_devices_t device, 5205 bool force, 5206 audio_patch_handle_t *patchHandle) 5207 { 5208 status_t status = NO_ERROR; 5209 5210 sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input); 5211 if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) { 5212 inputDesc->mDevice = device; 5213 5214 DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device); 5215 if (!deviceList.isEmpty()) { 5216 struct audio_patch patch; 5217 inputDesc->toAudioPortConfig(&patch.sinks[0]); 5218 // AUDIO_SOURCE_HOTWORD is for internal use only: 5219 // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL 5220 if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD && 5221 !inputDesc->isSoundTrigger()) { 5222 patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION; 5223 } 5224 patch.num_sinks = 1; 5225 //only one input device for now 5226 deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]); 5227 patch.num_sources = 1; 5228 ssize_t index; 5229 if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { 5230 index = mAudioPatches.indexOfKey(*patchHandle); 5231 } else { 5232 index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); 5233 } 5234 sp< AudioPatch> patchDesc; 5235 audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; 5236 if (index >= 0) { 5237 patchDesc = mAudioPatches.valueAt(index); 5238 afPatchHandle = patchDesc->mAfPatchHandle; 5239 } 5240 5241 status_t status = mpClientInterface->createAudioPatch(&patch, 5242 &afPatchHandle, 5243 0); 5244 ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d", 5245 status, afPatchHandle); 5246 if (status == NO_ERROR) { 5247 if (index < 0) { 5248 patchDesc = new AudioPatch(&patch, mUidCached); 5249 addAudioPatch(patchDesc->mHandle, patchDesc); 5250 } else { 5251 patchDesc->mPatch = patch; 5252 } 5253 patchDesc->mAfPatchHandle = afPatchHandle; 5254 if (patchHandle) { 5255 *patchHandle = patchDesc->mHandle; 5256 } 5257 inputDesc->setPatchHandle(patchDesc->mHandle); 5258 nextAudioPortGeneration(); 5259 mpClientInterface->onAudioPatchListUpdate(); 5260 } 5261 } 5262 } 5263 return status; 5264 } 5265 5266 status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input, 5267 audio_patch_handle_t *patchHandle) 5268 { 5269 sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input); 5270 ssize_t index; 5271 if (patchHandle) { 5272 index = mAudioPatches.indexOfKey(*patchHandle); 5273 } else { 5274 index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); 5275 } 5276 if (index < 0) { 5277 return INVALID_OPERATION; 5278 } 5279 sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); 5280 status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); 5281 ALOGV("resetInputDevice() releaseAudioPatch returned %d", status); 5282 inputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE); 5283 removeAudioPatch(patchDesc->mHandle); 5284 nextAudioPortGeneration(); 5285 mpClientInterface->onAudioPatchListUpdate(); 5286 return status; 5287 } 5288 5289 sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device, 5290 const String8& address, 5291 uint32_t& samplingRate, 5292 audio_format_t& format, 5293 audio_channel_mask_t& channelMask, 5294 audio_input_flags_t flags) 5295 { 5296 // Choose an input profile based on the requested capture parameters: select the first available 5297 // profile supporting all requested parameters. 5298 // 5299 // TODO: perhaps isCompatibleProfile should return a "matching" score so we can return 5300 // the best matching profile, not the first one. 5301 5302 for (size_t i = 0; i < mHwModules.size(); i++) 5303 { 5304 if (mHwModules[i]->mHandle == 0) { 5305 continue; 5306 } 5307 for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) 5308 { 5309 sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j]; 5310 // profile->log(); 5311 if (profile->isCompatibleProfile(device, address, samplingRate, 5312 &samplingRate /*updatedSamplingRate*/, 5313 format, 5314 &format /*updatedFormat*/, 5315 channelMask, 5316 &channelMask /*updatedChannelMask*/, 5317 (audio_output_flags_t) flags)) { 5318 5319 return profile; 5320 } 5321 } 5322 } 5323 return NULL; 5324 } 5325 5326 5327 audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource, 5328 AudioMix **policyMix) 5329 { 5330 audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; 5331 audio_devices_t selectedDeviceFromMix = 5332 mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix); 5333 5334 if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) { 5335 return selectedDeviceFromMix; 5336 } 5337 return getDeviceForInputSource(inputSource); 5338 } 5339 5340 audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource) 5341 { 5342 // Routing 5343 // Scan the whole RouteMap to see if we have an explicit route: 5344 // if the input source in the RouteMap is the same as the argument above, 5345 // and activity count is non-zero and the device in the route descriptor is available 5346 // then select this device. 5347 for (size_t routeIndex = 0; routeIndex < mInputRoutes.size(); routeIndex++) { 5348 sp<SessionRoute> route = mInputRoutes.valueAt(routeIndex); 5349 if ((inputSource == route->mSource) && route->isActive() && 5350 (mAvailableInputDevices.indexOf(route->mDeviceDescriptor) >= 0)) { 5351 return route->mDeviceDescriptor->type(); 5352 } 5353 } 5354 5355 return mEngine->getDeviceForInputSource(inputSource); 5356 } 5357 5358 float AudioPolicyManager::computeVolume(audio_stream_type_t stream, 5359 int index, 5360 audio_devices_t device) 5361 { 5362 float volumeDB = mVolumeCurves->volIndexToDb(stream, Volume::getDeviceCategory(device), index); 5363 5364 // handle the case of accessibility active while a ringtone is playing: if the ringtone is much 5365 // louder than the accessibility prompt, the prompt cannot be heard, thus masking the touch 5366 // exploration of the dialer UI. In this situation, bring the accessibility volume closer to 5367 // the ringtone volume 5368 if ((stream == AUDIO_STREAM_ACCESSIBILITY) 5369 && (AUDIO_MODE_RINGTONE == mEngine->getPhoneState()) 5370 && isStreamActive(AUDIO_STREAM_RING, 0)) { 5371 const float ringVolumeDB = computeVolume(AUDIO_STREAM_RING, index, device); 5372 return ringVolumeDB - 4 > volumeDB ? ringVolumeDB - 4 : volumeDB; 5373 } 5374 5375 // in-call: always cap earpiece volume by voice volume + some low headroom 5376 if ((stream != AUDIO_STREAM_VOICE_CALL) && (device & AUDIO_DEVICE_OUT_EARPIECE) && isInCall()) { 5377 switch (stream) { 5378 case AUDIO_STREAM_SYSTEM: 5379 case AUDIO_STREAM_RING: 5380 case AUDIO_STREAM_MUSIC: 5381 case AUDIO_STREAM_ALARM: 5382 case AUDIO_STREAM_NOTIFICATION: 5383 case AUDIO_STREAM_ENFORCED_AUDIBLE: 5384 case AUDIO_STREAM_DTMF: 5385 case AUDIO_STREAM_ACCESSIBILITY: { 5386 const float maxVoiceVolDb = computeVolume(AUDIO_STREAM_VOICE_CALL, index, device) 5387 + IN_CALL_EARPIECE_HEADROOM_DB; 5388 if (volumeDB > maxVoiceVolDb) { 5389 ALOGV("computeVolume() stream %d at vol=%f overriden by stream %d at vol=%f", 5390 stream, volumeDB, AUDIO_STREAM_VOICE_CALL, maxVoiceVolDb); 5391 volumeDB = maxVoiceVolDb; 5392 } 5393 } break; 5394 default: 5395 break; 5396 } 5397 } 5398 5399 // if a headset is connected, apply the following rules to ring tones and notifications 5400 // to avoid sound level bursts in user's ears: 5401 // - always attenuate notifications volume by 6dB 5402 // - attenuate ring tones volume by 6dB unless music is not playing and 5403 // speaker is part of the select devices 5404 // - if music is playing, always limit the volume to current music volume, 5405 // with a minimum threshold at -36dB so that notification is always perceived. 5406 const routing_strategy stream_strategy = getStrategy(stream); 5407 if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | 5408 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | 5409 AUDIO_DEVICE_OUT_WIRED_HEADSET | 5410 AUDIO_DEVICE_OUT_WIRED_HEADPHONE | 5411 AUDIO_DEVICE_OUT_USB_HEADSET)) && 5412 ((stream_strategy == STRATEGY_SONIFICATION) 5413 || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) 5414 || (stream == AUDIO_STREAM_SYSTEM) 5415 || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && 5416 (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) && 5417 mVolumeCurves->canBeMuted(stream)) { 5418 // when the phone is ringing we must consider that music could have been paused just before 5419 // by the music application and behave as if music was active if the last music track was 5420 // just stopped 5421 if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || 5422 mLimitRingtoneVolume) { 5423 volumeDB += SONIFICATION_HEADSET_VOLUME_FACTOR_DB; 5424 audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); 5425 float musicVolDB = computeVolume(AUDIO_STREAM_MUSIC, 5426 mVolumeCurves->getVolumeIndex(AUDIO_STREAM_MUSIC, 5427 musicDevice), 5428 musicDevice); 5429 float minVolDB = (musicVolDB > SONIFICATION_HEADSET_VOLUME_MIN_DB) ? 5430 musicVolDB : SONIFICATION_HEADSET_VOLUME_MIN_DB; 5431 if (volumeDB > minVolDB) { 5432 volumeDB = minVolDB; 5433 ALOGV("computeVolume limiting volume to %f musicVol %f", minVolDB, musicVolDB); 5434 } 5435 if (device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | 5436 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES)) { 5437 // on A2DP, also ensure notification volume is not too low compared to media when 5438 // intended to be played 5439 if ((volumeDB > -96.0f) && 5440 (musicVolDB - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB > volumeDB)) { 5441 ALOGV("computeVolume increasing volume for stream=%d device=0x%X from %f to %f", 5442 stream, device, 5443 volumeDB, musicVolDB - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB); 5444 volumeDB = musicVolDB - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB; 5445 } 5446 } 5447 } else if ((Volume::getDeviceForVolume(device) != AUDIO_DEVICE_OUT_SPEAKER) || 5448 stream_strategy != STRATEGY_SONIFICATION) { 5449 volumeDB += SONIFICATION_HEADSET_VOLUME_FACTOR_DB; 5450 } 5451 } 5452 5453 return volumeDB; 5454 } 5455 5456 status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, 5457 int index, 5458 const sp<AudioOutputDescriptor>& outputDesc, 5459 audio_devices_t device, 5460 int delayMs, 5461 bool force) 5462 { 5463 // do not change actual stream volume if the stream is muted 5464 if (outputDesc->mMuteCount[stream] != 0) { 5465 ALOGVV("checkAndSetVolume() stream %d muted count %d", 5466 stream, outputDesc->mMuteCount[stream]); 5467 return NO_ERROR; 5468 } 5469 audio_policy_forced_cfg_t forceUseForComm = 5470 mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION); 5471 // do not change in call volume if bluetooth is connected and vice versa 5472 if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) || 5473 (stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) { 5474 ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", 5475 stream, forceUseForComm); 5476 return INVALID_OPERATION; 5477 } 5478 5479 if (device == AUDIO_DEVICE_NONE) { 5480 device = outputDesc->device(); 5481 } 5482 5483 float volumeDb = computeVolume(stream, index, device); 5484 if (outputDesc->isFixedVolume(device)) { 5485 volumeDb = 0.0f; 5486 } 5487 5488 outputDesc->setVolume(volumeDb, stream, device, delayMs, force); 5489 5490 if (stream == AUDIO_STREAM_VOICE_CALL || 5491 stream == AUDIO_STREAM_BLUETOOTH_SCO) { 5492 float voiceVolume; 5493 // Force voice volume to max for bluetooth SCO as volume is managed by the headset 5494 if (stream == AUDIO_STREAM_VOICE_CALL) { 5495 voiceVolume = (float)index/(float)mVolumeCurves->getVolumeIndexMax(stream); 5496 } else { 5497 voiceVolume = 1.0; 5498 } 5499 5500 if (voiceVolume != mLastVoiceVolume) { 5501 mpClientInterface->setVoiceVolume(voiceVolume, delayMs); 5502 mLastVoiceVolume = voiceVolume; 5503 } 5504 } 5505 5506 return NO_ERROR; 5507 } 5508 5509 void AudioPolicyManager::applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc, 5510 audio_devices_t device, 5511 int delayMs, 5512 bool force) 5513 { 5514 ALOGVV("applyStreamVolumes() for device %08x", device); 5515 5516 for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) { 5517 checkAndSetVolume((audio_stream_type_t)stream, 5518 mVolumeCurves->getVolumeIndex((audio_stream_type_t)stream, device), 5519 outputDesc, 5520 device, 5521 delayMs, 5522 force); 5523 } 5524 } 5525 5526 void AudioPolicyManager::setStrategyMute(routing_strategy strategy, 5527 bool on, 5528 const sp<AudioOutputDescriptor>& outputDesc, 5529 int delayMs, 5530 audio_devices_t device) 5531 { 5532 ALOGVV("setStrategyMute() strategy %d, mute %d, output ID %d", 5533 strategy, on, outputDesc->getId()); 5534 for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) { 5535 if (getStrategy((audio_stream_type_t)stream) == strategy) { 5536 setStreamMute((audio_stream_type_t)stream, on, outputDesc, delayMs, device); 5537 } 5538 } 5539 } 5540 5541 void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, 5542 bool on, 5543 const sp<AudioOutputDescriptor>& outputDesc, 5544 int delayMs, 5545 audio_devices_t device) 5546 { 5547 if (device == AUDIO_DEVICE_NONE) { 5548 device = outputDesc->device(); 5549 } 5550 5551 ALOGVV("setStreamMute() stream %d, mute %d, mMuteCount %d device %04x", 5552 stream, on, outputDesc->mMuteCount[stream], device); 5553 5554 if (on) { 5555 if (outputDesc->mMuteCount[stream] == 0) { 5556 if (mVolumeCurves->canBeMuted(stream) && 5557 ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) || 5558 (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) { 5559 checkAndSetVolume(stream, 0, outputDesc, device, delayMs); 5560 } 5561 } 5562 // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored 5563 outputDesc->mMuteCount[stream]++; 5564 } else { 5565 if (outputDesc->mMuteCount[stream] == 0) { 5566 ALOGV("setStreamMute() unmuting non muted stream!"); 5567 return; 5568 } 5569 if (--outputDesc->mMuteCount[stream] == 0) { 5570 checkAndSetVolume(stream, 5571 mVolumeCurves->getVolumeIndex(stream, device), 5572 outputDesc, 5573 device, 5574 delayMs); 5575 } 5576 } 5577 } 5578 5579 void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, 5580 bool starting, bool stateChange) 5581 { 5582 if(!hasPrimaryOutput()) { 5583 return; 5584 } 5585 5586 // if the stream pertains to sonification strategy and we are in call we must 5587 // mute the stream if it is low visibility. If it is high visibility, we must play a tone 5588 // in the device used for phone strategy and play the tone if the selected device does not 5589 // interfere with the device used for phone strategy 5590 // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as 5591 // many times as there are active tracks on the output 5592 const routing_strategy stream_strategy = getStrategy(stream); 5593 if ((stream_strategy == STRATEGY_SONIFICATION) || 5594 ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { 5595 sp<SwAudioOutputDescriptor> outputDesc = mPrimaryOutput; 5596 ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", 5597 stream, starting, outputDesc->mDevice, stateChange); 5598 if (outputDesc->mRefCount[stream]) { 5599 int muteCount = 1; 5600 if (stateChange) { 5601 muteCount = outputDesc->mRefCount[stream]; 5602 } 5603 if (audio_is_low_visibility(stream)) { 5604 ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); 5605 for (int i = 0; i < muteCount; i++) { 5606 setStreamMute(stream, starting, mPrimaryOutput); 5607 } 5608 } else { 5609 ALOGV("handleIncallSonification() high visibility"); 5610 if (outputDesc->device() & 5611 getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) { 5612 ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); 5613 for (int i = 0; i < muteCount; i++) { 5614 setStreamMute(stream, starting, mPrimaryOutput); 5615 } 5616 } 5617 if (starting) { 5618 mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION, 5619 AUDIO_STREAM_VOICE_CALL); 5620 } else { 5621 mpClientInterface->stopTone(); 5622 } 5623 } 5624 } 5625 } 5626 } 5627 5628 audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr) 5629 { 5630 // flags to stream type mapping 5631 if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { 5632 return AUDIO_STREAM_ENFORCED_AUDIBLE; 5633 } 5634 if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { 5635 return AUDIO_STREAM_BLUETOOTH_SCO; 5636 } 5637 if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { 5638 return AUDIO_STREAM_TTS; 5639 } 5640 5641 // usage to stream type mapping 5642 switch (attr->usage) { 5643 case AUDIO_USAGE_MEDIA: 5644 case AUDIO_USAGE_GAME: 5645 case AUDIO_USAGE_ASSISTANT: 5646 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 5647 return AUDIO_STREAM_MUSIC; 5648 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 5649 return AUDIO_STREAM_ACCESSIBILITY; 5650 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 5651 return AUDIO_STREAM_SYSTEM; 5652 case AUDIO_USAGE_VOICE_COMMUNICATION: 5653 return AUDIO_STREAM_VOICE_CALL; 5654 5655 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 5656 return AUDIO_STREAM_DTMF; 5657 5658 case AUDIO_USAGE_ALARM: 5659 return AUDIO_STREAM_ALARM; 5660 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 5661 return AUDIO_STREAM_RING; 5662 5663 case AUDIO_USAGE_NOTIFICATION: 5664 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 5665 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 5666 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 5667 case AUDIO_USAGE_NOTIFICATION_EVENT: 5668 return AUDIO_STREAM_NOTIFICATION; 5669 5670 case AUDIO_USAGE_UNKNOWN: 5671 default: 5672 return AUDIO_STREAM_MUSIC; 5673 } 5674 } 5675 5676 bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) 5677 { 5678 // has flags that map to a strategy? 5679 if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) { 5680 return true; 5681 } 5682 5683 // has known usage? 5684 switch (paa->usage) { 5685 case AUDIO_USAGE_UNKNOWN: 5686 case AUDIO_USAGE_MEDIA: 5687 case AUDIO_USAGE_VOICE_COMMUNICATION: 5688 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 5689 case AUDIO_USAGE_ALARM: 5690 case AUDIO_USAGE_NOTIFICATION: 5691 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 5692 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 5693 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 5694 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 5695 case AUDIO_USAGE_NOTIFICATION_EVENT: 5696 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 5697 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 5698 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 5699 case AUDIO_USAGE_GAME: 5700 case AUDIO_USAGE_VIRTUAL_SOURCE: 5701 case AUDIO_USAGE_ASSISTANT: 5702 break; 5703 default: 5704 return false; 5705 } 5706 return true; 5707 } 5708 5709 bool AudioPolicyManager::isStrategyActive(const sp<AudioOutputDescriptor>& outputDesc, 5710 routing_strategy strategy, uint32_t inPastMs, 5711 nsecs_t sysTime) const 5712 { 5713 if ((sysTime == 0) && (inPastMs != 0)) { 5714 sysTime = systemTime(); 5715 } 5716 for (int i = 0; i < (int)AUDIO_STREAM_FOR_POLICY_CNT; i++) { 5717 if (((getStrategy((audio_stream_type_t)i) == strategy) || 5718 (NUM_STRATEGIES == strategy)) && 5719 outputDesc->isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) { 5720 return true; 5721 } 5722 } 5723 return false; 5724 } 5725 5726 audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) 5727 { 5728 return mEngine->getForceUse(usage); 5729 } 5730 5731 bool AudioPolicyManager::isInCall() 5732 { 5733 return isStateInCall(mEngine->getPhoneState()); 5734 } 5735 5736 bool AudioPolicyManager::isStateInCall(int state) 5737 { 5738 return is_state_in_call(state); 5739 } 5740 5741 void AudioPolicyManager::cleanUpForDevice(const sp<DeviceDescriptor>& deviceDesc) 5742 { 5743 for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--) { 5744 sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i); 5745 if (sourceDesc->mDevice->equals(deviceDesc)) { 5746 ALOGV("%s releasing audio source %d", __FUNCTION__, sourceDesc->getHandle()); 5747 stopAudioSource(sourceDesc->getHandle()); 5748 } 5749 } 5750 5751 for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) { 5752 sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i); 5753 bool release = false; 5754 for (size_t j = 0; j < patchDesc->mPatch.num_sources && !release; j++) { 5755 const struct audio_port_config *source = &patchDesc->mPatch.sources[j]; 5756 if (source->type == AUDIO_PORT_TYPE_DEVICE && 5757 source->ext.device.type == deviceDesc->type()) { 5758 release = true; 5759 } 5760 } 5761 for (size_t j = 0; j < patchDesc->mPatch.num_sinks && !release; j++) { 5762 const struct audio_port_config *sink = &patchDesc->mPatch.sinks[j]; 5763 if (sink->type == AUDIO_PORT_TYPE_DEVICE && 5764 sink->ext.device.type == deviceDesc->type()) { 5765 release = true; 5766 } 5767 } 5768 if (release) { 5769 ALOGV("%s releasing patch %u", __FUNCTION__, patchDesc->mHandle); 5770 releaseAudioPatch(patchDesc->mHandle, patchDesc->mUid); 5771 } 5772 } 5773 } 5774 5775 // Modify the list of surround sound formats supported. 5776 void AudioPolicyManager::filterSurroundFormats(FormatVector *formatsPtr) { 5777 FormatVector &formats = *formatsPtr; 5778 // TODO Set this based on Config properties. 5779 const bool alwaysForceAC3 = true; 5780 5781 audio_policy_forced_cfg_t forceUse = mEngine->getForceUse( 5782 AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND); 5783 ALOGD("%s: forced use = %d", __FUNCTION__, forceUse); 5784 5785 // Analyze original support for various formats. 5786 bool supportsAC3 = false; 5787 bool supportsOtherSurround = false; 5788 bool supportsIEC61937 = false; 5789 for (size_t formatIndex = 0; formatIndex < formats.size(); formatIndex++) { 5790 audio_format_t format = formats[formatIndex]; 5791 switch (format) { 5792 case AUDIO_FORMAT_AC3: 5793 supportsAC3 = true; 5794 break; 5795 case AUDIO_FORMAT_E_AC3: 5796 case AUDIO_FORMAT_DTS: 5797 case AUDIO_FORMAT_DTS_HD: 5798 supportsOtherSurround = true; 5799 break; 5800 case AUDIO_FORMAT_IEC61937: 5801 supportsIEC61937 = true; 5802 break; 5803 default: 5804 break; 5805 } 5806 } 5807 5808 // Modify formats based on surround preferences. 5809 // If NEVER, remove support for surround formats. 5810 if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) { 5811 if (supportsAC3 || supportsOtherSurround || supportsIEC61937) { 5812 // Remove surround sound related formats. 5813 for (size_t formatIndex = 0; formatIndex < formats.size(); ) { 5814 audio_format_t format = formats[formatIndex]; 5815 switch(format) { 5816 case AUDIO_FORMAT_AC3: 5817 case AUDIO_FORMAT_E_AC3: 5818 case AUDIO_FORMAT_DTS: 5819 case AUDIO_FORMAT_DTS_HD: 5820 case AUDIO_FORMAT_IEC61937: 5821 formats.removeAt(formatIndex); 5822 break; 5823 default: 5824 formatIndex++; // keep it 5825 break; 5826 } 5827 } 5828 supportsAC3 = false; 5829 supportsOtherSurround = false; 5830 supportsIEC61937 = false; 5831 } 5832 } else { // AUTO or ALWAYS 5833 // Most TVs support AC3 even if they do not report it in the EDID. 5834 if ((alwaysForceAC3 || (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS)) 5835 && !supportsAC3) { 5836 formats.add(AUDIO_FORMAT_AC3); 5837 supportsAC3 = true; 5838 } 5839 5840 // If ALWAYS, add support for raw surround formats if all are missing. 5841 // This assumes that if any of these formats are reported by the HAL 5842 // then the report is valid and should not be modified. 5843 if ((forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS) 5844 && !supportsOtherSurround) { 5845 formats.add(AUDIO_FORMAT_E_AC3); 5846 formats.add(AUDIO_FORMAT_DTS); 5847 formats.add(AUDIO_FORMAT_DTS_HD); 5848 supportsOtherSurround = true; 5849 } 5850 5851 // Add support for IEC61937 if any raw surround supported. 5852 // The HAL could do this but add it here, just in case. 5853 if ((supportsAC3 || supportsOtherSurround) && !supportsIEC61937) { 5854 formats.add(AUDIO_FORMAT_IEC61937); 5855 supportsIEC61937 = true; 5856 } 5857 } 5858 } 5859 5860 // Modify the list of channel masks supported. 5861 void AudioPolicyManager::filterSurroundChannelMasks(ChannelsVector *channelMasksPtr) { 5862 ChannelsVector &channelMasks = *channelMasksPtr; 5863 audio_policy_forced_cfg_t forceUse = mEngine->getForceUse( 5864 AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND); 5865 5866 // If NEVER, then remove support for channelMasks > stereo. 5867 if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) { 5868 for (size_t maskIndex = 0; maskIndex < channelMasks.size(); ) { 5869 audio_channel_mask_t channelMask = channelMasks[maskIndex]; 5870 if (channelMask & ~AUDIO_CHANNEL_OUT_STEREO) { 5871 ALOGI("%s: force NEVER, so remove channelMask 0x%08x", __FUNCTION__, channelMask); 5872 channelMasks.removeAt(maskIndex); 5873 } else { 5874 maskIndex++; 5875 } 5876 } 5877 // If ALWAYS, then make sure we at least support 5.1 5878 } else if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS) { 5879 bool supports5dot1 = false; 5880 // Are there any channel masks that can be considered "surround"? 5881 for (size_t maskIndex = 0; maskIndex < channelMasks.size(); maskIndex++) { 5882 audio_channel_mask_t channelMask = channelMasks[maskIndex]; 5883 if ((channelMask & AUDIO_CHANNEL_OUT_5POINT1) == AUDIO_CHANNEL_OUT_5POINT1) { 5884 supports5dot1 = true; 5885 break; 5886 } 5887 } 5888 // If not then add 5.1 support. 5889 if (!supports5dot1) { 5890 channelMasks.add(AUDIO_CHANNEL_OUT_5POINT1); 5891 ALOGI("%s: force ALWAYS, so adding channelMask for 5.1 surround", __FUNCTION__); 5892 } 5893 } 5894 } 5895 5896 void AudioPolicyManager::updateAudioProfiles(audio_devices_t device, 5897 audio_io_handle_t ioHandle, 5898 AudioProfileVector &profiles) 5899 { 5900 String8 reply; 5901 5902 // Format MUST be checked first to update the list of AudioProfile 5903 if (profiles.hasDynamicFormat()) { 5904 reply = mpClientInterface->getParameters( 5905 ioHandle, String8(AudioParameter::keyStreamSupportedFormats)); 5906 ALOGV("%s: supported formats %s", __FUNCTION__, reply.string()); 5907 AudioParameter repliedParameters(reply); 5908 if (repliedParameters.get( 5909 String8(AudioParameter::keyStreamSupportedFormats), reply) != NO_ERROR) { 5910 ALOGE("%s: failed to retrieve format, bailing out", __FUNCTION__); 5911 return; 5912 } 5913 FormatVector formats = formatsFromString(reply.string()); 5914 if (device == AUDIO_DEVICE_OUT_HDMI) { 5915 filterSurroundFormats(&formats); 5916 } 5917 profiles.setFormats(formats); 5918 } 5919 const FormatVector &supportedFormats = profiles.getSupportedFormats(); 5920 5921 for (size_t formatIndex = 0; formatIndex < supportedFormats.size(); formatIndex++) { 5922 audio_format_t format = supportedFormats[formatIndex]; 5923 ChannelsVector channelMasks; 5924 SampleRateVector samplingRates; 5925 AudioParameter requestedParameters; 5926 requestedParameters.addInt(String8(AudioParameter::keyFormat), format); 5927 5928 if (profiles.hasDynamicRateFor(format)) { 5929 reply = mpClientInterface->getParameters( 5930 ioHandle, 5931 requestedParameters.toString() + ";" + 5932 AudioParameter::keyStreamSupportedSamplingRates); 5933 ALOGV("%s: supported sampling rates %s", __FUNCTION__, reply.string()); 5934 AudioParameter repliedParameters(reply); 5935 if (repliedParameters.get( 5936 String8(AudioParameter::keyStreamSupportedSamplingRates), reply) == NO_ERROR) { 5937 samplingRates = samplingRatesFromString(reply.string()); 5938 } 5939 } 5940 if (profiles.hasDynamicChannelsFor(format)) { 5941 reply = mpClientInterface->getParameters(ioHandle, 5942 requestedParameters.toString() + ";" + 5943 AudioParameter::keyStreamSupportedChannels); 5944 ALOGV("%s: supported channel masks %s", __FUNCTION__, reply.string()); 5945 AudioParameter repliedParameters(reply); 5946 if (repliedParameters.get( 5947 String8(AudioParameter::keyStreamSupportedChannels), reply) == NO_ERROR) { 5948 channelMasks = channelMasksFromString(reply.string()); 5949 if (device == AUDIO_DEVICE_OUT_HDMI) { 5950 filterSurroundChannelMasks(&channelMasks); 5951 } 5952 } 5953 } 5954 profiles.addProfileFromHal(new AudioProfile(format, channelMasks, samplingRates)); 5955 } 5956 } 5957 5958 }; // namespace android 5959