/* * Copyright (C) 2009 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ #include "utils/Errors.h" #define LOG_TAG "APM_AudioPolicyManager" // Need to keep the log statements even in production builds // to enable VERBOSE logging dynamically. // You can enable VERBOSE logging as follows: // adb shell setprop log.tag.APM_AudioPolicyManager V #define LOG_NDEBUG 0 //#define VERY_VERBOSE_LOGGING #ifdef VERY_VERBOSE_LOGGING #define ALOGVV ALOGV #else #define ALOGVV(a...) do { } while(0) #endif #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "AudioPolicyManager.h" #include "TypeConverter.h" namespace android { namespace audio_flags = android::media::audiopolicy; using android::media::audio::common::AudioDevice; using android::media::audio::common::AudioDeviceAddress; using android::media::audio::common::AudioPortDeviceExt; using android::media::audio::common::AudioPortExt; using content::AttributionSourceState; //FIXME: workaround for truncated touch sounds // to be removed when the problem is handled by system UI #define TOUCH_SOUND_FIXED_DELAY_MS 100 // Largest difference in dB on earpiece in call between the voice volume and another // media / notification / system volume. constexpr float IN_CALL_EARPIECE_HEADROOM_DB = 3.f; template bool operator== (const SortedVector &left, const SortedVector &right) { if (left.size() != right.size()) { return false; } for (size_t index = 0; index < right.size(); index++) { if (left[index] != right[index]) { return false; } } return true; } template bool operator!= (const SortedVector &left, const SortedVector &right) { return !(left == right); } // ---------------------------------------------------------------------------- // AudioPolicyInterface implementation // ---------------------------------------------------------------------------- status_t AudioPolicyManager::setDeviceConnectionState(audio_policy_dev_state_t state, const android::media::audio::common::AudioPort& port, audio_format_t encodedFormat) { status_t status = setDeviceConnectionStateInt(state, port, encodedFormat); nextAudioPortGeneration(); return status; } status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char* device_address, const char* device_name, audio_format_t encodedFormat) { media::AudioPortFw aidlPort; if (status_t status = deviceToAudioPort(device, device_address, device_name, &aidlPort); status == OK) { return setDeviceConnectionState(state, aidlPort.hal, encodedFormat); } else { ALOGE("Failed to convert to AudioPort Parcelable: %s", statusToString(status).c_str()); return status; } } status_t AudioPolicyManager::broadcastDeviceConnectionState(const sp &device, media::DeviceConnectedState state) { audio_port_v7 devicePort; device->toAudioPort(&devicePort); status_t status = mpClientInterface->setDeviceConnectedState(&devicePort, state); ALOGE_IF(status != OK, "Error %d while setting connected state %d for device %s", status, static_cast(state), device->getDeviceTypeAddr().toString(false).c_str()); return status; } status_t AudioPolicyManager::setDeviceConnectionStateInt( audio_policy_dev_state_t state, const android::media::audio::common::AudioPort& port, audio_format_t encodedFormat) { if (port.ext.getTag() != AudioPortExt::device) { return BAD_VALUE; } audio_devices_t device_type; std::string device_address; if (status_t status = aidl2legacy_AudioDevice_audio_device( port.ext.get().device, &device_type, &device_address); status != OK) { return status; }; const char* device_name = port.name.c_str(); // connect/disconnect only 1 device at a time if (!audio_is_output_device(device_type) && !audio_is_input_device(device_type)) return BAD_VALUE; sp device = mHwModules.getDeviceDescriptor( device_type, device_address.c_str(), device_name, encodedFormat, state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE); if (device == nullptr) { return INVALID_OPERATION; } if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { device->setExtraAudioDescriptors(port.extraAudioDescriptors); } return setDeviceConnectionStateInt(device, state); } status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t deviceType, audio_policy_dev_state_t state, const char* device_address, const char* device_name, audio_format_t encodedFormat) { media::AudioPortFw aidlPort; if (status_t status = deviceToAudioPort(deviceType, device_address, device_name, &aidlPort); status == OK) { return setDeviceConnectionStateInt(state, aidlPort.hal, encodedFormat); } else { ALOGE("Failed to convert to AudioPort Parcelable: %s", statusToString(status).c_str()); return status; } } status_t AudioPolicyManager::setDeviceConnectionStateInt(const sp &device, audio_policy_dev_state_t state) { // handle output devices if (audio_is_output_device(device->type())) { SortedVector outputs; ssize_t index = mAvailableOutputDevices.indexOf(device); // save a copy of the opened output descriptors before any output is opened or closed // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies() mPreviousOutputs = mOutputs; bool wasLeUnicastActive = isLeUnicastActive(); switch (state) { // handle output device connection case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { if (index >= 0) { ALOGW("%s() device already connected: %s", __func__, device->toString().c_str()); return INVALID_OPERATION; } ALOGV("%s() connecting device %s format %x", __func__, device->toString().c_str(), device->getEncodedFormat()); // register new device as available if (mAvailableOutputDevices.add(device) < 0) { return NO_MEMORY; } // Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic // parameters on newly connected devices (instead of opening the outputs...) if (broadcastDeviceConnectionState( device, media::DeviceConnectedState::CONNECTED) != NO_ERROR) { mAvailableOutputDevices.remove(device); mHwModules.cleanUpForDevice(device); ALOGE("%s() device %s format %x connection failed", __func__, device->toString().c_str(), device->getEncodedFormat()); return INVALID_OPERATION; } if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) { mAvailableOutputDevices.remove(device); broadcastDeviceConnectionState(device, media::DeviceConnectedState::DISCONNECTED); mHwModules.cleanUpForDevice(device); return INVALID_OPERATION; } // Populate encapsulation information when a output device is connected. device->setEncapsulationInfoFromHal(mpClientInterface); // outputs should never be empty here ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():" "checkOutputsForDevice() returned no outputs but status OK"); ALOGV("%s() checkOutputsForDevice() returned %zu outputs", __func__, outputs.size()); } break; // handle output device disconnection case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { if (index < 0) { ALOGW("%s() device not connected: %s", __func__, device->toString().c_str()); return INVALID_OPERATION; } ALOGV("%s() disconnecting output device %s", __func__, device->toString().c_str()); // Notify the HAL to prepare to disconnect device broadcastDeviceConnectionState( device, media::DeviceConnectedState::PREPARE_TO_DISCONNECT); // remove device from available output devices mAvailableOutputDevices.remove(device); mOutputs.clearSessionRoutesForDevice(device); checkOutputsForDevice(device, state, outputs); // Send Disconnect to HALs broadcastDeviceConnectionState(device, media::DeviceConnectedState::DISCONNECTED); // Reset active device codec device->setEncodedFormat(AUDIO_FORMAT_DEFAULT); // remove device from mReportedFormatsMap cache mReportedFormatsMap.erase(device); // remove preferred mixer configurations mPreferredMixerAttrInfos.erase(device->getId()); } break; default: ALOGE("%s() invalid state: %x", __func__, state); return BAD_VALUE; } // Propagate device availability to Engine setEngineDeviceConnectionState(device, state); // No need to evaluate playback routing when connecting a remote submix // output device used by a dynamic policy of type recorder as no // playback use case is affected. bool doCheckForDeviceAndOutputChanges = true; if (device->type() == AUDIO_DEVICE_OUT_REMOTE_SUBMIX && device->address() != "0") { for (audio_io_handle_t output : outputs) { sp desc = mOutputs.valueFor(output); sp policyMix = desc->mPolicyMix.promote(); if (policyMix != nullptr && policyMix->mMixType == MIX_TYPE_RECORDERS && device->address() == policyMix->mDeviceAddress.c_str()) { doCheckForDeviceAndOutputChanges = false; break; } } } auto checkCloseOutputs = [&]() { // outputs must be closed after checkOutputForAllStrategies() is executed if (!outputs.isEmpty()) { for (audio_io_handle_t output : outputs) { sp desc = mOutputs.valueFor(output); // close unused outputs after device disconnection or direct outputs that have // been opened by checkOutputsForDevice() to query dynamic parameters // "outputs" vector never contains duplicated outputs if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) || (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && (desc->mDirectOpenCount == 0)) || (((desc->mFlags & AUDIO_OUTPUT_FLAG_SPATIALIZER) != 0) && !isOutputOnlyAvailableRouteToSomeDevice(desc))) { clearAudioSourcesForOutput(output); closeOutput(output); } } // check A2DP again after closing A2DP output to reset mA2dpSuspended if needed return true; } return false; }; if (doCheckForDeviceAndOutputChanges) { checkForDeviceAndOutputChanges(checkCloseOutputs); } else { checkCloseOutputs(); } (void)updateCallRouting(false /*fromCache*/); const DeviceVector msdOutDevices = getMsdAudioOutDevices(); const DeviceVector activeMediaDevices = mEngine->getActiveMediaDevices(mAvailableOutputDevices); std::map outputsToReopenWithDevices; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (desc->isActive() && ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput))) { DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/); // do not force device change on duplicated output because if device is 0, it will // also force a device 0 for the two outputs it is duplicated to which may override // a valid device selection on those outputs. bool force = (msdOutDevices.isEmpty() || msdOutDevices != desc->devices()) && !desc->isDuplicated() && (!device_distinguishes_on_address(device->type()) // always force when disconnecting (a non-duplicated device) || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); if (desc->mPreferredAttrInfo != nullptr && newDevices != desc->devices()) { // If the device is using preferred mixer attributes, the output need to reopen // with default configuration when the new selected devices are different from // current routing devices outputsToReopenWithDevices.emplace(mOutputs.keyAt(i), newDevices); continue; } setOutputDevices(__func__, desc, newDevices, force, 0); } if (!desc->isDuplicated() && desc->mProfile->hasDynamicAudioProfile() && !activeMediaDevices.empty() && desc->devices() != activeMediaDevices && desc->supportsDevicesForPlayback(activeMediaDevices)) { // Reopen the output to query the dynamic profiles when there is not active // clients or all active clients will be rerouted. Otherwise, set the flag // `mPendingReopenToQueryProfiles` in the SwOutputDescriptor so that the output // can be reopened to query dynamic profiles when all clients are inactive. if (areAllActiveTracksRerouted(desc)) { outputsToReopenWithDevices.emplace(mOutputs.keyAt(i), activeMediaDevices); } else { desc->mPendingReopenToQueryProfiles = true; } } if (!desc->supportsDevicesForPlayback(activeMediaDevices)) { // Clear the flag that previously set for re-querying profiles. desc->mPendingReopenToQueryProfiles = false; } } reopenOutputsWithDevices(outputsToReopenWithDevices); if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { cleanUpForDevice(device); } checkLeBroadcastRoutes(wasLeUnicastActive, nullptr, 0); mpClientInterface->onAudioPortListUpdate(); ALOGV("%s() completed for device: %s", __func__, device->toString().c_str()); return NO_ERROR; } // end if is output device // handle input devices if (audio_is_input_device(device->type())) { ssize_t index = mAvailableInputDevices.indexOf(device); switch (state) { // handle input device connection case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { if (index >= 0) { ALOGW("%s() device already connected: %s", __func__, device->toString().c_str()); return INVALID_OPERATION; } ALOGV("%s() connecting device %s", __func__, device->toString().c_str()); if (mAvailableInputDevices.add(device) < 0) { return NO_MEMORY; } // Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic // parameters on newly connected devices (instead of opening the inputs...) if (broadcastDeviceConnectionState( device, media::DeviceConnectedState::CONNECTED) != NO_ERROR) { mAvailableInputDevices.remove(device); mHwModules.cleanUpForDevice(device); ALOGE("%s() device %s format %x connection failed", __func__, device->toString().c_str(), device->getEncodedFormat()); return INVALID_OPERATION; } // Propagate device availability to Engine setEngineDeviceConnectionState(device, state); if (checkInputsForDevice(device, state) != NO_ERROR) { setEngineDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE); mAvailableInputDevices.remove(device); broadcastDeviceConnectionState(device, media::DeviceConnectedState::DISCONNECTED); mHwModules.cleanUpForDevice(device); return INVALID_OPERATION; } } break; // handle input device disconnection case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { if (index < 0) { ALOGW("%s() device not connected: %s", __func__, device->toString().c_str()); return INVALID_OPERATION; } ALOGV("%s() disconnecting input device %s", __func__, device->toString().c_str()); // Notify the HAL to prepare to disconnect device broadcastDeviceConnectionState( device, media::DeviceConnectedState::PREPARE_TO_DISCONNECT); mAvailableInputDevices.remove(device); checkInputsForDevice(device, state); // Set Disconnect to HALs broadcastDeviceConnectionState(device, media::DeviceConnectedState::DISCONNECTED); // remove device from mReportedFormatsMap cache mReportedFormatsMap.erase(device); // Propagate device availability to Engine setEngineDeviceConnectionState(device, state); } break; default: ALOGE("%s() invalid state: %x", __func__, state); return BAD_VALUE; } checkCloseInputs(); // As the input device list can impact the output device selection, update // getDeviceForStrategy() cache updateDevicesAndOutputs(); (void)updateCallRouting(false /*fromCache*/); // Reconnect Audio Source for (const auto &strategy : mEngine->getOrderedProductStrategies()) { auto attributes = mEngine->getAllAttributesForProductStrategy(strategy).front(); checkAudioSourceForAttributes(attributes); } if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) { cleanUpForDevice(device); } mpClientInterface->onAudioPortListUpdate(); ALOGV("%s() completed for device: %s", __func__, device->toString().c_str()); return NO_ERROR; } // end if is input device ALOGW("%s() invalid device: %s", __func__, device->toString().c_str()); return BAD_VALUE; } status_t AudioPolicyManager::deviceToAudioPort(audio_devices_t device, const char* device_address, const char* device_name, media::AudioPortFw* aidlPort) { const auto devDescr = sp::make(device, device_address); devDescr->setName(device_name); return devDescr->writeToParcelable(aidlPort); } void AudioPolicyManager::setEngineDeviceConnectionState(const sp device, audio_policy_dev_state_t state) { // the Engine does not have to know about remote submix devices used by dynamic audio policies if (audio_is_remote_submix_device(device->type()) && device->address() != "0") { return; } mEngine->setDeviceConnectionState(device, state); } audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device, const char *device_address) { sp devDesc = mHwModules.getDeviceDescriptor(device, device_address, "", AUDIO_FORMAT_DEFAULT, false /* allowToCreate */, (strlen(device_address) != 0)/*matchAddress*/); if (devDesc == 0) { ALOGV("getDeviceConnectionState() undeclared device, type %08x, address: %s", device, device_address); return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; } DeviceVector *deviceVector; if (audio_is_output_device(device)) { deviceVector = &mAvailableOutputDevices; } else if (audio_is_input_device(device)) { deviceVector = &mAvailableInputDevices; } else { ALOGW("%s() invalid device type %08x", __func__, device); return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; } return (deviceVector->getDevice( device, String8(device_address), AUDIO_FORMAT_DEFAULT) != 0) ? AUDIO_POLICY_DEVICE_STATE_AVAILABLE : AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; } status_t AudioPolicyManager::handleDeviceConfigChange(audio_devices_t device, const char *device_address, const char *device_name, audio_format_t encodedFormat) { ALOGV("handleDeviceConfigChange(() device: 0x%X, address %s name %s encodedFormat: 0x%X", device, device_address, device_name, encodedFormat); // connect/disconnect only 1 device at a time if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; // Check if the device is currently connected DeviceVector deviceList = mAvailableOutputDevices.getDevicesFromType(device); if (deviceList.empty()) { // Nothing to do: device is not connected return NO_ERROR; } sp devDesc = deviceList.itemAt(0); // For offloaded A2DP, Hw modules may have the capability to // configure codecs. // Handle two specific cases by sending a set parameter to // configure A2DP codecs. No need to toggle device state. // Case 1: A2DP active device switches from primary to primary // module // Case 2: A2DP device config changes on primary module. if (device_has_encoding_capability(device) && hasPrimaryOutput()) { sp module = mHwModules.getModuleForDeviceType(device, encodedFormat); audio_module_handle_t primaryHandle = mPrimaryOutput->getModuleHandle(); if (availablePrimaryOutputDevices().contains(devDesc) && (module != 0 && module->getHandle() == primaryHandle)) { bool isA2dp = audio_is_a2dp_out_device(device); const String8 supportKey = isA2dp ? String8(AudioParameter::keyReconfigA2dpSupported) : String8(AudioParameter::keyReconfigLeSupported); String8 reply = mpClientInterface->getParameters(AUDIO_IO_HANDLE_NONE, supportKey); AudioParameter repliedParameters(reply); int isReconfigSupported; repliedParameters.getInt(supportKey, isReconfigSupported); if (isReconfigSupported) { const String8 key = isA2dp ? String8(AudioParameter::keyReconfigA2dp) : String8(AudioParameter::keyReconfigLe); AudioParameter param; param.add(key, String8("true")); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); devDesc->setEncodedFormat(encodedFormat); return NO_ERROR; } } } auto musicStrategy = streamToStrategy(AUDIO_STREAM_MUSIC); uint32_t muteWaitMs = 0; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); // mute media strategies to avoid sending the music tail into // the earpiece or headset. if (desc->isStrategyActive(musicStrategy)) { uint32_t tempRecommendedMuteDuration = desc->getRecommendedMuteDurationMs(); uint32_t tempMuteDurationMs = tempRecommendedMuteDuration > 0 ? tempRecommendedMuteDuration : desc->latency() * 4; if (muteWaitMs < tempMuteDurationMs) { muteWaitMs = tempMuteDurationMs; } } setStrategyMute(musicStrategy, true, desc); setStrategyMute(musicStrategy, false, desc, MUTE_TIME_MS, mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA), nullptr, true /*fromCache*/).types()); } // Wait for the muted audio to propagate down the audio path see checkDeviceMuteStrategies(). // We assume that MUTE_TIME_MS is way larger than muteWaitMs so that unmuting still // happens after the actual device switch. if (muteWaitMs > 0) { ALOGW_IF(MUTE_TIME_MS < muteWaitMs * 2, "%s excessive mute wait %d", __func__, muteWaitMs); usleep(muteWaitMs * 1000); } // Toggle the device state: UNAVAILABLE -> AVAILABLE // This will force reading again the device configuration status_t status = setDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, device_address, device_name, devDesc->getEncodedFormat()); if (status != NO_ERROR) { ALOGW("handleDeviceConfigChange() error disabling connection state: %d", status); return status; } status = setDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, device_address, device_name, encodedFormat); if (status != NO_ERROR) { ALOGW("handleDeviceConfigChange() error enabling connection state: %d", status); return status; } return NO_ERROR; } status_t AudioPolicyManager::getHwOffloadFormatsSupportedForBluetoothMedia( audio_devices_t device, std::vector *formats) { ALOGV("getHwOffloadFormatsSupportedForBluetoothMedia()"); status_t status = NO_ERROR; std::unordered_set formatSet; sp primaryModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY); if (primaryModule == nullptr) { ALOGE("%s() unable to get primary module", __func__); return NO_INIT; } DeviceTypeSet audioDeviceSet; switch(device) { case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP: audioDeviceSet = getAudioDeviceOutAllA2dpSet(); break; case AUDIO_DEVICE_OUT_BLE_HEADSET: audioDeviceSet = getAudioDeviceOutLeAudioUnicastSet(); break; case AUDIO_DEVICE_OUT_BLE_BROADCAST: audioDeviceSet = getAudioDeviceOutLeAudioBroadcastSet(); break; default: ALOGE("%s() device type 0x%08x not supported", __func__, device); return BAD_VALUE; } DeviceVector declaredDevices = primaryModule->getDeclaredDevices().getDevicesFromTypes( audioDeviceSet); for (const auto& device : declaredDevices) { formatSet.insert(device->encodedFormats().begin(), device->encodedFormats().end()); } formats->assign(formatSet.begin(), formatSet.end()); return status; } DeviceVector AudioPolicyManager::selectBestRxSinkDevicesForCall(bool fromCache) { DeviceVector rxSinkdevices{}; rxSinkdevices = mEngine->getOutputDevicesForAttributes( attributes_initializer(AUDIO_USAGE_VOICE_COMMUNICATION), nullptr, fromCache); if (!rxSinkdevices.isEmpty() && mAvailableOutputDevices.contains(rxSinkdevices.itemAt(0))) { auto rxSinkDevice = rxSinkdevices.itemAt(0); auto telephonyRxModule = mHwModules.getModuleForDeviceType( AUDIO_DEVICE_IN_TELEPHONY_RX, AUDIO_FORMAT_DEFAULT); // retrieve Rx Source device descriptor sp rxSourceDevice = mAvailableInputDevices.getDevice( AUDIO_DEVICE_IN_TELEPHONY_RX, String8(), AUDIO_FORMAT_DEFAULT); // RX Telephony and Rx sink devices are declared by Primary Audio HAL if (isPrimaryModule(telephonyRxModule) && (telephonyRxModule->getHalVersionMajor() >= 3) && telephonyRxModule->supportsPatch(rxSourceDevice, rxSinkDevice)) { ALOGW("%s() device %s using HW Bridge", __func__, rxSinkDevice->toString().c_str()); return DeviceVector(rxSinkDevice); } } // Note that despite the fact that getNewOutputDevices() is called on the primary output, // the device returned is not necessarily reachable via this output // (filter later by setOutputDevices()) return getNewOutputDevices(mPrimaryOutput, fromCache); } status_t AudioPolicyManager::updateCallRouting(bool fromCache, uint32_t delayMs, uint32_t *waitMs) { if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) { DeviceVector rxDevices = selectBestRxSinkDevicesForCall(fromCache); return updateCallRoutingInternal(rxDevices, delayMs, waitMs); } return INVALID_OPERATION; } status_t AudioPolicyManager::updateCallRoutingInternal( const DeviceVector &rxDevices, uint32_t delayMs, uint32_t *waitMs) { bool createTxPatch = false; bool createRxPatch = false; uint32_t muteWaitMs = 0; if (hasPrimaryOutput() && mPrimaryOutput->devices().onlyContainsDevicesWithType(AUDIO_DEVICE_OUT_STUB)) { return INVALID_OPERATION; } audio_attributes_t attr = { .source = AUDIO_SOURCE_VOICE_COMMUNICATION }; auto txSourceDevice = mEngine->getInputDeviceForAttributes(attr); disconnectTelephonyAudioSource(mCallRxSourceClient); disconnectTelephonyAudioSource(mCallTxSourceClient); if (rxDevices.isEmpty()) { ALOGW("%s() no selected output device", __func__); return INVALID_OPERATION; } if (txSourceDevice == nullptr) { ALOGE("%s() selected input device not available", __func__); return INVALID_OPERATION; } ALOGV("%s device rxDevice %s txDevice %s", __func__, rxDevices.itemAt(0)->toString().c_str(), txSourceDevice->toString().c_str()); auto telephonyRxModule = mHwModules.getModuleForDeviceType(AUDIO_DEVICE_IN_TELEPHONY_RX, AUDIO_FORMAT_DEFAULT); auto telephonyTxModule = mHwModules.getModuleForDeviceType(AUDIO_DEVICE_OUT_TELEPHONY_TX, AUDIO_FORMAT_DEFAULT); // retrieve Rx Source and Tx Sink device descriptors sp rxSourceDevice = mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_TELEPHONY_RX, String8(), AUDIO_FORMAT_DEFAULT); sp txSinkDevice = mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT); // RX and TX Telephony device are declared by Primary Audio HAL if (isPrimaryModule(telephonyRxModule) && isPrimaryModule(telephonyTxModule) && (telephonyRxModule->getHalVersionMajor() >= 3)) { if (rxSourceDevice == 0 || txSinkDevice == 0) { // RX / TX Telephony device(s) is(are) not currently available ALOGE("%s() no telephony Tx and/or RX device", __func__); return INVALID_OPERATION; } // createAudioPatchInternal now supports both HW / SW bridging createRxPatch = true; createTxPatch = true; } else { // If the RX device is on the primary HW module, then use legacy routing method for // voice calls via setOutputDevice() on primary output. // Otherwise, create two audio patches for TX and RX path. createRxPatch = !(availablePrimaryOutputDevices().contains(rxDevices.itemAt(0))) && (rxSourceDevice != 0); // If the TX device is also on the primary HW module, setOutputDevice() will take care // of it due to legacy implementation. If not, create a patch. createTxPatch = !(availablePrimaryModuleInputDevices().contains(txSourceDevice)) && (txSinkDevice != 0); } // Use legacy routing method for voice calls via setOutputDevice() on primary output. // Otherwise, create two audio patches for TX and RX path. if (!createRxPatch) { if (!hasPrimaryOutput()) { ALOGW("%s() no primary output available", __func__); return INVALID_OPERATION; } muteWaitMs = setOutputDevices(__func__, mPrimaryOutput, rxDevices, true, delayMs); } else { // create RX path audio patch connectTelephonyRxAudioSource(delayMs); // If the TX device is on the primary HW module but RX device is // on other HW module, SinkMetaData of telephony input should handle it // assuming the device uses audio HAL V5.0 and above } if (createTxPatch) { // create TX path audio patch // terminate active capture if on the same HW module as the call TX source device // FIXME: would be better to refine to only inputs whose profile connects to the // call TX device but this information is not in the audio patch and logic here must be // symmetric to the one in startInput() for (const auto& activeDesc : mInputs.getActiveInputs()) { if (activeDesc->hasSameHwModuleAs(txSourceDevice)) { closeActiveClients(activeDesc); } } connectTelephonyTxAudioSource(txSourceDevice, txSinkDevice, delayMs); } if (waitMs != nullptr) { *waitMs = muteWaitMs; } return NO_ERROR; } bool AudioPolicyManager::isDeviceOfModule( const sp& devDesc, const char *moduleId) const { sp module = mHwModules.getModuleFromName(moduleId); if (module != 0) { return mAvailableOutputDevices.getDevicesFromHwModule(module->getHandle()) .indexOf(devDesc) != NAME_NOT_FOUND || mAvailableInputDevices.getDevicesFromHwModule(module->getHandle()) .indexOf(devDesc) != NAME_NOT_FOUND; } return false; } void AudioPolicyManager::connectTelephonyRxAudioSource(uint32_t delayMs) { disconnectTelephonyAudioSource(mCallRxSourceClient); const struct audio_port_config source = { .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_DEVICE, .ext.device.type = AUDIO_DEVICE_IN_TELEPHONY_RX, .ext.device.address = "" }; const auto aa = mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL); audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE; status_t status = startAudioSourceInternal(&source, &aa, &portId, 0 /*uid*/, true /*internal*/, true /*isCallRx*/, delayMs); ALOGE_IF(status != OK, "%s: failed to start audio source (%d)", __func__, status); mCallRxSourceClient = mAudioSources.valueFor(portId); ALOGE_IF(mCallRxSourceClient == nullptr, "%s failed to start Telephony Rx AudioSource", __func__); } void AudioPolicyManager::disconnectTelephonyAudioSource(sp &clientDesc) { if (clientDesc == nullptr) { return; } ALOGW_IF(stopAudioSource(clientDesc->portId()) != NO_ERROR, "%s error stopping audio source", __func__); clientDesc.clear(); } void AudioPolicyManager::connectTelephonyTxAudioSource( const sp &srcDevice, const sp &sinkDevice, uint32_t delayMs) { disconnectTelephonyAudioSource(mCallTxSourceClient); if (srcDevice == nullptr || sinkDevice == nullptr) { ALOGW("%s could not create patch, invalid sink and/or source device(s)", __func__); return; } PatchBuilder patchBuilder; patchBuilder.addSource(srcDevice).addSink(sinkDevice); ALOGV("%s between source %s and sink %s", __func__, srcDevice->toString().c_str(), sinkDevice->toString().c_str()); auto callTxSourceClientPortId = PolicyAudioPort::getNextUniqueId(); const auto aa = mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL); struct audio_port_config source = {}; srcDevice->toAudioPortConfig(&source); mCallTxSourceClient = new SourceClientDescriptor( callTxSourceClientPortId, mUidCached, aa, source, srcDevice, AUDIO_STREAM_PATCH, mCommunnicationStrategy, toVolumeSource(aa), true, false /*isCallRx*/, true /*isCallTx*/); mCallTxSourceClient->setPreferredDeviceId(sinkDevice->getId()); audio_patch_handle_t patchHandle = AUDIO_PATCH_HANDLE_NONE; status_t status = connectAudioSourceToSink( mCallTxSourceClient, sinkDevice, patchBuilder.patch(), patchHandle, mUidCached, delayMs); ALOGE_IF(status != NO_ERROR, "%s() error %d creating TX audio patch", __func__, status); if (status == NO_ERROR) { mAudioSources.add(callTxSourceClientPortId, mCallTxSourceClient); } } void AudioPolicyManager::setPhoneState(audio_mode_t state) { ALOGV("setPhoneState() state %d", state); // store previous phone state for management of sonification strategy below int oldState = mEngine->getPhoneState(); bool wasLeUnicastActive = isLeUnicastActive(); if (mEngine->setPhoneState(state) != NO_ERROR) { ALOGW("setPhoneState() invalid or same state %d", state); return; } /// Opens: can these line be executed after the switch of volume curves??? if (isStateInCall(oldState)) { ALOGV("setPhoneState() in call state management: new state is %d", state); // force reevaluating accessibility routing when call stops invalidateStreams({AUDIO_STREAM_ACCESSIBILITY}); } /** * Switching to or from incall state or switching between telephony and VoIP lead to force * routing command. */ bool force = ((isStateInCall(oldState) != isStateInCall(state)) || (isStateInCall(state) && (state != oldState))); // check for device and output changes triggered by new phone state checkForDeviceAndOutputChanges(); int delayMs = 0; if (isStateInCall(state)) { nsecs_t sysTime = systemTime(); auto musicStrategy = streamToStrategy(AUDIO_STREAM_MUSIC); auto sonificationStrategy = streamToStrategy(AUDIO_STREAM_ALARM); for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); // mute media and sonification strategies and delay device switch by the largest // latency of any output where either strategy is active. // This avoid sending the ring tone or music tail into the earpiece or headset. if ((desc->isStrategyActive(musicStrategy, SONIFICATION_HEADSET_MUSIC_DELAY, sysTime) || desc->isStrategyActive(sonificationStrategy, SONIFICATION_HEADSET_MUSIC_DELAY, sysTime)) && (delayMs < (int)desc->latency()*2)) { delayMs = desc->latency()*2; } setStrategyMute(musicStrategy, true, desc); setStrategyMute(musicStrategy, false, desc, MUTE_TIME_MS, mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA), nullptr, true /*fromCache*/).types()); setStrategyMute(sonificationStrategy, true, desc); setStrategyMute(sonificationStrategy, false, desc, MUTE_TIME_MS, mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_ALARM), nullptr, true /*fromCache*/).types()); } } if (state == AUDIO_MODE_IN_CALL) { (void)updateCallRouting(false /*fromCache*/, delayMs); } else { if (oldState == AUDIO_MODE_IN_CALL) { disconnectTelephonyAudioSource(mCallRxSourceClient); disconnectTelephonyAudioSource(mCallTxSourceClient); } if (hasPrimaryOutput()) { DeviceVector rxDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/); // force routing command to audio hardware when ending call // even if no device change is needed if (isStateInCall(oldState) && rxDevices.isEmpty()) { rxDevices = mPrimaryOutput->devices(); } setOutputDevices(__func__, mPrimaryOutput, rxDevices, force, 0); } } std::map outputsToReopen; // reevaluate routing on all outputs in case tracks have been started during the call for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/); if (state != AUDIO_MODE_NORMAL && oldState == AUDIO_MODE_NORMAL && desc->mPreferredAttrInfo != nullptr) { // If the output is using preferred mixer attributes and the audio mode is not normal, // the output need to reopen with default configuration. outputsToReopen.emplace(mOutputs.keyAt(i), newDevices); continue; } if (state != AUDIO_MODE_IN_CALL || (desc != mPrimaryOutput && !isTelephonyRxOrTx(desc))) { bool forceRouting = !newDevices.isEmpty(); setOutputDevices(__func__, desc, newDevices, forceRouting, 0 /*delayMs*/, nullptr, true /*requiresMuteCheck*/, !forceRouting /*requiresVolumeCheck*/); } } reopenOutputsWithDevices(outputsToReopen); checkLeBroadcastRoutes(wasLeUnicastActive, nullptr, delayMs); if (isStateInCall(state)) { ALOGV("setPhoneState() in call state management: new state is %d", state); // force reevaluating accessibility routing when call starts invalidateStreams({AUDIO_STREAM_ACCESSIBILITY}); } // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE mLimitRingtoneVolume = (state == AUDIO_MODE_RINGTONE && isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)); } audio_mode_t AudioPolicyManager::getPhoneState() { return mEngine->getPhoneState(); } void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) { ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState()); if (config == mEngine->getForceUse(usage)) { return; } if (mEngine->setForceUse(usage, config) != NO_ERROR) { ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage); return; } bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) || (usage == AUDIO_POLICY_FORCE_FOR_DOCK) || (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM); // check for device and output changes triggered by new force usage checkForDeviceAndOutputChanges(); // force client reconnection to reevaluate flag AUDIO_FLAG_AUDIBILITY_ENFORCED if (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM) { invalidateStreams({AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE}); } //FIXME: workaround for truncated touch sounds // to be removed when the problem is handled by system UI uint32_t delayMs = 0; if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) { delayMs = TOUCH_SOUND_FIXED_DELAY_MS; } updateCallAndOutputRouting(forceVolumeReeval, delayMs); updateInputRouting(); } void AudioPolicyManager::setSystemProperty(const char* property, const char* value) { ALOGV("setSystemProperty() property %s, value %s", property, value); } // Find an MSD output profile compatible with the parameters passed. // When "directOnly" is set, restrict search to profiles for direct outputs. sp AudioPolicyManager::getMsdProfileForOutput( const DeviceVector& devices, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, bool directOnly) { flags = getRelevantFlags(flags, directOnly); sp msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD); if (msdModule != nullptr) { // for the msd module check if there are patches to the output devices if (msdHasPatchesToAllDevices(devices.toTypeAddrVector())) { HwModuleCollection modules; modules.add(msdModule); return searchCompatibleProfileHwModules( modules, getMsdAudioOutDevices(), samplingRate, format, channelMask, flags, directOnly); } } return nullptr; } // Find an output profile compatible with the parameters passed. When "directOnly" is set, restrict // search to profiles for direct outputs. sp AudioPolicyManager::getProfileForOutput( const DeviceVector& devices, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, bool directOnly) { flags = getRelevantFlags(flags, directOnly); return searchCompatibleProfileHwModules( mHwModules, devices, samplingRate, format, channelMask, flags, directOnly); } audio_output_flags_t AudioPolicyManager::getRelevantFlags ( audio_output_flags_t flags, bool directOnly) { if (directOnly) { // only retain flags that will drive the direct output profile selection // if explicitly requested static const uint32_t kRelevantFlags = (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ); flags = (audio_output_flags_t)((flags & kRelevantFlags) | AUDIO_OUTPUT_FLAG_DIRECT); } return flags; } sp AudioPolicyManager::searchCompatibleProfileHwModules ( const HwModuleCollection& hwModules, const DeviceVector& devices, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, bool directOnly) { sp profile; for (const auto& hwModule : hwModules) { for (const auto& curProfile : hwModule->getOutputProfiles()) { if (curProfile->getCompatibilityScore(devices, samplingRate, NULL /*updatedSamplingRate*/, format, NULL /*updatedFormat*/, channelMask, NULL /*updatedChannelMask*/, flags) == IOProfile::NO_MATCH) { continue; } // reject profiles not corresponding to a device currently available if (!mAvailableOutputDevices.containsAtLeastOne(curProfile->getSupportedDevices())) { continue; } // reject profiles if connected device does not support codec if (!curProfile->devicesSupportEncodedFormats(devices.types())) { continue; } if (!directOnly) { return curProfile; } // when searching for direct outputs, if several profiles are compatible, give priority // to one with offload capability if (profile != 0 && ((curProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0)) { continue; } profile = curProfile; if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { break; } } } return profile; } sp AudioPolicyManager::getSpatializerOutputProfile( const audio_config_t *config __unused, const AudioDeviceTypeAddrVector &devices) const { for (const auto& hwModule : mHwModules) { for (const auto& curProfile : hwModule->getOutputProfiles()) { if (curProfile->getFlags() != AUDIO_OUTPUT_FLAG_SPATIALIZER) { continue; } if (!devices.empty()) { // reject profiles not corresponding to a device currently available DeviceVector supportedDevices = curProfile->getSupportedDevices(); if (!mAvailableOutputDevices.containsAtLeastOne(supportedDevices)) { continue; } if (supportedDevices.getDevicesFromDeviceTypeAddrVec(devices).size() != devices.size()) { continue; } } ALOGV("%s found profile %s", __func__, curProfile->getName().c_str()); return curProfile; } } return nullptr; } audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream) { DeviceVector devices = mEngine->getOutputDevicesForStream(stream, false /*fromCache*/); // Note that related method getOutputForAttr() uses getOutputForDevice() not selectOutput(). // We use selectOutput() here since we don't have the desired AudioTrack sample rate, // format, flags, etc. This may result in some discrepancy for functions that utilize // getOutput() solely on audio_stream_type such as AudioSystem::getOutputFrameCount() // and AudioSystem::getOutputSamplingRate(). SortedVector outputs = getOutputsForDevices(devices, mOutputs); audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE; if (stream == AUDIO_STREAM_MUSIC && mConfig->useDeepBufferForMedia()) { flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; } const audio_io_handle_t output = selectOutput(outputs, flags); ALOGV("getOutput() stream %d selected devices %s, output %d", stream, devices.toString().c_str(), output); return output; } status_t AudioPolicyManager::getAudioAttributes(audio_attributes_t *dstAttr, const audio_attributes_t *srcAttr, audio_stream_type_t srcStream) { if (srcAttr != NULL) { if (!isValidAttributes(srcAttr)) { ALOGE("%s invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", __func__, srcAttr->usage, srcAttr->content_type, srcAttr->flags, srcAttr->tags); return BAD_VALUE; } *dstAttr = *srcAttr; } else { if (srcStream < AUDIO_STREAM_MIN || srcStream >= AUDIO_STREAM_PUBLIC_CNT) { ALOGE("%s: invalid stream type", __func__); return BAD_VALUE; } *dstAttr = mEngine->getAttributesForStreamType(srcStream); } // Only honor audibility enforced when required. The client will be // forced to reconnect if the forced usage changes. if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { dstAttr->flags = static_cast( dstAttr->flags & ~AUDIO_FLAG_AUDIBILITY_ENFORCED); } return NO_ERROR; } status_t AudioPolicyManager::getOutputForAttrInt( audio_attributes_t *resultAttr, audio_io_handle_t *output, audio_session_t session, const audio_attributes_t *attr, audio_stream_type_t *stream, uid_t uid, audio_config_t *config, audio_output_flags_t *flags, audio_port_handle_t *selectedDeviceId, bool *isRequestedDeviceForExclusiveUse, std::vector> *secondaryMixes, output_type_t *outputType, bool *isSpatialized, bool *isBitPerfect) { DeviceVector outputDevices; const audio_port_handle_t requestedPortId = *selectedDeviceId; DeviceVector msdDevices = getMsdAudioOutDevices(); const sp requestedDevice = mAvailableOutputDevices.getDeviceFromId(requestedPortId); *outputType = API_OUTPUT_INVALID; *isSpatialized = false; status_t status = getAudioAttributes(resultAttr, attr, *stream); if (status != NO_ERROR) { return status; } if (auto it = mAllowedCapturePolicies.find(uid); it != end(mAllowedCapturePolicies)) { resultAttr->flags = static_cast(resultAttr->flags | it->second); } *stream = mEngine->getStreamTypeForAttributes(*resultAttr); ALOGV("%s() attributes=%s stream=%s session %d selectedDeviceId %d", __func__, toString(*resultAttr).c_str(), toString(*stream).c_str(), session, requestedPortId); bool usePrimaryOutputFromPolicyMixes = false; // The primary output is the explicit routing (eg. setPreferredDevice) if specified, // otherwise, fallback to the dynamic policies, if none match, query the engine. // Secondary outputs are always found by dynamic policies as the engine do not support them sp primaryMix; const audio_config_base_t clientConfig = {.sample_rate = config->sample_rate, .channel_mask = config->channel_mask, .format = config->format, }; status = mPolicyMixes.getOutputForAttr(*resultAttr, clientConfig, uid, session, *flags, mAvailableOutputDevices, requestedDevice, primaryMix, secondaryMixes, usePrimaryOutputFromPolicyMixes); if (status != OK) { return status; } // FIXME: in case of RENDER policy, the output capabilities should be checked if ((secondaryMixes != nullptr && !secondaryMixes->empty()) && (!audio_is_linear_pcm(config->format) || *flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) { ALOGD("%s: rejecting request as secondary mixes only support pcm", __func__); return BAD_VALUE; } if (usePrimaryOutputFromPolicyMixes) { sp policyMixDevice = mAvailableOutputDevices.getDevice(primaryMix->mDeviceType, primaryMix->mDeviceAddress, AUDIO_FORMAT_DEFAULT); sp policyDesc = primaryMix->getOutput(); bool tryDirectForFlags = policyDesc == nullptr || (policyDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) || (*flags & (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)); // if a direct output can be opened to deliver the track's multi-channel content to the // output rather than being downmixed by the primary output, then use this direct // output by by-passing the primary mix if possible, otherwise fall-through to primary // mix. bool tryDirectForChannelMask = policyDesc != nullptr && (audio_channel_count_from_out_mask(policyDesc->getConfig().channel_mask) < audio_channel_count_from_out_mask(config->channel_mask)); if (policyMixDevice != nullptr && (tryDirectForFlags || tryDirectForChannelMask)) { audio_io_handle_t newOutput; status = openDirectOutput( *stream, session, config, (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT), DeviceVector(policyMixDevice), &newOutput, *resultAttr); if (status == NO_ERROR) { policyDesc = mOutputs.valueFor(newOutput); primaryMix->setOutput(policyDesc); } else if (tryDirectForFlags) { ALOGW("%s, failed open direct, status: %d", __func__, status); policyDesc = nullptr; } // otherwise use primary if available. } if (policyDesc != nullptr) { policyDesc->mPolicyMix = primaryMix; *output = policyDesc->mIoHandle; *selectedDeviceId = policyMixDevice != nullptr ? policyMixDevice->getId() : AUDIO_PORT_HANDLE_NONE; if ((policyDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != AUDIO_OUTPUT_FLAG_DIRECT) { // Remove direct flag as it is not on a direct output. *flags = (audio_output_flags_t) (*flags & ~AUDIO_OUTPUT_FLAG_DIRECT); } ALOGV("getOutputForAttr() returns output %d", *output); if (resultAttr->usage == AUDIO_USAGE_VIRTUAL_SOURCE) { *outputType = API_OUT_MIX_PLAYBACK; } else { *outputType = API_OUTPUT_LEGACY; } return NO_ERROR; } else { if (policyMixDevice != nullptr) { ALOGE("%s, try to use primary mix but no output found", __func__); return INVALID_OPERATION; } // Fallback to default engine selection as the selected primary mix device is not // available. } } // Virtual sources must always be dynamicaly or explicitly routed if (resultAttr->usage == AUDIO_USAGE_VIRTUAL_SOURCE) { ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE"); return BAD_VALUE; } // explicit routing managed by getDeviceForStrategy in APM is now handled by engine // in order to let the choice of the order to future vendor engine outputDevices = mEngine->getOutputDevicesForAttributes(*resultAttr, requestedDevice, false); if ((resultAttr->flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } // Set incall music only if device was explicitly set, and fallback to the device which is // chosen by the engine if not. // FIXME: provide a more generic approach which is not device specific and move this back // to getOutputForDevice. // TODO: Remove check of AUDIO_STREAM_MUSIC once migration is completed on the app side. if (outputDevices.onlyContainsDevicesWithType(AUDIO_DEVICE_OUT_TELEPHONY_TX) && (*stream == AUDIO_STREAM_MUSIC || resultAttr->usage == AUDIO_USAGE_VOICE_COMMUNICATION) && audio_is_linear_pcm(config->format) && isCallAudioAccessible()) { if (requestedPortId != AUDIO_PORT_HANDLE_NONE) { *flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC; *isRequestedDeviceForExclusiveUse = true; } } ALOGV("%s() device %s, sampling rate %d, format %#x, channel mask %#x, flags %#x stream %s", __func__, outputDevices.toString().c_str(), config->sample_rate, config->format, config->channel_mask, *flags, toString(*stream).c_str()); *output = AUDIO_IO_HANDLE_NONE; if (!msdDevices.isEmpty()) { *output = getOutputForDevices(msdDevices, session, resultAttr, config, flags, isSpatialized); if (*output != AUDIO_IO_HANDLE_NONE && setMsdOutputPatches(&outputDevices) == NO_ERROR) { ALOGV("%s() Using MSD devices %s instead of devices %s", __func__, msdDevices.toString().c_str(), outputDevices.toString().c_str()); } else { *output = AUDIO_IO_HANDLE_NONE; } } if (*output == AUDIO_IO_HANDLE_NONE) { sp info = nullptr; if (outputDevices.size() == 1) { info = getPreferredMixerAttributesInfo( outputDevices.itemAt(0)->getId(), mEngine->getProductStrategyForAttributes(*resultAttr), true /*activeBitPerfectPreferred*/); // Only use preferred mixer if the uid matches or the preferred mixer is bit-perfect // and it is currently active. if (info != nullptr && info->getUid() != uid && (!info->isBitPerfect() || info->getActiveClientCount() == 0)) { info = nullptr; } if (com::android::media::audioserver:: fix_concurrent_playback_behavior_with_bit_perfect_client()) { if (info != nullptr && info->getUid() == uid && info->configMatches(*config) && (mEngine->getPhoneState() != AUDIO_MODE_NORMAL || std::any_of(gHighPriorityUseCases.begin(), gHighPriorityUseCases.end(), [this, &outputDevices](audio_usage_t usage) { return mOutputs.isUsageActiveOnDevice( usage, outputDevices[0]); }))) { // Bit-perfect request is not allowed when the phone mode is not normal or // there is any higher priority user case active. return INVALID_OPERATION; } } } *output = getOutputForDevices(outputDevices, session, resultAttr, config, flags, isSpatialized, info, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC); // The client will be active if the client is currently preferred mixer owner and the // requested configuration matches the preferred mixer configuration. *isBitPerfect = (info != nullptr && info->isBitPerfect() && info->getUid() == uid && *output != AUDIO_IO_HANDLE_NONE // When bit-perfect output is selected for the preferred mixer attributes owner, // only need to consider the config matches. && mOutputs.valueFor(*output)->isConfigurationMatched( clientConfig, AUDIO_OUTPUT_FLAG_NONE)); if (*isBitPerfect) { *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_BIT_PERFECT); } } if (*output == AUDIO_IO_HANDLE_NONE) { AudioProfileVector profiles; status_t ret = getProfilesForDevices(outputDevices, profiles, *flags, false /*isInput*/); if (ret == NO_ERROR && !profiles.empty()) { const auto channels = profiles[0]->getChannels(); if (!channels.empty() && (channels.find(config->channel_mask) == channels.end())) { config->channel_mask = *channels.begin(); } const auto sampleRates = profiles[0]->getSampleRates(); if (!sampleRates.empty() && (sampleRates.find(config->sample_rate) == sampleRates.end())) { config->sample_rate = *sampleRates.begin(); } config->format = profiles[0]->getFormat(); } return INVALID_OPERATION; } *selectedDeviceId = getFirstDeviceId(outputDevices); for (auto &outputDevice : outputDevices) { if (outputDevice->getId() == mConfig->getDefaultOutputDevice()->getId()) { *selectedDeviceId = outputDevice->getId(); break; } } if (outputDevices.onlyContainsDevicesWithType(AUDIO_DEVICE_OUT_TELEPHONY_TX)) { *outputType = API_OUTPUT_TELEPHONY_TX; } else { *outputType = API_OUTPUT_LEGACY; } ALOGV("%s returns output %d selectedDeviceId %d", __func__, *output, *selectedDeviceId); return NO_ERROR; } status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, audio_io_handle_t *output, audio_session_t session, audio_stream_type_t *stream, const AttributionSourceState& attributionSource, audio_config_t *config, audio_output_flags_t *flags, audio_port_handle_t *selectedDeviceId, audio_port_handle_t *portId, std::vector *secondaryOutputs, output_type_t *outputType, bool *isSpatialized, bool *isBitPerfect) { // The supplied portId must be AUDIO_PORT_HANDLE_NONE if (*portId != AUDIO_PORT_HANDLE_NONE) { return INVALID_OPERATION; } const uid_t uid = VALUE_OR_RETURN_STATUS( aidl2legacy_int32_t_uid_t(attributionSource.uid)); const audio_port_handle_t requestedPortId = *selectedDeviceId; audio_attributes_t resultAttr; bool isRequestedDeviceForExclusiveUse = false; std::vector> secondaryMixes; const sp requestedDevice = mAvailableOutputDevices.getDeviceFromId(requestedPortId); // Prevent from storing invalid requested device id in clients const audio_port_handle_t sanitizedRequestedPortId = requestedDevice != nullptr ? requestedPortId : AUDIO_PORT_HANDLE_NONE; *selectedDeviceId = sanitizedRequestedPortId; status_t status = getOutputForAttrInt(&resultAttr, output, session, attr, stream, uid, config, flags, selectedDeviceId, &isRequestedDeviceForExclusiveUse, secondaryOutputs != nullptr ? &secondaryMixes : nullptr, outputType, isSpatialized, isBitPerfect); if (status != NO_ERROR) { return status; } std::vector> weakSecondaryOutputDescs; if (secondaryOutputs != nullptr) { for (auto &secondaryMix : secondaryMixes) { sp outputDesc = secondaryMix->getOutput(); if (outputDesc != nullptr && outputDesc->mIoHandle != AUDIO_IO_HANDLE_NONE && outputDesc->mIoHandle != *output) { secondaryOutputs->push_back(outputDesc->mIoHandle); weakSecondaryOutputDescs.push_back(outputDesc); } } } audio_config_base_t clientConfig = {.sample_rate = config->sample_rate, .channel_mask = config->channel_mask, .format = config->format, }; *portId = PolicyAudioPort::getNextUniqueId(); sp outputDesc = mOutputs.valueFor(*output); sp clientDesc = new TrackClientDescriptor(*portId, uid, session, resultAttr, clientConfig, sanitizedRequestedPortId, *stream, mEngine->getProductStrategyForAttributes(resultAttr), toVolumeSource(resultAttr), *flags, isRequestedDeviceForExclusiveUse, std::move(weakSecondaryOutputDescs), outputDesc->mPolicyMix); outputDesc->addClient(clientDesc); ALOGV("%s() returns output %d requestedPortId %d selectedDeviceId %d for port ID %d", __func__, *output, requestedPortId, *selectedDeviceId, *portId); return NO_ERROR; } status_t AudioPolicyManager::openDirectOutput(audio_stream_type_t stream, audio_session_t session, const audio_config_t *config, audio_output_flags_t flags, const DeviceVector &devices, audio_io_handle_t *output, audio_attributes_t attributes) { *output = AUDIO_IO_HANDLE_NONE; // skip direct output selection if the request can obviously be attached to a mixed output // and not explicitly requested if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) && audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX && audio_channel_count_from_out_mask(config->channel_mask) <= 2) { return NAME_NOT_FOUND; } // Reject flag combinations that do not make sense. Note that the requested flags might not // have the 'DIRECT' flag set, however once a direct-capable profile is found, it will // combine the requested flags with its own flags, yielding an unsupported combination. if ((flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { return NAME_NOT_FOUND; } // Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled. // This prevents creating an offloaded track and tearing it down immediately after start // when audioflinger detects there is an active non offloadable effect. // FIXME: We should check the audio session here but we do not have it in this context. // This may prevent offloading in rare situations where effects are left active by apps // in the background. sp profile; if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || !(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) { profile = getProfileForOutput( devices, config->sample_rate, config->format, config->channel_mask, flags, true /* directOnly */); } if (profile == nullptr) { return NAME_NOT_FOUND; } // exclusive outputs for MMAP and Offload are enforced by different session ids. for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (!desc->isDuplicated() && (profile == desc->mProfile)) { // reuse direct output if currently open by the same client // and configured with same parameters if ((config->sample_rate == desc->getSamplingRate()) && (config->format == desc->getFormat()) && (config->channel_mask == desc->getChannelMask()) && (session == desc->mDirectClientSession)) { desc->mDirectOpenCount++; ALOGI("%s reusing direct output %d for session %d", __func__, mOutputs.keyAt(i), session); *output = mOutputs.keyAt(i); return NO_ERROR; } } } if (!profile->canOpenNewIo()) { if (!com::android::media::audioserver::direct_track_reprioritization()) { ALOGW("%s profile %s can't open new output maxOpenCount reached", __func__, profile->getName().c_str()); return NAME_NOT_FOUND; } else if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) != 0) { // MMAP gracefully handles lack of an exclusive track resource by mixing // above the audio framework. For AAudio to know that the limit is reached, // return an error. ALOGW("%s profile %s can't open new mmap output maxOpenCount reached", __func__, profile->getName().c_str()); return NAME_NOT_FOUND; } else { // Close outputs on this profile, if available, to free resources for this request for (int i = 0; i < mOutputs.size() && !profile->canOpenNewIo(); i++) { const auto desc = mOutputs.valueAt(i); if (desc->mProfile == profile) { ALOGV("%s closeOutput %d to prioritize session %d on profile %s", __func__, desc->mIoHandle, session, profile->getName().c_str()); closeOutput(desc->mIoHandle); } } } } // Unable to close streams to find free resources for this request if (!profile->canOpenNewIo()) { ALOGW("%s profile %s can't open new output maxOpenCount reached", __func__, profile->getName().c_str()); return NAME_NOT_FOUND; } auto outputDesc = sp::make(profile, mpClientInterface); // An MSD patch may be using the only output stream that can service this request. Release // all MSD patches to prioritize this request over any active output on MSD. releaseMsdOutputPatches(devices); status_t status = outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, &flags, output, attributes); // only accept an output with the requested parameters, unless the format can be IEC61937 // encapsulated and opened by AudioFlinger as wrapped IEC61937. const bool ignoreRequestedParametersCheck = audio_is_iec61937_compatible(config->format) && (flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO) && audio_has_proportional_frames(outputDesc->getFormat()); if (status != NO_ERROR || (!ignoreRequestedParametersCheck && ((config->sample_rate != 0 && config->sample_rate != outputDesc->getSamplingRate()) || (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->getFormat()) || (config->channel_mask != 0 && config->channel_mask != outputDesc->getChannelMask())))) { ALOGV("%s failed opening direct output: output %d sample rate %d %d," "format %d %d, channel mask %04x %04x", __func__, *output, config->sample_rate, outputDesc->getSamplingRate(), config->format, outputDesc->getFormat(), config->channel_mask, outputDesc->getChannelMask()); if (*output != AUDIO_IO_HANDLE_NONE) { outputDesc->close(); } // fall back to mixer output if possible when the direct output could not be open if (audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX) { return NAME_NOT_FOUND; } *output = AUDIO_IO_HANDLE_NONE; return BAD_VALUE; } outputDesc->mDirectOpenCount = 1; outputDesc->mDirectClientSession = session; addOutput(*output, outputDesc); // The version check is essentially to avoid making this call in the case of the HIDL HAL. if (auto hwModule = mHwModules.getModuleFromHandle(mPrimaryModuleHandle); hwModule && hwModule->getHalVersionMajor() >= 3) { setOutputDevices(__func__, outputDesc, devices, true, 0, NULL); } mPreviousOutputs = mOutputs; ALOGV("%s returns new direct output %d", __func__, *output); mpClientInterface->onAudioPortListUpdate(); return NO_ERROR; } audio_io_handle_t AudioPolicyManager::getOutputForDevices( const DeviceVector &devices, audio_session_t session, const audio_attributes_t *attr, const audio_config_t *config, audio_output_flags_t *flags, bool *isSpatialized, sp prefMixerConfigInfo, bool forceMutingHaptic) { audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; // Discard haptic channel mask when forcing muting haptic channels. audio_channel_mask_t channelMask = forceMutingHaptic ? static_cast(config->channel_mask & ~AUDIO_CHANNEL_HAPTIC_ALL) : config->channel_mask; // open a direct output if required by specified parameters //force direct flag if offload flag is set: offloading implies a direct output stream // and all common behaviors are driven by checking only the direct flag // this should normally be set appropriately in the policy configuration file if ((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT); } if ((*flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT); } audio_stream_type_t stream = mEngine->getStreamTypeForAttributes(*attr); // only allow deep buffering for music stream type if (stream != AUDIO_STREAM_MUSIC) { *flags = (audio_output_flags_t)(*flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); } else if (/* stream == AUDIO_STREAM_MUSIC && */ *flags == AUDIO_OUTPUT_FLAG_NONE && mConfig->useDeepBufferForMedia()) { // use DEEP_BUFFER as default output for music stream type *flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_DEEP_BUFFER; } if (stream == AUDIO_STREAM_TTS) { *flags = AUDIO_OUTPUT_FLAG_TTS; } else if (stream == AUDIO_STREAM_VOICE_CALL && audio_is_linear_pcm(config->format) && (*flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) == 0) { *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT); ALOGV("Set VoIP and Direct output flags for PCM format"); } // Attach the Ultrasound flag for the AUDIO_CONTENT_TYPE_ULTRASOUND if (attr->content_type == AUDIO_CONTENT_TYPE_ULTRASOUND) { *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_ULTRASOUND); } // Use the spatializer output if the content can be spatialized, no preferred mixer // was specified and offload or direct playback is not explicitly requested, and there is no // haptic channel included in playback *isSpatialized = false; if (mSpatializerOutput != nullptr && canBeSpatializedInt(attr, config, devices.toTypeAddrVector()) && prefMixerConfigInfo == nullptr && ((*flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT)) == 0) && checkHapticCompatibilityOnSpatializerOutput(config, session)) { *isSpatialized = true; return mSpatializerOutput->mIoHandle; } audio_config_t directConfig = *config; directConfig.channel_mask = channelMask; status_t status = openDirectOutput(stream, session, &directConfig, *flags, devices, &output, *attr); if (status != NAME_NOT_FOUND) { return output; } // A request for HW A/V sync cannot fallback to a mixed output because time // stamps are embedded in audio data if ((*flags & (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)) != 0) { return AUDIO_IO_HANDLE_NONE; } // A request for Tuner cannot fallback to a mixed output if ((directConfig.offload_info.content_id || directConfig.offload_info.sync_id)) { return AUDIO_IO_HANDLE_NONE; } // ignoring channel mask due to downmix capability in mixer // open a non direct output // for non direct outputs, only PCM is supported if (audio_is_linear_pcm(config->format)) { // get which output is suitable for the specified stream. The actual // routing change will happen when startOutput() will be called SortedVector outputs = getOutputsForDevices(devices, mOutputs); if (prefMixerConfigInfo != nullptr) { for (audio_io_handle_t outputHandle : outputs) { sp outputDesc = mOutputs.valueFor(outputHandle); if (outputDesc->mProfile == prefMixerConfigInfo->getProfile()) { output = outputHandle; break; } } if (output == AUDIO_IO_HANDLE_NONE) { // No output open with the preferred profile. Open a new one. audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.channel_mask = prefMixerConfigInfo->getConfigBase().channel_mask; config.sample_rate = prefMixerConfigInfo->getConfigBase().sample_rate; config.format = prefMixerConfigInfo->getConfigBase().format; sp preferredOutput = openOutputWithProfileAndDevice( prefMixerConfigInfo->getProfile(), devices, nullptr /*mixerConfig*/, &config, prefMixerConfigInfo->getFlags()); if (preferredOutput == nullptr) { ALOGE("%s failed to open output with preferred mixer config", __func__); } else { output = preferredOutput->mIoHandle; } } } else { // at this stage we should ignore the DIRECT flag as no direct output could be // found earlier *flags = (audio_output_flags_t) (*flags & ~AUDIO_OUTPUT_FLAG_DIRECT); if (com::android::media::audioserver:: fix_concurrent_playback_behavior_with_bit_perfect_client()) { // If the preferred mixer attributes is null, do not select the bit-perfect output // unless the bit-perfect output is the only output. // The bit-perfect output can exist while the passed in preferred mixer attributes // info is null when it is a high priority client. The high priority clients are // ringtone or alarm, which is not a bit-perfect use case. size_t i = 0; while (i < outputs.size() && outputs.size() > 1) { auto desc = mOutputs.valueFor(outputs[i]); // The output descriptor must not be null here. if (desc->isBitPerfect()) { outputs.removeItemsAt(i); } else { i += 1; } } } output = selectOutput( outputs, *flags, config->format, channelMask, config->sample_rate, session); } } ALOGW_IF((output == 0), "getOutputForDevices() could not find output for stream %d, " "sampling rate %d, format %#x, channels %#x, flags %#x", stream, config->sample_rate, config->format, channelMask, *flags); return output; } sp AudioPolicyManager::getMsdAudioInDevice() const { auto msdInDevices = mHwModules.getAvailableDevicesFromModuleName(AUDIO_HARDWARE_MODULE_ID_MSD, mAvailableInputDevices); return msdInDevices.isEmpty()? nullptr : msdInDevices.itemAt(0); } DeviceVector AudioPolicyManager::getMsdAudioOutDevices() const { return mHwModules.getAvailableDevicesFromModuleName(AUDIO_HARDWARE_MODULE_ID_MSD, mAvailableOutputDevices); } const AudioPatchCollection AudioPolicyManager::getMsdOutputPatches() const { AudioPatchCollection msdPatches; sp msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD); if (msdModule != 0) { for (size_t i = 0; i < mAudioPatches.size(); ++i) { sp patch = mAudioPatches.valueAt(i); for (size_t j = 0; j < patch->mPatch.num_sources; ++j) { const struct audio_port_config *source = &patch->mPatch.sources[j]; if (source->type == AUDIO_PORT_TYPE_DEVICE && source->ext.device.hw_module == msdModule->getHandle()) { msdPatches.addAudioPatch(patch->getHandle(), patch); } } } } return msdPatches; } bool AudioPolicyManager::isMsdPatch(const audio_patch_handle_t &handle) const { ssize_t index = mAudioPatches.indexOfKey(handle); if (index < 0) { return false; } const sp patch = mAudioPatches.valueAt(index); sp msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD); if (msdModule == nullptr) { return false; } const struct audio_port_config *sink = &patch->mPatch.sinks[0]; if (getMsdAudioOutDevices().contains(mAvailableOutputDevices.getDeviceFromId(sink->id))) { return true; } index = getMsdOutputPatches().indexOfKey(handle); if (index < 0) { return false; } return true; } status_t AudioPolicyManager::getMsdProfiles(bool hwAvSync, const InputProfileCollection &inputProfiles, const OutputProfileCollection &outputProfiles, const sp &sourceDevice, const sp &sinkDevice, AudioProfileVector& sourceProfiles, AudioProfileVector& sinkProfiles) const { if (inputProfiles.isEmpty()) { ALOGE("%s() no input profiles for source module", __func__); return NO_INIT; } if (outputProfiles.isEmpty()) { ALOGE("%s() no output profiles for sink module", __func__); return NO_INIT; } for (const auto &inProfile : inputProfiles) { if (hwAvSync == ((inProfile->getFlags() & AUDIO_INPUT_FLAG_HW_AV_SYNC) != 0) && inProfile->supportsDevice(sourceDevice)) { appendAudioProfiles(sourceProfiles, inProfile->getAudioProfiles()); } } for (const auto &outProfile : outputProfiles) { if (hwAvSync == ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) && outProfile->supportsDevice(sinkDevice)) { appendAudioProfiles(sinkProfiles, outProfile->getAudioProfiles()); } } return NO_ERROR; } status_t AudioPolicyManager::getBestMsdConfig(bool hwAvSync, const AudioProfileVector &sourceProfiles, const AudioProfileVector &sinkProfiles, audio_port_config *sourceConfig, audio_port_config *sinkConfig) const { // Compressed formats for MSD module, ordered from most preferred to least preferred. static const std::vector formatsOrder = {{ AUDIO_FORMAT_IEC60958, AUDIO_FORMAT_MAT_2_1, AUDIO_FORMAT_MAT_2_0, AUDIO_FORMAT_E_AC3, AUDIO_FORMAT_AC3, AUDIO_FORMAT_PCM_FLOAT, AUDIO_FORMAT_PCM_32_BIT, AUDIO_FORMAT_PCM_8_24_BIT, AUDIO_FORMAT_PCM_24_BIT_PACKED, AUDIO_FORMAT_PCM_16_BIT }}; static const std::vector channelMasksOrder = [](){ // Channel position masks for MSD module, 3D > 2D > 1D ordering (most preferred to least // preferred). std::vector masks = {{ AUDIO_CHANNEL_OUT_3POINT1POINT2, AUDIO_CHANNEL_OUT_3POINT0POINT2, AUDIO_CHANNEL_OUT_2POINT1POINT2, AUDIO_CHANNEL_OUT_2POINT0POINT2, AUDIO_CHANNEL_OUT_5POINT1, AUDIO_CHANNEL_OUT_STEREO }}; // insert index masks (higher counts most preferred) as preferred over position masks for (int i = 1; i <= AUDIO_CHANNEL_COUNT_MAX; i++) { masks.insert( masks.begin(), audio_channel_mask_for_index_assignment_from_count(i)); } return masks; }(); struct audio_config_base bestSinkConfig; status_t result = findBestMatchingOutputConfig(sourceProfiles, sinkProfiles, formatsOrder, channelMasksOrder, true /*preferHigherSamplingRates*/, bestSinkConfig); if (result != NO_ERROR) { ALOGD("%s() no matching config found for sink, hwAvSync: %d", __func__, hwAvSync); return result; } sinkConfig->sample_rate = bestSinkConfig.sample_rate; sinkConfig->channel_mask = bestSinkConfig.channel_mask; sinkConfig->format = bestSinkConfig.format; // For encoded streams force direct flag to prevent downstream mixing. sinkConfig->flags.output = static_cast( sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_DIRECT); if (audio_is_iec61937_compatible(sinkConfig->format)) { // For formats compatible with IEC61937 encapsulation, assume that // the input is IEC61937 framed (for proportional buffer sizing). // Add the AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO flag so downstream HAL can distinguish between // raw and IEC61937 framed streams. sinkConfig->flags.output = static_cast( sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO); } sourceConfig->sample_rate = bestSinkConfig.sample_rate; // Specify exact channel mask to prevent guessing by bit count in PatchPanel. sourceConfig->channel_mask = audio_channel_mask_get_representation(bestSinkConfig.channel_mask) == AUDIO_CHANNEL_REPRESENTATION_INDEX ? bestSinkConfig.channel_mask : audio_channel_mask_out_to_in(bestSinkConfig.channel_mask); sourceConfig->format = bestSinkConfig.format; // Copy input stream directly without any processing (e.g. resampling). sourceConfig->flags.input = static_cast( sourceConfig->flags.input | AUDIO_INPUT_FLAG_DIRECT); if (hwAvSync) { sinkConfig->flags.output = static_cast( sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); sourceConfig->flags.input = static_cast( sourceConfig->flags.input | AUDIO_INPUT_FLAG_HW_AV_SYNC); } const unsigned int config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE | AUDIO_PORT_CONFIG_CHANNEL_MASK | AUDIO_PORT_CONFIG_FORMAT | AUDIO_PORT_CONFIG_FLAGS; sinkConfig->config_mask |= config_mask; sourceConfig->config_mask |= config_mask; return NO_ERROR; } PatchBuilder AudioPolicyManager::buildMsdPatch(bool msdIsSource, const sp &device) const { PatchBuilder patchBuilder; sp msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD); ALOG_ASSERT(msdModule != nullptr, "MSD module not available"); sp deviceModule = mHwModules.getModuleForDevice(device, AUDIO_FORMAT_DEFAULT); if (deviceModule == nullptr) { ALOGE("%s() unable to get module for %s", __func__, device->toString().c_str()); return patchBuilder; } const InputProfileCollection inputProfiles = msdIsSource ? msdModule->getInputProfiles() : deviceModule->getInputProfiles(); const OutputProfileCollection outputProfiles = msdIsSource ? deviceModule->getOutputProfiles() : msdModule->getOutputProfiles(); const sp sourceDevice = msdIsSource ? getMsdAudioInDevice() : device; const sp sinkDevice = msdIsSource ? device : getMsdAudioOutDevices().itemAt(0); patchBuilder.addSource(sourceDevice).addSink(sinkDevice); audio_port_config sourceConfig = patchBuilder.patch()->sources[0]; audio_port_config sinkConfig = patchBuilder.patch()->sinks[0]; AudioProfileVector sourceProfiles; AudioProfileVector sinkProfiles; // TODO: Figure out whether MSD module has HW_AV_SYNC flag set in the AP config file. // For now, we just forcefully try with HwAvSync first. for (auto hwAvSync : { true, false }) { if (getMsdProfiles(hwAvSync, inputProfiles, outputProfiles, sourceDevice, sinkDevice, sourceProfiles, sinkProfiles) != NO_ERROR) { continue; } if (getBestMsdConfig(hwAvSync, sourceProfiles, sinkProfiles, &sourceConfig, &sinkConfig) == NO_ERROR) { // Found a matching config. Re-create PatchBuilder with this config. return (PatchBuilder()).addSource(sourceConfig).addSink(sinkConfig); } } ALOGV("%s() no matching config found. Fall through to default PCM patch" " supporting PCM format conversion.", __func__); return patchBuilder; } status_t AudioPolicyManager::setMsdOutputPatches(const DeviceVector *outputDevices) { DeviceVector devices; if (outputDevices != nullptr && outputDevices->size() > 0) { devices.add(*outputDevices); } else { // Use media strategy for unspecified output device. This should only // occur on checkForDeviceAndOutputChanges(). Device connection events may // therefore invalidate explicit routing requests. devices = mEngine->getOutputDevicesForAttributes( attributes_initializer(AUDIO_USAGE_MEDIA), nullptr, false /*fromCache*/); LOG_ALWAYS_FATAL_IF(devices.isEmpty(), "no output device to set MSD patch"); } std::vector patchesToCreate; for (auto i = 0u; i < devices.size(); ++i) { ALOGV("%s() for device %s", __func__, devices[i]->toString().c_str()); patchesToCreate.push_back(buildMsdPatch(true /*msdIsSource*/, devices[i])); } // Retain only the MSD patches associated with outputDevices request. // Tear down the others, and create new ones as needed. AudioPatchCollection patchesToRemove = getMsdOutputPatches(); for (auto it = patchesToCreate.begin(); it != patchesToCreate.end(); ) { auto retainedPatch = false; for (auto i = 0u; i < patchesToRemove.size(); ++i) { if (audio_patches_are_equal(it->patch(), &patchesToRemove[i]->mPatch)) { patchesToRemove.removeItemsAt(i); retainedPatch = true; break; } } if (retainedPatch) { it = patchesToCreate.erase(it); continue; } ++it; } if (patchesToCreate.size() == 0 && patchesToRemove.size() == 0) { return NO_ERROR; } for (auto i = 0u; i < patchesToRemove.size(); ++i) { auto ¤tPatch = patchesToRemove.valueAt(i); releaseAudioPatch(currentPatch->getHandle(), mUidCached); } status_t status = NO_ERROR; for (const auto &p : patchesToCreate) { auto currStatus = installPatch(__func__, -1 /*index*/, nullptr /*patchHandle*/, p.patch(), 0 /*delayMs*/, mUidCached, nullptr /*patchDescPtr*/); char message[256]; snprintf(message, sizeof(message), "%s() %s: creating MSD patch from device:IN_BUS to " "device:%#x (format:%#x channels:%#x samplerate:%d)", __func__, currStatus == NO_ERROR ? "Success" : "Error", p.patch()->sinks[0].ext.device.type, p.patch()->sources[0].format, p.patch()->sources[0].channel_mask, p.patch()->sources[0].sample_rate); if (currStatus == NO_ERROR) { ALOGD("%s", message); } else { ALOGE("%s", message); if (status == NO_ERROR) { status = currStatus; } } } return status; } void AudioPolicyManager::releaseMsdOutputPatches(const DeviceVector& devices) { AudioPatchCollection msdPatches = getMsdOutputPatches(); for (size_t i = 0; i < msdPatches.size(); i++) { const auto& patch = msdPatches[i]; for (size_t j = 0; j < patch->mPatch.num_sinks; ++j) { const struct audio_port_config *sink = &patch->mPatch.sinks[j]; if (sink->type == AUDIO_PORT_TYPE_DEVICE && devices.getDevice(sink->ext.device.type, String8(sink->ext.device.address), AUDIO_FORMAT_DEFAULT) != nullptr) { releaseAudioPatch(patch->getHandle(), mUidCached); break; } } } } bool AudioPolicyManager::msdHasPatchesToAllDevices(const AudioDeviceTypeAddrVector& devices) { DeviceVector devicesToCheck = mConfig->getOutputDevices().getDevicesFromDeviceTypeAddrVec(devices); AudioPatchCollection msdPatches = getMsdOutputPatches(); for (size_t i = 0; i < msdPatches.size(); i++) { const auto& patch = msdPatches[i]; for (size_t j = 0; j < patch->mPatch.num_sinks; ++j) { const struct audio_port_config *sink = &patch->mPatch.sinks[j]; if (sink->type == AUDIO_PORT_TYPE_DEVICE) { const auto& foundDevice = devicesToCheck.getDevice( sink->ext.device.type, String8(sink->ext.device.address), AUDIO_FORMAT_DEFAULT); if (foundDevice != nullptr) { devicesToCheck.remove(foundDevice); if (devicesToCheck.isEmpty()) { return true; } } } } } return false; } audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector& outputs, audio_output_flags_t flags, audio_format_t format, audio_channel_mask_t channelMask, uint32_t samplingRate, audio_session_t sessionId) { LOG_ALWAYS_FATAL_IF(!(format == AUDIO_FORMAT_INVALID || audio_is_linear_pcm(format)), "%s called with format %#x", __func__, format); // Return the output that haptic-generating attached to when 1) session id is specified, // 2) haptic-generating effect exists for given session id and 3) the output that // haptic-generating effect attached to is in given outputs. if (sessionId != AUDIO_SESSION_NONE) { audio_io_handle_t hapticGeneratingOutput = mEffects.getIoForSession( sessionId, FX_IID_HAPTICGENERATOR); if (outputs.indexOf(hapticGeneratingOutput) >= 0) { return hapticGeneratingOutput; } } // Flags disqualifying an output: the match must happen before calling selectOutput() static const audio_output_flags_t kExcludedFlags = (audio_output_flags_t) (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT); // Flags expressing a functional request: must be honored in priority over // other criteria static const audio_output_flags_t kFunctionalFlags = (audio_output_flags_t) (AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_INCALL_MUSIC | AUDIO_OUTPUT_FLAG_TTS | AUDIO_OUTPUT_FLAG_DIRECT_PCM | AUDIO_OUTPUT_FLAG_ULTRASOUND | AUDIO_OUTPUT_FLAG_SPATIALIZER); // Flags expressing a performance request: have lower priority than serving // requested sampling rate or channel mask static const audio_output_flags_t kPerformanceFlags = (audio_output_flags_t) (AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_RAW | AUDIO_OUTPUT_FLAG_SYNC); const audio_output_flags_t functionalFlags = (audio_output_flags_t)(flags & kFunctionalFlags); const audio_output_flags_t performanceFlags = (audio_output_flags_t)(flags & kPerformanceFlags); audio_io_handle_t bestOutput = (outputs.size() == 0) ? AUDIO_IO_HANDLE_NONE : outputs[0]; // select one output among several that provide a path to a particular device or set of // devices (the list was previously build by getOutputsForDevices()). // The priority is as follows: // 1: the output supporting haptic playback when requesting haptic playback // 2: the output with the highest number of requested functional flags // with tiebreak preferring the minimum number of extra functional flags // (see b/200293124, the incorrect selection of AUDIO_OUTPUT_FLAG_VOIP_RX). // 3: the output supporting the exact channel mask // 4: the output with a higher channel count than requested // 5: the output with the highest sampling rate if the requested sample rate is // greater than default sampling rate // 6: the output with the highest number of requested performance flags // 7: the output with the bit depth the closest to the requested one // 8: the primary output // 9: the first output in the list // matching criteria values in priority order for best matching output so far std::vector bestMatchCriteria(8, 0); const bool hasOrphanHaptic = mEffects.hasOrphansForSession(sessionId, FX_IID_HAPTICGENERATOR); const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); const uint32_t hapticChannelCount = audio_channel_count_from_out_mask( channelMask & AUDIO_CHANNEL_HAPTIC_ALL); for (audio_io_handle_t output : outputs) { sp outputDesc = mOutputs.valueFor(output); // matching criteria values in priority order for current output std::vector currentMatchCriteria(8, 0); if (outputDesc->isDuplicated()) { continue; } if ((kExcludedFlags & outputDesc->mFlags) != 0) { continue; } // If haptic channel is specified, use the haptic output if present. // When using haptic output, same audio format and sample rate are required. const uint32_t outputHapticChannelCount = audio_channel_count_from_out_mask( outputDesc->getChannelMask() & AUDIO_CHANNEL_HAPTIC_ALL); // skip if haptic channel specified but output does not support it, or output support haptic // but there is no haptic channel requested AND no orphan haptic effect exist if ((hapticChannelCount != 0 && outputHapticChannelCount == 0) || (hapticChannelCount == 0 && outputHapticChannelCount != 0 && !hasOrphanHaptic)) { continue; } // In the case of audio-coupled-haptic playback, there is no format conversion and // resampling in the framework, same format/channel/sampleRate for client and the output // thread is required. In the case of HapticGenerator effect, do not require format // matching. if ((outputHapticChannelCount >= hapticChannelCount && format == outputDesc->getFormat() && samplingRate == outputDesc->getSamplingRate()) || (outputHapticChannelCount != 0 && hasOrphanHaptic)) { currentMatchCriteria[0] = outputHapticChannelCount; } // functional flags match const int matchingFunctionalFlags = __builtin_popcount(outputDesc->mFlags & functionalFlags); const int totalFunctionalFlags = __builtin_popcount(outputDesc->mFlags & kFunctionalFlags); // Prefer matching functional flags, but subtract unnecessary functional flags. currentMatchCriteria[1] = 100 * (matchingFunctionalFlags + 1) - totalFunctionalFlags; // channel mask and channel count match uint32_t outputChannelCount = audio_channel_count_from_out_mask( outputDesc->getChannelMask()); if (channelMask != AUDIO_CHANNEL_NONE && channelCount > 2 && channelCount <= outputChannelCount) { if ((audio_channel_mask_get_representation(channelMask) == audio_channel_mask_get_representation(outputDesc->getChannelMask())) && ((channelMask & outputDesc->getChannelMask()) == channelMask)) { currentMatchCriteria[2] = outputChannelCount; } currentMatchCriteria[3] = outputChannelCount; } // sampling rate match if (samplingRate > SAMPLE_RATE_HZ_DEFAULT) { int diff; // avoid unsigned integer overflow. __builtin_sub_overflow(outputDesc->getSamplingRate(), samplingRate, &diff); // prefer the closest output sampling rate greater than or equal to target // if none exists, prefer the closest output sampling rate less than target. // // criteria is offset to make non-negative. currentMatchCriteria[4] = diff >= 0 ? -diff + 200'000'000 : diff + 100'000'000; } // performance flags match currentMatchCriteria[5] = popcount(outputDesc->mFlags & performanceFlags); // format match if (format != AUDIO_FORMAT_INVALID) { currentMatchCriteria[6] = PolicyAudioPort::kFormatDistanceMax - PolicyAudioPort::formatDistance(format, outputDesc->getFormat()); } // primary output match currentMatchCriteria[7] = outputDesc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY; // compare match criteria by priority then value if (std::lexicographical_compare(bestMatchCriteria.begin(), bestMatchCriteria.end(), currentMatchCriteria.begin(), currentMatchCriteria.end())) { bestMatchCriteria = currentMatchCriteria; bestOutput = output; std::stringstream result; std::copy(bestMatchCriteria.begin(), bestMatchCriteria.end(), std::ostream_iterator(result, " ")); ALOGV("%s new bestOutput %d criteria %s", __func__, bestOutput, result.str().c_str()); } } return bestOutput; } status_t AudioPolicyManager::startOutput(audio_port_handle_t portId) { ALOGV("%s portId %d", __FUNCTION__, portId); sp outputDesc = mOutputs.getOutputForClient(portId); if (outputDesc == 0) { ALOGW("startOutput() no output for client %d", portId); return BAD_VALUE; } sp client = outputDesc->getClient(portId); ALOGV("startOutput() output %d, stream %d, session %d", outputDesc->mIoHandle, client->stream(), client->session()); if (com::android::media::audioserver::fix_concurrent_playback_behavior_with_bit_perfect_client() && gHighPriorityUseCases.count(client->attributes().usage) != 0 && outputDesc->isBitPerfect()) { // Usually, APM selects bit-perfect output for high priority use cases only when // bit-perfect output is the only output that can be routed to the selected device. // However, here is no need to play high priority use cases such as ringtone and alarm // on the bit-perfect path. Reopen the output and return DEAD_OBJECT so that the client // can attach to new output. ALOGD("%s: reopen bit-perfect output as high priority use case(%d) is starting", __func__, client->stream()); reopenOutput(outputDesc, nullptr /*config*/, AUDIO_OUTPUT_FLAG_NONE, __func__); return DEAD_OBJECT; } status_t status = outputDesc->start(); if (status != NO_ERROR) { return status; } uint32_t delayMs; status = startSource(outputDesc, client, &delayMs); if (status != NO_ERROR) { outputDesc->stop(); if (status == DEAD_OBJECT) { sp desc = reopenOutput(outputDesc, nullptr /*config*/, AUDIO_OUTPUT_FLAG_NONE, __func__); if (desc == nullptr) { // This is not common, it may indicate something wrong with the HAL. ALOGE("%s unable to open output with default config", __func__); return status; } } return status; } // If the client is the first one active on preferred mixer parameters, reopen the output // if the current mixer parameters doesn't match the preferred one. if (outputDesc->devices().size() == 1) { sp info = getPreferredMixerAttributesInfo( outputDesc->devices()[0]->getId(), client->strategy()); if (info != nullptr && info->getUid() == client->uid()) { if (info->getActiveClientCount() == 0 && !outputDesc->isConfigurationMatched( info->getConfigBase(), info->getFlags())) { stopSource(outputDesc, client); outputDesc->stop(); audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.channel_mask = info->getConfigBase().channel_mask; config.sample_rate = info->getConfigBase().sample_rate; config.format = info->getConfigBase().format; sp desc = reopenOutput(outputDesc, &config, info->getFlags(), __func__); if (desc == nullptr) { return BAD_VALUE; } desc->mPreferredAttrInfo = info; // Intentionally return error to let the client side resending request for // creating and starting. return DEAD_OBJECT; } info->increaseActiveClient(); if (info->getActiveClientCount() == 1 && info->isBitPerfect()) { // If it is first bit-perfect client, reroute all clients that will be routed to // the bit-perfect sink so that it is guaranteed only bit-perfect stream is active. PortHandleVector clientsToInvalidate; for (size_t i = 0; i < mOutputs.size(); i++) { if (mOutputs[i] == outputDesc || mOutputs[i]->devices().filter(outputDesc->devices()).isEmpty()) { continue; } for (const auto& c : mOutputs[i]->getClientIterable()) { clientsToInvalidate.push_back(c->portId()); } } if (!clientsToInvalidate.empty()) { ALOGD("%s Invalidate clients due to first bit-perfect client started", __func__); mpClientInterface->invalidateTracks(clientsToInvalidate); } } } } if (client->hasPreferredDevice()) { // playback activity with preferred device impacts routing occurred, inform upper layers mpClientInterface->onRoutingUpdated(); } if (delayMs != 0) { usleep(delayMs * 1000); } if (status == NO_ERROR && outputDesc->mPreferredAttrInfo != nullptr && outputDesc->isBitPerfect() && com::android::media::audioserver:: fix_concurrent_playback_behavior_with_bit_perfect_client()) { // A new client is started on bit-perfect output, update all clients internal mute. updateClientsInternalMute(outputDesc); } return status; } bool AudioPolicyManager::isLeUnicastActive() const { if (isInCall()) { return true; } return isAnyDeviceTypeActive(getAudioDeviceOutLeAudioUnicastSet()); } bool AudioPolicyManager::isAnyDeviceTypeActive(const DeviceTypeSet& deviceTypes) const { if (mAvailableOutputDevices.getDevicesFromTypes(deviceTypes).isEmpty()) { return false; } bool active = mOutputs.isAnyDeviceTypeActive(deviceTypes); ALOGV("%s active %d", __func__, active); return active; } status_t AudioPolicyManager::startSource(const sp& outputDesc, const sp& client, uint32_t *delayMs) { // cannot start playback of STREAM_TTS if any other output is being used uint32_t beaconMuteLatency = 0; *delayMs = 0; audio_stream_type_t stream = client->stream(); auto clientVolSrc = client->volumeSource(); auto clientStrategy = client->strategy(); auto clientAttr = client->attributes(); if (stream == AUDIO_STREAM_TTS) { ALOGV("\t found BEACON stream"); if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive( toVolumeSource(AUDIO_STREAM_TTS, false) /*sourceToIgnore*/)) { return INVALID_OPERATION; } else { beaconMuteLatency = handleEventForBeacon(STARTING_BEACON); } } else { // some playback other than beacon starts beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); } // force device change if the output is inactive and no audio patch is already present. // check active before incrementing usage count bool force = !outputDesc->isActive() && !outputDesc->isRouted(); DeviceVector devices; sp policyMix = outputDesc->mPolicyMix.promote(); const char *address = NULL; if (policyMix != nullptr) { audio_devices_t newDeviceType; address = policyMix->mDeviceAddress.c_str(); if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) { newDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; } else { newDeviceType = policyMix->mDeviceType; } sp device = mAvailableOutputDevices.getDevice(newDeviceType, String8(address), AUDIO_FORMAT_DEFAULT); ALOG_ASSERT(device, "%s: no device found t=%u, a=%s", __func__, newDeviceType, address); devices.add(device); } // requiresMuteCheck is false when we can bypass mute strategy. // It covers a common case when there is no materially active audio // and muting would result in unnecessary delay and dropped audio. const uint32_t outputLatencyMs = outputDesc->latency(); bool requiresMuteCheck = outputDesc->isActive(outputLatencyMs * 2); // account for drain bool wasLeUnicastActive = isLeUnicastActive(); // increment usage count for this stream on the requested output: // NOTE that the usage count is the same for duplicated output and hardware output which is // necessary for a correct control of hardware output routing by startOutput() and stopOutput() outputDesc->setClientActive(client, true); if (client->hasPreferredDevice(true)) { if (outputDesc->sameExclusivePreferredDevicesCount() > 0) { // Preferred device may be exclusive, use only if no other active clients on this output devices = DeviceVector( mAvailableOutputDevices.getDeviceFromId(client->preferredDeviceId())); } else { devices = getNewOutputDevices(outputDesc, false /*fromCache*/); } if (devices != outputDesc->devices()) { checkStrategyRoute(clientStrategy, outputDesc->mIoHandle); } } if (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_MEDIA))) { selectOutputForMusicEffects(); } if (outputDesc->getActivityCount(clientVolSrc) == 1 || !devices.isEmpty()) { // starting an output being rerouted? if (devices.isEmpty()) { devices = getNewOutputDevices(outputDesc, false /*fromCache*/); } bool shouldWait = (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM)) || followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_NOTIFICATION)) || (beaconMuteLatency > 0)); uint32_t waitMs = beaconMuteLatency; const bool needToCloseBitPerfectOutput = (com::android::media::audioserver:: fix_concurrent_playback_behavior_with_bit_perfect_client() && gHighPriorityUseCases.count(clientAttr.usage) != 0); std::vector> outputsToReopen; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (desc != outputDesc) { // An output has a shared device if // - managed by the same hw module // - supports the currently selected device const bool sharedDevice = outputDesc->sharesHwModuleWith(desc) && (!desc->filterSupportedDevices(devices).isEmpty()); // force a device change if any other output is: // - managed by the same hw module // - supports currently selected device // - has a current device selection that differs from selected device. // - has an active audio patch // In this case, the audio HAL must receive the new device selection so that it can // change the device currently selected by the other output. if (sharedDevice && desc->devices() != devices && desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) { force = true; } // wait for audio on other active outputs to be presented when starting // a notification so that audio focus effect can propagate, or that a mute/unmute // event occurred for beacon const uint32_t latencyMs = desc->latency(); const bool isActive = desc->isActive(latencyMs * 2); // account for drain if (shouldWait && isActive && (waitMs < latencyMs)) { waitMs = latencyMs; } // Require mute check if another output is on a shared device // and currently active to have proper drain and avoid pops. // Note restoring AudioTracks onto this output needs to invoke // a volume ramp if there is no mute. requiresMuteCheck |= sharedDevice && isActive; if (needToCloseBitPerfectOutput && desc->isBitPerfect()) { outputsToReopen.push_back(desc); } } } if (outputDesc->mPreferredAttrInfo != nullptr && devices != outputDesc->devices()) { // If the output is open with preferred mixer attributes, but the routed device is // changed when calling this function, returning DEAD_OBJECT to indicate routing // changed. return DEAD_OBJECT; } for (auto& outputToReopen : outputsToReopen) { reopenOutput(outputToReopen, nullptr /*config*/, AUDIO_OUTPUT_FLAG_NONE, __func__); } const uint32_t muteWaitMs = setOutputDevices(__func__, outputDesc, devices, force, 0, nullptr, requiresMuteCheck); // apply volume rules for current stream and device if necessary auto &curves = getVolumeCurves(client->attributes()); if (NO_ERROR != checkAndSetVolume(curves, client->volumeSource(), curves.getVolumeIndex(outputDesc->devices().types()), outputDesc, outputDesc->devices().types(), 0 /*delay*/, outputDesc->useHwGain() /*force*/)) { // request AudioService to reinitialize the volume curves asynchronously ALOGE("checkAndSetVolume failed, requesting volume range init"); mpClientInterface->onVolumeRangeInitRequest(); }; // update the outputs if starting an output with a stream that can affect notification // routing handleNotificationRoutingForStream(stream); // force reevaluating accessibility routing when ringtone or alarm starts if (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM))) { invalidateStreams({AUDIO_STREAM_ACCESSIBILITY}); } if (waitMs > muteWaitMs) { *delayMs = waitMs - muteWaitMs; } // FIXME: A device change (muteWaitMs > 0) likely introduces a volume change. // A volume change enacted by APM with 0 delay is not synchronous, as it goes // via AudioCommandThread to AudioFlinger. Hence it is possible that the volume // change occurs after the MixerThread starts and causes a stream volume // glitch. // // We do not introduce additional delay here. } if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE && mEngine->getForceUse( AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), true, outputDesc); } // Automatically enable the remote submix input when output is started on a re routing mix // of type MIX_TYPE_RECORDERS if (isSingleDeviceType(devices.types(), &audio_is_remote_submix_device) && policyMix != NULL && policyMix->mMixType == MIX_TYPE_RECORDERS) { setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address, "remote-submix", AUDIO_FORMAT_DEFAULT); } checkLeBroadcastRoutes(wasLeUnicastActive, outputDesc, *delayMs); return NO_ERROR; } void AudioPolicyManager::checkLeBroadcastRoutes(bool wasUnicastActive, sp ignoredOutput, uint32_t delayMs) { bool isUnicastActive = isLeUnicastActive(); if (wasUnicastActive != isUnicastActive) { std::map outputsToReopen; //reroute all outputs routed to LE broadcast if LE unicast activy changed on any output for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (desc != ignoredOutput && desc->isActive() && ((isUnicastActive && !desc->devices(). getDevicesFromType(AUDIO_DEVICE_OUT_BLE_BROADCAST).isEmpty()) || (wasUnicastActive && !desc->devices().getDevicesFromTypes( getAudioDeviceOutLeAudioUnicastSet()).isEmpty()))) { DeviceVector newDevices = getNewOutputDevices(desc, false /*fromCache*/); bool force = desc->devices() != newDevices; if (desc->mPreferredAttrInfo != nullptr && force) { // If the device is using preferred mixer attributes, the output need to reopen // with default configuration when the new selected devices are different from // current routing devices. outputsToReopen.emplace(mOutputs.keyAt(i), newDevices); continue; } setOutputDevices(__func__, desc, newDevices, force, delayMs); // re-apply device specific volume if not done by setOutputDevice() if (!force) { applyStreamVolumes(desc, newDevices.types(), delayMs); } } } reopenOutputsWithDevices(outputsToReopen); } } status_t AudioPolicyManager::stopOutput(audio_port_handle_t portId) { ALOGV("%s portId %d", __FUNCTION__, portId); sp outputDesc = mOutputs.getOutputForClient(portId); if (outputDesc == 0) { ALOGW("stopOutput() no output for client %d", portId); return BAD_VALUE; } sp client = outputDesc->getClient(portId); if (client->hasPreferredDevice(true)) { // playback activity with preferred device impacts routing occurred, inform upper layers mpClientInterface->onRoutingUpdated(); } ALOGV("stopOutput() output %d, stream %d, session %d", outputDesc->mIoHandle, client->stream(), client->session()); status_t status = stopSource(outputDesc, client); if (status == NO_ERROR ) { outputDesc->stop(); } else { return status; } if (outputDesc->devices().size() == 1) { sp info = getPreferredMixerAttributesInfo( outputDesc->devices()[0]->getId(), client->strategy()); bool outputReopened = false; if (info != nullptr && info->getUid() == client->uid()) { info->decreaseActiveClient(); if (info->getActiveClientCount() == 0) { reopenOutput(outputDesc, nullptr /*config*/, AUDIO_OUTPUT_FLAG_NONE, __func__); outputReopened = true; } } if (com::android::media::audioserver:: fix_concurrent_playback_behavior_with_bit_perfect_client() && !outputReopened && outputDesc->isBitPerfect()) { // Only need to update the clients' internal mute when the output is bit-perfect and it // is not reopened. updateClientsInternalMute(outputDesc); } } return status; } status_t AudioPolicyManager::stopSource(const sp& outputDesc, const sp& client) { // always handle stream stop, check which stream type is stopping audio_stream_type_t stream = client->stream(); auto clientVolSrc = client->volumeSource(); bool wasLeUnicastActive = isLeUnicastActive(); handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); if (outputDesc->getActivityCount(clientVolSrc) > 0) { if (outputDesc->getActivityCount(clientVolSrc) == 1) { // Automatically disable the remote submix input when output is stopped on a // re routing mix of type MIX_TYPE_RECORDERS sp policyMix = outputDesc->mPolicyMix.promote(); if (isSingleDeviceType( outputDesc->devices().types(), &audio_is_remote_submix_device) && policyMix != nullptr && policyMix->mMixType == MIX_TYPE_RECORDERS) { setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, policyMix->mDeviceAddress, "remote-submix", AUDIO_FORMAT_DEFAULT); } } bool forceDeviceUpdate = false; if (client->hasPreferredDevice(true) && outputDesc->sameExclusivePreferredDevicesCount() < 2) { checkStrategyRoute(client->strategy(), AUDIO_IO_HANDLE_NONE); forceDeviceUpdate = true; } // decrement usage count of this stream on the output outputDesc->setClientActive(client, false); // store time at which the stream was stopped - see isStreamActive() if (outputDesc->getActivityCount(clientVolSrc) == 0 || forceDeviceUpdate) { outputDesc->setStopTime(client, systemTime()); DeviceVector newDevices = getNewOutputDevices(outputDesc, false /*fromCache*/); // If the routing does not change, if an output is routed on a device using HwGain // (aka setAudioPortConfig) and there are still active clients following different // volume group(s), force reapply volume bool requiresVolumeCheck = outputDesc->getActivityCount(clientVolSrc) == 0 && outputDesc->useHwGain() && outputDesc->isAnyActive(VOLUME_SOURCE_NONE); // delay the device switch by twice the latency because stopOutput() is executed when // the track stop() command is received and at that time the audio track buffer can // still contain data that needs to be drained. The latency only covers the audio HAL // and kernel buffers. Also the latency does not always include additional delay in the // audio path (audio DSP, CODEC ...) setOutputDevices(__func__, outputDesc, newDevices, false, outputDesc->latency()*2, nullptr, true /*requiresMuteCheck*/, requiresVolumeCheck); // force restoring the device selection on other active outputs if it differs from the // one being selected for this output std::map outputsToReopen; uint32_t delayMs = outputDesc->latency()*2; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (desc != outputDesc && desc->isActive() && outputDesc->sharesHwModuleWith(desc) && (newDevices != desc->devices())) { DeviceVector newDevices2 = getNewOutputDevices(desc, false /*fromCache*/); bool force = desc->devices() != newDevices2; if (desc->mPreferredAttrInfo != nullptr && force) { // If the device is using preferred mixer attributes, the output need to // reopen with default configuration when the new selected devices are // different from current routing devices. outputsToReopen.emplace(mOutputs.keyAt(i), newDevices2); continue; } setOutputDevices(__func__, desc, newDevices2, force, delayMs); // re-apply device specific volume if not done by setOutputDevice() if (!force) { applyStreamVolumes(desc, newDevices2.types(), delayMs); } } } reopenOutputsWithDevices(outputsToReopen); // update the outputs if stopping one with a stream that can affect notification routing handleNotificationRoutingForStream(stream); } if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE && mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), false, outputDesc); } if (followsSameRouting(client->attributes(), attributes_initializer(AUDIO_USAGE_MEDIA))) { selectOutputForMusicEffects(); } checkLeBroadcastRoutes(wasLeUnicastActive, outputDesc, outputDesc->latency()*2); return NO_ERROR; } else { ALOGW("stopOutput() refcount is already 0"); return INVALID_OPERATION; } } bool AudioPolicyManager::releaseOutput(audio_port_handle_t portId) { ALOGV("%s portId %d", __FUNCTION__, portId); sp outputDesc = mOutputs.getOutputForClient(portId); if (outputDesc == 0) { // If an output descriptor is closed due to a device routing change, // then there are race conditions with releaseOutput from tracks // that may be destroyed (with no PlaybackThread) or a PlaybackThread // destroyed shortly thereafter. // // Here we just log a warning, instead of a fatal error. ALOGW("releaseOutput() no output for client %d", portId); return false; } ALOGV("releaseOutput() %d", outputDesc->mIoHandle); sp client = outputDesc->getClient(portId); if (outputDesc->isClientActive(client)) { ALOGW("releaseOutput() inactivates portId %d in good faith", portId); stopOutput(portId); } if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { if (outputDesc->mDirectOpenCount <= 0) { ALOGW("releaseOutput() invalid open count %d for output %d", outputDesc->mDirectOpenCount, outputDesc->mIoHandle); return false; } if (--outputDesc->mDirectOpenCount == 0) { closeOutput(outputDesc->mIoHandle); mpClientInterface->onAudioPortListUpdate(); } } outputDesc->removeClient(portId); if (outputDesc->mPendingReopenToQueryProfiles && outputDesc->getClientCount() == 0) { // The output is pending reopened to query dynamic profiles and // there is no active clients closeOutput(outputDesc->mIoHandle); sp newOutputDesc = openOutputWithProfileAndDevice( outputDesc->mProfile, mEngine->getActiveMediaDevices(mAvailableOutputDevices)); if (newOutputDesc == nullptr) { ALOGE("%s failed to open output", __func__); } return true; } return false; } status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, audio_io_handle_t *input, audio_unique_id_t riid, audio_session_t session, const AttributionSourceState& attributionSource, audio_config_base_t *config, audio_input_flags_t flags, audio_port_handle_t *selectedDeviceId, input_type_t *inputType, audio_port_handle_t *portId, uint32_t *virtualDeviceId) { ALOGV("%s() source %d, sampling rate %d, format %#x, channel mask %#x, session %d, " "flags %#x attributes=%s requested device ID %d", __func__, attr->source, config->sample_rate, config->format, config->channel_mask, session, flags, toString(*attr).c_str(), *selectedDeviceId); status_t status = NO_ERROR; audio_attributes_t attributes = *attr; sp policyMix; sp device; sp inputDesc; sp previousInputDesc; sp clientDesc; audio_port_handle_t requestedDeviceId = *selectedDeviceId; uid_t uid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(attributionSource.uid)); bool isSoundTrigger; // The supplied portId must be AUDIO_PORT_HANDLE_NONE if (*portId != AUDIO_PORT_HANDLE_NONE) { return INVALID_OPERATION; } if (attr->source == AUDIO_SOURCE_DEFAULT) { attributes.source = AUDIO_SOURCE_MIC; } // Explicit routing? sp explicitRoutingDevice = mAvailableInputDevices.getDeviceFromId(*selectedDeviceId); // special case for mmap capture: if an input IO handle is specified, we reuse this input if // possible if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) == AUDIO_INPUT_FLAG_MMAP_NOIRQ && *input != AUDIO_IO_HANDLE_NONE) { ssize_t index = mInputs.indexOfKey(*input); if (index < 0) { ALOGW("getInputForAttr() unknown MMAP input %d", *input); status = BAD_VALUE; goto error; } sp inputDesc = mInputs.valueAt(index); RecordClientVector clients = inputDesc->getClientsForSession(session); if (clients.size() == 0) { ALOGW("getInputForAttr() unknown session %d on input %d", session, *input); status = BAD_VALUE; goto error; } // For MMAP mode, the first call to getInputForAttr() is made on behalf of audioflinger. // The second call is for the first active client and sets the UID. Any further call // corresponds to a new client and is only permitted from the same UID. // If the first UID is silenced, allow a new UID connection and replace with new UID if (clients.size() > 1) { for (const auto& client : clients) { // The client map is ordered by key values (portId) and portIds are allocated // incrementaly. So the first client in this list is the one opened by audio flinger // when the mmap stream is created and should be ignored as it does not correspond // to an actual client if (client == *clients.cbegin()) { continue; } if (uid != client->uid() && !client->isSilenced()) { ALOGW("getInputForAttr() bad uid %d for client %d uid %d", uid, client->portId(), client->uid()); status = INVALID_OPERATION; goto error; } } } *inputType = API_INPUT_LEGACY; device = inputDesc->getDevice(); ALOGV("%s reusing MMAP input %d for session %d", __FUNCTION__, *input, session); goto exit; } *input = AUDIO_IO_HANDLE_NONE; *inputType = API_INPUT_INVALID; if (attributes.source == AUDIO_SOURCE_REMOTE_SUBMIX && extractAddressFromAudioAttributes(attributes).has_value()) { status = mPolicyMixes.getInputMixForAttr(attributes, &policyMix); if (status != NO_ERROR) { ALOGW("%s could not find input mix for attr %s", __func__, toString(attributes).c_str()); goto error; } device = mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_REMOTE_SUBMIX, String8(attr->tags + strlen("addr=")), AUDIO_FORMAT_DEFAULT); if (device == nullptr) { ALOGW("%s could not find in Remote Submix device for source %d, tags %s", __func__, attributes.source, attributes.tags); status = BAD_VALUE; goto error; } if (is_mix_loopback_render(policyMix->mRouteFlags)) { *inputType = API_INPUT_MIX_PUBLIC_CAPTURE_PLAYBACK; } else { *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; } if (virtualDeviceId) { *virtualDeviceId = policyMix->mVirtualDeviceId; } } else { if (explicitRoutingDevice != nullptr) { device = explicitRoutingDevice; } else { // Prevent from storing invalid requested device id in clients requestedDeviceId = AUDIO_PORT_HANDLE_NONE; device = mEngine->getInputDeviceForAttributes(attributes, uid, session, &policyMix); ALOGV_IF(device != nullptr, "%s found device type is 0x%X", __FUNCTION__, device->type()); } if (device == nullptr) { ALOGW("getInputForAttr() could not find device for source %d", attributes.source); status = BAD_VALUE; goto error; } if (device->type() == AUDIO_DEVICE_IN_ECHO_REFERENCE) { *inputType = API_INPUT_MIX_CAPTURE; } else if (policyMix) { ALOG_ASSERT(policyMix->mMixType == MIX_TYPE_RECORDERS, "Invalid Mix Type"); // there is an external policy, but this input is attached to a mix of recorders, // meaning it receives audio injected into the framework, so the recorder doesn't // know about it and is therefore considered "legacy" *inputType = API_INPUT_LEGACY; if (virtualDeviceId) { *virtualDeviceId = policyMix->mVirtualDeviceId; } } else if (audio_is_remote_submix_device(device->type())) { *inputType = API_INPUT_MIX_CAPTURE; } else if (device->type() == AUDIO_DEVICE_IN_TELEPHONY_RX) { *inputType = API_INPUT_TELEPHONY_RX; } else { *inputType = API_INPUT_LEGACY; } } *input = getInputForDevice(device, session, attributes, config, flags, policyMix); if (*input == AUDIO_IO_HANDLE_NONE) { status = INVALID_OPERATION; AudioProfileVector profiles; status_t ret = getProfilesForDevices( DeviceVector(device), profiles, flags, true /*isInput*/); if (ret == NO_ERROR && !profiles.empty()) { const auto channels = profiles[0]->getChannels(); if (!channels.empty() && (channels.find(config->channel_mask) == channels.end())) { config->channel_mask = *channels.begin(); } const auto sampleRates = profiles[0]->getSampleRates(); if (!sampleRates.empty() && (sampleRates.find(config->sample_rate) == sampleRates.end())) { config->sample_rate = *sampleRates.begin(); } config->format = profiles[0]->getFormat(); } goto error; } if (policyMix != nullptr && virtualDeviceId != nullptr) { *virtualDeviceId = policyMix->mVirtualDeviceId; } exit: *selectedDeviceId = mAvailableInputDevices.contains(device) ? device->getId() : AUDIO_PORT_HANDLE_NONE; isSoundTrigger = attributes.source == AUDIO_SOURCE_HOTWORD && mSoundTriggerSessions.indexOfKey(session) >= 0; *portId = PolicyAudioPort::getNextUniqueId(); clientDesc = new RecordClientDescriptor(*portId, riid, uid, session, attributes, *config, requestedDeviceId, attributes.source, flags, isSoundTrigger); inputDesc = mInputs.valueFor(*input); // Move (if found) effect for the client session to its input mEffects.moveEffectsForIo(session, *input, &mInputs, mpClientInterface); inputDesc->addClient(clientDesc); ALOGV("getInputForAttr() returns input %d type %d selectedDeviceId %d for port ID %d", *input, *inputType, *selectedDeviceId, *portId); return NO_ERROR; error: return status; } audio_io_handle_t AudioPolicyManager::getInputForDevice(const sp &device, audio_session_t session, const audio_attributes_t &attributes, audio_config_base_t *config, audio_input_flags_t flags, const sp &policyMix) { audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; audio_source_t halInputSource = attributes.source; bool isSoundTrigger = false; if (attributes.source == AUDIO_SOURCE_HOTWORD) { ssize_t index = mSoundTriggerSessions.indexOfKey(session); if (index >= 0) { input = mSoundTriggerSessions.valueFor(session); isSoundTrigger = true; flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD); ALOGV("SoundTrigger capture on session %d input %d", session, input); } else { halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION; } } else if (attributes.source == AUDIO_SOURCE_VOICE_COMMUNICATION && audio_is_linear_pcm(config->format)) { flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_VOIP_TX); } if (attributes.source == AUDIO_SOURCE_ULTRASOUND) { flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_ULTRASOUND); } // sampling rate and flags may be updated by getInputProfile uint32_t profileSamplingRate = (config->sample_rate == 0) ? SAMPLE_RATE_HZ_DEFAULT : config->sample_rate; audio_format_t profileFormat = config->format; audio_channel_mask_t profileChannelMask = config->channel_mask; audio_input_flags_t profileFlags = flags; // find a compatible input profile (not necessarily identical in parameters) sp profile = getInputProfile( device, profileSamplingRate, profileFormat, profileChannelMask, profileFlags); if (profile == nullptr) { return input; } // Pick input sampling rate if not specified by client uint32_t samplingRate = config->sample_rate; if (samplingRate == 0) { samplingRate = profileSamplingRate; } if (profile->getModuleHandle() == 0) { ALOGE("getInputForAttr(): HW module %s not opened", profile->getModuleName()); return input; } // Reuse an already opened input if a client with the same session ID already exists // on that input for (size_t i = 0; i < mInputs.size(); i++) { sp desc = mInputs.valueAt(i); if (desc->mProfile != profile) { continue; } RecordClientVector clients = desc->clientsList(); for (const auto &client : clients) { if (session == client->session()) { return desc->mIoHandle; } } } if (!profile->canOpenNewIo()) { for (size_t i = 0; i < mInputs.size(); ) { sp desc = mInputs.valueAt(i); if (desc->mProfile != profile) { i++; continue; } // if sound trigger, reuse input if used by other sound trigger on same session // else // reuse input if active client app is not in IDLE state // RecordClientVector clients = desc->clientsList(); bool doClose = false; for (const auto& client : clients) { if (isSoundTrigger != client->isSoundTrigger()) { continue; } if (client->isSoundTrigger()) { if (session == client->session()) { return desc->mIoHandle; } continue; } if (client->active() && client->appState() != APP_STATE_IDLE) { return desc->mIoHandle; } doClose = true; } if (doClose) { closeInput(desc->mIoHandle); } else { i++; } } } sp inputDesc = new AudioInputDescriptor(profile, mpClientInterface); audio_config_t lConfig = AUDIO_CONFIG_INITIALIZER; lConfig.sample_rate = profileSamplingRate; lConfig.channel_mask = profileChannelMask; lConfig.format = profileFormat; status_t status = inputDesc->open(&lConfig, device, halInputSource, profileFlags, &input); // only accept input with the exact requested set of parameters if (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE || (profileSamplingRate != lConfig.sample_rate) || !audio_formats_match(profileFormat, lConfig.format) || (profileChannelMask != lConfig.channel_mask)) { ALOGW("getInputForAttr() failed opening input: sampling rate %d" ", format %#x, channel mask %#x", profileSamplingRate, profileFormat, profileChannelMask); if (input != AUDIO_IO_HANDLE_NONE) { inputDesc->close(); } return AUDIO_IO_HANDLE_NONE; } inputDesc->mPolicyMix = policyMix; addInput(input, inputDesc); mpClientInterface->onAudioPortListUpdate(); return input; } status_t AudioPolicyManager::startInput(audio_port_handle_t portId) { ALOGV("%s portId %d", __FUNCTION__, portId); sp inputDesc = mInputs.getInputForClient(portId); if (inputDesc == 0) { ALOGW("%s no input for client %d", __FUNCTION__, portId); return DEAD_OBJECT; } audio_io_handle_t input = inputDesc->mIoHandle; sp client = inputDesc->getClient(portId); if (client->active()) { ALOGW("%s input %d client %d already started", __FUNCTION__, input, client->portId()); return INVALID_OPERATION; } audio_session_t session = client->session(); ALOGV("%s input:%d, session:%d)", __FUNCTION__, input, session); Vector> activeInputs = mInputs.getActiveInputs(); status_t status = inputDesc->start(); if (status != NO_ERROR) { return status; } // increment activity count before calling getNewInputDevice() below as only active sessions // are considered for device selection inputDesc->setClientActive(client, true); // indicate active capture to sound trigger service if starting capture from a mic on // primary HW module sp device = getNewInputDevice(inputDesc); if (device != nullptr) { status = setInputDevice(input, device, true /* force */); } else { ALOGW("%s no new input device can be found for descriptor %d", __FUNCTION__, inputDesc->getId()); status = BAD_VALUE; } if (status == NO_ERROR && inputDesc->activeCount() == 1) { sp policyMix = inputDesc->mPolicyMix.promote(); // if input maps to a dynamic policy with an activity listener, notify of state change if ((policyMix != nullptr) && ((policyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) { mpClientInterface->onDynamicPolicyMixStateUpdate(policyMix->mDeviceAddress, MIX_STATE_MIXING); } DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices(); if (primaryInputDevices.contains(device) && mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1) { mpClientInterface->setSoundTriggerCaptureState(true); } // automatically enable the remote submix output when input is started if not // used by a policy mix of type MIX_TYPE_RECORDERS // For remote submix (a virtual device), we open only one input per capture request. if (audio_is_remote_submix_device(inputDesc->getDeviceType())) { String8 address = String8(""); if (policyMix == nullptr) { address = String8("0"); } else if (policyMix->mMixType == MIX_TYPE_PLAYERS) { address = policyMix->mDeviceAddress; } if (address != "") { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address, "remote-submix", AUDIO_FORMAT_DEFAULT); } } } else if (status != NO_ERROR) { // Restore client activity state. inputDesc->setClientActive(client, false); inputDesc->stop(); } ALOGV("%s input %d source = %d status = %d exit", __FUNCTION__, input, client->source(), status); return status; } status_t AudioPolicyManager::stopInput(audio_port_handle_t portId) { ALOGV("%s portId %d", __FUNCTION__, portId); sp inputDesc = mInputs.getInputForClient(portId); if (inputDesc == 0) { ALOGW("%s no input for client %d", __FUNCTION__, portId); return BAD_VALUE; } audio_io_handle_t input = inputDesc->mIoHandle; sp client = inputDesc->getClient(portId); if (!client->active()) { ALOGW("%s input %d client %d already stopped", __FUNCTION__, input, client->portId()); return INVALID_OPERATION; } auto old_source = inputDesc->source(); inputDesc->setClientActive(client, false); inputDesc->stop(); if (inputDesc->isActive()) { auto current_source = inputDesc->source(); setInputDevice(input, getNewInputDevice(inputDesc), old_source != current_source /* force */); } else { sp policyMix = inputDesc->mPolicyMix.promote(); // if input maps to a dynamic policy with an activity listener, notify of state change if ((policyMix != nullptr) && ((policyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) { mpClientInterface->onDynamicPolicyMixStateUpdate(policyMix->mDeviceAddress, MIX_STATE_IDLE); } // automatically disable the remote submix output when input is stopped if not // used by a policy mix of type MIX_TYPE_RECORDERS if (audio_is_remote_submix_device(inputDesc->getDeviceType())) { String8 address = String8(""); if (policyMix == nullptr) { address = String8("0"); } else if (policyMix->mMixType == MIX_TYPE_PLAYERS) { address = policyMix->mDeviceAddress; } if (address != "") { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address, "remote-submix", AUDIO_FORMAT_DEFAULT); } } resetInputDevice(input); // indicate inactive capture to sound trigger service if stopping capture from a mic on // primary HW module DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices(); if (primaryInputDevices.contains(inputDesc->getDevice()) && mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) { mpClientInterface->setSoundTriggerCaptureState(false); } inputDesc->clearPreemptedSessions(); } return NO_ERROR; } void AudioPolicyManager::releaseInput(audio_port_handle_t portId) { ALOGV("%s portId %d", __FUNCTION__, portId); sp inputDesc = mInputs.getInputForClient(portId); if (inputDesc == 0) { ALOGW("%s no input for client %d", __FUNCTION__, portId); return; } sp client = inputDesc->getClient(portId); audio_io_handle_t input = inputDesc->mIoHandle; ALOGV("%s %d", __FUNCTION__, input); inputDesc->removeClient(portId); // If no more clients are present in this session, park effects to an orphan chain RecordClientVector clientsOnSession = inputDesc->getClientsForSession(client->session()); if (clientsOnSession.size() == 0) { mEffects.putOrphanEffects(client->session(), input, &mInputs, mpClientInterface); } if (inputDesc->getClientCount() > 0) { ALOGV("%s(%d) %zu clients remaining", __func__, portId, inputDesc->getClientCount()); return; } closeInput(input); mpClientInterface->onAudioPortListUpdate(); ALOGV("%s exit", __FUNCTION__); } void AudioPolicyManager::closeActiveClients(const sp& input) { RecordClientVector clients = input->clientsList(true); for (const auto& client : clients) { closeClient(client->portId()); } } void AudioPolicyManager::closeClient(audio_port_handle_t portId) { stopInput(portId); releaseInput(portId); } bool AudioPolicyManager::checkCloseInput(const sp& input) { if (input->clientsList().size() == 0 || !mAvailableInputDevices.containsAtLeastOne(input->supportedDevices())) { return true; } for (const auto& client : input->clientsList()) { sp device = mEngine->getInputDeviceForAttributes(client->attributes(), client->uid(), client->session()); if (!input->supportedDevices().contains(device)) { return true; } } setInputDevice(input->mIoHandle, getNewInputDevice(input)); return false; } void AudioPolicyManager::checkCloseInputs() { // After connecting or disconnecting an input device, close input if: // - it has no client (was just opened to check profile) OR // - none of its supported devices are connected anymore OR // - one of its clients cannot be routed to one of its supported // devices anymore. Otherwise update device selection std::vector inputsToClose; for (size_t i = 0; i < mInputs.size(); i++) { if (checkCloseInput(mInputs.valueAt(i))) { inputsToClose.push_back(mInputs.keyAt(i)); } } for (const audio_io_handle_t handle : inputsToClose) { ALOGV("%s closing input %d", __func__, handle); closeInput(handle); } } status_t AudioPolicyManager::setDeviceAbsoluteVolumeEnabled(audio_devices_t deviceType, const char *address __unused, bool enabled, audio_stream_type_t streamToDriveAbs) { audio_attributes_t attributesToDriveAbs = mEngine->getAttributesForStreamType(streamToDriveAbs); if (attributesToDriveAbs == AUDIO_ATTRIBUTES_INITIALIZER) { ALOGW("%s: no attributes for stream %s, bailing out", __func__, toString(streamToDriveAbs).c_str()); return BAD_VALUE; } if (enabled) { mAbsoluteVolumeDrivingStreams[deviceType] = attributesToDriveAbs; } else { mAbsoluteVolumeDrivingStreams.erase(deviceType); } return NO_ERROR; } void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) { ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); if (indexMin < 0 || indexMax < 0) { ALOGE("%s for stream %d: invalid min %d or max %d", __func__, stream , indexMin, indexMax); return; } getVolumeCurves(stream).initVolume(indexMin, indexMax); // initialize other private stream volumes which follow this one for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) { if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) { continue; } getVolumeCurves((audio_stream_type_t)curStream).initVolume(indexMin, indexMax); } } status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, int index, audio_devices_t device) { auto attributes = mEngine->getAttributesForStreamType(stream); if (attributes == AUDIO_ATTRIBUTES_INITIALIZER) { ALOGW("%s: no group for stream %s, bailing out", __func__, toString(stream).c_str()); return NO_ERROR; } ALOGV("%s: stream %s attributes=%s, index %d , device 0x%X", __func__, toString(stream).c_str(), toString(attributes).c_str(), index, device); return setVolumeIndexForAttributes(attributes, index, device); } status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream, int *index, audio_devices_t device) { // if device is AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME, return volume for device selected for this // stream by the engine. DeviceTypeSet deviceTypes = {device}; if (device == AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) { deviceTypes = mEngine->getOutputDevicesForStream( stream, true /*fromCache*/).types(); } return getVolumeIndex(getVolumeCurves(stream), *index, deviceTypes); } status_t AudioPolicyManager::setVolumeIndexForAttributes(const audio_attributes_t &attributes, int index, audio_devices_t device) { // Get Volume group matching the Audio Attributes auto group = mEngine->getVolumeGroupForAttributes(attributes); if (group == VOLUME_GROUP_NONE) { ALOGD("%s: no group matching with %s", __FUNCTION__, toString(attributes).c_str()); return BAD_VALUE; } ALOGV("%s: group %d matching with %s index %d", __FUNCTION__, group, toString(attributes).c_str(), index); if (mEngine->getStreamTypeForAttributes(attributes) == AUDIO_STREAM_PATCH) { ALOGV("%s: cannot change volume for PATCH stream, attrs: %s", __FUNCTION__, toString(attributes).c_str()); return NO_ERROR; } status_t status = NO_ERROR; IVolumeCurves &curves = getVolumeCurves(attributes); VolumeSource vs = toVolumeSource(group); // AUDIO_STREAM_BLUETOOTH_SCO is only used for volume control so we remap // to AUDIO_STREAM_VOICE_CALL to match with relevant playback activity VolumeSource activityVs = (vs == toVolumeSource(AUDIO_STREAM_BLUETOOTH_SCO, false)) ? toVolumeSource(AUDIO_STREAM_VOICE_CALL, false) : vs; product_strategy_t strategy = mEngine->getProductStrategyForAttributes(attributes); status = setVolumeCurveIndex(index, device, curves); if (status != NO_ERROR) { ALOGE("%s failed to set curve index for group %d device 0x%X", __func__, group, device); return status; } DeviceTypeSet curSrcDevices; auto curCurvAttrs = curves.getAttributes(); if (!curCurvAttrs.empty() && curCurvAttrs.front() != defaultAttr) { auto attr = curCurvAttrs.front(); curSrcDevices = mEngine->getOutputDevicesForAttributes(attr, nullptr, false).types(); } else if (!curves.getStreamTypes().empty()) { auto stream = curves.getStreamTypes().front(); curSrcDevices = mEngine->getOutputDevicesForStream(stream, false).types(); } else { ALOGE("%s: Invalid src %d: no valid attributes nor stream",__func__, vs); return BAD_VALUE; } audio_devices_t curSrcDevice = Volume::getDeviceForVolume(curSrcDevices); resetDeviceTypes(curSrcDevices, curSrcDevice); // update volume on all outputs and streams matching the following: // - The requested stream (or a stream matching for volume control) is active on the output // - The device (or devices) selected by the engine for this stream includes // the requested device // - For non default requested device, currently selected device on the output is either the // requested device or one of the devices selected by the engine for this stream // - For default requested device (AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME), apply volume only if // no specific device volume value exists for currently selected device. // - Only apply the volume if the requested device is the desired device for volume control. for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); DeviceTypeSet curDevices = desc->devices().types(); if (curDevices.erase(AUDIO_DEVICE_OUT_SPEAKER_SAFE)) { curDevices.insert(AUDIO_DEVICE_OUT_SPEAKER); } if (!(desc->isActive(activityVs) || isInCallOrScreening())) { continue; } if (device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME && curDevices.find(device) == curDevices.end()) { continue; } bool applyVolume = false; if (device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) { curSrcDevices.insert(device); applyVolume = (curSrcDevices.find( Volume::getDeviceForVolume(curDevices)) != curSrcDevices.end()) && Volume::getDeviceForVolume(curSrcDevices) == device; } else { applyVolume = !curves.hasVolumeIndexForDevice(curSrcDevice); } if (!applyVolume) { continue; // next output } // Inter / intra volume group priority management: Loop on strategies arranged by priority // If a higher priority strategy is active, and the output is routed to a device with a // HW Gain management, do not change the volume if (desc->useHwGain()) { applyVolume = false; // If the volume source is active with higher priority source, ensure at least Sw Muted desc->setSwMute((index == 0), vs, curves.getStreamTypes(), curDevices, 0 /*delayMs*/); for (const auto &productStrategy : mEngine->getOrderedProductStrategies()) { auto activeClients = desc->clientsList(true /*activeOnly*/, productStrategy, false /*preferredDevice*/); if (activeClients.empty()) { continue; } bool isPreempted = false; bool isHigherPriority = productStrategy < strategy; for (const auto &client : activeClients) { if (isHigherPriority && (client->volumeSource() != activityVs)) { ALOGV("%s: Strategy=%d (\nrequester:\n" " group %d, volumeGroup=%d attributes=%s)\n" " higher priority source active:\n" " volumeGroup=%d attributes=%s) \n" " on output %zu, bailing out", __func__, productStrategy, group, group, toString(attributes).c_str(), client->volumeSource(), toString(client->attributes()).c_str(), i); applyVolume = false; isPreempted = true; break; } // However, continue for loop to ensure no higher prio clients running on output if (client->volumeSource() == activityVs) { applyVolume = true; } } if (isPreempted || applyVolume) { break; } } if (!applyVolume) { continue; // next output } } //FIXME: workaround for truncated touch sounds // delayed volume change for system stream to be removed when the problem is // handled by system UI status_t volStatus = checkAndSetVolume( curves, vs, index, desc, curDevices, ((vs == toVolumeSource(AUDIO_STREAM_SYSTEM, false))? TOUCH_SOUND_FIXED_DELAY_MS : 0)); if (volStatus != NO_ERROR) { status = volStatus; } } // update voice volume if the an active call route exists if (mCallRxSourceClient != nullptr && mCallRxSourceClient->isConnected() && (curSrcDevices.find( Volume::getDeviceForVolume({mCallRxSourceClient->sinkDevice()->type()})) != curSrcDevices.end())) { bool isVoiceVolSrc; bool isBtScoVolSrc; if (isVolumeConsistentForCalls(vs, {mCallRxSourceClient->sinkDevice()->type()}, isVoiceVolSrc, isBtScoVolSrc, __func__) && (isVoiceVolSrc || isBtScoVolSrc)) { bool voiceVolumeManagedByHost = isVoiceVolSrc && !audio_is_ble_out_device(mCallRxSourceClient->sinkDevice()->type()); setVoiceVolume(index, curves, voiceVolumeManagedByHost, 0); } } mpClientInterface->onAudioVolumeGroupChanged(group, 0 /*flags*/); return status; } status_t AudioPolicyManager::setVolumeCurveIndex(int index, audio_devices_t device, IVolumeCurves &volumeCurves) { // VOICE_CALL stream has minVolumeIndex > 0 but can be muted directly by an // app that has MODIFY_PHONE_STATE permission. bool hasVoice = hasVoiceStream(volumeCurves.getStreamTypes()); if (((index < volumeCurves.getVolumeIndexMin()) && !(hasVoice && index == 0)) || (index > volumeCurves.getVolumeIndexMax())) { ALOGE("%s: wrong index %d min=%d max=%d, device 0x%X", __FUNCTION__, index, volumeCurves.getVolumeIndexMin(), volumeCurves.getVolumeIndexMax(), device); return BAD_VALUE; } if (!audio_is_output_device(device)) { return BAD_VALUE; } // Force max volume if stream cannot be muted if (!volumeCurves.canBeMuted()) index = volumeCurves.getVolumeIndexMax(); ALOGV("%s device %08x, index %d", __FUNCTION__ , device, index); volumeCurves.addCurrentVolumeIndex(device, index); return NO_ERROR; } status_t AudioPolicyManager::getVolumeIndexForAttributes(const audio_attributes_t &attr, int &index, audio_devices_t device) { // if device is AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME, return volume for device selected for this // stream by the engine. DeviceTypeSet deviceTypes = {device}; if (device == AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) { deviceTypes = mEngine->getOutputDevicesForAttributes( attr, nullptr, true /*fromCache*/).types(); } return getVolumeIndex(getVolumeCurves(attr), index, deviceTypes); } status_t AudioPolicyManager::getVolumeIndex(const IVolumeCurves &curves, int &index, const DeviceTypeSet& deviceTypes) const { if (!isSingleDeviceType(deviceTypes, audio_is_output_device)) { return BAD_VALUE; } index = curves.getVolumeIndex(deviceTypes); ALOGV("%s: device %s index %d", __FUNCTION__, dumpDeviceTypes(deviceTypes).c_str(), index); return NO_ERROR; } status_t AudioPolicyManager::getMinVolumeIndexForAttributes(const audio_attributes_t &attr, int &index) { index = getVolumeCurves(attr).getVolumeIndexMin(); return NO_ERROR; } status_t AudioPolicyManager::getMaxVolumeIndexForAttributes(const audio_attributes_t &attr, int &index) { index = getVolumeCurves(attr).getVolumeIndexMax(); return NO_ERROR; } audio_io_handle_t AudioPolicyManager::selectOutputForMusicEffects() { // select one output among several suitable for global effects. // The priority is as follows: // 1: An offloaded output. If the effect ends up not being offloadable, // AudioFlinger will invalidate the track and the offloaded output // will be closed causing the effect to be moved to a PCM output. // 2: A deep buffer output // 3: The primary output // 4: the first output in the list DeviceVector devices = mEngine->getOutputDevicesForAttributes( attributes_initializer(AUDIO_USAGE_MEDIA), nullptr, false /*fromCache*/); SortedVector outputs = getOutputsForDevices(devices, mOutputs); if (outputs.size() == 0) { return AUDIO_IO_HANDLE_NONE; } audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; bool activeOnly = true; while (output == AUDIO_IO_HANDLE_NONE) { audio_io_handle_t outputOffloaded = AUDIO_IO_HANDLE_NONE; audio_io_handle_t outputDeepBuffer = AUDIO_IO_HANDLE_NONE; audio_io_handle_t outputPrimary = AUDIO_IO_HANDLE_NONE; for (audio_io_handle_t output : outputs) { sp desc = mOutputs.valueFor(output); if (activeOnly && !desc->isActive(toVolumeSource(AUDIO_STREAM_MUSIC))) { continue; } ALOGV("selectOutputForMusicEffects activeOnly %d output %d flags 0x%08x", activeOnly, output, desc->mFlags); if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { outputOffloaded = output; } if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { outputDeepBuffer = output; } if ((desc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) != 0) { outputPrimary = output; } } if (outputOffloaded != AUDIO_IO_HANDLE_NONE) { output = outputOffloaded; } else if (outputDeepBuffer != AUDIO_IO_HANDLE_NONE) { output = outputDeepBuffer; } else if (outputPrimary != AUDIO_IO_HANDLE_NONE) { output = outputPrimary; } else { output = outputs[0]; } activeOnly = false; } if (output != mMusicEffectOutput) { mEffects.moveEffects(AUDIO_SESSION_OUTPUT_MIX, mMusicEffectOutput, output, mpClientInterface); mMusicEffectOutput = output; } ALOGV("selectOutputForMusicEffects selected output %d", output); return output; } audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc __unused) { return selectOutputForMusicEffects(); } status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc, audio_io_handle_t io, product_strategy_t strategy, int session, int id) { if (session != AUDIO_SESSION_DEVICE && io != AUDIO_IO_HANDLE_NONE) { ssize_t index = mOutputs.indexOfKey(io); if (index < 0) { index = mInputs.indexOfKey(io); if (index < 0) { ALOGW("registerEffect() unknown io %d", io); return INVALID_OPERATION; } } } bool isMusicEffect = (session != AUDIO_SESSION_OUTPUT_STAGE) && ((strategy == streamToStrategy(AUDIO_STREAM_MUSIC) || strategy == PRODUCT_STRATEGY_NONE)); return mEffects.registerEffect(desc, io, session, id, isMusicEffect); } status_t AudioPolicyManager::unregisterEffect(int id) { if (mEffects.getEffect(id) == nullptr) { return INVALID_OPERATION; } if (mEffects.isEffectEnabled(id)) { ALOGW("%s effect %d enabled", __FUNCTION__, id); setEffectEnabled(id, false); } return mEffects.unregisterEffect(id); } status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled) { sp effect = mEffects.getEffect(id); if (effect == nullptr) { return INVALID_OPERATION; } status_t status = mEffects.setEffectEnabled(id, enabled); if (status == NO_ERROR) { mInputs.trackEffectEnabled(effect, enabled); } return status; } status_t AudioPolicyManager::moveEffectsToIo(const std::vector& ids, audio_io_handle_t io) { mEffects.moveEffects(ids, io); return NO_ERROR; } bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const { auto vs = toVolumeSource(stream, false); return vs != VOLUME_SOURCE_NONE ? mOutputs.isActive(vs, inPastMs) : false; } bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const { auto vs = toVolumeSource(stream, false); return vs != VOLUME_SOURCE_NONE ? mOutputs.isActiveRemotely(vs, inPastMs) : false; } bool AudioPolicyManager::isSourceActive(audio_source_t source) const { for (size_t i = 0; i < mInputs.size(); i++) { const sp inputDescriptor = mInputs.valueAt(i); if (inputDescriptor->isSourceActive(source)) { return true; } } return false; } // Register a list of custom mixes with their attributes and format. // When a mix is registered, corresponding input and output profiles are // added to the remote submix hw module. The profile contains only the // parameters (sampling rate, format...) specified by the mix. // The corresponding input remote submix device is also connected. // // When a remote submix device is connected, the address is checked to select the // appropriate profile and the corresponding input or output stream is opened. // // When capture starts, getInputForAttr() will: // - 1 look for a mix matching the address passed in attribtutes tags if any // - 2 if none found, getDeviceForInputSource() will: // - 2.1 look for a mix matching the attributes source // - 2.2 if none found, default to device selection by policy rules // At this time, the corresponding output remote submix device is also connected // and active playback use cases can be transferred to this mix if needed when reconnecting // after AudioTracks are invalidated // // When playback starts, getOutputForAttr() will: // - 1 look for a mix matching the address passed in attribtutes tags if any // - 2 if none found, look for a mix matching the attributes usage // - 3 if none found, default to device and output selection by policy rules. status_t AudioPolicyManager::registerPolicyMixes(const Vector& mixes) { ALOGV("registerPolicyMixes() %zu mix(es)", mixes.size()); status_t res = NO_ERROR; bool checkOutputs = false; sp rSubmixModule; Vector registeredMixes; // examine each mix's route type for (size_t i = 0; i < mixes.size(); i++) { AudioMix mix = mixes[i]; // Only capture of playback is allowed in LOOP_BACK & RENDER mode if (is_mix_loopback_render(mix.mRouteFlags) && mix.mMixType != MIX_TYPE_PLAYERS) { ALOGE("Unsupported Policy Mix %zu of %zu: " "Only capture of playback is allowed in LOOP_BACK & RENDER mode", i, mixes.size()); res = INVALID_OPERATION; break; } // LOOP_BACK and LOOP_BACK | RENDER have the same remote submix backend and are handled // in the same way. if ((mix.mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) { ALOGV("registerPolicyMixes() mix %zu of %zu is LOOP_BACK %d", i, mixes.size(), mix.mRouteFlags); if (rSubmixModule == 0) { rSubmixModule = mHwModules.getModuleFromName( AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX); if (rSubmixModule == 0) { ALOGE("Unable to find audio module for submix, aborting mix %zu registration", i); res = INVALID_OPERATION; break; } } String8 address = mix.mDeviceAddress; audio_devices_t deviceTypeToMakeAvailable; if (mix.mMixType == MIX_TYPE_PLAYERS) { mix.mDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; deviceTypeToMakeAvailable = AUDIO_DEVICE_IN_REMOTE_SUBMIX; } else { mix.mDeviceType = AUDIO_DEVICE_IN_REMOTE_SUBMIX; deviceTypeToMakeAvailable = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; } if (mPolicyMixes.registerMix(mix, 0 /*output desc*/) != NO_ERROR) { ALOGE("Error registering mix %zu for address %s", i, address.c_str()); res = INVALID_OPERATION; break; } audio_config_t outputConfig = mix.mFormat; audio_config_t inputConfig = mix.mFormat; // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL // in stereo and let audio flinger do the channel conversion if needed. outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO; inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO; rSubmixModule->addOutputProfile(address.c_str(), &outputConfig, AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address, audio_is_linear_pcm(outputConfig.format) ? AUDIO_OUTPUT_FLAG_NONE : AUDIO_OUTPUT_FLAG_DIRECT); rSubmixModule->addInputProfile(address.c_str(), &inputConfig, AUDIO_DEVICE_IN_REMOTE_SUBMIX, address, audio_is_linear_pcm(inputConfig.format) ? AUDIO_INPUT_FLAG_NONE : AUDIO_INPUT_FLAG_DIRECT); if ((res = setDeviceConnectionStateInt(deviceTypeToMakeAvailable, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address.c_str(), "remote-submix", AUDIO_FORMAT_DEFAULT)) != NO_ERROR) { ALOGE("Failed to set remote submix device available, type %u, address %s", mix.mDeviceType, address.c_str()); break; } } else if ((mix.mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) { String8 address = mix.mDeviceAddress; audio_devices_t type = mix.mDeviceType; ALOGV(" registerPolicyMixes() mix %zu of %zu is RENDER, dev=0x%X addr=%s", i, mixes.size(), type, address.c_str()); sp device = mHwModules.getDeviceDescriptor( mix.mDeviceType, mix.mDeviceAddress, String8(), AUDIO_FORMAT_DEFAULT); if (device == nullptr) { res = INVALID_OPERATION; break; } bool foundOutput = false; // First try to find an already opened output supporting the device for (size_t j = 0 ; j < mOutputs.size() && !foundOutput && res == NO_ERROR; j++) { sp desc = mOutputs.valueAt(j); if (!desc->isDuplicated() && desc->supportedDevices().contains(device)) { if (mPolicyMixes.registerMix(mix, desc) != NO_ERROR) { ALOGE("Could not register mix RENDER, dev=0x%X addr=%s", type, address.c_str()); res = INVALID_OPERATION; } else { foundOutput = true; } } } // If no output found, try to find a direct output profile supporting the device for (size_t i = 0; i < mHwModules.size() && !foundOutput && res == NO_ERROR; i++) { sp module = mHwModules[i]; for (size_t j = 0; j < module->getOutputProfiles().size() && !foundOutput && res == NO_ERROR; j++) { sp profile = module->getOutputProfiles()[j]; if (profile->isDirectOutput() && profile->supportsDevice(device)) { if (mPolicyMixes.registerMix(mix, nullptr) != NO_ERROR) { ALOGE("Could not register mix RENDER, dev=0x%X addr=%s", type, address.c_str()); res = INVALID_OPERATION; } else { foundOutput = true; } } } } if (res != NO_ERROR) { ALOGE(" Error registering mix %zu for device 0x%X addr %s", i, type, address.c_str()); res = INVALID_OPERATION; break; } else if (!foundOutput) { ALOGE(" Output not found for mix %zu for device 0x%X addr %s", i, type, address.c_str()); res = INVALID_OPERATION; break; } else { checkOutputs = true; registeredMixes.add(mix); } } } if (res != NO_ERROR) { if (audio_flags::audio_mix_ownership()) { // Only unregister mixes that were actually registered to not accidentally unregister // mixes that already existed previously. unregisterPolicyMixes(registeredMixes); registeredMixes.clear(); } else { unregisterPolicyMixes(mixes); } } else if (checkOutputs) { checkForDeviceAndOutputChanges(); updateCallAndOutputRouting(); } return res; } status_t AudioPolicyManager::unregisterPolicyMixes(Vector mixes) { ALOGV("unregisterPolicyMixes() num mixes %zu", mixes.size()); status_t res = NO_ERROR; bool checkOutputs = false; sp rSubmixModule; // examine each mix's route type for (const auto& mix : mixes) { if ((mix.mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) { if (rSubmixModule == 0) { rSubmixModule = mHwModules.getModuleFromName( AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX); if (rSubmixModule == 0) { res = INVALID_OPERATION; continue; } } String8 address = mix.mDeviceAddress; if (mPolicyMixes.unregisterMix(mix) != NO_ERROR) { res = INVALID_OPERATION; continue; } for (auto device: {AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_DEVICE_OUT_REMOTE_SUBMIX}) { if (getDeviceConnectionState(device, address.c_str()) == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { status_t currentRes = setDeviceConnectionStateInt(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address.c_str(), "remote-submix", AUDIO_FORMAT_DEFAULT); if (!audio_flags::audio_mix_ownership()) { res = currentRes; } if (currentRes != OK) { ALOGE("Error making RemoteSubmix device unavailable for mix " "with type %d, address %s", device, address.c_str()); res = INVALID_OPERATION; } } } rSubmixModule->removeOutputProfile(address.c_str()); rSubmixModule->removeInputProfile(address.c_str()); } else if ((mix.mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) { if (mPolicyMixes.unregisterMix(mix) != NO_ERROR) { res = INVALID_OPERATION; continue; } else { checkOutputs = true; } } } if (res == NO_ERROR && checkOutputs) { checkForDeviceAndOutputChanges(); updateCallAndOutputRouting(); } return res; } status_t AudioPolicyManager::getRegisteredPolicyMixes(std::vector& _aidl_return) { if (!audio_flags::audio_mix_test_api()) { return INVALID_OPERATION; } _aidl_return.clear(); _aidl_return.reserve(mPolicyMixes.size()); for (const auto &policyMix: mPolicyMixes) { _aidl_return.emplace_back(policyMix->mCriteria, policyMix->mMixType, policyMix->mFormat, policyMix->mRouteFlags, policyMix->mDeviceAddress, policyMix->mCbFlags); _aidl_return.back().mDeviceType = policyMix->mDeviceType; _aidl_return.back().mToken = policyMix->mToken; _aidl_return.back().mVirtualDeviceId = policyMix->mVirtualDeviceId; } ALOGVV("%s() returning %zu registered mixes", __func__, _aidl_return.size()); return OK; } status_t AudioPolicyManager::updatePolicyMix( const AudioMix& mix, const std::vector& updatedCriteria) { status_t res = mPolicyMixes.updateMix(mix, updatedCriteria); if (res == NO_ERROR) { checkForDeviceAndOutputChanges(); updateCallAndOutputRouting(); } return res; } void AudioPolicyManager::dumpManualSurroundFormats(String8 *dst) const { size_t i = 0; constexpr size_t audioFormatPrefixLen = sizeof("AUDIO_FORMAT_"); for (const auto& fmt : mManualSurroundFormats) { if (i++ != 0) dst->append(", "); std::string sfmt; FormatConverter::toString(fmt, sfmt); dst->append(sfmt.size() >= audioFormatPrefixLen ? sfmt.c_str() + audioFormatPrefixLen - 1 : sfmt.c_str()); } } // Returns true if all devices types match the predicate and are supported by one HW module bool AudioPolicyManager::areAllDevicesSupported( const AudioDeviceTypeAddrVector& devices, std::function predicate, const char *context, bool matchAddress) { for (size_t i = 0; i < devices.size(); i++) { sp devDesc = mHwModules.getDeviceDescriptor( devices[i].mType, devices[i].getAddress(), String8(), AUDIO_FORMAT_DEFAULT, false /*allowToCreate*/, matchAddress); if (devDesc == nullptr || (predicate != nullptr && !predicate(devices[i].mType))) { ALOGE("%s: device type %#x address %s not supported or not match predicate", context, devices[i].mType, devices[i].getAddress()); return false; } } return true; } void AudioPolicyManager::changeOutputDevicesMuteState( const AudioDeviceTypeAddrVector& devices) { ALOGVV("%s() num devices %zu", __func__, devices.size()); std::vector> outputs = getSoftwareOutputsForDevices(devices); for (size_t i = 0; i < outputs.size(); i++) { sp outputDesc = outputs[i]; DeviceVector prevDevices = outputDesc->devices(); checkDeviceMuteStrategies(outputDesc, prevDevices, 0 /* delayMs */); } } std::vector> AudioPolicyManager::getSoftwareOutputsForDevices( const AudioDeviceTypeAddrVector& devices) const { std::vector> outputs; DeviceVector deviceDescriptors; for (size_t j = 0; j < devices.size(); j++) { sp desc = mHwModules.getDeviceDescriptor( devices[j].mType, devices[j].getAddress(), String8(), AUDIO_FORMAT_DEFAULT); if (desc == nullptr || !audio_is_output_device(devices[j].mType)) { ALOGE("%s: device type %#x address %s not supported or not an output device", __func__, devices[j].mType, devices[j].getAddress()); continue; } deviceDescriptors.add(desc); } for (size_t i = 0; i < mOutputs.size(); i++) { if (!mOutputs.valueAt(i)->supportsAtLeastOne(deviceDescriptors)) { continue; } outputs.push_back(mOutputs.valueAt(i)); } return outputs; } status_t AudioPolicyManager::setUidDeviceAffinities(uid_t uid, const AudioDeviceTypeAddrVector& devices) { ALOGV("%s() uid=%d num devices %zu", __FUNCTION__, uid, devices.size()); if (!areAllDevicesSupported(devices, audio_is_output_device, __func__)) { return BAD_VALUE; } status_t res = mPolicyMixes.setUidDeviceAffinities(uid, devices); if (res != NO_ERROR) { ALOGE("%s() Could not set all device affinities for uid = %d", __FUNCTION__, uid); return res; } checkForDeviceAndOutputChanges(); updateCallAndOutputRouting(); return NO_ERROR; } status_t AudioPolicyManager::removeUidDeviceAffinities(uid_t uid) { ALOGV("%s() uid=%d", __FUNCTION__, uid); status_t res = mPolicyMixes.removeUidDeviceAffinities(uid); if (res != NO_ERROR) { ALOGE("%s() Could not remove all device affinities for uid = %d", __FUNCTION__, uid); return INVALID_OPERATION; } checkForDeviceAndOutputChanges(); updateCallAndOutputRouting(); return res; } status_t AudioPolicyManager::setDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role, const AudioDeviceTypeAddrVector &devices) { ALOGV("%s() strategy=%d role=%d %s", __func__, strategy, role, dumpAudioDeviceTypeAddrVector(devices).c_str()); if (!areAllDevicesSupported(devices, audio_is_output_device, __func__)) { return BAD_VALUE; } status_t status = mEngine->setDevicesRoleForStrategy(strategy, role, devices); if (status != NO_ERROR) { ALOGW("Engine could not set preferred devices %s for strategy %d role %d", dumpAudioDeviceTypeAddrVector(devices).c_str(), strategy, role); return status; } checkForDeviceAndOutputChanges(); bool forceVolumeReeval = false; // FIXME: workaround for truncated touch sounds // to be removed when the problem is handled by system UI uint32_t delayMs = 0; if (strategy == mCommunnicationStrategy) { forceVolumeReeval = true; delayMs = TOUCH_SOUND_FIXED_DELAY_MS; updateInputRouting(); } updateCallAndOutputRouting(forceVolumeReeval, delayMs); return NO_ERROR; } void AudioPolicyManager::updateCallAndOutputRouting(bool forceVolumeReeval, uint32_t delayMs, bool skipDelays) { uint32_t waitMs = 0; bool wasLeUnicastActive = isLeUnicastActive(); if (updateCallRouting(true /*fromCache*/, delayMs, &waitMs) == NO_ERROR) { // Only apply special touch sound delay once delayMs = 0; } std::map outputsToReopen; for (size_t i = 0; i < mOutputs.size(); i++) { sp outputDesc = mOutputs.valueAt(i); DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/); if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput && !isTelephonyRxOrTx(outputDesc))) { // As done in setDeviceConnectionState, we could also fix default device issue by // preventing the force re-routing in case of default dev that distinguishes on address. // Let's give back to engine full device choice decision however. bool forceRouting = !newDevices.isEmpty(); if (outputDesc->mPreferredAttrInfo != nullptr && newDevices != outputDesc->devices()) { // If the device is using preferred mixer attributes, the output need to reopen // with default configuration when the new selected devices are different from // current routing devices. outputsToReopen.emplace(mOutputs.keyAt(i), newDevices); continue; } waitMs = setOutputDevices(__func__, outputDesc, newDevices, forceRouting, delayMs, nullptr, !skipDelays /*requiresMuteCheck*/, !forceRouting /*requiresVolumeCheck*/, skipDelays); // Only apply special touch sound delay once delayMs = 0; } if (forceVolumeReeval && !newDevices.isEmpty()) { applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true); } } reopenOutputsWithDevices(outputsToReopen); checkLeBroadcastRoutes(wasLeUnicastActive, nullptr, delayMs); } void AudioPolicyManager::updateInputRouting() { for (const auto& activeDesc : mInputs.getActiveInputs()) { // Skip for hotword recording as the input device switch // is handled within sound trigger HAL if (activeDesc->isSoundTrigger() && activeDesc->source() == AUDIO_SOURCE_HOTWORD) { continue; } auto newDevice = getNewInputDevice(activeDesc); // Force new input selection if the new device can not be reached via current input if (activeDesc->mProfile->getSupportedDevices().contains(newDevice)) { setInputDevice(activeDesc->mIoHandle, newDevice); } else { closeInput(activeDesc->mIoHandle); } } } status_t AudioPolicyManager::removeDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role, const AudioDeviceTypeAddrVector &devices) { ALOGV("%s() strategy=%d role=%d %s", __func__, strategy, role, dumpAudioDeviceTypeAddrVector(devices).c_str()); if (!areAllDevicesSupported( devices, audio_is_output_device, __func__, /*matchAddress*/false)) { return BAD_VALUE; } status_t status = mEngine->removeDevicesRoleForStrategy(strategy, role, devices); if (status != NO_ERROR) { ALOGW("Engine could not remove devices %s for strategy %d role %d", dumpAudioDeviceTypeAddrVector(devices).c_str(), strategy, role); return status; } checkForDeviceAndOutputChanges(); bool forceVolumeReeval = false; // TODO(b/263479999): workaround for truncated touch sounds // to be removed when the problem is handled by system UI uint32_t delayMs = 0; if (strategy == mCommunnicationStrategy) { forceVolumeReeval = true; delayMs = TOUCH_SOUND_FIXED_DELAY_MS; updateInputRouting(); } updateCallAndOutputRouting(forceVolumeReeval, delayMs); return NO_ERROR; } status_t AudioPolicyManager::clearDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role) { ALOGV("%s() strategy=%d role=%d", __func__, strategy, role); status_t status = mEngine->clearDevicesRoleForStrategy(strategy, role); if (status != NO_ERROR) { ALOGW_IF(status != NAME_NOT_FOUND, "Engine could not remove device role for strategy %d status %d", strategy, status); return status; } checkForDeviceAndOutputChanges(); bool forceVolumeReeval = false; // FIXME: workaround for truncated touch sounds // to be removed when the problem is handled by system UI uint32_t delayMs = 0; if (strategy == mCommunnicationStrategy) { forceVolumeReeval = true; delayMs = TOUCH_SOUND_FIXED_DELAY_MS; updateInputRouting(); } updateCallAndOutputRouting(forceVolumeReeval, delayMs); return NO_ERROR; } status_t AudioPolicyManager::getDevicesForRoleAndStrategy(product_strategy_t strategy, device_role_t role, AudioDeviceTypeAddrVector &devices) { return mEngine->getDevicesForRoleAndStrategy(strategy, role, devices); } status_t AudioPolicyManager::setDevicesRoleForCapturePreset( audio_source_t audioSource, device_role_t role, const AudioDeviceTypeAddrVector &devices) { ALOGV("%s() audioSource=%d role=%d %s", __func__, audioSource, role, dumpAudioDeviceTypeAddrVector(devices).c_str()); if (!areAllDevicesSupported(devices, audio_call_is_input_device, __func__)) { return BAD_VALUE; } status_t status = mEngine->setDevicesRoleForCapturePreset(audioSource, role, devices); ALOGW_IF(status != NO_ERROR, "Engine could not set preferred devices %s for audio source %d role %d", dumpAudioDeviceTypeAddrVector(devices).c_str(), audioSource, role); return status; } status_t AudioPolicyManager::addDevicesRoleForCapturePreset( audio_source_t audioSource, device_role_t role, const AudioDeviceTypeAddrVector &devices) { ALOGV("%s() audioSource=%d role=%d %s", __func__, audioSource, role, dumpAudioDeviceTypeAddrVector(devices).c_str()); if (!areAllDevicesSupported(devices, audio_call_is_input_device, __func__)) { return BAD_VALUE; } status_t status = mEngine->addDevicesRoleForCapturePreset(audioSource, role, devices); ALOGW_IF(status != NO_ERROR, "Engine could not add preferred devices %s for audio source %d role %d", dumpAudioDeviceTypeAddrVector(devices).c_str(), audioSource, role); updateInputRouting(); return status; } status_t AudioPolicyManager::removeDevicesRoleForCapturePreset( audio_source_t audioSource, device_role_t role, const AudioDeviceTypeAddrVector& devices) { ALOGV("%s() audioSource=%d role=%d devices=%s", __func__, audioSource, role, dumpAudioDeviceTypeAddrVector(devices).c_str()); if (!areAllDevicesSupported( devices, audio_call_is_input_device, __func__, /*matchAddress*/false)) { return BAD_VALUE; } status_t status = mEngine->removeDevicesRoleForCapturePreset( audioSource, role, devices); ALOGW_IF(status != NO_ERROR && status != NAME_NOT_FOUND, "Engine could not remove devices role (%d) for capture preset %d", role, audioSource); if (status == NO_ERROR) { updateInputRouting(); } return status; } status_t AudioPolicyManager::clearDevicesRoleForCapturePreset(audio_source_t audioSource, device_role_t role) { ALOGV("%s() audioSource=%d role=%d", __func__, audioSource, role); status_t status = mEngine->clearDevicesRoleForCapturePreset(audioSource, role); ALOGW_IF(status != NO_ERROR && status != NAME_NOT_FOUND, "Engine could not clear devices role (%d) for capture preset %d", role, audioSource); if (status == NO_ERROR) { updateInputRouting(); } return status; } status_t AudioPolicyManager::getDevicesForRoleAndCapturePreset( audio_source_t audioSource, device_role_t role, AudioDeviceTypeAddrVector &devices) { return mEngine->getDevicesForRoleAndCapturePreset(audioSource, role, devices); } status_t AudioPolicyManager::setUserIdDeviceAffinities(int userId, const AudioDeviceTypeAddrVector& devices) { ALOGV("%s() userId=%d num devices %zu", __func__, userId, devices.size()); if (!areAllDevicesSupported(devices, audio_is_output_device, __func__)) { return BAD_VALUE; } status_t status = mPolicyMixes.setUserIdDeviceAffinities(userId, devices); if (status != NO_ERROR) { ALOGE("%s() could not set device affinity for userId %d", __FUNCTION__, userId); return status; } // reevaluate outputs for all devices checkForDeviceAndOutputChanges(); changeOutputDevicesMuteState(devices); updateCallAndOutputRouting(false /* forceVolumeReeval */, 0 /* delayMs */, true /* skipDelays */); changeOutputDevicesMuteState(devices); return NO_ERROR; } status_t AudioPolicyManager::removeUserIdDeviceAffinities(int userId) { ALOGV("%s() userId=%d", __FUNCTION__, userId); AudioDeviceTypeAddrVector devices; mPolicyMixes.getDevicesForUserId(userId, devices); status_t status = mPolicyMixes.removeUserIdDeviceAffinities(userId); if (status != NO_ERROR) { ALOGE("%s() Could not remove all device affinities fo userId = %d", __FUNCTION__, userId); return status; } // reevaluate outputs for all devices checkForDeviceAndOutputChanges(); changeOutputDevicesMuteState(devices); updateCallAndOutputRouting(false /* forceVolumeReeval */, 0 /* delayMs */, true /* skipDelays */); changeOutputDevicesMuteState(devices); return NO_ERROR; } void AudioPolicyManager::dump(String8 *dst) const { dst->appendFormat("\nAudioPolicyManager Dump: %p\n", this); dst->appendFormat(" Primary Output I/O handle: %d\n", hasPrimaryOutput() ? mPrimaryOutput->mIoHandle : AUDIO_IO_HANDLE_NONE); std::string stateLiteral; AudioModeConverter::toString(mEngine->getPhoneState(), stateLiteral); dst->appendFormat(" Phone state: %s\n", stateLiteral.c_str()); const char* forceUses[AUDIO_POLICY_FORCE_USE_CNT] = { "communications", "media", "record", "dock", "system", "HDMI system audio", "encoded surround output", "vibrate ringing" }; for (audio_policy_force_use_t i = AUDIO_POLICY_FORCE_FOR_COMMUNICATION; i < AUDIO_POLICY_FORCE_USE_CNT; i = (audio_policy_force_use_t)((int)i + 1)) { audio_policy_forced_cfg_t forceUseValue = mEngine->getForceUse(i); dst->appendFormat(" Force use for %s: %d", forceUses[i], forceUseValue); if (i == AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND && forceUseValue == AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL) { dst->append(" (MANUAL: "); dumpManualSurroundFormats(dst); dst->append(")"); } dst->append("\n"); } dst->appendFormat(" TTS output %savailable\n", mTtsOutputAvailable ? "" : "not "); dst->appendFormat(" Master mono: %s\n", mMasterMono ? "on" : "off"); dst->appendFormat(" Communication Strategy id: %d\n", mCommunnicationStrategy); dst->appendFormat(" Config source: %s\n", mConfig->getSource().c_str()); dst->append("\n"); mAvailableOutputDevices.dump(dst, String8("Available output"), 1); dst->append("\n"); mAvailableInputDevices.dump(dst, String8("Available input"), 1); mHwModules.dump(dst); mOutputs.dump(dst); mInputs.dump(dst); mEffects.dump(dst, 1); mAudioPatches.dump(dst); mPolicyMixes.dump(dst); mAudioSources.dump(dst); dst->appendFormat(" AllowedCapturePolicies:\n"); for (auto& policy : mAllowedCapturePolicies) { dst->appendFormat(" - uid=%d flag_mask=%#x\n", policy.first, policy.second); } dst->appendFormat(" Preferred mixer audio configuration:\n"); for (const auto it : mPreferredMixerAttrInfos) { dst->appendFormat(" - device port id: %d\n", it.first); for (const auto preferredMixerInfoIt : it.second) { dst->appendFormat(" - strategy: %d; ", preferredMixerInfoIt.first); preferredMixerInfoIt.second->dump(dst); } } dst->appendFormat("\nPolicy Engine dump:\n"); mEngine->dump(dst); dst->appendFormat("\nAbsolute volume devices with driving streams:\n"); for (const auto it : mAbsoluteVolumeDrivingStreams) { dst->appendFormat(" - device type: %s, driving stream %d\n", dumpDeviceTypes({it.first}).c_str(), mEngine->getVolumeGroupForAttributes(it.second)); } } status_t AudioPolicyManager::dump(int fd) { String8 result; dump(&result); write(fd, result.c_str(), result.size()); return NO_ERROR; } status_t AudioPolicyManager::setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t capturePolicy) { mAllowedCapturePolicies[uid] = capturePolicy; return NO_ERROR; } // This function checks for the parameters which can be offloaded. // This can be enhanced depending on the capability of the DSP and policy // of the system. audio_offload_mode_t AudioPolicyManager::getOffloadSupport(const audio_offload_info_t& offloadInfo) { ALOGV("%s: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," " BitRate=%u, duration=%" PRId64 " us, has_video=%d", __func__, offloadInfo.sample_rate, offloadInfo.channel_mask, offloadInfo.format, offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, offloadInfo.has_video); if (!isOffloadPossible(offloadInfo)) { return AUDIO_OFFLOAD_NOT_SUPPORTED; } // See if there is a profile to support this. // AUDIO_DEVICE_NONE sp profile = getProfileForOutput(DeviceVector() /*ignore device */, offloadInfo.sample_rate, offloadInfo.format, offloadInfo.channel_mask, AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, true /* directOnly */); ALOGV("%s: profile %sfound%s", __func__, profile != nullptr ? "" : "NOT ", (profile != nullptr && (profile->getFlags() & AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD) != 0) ? ", supports gapless" : ""); if (profile == nullptr) { return AUDIO_OFFLOAD_NOT_SUPPORTED; } if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD) != 0) { return AUDIO_OFFLOAD_GAPLESS_SUPPORTED; } return AUDIO_OFFLOAD_SUPPORTED; } bool AudioPolicyManager::isDirectOutputSupported(const audio_config_base_t& config, const audio_attributes_t& attributes) { audio_output_flags_t output_flags = AUDIO_OUTPUT_FLAG_NONE; audio_flags_to_audio_output_flags(attributes.flags, &output_flags); DeviceVector outputDevices = mEngine->getOutputDevicesForAttributes(attributes); sp profile = getProfileForOutput(outputDevices, config.sample_rate, config.format, config.channel_mask, output_flags, true /* directOnly */); ALOGV("%s() profile %sfound with name: %s, " "sample rate: %u, format: 0x%x, channel_mask: 0x%x, output flags: 0x%x", __FUNCTION__, profile != 0 ? "" : "NOT ", (profile != 0 ? profile->getTagName().c_str() : "null"), config.sample_rate, config.format, config.channel_mask, output_flags); // also try the MSD module if compatible profile not found if (profile == nullptr) { profile = getMsdProfileForOutput(outputDevices, config.sample_rate, config.format, config.channel_mask, output_flags, true /* directOnly */); ALOGV("%s() MSD profile %sfound with name: %s, " "sample rate: %u, format: 0x%x, channel_mask: 0x%x, output flags: 0x%x", __FUNCTION__, profile != 0 ? "" : "NOT ", (profile != 0 ? profile->getTagName().c_str() : "null"), config.sample_rate, config.format, config.channel_mask, output_flags); } return (profile != nullptr); } bool AudioPolicyManager::isOffloadPossible(const audio_offload_info_t &offloadInfo, bool durationIgnored) { if (mMasterMono) { return false; // no offloading if mono is set. } // Check if offload has been disabled if (property_get_bool("audio.offload.disable", false /* default_value */)) { ALOGV("%s: offload disabled by audio.offload.disable", __func__); return false; } // Check if stream type is music, then only allow offload as of now. if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) { ALOGV("%s: stream_type != MUSIC, returning false", __func__); return false; } //TODO: enable audio offloading with video when ready const bool allowOffloadWithVideo = property_get_bool("audio.offload.video", false /* default_value */); if (offloadInfo.has_video && !allowOffloadWithVideo) { ALOGV("%s: has_video == true, returning false", __func__); return false; } //If duration is less than minimum value defined in property, return false const int min_duration_secs = property_get_int32( "audio.offload.min.duration.secs", -1 /* default_value */); if (!durationIgnored) { if (min_duration_secs >= 0) { if (offloadInfo.duration_us < min_duration_secs * 1000000LL) { ALOGV("%s: Offload denied by duration < audio.offload.min.duration.secs(=%d)", __func__, min_duration_secs); return false; } } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { ALOGV("%s: Offload denied by duration < default min(=%u)", __func__, OFFLOAD_DEFAULT_MIN_DURATION_SECS); return false; } } // Do not allow offloading if one non offloadable effect is enabled. This prevents from // creating an offloaded track and tearing it down immediately after start when audioflinger // detects there is an active non offloadable effect. // FIXME: We should check the audio session here but we do not have it in this context. // This may prevent offloading in rare situations where effects are left active by apps // in the background. if (mEffects.isNonOffloadableEffectEnabled()) { return false; } return true; } audio_direct_mode_t AudioPolicyManager::getDirectPlaybackSupport(const audio_attributes_t *attr, const audio_config_t *config) { audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER; offloadInfo.format = config->format; offloadInfo.sample_rate = config->sample_rate; offloadInfo.channel_mask = config->channel_mask; offloadInfo.stream_type = mEngine->getStreamTypeForAttributes(*attr); offloadInfo.has_video = false; offloadInfo.is_streaming = false; const bool offloadPossible = isOffloadPossible(offloadInfo, true /*durationIgnored*/); audio_direct_mode_t directMode = AUDIO_DIRECT_NOT_SUPPORTED; audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE; audio_flags_to_audio_output_flags(attr->flags, &flags); // only retain flags that will drive compressed offload or passthrough uint32_t relevantFlags = AUDIO_OUTPUT_FLAG_HW_AV_SYNC; if (offloadPossible) { relevantFlags |= AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD; } flags = (audio_output_flags_t)((flags & relevantFlags) | AUDIO_OUTPUT_FLAG_DIRECT); DeviceVector engineOutputDevices = mEngine->getOutputDevicesForAttributes(*attr); for (const auto& hwModule : mHwModules) { DeviceVector outputDevices = engineOutputDevices; // the MSD module checks for different conditions and output devices if (strcmp(hwModule->getName(), AUDIO_HARDWARE_MODULE_ID_MSD) == 0) { if (!msdHasPatchesToAllDevices(engineOutputDevices.toTypeAddrVector())) { continue; } outputDevices = getMsdAudioOutDevices(); } for (const auto& curProfile : hwModule->getOutputProfiles()) { if (curProfile->getCompatibilityScore(outputDevices, config->sample_rate, nullptr /*updatedSamplingRate*/, config->format, nullptr /*updatedFormat*/, config->channel_mask, nullptr /*updatedChannelMask*/, flags) == IOProfile::NO_MATCH) { continue; } // reject profiles not corresponding to a device currently available if (!mAvailableOutputDevices.containsAtLeastOne(curProfile->getSupportedDevices())) { continue; } if (offloadPossible && ((curProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != AUDIO_OUTPUT_FLAG_NONE)) { if ((directMode & AUDIO_DIRECT_OFFLOAD_GAPLESS_SUPPORTED) != AUDIO_DIRECT_NOT_SUPPORTED) { // Already reports offload gapless supported. No need to report offload support. continue; } if ((curProfile->getFlags() & AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD) != AUDIO_OUTPUT_FLAG_NONE) { // If offload gapless is reported, no need to report offload support. directMode = (audio_direct_mode_t) ((directMode & ~AUDIO_DIRECT_OFFLOAD_SUPPORTED) | AUDIO_DIRECT_OFFLOAD_GAPLESS_SUPPORTED); } else { directMode = (audio_direct_mode_t)(directMode | AUDIO_DIRECT_OFFLOAD_SUPPORTED); } } else { directMode = (audio_direct_mode_t) (directMode | AUDIO_DIRECT_BITSTREAM_SUPPORTED); } } } return directMode; } status_t AudioPolicyManager::getDirectProfilesForAttributes(const audio_attributes_t* attr, AudioProfileVector& audioProfilesVector) { if (mEffects.isNonOffloadableEffectEnabled()) { return OK; } DeviceVector devices; status_t status = getDevicesForAttributes(*attr, devices, false /* forVolume */); if (status != OK) { return status; } ALOGV("%s: found %zu output devices for attributes.", __func__, devices.size()); if (devices.empty()) { return OK; // no output devices for the attributes } return getProfilesForDevices(devices, audioProfilesVector, AUDIO_OUTPUT_FLAG_DIRECT /*flags*/, false /*isInput*/); } status_t AudioPolicyManager::getSupportedMixerAttributes( audio_port_handle_t portId, std::vector &mixerAttrs) { ALOGV("%s, portId=%d", __func__, portId); sp deviceDescriptor = mAvailableOutputDevices.getDeviceFromId(portId); if (deviceDescriptor == nullptr) { ALOGE("%s the requested device is currently unavailable", __func__); return BAD_VALUE; } if (!audio_is_usb_out_device(deviceDescriptor->type())) { ALOGE("%s the requested device(type=%#x) is not usb device", __func__, deviceDescriptor->type()); return BAD_VALUE; } for (const auto& hwModule : mHwModules) { for (const auto& curProfile : hwModule->getOutputProfiles()) { if (curProfile->supportsDevice(deviceDescriptor)) { curProfile->toSupportedMixerAttributes(&mixerAttrs); } } } return NO_ERROR; } status_t AudioPolicyManager::setPreferredMixerAttributes( const audio_attributes_t *attr, audio_port_handle_t portId, uid_t uid, const audio_mixer_attributes_t *mixerAttributes) { ALOGV("%s, attr=%s, mixerAttributes={format=%#x, channelMask=%#x, samplingRate=%u, " "mixerBehavior=%d}, uid=%d, portId=%u", __func__, toString(*attr).c_str(), mixerAttributes->config.format, mixerAttributes->config.channel_mask, mixerAttributes->config.sample_rate, mixerAttributes->mixer_behavior, uid, portId); if (attr->usage != AUDIO_USAGE_MEDIA) { ALOGE("%s failed, only media is allowed, the given usage is %d", __func__, attr->usage); return BAD_VALUE; } sp deviceDescriptor = mAvailableOutputDevices.getDeviceFromId(portId); if (deviceDescriptor == nullptr) { ALOGE("%s the requested device is currently unavailable", __func__); return BAD_VALUE; } if (!audio_is_usb_out_device(deviceDescriptor->type())) { ALOGE("%s(%d), type=%d, is not a usb output device", __func__, portId, deviceDescriptor->type()); return BAD_VALUE; } audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE; audio_flags_to_audio_output_flags(attr->flags, &flags); flags = (audio_output_flags_t) (flags | audio_output_flags_from_mixer_behavior(mixerAttributes->mixer_behavior)); sp profile = nullptr; DeviceVector devices(deviceDescriptor); for (const auto& hwModule : mHwModules) { for (const auto& curProfile : hwModule->getOutputProfiles()) { if (curProfile->hasDynamicAudioProfile() && curProfile->getCompatibilityScore( devices, mixerAttributes->config.sample_rate, nullptr /*updatedSamplingRate*/, mixerAttributes->config.format, nullptr /*updatedFormat*/, mixerAttributes->config.channel_mask, nullptr /*updatedChannelMask*/, flags, false /*exactMatchRequiredForInputFlags*/) != IOProfile::NO_MATCH) { profile = curProfile; break; } } } if (profile == nullptr) { ALOGE("%s, there is no compatible profile found", __func__); return BAD_VALUE; } sp mixerAttrInfo = sp::make( uid, portId, profile, flags, *mixerAttributes); const product_strategy_t strategy = mEngine->getProductStrategyForAttributes(*attr); mPreferredMixerAttrInfos[portId][strategy] = mixerAttrInfo; // If 1) there is any client from the preferred mixer configuration owner that is currently // active and matches the strategy and 2) current output is on the preferred device and the // mixer configuration doesn't match the preferred one, reopen output with preferred mixer // configuration. std::vector outputsToReopen; for (size_t i = 0; i < mOutputs.size(); i++) { const auto output = mOutputs.valueAt(i); if (output->mProfile == profile && output->devices().onlyContainsDevice(deviceDescriptor)) { if (output->isConfigurationMatched(mixerAttributes->config, flags)) { output->mPreferredAttrInfo = mixerAttrInfo; } else { for (const auto &client: output->getActiveClients()) { if (client->uid() == uid && client->strategy() == strategy) { client->setIsInvalid(); outputsToReopen.push_back(output->mIoHandle); } } } } } audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = mixerAttributes->config.sample_rate; config.channel_mask = mixerAttributes->config.channel_mask; config.format = mixerAttributes->config.format; for (const auto output : outputsToReopen) { sp desc = reopenOutput(mOutputs.valueFor(output), &config, flags, __func__); if (desc == nullptr) { ALOGE("%s, failed to reopen output with preferred mixer attributes", __func__); continue; } desc->mPreferredAttrInfo = mixerAttrInfo; } return NO_ERROR; } sp AudioPolicyManager::getPreferredMixerAttributesInfo( audio_port_handle_t devicePortId, product_strategy_t strategy, bool activeBitPerfectPreferred) { auto it = mPreferredMixerAttrInfos.find(devicePortId); if (it == mPreferredMixerAttrInfos.end()) { return nullptr; } if (activeBitPerfectPreferred) { for (auto [strategy, info] : it->second) { if (info->isBitPerfect() && info->getActiveClientCount() != 0) { return info; } } } auto strategyMatchedMixerAttrInfoIt = it->second.find(strategy); return strategyMatchedMixerAttrInfoIt == it->second.end() ? nullptr : strategyMatchedMixerAttrInfoIt->second; } status_t AudioPolicyManager::getPreferredMixerAttributes( const audio_attributes_t *attr, audio_port_handle_t portId, audio_mixer_attributes_t* mixerAttributes) { sp info = getPreferredMixerAttributesInfo( portId, mEngine->getProductStrategyForAttributes(*attr)); if (info == nullptr) { return NAME_NOT_FOUND; } *mixerAttributes = info->getMixerAttributes(); return NO_ERROR; } status_t AudioPolicyManager::clearPreferredMixerAttributes(const audio_attributes_t *attr, audio_port_handle_t portId, uid_t uid) { const product_strategy_t strategy = mEngine->getProductStrategyForAttributes(*attr); const auto preferredMixerAttrInfo = getPreferredMixerAttributesInfo(portId, strategy); if (preferredMixerAttrInfo == nullptr) { return NAME_NOT_FOUND; } if (preferredMixerAttrInfo->getUid() != uid) { ALOGE("%s, requested uid=%d, owned uid=%d", __func__, uid, preferredMixerAttrInfo->getUid()); return PERMISSION_DENIED; } mPreferredMixerAttrInfos[portId].erase(strategy); if (mPreferredMixerAttrInfos[portId].empty()) { mPreferredMixerAttrInfos.erase(portId); } // Reconfig existing output std::vector potentialOutputsToReopen; for (size_t i = 0; i < mOutputs.size(); i++) { if (mOutputs.valueAt(i)->mProfile == preferredMixerAttrInfo->getProfile()) { potentialOutputsToReopen.push_back(mOutputs.keyAt(i)); } } for (const auto output : potentialOutputsToReopen) { sp desc = mOutputs.valueFor(output); if (desc->isConfigurationMatched(preferredMixerAttrInfo->getConfigBase(), preferredMixerAttrInfo->getFlags())) { reopenOutput(desc, nullptr /*config*/, AUDIO_OUTPUT_FLAG_NONE, __func__); } } return NO_ERROR; } status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role, audio_port_type_t type, unsigned int *num_ports, struct audio_port_v7 *ports, unsigned int *generation) { if (num_ports == nullptr || (*num_ports != 0 && ports == nullptr) || generation == nullptr) { return BAD_VALUE; } ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports); if (ports == nullptr) { *num_ports = 0; } size_t portsWritten = 0; size_t portsMax = *num_ports; *num_ports = 0; if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) { // do not report devices with type AUDIO_DEVICE_IN_STUB or AUDIO_DEVICE_OUT_STUB // as they are used by stub HALs by convention if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { for (const auto& dev : mAvailableOutputDevices) { if (dev->type() == AUDIO_DEVICE_OUT_STUB) { continue; } if (portsWritten < portsMax) { dev->toAudioPort(&ports[portsWritten++]); } (*num_ports)++; } } if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { for (const auto& dev : mAvailableInputDevices) { if (dev->type() == AUDIO_DEVICE_IN_STUB) { continue; } if (portsWritten < portsMax) { dev->toAudioPort(&ports[portsWritten++]); } (*num_ports)++; } } } if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) { if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) { mInputs[i]->toAudioPort(&ports[portsWritten++]); } *num_ports += mInputs.size(); } if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { size_t numOutputs = 0; for (size_t i = 0; i < mOutputs.size(); i++) { if (!mOutputs[i]->isDuplicated()) { numOutputs++; if (portsWritten < portsMax) { mOutputs[i]->toAudioPort(&ports[portsWritten++]); } } } *num_ports += numOutputs; } } *generation = curAudioPortGeneration(); ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports); return NO_ERROR; } status_t AudioPolicyManager::listDeclaredDevicePorts(media::AudioPortRole role, std::vector* _aidl_return) { auto pushPort = [&](const sp& dev) -> status_t { audio_port_v7 port; dev->toAudioPort(&port); auto aidlPort = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_v7_AudioPortFw(port)); _aidl_return->push_back(std::move(aidlPort)); return OK; }; for (const auto& module : mHwModules) { for (const auto& dev : module->getDeclaredDevices()) { if (role == media::AudioPortRole::NONE || ((role == media::AudioPortRole::SOURCE) == audio_is_input_device(dev->type()))) { RETURN_STATUS_IF_ERROR(pushPort(dev)); } } } return OK; } status_t AudioPolicyManager::getAudioPort(struct audio_port_v7 *port) { if (port == nullptr || port->id == AUDIO_PORT_HANDLE_NONE) { return BAD_VALUE; } sp dev = mAvailableOutputDevices.getDeviceFromId(port->id); if (dev != 0) { dev->toAudioPort(port); return NO_ERROR; } dev = mAvailableInputDevices.getDeviceFromId(port->id); if (dev != 0) { dev->toAudioPort(port); return NO_ERROR; } sp out = mOutputs.getOutputFromId(port->id); if (out != 0) { out->toAudioPort(port); return NO_ERROR; } sp in = mInputs.getInputFromId(port->id); if (in != 0) { in->toAudioPort(port); return NO_ERROR; } return BAD_VALUE; } status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, audio_patch_handle_t *handle, uid_t uid) { ALOGV("%s", __func__); if (handle == NULL || patch == NULL) { return BAD_VALUE; } ALOGV("%s num sources %d num sinks %d", __func__, patch->num_sources, patch->num_sinks); if (!audio_patch_is_valid(patch)) { return BAD_VALUE; } // only one source per audio patch supported for now if (patch->num_sources > 1) { return INVALID_OPERATION; } if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) { return INVALID_OPERATION; } for (size_t i = 0; i < patch->num_sinks; i++) { if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) { return INVALID_OPERATION; } } sp srcDevice = mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); sp sinkDevice = mAvailableOutputDevices.getDeviceFromId(patch->sinks[0].id); if (srcDevice == nullptr || sinkDevice == nullptr) { ALOGW("%s could not create patch, invalid sink and/or source device(s)", __func__); return BAD_VALUE; } ALOGV("%s between source %s and sink %s", __func__, srcDevice->toString().c_str(), sinkDevice->toString().c_str()); audio_port_handle_t portId = PolicyAudioPort::getNextUniqueId(); // Default attributes, default volume priority, not to infer with non raw audio patches. audio_attributes_t attributes = attributes_initializer(AUDIO_USAGE_MEDIA); const struct audio_port_config *source = &patch->sources[0]; sp sourceDesc = new SourceClientDescriptor( portId, uid, attributes, *source, srcDevice, AUDIO_STREAM_PATCH, mEngine->getProductStrategyForAttributes(attributes), toVolumeSource(attributes), true, false /*isCallRx*/, false /*isCallTx*/); sourceDesc->setPreferredDeviceId(sinkDevice->getId()); status_t status = connectAudioSourceToSink(sourceDesc, sinkDevice, patch, *handle, uid, 0 /* delayMs */); if (status != NO_ERROR) { return INVALID_OPERATION; } mAudioSources.add(portId, sourceDesc); return NO_ERROR; } status_t AudioPolicyManager::connectAudioSourceToSink( const sp& sourceDesc, const sp &sinkDevice, const struct audio_patch *patch, audio_patch_handle_t &handle, uid_t uid, uint32_t delayMs) { status_t status = createAudioPatchInternal(patch, &handle, uid, delayMs, sourceDesc); if (status != NO_ERROR || mAudioPatches.indexOfKey(handle) < 0) { ALOGW("%s patch panel could not connect device patch, error %d", __func__, status); return INVALID_OPERATION; } sourceDesc->connect(handle, sinkDevice); if (isMsdPatch(handle)) { return NO_ERROR; } // SW Bridge? (@todo: HW bridge, keep track of HwOutput for device selection "reconsideration") sp swOutput = sourceDesc->swOutput().promote(); ALOG_ASSERT(swOutput != nullptr, "%s: a swOutput shall always be associated", __func__); if (swOutput->getClient(sourceDesc->portId()) != nullptr) { ALOGW("%s source portId has already been attached to outputDesc", __func__); goto FailurePatchAdded; } status = swOutput->start(); if (status != NO_ERROR) { goto FailureSourceAdded; } swOutput->addClient(sourceDesc); status = startSource(swOutput, sourceDesc, &delayMs); if (status != NO_ERROR) { ALOGW("%s failed to start source, error %d", __FUNCTION__, status); goto FailureSourceActive; } if (delayMs != 0) { usleep(delayMs * 1000); } return NO_ERROR; FailureSourceActive: swOutput->stop(); releaseOutput(sourceDesc->portId()); FailureSourceAdded: sourceDesc->setSwOutput(nullptr); FailurePatchAdded: releaseAudioPatchInternal(handle); return INVALID_OPERATION; } status_t AudioPolicyManager::createAudioPatchInternal(const struct audio_patch *patch, audio_patch_handle_t *handle, uid_t uid, uint32_t delayMs, const sp& sourceDesc) { ALOGV("%s num sources %d num sinks %d", __func__, patch->num_sources, patch->num_sinks); sp patchDesc; ssize_t index = mAudioPatches.indexOfKey(*handle); ALOGV("%s source id %d role %d type %d", __func__, patch->sources[0].id, patch->sources[0].role, patch->sources[0].type); #if LOG_NDEBUG == 0 for (size_t i = 0; i < patch->num_sinks; i++) { ALOGV("%s sink %zu: id %d role %d type %d", __func__ ,i, patch->sinks[i].id, patch->sinks[i].role, patch->sinks[i].type); } #endif if (index >= 0) { patchDesc = mAudioPatches.valueAt(index); ALOGV("%s mUidCached %d patchDesc->mUid %d uid %d", __func__, mUidCached, patchDesc->getUid(), uid); if (patchDesc->getUid() != mUidCached && uid != patchDesc->getUid()) { return INVALID_OPERATION; } } else { *handle = AUDIO_PATCH_HANDLE_NONE; } if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { sp outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("%s output not found for id %d", __func__, patch->sources[0].id); return BAD_VALUE; } ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports", outputDesc->mIoHandle); if (patchDesc != 0) { if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { ALOGV("%s source id differs for patch current id %d new id %d", __func__, patchDesc->mPatch.sources[0].id, patch->sources[0].id); return BAD_VALUE; } } DeviceVector devices; for (size_t i = 0; i < patch->num_sinks; i++) { // Only support mix to devices connection // TODO add support for mix to mix connection if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { ALOGV("%s source mix but sink is not a device", __func__); return INVALID_OPERATION; } sp devDesc = mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); if (devDesc == 0) { ALOGV("%s out device not found for id %d", __func__, patch->sinks[i].id); return BAD_VALUE; } if (outputDesc->mProfile->getCompatibilityScore( DeviceVector(devDesc), patch->sources[0].sample_rate, nullptr, // updatedSamplingRate patch->sources[0].format, nullptr, // updatedFormat patch->sources[0].channel_mask, nullptr, // updatedChannelMask AUDIO_OUTPUT_FLAG_NONE /*FIXME*/) == IOProfile::NO_MATCH) { ALOGV("%s profile not supported for device %08x", __func__, devDesc->type()); return INVALID_OPERATION; } devices.add(devDesc); } if (devices.size() == 0) { return INVALID_OPERATION; } // TODO: reconfigure output format and channels here ALOGV("%s setting device %s on output %d", __func__, dumpDeviceTypes(devices.types()).c_str(), outputDesc->mIoHandle); setOutputDevices(__func__, outputDesc, devices, true, 0, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { ALOGW("%s setOutputDevice() did not reuse the patch provided", __func__); } patchDesc = mAudioPatches.valueAt(index); patchDesc->setUid(uid); ALOGV("%s success", __func__); } else { ALOGW("%s setOutputDevice() failed to create a patch", __func__); return INVALID_OPERATION; } } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { // input device to input mix connection // only one sink supported when connecting an input device to a mix if (patch->num_sinks > 1) { return INVALID_OPERATION; } sp inputDesc = mInputs.getInputFromId(patch->sinks[0].id); if (inputDesc == NULL) { return BAD_VALUE; } if (patchDesc != 0) { if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) { return BAD_VALUE; } } sp device = mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); if (device == 0) { return BAD_VALUE; } if (inputDesc->mProfile->getCompatibilityScore( DeviceVector(device), patch->sinks[0].sample_rate, nullptr, /*updatedSampleRate*/ patch->sinks[0].format, nullptr, /*updatedFormat*/ patch->sinks[0].channel_mask, nullptr, /*updatedChannelMask*/ // FIXME for the parameter type, // and the NONE (audio_output_flags_t) AUDIO_INPUT_FLAG_NONE) == IOProfile::NO_MATCH) { return INVALID_OPERATION; } // TODO: reconfigure output format and channels here ALOGV("%s setting device %s on output %d", __func__, device->toString().c_str(), inputDesc->mIoHandle); setInputDevice(inputDesc->mIoHandle, device, true, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { ALOGW("%s setInputDevice() did not reuse the patch provided", __func__); } patchDesc = mAudioPatches.valueAt(index); patchDesc->setUid(uid); ALOGV("%s success", __func__); } else { ALOGW("%s setInputDevice() failed to create a patch", __func__); return INVALID_OPERATION; } } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { // device to device connection if (patchDesc != 0) { if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) { return BAD_VALUE; } } sp srcDevice = mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); if (srcDevice == 0) { return BAD_VALUE; } //update source and sink with our own data as the data passed in the patch may // be incomplete. PatchBuilder patchBuilder; audio_port_config sourcePortConfig = {}; // if first sink is to MSD, establish single MSD patch if (getMsdAudioOutDevices().contains( mAvailableOutputDevices.getDeviceFromId(patch->sinks[0].id))) { ALOGV("%s patching to MSD", __FUNCTION__); patchBuilder = buildMsdPatch(false /*msdIsSource*/, srcDevice); goto installPatch; } srcDevice->toAudioPortConfig(&sourcePortConfig, &patch->sources[0]); patchBuilder.addSource(sourcePortConfig); for (size_t i = 0; i < patch->num_sinks; i++) { if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { ALOGV("%s source device but one sink is not a device", __func__); return INVALID_OPERATION; } sp sinkDevice = mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); if (sinkDevice == 0) { return BAD_VALUE; } audio_port_config sinkPortConfig = {}; sinkDevice->toAudioPortConfig(&sinkPortConfig, &patch->sinks[i]); patchBuilder.addSink(sinkPortConfig); // Whatever Sw or Hw bridge, we do attach an SwOutput to an Audio Source for // volume management purpose (tracking activity) // In case of Hw bridge, it is a Work Around. The mixPort used is the one declared // in config XML to reach the sink so that is can be declared as available. audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; sp outputDesc; if (!sourceDesc->isInternal()) { // take care of dynamic routing for SwOutput selection, audio_attributes_t attributes = sourceDesc->attributes(); audio_stream_type_t stream = sourceDesc->stream(); audio_attributes_t resultAttr; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = sourceDesc->config().sample_rate; audio_channel_mask_t sourceMask = sourceDesc->config().channel_mask; config.channel_mask = (audio_channel_mask_get_representation(sourceMask) == AUDIO_CHANNEL_REPRESENTATION_INDEX) ? sourceMask : audio_channel_mask_in_to_out(sourceMask); config.format = sourceDesc->config().format; audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE; audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE; bool isRequestedDeviceForExclusiveUse = false; output_type_t outputType; bool isSpatialized; bool isBitPerfect; getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE, &attributes, &stream, sourceDesc->uid(), &config, &flags, &selectedDeviceId, &isRequestedDeviceForExclusiveUse, nullptr, &outputType, &isSpatialized, &isBitPerfect); if (output == AUDIO_IO_HANDLE_NONE) { ALOGV("%s no output for device %s", __FUNCTION__, sinkDevice->toString().c_str()); return INVALID_OPERATION; } outputDesc = mOutputs.valueFor(output); if (outputDesc->isDuplicated()) { ALOGE("%s output is duplicated", __func__); return INVALID_OPERATION; } bool closeOutput = outputDesc->mDirectOpenCount != 0; sourceDesc->setSwOutput(outputDesc, closeOutput); } else { // Same for "raw patches" aka created from createAudioPatch API SortedVector outputs = getOutputsForDevices(DeviceVector(sinkDevice), mOutputs); // if the sink device is reachable via an opened output stream, request to // go via this output stream by adding a second source to the patch // description output = selectOutput(outputs); if (output == AUDIO_IO_HANDLE_NONE) { ALOGE("%s no output available for internal patch sink", __func__); return INVALID_OPERATION; } outputDesc = mOutputs.valueFor(output); if (outputDesc->isDuplicated()) { ALOGV("%s output for device %s is duplicated", __func__, sinkDevice->toString().c_str()); return INVALID_OPERATION; } sourceDesc->setSwOutput(outputDesc, /* closeOutput= */ false); } // create a software bridge in PatchPanel if: // - source and sink devices are on different HW modules OR // - audio HAL version is < 3.0 // - audio HAL version is >= 3.0 but no route has been declared between devices // - called from startAudioSource (aka sourceDesc is not internal) and source device // does not have a gain controller if (!srcDevice->hasSameHwModuleAs(sinkDevice) || (srcDevice->getModuleVersionMajor() < 3) || !srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice) || (!sourceDesc->isInternal() && srcDevice->getAudioPort()->getGains().size() == 0)) { // support only one sink device for now to simplify output selection logic if (patch->num_sinks > 1) { return INVALID_OPERATION; } sourceDesc->setUseSwBridge(); if (outputDesc != nullptr) { audio_port_config srcMixPortConfig = {}; outputDesc->toAudioPortConfig(&srcMixPortConfig, nullptr); // for volume control, we may need a valid stream srcMixPortConfig.ext.mix.usecase.stream = (!sourceDesc->isInternal() || sourceDesc->isCallTx()) ? mEngine->getStreamTypeForAttributes(sourceDesc->attributes()) : AUDIO_STREAM_PATCH; patchBuilder.addSource(srcMixPortConfig); } } } // TODO: check from routing capabilities in config file and other conflicting patches installPatch: status_t status = installPatch( __func__, index, handle, patchBuilder.patch(), delayMs, uid, &patchDesc); if (status != NO_ERROR) { ALOGW("%s patch panel could not connect device patch, error %d", __func__, status); return INVALID_OPERATION; } } else { return BAD_VALUE; } } else { return BAD_VALUE; } return NO_ERROR; } status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle, uid_t uid) { ALOGV("%s patch %d", __func__, handle); ssize_t index = mAudioPatches.indexOfKey(handle); if (index < 0) { return BAD_VALUE; } sp patchDesc = mAudioPatches.valueAt(index); ALOGV("%s() mUidCached %d patchDesc->mUid %d uid %d", __func__, mUidCached, patchDesc->getUid(), uid); if (patchDesc->getUid() != mUidCached && uid != patchDesc->getUid()) { return INVALID_OPERATION; } audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE; for (size_t i = 0; i < mAudioSources.size(); i++) { sp sourceDesc = mAudioSources.valueAt(i); if (sourceDesc != nullptr && sourceDesc->getPatchHandle() == handle) { portId = sourceDesc->portId(); break; } } return portId != AUDIO_PORT_HANDLE_NONE ? stopAudioSource(portId) : releaseAudioPatchInternal(handle); } status_t AudioPolicyManager::releaseAudioPatchInternal(audio_patch_handle_t handle, uint32_t delayMs, const sp& sourceDesc) { ALOGV("%s patch %d", __func__, handle); if (mAudioPatches.indexOfKey(handle) < 0) { ALOGE("%s: no patch found with handle=%d", __func__, handle); return BAD_VALUE; } sp patchDesc = mAudioPatches.valueFor(handle); struct audio_patch *patch = &patchDesc->mPatch; patchDesc->setUid(mUidCached); if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { sp outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("%s output not found for id %d", __func__, patch->sources[0].id); return BAD_VALUE; } setOutputDevices(__func__, outputDesc, getNewOutputDevices(outputDesc, true /*fromCache*/), true, 0, NULL); } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { sp inputDesc = mInputs.getInputFromId(patch->sinks[0].id); if (inputDesc == NULL) { ALOGV("%s input not found for id %d", __func__, patch->sinks[0].id); return BAD_VALUE; } setInputDevice(inputDesc->mIoHandle, getNewInputDevice(inputDesc), true, NULL); } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { status_t status = mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), delayMs); ALOGV("%s patch panel returned %d patchHandle %d", __func__, status, patchDesc->getAfHandle()); removeAudioPatch(patchDesc->getHandle()); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); // SW or HW Bridge sp outputDesc = nullptr; audio_patch_handle_t patchHandle = AUDIO_PATCH_HANDLE_NONE; if (patch->num_sources > 1 && patch->sources[1].type == AUDIO_PORT_TYPE_MIX) { outputDesc = mOutputs.getOutputFromId(patch->sources[1].id); } else if (patch->num_sources == 1 && sourceDesc != nullptr) { outputDesc = sourceDesc->swOutput().promote(); } if (outputDesc == nullptr) { ALOGW("%s no output for id %d", __func__, patch->sources[0].id); // releaseOutput has already called closeOutput in case of direct output return NO_ERROR; } patchHandle = outputDesc->getPatchHandle(); // While using a HwBridge, force reconsidering device only if not reusing an existing // output and no more activity on output (will force to close). const bool force = sourceDesc->canCloseOutput() && !outputDesc->isActive(); // APM pattern is to have always outputs opened / patch realized for reachable devices. // Update device may result to NONE (empty), coupled with force, it releases the patch. // Reconsider device only for cases: // 1 / Active Output // 2 / Inactive Output previously hosting HwBridge // 3 / Inactive Output previously hosting SwBridge that can be closed. bool updateDevice = outputDesc->isActive() || !sourceDesc->useSwBridge() || sourceDesc->canCloseOutput(); setOutputDevices(__func__, outputDesc, updateDevice ? getNewOutputDevices(outputDesc, true /*fromCache*/) : outputDesc->devices(), force, 0, patchHandle == AUDIO_PATCH_HANDLE_NONE ? nullptr : &patchHandle); } else { return BAD_VALUE; } } else { return BAD_VALUE; } return NO_ERROR; } status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches, struct audio_patch *patches, unsigned int *generation) { if (generation == NULL) { return BAD_VALUE; } *generation = curAudioPortGeneration(); return mAudioPatches.listAudioPatches(num_patches, patches); } status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config) { ALOGV("setAudioPortConfig()"); if (config == NULL) { return BAD_VALUE; } ALOGV("setAudioPortConfig() on port handle %d", config->id); // Only support gain configuration for now if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) { return INVALID_OPERATION; } sp audioPortConfig; if (config->type == AUDIO_PORT_TYPE_MIX) { if (config->role == AUDIO_PORT_ROLE_SOURCE) { sp outputDesc = mOutputs.getOutputFromId(config->id); if (outputDesc == NULL) { return BAD_VALUE; } ALOG_ASSERT(!outputDesc->isDuplicated(), "setAudioPortConfig() called on duplicated output %d", outputDesc->mIoHandle); audioPortConfig = outputDesc; } else if (config->role == AUDIO_PORT_ROLE_SINK) { sp inputDesc = mInputs.getInputFromId(config->id); if (inputDesc == NULL) { return BAD_VALUE; } audioPortConfig = inputDesc; } else { return BAD_VALUE; } } else if (config->type == AUDIO_PORT_TYPE_DEVICE) { sp deviceDesc; if (config->role == AUDIO_PORT_ROLE_SOURCE) { deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id); } else if (config->role == AUDIO_PORT_ROLE_SINK) { deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id); } else { return BAD_VALUE; } if (deviceDesc == NULL) { return BAD_VALUE; } audioPortConfig = deviceDesc; } else { return BAD_VALUE; } struct audio_port_config backupConfig = {}; status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig); if (status == NO_ERROR) { struct audio_port_config newConfig = {}; audioPortConfig->toAudioPortConfig(&newConfig, config); status = mpClientInterface->setAudioPortConfig(&newConfig, 0); } if (status != NO_ERROR) { audioPortConfig->applyAudioPortConfig(&backupConfig); } return status; } void AudioPolicyManager::releaseResourcesForUid(uid_t uid) { clearAudioSources(uid); clearAudioPatches(uid); clearSessionRoutes(uid); } void AudioPolicyManager::clearAudioPatches(uid_t uid) { for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) { sp patchDesc = mAudioPatches.valueAt(i); if (patchDesc->getUid() == uid) { releaseAudioPatch(mAudioPatches.keyAt(i), uid); } } } void AudioPolicyManager::checkStrategyRoute(product_strategy_t ps, audio_io_handle_t ouptutToSkip) { // Take the first attributes following the product strategy as it is used to retrieve the routed // device. All attributes wihin a strategy follows the same "routing strategy" auto attributes = mEngine->getAllAttributesForProductStrategy(ps).front(); DeviceVector devices = mEngine->getOutputDevicesForAttributes(attributes, nullptr, false); SortedVector outputs = getOutputsForDevices(devices, mOutputs); std::map outputsToReopen; for (size_t j = 0; j < mOutputs.size(); j++) { if (mOutputs.keyAt(j) == ouptutToSkip) { continue; } sp outputDesc = mOutputs.valueAt(j); if (!outputDesc->isStrategyActive(ps)) { continue; } // If the default device for this strategy is on another output mix, // invalidate all tracks in this strategy to force re connection. // Otherwise select new device on the output mix. if (outputs.indexOf(mOutputs.keyAt(j)) < 0) { invalidateStreams(mEngine->getStreamTypesForProductStrategy(ps)); } else { DeviceVector newDevices = getNewOutputDevices(outputDesc, false /*fromCache*/); if (outputDesc->mPreferredAttrInfo != nullptr && outputDesc->devices() != newDevices) { // If the device is using preferred mixer attributes, the output need to reopen // with default configuration when the new selected devices are different from // current routing devices. outputsToReopen.emplace(mOutputs.keyAt(j), newDevices); continue; } setOutputDevices(__func__, outputDesc, newDevices, false); } } reopenOutputsWithDevices(outputsToReopen); } void AudioPolicyManager::clearSessionRoutes(uid_t uid) { // remove output routes associated with this uid std::vector affectedStrategies; for (size_t i = 0; i < mOutputs.size(); i++) { sp outputDesc = mOutputs.valueAt(i); for (const auto& client : outputDesc->getClientIterable()) { if (client->hasPreferredDevice() && client->uid() == uid) { client->setPreferredDeviceId(AUDIO_PORT_HANDLE_NONE); auto clientStrategy = client->strategy(); if (std::find(begin(affectedStrategies), end(affectedStrategies), clientStrategy) != end(affectedStrategies)) { continue; } affectedStrategies.push_back(client->strategy()); } } } // reroute outputs if necessary for (const auto& strategy : affectedStrategies) { checkStrategyRoute(strategy, AUDIO_IO_HANDLE_NONE); } // remove input routes associated with this uid SortedVector affectedSources; for (size_t i = 0; i < mInputs.size(); i++) { sp inputDesc = mInputs.valueAt(i); for (const auto& client : inputDesc->getClientIterable()) { if (client->hasPreferredDevice() && client->uid() == uid) { client->setPreferredDeviceId(AUDIO_PORT_HANDLE_NONE); affectedSources.add(client->source()); } } } // reroute inputs if necessary SortedVector inputsToClose; for (size_t i = 0; i < mInputs.size(); i++) { sp inputDesc = mInputs.valueAt(i); if (affectedSources.indexOf(inputDesc->source()) >= 0) { inputsToClose.add(inputDesc->mIoHandle); } } for (const auto& input : inputsToClose) { closeInput(input); } } void AudioPolicyManager::clearAudioSources(uid_t uid) { for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--) { sp sourceDesc = mAudioSources.valueAt(i); if (sourceDesc->uid() == uid) { stopAudioSource(mAudioSources.keyAt(i)); } } } status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session, audio_io_handle_t *ioHandle, audio_devices_t *device) { *session = (audio_session_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); audio_attributes_t attr = { .source = AUDIO_SOURCE_HOTWORD }; sp deviceDesc = mEngine->getInputDeviceForAttributes(attr); if (deviceDesc == nullptr) { return INVALID_OPERATION; } *device = deviceDesc->type(); return mSoundTriggerSessions.acquireSession(*session, *ioHandle); } status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source, const audio_attributes_t *attributes, audio_port_handle_t *portId, uid_t uid) { return startAudioSourceInternal(source, attributes, portId, uid, false /*internal*/, false /*isCallRx*/, 0 /*delayMs*/); } status_t AudioPolicyManager::startAudioSourceInternal(const struct audio_port_config *source, const audio_attributes_t *attributes, audio_port_handle_t *portId, uid_t uid, bool internal, bool isCallRx, uint32_t delayMs) { ALOGV("%s", __FUNCTION__); *portId = AUDIO_PORT_HANDLE_NONE; if (source == NULL || attributes == NULL || portId == NULL) { ALOGW("%s invalid argument: source %p attributes %p handle %p", __FUNCTION__, source, attributes, portId); return BAD_VALUE; } if (source->role != AUDIO_PORT_ROLE_SOURCE || source->type != AUDIO_PORT_TYPE_DEVICE) { ALOGW("%s INVALID_OPERATION source->role %d source->type %d", __FUNCTION__, source->role, source->type); return INVALID_OPERATION; } sp srcDevice = mAvailableInputDevices.getDevice(source->ext.device.type, String8(source->ext.device.address), AUDIO_FORMAT_DEFAULT); if (srcDevice == 0) { ALOGW("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type); return BAD_VALUE; } *portId = PolicyAudioPort::getNextUniqueId(); sp sourceDesc = new SourceClientDescriptor(*portId, uid, *attributes, *source, srcDevice, mEngine->getStreamTypeForAttributes(*attributes), mEngine->getProductStrategyForAttributes(*attributes), toVolumeSource(*attributes), internal, isCallRx, false); status_t status = connectAudioSource(sourceDesc, delayMs); if (status == NO_ERROR) { mAudioSources.add(*portId, sourceDesc); } return status; } status_t AudioPolicyManager::connectAudioSource(const sp& sourceDesc, uint32_t delayMs) { ALOGV("%s handle %d", __FUNCTION__, sourceDesc->portId()); // make sure we only have one patch per source. disconnectAudioSource(sourceDesc); audio_attributes_t attributes = sourceDesc->attributes(); // May the device (dynamic) have been disconnected/reconnected, id has changed. sp srcDevice = mAvailableInputDevices.getDevice( sourceDesc->srcDevice()->type(), String8(sourceDesc->srcDevice()->address().c_str()), AUDIO_FORMAT_DEFAULT); DeviceVector sinkDevices = mEngine->getOutputDevicesForAttributes(attributes, nullptr, false /*fromCache*/); ALOG_ASSERT(!sinkDevices.isEmpty(), "connectAudioSource(): no device found for attributes"); sp sinkDevice = sinkDevices.itemAt(0); if (!mAvailableOutputDevices.contains(sinkDevice)) { ALOGE("%s Device %s not available", __func__, sinkDevice->toString().c_str()); return INVALID_OPERATION; } PatchBuilder patchBuilder; patchBuilder.addSink(sinkDevice).addSource(srcDevice); audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE; return connectAudioSourceToSink( sourceDesc, sinkDevice, patchBuilder.patch(), handle, mUidCached, delayMs); } status_t AudioPolicyManager::stopAudioSource(audio_port_handle_t portId) { sp sourceDesc = mAudioSources.valueFor(portId); ALOGV("%s port ID %d", __FUNCTION__, portId); if (sourceDesc == 0) { ALOGW("%s unknown source for port ID %d", __FUNCTION__, portId); return BAD_VALUE; } status_t status = disconnectAudioSource(sourceDesc); mAudioSources.removeItem(portId); return status; } status_t AudioPolicyManager::setMasterMono(bool mono) { if (mMasterMono == mono) { return NO_ERROR; } mMasterMono = mono; // if enabling mono we close all offloaded devices, which will invalidate the // corresponding AudioTrack. The AudioTrack client/MediaPlayer is responsible // for recreating the new AudioTrack as non-offloaded PCM. // // If disabling mono, we leave all tracks as is: we don't know which clients // and tracks are able to be recreated as offloaded. The next "song" should // play back offloaded. if (mMasterMono) { Vector offloaded; for (size_t i = 0; i < mOutputs.size(); ++i) { sp desc = mOutputs.valueAt(i); if (desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { offloaded.push(desc->mIoHandle); } } for (const auto& handle : offloaded) { closeOutput(handle); } } // update master mono for all remaining outputs for (size_t i = 0; i < mOutputs.size(); ++i) { updateMono(mOutputs.keyAt(i)); } return NO_ERROR; } status_t AudioPolicyManager::getMasterMono(bool *mono) { *mono = mMasterMono; return NO_ERROR; } float AudioPolicyManager::getStreamVolumeDB( audio_stream_type_t stream, int index, audio_devices_t device) { return computeVolume(getVolumeCurves(stream), toVolumeSource(stream), index, {device}); } status_t AudioPolicyManager::getSurroundFormats(unsigned int *numSurroundFormats, audio_format_t *surroundFormats, bool *surroundFormatsEnabled) { if (numSurroundFormats == nullptr || (*numSurroundFormats != 0 && (surroundFormats == nullptr || surroundFormatsEnabled == nullptr))) { return BAD_VALUE; } ALOGV("%s() numSurroundFormats %d surroundFormats %p surroundFormatsEnabled %p", __func__, *numSurroundFormats, surroundFormats, surroundFormatsEnabled); size_t formatsWritten = 0; size_t formatsMax = *numSurroundFormats; *numSurroundFormats = mConfig->getSurroundFormats().size(); audio_policy_forced_cfg_t forceUse = mEngine->getForceUse( AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND); for (const auto& format: mConfig->getSurroundFormats()) { if (formatsWritten < formatsMax) { surroundFormats[formatsWritten] = format.first; bool formatEnabled = true; switch (forceUse) { case AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL: formatEnabled = mManualSurroundFormats.count(format.first) != 0; break; case AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER: formatEnabled = false; break; default: // AUTO or ALWAYS => true break; } surroundFormatsEnabled[formatsWritten++] = formatEnabled; } } return NO_ERROR; } status_t AudioPolicyManager::getReportedSurroundFormats(unsigned int *numSurroundFormats, audio_format_t *surroundFormats) { if (numSurroundFormats == nullptr || (*numSurroundFormats != 0 && surroundFormats == nullptr)) { return BAD_VALUE; } ALOGV("%s() numSurroundFormats %d surroundFormats %p", __func__, *numSurroundFormats, surroundFormats); size_t formatsWritten = 0; size_t formatsMax = *numSurroundFormats; std::unordered_set formats; // Uses primary surround formats only // Return formats from all device profiles that have already been resolved by // checkOutputsForDevice(). for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { sp device = mAvailableOutputDevices[i]; audio_devices_t deviceType = device->type(); // Enabling/disabling formats are applied to only HDMI devices. So, this function // returns formats reported by HDMI devices. if (deviceType != AUDIO_DEVICE_OUT_HDMI && deviceType != AUDIO_DEVICE_OUT_HDMI_ARC && deviceType != AUDIO_DEVICE_OUT_HDMI_EARC) { continue; } // Formats reported by sink devices std::unordered_set formatset; if (auto it = mReportedFormatsMap.find(device); it != mReportedFormatsMap.end()) { formatset.insert(it->second.begin(), it->second.end()); } // Formats hard-coded in the in policy configuration file (if any). FormatVector encodedFormats = device->encodedFormats(); formatset.insert(encodedFormats.begin(), encodedFormats.end()); // Filter the formats which are supported by the vendor hardware. for (auto it = formatset.begin(); it != formatset.end(); ++it) { if (mConfig->getSurroundFormats().count(*it) != 0) { formats.insert(*it); } else { for (const auto& pair : mConfig->getSurroundFormats()) { if (pair.second.count(*it) != 0) { formats.insert(pair.first); break; } } } } } *numSurroundFormats = formats.size(); for (const auto& format: formats) { if (formatsWritten < formatsMax) { surroundFormats[formatsWritten++] = format; } } return NO_ERROR; } status_t AudioPolicyManager::setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled) { ALOGV("%s() format 0x%X enabled %d", __func__, audioFormat, enabled); const auto& formatIter = mConfig->getSurroundFormats().find(audioFormat); if (formatIter == mConfig->getSurroundFormats().end()) { ALOGW("%s() format 0x%X is not a known surround format", __func__, audioFormat); return BAD_VALUE; } if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND) != AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL) { ALOGW("%s() not in manual mode for surround sound format selection", __func__); return INVALID_OPERATION; } if ((mManualSurroundFormats.count(audioFormat) != 0) == enabled) { return NO_ERROR; } std::unordered_set surroundFormatsBackup(mManualSurroundFormats); if (enabled) { mManualSurroundFormats.insert(audioFormat); for (const auto& subFormat : formatIter->second) { mManualSurroundFormats.insert(subFormat); } } else { mManualSurroundFormats.erase(audioFormat); for (const auto& subFormat : formatIter->second) { mManualSurroundFormats.erase(subFormat); } } sp outputDesc; bool profileUpdated = false; DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromTypes( {AUDIO_DEVICE_OUT_HDMI, AUDIO_DEVICE_OUT_HDMI_ARC, AUDIO_DEVICE_OUT_HDMI_EARC}); for (size_t i = 0; i < hdmiOutputDevices.size(); i++) { // Simulate reconnection to update enabled surround sound formats. String8 address = String8(hdmiOutputDevices[i]->address().c_str()); std::string name = hdmiOutputDevices[i]->getName(); status_t status = setDeviceConnectionStateInt(hdmiOutputDevices[i]->type(), AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT); if (status != NO_ERROR) { continue; } status = setDeviceConnectionStateInt(hdmiOutputDevices[i]->type(), AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT); profileUpdated |= (status == NO_ERROR); } // FIXME: Why doing this for input HDMI devices if we don't augment their reported formats? DeviceVector hdmiInputDevices = mAvailableInputDevices.getDevicesFromType( AUDIO_DEVICE_IN_HDMI); for (size_t i = 0; i < hdmiInputDevices.size(); i++) { // Simulate reconnection to update enabled surround sound formats. String8 address = String8(hdmiInputDevices[i]->address().c_str()); std::string name = hdmiInputDevices[i]->getName(); status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT); if (status != NO_ERROR) { continue; } status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT); profileUpdated |= (status == NO_ERROR); } if (!profileUpdated) { ALOGW("%s() no audio profiles updated, undoing surround formats change", __func__); mManualSurroundFormats = std::move(surroundFormatsBackup); } return profileUpdated ? NO_ERROR : INVALID_OPERATION; } void AudioPolicyManager::setAppState(audio_port_handle_t portId, app_state_t state) { ALOGV("%s(portId:%d, state:%d)", __func__, portId, state); for (size_t i = 0; i < mInputs.size(); i++) { mInputs.valueAt(i)->setAppState(portId, state); } } bool AudioPolicyManager::isHapticPlaybackSupported() { for (const auto& hwModule : mHwModules) { const OutputProfileCollection &outputProfiles = hwModule->getOutputProfiles(); for (const auto &outProfile : outputProfiles) { struct audio_port audioPort; outProfile->toAudioPort(&audioPort); for (size_t i = 0; i < audioPort.num_channel_masks; i++) { if (audioPort.channel_masks[i] & AUDIO_CHANNEL_HAPTIC_ALL) { return true; } } } } return false; } bool AudioPolicyManager::isUltrasoundSupported() { bool hasUltrasoundOutput = false; bool hasUltrasoundInput = false; for (const auto& hwModule : mHwModules) { const OutputProfileCollection &outputProfiles = hwModule->getOutputProfiles(); if (!hasUltrasoundOutput) { for (const auto &outProfile : outputProfiles) { if (outProfile->getFlags() & AUDIO_OUTPUT_FLAG_ULTRASOUND) { hasUltrasoundOutput = true; break; } } } const InputProfileCollection &inputProfiles = hwModule->getInputProfiles(); if (!hasUltrasoundInput) { for (const auto &inputProfile : inputProfiles) { if (inputProfile->getFlags() & AUDIO_INPUT_FLAG_ULTRASOUND) { hasUltrasoundInput = true; break; } } } if (hasUltrasoundOutput && hasUltrasoundInput) return true; } return false; } bool AudioPolicyManager::isHotwordStreamSupported(bool lookbackAudio) { const auto mask = AUDIO_INPUT_FLAG_HOTWORD_TAP | (lookbackAudio ? AUDIO_INPUT_FLAG_HW_LOOKBACK : 0); for (const auto& hwModule : mHwModules) { const InputProfileCollection &inputProfiles = hwModule->getInputProfiles(); for (const auto &inputProfile : inputProfiles) { if ((inputProfile->getFlags() & mask) == mask) { return true; } } } return false; } bool AudioPolicyManager::isCallScreenModeSupported() { return mConfig->isCallScreenModeSupported(); } status_t AudioPolicyManager::disconnectAudioSource(const sp& sourceDesc) { ALOGV("%s port Id %d", __FUNCTION__, sourceDesc->portId()); if (!sourceDesc->isConnected()) { ALOGV("%s port Id %d already disconnected", __FUNCTION__, sourceDesc->portId()); return NO_ERROR; } sp swOutput = sourceDesc->swOutput().promote(); if (swOutput != 0) { status_t status = stopSource(swOutput, sourceDesc); if (status == NO_ERROR) { swOutput->stop(); } if (releaseOutput(sourceDesc->portId())) { // The output descriptor is reopened to query dynamic profiles. In that case, there is // no need to release audio patch here but just return NO_ERROR. return NO_ERROR; } } else { sp hwOutputDesc = sourceDesc->hwOutput().promote(); if (hwOutputDesc != 0) { // close Hwoutput and remove from mHwOutputs } else { ALOGW("%s source has neither SW nor HW output", __FUNCTION__); } } status_t status = releaseAudioPatchInternal(sourceDesc->getPatchHandle(), 0, sourceDesc); sourceDesc->disconnect(); return status; } sp AudioPolicyManager::getSourceForAttributesOnOutput( audio_io_handle_t output, const audio_attributes_t &attr) { sp source; for (size_t i = 0; i < mAudioSources.size(); i++) { sp sourceDesc = mAudioSources.valueAt(i); sp outputDesc = sourceDesc->swOutput().promote(); if (followsSameRouting(attr, sourceDesc->attributes()) && outputDesc != 0 && outputDesc->mIoHandle == output) { source = sourceDesc; break; } } return source; } bool AudioPolicyManager::canBeSpatializedInt(const audio_attributes_t *attr, const audio_config_t *config, const AudioDeviceTypeAddrVector &devices) const { // The caller can have the audio attributes criteria ignored by either passing a null ptr or // the AUDIO_ATTRIBUTES_INITIALIZER value. // If attributes are specified, current policy is to only allow spatialization for media // and game usages. if (attr != nullptr && *attr != AUDIO_ATTRIBUTES_INITIALIZER) { if (attr->usage != AUDIO_USAGE_MEDIA && attr->usage != AUDIO_USAGE_GAME) { return false; } if ((attr->flags & (AUDIO_FLAG_CONTENT_SPATIALIZED | AUDIO_FLAG_NEVER_SPATIALIZE)) != 0) { return false; } } // The caller can have the audio config criteria ignored by either passing a null ptr or // the AUDIO_CONFIG_INITIALIZER value. // If an audio config is specified, current policy is to only allow spatialization for // some positional channel masks and PCM format and for stereo if low latency performance // mode is not requested. if (config != nullptr && *config != AUDIO_CONFIG_INITIALIZER) { static const bool stereo_spatialization_enabled = property_get_bool("ro.audio.stereo_spatialization_enabled", false); const bool channel_mask_spatialized = (stereo_spatialization_enabled && com_android_media_audio_stereo_spatialization()) ? audio_channel_mask_contains_stereo(config->channel_mask) : audio_is_channel_mask_spatialized(config->channel_mask); if (!channel_mask_spatialized) { return false; } if (!audio_is_linear_pcm(config->format)) { return false; } if (config->channel_mask == AUDIO_CHANNEL_OUT_STEREO && ((attr->flags & AUDIO_FLAG_LOW_LATENCY) != 0)) { return false; } } sp profile = getSpatializerOutputProfile(config, devices); if (profile == nullptr) { return false; } return true; } // The Spatializer output is compatible with Haptic use cases if: // 1. the Spatializer output thread supports Haptic, and format/sampleRate are same // with client if client haptic channel bits were set, or // 2. the Spatializer output thread does not support Haptic, and client did not ask haptic by // including the haptic bits or creating the HapticGenerator effect for same session. bool AudioPolicyManager::checkHapticCompatibilityOnSpatializerOutput( const audio_config_t* config, audio_session_t sessionId) const { const auto clientHapticChannel = audio_channel_count_from_out_mask(config->channel_mask & AUDIO_CHANNEL_HAPTIC_ALL); const auto threadOutputHapticChannel = audio_channel_count_from_out_mask( mSpatializerOutput->getChannelMask() & AUDIO_CHANNEL_HAPTIC_ALL); if (threadOutputHapticChannel) { // check format and sampleRate match if client haptic channel mask exist if (clientHapticChannel) { return mSpatializerOutput->getFormat() == config->format && mSpatializerOutput->getSamplingRate() == config->sample_rate; } return true; } else { // in the case of the Spatializer output channel mask does not have haptic channel bits, it // means haptic use cases (either the client channelmask includes haptic bits, or created a // HapticGenerator effect for this session) are not supported. return clientHapticChannel == 0 && !mEffects.hasOrphansForSession(sessionId, FX_IID_HAPTICGENERATOR); } } void AudioPolicyManager::checkVirtualizerClientRoutes() { std::set streamsToInvalidate; for (size_t i = 0; i < mOutputs.size(); i++) { const sp& desc = mOutputs[i]; for (const sp& client : desc->getClientIterable()) { audio_attributes_t attr = client->attributes(); DeviceVector devices = mEngine->getOutputDevicesForAttributes(attr, nullptr, false); AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector(); audio_config_base_t clientConfig = client->config(); audio_config_t config = audio_config_initializer(&clientConfig); if (desc != mSpatializerOutput && canBeSpatializedInt(&attr, &config, devicesTypeAddress)) { streamsToInvalidate.insert(client->stream()); } } } invalidateStreams(StreamTypeVector(streamsToInvalidate.begin(), streamsToInvalidate.end())); } bool AudioPolicyManager::isOutputOnlyAvailableRouteToSomeDevice( const sp& outputDesc) { if (outputDesc->isDuplicated()) { return false; } DeviceVector devices = outputDesc->supportedDevices(); for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (desc == outputDesc || desc->isDuplicated()) { continue; } DeviceVector sharedDevices = desc->filterSupportedDevices(devices); if (!sharedDevices.isEmpty() && (desc->devicesSupportEncodedFormats(sharedDevices.types()) == outputDesc->devicesSupportEncodedFormats(sharedDevices.types()))) { return false; } } return true; } status_t AudioPolicyManager::getSpatializerOutput(const audio_config_base_t *mixerConfig, const audio_attributes_t *attr, audio_io_handle_t *output) { *output = AUDIO_IO_HANDLE_NONE; DeviceVector devices = mEngine->getOutputDevicesForAttributes(*attr, nullptr, false); AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector(); audio_config_t *configPtr = nullptr; audio_config_t config; if (mixerConfig != nullptr) { config = audio_config_initializer(mixerConfig); configPtr = &config; } if (!canBeSpatializedInt(attr, configPtr, devicesTypeAddress)) { ALOGV("%s provided attributes or mixer config cannot be spatialized", __func__); return BAD_VALUE; } sp profile = getSpatializerOutputProfile(configPtr, devicesTypeAddress); if (profile == nullptr) { ALOGV("%s no suitable output profile for provided attributes or mixer config", __func__); return BAD_VALUE; } std::vector> spatializerOutputs; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (!desc->isDuplicated() && (desc->mFlags & AUDIO_OUTPUT_FLAG_SPATIALIZER) != 0) { spatializerOutputs.push_back(desc); ALOGV("%s adding opened spatializer Output %d", __func__, desc->mIoHandle); } } mSpatializerOutput.clear(); bool outputsChanged = false; for (const auto& desc : spatializerOutputs) { if (desc->mProfile == profile && (configPtr == nullptr || configPtr->channel_mask == desc->mMixerChannelMask)) { mSpatializerOutput = desc; ALOGV("%s reusing current spatializer output %d", __func__, desc->mIoHandle); } else { ALOGV("%s closing spatializerOutput output %d to match channel mask %#x" " and devices %s", __func__, desc->mIoHandle, configPtr != nullptr ? configPtr->channel_mask : 0, devices.toString().c_str()); closeOutput(desc->mIoHandle); outputsChanged = true; } } if (mSpatializerOutput == nullptr) { sp desc = openOutputWithProfileAndDevice(profile, devices, mixerConfig); if (desc != nullptr) { mSpatializerOutput = desc; outputsChanged = true; } } checkVirtualizerClientRoutes(); if (outputsChanged) { mPreviousOutputs = mOutputs; mpClientInterface->onAudioPortListUpdate(); } if (mSpatializerOutput == nullptr) { ALOGV("%s could not open spatializer output with requested config", __func__); return BAD_VALUE; } *output = mSpatializerOutput->mIoHandle; ALOGV("%s returning new spatializer output %d", __func__, *output); return OK; } status_t AudioPolicyManager::releaseSpatializerOutput(audio_io_handle_t output) { if (mSpatializerOutput == nullptr) { return INVALID_OPERATION; } if (mSpatializerOutput->mIoHandle != output) { return BAD_VALUE; } if (!isOutputOnlyAvailableRouteToSomeDevice(mSpatializerOutput)) { ALOGV("%s closing spatializer output %d", __func__, mSpatializerOutput->mIoHandle); closeOutput(mSpatializerOutput->mIoHandle); //from now on mSpatializerOutput is null checkVirtualizerClientRoutes(); } return NO_ERROR; } // ---------------------------------------------------------------------------- // AudioPolicyManager // ---------------------------------------------------------------------------- uint32_t AudioPolicyManager::nextAudioPortGeneration() { return mAudioPortGeneration++; } AudioPolicyManager::AudioPolicyManager(const sp& config, EngineInstance&& engine, AudioPolicyClientInterface *clientInterface) : mUidCached(AID_AUDIOSERVER), // no need to call getuid(), there's only one of us running. mConfig(config), mEngine(std::move(engine)), mpClientInterface(clientInterface), mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), mA2dpSuspended(false), mAudioPortGeneration(1), mBeaconMuteRefCount(0), mBeaconPlayingRefCount(0), mBeaconMuted(false), mTtsOutputAvailable(false), mMasterMono(false), mMusicEffectOutput(AUDIO_IO_HANDLE_NONE) { } status_t AudioPolicyManager::initialize() { if (mEngine == nullptr) { return NO_INIT; } mEngine->setObserver(this); status_t status = mEngine->initCheck(); if (status != NO_ERROR) { LOG_FATAL("Policy engine not initialized(err=%d)", status); return status; } // The actual device selection cache will be updated when calling `updateDevicesAndOutputs` // at the end of this function. mEngine->initializeDeviceSelectionCache(); mCommunnicationStrategy = mEngine->getProductStrategyForAttributes( mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL)); // after parsing the config, mConfig contain all known devices; // open all output streams needed to access attached devices onNewAudioModulesAvailableInt(nullptr /*newDevices*/); // make sure default device is reachable if (const auto defaultOutputDevice = mConfig->getDefaultOutputDevice(); defaultOutputDevice == nullptr || !mAvailableOutputDevices.contains(defaultOutputDevice)) { ALOGE_IF(defaultOutputDevice != nullptr, "Default device %s is unreachable", defaultOutputDevice->toString().c_str()); status = NO_INIT; } ALOGW_IF(mPrimaryOutput == nullptr, "The policy configuration does not declare a primary output"); // Silence ALOGV statements property_set("log.tag." LOG_TAG, "D"); updateDevicesAndOutputs(); return status; } AudioPolicyManager::~AudioPolicyManager() { for (size_t i = 0; i < mOutputs.size(); i++) { mOutputs.valueAt(i)->close(); } for (size_t i = 0; i < mInputs.size(); i++) { mInputs.valueAt(i)->close(); } mAvailableOutputDevices.clear(); mAvailableInputDevices.clear(); mOutputs.clear(); mInputs.clear(); mHwModules.clear(); mManualSurroundFormats.clear(); mConfig.clear(); } status_t AudioPolicyManager::initCheck() { return hasPrimaryOutput() ? NO_ERROR : NO_INIT; } // --- void AudioPolicyManager::onNewAudioModulesAvailable() { DeviceVector newDevices; onNewAudioModulesAvailableInt(&newDevices); if (!newDevices.empty()) { nextAudioPortGeneration(); mpClientInterface->onAudioPortListUpdate(); } } void AudioPolicyManager::onNewAudioModulesAvailableInt(DeviceVector *newDevices) { for (const auto& hwModule : mConfig->getHwModules()) { if (std::find(mHwModules.begin(), mHwModules.end(), hwModule) != mHwModules.end()) { continue; } if (hwModule->getHandle() == AUDIO_MODULE_HANDLE_NONE) { if (audio_module_handle_t handle = mpClientInterface->loadHwModule(hwModule->getName()); handle != AUDIO_MODULE_HANDLE_NONE) { hwModule->setHandle(handle); } else { ALOGW("could not load HW module %s", hwModule->getName()); continue; } } mHwModules.push_back(hwModule); // open all output streams needed to access attached devices. // direct outputs are closed immediately after checking the availability of attached devices // This also validates mAvailableOutputDevices list for (const auto& outProfile : hwModule->getOutputProfiles()) { if (!outProfile->canOpenNewIo()) { ALOGE("Invalid Output profile max open count %u for profile %s", outProfile->maxOpenCount, outProfile->getTagName().c_str()); continue; } if (!outProfile->hasSupportedDevices()) { ALOGW("Output profile contains no device on module %s", hwModule->getName()); continue; } if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_TTS) != 0 || (outProfile->getFlags() & AUDIO_OUTPUT_FLAG_ULTRASOUND) != 0) { mTtsOutputAvailable = true; } const DeviceVector &supportedDevices = outProfile->getSupportedDevices(); DeviceVector availProfileDevices = supportedDevices.filter(mConfig->getOutputDevices()); sp supportedDevice = 0; if (supportedDevices.contains(mConfig->getDefaultOutputDevice())) { supportedDevice = mConfig->getDefaultOutputDevice(); } else { // choose first device present in profile's SupportedDevices also part of // mAvailableOutputDevices. if (availProfileDevices.isEmpty()) { continue; } supportedDevice = availProfileDevices.itemAt(0); } if (!mConfig->getOutputDevices().contains(supportedDevice)) { continue; } if (outProfile->isMmap() && !outProfile->hasDynamicAudioProfile() && availProfileDevices.areAllDevicesAttached()) { ALOGV("%s skip opening output for mmap profile %s", __func__, outProfile->getTagName().c_str()); continue; } sp outputDesc = new SwAudioOutputDescriptor(outProfile, mpClientInterface); audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE; audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER; status_t status = outputDesc->open(nullptr /* halConfig */, nullptr /* mixerConfig */, DeviceVector(supportedDevice), AUDIO_STREAM_DEFAULT, &flags, &output, attributes); if (status != NO_ERROR) { ALOGW("Cannot open output stream for devices %s on hw module %s", supportedDevice->toString().c_str(), hwModule->getName()); continue; } for (const auto &device : availProfileDevices) { // give a valid ID to an attached device once confirmed it is reachable if (!device->isAttached()) { device->attach(hwModule); mAvailableOutputDevices.add(device); device->setEncapsulationInfoFromHal(mpClientInterface); if (newDevices) newDevices->add(device); setEngineDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE); } } if (mPrimaryOutput == nullptr && outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) { mPrimaryOutput = outputDesc; mPrimaryModuleHandle = mPrimaryOutput->getModuleHandle(); } if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { outputDesc->close(); } else { addOutput(output, outputDesc); setOutputDevices(__func__, outputDesc, DeviceVector(supportedDevice), true, 0, NULL); } } // open input streams needed to access attached devices to validate // mAvailableInputDevices list for (const auto& inProfile : hwModule->getInputProfiles()) { if (!inProfile->canOpenNewIo()) { ALOGE("Invalid Input profile max open count %u for profile %s", inProfile->maxOpenCount, inProfile->getTagName().c_str()); continue; } if (!inProfile->hasSupportedDevices()) { ALOGW("Input profile contains no device on module %s", hwModule->getName()); continue; } // chose first device present in profile's SupportedDevices also part of // available input devices const DeviceVector &supportedDevices = inProfile->getSupportedDevices(); DeviceVector availProfileDevices = supportedDevices.filter(mConfig->getInputDevices()); if (availProfileDevices.isEmpty()) { ALOGV("%s: Input device list is empty! for profile %s", __func__, inProfile->getTagName().c_str()); continue; } if (inProfile->isMmap() && !inProfile->hasDynamicAudioProfile() && availProfileDevices.areAllDevicesAttached()) { ALOGV("%s skip opening input for mmap profile %s", __func__, inProfile->getTagName().c_str()); continue; } sp inputDesc = new AudioInputDescriptor(inProfile, mpClientInterface); audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; status_t status = inputDesc->open(nullptr, availProfileDevices.itemAt(0), AUDIO_SOURCE_MIC, (audio_input_flags_t) inProfile->getFlags(), &input); if (status != NO_ERROR) { ALOGW("%s: Cannot open input stream for device %s for profile %s on hw module %s", __func__, availProfileDevices.toString().c_str(), inProfile->getTagName().c_str(), hwModule->getName()); continue; } for (const auto &device : availProfileDevices) { // give a valid ID to an attached device once confirmed it is reachable if (!device->isAttached()) { device->attach(hwModule); device->importAudioPortAndPickAudioProfile(inProfile, true); mAvailableInputDevices.add(device); if (newDevices) newDevices->add(device); setEngineDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE); } } inputDesc->close(); } } // Check if spatializer outputs can be closed until used. // mOutputs vector never contains duplicated outputs at this point. std::vector outputsClosed; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if ((desc->mFlags & AUDIO_OUTPUT_FLAG_SPATIALIZER) != 0 && !isOutputOnlyAvailableRouteToSomeDevice(desc)) { outputsClosed.push_back(desc->mIoHandle); nextAudioPortGeneration(); ssize_t index = mAudioPatches.indexOfKey(desc->getPatchHandle()); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); (void) /*status_t status*/ mpClientInterface->releaseAudioPatch( patchDesc->getAfHandle(), 0); mAudioPatches.removeItemsAt(index); mpClientInterface->onAudioPatchListUpdate(); } desc->close(); } } for (auto output : outputsClosed) { removeOutput(output); } } void AudioPolicyManager::addOutput(audio_io_handle_t output, const sp& outputDesc) { mOutputs.add(output, outputDesc); applyStreamVolumes(outputDesc, DeviceTypeSet(), 0 /* delayMs */, true /* force */); updateMono(output); // update mono status when adding to output list selectOutputForMusicEffects(); nextAudioPortGeneration(); } void AudioPolicyManager::removeOutput(audio_io_handle_t output) { if (mPrimaryOutput != 0 && mPrimaryOutput == mOutputs.valueFor(output)) { ALOGV("%s: removing primary output", __func__); mPrimaryOutput = nullptr; } mOutputs.removeItem(output); selectOutputForMusicEffects(); } void AudioPolicyManager::addInput(audio_io_handle_t input, const sp& inputDesc) { mInputs.add(input, inputDesc); nextAudioPortGeneration(); } status_t AudioPolicyManager::checkOutputsForDevice(const sp& device, audio_policy_dev_state_t state, SortedVector& outputs) { audio_devices_t deviceType = device->type(); const String8 &address = String8(device->address().c_str()); sp desc; if (audio_device_is_digital(deviceType)) { // erase all current sample rates, formats and channel masks device->clearAudioProfiles(); } if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { // first call getAudioPort to get the supported attributes from the HAL struct audio_port_v7 port = {}; device->toAudioPort(&port); status_t status = mpClientInterface->getAudioPort(&port); if (status == NO_ERROR) { device->importAudioPort(port); } // then list already open outputs that can be routed to this device for (size_t i = 0; i < mOutputs.size(); i++) { desc = mOutputs.valueAt(i); if (!desc->isDuplicated() && desc->supportsDevice(device) && desc->devicesSupportEncodedFormats({deviceType})) { ALOGV("checkOutputsForDevice(): adding opened output %d on device %s", mOutputs.keyAt(i), device->toString().c_str()); outputs.add(mOutputs.keyAt(i)); } } // then look for output profiles that can be routed to this device SortedVector< sp > profiles; for (const auto& hwModule : mHwModules) { for (size_t j = 0; j < hwModule->getOutputProfiles().size(); j++) { sp profile = hwModule->getOutputProfiles()[j]; if (profile->supportsDevice(device)) { profiles.add(profile); ALOGV("%s(): adding profile %s from module %s", __func__, profile->getTagName().c_str(), hwModule->getName()); } } } ALOGV(" found %zu profiles, %zu outputs", profiles.size(), outputs.size()); if (profiles.isEmpty() && outputs.isEmpty()) { ALOGW("checkOutputsForDevice(): No output available for device %04x", deviceType); return BAD_VALUE; } // open outputs for matching profiles if needed. Direct outputs are also opened to // query for dynamic parameters and will be closed later by setDeviceConnectionState() for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { sp profile = profiles[profile_index]; // nothing to do if one output is already opened for this profile size_t j; for (j = 0; j < outputs.size(); j++) { desc = mOutputs.valueFor(outputs.itemAt(j)); if (!desc->isDuplicated() && desc->mProfile == profile) { // matching profile: save the sample rates, format and channel masks supported // by the profile in our device descriptor if (audio_device_is_digital(deviceType)) { device->importAudioPortAndPickAudioProfile(profile); } break; } } if (j != outputs.size()) { continue; } if (profile->isMmap() && !profile->hasDynamicAudioProfile()) { ALOGV("%s skip opening output for mmap profile %s", __func__, profile->getTagName().c_str()); continue; } if (!profile->canOpenNewIo()) { ALOGW("Max Output number %u already opened for this profile %s", profile->maxOpenCount, profile->getTagName().c_str()); continue; } ALOGV("opening output for device %08x with params %s profile %p name %s", deviceType, address.c_str(), profile.get(), profile->getName().c_str()); desc = openOutputWithProfileAndDevice(profile, DeviceVector(device)); audio_io_handle_t output = desc == nullptr ? AUDIO_IO_HANDLE_NONE : desc->mIoHandle; if (output == AUDIO_IO_HANDLE_NONE) { ALOGW("checkOutputsForDevice() could not open output for device %x", deviceType); profiles.removeAt(profile_index); profile_index--; } else { outputs.add(output); // Load digital format info only for digital devices if (audio_device_is_digital(deviceType)) { // TODO: when getAudioPort is ready, it may not be needed to import the audio // port but just pick audio profile device->importAudioPortAndPickAudioProfile(profile); } if (device_distinguishes_on_address(deviceType)) { ALOGV("checkOutputsForDevice(): setOutputDevices %s", device->toString().c_str()); setOutputDevices(__func__, desc, DeviceVector(device), true/*force*/, 0/*delay*/, NULL/*patch handle*/); } ALOGV("checkOutputsForDevice(): adding output %d", output); } } if (profiles.isEmpty()) { ALOGW("checkOutputsForDevice(): No output available for device %04x", deviceType); return BAD_VALUE; } } else { // Disconnect // check if one opened output is not needed any more after disconnecting one device for (size_t i = 0; i < mOutputs.size(); i++) { desc = mOutputs.valueAt(i); if (!desc->isDuplicated()) { // exact match on device if (device_distinguishes_on_address(deviceType) && desc->supportsDevice(device) && desc->containsSingleDeviceSupportingEncodedFormats(device)) { outputs.add(mOutputs.keyAt(i)); } else if (!mAvailableOutputDevices.containsAtLeastOne(desc->supportedDevices())) { ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); } } } // Clear any profiles associated with the disconnected device. for (const auto& hwModule : mHwModules) { for (size_t j = 0; j < hwModule->getOutputProfiles().size(); j++) { sp profile = hwModule->getOutputProfiles()[j]; if (!profile->supportsDevice(device)) { continue; } ALOGV("%s(): clearing direct output profile %s on module %s", __func__, profile->getTagName().c_str(), hwModule->getName()); profile->clearAudioProfiles(); if (!profile->hasDynamicAudioProfile()) { continue; } // When a device is disconnected, if there is an IOProfile that contains dynamic // profiles and supports the disconnected device, call getAudioPort to repopulate // the capabilities of the devices that is supported by the IOProfile. for (const auto& supportedDevice : profile->getSupportedDevices()) { if (supportedDevice == device || !mAvailableOutputDevices.contains(supportedDevice)) { continue; } struct audio_port_v7 port; supportedDevice->toAudioPort(&port); status_t status = mpClientInterface->getAudioPort(&port); if (status == NO_ERROR) { supportedDevice->importAudioPort(port); } } } } } return NO_ERROR; } status_t AudioPolicyManager::checkInputsForDevice(const sp& device, audio_policy_dev_state_t state) { if (audio_device_is_digital(device->type())) { // erase all current sample rates, formats and channel masks device->clearAudioProfiles(); } if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { sp desc; // first call getAudioPort to get the supported attributes from the HAL struct audio_port_v7 port = {}; device->toAudioPort(&port); status_t status = mpClientInterface->getAudioPort(&port); if (status == NO_ERROR) { device->importAudioPort(port); } // look for input profiles that can be routed to this device SortedVector< sp > profiles; for (const auto& hwModule : mHwModules) { for (size_t profile_index = 0; profile_index < hwModule->getInputProfiles().size(); profile_index++) { sp profile = hwModule->getInputProfiles()[profile_index]; if (profile->supportsDevice(device)) { profiles.add(profile); ALOGV("%s : adding profile %s from module %s", __func__, profile->getTagName().c_str(), hwModule->getName()); } } } if (profiles.isEmpty()) { ALOGW("%s: No input profile available for device %s", __func__, device->toString().c_str()); return BAD_VALUE; } // open inputs for matching profiles if needed. Direct inputs are also opened to // query for dynamic parameters and will be closed later by setDeviceConnectionState() for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) { sp profile = profiles[profile_index]; // nothing to do if one input is already opened for this profile size_t input_index; for (input_index = 0; input_index < mInputs.size(); input_index++) { desc = mInputs.valueAt(input_index); if (desc->mProfile == profile) { if (audio_device_is_digital(device->type())) { device->importAudioPortAndPickAudioProfile(profile); } break; } } if (input_index != mInputs.size()) { continue; } if (profile->isMmap() && !profile->hasDynamicAudioProfile()) { ALOGV("%s skip opening input for mmap profile %s", __func__, profile->getTagName().c_str()); continue; } if (!profile->canOpenNewIo()) { ALOGW("%s Max Input number %u already opened for this profile %s", __func__, profile->maxOpenCount, profile->getTagName().c_str()); continue; } desc = new AudioInputDescriptor(profile, mpClientInterface); audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; ALOGV("%s opening input for profile %s", __func__, profile->getTagName().c_str()); status = desc->open(nullptr, device, AUDIO_SOURCE_MIC, (audio_input_flags_t) profile->getFlags(), &input); if (status == NO_ERROR) { const String8& address = String8(device->address().c_str()); if (!address.empty()) { char *param = audio_device_address_to_parameter(device->type(), address); mpClientInterface->setParameters(input, String8(param)); free(param); } updateAudioProfiles(device, input, profile); if (!profile->hasValidAudioProfile()) { ALOGW("%s direct input missing param for profile %s", __func__, profile->getTagName().c_str()); desc->close(); input = AUDIO_IO_HANDLE_NONE; } if (input != AUDIO_IO_HANDLE_NONE) { addInput(input, desc); } } // endif input != 0 if (input == AUDIO_IO_HANDLE_NONE) { ALOGW("%s could not open input for device %s on profile %s", __func__, device->toString().c_str(), profile->getTagName().c_str()); profiles.removeAt(profile_index); profile_index--; } else { if (audio_device_is_digital(device->type())) { device->importAudioPortAndPickAudioProfile(profile); } ALOGV("%s: adding input %d for profile %s", __func__, input, profile->getTagName().c_str()); if (checkCloseInput(desc)) { ALOGV("%s: closing input %d for profile %s", __func__, input, profile->getTagName().c_str()); closeInput(input); } } } // end scan profiles if (profiles.isEmpty()) { ALOGW("%s: No input available for device %s", __func__, device->toString().c_str()); return BAD_VALUE; } } else { // Disconnect // Clear any profiles associated with the disconnected device. for (const auto& hwModule : mHwModules) { for (size_t profile_index = 0; profile_index < hwModule->getInputProfiles().size(); profile_index++) { sp profile = hwModule->getInputProfiles()[profile_index]; if (profile->supportsDevice(device)) { ALOGV("%s: clearing direct input profile %s on module %s", __func__, profile->getTagName().c_str(), hwModule->getName()); profile->clearAudioProfiles(); } } } } // end disconnect return NO_ERROR; } void AudioPolicyManager::closeOutput(audio_io_handle_t output) { ALOGV("closeOutput(%d)", output); sp closingOutput = mOutputs.valueFor(output); if (closingOutput == NULL) { ALOGW("closeOutput() unknown output %d", output); return; } const bool closingOutputWasActive = closingOutput->isActive(); mPolicyMixes.closeOutput(closingOutput, mOutputs); // look for duplicated outputs connected to the output being removed. for (size_t i = 0; i < mOutputs.size(); i++) { sp dupOutput = mOutputs.valueAt(i); if (dupOutput->isDuplicated() && (dupOutput->mOutput1 == closingOutput || dupOutput->mOutput2 == closingOutput)) { sp remainingOutput = dupOutput->mOutput1 == closingOutput ? dupOutput->mOutput2 : dupOutput->mOutput1; // As all active tracks on duplicated output will be deleted, // and as they were also referenced on the other output, the reference // count for their stream type must be adjusted accordingly on // the other output. const bool wasActive = remainingOutput->isActive(); // Note: no-op on the closing output where all clients has already been set inactive dupOutput->setAllClientsInactive(); // stop() will be a no op if the output is still active but is needed in case all // active streams refcounts where cleared above if (wasActive) { remainingOutput->stop(); } audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); mpClientInterface->closeOutput(duplicatedOutput); removeOutput(duplicatedOutput); } } nextAudioPortGeneration(); ssize_t index = mAudioPatches.indexOfKey(closingOutput->getPatchHandle()); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); (void) /*status_t status*/ mpClientInterface->releaseAudioPatch( patchDesc->getAfHandle(), 0); mAudioPatches.removeItemsAt(index); mpClientInterface->onAudioPatchListUpdate(); } if (closingOutputWasActive) { closingOutput->stop(); } closingOutput->close(); if (closingOutput->isBitPerfect()) { for (const auto device : closingOutput->devices()) { device->setPreferredConfig(nullptr); } } removeOutput(output); mPreviousOutputs = mOutputs; if (closingOutput == mSpatializerOutput) { mSpatializerOutput.clear(); } // MSD patches may have been released to support a non-MSD direct output. Reset MSD patch if // no direct outputs are open. if (!getMsdAudioOutDevices().isEmpty()) { bool directOutputOpen = false; for (size_t i = 0; i < mOutputs.size(); i++) { if (mOutputs[i]->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { directOutputOpen = true; break; } } if (!directOutputOpen) { ALOGV("no direct outputs open, reset MSD patches"); // TODO: The MSD patches to be established here may differ to current MSD patches due to // how output devices for patching are resolved. Avoid by caching and reusing the // arguments to mEngine->getOutputDevicesForAttributes() when resolving which output // devices to patch to. This may be complicated by the fact that devices may become // unavailable. setMsdOutputPatches(); } } if (closingOutput->mPreferredAttrInfo != nullptr) { closingOutput->mPreferredAttrInfo->resetActiveClient(); } } void AudioPolicyManager::closeInput(audio_io_handle_t input) { ALOGV("closeInput(%d)", input); sp inputDesc = mInputs.valueFor(input); if (inputDesc == NULL) { ALOGW("closeInput() unknown input %d", input); return; } nextAudioPortGeneration(); sp device = inputDesc->getDevice(); ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); (void) /*status_t status*/ mpClientInterface->releaseAudioPatch( patchDesc->getAfHandle(), 0); mAudioPatches.removeItemsAt(index); mpClientInterface->onAudioPatchListUpdate(); } mEffects.putOrphanEffectsForIo(input); inputDesc->close(); mInputs.removeItem(input); DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices(); if (primaryInputDevices.contains(device) && mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) { mpClientInterface->setSoundTriggerCaptureState(false); } } SortedVector AudioPolicyManager::getOutputsForDevices( const DeviceVector &devices, const SwAudioOutputCollection& openOutputs) { SortedVector outputs; ALOGVV("%s() devices %s", __func__, devices.toString().c_str()); for (size_t i = 0; i < openOutputs.size(); i++) { ALOGVV("output %zu isDuplicated=%d device=%s", i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices().toString().c_str()); if (openOutputs.valueAt(i)->supportsAllDevices(devices) && openOutputs.valueAt(i)->devicesSupportEncodedFormats(devices.types())) { ALOGVV("%s() found output %d", __func__, openOutputs.keyAt(i)); outputs.add(openOutputs.keyAt(i)); } } return outputs; } void AudioPolicyManager::checkForDeviceAndOutputChanges(std::function onOutputsChecked) { // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP // output is suspended before any tracks are moved to it checkA2dpSuspend(); checkOutputForAllStrategies(); checkSecondaryOutputs(); if (onOutputsChecked != nullptr && onOutputsChecked()) checkA2dpSuspend(); updateDevicesAndOutputs(); if (mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD) != 0) { // TODO: The MSD patches to be established here may differ to current MSD patches due to how // output devices for patching are resolved. Nevertheless, AudioTracks affected by device // configuration changes will ultimately be rerouted correctly. We can still avoid // unnecessary rerouting by caching and reusing the arguments to // mEngine->getOutputDevicesForAttributes() when resolving which output devices to patch to. // This may be complicated by the fact that devices may become unavailable. setMsdOutputPatches(); } // an event that changed routing likely occurred, inform upper layers mpClientInterface->onRoutingUpdated(); } bool AudioPolicyManager::followsSameRouting(const audio_attributes_t &lAttr, const audio_attributes_t &rAttr) const { return mEngine->getProductStrategyForAttributes(lAttr) == mEngine->getProductStrategyForAttributes(rAttr); } void AudioPolicyManager::checkAudioSourceForAttributes(const audio_attributes_t &attr) { for (size_t i = 0; i < mAudioSources.size(); i++) { sp sourceDesc = mAudioSources.valueAt(i); if (sourceDesc != nullptr && followsSameRouting(attr, sourceDesc->attributes()) && sourceDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE && !sourceDesc->isCallRx() && !sourceDesc->isInternal()) { connectAudioSource(sourceDesc, 0 /*delayMs*/); } } } void AudioPolicyManager::clearAudioSourcesForOutput(audio_io_handle_t output) { for (size_t i = 0; i < mAudioSources.size(); i++) { sp sourceDesc = mAudioSources.valueAt(i); if (sourceDesc != nullptr && sourceDesc->swOutput().promote() != nullptr && sourceDesc->swOutput().promote()->mIoHandle == output) { disconnectAudioSource(sourceDesc); } } } void AudioPolicyManager::checkOutputForAttributes(const audio_attributes_t &attr) { auto psId = mEngine->getProductStrategyForAttributes(attr); DeviceVector oldDevices = mEngine->getOutputDevicesForAttributes(attr, 0, true /*fromCache*/); DeviceVector newDevices = mEngine->getOutputDevicesForAttributes(attr, 0, false /*fromCache*/); SortedVector srcOutputs = getOutputsForDevices(oldDevices, mPreviousOutputs); SortedVector dstOutputs = getOutputsForDevices(newDevices, mOutputs); uint32_t maxLatency = 0; bool unneededUsePrimaryOutputFromPolicyMixes = false; std::vector> invalidatedOutputs; // take into account dynamic audio policies related changes: if a client is now associated // to a different policy mix than at creation time, invalidate corresponding stream // invalidate clients on outputs that do not support all the newly selected devices for the // strategy for (size_t i = 0; i < mPreviousOutputs.size(); i++) { const sp& desc = mPreviousOutputs.valueAt(i); if (desc->isDuplicated() || desc->getClientCount() == 0) { continue; } for (const sp& client : desc->getClientIterable()) { if (mEngine->getProductStrategyForAttributes(client->attributes()) != psId) { continue; } if (!desc->supportsAllDevices(newDevices)) { invalidatedOutputs.push_back(desc); break; } sp primaryMix; status_t status = mPolicyMixes.getOutputForAttr(client->attributes(), client->config(), client->uid(), client->session(), client->flags(), mAvailableOutputDevices, nullptr /* requestedDevice */, primaryMix, nullptr /* secondaryMixes */, unneededUsePrimaryOutputFromPolicyMixes); if (status == OK) { if (client->getPrimaryMix() != primaryMix || client->hasLostPrimaryMix()) { if (desc->isStrategyActive(psId) && maxLatency < desc->latency()) { maxLatency = desc->latency(); } invalidatedOutputs.push_back(desc); break; } } } } if (srcOutputs != dstOutputs || !invalidatedOutputs.empty()) { // get maximum latency of all source outputs to determine the minimum mute time guaranteeing // audio from invalidated tracks will be rendered when unmuting for (audio_io_handle_t srcOut : srcOutputs) { sp desc = mPreviousOutputs.valueFor(srcOut); if (desc == nullptr) continue; if (desc->isStrategyActive(psId) && maxLatency < desc->latency()) { maxLatency = desc->latency(); } bool invalidate = false; for (auto client : desc->clientsList(false /*activeOnly*/)) { if (desc->isDuplicated() || !desc->mProfile->isDirectOutput()) { // a client on a non direct outputs has necessarily a linear PCM format // so we can call selectOutput() safely const audio_io_handle_t newOutput = selectOutput(dstOutputs, client->flags(), client->config().format, client->config().channel_mask, client->config().sample_rate, client->session()); if (newOutput != srcOut) { invalidate = true; break; } } else { sp profile = getProfileForOutput(newDevices, client->config().sample_rate, client->config().format, client->config().channel_mask, client->flags(), true /* directOnly */); if (profile != desc->mProfile) { invalidate = true; break; } } } // mute strategy while moving tracks from one output to another if (invalidate) { invalidatedOutputs.push_back(desc); if (desc->isStrategyActive(psId)) { setStrategyMute(psId, true, desc); setStrategyMute(psId, false, desc, maxLatency * LATENCY_MUTE_FACTOR, newDevices.types()); } } sp source = getSourceForAttributesOnOutput(srcOut, attr); if (source != nullptr && !source->isCallRx() && !source->isInternal()) { connectAudioSource(source, 0 /*delayMs*/); } } ALOGV_IF(!(srcOutputs.isEmpty() || dstOutputs.isEmpty()), "%s: strategy %d, moving from output %s to output %s", __func__, psId, std::to_string(srcOutputs[0]).c_str(), std::to_string(dstOutputs[0]).c_str()); // Move effects associated to this stream from previous output to new output if (followsSameRouting(attr, attributes_initializer(AUDIO_USAGE_MEDIA))) { selectOutputForMusicEffects(); } // Move tracks associated to this stream (and linked) from previous output to new output if (!invalidatedOutputs.empty()) { invalidateStreams(mEngine->getStreamTypesForProductStrategy(psId)); for (sp desc : invalidatedOutputs) { desc->setTracksInvalidatedStatusByStrategy(psId); } } } } void AudioPolicyManager::checkOutputForAllStrategies() { for (const auto &strategy : mEngine->getOrderedProductStrategies()) { auto attributes = mEngine->getAllAttributesForProductStrategy(strategy).front(); checkOutputForAttributes(attributes); checkAudioSourceForAttributes(attributes); } } void AudioPolicyManager::checkSecondaryOutputs() { PortHandleVector clientsToInvalidate; TrackSecondaryOutputsMap trackSecondaryOutputs; bool unneededUsePrimaryOutputFromPolicyMixes = false; for (size_t i = 0; i < mOutputs.size(); i++) { const sp& outputDescriptor = mOutputs[i]; for (const sp& client : outputDescriptor->getClientIterable()) { sp primaryMix; std::vector> secondaryMixes; status_t status = mPolicyMixes.getOutputForAttr(client->attributes(), client->config(), client->uid(), client->session(), client->flags(), mAvailableOutputDevices, nullptr /* requestedDevice */, primaryMix, &secondaryMixes, unneededUsePrimaryOutputFromPolicyMixes); std::vector> secondaryDescs; for (auto &secondaryMix : secondaryMixes) { sp outputDesc = secondaryMix->getOutput(); if (outputDesc != nullptr && outputDesc->mIoHandle != AUDIO_IO_HANDLE_NONE && outputDesc != outputDescriptor) { secondaryDescs.push_back(outputDesc); } } if (status != OK && (client->flags() & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == AUDIO_OUTPUT_FLAG_NONE) { // When it failed to query secondary output, only invalidate the client that is not // MMAP. The reason is that MMAP stream will not support secondary output. clientsToInvalidate.push_back(client->portId()); } else if (!std::equal( client->getSecondaryOutputs().begin(), client->getSecondaryOutputs().end(), secondaryDescs.begin(), secondaryDescs.end())) { if (client->flags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD || !audio_is_linear_pcm(client->config().format)) { // If the format is not PCM, the tracks should be invalidated to get correct // behavior when the secondary output is changed. clientsToInvalidate.push_back(client->portId()); } else { std::vector> weakSecondaryDescs; std::vector secondaryOutputIds; for (const auto &secondaryDesc: secondaryDescs) { secondaryOutputIds.push_back(secondaryDesc->mIoHandle); weakSecondaryDescs.push_back(secondaryDesc); } trackSecondaryOutputs.emplace(client->portId(), secondaryOutputIds); client->setSecondaryOutputs(std::move(weakSecondaryDescs)); } } } } if (!trackSecondaryOutputs.empty()) { mpClientInterface->updateSecondaryOutputs(trackSecondaryOutputs); } if (!clientsToInvalidate.empty()) { ALOGD("%s Invalidate clients due to fail getting output for attr", __func__); mpClientInterface->invalidateTracks(clientsToInvalidate); } } bool AudioPolicyManager::isScoRequestedForComm() const { AudioDeviceTypeAddrVector devices; mEngine->getDevicesForRoleAndStrategy(mCommunnicationStrategy, DEVICE_ROLE_PREFERRED, devices); for (const auto &device : devices) { if (audio_is_bluetooth_out_sco_device(device.mType)) { return true; } } return false; } bool AudioPolicyManager::isHearingAidUsedForComm() const { DeviceVector devices = mEngine->getOutputDevicesForStream(AUDIO_STREAM_VOICE_CALL, true /*fromCache*/); for (const auto &device : devices) { if (device->type() == AUDIO_DEVICE_OUT_HEARING_AID) { return true; } } return false; } void AudioPolicyManager::checkA2dpSuspend() { audio_io_handle_t a2dpOutput = mOutputs.getA2dpOutput(); if (a2dpOutput == 0 || mOutputs.isA2dpOffloadedOnPrimary()) { mA2dpSuspended = false; return; } bool isScoConnected = (mAvailableInputDevices.types().count(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) != 0 || !Intersection(mAvailableOutputDevices.types(), getAudioDeviceOutAllScoSet()).empty()); bool isScoRequested = isScoRequestedForComm(); // if suspended, restore A2DP output if: // ((SCO device is NOT connected) || // ((SCO is not requested) && // (phone state is NOT in call) && (phone state is NOT ringing))) // // if not suspended, suspend A2DP output if: // (SCO device is connected) && // ((SCO is requested) || // ((phone state is in call) || (phone state is ringing))) // if (mA2dpSuspended) { if (!isScoConnected || (!isScoRequested && (mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) && (mEngine->getPhoneState() != AUDIO_MODE_RINGTONE))) { mpClientInterface->restoreOutput(a2dpOutput); mA2dpSuspended = false; } } else { if (isScoConnected && (isScoRequested || (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) || (mEngine->getPhoneState() == AUDIO_MODE_RINGTONE))) { mpClientInterface->suspendOutput(a2dpOutput); mA2dpSuspended = true; } } } DeviceVector AudioPolicyManager::getNewOutputDevices(const sp& outputDesc, bool fromCache) { if (outputDesc == nullptr) { return DeviceVector{}; } ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle()); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); if (patchDesc->getUid() != mUidCached) { ALOGV("%s device %s forced by patch %d", __func__, outputDesc->devices().toString().c_str(), outputDesc->getPatchHandle()); return outputDesc->devices(); } } // Do not retrieve engine device for outputs through MSD // TODO: support explicit routing requests by resetting MSD patch to engine device. if (outputDesc->devices() == getMsdAudioOutDevices()) { return outputDesc->devices(); } // Honor explicit routing requests only if no client using default routing is active on this // input: a specific app can not force routing for other apps by setting a preferred device. bool active; // unused sp device = findPreferredDevice(outputDesc, PRODUCT_STRATEGY_NONE, active, mAvailableOutputDevices); if (device != nullptr) { return DeviceVector(device); } // Legacy Engine cannot take care of bus devices and mix, so we need to handle the conflict // of setForceUse / Default Bus device here device = mPolicyMixes.getDeviceAndMixForOutput(outputDesc, mAvailableOutputDevices); if (device != nullptr) { return DeviceVector(device); } DeviceVector devices; for (const auto &productStrategy : mEngine->getOrderedProductStrategies()) { StreamTypeVector streams = mEngine->getStreamTypesForProductStrategy(productStrategy); auto hasStreamActive = [&](auto stream) { return hasStream(streams, stream) && isStreamActive(stream, 0); }; auto doGetOutputDevicesForVoice = [&]() { return hasVoiceStream(streams) && (outputDesc == mPrimaryOutput || outputDesc->isActive(toVolumeSource(AUDIO_STREAM_VOICE_CALL, false))) && (isInCall() || mOutputs.isStrategyActiveOnSameModule(productStrategy, outputDesc)) && !isStreamActive(AUDIO_STREAM_ENFORCED_AUDIBLE, 0); }; // With low-latency playing on speaker, music on WFD, when the first low-latency // output is stopped, getNewOutputDevices checks for a product strategy // from the list, as STRATEGY_SONIFICATION comes prior to STRATEGY_MEDIA. // If an ALARM or ENFORCED_AUDIBLE stream is supported by the product strategy, // devices are returned for STRATEGY_SONIFICATION without checking whether the // stream is associated to the output descriptor. if (doGetOutputDevicesForVoice() || outputDesc->isStrategyActive(productStrategy) || ((hasStreamActive(AUDIO_STREAM_ALARM) || hasStreamActive(AUDIO_STREAM_ENFORCED_AUDIBLE)) && mOutputs.isStrategyActiveOnSameModule(productStrategy, outputDesc))) { // Retrieval of devices for voice DL is done on primary output profile, cannot // check the route (would force modifying configuration file for this profile) auto attr = mEngine->getAllAttributesForProductStrategy(productStrategy).front(); devices = mEngine->getOutputDevicesForAttributes(attr, nullptr, fromCache); break; } } ALOGV("%s selected devices %s", __func__, devices.toString().c_str()); return devices; } sp AudioPolicyManager::getNewInputDevice( const sp& inputDesc) { sp device; ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); if (patchDesc->getUid() != mUidCached) { ALOGV("getNewInputDevice() device %s forced by patch %d", inputDesc->getDevice()->toString().c_str(), inputDesc->getPatchHandle()); return inputDesc->getDevice(); } } // Honor explicit routing requests only if no client using default routing is active on this // input: a specific app can not force routing for other apps by setting a preferred device. bool active; device = findPreferredDevice(inputDesc, AUDIO_SOURCE_DEFAULT, active, mAvailableInputDevices); if (device != nullptr) { return device; } // If we are not in call and no client is active on this input, this methods returns // a null sp<>, causing the patch on the input stream to be released. audio_attributes_t attributes; uid_t uid; audio_session_t session; sp topClient = inputDesc->getHighestPriorityClient(); if (topClient != nullptr) { attributes = topClient->attributes(); uid = topClient->uid(); session = topClient->session(); } else { attributes = { .source = AUDIO_SOURCE_DEFAULT }; uid = 0; session = AUDIO_SESSION_NONE; } if (attributes.source == AUDIO_SOURCE_DEFAULT && isInCall()) { attributes.source = AUDIO_SOURCE_VOICE_COMMUNICATION; } if (attributes.source != AUDIO_SOURCE_DEFAULT) { device = mEngine->getInputDeviceForAttributes(attributes, uid, session); } return device; } bool AudioPolicyManager::streamsMatchForvolume(audio_stream_type_t stream1, audio_stream_type_t stream2) { return (stream1 == stream2); } status_t AudioPolicyManager::getDevicesForAttributes( const audio_attributes_t &attr, AudioDeviceTypeAddrVector *devices, bool forVolume) { if (devices == nullptr) { return BAD_VALUE; } DeviceVector curDevices; if (status_t status = getDevicesForAttributes(attr, curDevices, forVolume); status != OK) { return status; } for (const auto& device : curDevices) { devices->push_back(device->getDeviceTypeAddr()); } return NO_ERROR; } void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) { switch(stream) { case AUDIO_STREAM_MUSIC: checkOutputForAttributes(attributes_initializer(AUDIO_USAGE_NOTIFICATION)); updateDevicesAndOutputs(); break; default: break; } } uint32_t AudioPolicyManager::handleEventForBeacon(int event) { // skip beacon mute management if a dedicated TTS output is available if (mTtsOutputAvailable) { return 0; } switch(event) { case STARTING_OUTPUT: mBeaconMuteRefCount++; break; case STOPPING_OUTPUT: if (mBeaconMuteRefCount > 0) { mBeaconMuteRefCount--; } break; case STARTING_BEACON: mBeaconPlayingRefCount++; break; case STOPPING_BEACON: if (mBeaconPlayingRefCount > 0) { mBeaconPlayingRefCount--; } break; } if (mBeaconMuteRefCount > 0) { // any playback causes beacon to be muted return setBeaconMute(true); } else { // no other playback: unmute when beacon starts playing, mute when it stops return setBeaconMute(mBeaconPlayingRefCount == 0); } } uint32_t AudioPolicyManager::setBeaconMute(bool mute) { ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d", mute, mBeaconMuteRefCount, mBeaconPlayingRefCount); // keep track of muted state to avoid repeating mute/unmute operations if (mBeaconMuted != mute) { // mute/unmute AUDIO_STREAM_TTS on all outputs ALOGV("\t muting %d", mute); uint32_t maxLatency = 0; auto ttsVolumeSource = toVolumeSource(AUDIO_STREAM_TTS, false); if (ttsVolumeSource == VOLUME_SOURCE_NONE) { ALOGV("\t no tts volume source available"); return 0; } for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); setVolumeSourceMute(ttsVolumeSource, mute/*on*/, desc, 0 /*delay*/, DeviceTypeSet()); const uint32_t latency = desc->latency() * 2; if (desc->isActive(latency * 2) && latency > maxLatency) { maxLatency = latency; } } mBeaconMuted = mute; return maxLatency; } return 0; } void AudioPolicyManager::updateDevicesAndOutputs() { mEngine->updateDeviceSelectionCache(); mPreviousOutputs = mOutputs; } uint32_t AudioPolicyManager::checkDeviceMuteStrategies(const sp& outputDesc, const DeviceVector &prevDevices, uint32_t delayMs) { // mute/unmute strategies using an incompatible device combination // if muting, wait for the audio in pcm buffer to be drained before proceeding // if unmuting, unmute only after the specified delay if (outputDesc->isDuplicated()) { return 0; } uint32_t muteWaitMs = 0; DeviceVector devices = outputDesc->devices(); bool shouldMute = outputDesc->isActive() && (devices.size() >= 2); auto productStrategies = mEngine->getOrderedProductStrategies(); for (const auto &productStrategy : productStrategies) { auto attributes = mEngine->getAllAttributesForProductStrategy(productStrategy).front(); DeviceVector curDevices = mEngine->getOutputDevicesForAttributes(attributes, nullptr, false/*fromCache*/); curDevices = curDevices.filter(outputDesc->supportedDevices()); bool mute = shouldMute && curDevices.containsAtLeastOne(devices) && curDevices != devices; bool doMute = false; if (mute && !outputDesc->isStrategyMutedByDevice(productStrategy)) { doMute = true; outputDesc->setStrategyMutedByDevice(productStrategy, true); } else if (!mute && outputDesc->isStrategyMutedByDevice(productStrategy)) { doMute = true; outputDesc->setStrategyMutedByDevice(productStrategy, false); } if (doMute) { for (size_t j = 0; j < mOutputs.size(); j++) { sp desc = mOutputs.valueAt(j); // skip output if it does not share any device with current output if (!desc->supportedDevices().containsAtLeastOne(outputDesc->supportedDevices())) { continue; } ALOGVV("%s() output %s %s (curDevice %s)", __func__, desc->info().c_str(), mute ? "muting" : "unmuting", curDevices.toString().c_str()); setStrategyMute(productStrategy, mute, desc, mute ? 0 : delayMs); if (desc->isStrategyActive(productStrategy)) { if (mute) { // FIXME: should not need to double latency if volume could be applied // immediately by the audioflinger mixer. We must account for the delay // between now and the next time the audioflinger thread for this output // will process a buffer (which corresponds to one buffer size, // usually 1/2 or 1/4 of the latency). if (muteWaitMs < desc->latency() * 2) { muteWaitMs = desc->latency() * 2; } } } } } } // temporary mute output if device selection changes to avoid volume bursts due to // different per device volumes if (outputDesc->isActive() && (devices != prevDevices)) { uint32_t tempMuteWaitMs = outputDesc->latency() * 2; if (muteWaitMs < tempMuteWaitMs) { muteWaitMs = tempMuteWaitMs; } // If recommended duration is defined, replace temporary mute duration to avoid // truncated notifications at beginning, which depends on duration of changing path in HAL. // Otherwise, temporary mute duration is conservatively set to 4 times the reported latency. uint32_t tempRecommendedMuteDuration = outputDesc->getRecommendedMuteDurationMs(); uint32_t tempMuteDurationMs = tempRecommendedMuteDuration > 0 ? tempRecommendedMuteDuration : outputDesc->latency() * 4; for (const auto &activeVs : outputDesc->getActiveVolumeSources()) { // make sure that we do not start the temporary mute period too early in case of // delayed device change setVolumeSourceMute(activeVs, true, outputDesc, delayMs); setVolumeSourceMute(activeVs, false, outputDesc, delayMs + tempMuteDurationMs, devices.types()); } } // wait for the PCM output buffers to empty before proceeding with the rest of the command if (muteWaitMs > delayMs) { muteWaitMs -= delayMs; usleep(muteWaitMs * 1000); return muteWaitMs; } return 0; } uint32_t AudioPolicyManager::setOutputDevices(const char *caller, const sp& outputDesc, const DeviceVector &devices, bool force, int delayMs, audio_patch_handle_t *patchHandle, bool requiresMuteCheck, bool requiresVolumeCheck, bool skipMuteDelay) { // TODO(b/262404095): Consider if the output need to be reopened. std::string logPrefix = std::string("caller ") + caller + outputDesc->info(); ALOGV("%s %s device %s delayMs %d", __func__, logPrefix.c_str(), devices.toString().c_str(), delayMs); uint32_t muteWaitMs; if (outputDesc->isDuplicated()) { muteWaitMs = setOutputDevices(__func__, outputDesc->subOutput1(), devices, force, delayMs, nullptr /* patchHandle */, requiresMuteCheck, skipMuteDelay); muteWaitMs += setOutputDevices(__func__, outputDesc->subOutput2(), devices, force, delayMs, nullptr /* patchHandle */, requiresMuteCheck, skipMuteDelay); return muteWaitMs; } // filter devices according to output selected DeviceVector filteredDevices = outputDesc->filterSupportedDevices(devices); DeviceVector prevDevices = outputDesc->devices(); DeviceVector availPrevDevices = mAvailableOutputDevices.filter(prevDevices); ALOGV("%s %s prevDevice %s", __func__, logPrefix.c_str(), prevDevices.toString().c_str()); if (!filteredDevices.isEmpty()) { outputDesc->setDevices(filteredDevices); } // if the outputs are not materially active, there is no need to mute. if (requiresMuteCheck) { muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevices, delayMs); } else { ALOGV("%s: %s suppressing checkDeviceMuteStrategies", __func__, logPrefix.c_str()); muteWaitMs = 0; } bool outputRouted = outputDesc->isRouted(); // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current // output profile or if new device is not supported AND previous device(s) is(are) still // available (otherwise reset device must be done on the output) if (!devices.isEmpty() && filteredDevices.isEmpty() && !availPrevDevices.empty()) { ALOGV("%s: %s unsupported device %s for output", __func__, logPrefix.c_str(), devices.toString().c_str()); // restore previous device after evaluating strategy mute state outputDesc->setDevices(prevDevices); return muteWaitMs; } // Do not change the routing if: // the requested device is AUDIO_DEVICE_NONE // OR the requested device is the same as current device // AND force is not specified // AND the output is connected by a valid audio patch. // Doing this check here allows the caller to call setOutputDevices() without conditions if ((filteredDevices.isEmpty() || filteredDevices == prevDevices) && !force && outputRouted) { ALOGV("%s %s setting same device %s or null device, force=%d, patch handle=%d", __func__, logPrefix.c_str(), filteredDevices.toString().c_str(), force, outputDesc->getPatchHandle()); if (requiresVolumeCheck && !filteredDevices.isEmpty()) { ALOGV("%s %s setting same device on routed output, force apply volumes", __func__, logPrefix.c_str()); applyStreamVolumes(outputDesc, filteredDevices.types(), delayMs, true /*force*/); } return muteWaitMs; } ALOGV("%s %s changing device to %s", __func__, logPrefix.c_str(), filteredDevices.toString().c_str()); // do the routing if (filteredDevices.isEmpty() || mAvailableOutputDevices.filter(filteredDevices).empty()) { resetOutputDevice(outputDesc, delayMs, NULL); } else { PatchBuilder patchBuilder; patchBuilder.addSource(outputDesc); ALOG_ASSERT(filteredDevices.size() <= AUDIO_PATCH_PORTS_MAX, "Too many sink ports"); for (const auto &filteredDevice : filteredDevices) { patchBuilder.addSink(filteredDevice); } // Add half reported latency to delayMs when muteWaitMs is null in order // to avoid disordered sequence of muting volume and changing devices. int actualDelayMs = !skipMuteDelay && muteWaitMs == 0 ? (delayMs + (outputDesc->latency() / 2)) : delayMs; installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), actualDelayMs); } // Since the mute is skip, also skip the apply stream volume as that will be applied externally if (!skipMuteDelay) { // update stream volumes according to new device applyStreamVolumes(outputDesc, filteredDevices.types(), delayMs); } return muteWaitMs; } status_t AudioPolicyManager::resetOutputDevice(const sp& outputDesc, int delayMs, audio_patch_handle_t *patchHandle) { ssize_t index; if (patchHandle == nullptr && !outputDesc->isRouted()) { return INVALID_OPERATION; } if (patchHandle) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle()); } if (index < 0) { return INVALID_OPERATION; } sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), delayMs); ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status); outputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE); removeAudioPatch(patchDesc->getHandle()); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); return status; } status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input, const sp &device, bool force, audio_patch_handle_t *patchHandle) { status_t status = NO_ERROR; sp inputDesc = mInputs.valueFor(input); if ((device != nullptr) && ((device != inputDesc->getDevice()) || force)) { inputDesc->setDevice(device); if (mAvailableInputDevices.contains(device)) { PatchBuilder patchBuilder; patchBuilder.addSink(inputDesc, // AUDIO_SOURCE_HOTWORD is for internal use only: // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL [inputDesc](const PatchBuilder::mix_usecase_t& usecase) { auto result = usecase; if (result.source == AUDIO_SOURCE_HOTWORD && !inputDesc->isSoundTrigger()) { result.source = AUDIO_SOURCE_VOICE_RECOGNITION; } return result; }); //only one input device for now if (audio_is_remote_submix_device(device->type())) { // remote submix HAL does not support audio conversion, need source device // audio config to match the sink input descriptor audio config, otherwise AIDL // HAL patching will fail audio_port_config srcDevicePortConfig = {}; device->toAudioPortConfig(&srcDevicePortConfig, nullptr); srcDevicePortConfig.sample_rate = inputDesc->getSamplingRate(); srcDevicePortConfig.channel_mask = inputDesc->getChannelMask(); srcDevicePortConfig.format = inputDesc->getFormat(); patchBuilder.addSource(srcDevicePortConfig); } else { patchBuilder.addSource(device); } status = installPatch(__func__, patchHandle, inputDesc.get(), patchBuilder.patch(), 0); } } return status; } status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input, audio_patch_handle_t *patchHandle) { sp inputDesc = mInputs.valueFor(input); ssize_t index; if (patchHandle) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); } if (index < 0) { return INVALID_OPERATION; } sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), 0); ALOGV("resetInputDevice() releaseAudioPatch returned %d", status); inputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE); removeAudioPatch(patchDesc->getHandle()); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); return status; } sp AudioPolicyManager::getInputProfile(const sp &device, uint32_t& samplingRate, audio_format_t& format, audio_channel_mask_t& channelMask, audio_input_flags_t flags) { // Choose an input profile based on the requested capture parameters: select the first available // profile supporting all requested parameters. // The flags can be ignored if it doesn't contain a much match flag. using underlying_input_flag_t = std::underlying_type_t; const underlying_input_flag_t mustMatchFlag = AUDIO_INPUT_FLAG_MMAP_NOIRQ | AUDIO_INPUT_FLAG_HOTWORD_TAP | AUDIO_INPUT_FLAG_HW_LOOKBACK; const underlying_input_flag_t oriFlags = flags; for (;;) { sp firstInexact = nullptr; uint32_t updatedSamplingRate = 0; audio_format_t updatedFormat = AUDIO_FORMAT_INVALID; audio_channel_mask_t updatedChannelMask = AUDIO_CHANNEL_INVALID; for (const auto& hwModule : mHwModules) { for (const auto& profile : hwModule->getInputProfiles()) { // profile->log(); //updatedFormat = format; if (profile->getCompatibilityScore( DeviceVector(device), samplingRate, &updatedSamplingRate, format, &updatedFormat, channelMask, &updatedChannelMask, // FIXME ugly cast (audio_output_flags_t) flags, true /*exactMatchRequiredForInputFlags*/) == IOProfile::EXACT_MATCH) { samplingRate = updatedSamplingRate; format = updatedFormat; channelMask = updatedChannelMask; return profile; } if (firstInexact == nullptr && profile->getCompatibilityScore( DeviceVector(device), samplingRate, &updatedSamplingRate, format, &updatedFormat, channelMask, &updatedChannelMask, // FIXME ugly cast (audio_output_flags_t) flags, false /*exactMatchRequiredForInputFlags*/) != IOProfile::NO_MATCH) { firstInexact = profile; } } } if (firstInexact != nullptr) { samplingRate = updatedSamplingRate; format = updatedFormat; channelMask = updatedChannelMask; return firstInexact; } else if (flags & AUDIO_INPUT_FLAG_RAW) { flags = (audio_input_flags_t) (flags & ~AUDIO_INPUT_FLAG_RAW); // retry } else if ((flags & mustMatchFlag) == AUDIO_INPUT_FLAG_NONE && flags != AUDIO_INPUT_FLAG_NONE && audio_is_linear_pcm(format)) { flags = AUDIO_INPUT_FLAG_NONE; } else { // fail ALOGW("%s could not find profile for device %s, sampling rate %u, format %#x, " "channel mask 0x%X, flags %#x", __func__, device->toString().c_str(), samplingRate, format, channelMask, oriFlags); break; } } return nullptr; } float AudioPolicyManager::adjustDeviceAttenuationForAbsVolume(IVolumeCurves &curves, VolumeSource volumeSource, int index, const DeviceTypeSet &deviceTypes) { audio_devices_t volumeDevice = Volume::getDeviceForVolume(deviceTypes); device_category deviceCategory = Volume::getDeviceCategory({volumeDevice}); float volumeDb = curves.volIndexToDb(deviceCategory, index); if (com_android_media_audio_abs_volume_index_fix()) { if (mAbsoluteVolumeDrivingStreams.find(volumeDevice) != mAbsoluteVolumeDrivingStreams.end()) { audio_attributes_t attributesToDriveAbs = mAbsoluteVolumeDrivingStreams[volumeDevice]; auto groupToDriveAbs = mEngine->getVolumeGroupForAttributes(attributesToDriveAbs); if (groupToDriveAbs == VOLUME_GROUP_NONE) { ALOGD("%s: no group matching with %s", __FUNCTION__, toString(attributesToDriveAbs).c_str()); return volumeDb; } float volumeDbMax = curves.volIndexToDb(deviceCategory, curves.getVolumeIndexMax()); VolumeSource vsToDriveAbs = toVolumeSource(groupToDriveAbs); if (vsToDriveAbs == volumeSource) { // attenuation is applied by the abs volume controller return volumeDbMax; } else { IVolumeCurves &curvesAbs = getVolumeCurves(vsToDriveAbs); int indexAbs = curvesAbs.getVolumeIndex({volumeDevice}); float volumeDbAbs = curvesAbs.volIndexToDb(deviceCategory, indexAbs); float volumeDbAbsMax = curvesAbs.volIndexToDb(deviceCategory, curvesAbs.getVolumeIndexMax()); float newVolumeDb = fminf(volumeDb + volumeDbAbsMax - volumeDbAbs, volumeDbMax); ALOGV("%s: abs vol stream %d with attenuation %f is adjusting stream %d from " "attenuation %f to attenuation %f %f", __func__, vsToDriveAbs, volumeDbAbs, volumeSource, volumeDb, newVolumeDb, volumeDbMax); return newVolumeDb; } } return volumeDb; } else { return volumeDb; } } float AudioPolicyManager::computeVolume(IVolumeCurves &curves, VolumeSource volumeSource, int index, const DeviceTypeSet& deviceTypes, bool computeInternalInteraction) { float volumeDb = adjustDeviceAttenuationForAbsVolume(curves, volumeSource, index, deviceTypes); ALOGV("%s volume source %d, index %d, devices %s, compute internal %b ", __func__, volumeSource, index, dumpDeviceTypes(deviceTypes).c_str(), computeInternalInteraction); if (!computeInternalInteraction) { return volumeDb; } // handle the case of accessibility active while a ringtone is playing: if the ringtone is much // louder than the accessibility prompt, the prompt cannot be heard, thus masking the touch // exploration of the dialer UI. In this situation, bring the accessibility volume closer to // the ringtone volume const auto callVolumeSrc = toVolumeSource(AUDIO_STREAM_VOICE_CALL, false); const auto ringVolumeSrc = toVolumeSource(AUDIO_STREAM_RING, false); const auto musicVolumeSrc = toVolumeSource(AUDIO_STREAM_MUSIC, false); const auto alarmVolumeSrc = toVolumeSource(AUDIO_STREAM_ALARM, false); const auto a11yVolumeSrc = toVolumeSource(AUDIO_STREAM_ACCESSIBILITY, false); if (AUDIO_MODE_RINGTONE == mEngine->getPhoneState() && mOutputs.isActive(ringVolumeSrc, 0)) { auto &ringCurves = getVolumeCurves(AUDIO_STREAM_RING); const float ringVolumeDb = computeVolume(ringCurves, ringVolumeSrc, index, deviceTypes, /* computeInternalInteraction= */ false); return ringVolumeDb - 4 > volumeDb ? ringVolumeDb - 4 : volumeDb; } // in-call: always cap volume by voice volume + some low headroom if ((volumeSource != callVolumeSrc && (isInCall() || mOutputs.isActiveLocally(callVolumeSrc))) && (volumeSource == toVolumeSource(AUDIO_STREAM_SYSTEM, false) || volumeSource == ringVolumeSrc || volumeSource == musicVolumeSrc || volumeSource == alarmVolumeSrc || volumeSource == toVolumeSource(AUDIO_STREAM_NOTIFICATION, false) || volumeSource == toVolumeSource(AUDIO_STREAM_ENFORCED_AUDIBLE, false) || volumeSource == toVolumeSource(AUDIO_STREAM_DTMF, false) || volumeSource == a11yVolumeSrc)) { auto &voiceCurves = getVolumeCurves(callVolumeSrc); int voiceVolumeIndex = voiceCurves.getVolumeIndex(deviceTypes); const float maxVoiceVolDb = computeVolume(voiceCurves, callVolumeSrc, voiceVolumeIndex, deviceTypes, /* computeInternalInteraction= */ false) + IN_CALL_EARPIECE_HEADROOM_DB; // FIXME: Workaround for call screening applications until a proper audio mode is defined // to support this scenario : Exempt the RING stream from the audio cap if the audio was // programmatically muted. // VOICE_CALL stream has minVolumeIndex > 0 : Users cannot set the volume of voice calls to // 0. We don't want to cap volume when the system has programmatically muted the voice call // stream. See setVolumeCurveIndex() for more information. bool exemptFromCapping = ((volumeSource == ringVolumeSrc) || (volumeSource == a11yVolumeSrc)) && (voiceVolumeIndex == 0); ALOGV_IF(exemptFromCapping, "%s volume source %d at vol=%f not capped", __func__, volumeSource, volumeDb); if ((volumeDb > maxVoiceVolDb) && !exemptFromCapping) { ALOGV("%s volume source %d at vol=%f overriden by volume group %d at vol=%f", __func__, volumeSource, volumeDb, callVolumeSrc, maxVoiceVolDb); volumeDb = maxVoiceVolDb; } } // if a headset is connected, apply the following rules to ring tones and notifications // to avoid sound level bursts in user's ears: // - always attenuate notifications volume by 6dB // - attenuate ring tones volume by 6dB unless music is not playing and // speaker is part of the select devices // - if music is playing, always limit the volume to current music volume, // with a minimum threshold at -36dB so that notification is always perceived. if (!Intersection(deviceTypes, {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADSET, AUDIO_DEVICE_OUT_WIRED_HEADPHONE, AUDIO_DEVICE_OUT_USB_HEADSET, AUDIO_DEVICE_OUT_HEARING_AID, AUDIO_DEVICE_OUT_BLE_HEADSET}).empty() && ((volumeSource == alarmVolumeSrc || volumeSource == ringVolumeSrc) || (volumeSource == toVolumeSource(AUDIO_STREAM_NOTIFICATION, false)) || (volumeSource == toVolumeSource(AUDIO_STREAM_SYSTEM, false)) || ((volumeSource == toVolumeSource(AUDIO_STREAM_ENFORCED_AUDIBLE, false)) && (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) && curves.canBeMuted()) { // when the phone is ringing we must consider that music could have been paused just before // by the music application and behave as if music was active if the last music track was // just stopped if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || mLimitRingtoneVolume) { volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB; DeviceTypeSet musicDevice = mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA), nullptr, true /*fromCache*/).types(); auto &musicCurves = getVolumeCurves(AUDIO_STREAM_MUSIC); float musicVolDb = computeVolume(musicCurves, musicVolumeSrc, musicCurves.getVolumeIndex(musicDevice), musicDevice, /* computeInternalInteraction= */ false); float minVolDb = (musicVolDb > SONIFICATION_HEADSET_VOLUME_MIN_DB) ? musicVolDb : SONIFICATION_HEADSET_VOLUME_MIN_DB; if (volumeDb > minVolDb) { volumeDb = minVolDb; ALOGV("computeVolume limiting volume to %f musicVol %f", minVolDb, musicVolDb); } if (Volume::getDeviceForVolume(deviceTypes) != AUDIO_DEVICE_OUT_SPEAKER && !Intersection(deviceTypes, {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES, AUDIO_DEVICE_OUT_BLE_HEADSET}).empty()) { // on A2DP/BLE, also ensure notification volume is not too low compared to media // when intended to be played. if ((volumeDb > -96.0f) && (musicVolDb - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB > volumeDb)) { ALOGV("%s increasing volume for volume source=%d device=%s from %f to %f", __func__, volumeSource, dumpDeviceTypes(deviceTypes).c_str(), volumeDb, musicVolDb - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB); volumeDb = musicVolDb - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB; } } } else if ((Volume::getDeviceForVolume(deviceTypes) != AUDIO_DEVICE_OUT_SPEAKER) || (!(volumeSource == alarmVolumeSrc || volumeSource == ringVolumeSrc))) { volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB; } } return volumeDb; } int AudioPolicyManager::rescaleVolumeIndex(int srcIndex, VolumeSource fromVolumeSource, VolumeSource toVolumeSource) { if (fromVolumeSource == toVolumeSource) { return srcIndex; } auto &srcCurves = getVolumeCurves(fromVolumeSource); auto &dstCurves = getVolumeCurves(toVolumeSource); float minSrc = (float)srcCurves.getVolumeIndexMin(); float maxSrc = (float)srcCurves.getVolumeIndexMax(); float minDst = (float)dstCurves.getVolumeIndexMin(); float maxDst = (float)dstCurves.getVolumeIndexMax(); // preserve mute request or correct range if (srcIndex < minSrc) { if (srcIndex == 0) { return 0; } srcIndex = minSrc; } else if (srcIndex > maxSrc) { srcIndex = maxSrc; } return (int)(minDst + ((srcIndex - minSrc) * (maxDst - minDst)) / (maxSrc - minSrc)); } status_t AudioPolicyManager::checkAndSetVolume(IVolumeCurves &curves, VolumeSource volumeSource, int index, const sp& outputDesc, DeviceTypeSet deviceTypes, int delayMs, bool force) { // APM is single threaded, and single instance. static std::set invalidCurvesReported; // do not change actual attributes volume if the attributes is muted if (outputDesc->isMuted(volumeSource)) { ALOGVV("%s: volume source %d muted count %d active=%d", __func__, volumeSource, outputDesc->getMuteCount(volumeSource), outputDesc->isActive(volumeSource)); return NO_ERROR; } bool isVoiceVolSrc; bool isBtScoVolSrc; if (!isVolumeConsistentForCalls( volumeSource, deviceTypes, isVoiceVolSrc, isBtScoVolSrc, __func__)) { // Do not return an error here as AudioService will always set both voice call // and Bluetooth SCO volumes due to stream aliasing. return NO_ERROR; } if (deviceTypes.empty()) { deviceTypes = outputDesc->devices().types(); index = curves.getVolumeIndex(deviceTypes); ALOGV("%s if deviceTypes is change from none to device %s, need get index %d", __func__, dumpDeviceTypes(deviceTypes).c_str(), index); } if (curves.getVolumeIndexMin() < 0 || curves.getVolumeIndexMax() < 0) { if (!invalidCurvesReported.count(&curves)) { invalidCurvesReported.insert(&curves); String8 dump; curves.dump(&dump); ALOGE("invalid volume index range in the curve:\n%s", dump.c_str()); } return BAD_VALUE; } float volumeDb = computeVolume(curves, volumeSource, index, deviceTypes); if (outputDesc->isFixedVolume(deviceTypes) || // Force VoIP volume to max for bluetooth SCO/BLE device except if muted (index != 0 && (isVoiceVolSrc || isBtScoVolSrc) && (isSingleDeviceType(deviceTypes, audio_is_bluetooth_out_sco_device) || isSingleDeviceType(deviceTypes, audio_is_ble_out_device)))) { volumeDb = 0.0f; } const bool muted = (index == 0) && (volumeDb != 0.0f); outputDesc->setVolume(volumeDb, muted, volumeSource, curves.getStreamTypes(), deviceTypes, delayMs, force, isVoiceVolSrc); if (outputDesc == mPrimaryOutput && (isVoiceVolSrc || isBtScoVolSrc)) { bool voiceVolumeManagedByHost = isVoiceVolSrc && !isSingleDeviceType(deviceTypes, audio_is_ble_out_device); setVoiceVolume(index, curves, voiceVolumeManagedByHost, delayMs); } return NO_ERROR; } void AudioPolicyManager::setVoiceVolume( int index, IVolumeCurves &curves, bool voiceVolumeManagedByHost, int delayMs) { float voiceVolume; // Force voice volume to max or mute for Bluetooth SCO/BLE as other attenuations are managed // by the headset if (voiceVolumeManagedByHost) { voiceVolume = (float)index/(float)curves.getVolumeIndexMax(); } else { voiceVolume = index == 0 ? 0.0 : 1.0; } if (voiceVolume != mLastVoiceVolume) { mpClientInterface->setVoiceVolume(voiceVolume, delayMs); mLastVoiceVolume = voiceVolume; } } bool AudioPolicyManager::isVolumeConsistentForCalls(VolumeSource volumeSource, const DeviceTypeSet& deviceTypes, bool& isVoiceVolSrc, bool& isBtScoVolSrc, const char* caller) { const VolumeSource callVolSrc = toVolumeSource(AUDIO_STREAM_VOICE_CALL, false); const VolumeSource btScoVolSrc = toVolumeSource(AUDIO_STREAM_BLUETOOTH_SCO, false); const bool isScoRequested = isScoRequestedForComm(); const bool isHAUsed = isHearingAidUsedForComm(); isVoiceVolSrc = (volumeSource != VOLUME_SOURCE_NONE) && (callVolSrc == volumeSource); isBtScoVolSrc = (volumeSource != VOLUME_SOURCE_NONE) && (btScoVolSrc == volumeSource); if ((callVolSrc != btScoVolSrc) && ((isVoiceVolSrc && isScoRequested) || (isBtScoVolSrc && !(isScoRequested || isHAUsed))) && !isSingleDeviceType(deviceTypes, AUDIO_DEVICE_OUT_TELEPHONY_TX)) { ALOGV("%s cannot set volume group %d volume when is%srequested for comm", caller, volumeSource, isScoRequested ? " " : " not "); return false; } return true; } void AudioPolicyManager::applyStreamVolumes(const sp& outputDesc, const DeviceTypeSet& deviceTypes, int delayMs, bool force) { ALOGVV("applyStreamVolumes() for device %s", dumpDeviceTypes(deviceTypes).c_str()); for (const auto &volumeGroup : mEngine->getVolumeGroups()) { auto &curves = getVolumeCurves(toVolumeSource(volumeGroup)); checkAndSetVolume(curves, toVolumeSource(volumeGroup), curves.getVolumeIndex(deviceTypes), outputDesc, deviceTypes, delayMs, force); } } void AudioPolicyManager::setStrategyMute(product_strategy_t strategy, bool on, const sp& outputDesc, int delayMs, DeviceTypeSet deviceTypes) { std::vector sourcesToMute; for (auto attributes: mEngine->getAllAttributesForProductStrategy(strategy)) { ALOGVV("%s() attributes %s, mute %d, output ID %d", __func__, toString(attributes).c_str(), on, outputDesc->getId()); VolumeSource source = toVolumeSource(attributes, false); if ((source != VOLUME_SOURCE_NONE) && (std::find(begin(sourcesToMute), end(sourcesToMute), source) == end(sourcesToMute))) { sourcesToMute.push_back(source); } } for (auto source : sourcesToMute) { setVolumeSourceMute(source, on, outputDesc, delayMs, deviceTypes); } } void AudioPolicyManager::setVolumeSourceMute(VolumeSource volumeSource, bool on, const sp& outputDesc, int delayMs, DeviceTypeSet deviceTypes) { if (deviceTypes.empty()) { deviceTypes = outputDesc->devices().types(); } auto &curves = getVolumeCurves(volumeSource); if (on) { if (!outputDesc->isMuted(volumeSource)) { if (curves.canBeMuted() && (volumeSource != toVolumeSource(AUDIO_STREAM_ENFORCED_AUDIBLE, false) || (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) { checkAndSetVolume(curves, volumeSource, 0, outputDesc, deviceTypes, delayMs); } } // increment mMuteCount after calling checkAndSetVolume() so that volume change is not // ignored outputDesc->incMuteCount(volumeSource); } else { if (!outputDesc->isMuted(volumeSource)) { ALOGV("%s unmuting non muted attributes!", __func__); return; } if (outputDesc->decMuteCount(volumeSource) == 0) { checkAndSetVolume(curves, volumeSource, curves.getVolumeIndex(deviceTypes), outputDesc, deviceTypes, delayMs); } } } bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) { // has flags that map to a stream type? if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) { return true; } // has known usage? switch (paa->usage) { case AUDIO_USAGE_UNKNOWN: case AUDIO_USAGE_MEDIA: case AUDIO_USAGE_VOICE_COMMUNICATION: case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: case AUDIO_USAGE_ALARM: case AUDIO_USAGE_NOTIFICATION: case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: case AUDIO_USAGE_NOTIFICATION_EVENT: case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: case AUDIO_USAGE_ASSISTANCE_SONIFICATION: case AUDIO_USAGE_GAME: case AUDIO_USAGE_VIRTUAL_SOURCE: case AUDIO_USAGE_ASSISTANT: case AUDIO_USAGE_CALL_ASSISTANT: case AUDIO_USAGE_EMERGENCY: case AUDIO_USAGE_SAFETY: case AUDIO_USAGE_VEHICLE_STATUS: case AUDIO_USAGE_ANNOUNCEMENT: break; default: return false; } return true; } audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) { return mEngine->getForceUse(usage); } bool AudioPolicyManager::isInCall() const { return isStateInCall(mEngine->getPhoneState()); } bool AudioPolicyManager::isStateInCall(int state) const { return is_state_in_call(state); } bool AudioPolicyManager::isCallAudioAccessible() const { audio_mode_t mode = mEngine->getPhoneState(); return (mode == AUDIO_MODE_IN_CALL) || (mode == AUDIO_MODE_CALL_SCREEN) || (mode == AUDIO_MODE_CALL_REDIRECT); } bool AudioPolicyManager::isInCallOrScreening() const { audio_mode_t mode = mEngine->getPhoneState(); return isStateInCall(mode) || mode == AUDIO_MODE_CALL_SCREEN; } void AudioPolicyManager::cleanUpForDevice(const sp& deviceDesc) { for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--) { sp sourceDesc = mAudioSources.valueAt(i); if (sourceDesc->isConnected() && (sourceDesc->srcDevice()->equals(deviceDesc) || sourceDesc->sinkDevice()->equals(deviceDesc)) && !sourceDesc->isCallRx()) { disconnectAudioSource(sourceDesc); } } for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) { sp patchDesc = mAudioPatches.valueAt(i); bool release = false; for (size_t j = 0; j < patchDesc->mPatch.num_sources && !release; j++) { const struct audio_port_config *source = &patchDesc->mPatch.sources[j]; if (source->type == AUDIO_PORT_TYPE_DEVICE && source->ext.device.type == deviceDesc->type()) { release = true; } } const char *address = deviceDesc->address().c_str(); for (size_t j = 0; j < patchDesc->mPatch.num_sinks && !release; j++) { const struct audio_port_config *sink = &patchDesc->mPatch.sinks[j]; if (sink->type == AUDIO_PORT_TYPE_DEVICE && sink->ext.device.type == deviceDesc->type() && (strnlen(address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0 || strncmp(sink->ext.device.address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) { release = true; } } if (release) { ALOGV("%s releasing patch %u", __FUNCTION__, patchDesc->getHandle()); releaseAudioPatch(patchDesc->getHandle(), patchDesc->getUid()); } } mInputs.clearSessionRoutesForDevice(deviceDesc); mHwModules.cleanUpForDevice(deviceDesc); } void AudioPolicyManager::modifySurroundFormats( const sp& devDesc, FormatVector *formatsPtr) { std::unordered_set enforcedSurround( devDesc->encodedFormats().begin(), devDesc->encodedFormats().end()); std::unordered_set allSurround; // A flat set of all known surround formats for (const auto& pair : mConfig->getSurroundFormats()) { allSurround.insert(pair.first); for (const auto& subformat : pair.second) allSurround.insert(subformat); } audio_policy_forced_cfg_t forceUse = mEngine->getForceUse( AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND); ALOGD("%s: forced use = %d", __FUNCTION__, forceUse); // This is the resulting set of formats depending on the surround mode: // 'all surround' = allSurround // 'enforced surround' = enforcedSurround [may include IEC69137 which isn't raw surround fmt] // 'non-surround' = not in 'all surround' and not in 'enforced surround' // 'manual surround' = mManualSurroundFormats // AUTO: formats v 'enforced surround' // ALWAYS: formats v 'all surround' v 'enforced surround' // NEVER: formats ^ 'non-surround' // MANUAL: formats ^ ('non-surround' v 'manual surround' v (IEC69137 ^ 'enforced surround')) std::unordered_set formatSet; if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL || forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) { // formatSet is (formats ^ 'non-surround') for (auto formatIter = formatsPtr->begin(); formatIter != formatsPtr->end(); ++formatIter) { if (allSurround.count(*formatIter) == 0 && enforcedSurround.count(*formatIter) == 0) { formatSet.insert(*formatIter); } } } else { formatSet.insert(formatsPtr->begin(), formatsPtr->end()); } formatsPtr->clear(); // Re-filled from the formatSet at the end. if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL) { formatSet.insert(mManualSurroundFormats.begin(), mManualSurroundFormats.end()); // Enable IEC61937 when in MANUAL mode if it's enforced for this device. if (enforcedSurround.count(AUDIO_FORMAT_IEC61937) != 0) { formatSet.insert(AUDIO_FORMAT_IEC61937); } } else if (forceUse != AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) { // AUTO or ALWAYS if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS) { formatSet.insert(allSurround.begin(), allSurround.end()); } formatSet.insert(enforcedSurround.begin(), enforcedSurround.end()); } for (const auto& format : formatSet) { formatsPtr->push_back(format); } } void AudioPolicyManager::modifySurroundChannelMasks(ChannelMaskSet *channelMasksPtr) { ChannelMaskSet &channelMasks = *channelMasksPtr; audio_policy_forced_cfg_t forceUse = mEngine->getForceUse( AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND); // If NEVER, then remove support for channelMasks > stereo. if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) { for (auto it = channelMasks.begin(); it != channelMasks.end();) { audio_channel_mask_t channelMask = *it; if (channelMask & ~AUDIO_CHANNEL_OUT_STEREO) { ALOGV("%s: force NEVER, so remove channelMask 0x%08x", __FUNCTION__, channelMask); it = channelMasks.erase(it); } else { ++it; } } // If ALWAYS or MANUAL, then make sure we at least support 5.1 } else if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS || forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL) { bool supports5dot1 = false; // Are there any channel masks that can be considered "surround"? for (audio_channel_mask_t channelMask : channelMasks) { if ((channelMask & AUDIO_CHANNEL_OUT_5POINT1) == AUDIO_CHANNEL_OUT_5POINT1) { supports5dot1 = true; break; } } // If not then add 5.1 support. if (!supports5dot1) { channelMasks.insert(AUDIO_CHANNEL_OUT_5POINT1); ALOGV("%s: force MANUAL or ALWAYS, so adding channelMask for 5.1 surround", __func__); } } } void AudioPolicyManager::updateAudioProfiles(const sp& devDesc, audio_io_handle_t ioHandle, const sp& profile) { if (!profile->hasDynamicAudioProfile()) { return; } audio_port_v7 devicePort; devDesc->toAudioPort(&devicePort); audio_port_v7 mixPort; profile->toAudioPort(&mixPort); mixPort.ext.mix.handle = ioHandle; status_t status = mpClientInterface->getAudioMixPort(&devicePort, &mixPort); if (status != NO_ERROR) { ALOGE("%s failed to query the attributes of the mix port", __func__); return; } std::set supportedFormats; for (size_t i = 0; i < mixPort.num_audio_profiles; ++i) { supportedFormats.insert(mixPort.audio_profiles[i].format); } FormatVector formats(supportedFormats.begin(), supportedFormats.end()); mReportedFormatsMap[devDesc] = formats; if (devDesc->type() == AUDIO_DEVICE_OUT_HDMI || devDesc->type() == AUDIO_DEVICE_OUT_HDMI_ARC || devDesc->type() == AUDIO_DEVICE_OUT_HDMI_EARC || isDeviceOfModule(devDesc,AUDIO_HARDWARE_MODULE_ID_MSD)) { modifySurroundFormats(devDesc, &formats); size_t modifiedNumProfiles = 0; for (size_t i = 0; i < mixPort.num_audio_profiles; ++i) { if (std::find(formats.begin(), formats.end(), mixPort.audio_profiles[i].format) == formats.end()) { // Skip the format that is not present after modifying surround formats. continue; } memcpy(&mixPort.audio_profiles[modifiedNumProfiles], &mixPort.audio_profiles[i], sizeof(struct audio_profile)); ChannelMaskSet channels(mixPort.audio_profiles[modifiedNumProfiles].channel_masks, mixPort.audio_profiles[modifiedNumProfiles].channel_masks + mixPort.audio_profiles[modifiedNumProfiles].num_channel_masks); modifySurroundChannelMasks(&channels); std::copy(channels.begin(), channels.end(), std::begin(mixPort.audio_profiles[modifiedNumProfiles].channel_masks)); mixPort.audio_profiles[modifiedNumProfiles++].num_channel_masks = channels.size(); } mixPort.num_audio_profiles = modifiedNumProfiles; } profile->importAudioPort(mixPort); } status_t AudioPolicyManager::installPatch(const char *caller, audio_patch_handle_t *patchHandle, AudioIODescriptorInterface *ioDescriptor, const struct audio_patch *patch, int delayMs) { ssize_t index = mAudioPatches.indexOfKey( patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE ? *patchHandle : ioDescriptor->getPatchHandle()); sp patchDesc; status_t status = installPatch( caller, index, patchHandle, patch, delayMs, mUidCached, &patchDesc); if (status == NO_ERROR) { ioDescriptor->setPatchHandle(patchDesc->getHandle()); } return status; } status_t AudioPolicyManager::installPatch(const char *caller, ssize_t index, audio_patch_handle_t *patchHandle, const struct audio_patch *patch, int delayMs, uid_t uid, sp *patchDescPtr) { sp patchDesc; audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; if (index >= 0) { patchDesc = mAudioPatches.valueAt(index); afPatchHandle = patchDesc->getAfHandle(); } status_t status = mpClientInterface->createAudioPatch(patch, &afPatchHandle, delayMs); ALOGV("%s() AF::createAudioPatch returned %d patchHandle %d num_sources %d num_sinks %d", caller, status, afPatchHandle, patch->num_sources, patch->num_sinks); if (status == NO_ERROR) { if (index < 0) { patchDesc = new AudioPatch(patch, uid); addAudioPatch(patchDesc->getHandle(), patchDesc); } else { patchDesc->mPatch = *patch; } patchDesc->setAfHandle(afPatchHandle); if (patchHandle) { *patchHandle = patchDesc->getHandle(); } nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } if (patchDescPtr) *patchDescPtr = patchDesc; return status; } bool AudioPolicyManager::areAllActiveTracksRerouted(const sp& output) { const TrackClientVector activeClients = output->getActiveClients(); if (activeClients.empty()) { return true; } ssize_t index = mAudioPatches.indexOfKey(output->getPatchHandle()); if (index < 0) { ALOGE("%s, no audio patch found while there are active clients on output %d", __func__, output->getId()); return false; } sp patchDesc = mAudioPatches.valueAt(index); DeviceVector routedDevices; for (int i = 0; i < patchDesc->mPatch.num_sinks; ++i) { sp device = mAvailableOutputDevices.getDeviceFromId( patchDesc->mPatch.sinks[i].id); if (device == nullptr) { ALOGE("%s, no audio device found with id(%d)", __func__, patchDesc->mPatch.sinks[i].id); return false; } routedDevices.add(device); } for (const auto& client : activeClients) { if (client->isInvalid()) { // No need to take care about invalidated clients. continue; } sp preferredDevice = mAvailableOutputDevices.getDeviceFromId(client->preferredDeviceId()); if (mEngine->getOutputDevicesForAttributes( client->attributes(), preferredDevice, false) == routedDevices) { return false; } } return true; } sp AudioPolicyManager::openOutputWithProfileAndDevice( const sp& profile, const DeviceVector& devices, const audio_config_base_t *mixerConfig, const audio_config_t *halConfig, audio_output_flags_t flags) { for (const auto& device : devices) { // TODO: This should be checking if the profile supports the device combo. if (!profile->supportsDevice(device)) { ALOGE("%s profile(%s) doesn't support device %#x", __func__, profile->getName().c_str(), device->type()); return nullptr; } } sp desc = new SwAudioOutputDescriptor(profile, mpClientInterface); audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER; status_t status = desc->open(halConfig, mixerConfig, devices, AUDIO_STREAM_DEFAULT, &flags, &output, attributes); if (status != NO_ERROR) { ALOGE("%s failed to open output %d", __func__, status); return nullptr; } if ((flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) == AUDIO_OUTPUT_FLAG_BIT_PERFECT) { auto portConfig = desc->getConfig(); for (const auto& device : devices) { device->setPreferredConfig(&portConfig); } } // Here is where the out_set_parameters() for card & device gets called sp device = devices.getDeviceForOpening(); const audio_devices_t deviceType = device->type(); const String8 &address = String8(device->address().c_str()); if (!address.empty()) { char *param = audio_device_address_to_parameter(deviceType, address.c_str()); mpClientInterface->setParameters(output, String8(param)); free(param); } updateAudioProfiles(device, output, profile); if (!profile->hasValidAudioProfile()) { ALOGW("%s() missing param", __func__); desc->close(); return nullptr; } else if (profile->hasDynamicAudioProfile() && halConfig == nullptr) { // Reopen the output with the best audio profile picked by APM when the profile supports // dynamic audio profile and the hal config is not specified. desc->close(); output = AUDIO_IO_HANDLE_NONE; audio_config_t config = AUDIO_CONFIG_INITIALIZER; profile->pickAudioProfile( config.sample_rate, config.channel_mask, config.format); config.offload_info.sample_rate = config.sample_rate; config.offload_info.channel_mask = config.channel_mask; config.offload_info.format = config.format; status = desc->open(&config, mixerConfig, devices, AUDIO_STREAM_DEFAULT, &flags, &output, attributes); if (status != NO_ERROR) { return nullptr; } } addOutput(output, desc); // The version check is essentially to avoid making this call in the case of the HIDL HAL. if (auto hwModule = mHwModules.getModuleFromHandle(mPrimaryModuleHandle); hwModule && hwModule->getHalVersionMajor() >= 3) { setOutputDevices(__func__, desc, devices, true, 0, NULL); } sp speaker = mAvailableOutputDevices.getDevice( AUDIO_DEVICE_OUT_SPEAKER, String8(""), AUDIO_FORMAT_DEFAULT); if (audio_is_remote_submix_device(deviceType) && address != "0") { sp policyMix; if (mPolicyMixes.getAudioPolicyMix(deviceType, address, policyMix) == NO_ERROR) { policyMix->setOutput(desc); desc->mPolicyMix = policyMix; } else { ALOGW("checkOutputsForDevice() cannot find policy for address %s", address.c_str()); } } else if (hasPrimaryOutput() && speaker != nullptr && mPrimaryOutput->supportsDevice(speaker) && !desc->supportsDevice(speaker) && ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) { // no duplicated output for: // - direct outputs // - outputs used by dynamic policy mixes // - outputs that supports SPEAKER while the primary output does not. audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE; //TODO: configure audio effect output stage here // open a duplicating output thread for the new output and the primary output sp dupOutputDesc = new SwAudioOutputDescriptor(nullptr, mpClientInterface); status = dupOutputDesc->openDuplicating(mPrimaryOutput, desc, &duplicatedOutput); if (status == NO_ERROR) { // add duplicated output descriptor addOutput(duplicatedOutput, dupOutputDesc); } else { ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", mPrimaryOutput->mIoHandle, output); desc->close(); removeOutput(output); nextAudioPortGeneration(); return nullptr; } } if (mPrimaryOutput == nullptr && profile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) { ALOGV("%s(): re-assigning mPrimaryOutput", __func__); mPrimaryOutput = desc; mPrimaryModuleHandle = mPrimaryOutput->getModuleHandle(); } return desc; } status_t AudioPolicyManager::getDevicesForAttributes( const audio_attributes_t &attr, DeviceVector &devices, bool forVolume) { // Devices are determined in the following precedence: // // 1) Devices associated with a dynamic policy matching the attributes. This is often // a remote submix from MIX_ROUTE_FLAG_LOOP_BACK. // // If no such dynamic policy then // 2) Devices containing an active client using setPreferredDevice // with same strategy as the attributes. // (from the default Engine::getOutputDevicesForAttributes() implementation). // // If no corresponding active client with setPreferredDevice then // 3) Devices associated with the strategy determined by the attributes // (from the default Engine::getOutputDevicesForAttributes() implementation). // // See related getOutputForAttrInt(). // check dynamic policies but only for primary descriptors (secondary not used for audible // audio routing, only used for duplication for playback capture) sp policyMix; bool unneededUsePrimaryOutputFromPolicyMixes = false; status_t status = mPolicyMixes.getOutputForAttr(attr, AUDIO_CONFIG_BASE_INITIALIZER, 0 /*uid unknown here*/, AUDIO_SESSION_NONE, AUDIO_OUTPUT_FLAG_NONE, mAvailableOutputDevices, nullptr /* requestedDevice */, policyMix, nullptr /* secondaryMixes */, unneededUsePrimaryOutputFromPolicyMixes); if (status != OK) { return status; } if (policyMix != nullptr && policyMix->getOutput() != nullptr && // For volume control, skip LOOPBACK mixes which use AUDIO_DEVICE_OUT_REMOTE_SUBMIX // as they are unaffected by device/stream volume // (per SwAudioOutputDescriptor::isFixedVolume()). (!forVolume || policyMix->mDeviceType != AUDIO_DEVICE_OUT_REMOTE_SUBMIX) ) { sp deviceDesc = mAvailableOutputDevices.getDevice( policyMix->mDeviceType, policyMix->mDeviceAddress, AUDIO_FORMAT_DEFAULT); devices.add(deviceDesc); } else { // The default Engine::getOutputDevicesForAttributes() uses findPreferredDevice() // which selects setPreferredDevice if active. This means forVolume call // will take an active setPreferredDevice, if such exists. devices = mEngine->getOutputDevicesForAttributes( attr, nullptr /* preferredDevice */, false /* fromCache */); } if (forVolume) { // We alias the device AUDIO_DEVICE_OUT_SPEAKER_SAFE to AUDIO_DEVICE_OUT_SPEAKER // for single volume control in AudioService (such relationship should exist if // SPEAKER_SAFE is present). // // (This is unrelated to a different device grouping as Volume::getDeviceCategory) DeviceVector speakerSafeDevices = devices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER_SAFE); if (!speakerSafeDevices.isEmpty()) { devices.merge(mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER)); devices.remove(speakerSafeDevices); } } return NO_ERROR; } status_t AudioPolicyManager::getProfilesForDevices(const DeviceVector& devices, AudioProfileVector& audioProfiles, uint32_t flags, bool isInput) { for (const auto& hwModule : mHwModules) { // the MSD module checks for different conditions if (strcmp(hwModule->getName(), AUDIO_HARDWARE_MODULE_ID_MSD) == 0) { continue; } IOProfileCollection ioProfiles = isInput ? hwModule->getInputProfiles() : hwModule->getOutputProfiles(); for (const auto& profile : ioProfiles) { if (!profile->areAllDevicesSupported(devices) || !profile->isCompatibleProfileForFlags( flags, false /*exactMatchRequiredForInputFlags*/)) { continue; } audioProfiles.addAllValidProfiles(profile->asAudioPort()->getAudioProfiles()); } } if (!isInput) { // add the direct profiles from MSD if present and has audio patches to all the output(s) const auto &msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD); if (msdModule != nullptr) { if (msdHasPatchesToAllDevices(devices.toTypeAddrVector())) { ALOGV("%s: MSD audio patches set to all output devices.", __func__); for (const auto &profile: msdModule->getOutputProfiles()) { if (!profile->asAudioPort()->isDirectOutput()) { continue; } audioProfiles.addAllValidProfiles(profile->asAudioPort()->getAudioProfiles()); } } else { ALOGV("%s: MSD audio patches NOT set to all output devices.", __func__); } } } return NO_ERROR; } sp AudioPolicyManager::reopenOutput(sp outputDesc, const audio_config_t *config, audio_output_flags_t flags, const char* caller) { closeOutput(outputDesc->mIoHandle); sp preferredOutput = openOutputWithProfileAndDevice( outputDesc->mProfile, outputDesc->devices(), nullptr /*mixerConfig*/, config, flags); if (preferredOutput == nullptr) { ALOGE("%s failed to reopen output device=%d, caller=%s", __func__, outputDesc->devices()[0]->getId(), caller); } return preferredOutput; } void AudioPolicyManager::reopenOutputsWithDevices( const std::map &outputsToReopen) { for (const auto& [output, devices] : outputsToReopen) { sp desc = mOutputs.valueFor(output); closeOutput(output); openOutputWithProfileAndDevice(desc->mProfile, devices); } } PortHandleVector AudioPolicyManager::getClientsForStream( audio_stream_type_t streamType) const { PortHandleVector clients; for (size_t i = 0; i < mOutputs.size(); ++i) { PortHandleVector clientsForStream = mOutputs.valueAt(i)->getClientsForStream(streamType); clients.insert(clients.end(), clientsForStream.begin(), clientsForStream.end()); } return clients; } void AudioPolicyManager::invalidateStreams(StreamTypeVector streams) const { PortHandleVector clients; for (auto stream : streams) { PortHandleVector clientsForStream = getClientsForStream(stream); clients.insert(clients.end(), clientsForStream.begin(), clientsForStream.end()); } mpClientInterface->invalidateTracks(clients); } void AudioPolicyManager::updateClientsInternalMute( const sp &desc) { if (!desc->isBitPerfect() || !com::android::media::audioserver:: fix_concurrent_playback_behavior_with_bit_perfect_client()) { // This is only used for bit perfect output now. return; } sp bitPerfectClient = nullptr; bool bitPerfectClientInternalMute = false; std::vector clientsInternalMute; for (const sp& client : desc->getActiveClients()) { if ((client->flags() & AUDIO_OUTPUT_FLAG_BIT_PERFECT) != AUDIO_OUTPUT_FLAG_NONE) { bitPerfectClient = client; continue; } bool muted = false; if (client->stream() == AUDIO_STREAM_SYSTEM) { // System sound is muted. muted = true; } else { bitPerfectClientInternalMute = true; } if (client->setInternalMute(muted)) { auto result = legacy2aidl_audio_port_handle_t_int32_t(client->portId()); if (!result.ok()) { ALOGE("%s, failed to convert port id(%d) to aidl", __func__, client->portId()); continue; } media::TrackInternalMuteInfo info; info.portId = result.value(); info.muted = client->getInternalMute(); clientsInternalMute.push_back(std::move(info)); } } if (bitPerfectClient != nullptr && bitPerfectClient->setInternalMute(bitPerfectClientInternalMute)) { auto result = legacy2aidl_audio_port_handle_t_int32_t(bitPerfectClient->portId()); if (result.ok()) { media::TrackInternalMuteInfo info; info.portId = result.value(); info.muted = bitPerfectClient->getInternalMute(); clientsInternalMute.push_back(std::move(info)); } else { ALOGE("%s, failed to convert port id(%d) of bit perfect client to aidl", __func__, bitPerfectClient->portId()); } } if (!clientsInternalMute.empty()) { if (status_t status = mpClientInterface->setTracksInternalMute(clientsInternalMute); status != NO_ERROR) { ALOGE("%s, failed to update tracks internal mute, err=%d", __func__, status); } } } } // namespace android