/* * 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. */ #define LOG_TAG "AudioPolicyManager" //#define LOG_NDEBUG 0 //#define VERY_VERBOSE_LOGGING #ifdef VERY_VERBOSE_LOGGING #define ALOGVV ALOGV #else #define ALOGVV(a...) do { } while(0) #endif // A device mask for all audio input devices that are considered "virtual" when evaluating // active inputs in getActiveInput() #define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER) // A device mask for all audio output devices that are considered "remote" when evaluating // active output devices in isStreamActiveRemotely() #define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX // A device mask for all audio input and output devices where matching inputs/outputs on device // type alone is not enough: the address must match too #define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \ AUDIO_DEVICE_OUT_REMOTE_SUBMIX) #include #include #include #include #include #include #include #include #include #include "AudioPolicyManager.h" #include "audio_policy_conf.h" namespace android { // ---------------------------------------------------------------------------- // Definitions for audio_policy.conf file parsing // ---------------------------------------------------------------------------- struct StringToEnum { const char *name; uint32_t value; }; #define STRING_TO_ENUM(string) { #string, string } #define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) const StringToEnum sDeviceNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE), STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER), STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE), STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET), STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE), STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO), STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET), STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT), STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO), STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP), STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES), STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP), STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL), STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI), STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET), STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET), STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY), STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE), STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB), STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX), STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX), STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE), STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC), STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF), STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM), STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE), STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT), STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC), STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET), STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO), STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET), STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL), STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI), STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX), STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL), STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC), STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX), STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET), STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET), STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY), STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE), STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER), STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER), STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE), STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF), STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP), STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK), }; const StringToEnum sOutputFlagNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT), STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY), STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST), STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER), STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD), STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING), STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC), }; const StringToEnum sInputFlagNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST), STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD), }; const StringToEnum sFormatNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT), STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT), STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT), STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT), STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT), STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED), STRING_TO_ENUM(AUDIO_FORMAT_MP3), STRING_TO_ENUM(AUDIO_FORMAT_AAC), STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN), STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC), STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR), STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP), STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1), STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE), STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC), STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD), STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2), STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD), STRING_TO_ENUM(AUDIO_FORMAT_VORBIS), STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1), STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2), STRING_TO_ENUM(AUDIO_FORMAT_OPUS), STRING_TO_ENUM(AUDIO_FORMAT_AC3), STRING_TO_ENUM(AUDIO_FORMAT_E_AC3), }; const StringToEnum sOutChannelsNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), }; const StringToEnum sInChannelsNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO), STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO), STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK), }; const StringToEnum sGainModeNameToEnumTable[] = { STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT), STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS), STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP), }; uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table, size_t size, const char *name) { for (size_t i = 0; i < size; i++) { if (strcmp(table[i].name, name) == 0) { ALOGV("stringToEnum() found %s", table[i].name); return table[i].value; } } return 0; } const char *AudioPolicyManager::enumToString(const struct StringToEnum *table, size_t size, uint32_t value) { for (size_t i = 0; i < size; i++) { if (table[i].value == value) { return table[i].name; } } return ""; } bool AudioPolicyManager::stringToBool(const char *value) { return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0)); } // ---------------------------------------------------------------------------- // AudioPolicyInterface implementation // ---------------------------------------------------------------------------- status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address) { String8 address = (device_address == NULL) ? String8("") : String8(device_address); // handle legacy remote submix case where the address was not always specified if (deviceDistinguishesOnAddress(device) && (address.length() == 0)) { address = String8("0"); } ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, address.string()); // connect/disconnect only 1 device at a time if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; // handle output devices if (audio_is_output_device(device)) { SortedVector outputs; sp devDesc = new DeviceDescriptor(String8(""), device); devDesc->mAddress = address; ssize_t index = mAvailableOutputDevices.indexOf(devDesc); // 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; switch (state) { // handle output device connection case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: if (index >= 0) { ALOGW("setDeviceConnectionState() device already connected: %x", device); return INVALID_OPERATION; } ALOGV("setDeviceConnectionState() connecting device %x", device); // register new device as available index = mAvailableOutputDevices.add(devDesc); if (index >= 0) { sp module = getModuleForDevice(device); if (module == 0) { ALOGD("setDeviceConnectionState() could not find HW module for device %08x", device); mAvailableOutputDevices.remove(devDesc); return INVALID_OPERATION; } mAvailableOutputDevices[index]->mId = nextUniqueId(); mAvailableOutputDevices[index]->mModule = module; } else { return NO_MEMORY; } if (checkOutputsForDevice(devDesc, state, outputs, address) != NO_ERROR) { mAvailableOutputDevices.remove(devDesc); return INVALID_OPERATION; } // outputs should never be empty here ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():" "checkOutputsForDevice() returned no outputs but status OK"); ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs", outputs.size()); break; // handle output device disconnection case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { if (index < 0) { ALOGW("setDeviceConnectionState() device not connected: %x", device); return INVALID_OPERATION; } ALOGV("setDeviceConnectionState() disconnecting output device %x", device); // Set Disconnect to HALs AudioParameter param = AudioParameter(address); param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); // remove device from available output devices mAvailableOutputDevices.remove(devDesc); checkOutputsForDevice(devDesc, state, outputs, address); } break; default: ALOGE("setDeviceConnectionState() invalid state: %x", state); return BAD_VALUE; } // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP // output is suspended before any tracks are moved to it checkA2dpSuspend(); checkOutputForAllStrategies(); // outputs must be closed after checkOutputForAllStrategies() is executed if (!outputs.isEmpty()) { for (size_t i = 0; i < outputs.size(); i++) { sp desc = mOutputs.valueFor(outputs[i]); // close unused outputs after device disconnection or direct outputs that have been // opened by checkOutputsForDevice() to query dynamic parameters if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) || (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && (desc->mDirectOpenCount == 0))) { closeOutput(outputs[i]); } } // check again after closing A2DP output to reset mA2dpSuspended if needed checkA2dpSuspend(); } updateDevicesAndOutputs(); if (mPhoneState == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); updateCallRouting(newDevice); } for (size_t i = 0; i < mOutputs.size(); i++) { audio_io_handle_t output = mOutputs.keyAt(i); if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i), 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 = !mOutputs.valueAt(i)->isDuplicated() && (!deviceDistinguishesOnAddress(device) // always force when disconnecting (a non-duplicated device) || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); setOutputDevice(output, newDevice, force, 0); } } mpClientInterface->onAudioPortListUpdate(); return NO_ERROR; } // end if is output device // handle input devices if (audio_is_input_device(device)) { SortedVector inputs; sp devDesc = new DeviceDescriptor(String8(""), device); devDesc->mAddress = address; ssize_t index = mAvailableInputDevices.indexOf(devDesc); switch (state) { // handle input device connection case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: { if (index >= 0) { ALOGW("setDeviceConnectionState() device already connected: %d", device); return INVALID_OPERATION; } sp module = getModuleForDevice(device); if (module == NULL) { ALOGW("setDeviceConnectionState(): could not find HW module for device %08x", device); return INVALID_OPERATION; } if (checkInputsForDevice(device, state, inputs, address) != NO_ERROR) { return INVALID_OPERATION; } index = mAvailableInputDevices.add(devDesc); if (index >= 0) { mAvailableInputDevices[index]->mId = nextUniqueId(); mAvailableInputDevices[index]->mModule = module; } else { return NO_MEMORY; } } break; // handle input device disconnection case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: { if (index < 0) { ALOGW("setDeviceConnectionState() device not connected: %d", device); return INVALID_OPERATION; } ALOGV("setDeviceConnectionState() disconnecting input device %x", device); // Set Disconnect to HALs AudioParameter param = AudioParameter(address); param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); checkInputsForDevice(device, state, inputs, address); mAvailableInputDevices.remove(devDesc); } break; default: ALOGE("setDeviceConnectionState() invalid state: %x", state); return BAD_VALUE; } closeAllInputs(); if (mPhoneState == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); updateCallRouting(newDevice); } mpClientInterface->onAudioPortListUpdate(); return NO_ERROR; } // end if is input device ALOGW("setDeviceConnectionState() invalid device: %x", device); return BAD_VALUE; } audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device, const char *device_address) { audio_policy_dev_state_t state = AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; sp devDesc = new DeviceDescriptor(String8(""), device); devDesc->mAddress = (device_address == NULL) ? String8("") : String8(device_address); // handle legacy remote submix case where the address was not always specified if (deviceDistinguishesOnAddress(device) && (devDesc->mAddress.length() == 0)) { devDesc->mAddress = String8("0"); } ssize_t index; DeviceVector *deviceVector; if (audio_is_output_device(device)) { deviceVector = &mAvailableOutputDevices; } else if (audio_is_input_device(device)) { deviceVector = &mAvailableInputDevices; } else { ALOGW("getDeviceConnectionState() invalid device type %08x", device); return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; } index = deviceVector->indexOf(devDesc); if (index >= 0) { return AUDIO_POLICY_DEVICE_STATE_AVAILABLE; } else { return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; } } void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs) { bool createTxPatch = false; struct audio_patch patch; patch.num_sources = 1; patch.num_sinks = 1; status_t status; audio_patch_handle_t afPatchHandle; DeviceVector deviceList; audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice); // release existing RX patch if any if (mCallRxPatch != 0) { mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); mCallRxPatch.clear(); } // release TX patch if any if (mCallTxPatch != 0) { mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); mCallTxPatch.clear(); } // 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. if (availablePrimaryOutputDevices() & rxDevice) { setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs); // 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. if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN) == AUDIO_DEVICE_NONE) { createTxPatch = true; } } else { // create RX path audio patch deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice); ALOG_ASSERT(!deviceList.isEmpty(), "updateCallRouting() selected device not in output device list"); sp rxSinkDeviceDesc = deviceList.itemAt(0); deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX); ALOG_ASSERT(!deviceList.isEmpty(), "updateCallRouting() no telephony RX device"); sp rxSourceDeviceDesc = deviceList.itemAt(0); rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); // request to reuse existing output stream if one is already opened to reach the RX device SortedVector outputs = getOutputsForDevice(rxDevice, mOutputs); audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); if (output != AUDIO_IO_HANDLE_NONE) { sp outputDesc = mOutputs.valueFor(output); ALOG_ASSERT(!outputDesc->isDuplicated(), "updateCallRouting() RX device output is duplicated"); outputDesc->toAudioPortConfig(&patch.sources[1]); patch.num_sources = 2; } afPatchHandle = AUDIO_PATCH_HANDLE_NONE; status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch", status); if (status == NO_ERROR) { mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(), &patch, mUidCached); mCallRxPatch->mAfPatchHandle = afPatchHandle; mCallRxPatch->mUid = mUidCached; } createTxPatch = true; } if (createTxPatch) { struct audio_patch patch; patch.num_sources = 1; patch.num_sinks = 1; deviceList = mAvailableInputDevices.getDevicesFromType(txDevice); ALOG_ASSERT(!deviceList.isEmpty(), "updateCallRouting() selected device not in input device list"); sp txSourceDeviceDesc = deviceList.itemAt(0); txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]); deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX); ALOG_ASSERT(!deviceList.isEmpty(), "updateCallRouting() no telephony TX device"); sp txSinkDeviceDesc = deviceList.itemAt(0); txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); SortedVector outputs = getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs); audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); // request to reuse existing output stream if one is already opened to reach the TX // path output device if (output != AUDIO_IO_HANDLE_NONE) { sp outputDesc = mOutputs.valueFor(output); ALOG_ASSERT(!outputDesc->isDuplicated(), "updateCallRouting() RX device output is duplicated"); outputDesc->toAudioPortConfig(&patch.sources[1]); patch.num_sources = 2; } afPatchHandle = AUDIO_PATCH_HANDLE_NONE; status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch", status); if (status == NO_ERROR) { mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(), &patch, mUidCached); mCallTxPatch->mAfPatchHandle = afPatchHandle; mCallTxPatch->mUid = mUidCached; } } } void AudioPolicyManager::setPhoneState(audio_mode_t state) { ALOGV("setPhoneState() state %d", state); if (state < 0 || state >= AUDIO_MODE_CNT) { ALOGW("setPhoneState() invalid state %d", state); return; } if (state == mPhoneState ) { ALOGW("setPhoneState() setting same state %d", state); return; } // if leaving call state, handle special case of active streams // pertaining to sonification strategy see handleIncallSonification() if (isInCall()) { ALOGV("setPhoneState() in call state management: new state is %d", state); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } handleIncallSonification((audio_stream_type_t)stream, false, true); } } // store previous phone state for management of sonification strategy below int oldState = mPhoneState; mPhoneState = state; bool force = false; // are we entering or starting a call if (!isStateInCall(oldState) && isStateInCall(state)) { ALOGV(" Entering call in setPhoneState()"); // force routing command to audio hardware when starting a call // even if no device change is needed force = true; for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]; } } else if (isStateInCall(oldState) && !isStateInCall(state)) { ALOGV(" Exiting call in setPhoneState()"); // force routing command to audio hardware when exiting a call // even if no device change is needed force = true; for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = sVolumeProfiles[AUDIO_STREAM_DTMF][j]; } } else if (isStateInCall(state) && (state != oldState)) { ALOGV(" Switching between telephony and VoIP in setPhoneState()"); // force routing command to audio hardware when switching between telephony and VoIP // even if no device change is needed force = true; } // check for device and output changes triggered by new phone state checkA2dpSuspend(); checkOutputForAllStrategies(); updateDevicesAndOutputs(); sp hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); int delayMs = 0; if (isStateInCall(state)) { nsecs_t sysTime = systemTime(); 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(STRATEGY_MEDIA, SONIFICATION_HEADSET_MUSIC_DELAY, sysTime) || desc->isStrategyActive(STRATEGY_SONIFICATION, SONIFICATION_HEADSET_MUSIC_DELAY, sysTime)) && (delayMs < (int)desc->mLatency*2)) { delayMs = desc->mLatency*2; } setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); } } // Note that despite the fact that getNewOutputDevice() is called on the primary output, // the device returned is not necessarily reachable via this output audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); // force routing command to audio hardware when ending call // even if no device change is needed if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) { rxDevice = hwOutputDesc->device(); } if (state == AUDIO_MODE_IN_CALL) { updateCallRouting(rxDevice, delayMs); } else if (oldState == AUDIO_MODE_IN_CALL) { if (mCallRxPatch != 0) { mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0); mCallRxPatch.clear(); } if (mCallTxPatch != 0) { mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0); mCallTxPatch.clear(); } setOutputDevice(mPrimaryOutput, rxDevice, force, 0); } else { setOutputDevice(mPrimaryOutput, rxDevice, force, 0); } // if entering in call state, handle special case of active streams // pertaining to sonification strategy see handleIncallSonification() if (isStateInCall(state)) { ALOGV("setPhoneState() in call state management: new state is %d", state); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } handleIncallSonification((audio_stream_type_t)stream, true, true); } } // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE if (state == AUDIO_MODE_RINGTONE && isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { mLimitRingtoneVolume = true; } else { mLimitRingtoneVolume = false; } } 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, mPhoneState); bool forceVolumeReeval = false; switch(usage) { case AUDIO_POLICY_FORCE_FOR_COMMUNICATION: if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_NONE) { ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); return; } forceVolumeReeval = true; mForceUse[usage] = config; break; case AUDIO_POLICY_FORCE_FOR_MEDIA: if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && config != AUDIO_POLICY_FORCE_ANALOG_DOCK && config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) { ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config); return; } mForceUse[usage] = config; break; case AUDIO_POLICY_FORCE_FOR_RECORD: if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && config != AUDIO_POLICY_FORCE_NONE) { ALOGW("setForceUse() invalid config %d for FOR_RECORD", config); return; } mForceUse[usage] = config; break; case AUDIO_POLICY_FORCE_FOR_DOCK: if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK && config != AUDIO_POLICY_FORCE_BT_DESK_DOCK && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && config != AUDIO_POLICY_FORCE_ANALOG_DOCK && config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) { ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); } forceVolumeReeval = true; mForceUse[usage] = config; break; case AUDIO_POLICY_FORCE_FOR_SYSTEM: if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); } forceVolumeReeval = true; mForceUse[usage] = config; break; case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO: if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) { ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config); } mForceUse[usage] = config; break; default: ALOGW("setForceUse() invalid usage %d", usage); break; } // check for device and output changes triggered by new force usage checkA2dpSuspend(); checkOutputForAllStrategies(); updateDevicesAndOutputs(); if (mPhoneState == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/); updateCallRouting(newDevice); } for (size_t i = 0; i < mOutputs.size(); i++) { audio_io_handle_t output = mOutputs.keyAt(i); audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/); if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); } if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { applyStreamVolumes(output, newDevice, 0, true); } } audio_io_handle_t activeInput = getActiveInput(); if (activeInput != 0) { setInputDevice(activeInput, getNewInputDevice(activeInput)); } } audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) { return mForceUse[usage]; } void AudioPolicyManager::setSystemProperty(const char* property, const char* value) { ALOGV("setSystemProperty() property %s, value %s", property, value); } // Find a direct output profile compatible with the parameters passed, even if the input flags do // not explicitly request a direct output sp AudioPolicyManager::getProfileForDirectOutput( audio_devices_t device, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags) { for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { sp profile = mHwModules[i]->mOutputProfiles[j]; bool found = profile->isCompatibleProfile(device, String8(""), samplingRate, NULL /*updatedSamplingRate*/, format, channelMask, flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ? AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT); if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) { return profile; } } } return 0; } audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { routing_strategy strategy = getStrategy(stream); audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x", device, stream, samplingRate, format, channelMask, flags); return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE, stream, samplingRate,format, channelMask, flags, offloadInfo); } status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, audio_io_handle_t *output, audio_session_t session, audio_stream_type_t *stream, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { audio_attributes_t attributes; if (attr != NULL) { if (!isValidAttributes(attr)) { ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", attr->usage, attr->content_type, attr->flags, attr->tags); return BAD_VALUE; } attributes = *attr; } else { if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) { ALOGE("getOutputForAttr(): invalid stream type"); return BAD_VALUE; } stream_type_to_audio_attributes(*stream, &attributes); } for (size_t i = 0; i < mPolicyMixes.size(); i++) { sp desc; if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) { for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) { if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule && mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) || (RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule && mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) { desc = mPolicyMixes[i]->mOutput; break; } if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && strncmp(attributes.tags + strlen("addr="), mPolicyMixes[i]->mMix.mRegistrationId.string(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { desc = mPolicyMixes[i]->mOutput; break; } } } else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) { if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE && strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && strncmp(attributes.tags + strlen("addr="), mPolicyMixes[i]->mMix.mRegistrationId.string(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { desc = mPolicyMixes[i]->mOutput; } } if (desc != 0) { if (!audio_is_linear_pcm(format)) { return BAD_VALUE; } desc->mPolicyMix = &mPolicyMixes[i]->mMix; *stream = streamTypefromAttributesInt(&attributes); *output = desc->mIoHandle; ALOGV("getOutputForAttr() returns output %d", *output); return NO_ERROR; } } if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) { ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE"); return BAD_VALUE; } ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x", attributes.usage, attributes.content_type, attributes.tags, attributes.flags); routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes); audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x", device, samplingRate, format, channelMask, flags); *stream = streamTypefromAttributesInt(&attributes); *output = getOutputForDevice(device, session, *stream, samplingRate, format, channelMask, flags, offloadInfo); if (*output == AUDIO_IO_HANDLE_NONE) { return INVALID_OPERATION; } return NO_ERROR; } audio_io_handle_t AudioPolicyManager::getOutputForDevice( audio_devices_t device, audio_session_t session __unused, audio_stream_type_t stream, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; uint32_t latency = 0; status_t status; #ifdef AUDIO_POLICY_TEST if (mCurOutput != 0) { ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d", mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput); if (mTestOutputs[mCurOutput] == 0) { ALOGV("getOutput() opening test output"); sp outputDesc = new AudioOutputDescriptor(NULL); outputDesc->mDevice = mTestDevice; outputDesc->mLatency = mTestLatencyMs; outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0); outputDesc->mRefCount[stream] = 0; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = mTestSamplingRate; config.channel_mask = mTestChannels; config.format = mTestFormat; if (offloadInfo != NULL) { config.offload_info = *offloadInfo; } status = mpClientInterface->openOutput(0, &mTestOutputs[mCurOutput], &config, &outputDesc->mDevice, String8(""), &outputDesc->mLatency, outputDesc->mFlags); if (status == NO_ERROR) { outputDesc->mSamplingRate = config.sample_rate; outputDesc->mFormat = config.format; outputDesc->mChannelMask = config.channel_mask; AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"),mCurOutput); mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString()); addOutput(mTestOutputs[mCurOutput], outputDesc); } } return mTestOutputs[mCurOutput]; } #endif //AUDIO_POLICY_TEST // 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); } // only allow deep buffering for music stream type if (stream != AUDIO_STREAM_MUSIC) { flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); } sp profile; // 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(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE && audio_channel_count_from_out_mask(channelMask) <= 2) { goto non_direct_output; } // 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 (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || !isNonOffloadableEffectEnabled()) { profile = getProfileForDirectOutput(device, samplingRate, format, channelMask, (audio_output_flags_t)flags); } if (profile != 0) { sp outputDesc = NULL; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (!desc->isDuplicated() && (profile == desc->mProfile)) { outputDesc = desc; // reuse direct output if currently open and configured with same parameters if ((samplingRate == outputDesc->mSamplingRate) && (format == outputDesc->mFormat) && (channelMask == outputDesc->mChannelMask)) { outputDesc->mDirectOpenCount++; ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i)); return mOutputs.keyAt(i); } } } // close direct output if currently open and configured with different parameters if (outputDesc != NULL) { closeOutput(outputDesc->mIoHandle); } outputDesc = new AudioOutputDescriptor(profile); outputDesc->mDevice = device; outputDesc->mLatency = 0; outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = samplingRate; config.channel_mask = channelMask; config.format = format; if (offloadInfo != NULL) { config.offload_info = *offloadInfo; } status = mpClientInterface->openOutput(profile->mModule->mHandle, &output, &config, &outputDesc->mDevice, String8(""), &outputDesc->mLatency, outputDesc->mFlags); // only accept an output with the requested parameters if (status != NO_ERROR || (samplingRate != 0 && samplingRate != config.sample_rate) || (format != AUDIO_FORMAT_DEFAULT && format != config.format) || (channelMask != 0 && channelMask != config.channel_mask)) { ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d," "format %d %d, channelMask %04x %04x", output, samplingRate, outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask, outputDesc->mChannelMask); if (output != AUDIO_IO_HANDLE_NONE) { mpClientInterface->closeOutput(output); } return AUDIO_IO_HANDLE_NONE; } outputDesc->mSamplingRate = config.sample_rate; outputDesc->mChannelMask = config.channel_mask; outputDesc->mFormat = config.format; outputDesc->mRefCount[stream] = 0; outputDesc->mStopTime[stream] = 0; outputDesc->mDirectOpenCount = 1; audio_io_handle_t srcOutput = getOutputForEffect(); addOutput(output, outputDesc); audio_io_handle_t dstOutput = getOutputForEffect(); if (dstOutput == output) { mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput); } mPreviousOutputs = mOutputs; ALOGV("getOutput() returns new direct output %d", output); mpClientInterface->onAudioPortListUpdate(); return output; } non_direct_output: // 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(format)) { // get which output is suitable for the specified stream. The actual // routing change will happen when startOutput() will be called SortedVector outputs = getOutputsForDevice(device, mOutputs); // 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); output = selectOutput(outputs, flags, format); } ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); ALOGV("getOutput() returns output %d", output); return output; } audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector& outputs, audio_output_flags_t flags, audio_format_t format) { // select one output among several that provide a path to a particular device or set of // devices (the list was previously build by getOutputsForDevice()). // The priority is as follows: // 1: the output with the highest number of requested policy flags // 2: the primary output // 3: the first output in the list if (outputs.size() == 0) { return 0; } if (outputs.size() == 1) { return outputs[0]; } int maxCommonFlags = 0; audio_io_handle_t outputFlags = 0; audio_io_handle_t outputPrimary = 0; for (size_t i = 0; i < outputs.size(); i++) { sp outputDesc = mOutputs.valueFor(outputs[i]); if (!outputDesc->isDuplicated()) { // if a valid format is specified, skip output if not compatible if (format != AUDIO_FORMAT_INVALID) { if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { if (format != outputDesc->mFormat) { continue; } } else if (!audio_is_linear_pcm(format)) { continue; } } int commonFlags = popcount(outputDesc->mProfile->mFlags & flags); if (commonFlags > maxCommonFlags) { outputFlags = outputs[i]; maxCommonFlags = commonFlags; ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags); } if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { outputPrimary = outputs[i]; } } } if (outputFlags != 0) { return outputFlags; } if (outputPrimary != 0) { return outputPrimary; } return outputs[0]; } status_t AudioPolicyManager::startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session) { ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); ssize_t index = mOutputs.indexOfKey(output); if (index < 0) { ALOGW("startOutput() unknown output %d", output); return BAD_VALUE; } // cannot start playback of STREAM_TTS if any other output is being used uint32_t beaconMuteLatency = 0; if (stream == AUDIO_STREAM_TTS) { ALOGV("\t found BEACON stream"); if (isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) { return INVALID_OPERATION; } else { beaconMuteLatency = handleEventForBeacon(STARTING_BEACON); } } else { // some playback other than beacon starts beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); } sp outputDesc = mOutputs.valueAt(index); // 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->changeRefCount(stream, 1); if (outputDesc->mRefCount[stream] == 1) { // starting an output being rerouted? audio_devices_t newDevice; if (outputDesc->mPolicyMix != NULL) { newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; } else { newDevice = getNewOutputDevice(output, false /*fromCache*/); } routing_strategy strategy = getStrategy(stream); bool shouldWait = (strategy == STRATEGY_SONIFICATION) || (strategy == STRATEGY_SONIFICATION_RESPECTFUL) || (beaconMuteLatency > 0); uint32_t waitMs = beaconMuteLatency; bool force = false; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); if (desc != outputDesc) { // force a device change if any other output is managed by the same hw // module and has a current device selection that differs from selected device. // In this case, the audio HAL must receive the new device selection so that it can // change the device currently selected by the other active output. if (outputDesc->sharesHwModuleWith(desc) && desc->device() != newDevice) { 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 uint32_t latency = desc->latency(); if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) { waitMs = latency; } } } uint32_t muteWaitMs = setOutputDevice(output, newDevice, force); // handle special case for sonification while in call if (isInCall()) { handleIncallSonification(stream, true, false); } // apply volume rules for current stream and device if necessary checkAndSetVolume(stream, mStreams[stream].getVolumeIndex(newDevice), output, newDevice); // update the outputs if starting an output with a stream that can affect notification // routing handleNotificationRoutingForStream(stream); // Automatically enable the remote submix input when output is started on a re routing mix // of type MIX_TYPE_RECORDERS if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL && outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { setDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, outputDesc->mPolicyMix->mRegistrationId); } if (waitMs > muteWaitMs) { usleep((waitMs - muteWaitMs) * 2 * 1000); } } return NO_ERROR; } status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session) { ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session); ssize_t index = mOutputs.indexOfKey(output); if (index < 0) { ALOGW("stopOutput() unknown output %d", output); return BAD_VALUE; } sp outputDesc = mOutputs.valueAt(index); // always handle stream stop, check which stream type is stopping handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); // handle special case for sonification while in call if (isInCall()) { handleIncallSonification(stream, false, false); } if (outputDesc->mRefCount[stream] > 0) { // decrement usage count of this stream on the output outputDesc->changeRefCount(stream, -1); // store time at which the stream was stopped - see isStreamActive() if (outputDesc->mRefCount[stream] == 0) { // Automatically disable the remote submix input when output is stopped on a // re routing mix of type MIX_TYPE_RECORDERS if (audio_is_remote_submix_device(outputDesc->mDevice) && outputDesc->mPolicyMix != NULL && outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { setDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, outputDesc->mPolicyMix->mRegistrationId); } outputDesc->mStopTime[stream] = systemTime(); audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/); // 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 ...) setOutputDevice(output, newDevice, false, outputDesc->mLatency*2); // force restoring the device selection on other active outputs if it differs from the // one being selected for this output for (size_t i = 0; i < mOutputs.size(); i++) { audio_io_handle_t curOutput = mOutputs.keyAt(i); sp desc = mOutputs.valueAt(i); if (curOutput != output && desc->isActive() && outputDesc->sharesHwModuleWith(desc) && (newDevice != desc->device())) { setOutputDevice(curOutput, getNewOutputDevice(curOutput, false /*fromCache*/), true, outputDesc->mLatency*2); } } // update the outputs if stopping one with a stream that can affect notification routing handleNotificationRoutingForStream(stream); } return NO_ERROR; } else { ALOGW("stopOutput() refcount is already 0 for output %d", output); return INVALID_OPERATION; } } void AudioPolicyManager::releaseOutput(audio_io_handle_t output, audio_stream_type_t stream __unused, audio_session_t session __unused) { ALOGV("releaseOutput() %d", output); ssize_t index = mOutputs.indexOfKey(output); if (index < 0) { ALOGW("releaseOutput() releasing unknown output %d", output); return; } #ifdef AUDIO_POLICY_TEST int testIndex = testOutputIndex(output); if (testIndex != 0) { sp outputDesc = mOutputs.valueAt(index); if (outputDesc->isActive()) { mpClientInterface->closeOutput(output); mOutputs.removeItem(output); mTestOutputs[testIndex] = 0; } return; } #endif //AUDIO_POLICY_TEST sp desc = mOutputs.valueAt(index); if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { if (desc->mDirectOpenCount <= 0) { ALOGW("releaseOutput() invalid open count %d for output %d", desc->mDirectOpenCount, output); return; } if (--desc->mDirectOpenCount == 0) { closeOutput(output); // If effects where present on the output, audioflinger moved them to the primary // output by default: move them back to the appropriate output. audio_io_handle_t dstOutput = getOutputForEffect(); if (dstOutput != mPrimaryOutput) { mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput); } mpClientInterface->onAudioPortListUpdate(); } } } status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, audio_io_handle_t *input, audio_session_t session, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_input_flags_t flags) { ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x," "session %d, flags %#x", attr->source, samplingRate, format, channelMask, session, flags); *input = AUDIO_IO_HANDLE_NONE; audio_devices_t device; // handle legacy remote submix case where the address was not always specified String8 address = String8(""); bool isSoundTrigger = false; audio_source_t halInputSource = attr->source; AudioMix *policyMix = NULL; if (attr->source == AUDIO_SOURCE_REMOTE_SUBMIX && strncmp(attr->tags, "addr=", strlen("addr=")) == 0) { device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; address = String8(attr->tags + strlen("addr=")); ssize_t index = mPolicyMixes.indexOfKey(address); if (index < 0) { ALOGW("getInputForAttr() no policy for address %s", address.string()); return BAD_VALUE; } if (mPolicyMixes[index]->mMix.mMixType != MIX_TYPE_PLAYERS) { ALOGW("getInputForAttr() bad policy mix type for address %s", address.string()); return BAD_VALUE; } policyMix = &mPolicyMixes[index]->mMix; } else { device = getDeviceForInputSource(attr->source, &policyMix); if (device == AUDIO_DEVICE_NONE) { ALOGW("getInputForAttr() could not find device for source %d", attr->source); return BAD_VALUE; } if (policyMix != NULL) { address = policyMix->mRegistrationId; } else if (audio_is_remote_submix_device(device)) { address = String8("0"); } // adapt channel selection to input source switch (attr->source) { case AUDIO_SOURCE_VOICE_UPLINK: channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK; break; case AUDIO_SOURCE_VOICE_DOWNLINK: channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK; break; case AUDIO_SOURCE_VOICE_CALL: channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK; break; default: break; } if (attr->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; } } } sp profile = getInputProfile(device, address, samplingRate, format, channelMask, flags); if (profile == 0) { //retry without flags audio_input_flags_t log_flags = flags; flags = AUDIO_INPUT_FLAG_NONE; profile = getInputProfile(device, address, samplingRate, format, channelMask, flags); if (profile == 0) { ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u," "format %#x, channelMask 0x%X, flags %#x", device, samplingRate, format, channelMask, log_flags); return BAD_VALUE; } } if (profile->mModule->mHandle == 0) { ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName); return NO_INIT; } audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = samplingRate; config.channel_mask = channelMask; config.format = format; status_t status = mpClientInterface->openInput(profile->mModule->mHandle, input, &config, &device, address, halInputSource, flags); // only accept input with the exact requested set of parameters if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE || (samplingRate != config.sample_rate) || (format != config.format) || (channelMask != config.channel_mask)) { ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x", samplingRate, format, channelMask); if (*input != AUDIO_IO_HANDLE_NONE) { mpClientInterface->closeInput(*input); } return BAD_VALUE; } sp inputDesc = new AudioInputDescriptor(profile); inputDesc->mInputSource = attr->source; inputDesc->mRefCount = 0; inputDesc->mOpenRefCount = 1; inputDesc->mSamplingRate = samplingRate; inputDesc->mFormat = format; inputDesc->mChannelMask = channelMask; inputDesc->mDevice = device; inputDesc->mSessions.add(session); inputDesc->mIsSoundTrigger = isSoundTrigger; inputDesc->mPolicyMix = policyMix; addInput(*input, inputDesc); mpClientInterface->onAudioPortListUpdate(); return NO_ERROR; } status_t AudioPolicyManager::startInput(audio_io_handle_t input, audio_session_t session) { ALOGV("startInput() input %d", input); ssize_t index = mInputs.indexOfKey(input); if (index < 0) { ALOGW("startInput() unknown input %d", input); return BAD_VALUE; } sp inputDesc = mInputs.valueAt(index); index = inputDesc->mSessions.indexOf(session); if (index < 0) { ALOGW("startInput() unknown session %d on input %d", session, input); return BAD_VALUE; } // virtual input devices are compatible with other input devices if (!isVirtualInputDevice(inputDesc->mDevice)) { // for a non-virtual input device, check if there is another (non-virtual) active input audio_io_handle_t activeInput = getActiveInput(); if (activeInput != 0 && activeInput != input) { // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed, // otherwise the active input continues and the new input cannot be started. sp activeDesc = mInputs.valueFor(activeInput); if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) { ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput); stopInput(activeInput, activeDesc->mSessions.itemAt(0)); releaseInput(activeInput, activeDesc->mSessions.itemAt(0)); } else { ALOGE("startInput(%d) failed: other input %d already started", input, activeInput); return INVALID_OPERATION; } } } if (inputDesc->mRefCount == 0) { if (activeInputsCount() == 0) { SoundTrigger::setCaptureState(true); } setInputDevice(input, getNewInputDevice(input), true /* force */); // 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->mDevice)) { String8 address = String8(""); if (inputDesc->mPolicyMix == NULL) { address = String8("0"); } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { address = inputDesc->mPolicyMix->mRegistrationId; } if (address != "") { setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address); } } } ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource); inputDesc->mRefCount++; return NO_ERROR; } status_t AudioPolicyManager::stopInput(audio_io_handle_t input, audio_session_t session) { ALOGV("stopInput() input %d", input); ssize_t index = mInputs.indexOfKey(input); if (index < 0) { ALOGW("stopInput() unknown input %d", input); return BAD_VALUE; } sp inputDesc = mInputs.valueAt(index); index = inputDesc->mSessions.indexOf(session); if (index < 0) { ALOGW("stopInput() unknown session %d on input %d", session, input); return BAD_VALUE; } if (inputDesc->mRefCount == 0) { ALOGW("stopInput() input %d already stopped", input); return INVALID_OPERATION; } inputDesc->mRefCount--; if (inputDesc->mRefCount == 0) { // 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->mDevice)) { String8 address = String8(""); if (inputDesc->mPolicyMix == NULL) { address = String8("0"); } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) { address = inputDesc->mPolicyMix->mRegistrationId; } if (address != "") { setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address); } } resetInputDevice(input); if (activeInputsCount() == 0) { SoundTrigger::setCaptureState(false); } } return NO_ERROR; } void AudioPolicyManager::releaseInput(audio_io_handle_t input, audio_session_t session) { ALOGV("releaseInput() %d", input); ssize_t index = mInputs.indexOfKey(input); if (index < 0) { ALOGW("releaseInput() releasing unknown input %d", input); return; } sp inputDesc = mInputs.valueAt(index); ALOG_ASSERT(inputDesc != 0); index = inputDesc->mSessions.indexOf(session); if (index < 0) { ALOGW("releaseInput() unknown session %d on input %d", session, input); return; } inputDesc->mSessions.remove(session); if (inputDesc->mOpenRefCount == 0) { ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount); return; } inputDesc->mOpenRefCount--; if (inputDesc->mOpenRefCount > 0) { ALOGV("releaseInput() exit > 0"); return; } closeInput(input); mpClientInterface->onAudioPortListUpdate(); ALOGV("releaseInput() exit"); } void AudioPolicyManager::closeAllInputs() { bool patchRemoved = false; for(size_t input_index = 0; input_index < mInputs.size(); input_index++) { sp inputDesc = mInputs.valueAt(input_index); ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); if (patch_index >= 0) { sp patchDesc = mAudioPatches.valueAt(patch_index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); mAudioPatches.removeItemsAt(patch_index); patchRemoved = true; } mpClientInterface->closeInput(mInputs.keyAt(input_index)); } mInputs.clear(); nextAudioPortGeneration(); if (patchRemoved) { mpClientInterface->onAudioPatchListUpdate(); } } 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 || indexMin >= indexMax) { ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax); return; } mStreams[stream].mIndexMin = indexMin; mStreams[stream].mIndexMax = indexMax; //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now if (stream == AUDIO_STREAM_MUSIC) { mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMin = indexMin; mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMax = indexMax; } } status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, int index, audio_devices_t device) { if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { return BAD_VALUE; } if (!audio_is_output_device(device)) { return BAD_VALUE; } // Force max volume if stream cannot be muted if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", stream, device, index); // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and // clear all device specific values if (device == AUDIO_DEVICE_OUT_DEFAULT) { mStreams[stream].mIndexCur.clear(); } mStreams[stream].mIndexCur.add(device, index); // update volume on all outputs whose current device is also selected by the same // strategy as the device specified by the caller audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE; if (stream == AUDIO_STREAM_MUSIC) { mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexCur.add(device, index); accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/); } if ((device != AUDIO_DEVICE_OUT_DEFAULT) && (device & (strategyDevice | accessibilityDevice)) == 0) { return NO_ERROR; } status_t status = NO_ERROR; for (size_t i = 0; i < mOutputs.size(); i++) { audio_devices_t curDevice = getDeviceForVolume(mOutputs.valueAt(i)->device()); if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) { status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); if (volStatus != NO_ERROR) { status = volStatus; } } if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) { status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY, index, mOutputs.keyAt(i), curDevice); } } return status; } status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream, int *index, audio_devices_t device) { if (index == NULL) { return BAD_VALUE; } if (!audio_is_output_device(device)) { return BAD_VALUE; } // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to // the strategy the stream belongs to. if (device == AUDIO_DEVICE_OUT_DEFAULT) { device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); } device = getDeviceForVolume(device); *index = mStreams[stream].getVolumeIndex(device); ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); return NO_ERROR; } audio_io_handle_t AudioPolicyManager::selectOutputForEffects( const SortedVector& outputs) { // 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 first output in the list if (outputs.size() == 0) { return 0; } audio_io_handle_t outputOffloaded = 0; audio_io_handle_t outputDeepBuffer = 0; for (size_t i = 0; i < outputs.size(); i++) { sp desc = mOutputs.valueFor(outputs[i]); ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags); if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { outputOffloaded = outputs[i]; } if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) { outputDeepBuffer = outputs[i]; } } ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d", outputOffloaded, outputDeepBuffer); if (outputOffloaded != 0) { return outputOffloaded; } if (outputDeepBuffer != 0) { return outputDeepBuffer; } return outputs[0]; } audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc) { // apply simple rule where global effects are attached to the same output as MUSIC streams routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC); audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); SortedVector dstOutputs = getOutputsForDevice(device, mOutputs); audio_io_handle_t output = selectOutputForEffects(dstOutputs); ALOGV("getOutputForEffect() got output %d for fx %s flags %x", output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags); return output; } status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc, audio_io_handle_t io, uint32_t strategy, int session, int id) { 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; } } if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", desc->name, desc->memoryUsage); return INVALID_OPERATION; } mTotalEffectsMemory += desc->memoryUsage; ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", desc->name, io, strategy, session, id); ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); sp effectDesc = new EffectDescriptor(); memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t)); effectDesc->mIo = io; effectDesc->mStrategy = (routing_strategy)strategy; effectDesc->mSession = session; effectDesc->mEnabled = false; mEffects.add(id, effectDesc); return NO_ERROR; } status_t AudioPolicyManager::unregisterEffect(int id) { ssize_t index = mEffects.indexOfKey(id); if (index < 0) { ALOGW("unregisterEffect() unknown effect ID %d", id); return INVALID_OPERATION; } sp effectDesc = mEffects.valueAt(index); setEffectEnabled(effectDesc, false); if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) { ALOGW("unregisterEffect() memory %d too big for total %d", effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); effectDesc->mDesc.memoryUsage = mTotalEffectsMemory; } mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage; ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d", effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); mEffects.removeItem(id); return NO_ERROR; } status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled) { ssize_t index = mEffects.indexOfKey(id); if (index < 0) { ALOGW("unregisterEffect() unknown effect ID %d", id); return INVALID_OPERATION; } return setEffectEnabled(mEffects.valueAt(index), enabled); } status_t AudioPolicyManager::setEffectEnabled(const sp& effectDesc, bool enabled) { if (enabled == effectDesc->mEnabled) { ALOGV("setEffectEnabled(%s) effect already %s", enabled?"true":"false", enabled?"enabled":"disabled"); return INVALID_OPERATION; } if (enabled) { if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS", effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10); return INVALID_OPERATION; } mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad; ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); } else { if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) { ALOGW("setEffectEnabled(false) CPU load %d too high for total %d", effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; } mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad; ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); } effectDesc->mEnabled = enabled; return NO_ERROR; } bool AudioPolicyManager::isNonOffloadableEffectEnabled() { for (size_t i = 0; i < mEffects.size(); i++) { sp effectDesc = mEffects.valueAt(i); if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) && ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) { ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d", effectDesc->mDesc.name, effectDesc->mSession); return true; } } return false; } bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const { nsecs_t sysTime = systemTime(); for (size_t i = 0; i < mOutputs.size(); i++) { const sp outputDesc = mOutputs.valueAt(i); if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) { return true; } } return false; } bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const { nsecs_t sysTime = systemTime(); for (size_t i = 0; i < mOutputs.size(); i++) { const sp outputDesc = mOutputs.valueAt(i); if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && outputDesc->isStreamActive(stream, inPastMs, sysTime)) { // do not consider re routing (when the output is going to a dynamic policy) // as "remote playback" if (outputDesc->mPolicyMix == NULL) { return true; } } } return 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->mInputSource == (int)source || (source == AUDIO_SOURCE_VOICE_RECOGNITION && inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) && (inputDescriptor->mRefCount > 0)) { 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(Vector mixes) { sp module; for (size_t i = 0; i < mHwModules.size(); i++) { if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 && mHwModules[i]->mHandle != 0) { module = mHwModules[i]; break; } } if (module == 0) { return INVALID_OPERATION; } ALOGV("registerPolicyMixes() num mixes %d", mixes.size()); for (size_t i = 0; i < mixes.size(); i++) { String8 address = mixes[i].mRegistrationId; ssize_t index = mPolicyMixes.indexOfKey(address); if (index >= 0) { ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string()); continue; } audio_config_t outputConfig = mixes[i].mFormat; audio_config_t inputConfig = mixes[i].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; module->addOutputProfile(address, &outputConfig, AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address); module->addInputProfile(address, &inputConfig, AUDIO_DEVICE_IN_REMOTE_SUBMIX, address); sp policyMix = new AudioPolicyMix(); policyMix->mMix = mixes[i]; mPolicyMixes.add(address, policyMix); if (mixes[i].mMixType == MIX_TYPE_PLAYERS) { setDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address.string()); } else { setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, address.string()); } } return NO_ERROR; } status_t AudioPolicyManager::unregisterPolicyMixes(Vector mixes) { sp module; for (size_t i = 0; i < mHwModules.size(); i++) { if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 && mHwModules[i]->mHandle != 0) { module = mHwModules[i]; break; } } if (module == 0) { return INVALID_OPERATION; } ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size()); for (size_t i = 0; i < mixes.size(); i++) { String8 address = mixes[i].mRegistrationId; ssize_t index = mPolicyMixes.indexOfKey(address); if (index < 0) { ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string()); continue; } mPolicyMixes.removeItemsAt(index); if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { setDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address.string()); } if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, address.string()); } module->removeOutputProfile(address); module->removeInputProfile(address); } return NO_ERROR; } status_t AudioPolicyManager::dump(int fd) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this); result.append(buffer); snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput); result.append(buffer); snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState); result.append(buffer); snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); result.append(buffer); snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]); result.append(buffer); snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]); result.append(buffer); snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]); result.append(buffer); snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]); result.append(buffer); snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO]); result.append(buffer); snprintf(buffer, SIZE, " Available output devices:\n"); result.append(buffer); write(fd, result.string(), result.size()); for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { mAvailableOutputDevices[i]->dump(fd, 2, i); } snprintf(buffer, SIZE, "\n Available input devices:\n"); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { mAvailableInputDevices[i]->dump(fd, 2, i); } snprintf(buffer, SIZE, "\nHW Modules dump:\n"); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mHwModules.size(); i++) { snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1); write(fd, buffer, strlen(buffer)); mHwModules[i]->dump(fd); } snprintf(buffer, SIZE, "\nOutputs dump:\n"); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mOutputs.size(); i++) { snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i)); write(fd, buffer, strlen(buffer)); mOutputs.valueAt(i)->dump(fd); } snprintf(buffer, SIZE, "\nInputs dump:\n"); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mInputs.size(); i++) { snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i)); write(fd, buffer, strlen(buffer)); mInputs.valueAt(i)->dump(fd); } snprintf(buffer, SIZE, "\nStreams dump:\n"); write(fd, buffer, strlen(buffer)); snprintf(buffer, SIZE, " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n"); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) { snprintf(buffer, SIZE, " %02zu ", i); write(fd, buffer, strlen(buffer)); mStreams[i].dump(fd); } snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n", (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory); write(fd, buffer, strlen(buffer)); snprintf(buffer, SIZE, "Registered effects:\n"); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mEffects.size(); i++) { snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i)); write(fd, buffer, strlen(buffer)); mEffects.valueAt(i)->dump(fd); } snprintf(buffer, SIZE, "\nAudio Patches:\n"); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mAudioPatches.size(); i++) { mAudioPatches[i]->dump(fd, 2, i); } 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. bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo) { ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," " BitRate=%u, duration=%" PRId64 " us, has_video=%d", offloadInfo.sample_rate, offloadInfo.channel_mask, offloadInfo.format, offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, offloadInfo.has_video); // Check if offload has been disabled char propValue[PROPERTY_VALUE_MAX]; if (property_get("audio.offload.disable", propValue, "0")) { if (atoi(propValue) != 0) { ALOGV("offload disabled by audio.offload.disable=%s", propValue ); return false; } } // Check if stream type is music, then only allow offload as of now. if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) { ALOGV("isOffloadSupported: stream_type != MUSIC, returning false"); return false; } //TODO: enable audio offloading with video when ready if (offloadInfo.has_video) { ALOGV("isOffloadSupported: has_video == true, returning false"); return false; } //If duration is less than minimum value defined in property, return false if (property_get("audio.offload.min.duration.secs", propValue, NULL)) { if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) { ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); return false; } } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { ALOGV("Offload denied by duration < default min(=%u)", 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 (isNonOffloadableEffectEnabled()) { return false; } // See if there is a profile to support this. // AUDIO_DEVICE_NONE sp profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */, offloadInfo.sample_rate, offloadInfo.format, offloadInfo.channel_mask, AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT "); return (profile != 0); } status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role, audio_port_type_t type, unsigned int *num_ports, struct audio_port *ports, unsigned int *generation) { if (num_ports == NULL || (*num_ports != 0 && ports == NULL) || generation == NULL) { return BAD_VALUE; } ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports); if (ports == NULL) { *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) { if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) { for (size_t i = 0; i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) { mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]); } *num_ports += mAvailableOutputDevices.size(); } if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) { for (size_t i = 0; i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) { mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]); } *num_ports += mAvailableInputDevices.size(); } } 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::getAudioPort(struct audio_port *port __unused) { return NO_ERROR; } sp AudioPolicyManager::getOutputFromId( audio_port_handle_t id) const { sp outputDesc = NULL; for (size_t i = 0; i < mOutputs.size(); i++) { outputDesc = mOutputs.valueAt(i); if (outputDesc->mId == id) { break; } } return outputDesc; } sp AudioPolicyManager::getInputFromId( audio_port_handle_t id) const { sp inputDesc = NULL; for (size_t i = 0; i < mInputs.size(); i++) { inputDesc = mInputs.valueAt(i); if (inputDesc->mId == id) { break; } } return inputDesc; } sp AudioPolicyManager::getModuleForDevice( audio_devices_t device) const { sp module; for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } if (audio_is_output_device(device)) { for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) { return mHwModules[i]; } } } else { for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) { return mHwModules[i]; } } } } return module; } sp AudioPolicyManager::getModuleFromName(const char *name) const { sp module; for (size_t i = 0; i < mHwModules.size(); i++) { if (strcmp(mHwModules[i]->mName, name) == 0) { return mHwModules[i]; } } return module; } audio_devices_t AudioPolicyManager::availablePrimaryOutputDevices() { sp outputDesc = mOutputs.valueFor(mPrimaryOutput); audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types(); return devices & mAvailableOutputDevices.types(); } audio_devices_t AudioPolicyManager::availablePrimaryInputDevices() { audio_module_handle_t primaryHandle = mOutputs.valueFor(mPrimaryOutput)->mProfile->mModule->mHandle; audio_devices_t devices = AUDIO_DEVICE_NONE; for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { if (mAvailableInputDevices[i]->mModule->mHandle == primaryHandle) { devices |= mAvailableInputDevices[i]->mDeviceType; } } return devices; } status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, audio_patch_handle_t *handle, uid_t uid) { ALOGV("createAudioPatch()"); if (handle == NULL || patch == NULL) { return BAD_VALUE; } ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks); if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX || patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) { 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 patchDesc; ssize_t index = mAudioPatches.indexOfKey(*handle); ALOGV("createAudioPatch source id %d role %d type %d", 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("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id, patch->sinks[i].role, patch->sinks[i].type); } #endif if (index >= 0) { patchDesc = mAudioPatches.valueAt(index); ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", mUidCached, patchDesc->mUid, uid); if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { return INVALID_OPERATION; } } else { *handle = 0; } if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { sp outputDesc = getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("createAudioPatch() output not found for id %d", 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("createAudioPatch() source id differs for patch current id %d new id %d", 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("createAudioPatch() source mix but sink is not a device"); return INVALID_OPERATION; } sp devDesc = mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); if (devDesc == 0) { ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id); return BAD_VALUE; } if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType, devDesc->mAddress, patch->sources[0].sample_rate, NULL, // updatedSamplingRate patch->sources[0].format, patch->sources[0].channel_mask, AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) { ALOGV("createAudioPatch() profile not supported for device %08x", devDesc->mDeviceType); return INVALID_OPERATION; } devices.add(devDesc); } if (devices.size() == 0) { return INVALID_OPERATION; } // TODO: reconfigure output format and channels here ALOGV("createAudioPatch() setting device %08x on output %d", devices.types(), outputDesc->mIoHandle); setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided"); } patchDesc = mAudioPatches.valueAt(index); patchDesc->mUid = uid; ALOGV("createAudioPatch() success"); } else { ALOGW("createAudioPatch() setOutputDevice() failed to create a patch"); 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 = 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 devDesc = mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); if (devDesc == 0) { return BAD_VALUE; } if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType, devDesc->mAddress, patch->sinks[0].sample_rate, NULL, /*updatedSampleRate*/ patch->sinks[0].format, patch->sinks[0].channel_mask, // FIXME for the parameter type, // and the NONE (audio_output_flags_t) AUDIO_INPUT_FLAG_NONE)) { return INVALID_OPERATION; } // TODO: reconfigure output format and channels here ALOGV("createAudioPatch() setting device %08x on output %d", devDesc->mDeviceType, inputDesc->mIoHandle); setInputDevice(inputDesc->mIoHandle, devDesc->mDeviceType, true, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided"); } patchDesc = mAudioPatches.valueAt(index); patchDesc->mUid = uid; ALOGV("createAudioPatch() success"); } else { ALOGW("createAudioPatch() setInputDevice() failed to create a patch"); 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 srcDeviceDesc = mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); if (srcDeviceDesc == 0) { return BAD_VALUE; } //update source and sink with our own data as the data passed in the patch may // be incomplete. struct audio_patch newPatch = *patch; srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]); for (size_t i = 0; i < patch->num_sinks; i++) { if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) { ALOGV("createAudioPatch() source device but one sink is not a device"); return INVALID_OPERATION; } sp sinkDeviceDesc = mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id); if (sinkDeviceDesc == 0) { return BAD_VALUE; } sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]); if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) { // only one sink supported when connected devices across HW modules if (patch->num_sinks > 1) { return INVALID_OPERATION; } SortedVector outputs = getOutputsForDevice(sinkDeviceDesc->mDeviceType, 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 audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); if (output != AUDIO_IO_HANDLE_NONE) { sp outputDesc = mOutputs.valueFor(output); if (outputDesc->isDuplicated()) { return INVALID_OPERATION; } outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]); newPatch.num_sources = 2; } } } // TODO: check from routing capabilities in config file and other conflicting patches audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; if (index >= 0) { afPatchHandle = patchDesc->mAfPatchHandle; } status_t status = mpClientInterface->createAudioPatch(&newPatch, &afPatchHandle, 0); ALOGV("createAudioPatch() patch panel returned %d patchHandle %d", status, afPatchHandle); if (status == NO_ERROR) { if (index < 0) { patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), &newPatch, uid); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = newPatch; } patchDesc->mAfPatchHandle = afPatchHandle; *handle = patchDesc->mHandle; nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } else { ALOGW("createAudioPatch() patch panel could not connect device patch, error %d", 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("releaseAudioPatch() patch %d", handle); ssize_t index = mAudioPatches.indexOfKey(handle); if (index < 0) { return BAD_VALUE; } sp patchDesc = mAudioPatches.valueAt(index); ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d", mUidCached, patchDesc->mUid, uid); if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) { return INVALID_OPERATION; } struct audio_patch *patch = &patchDesc->mPatch; patchDesc->mUid = mUidCached; if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { sp outputDesc = getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id); return BAD_VALUE; } setOutputDevice(outputDesc->mIoHandle, getNewOutputDevice(outputDesc->mIoHandle, 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 = getInputFromId(patch->sinks[0].id); if (inputDesc == NULL) { ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id); return BAD_VALUE; } setInputDevice(inputDesc->mIoHandle, getNewInputDevice(inputDesc->mIoHandle), true, NULL); } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle; status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d", status, patchDesc->mAfPatchHandle); removeAudioPatch(patchDesc->mHandle); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } 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 (num_patches == NULL || (*num_patches != 0 && patches == NULL) || generation == NULL) { return BAD_VALUE; } ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu", *num_patches, patches, mAudioPatches.size()); if (patches == NULL) { *num_patches = 0; } size_t patchesWritten = 0; size_t patchesMax = *num_patches; for (size_t i = 0; i < mAudioPatches.size() && patchesWritten < patchesMax; i++) { patches[patchesWritten] = mAudioPatches[i]->mPatch; patches[patchesWritten++].id = mAudioPatches[i]->mHandle; ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d", i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks); } *num_patches = mAudioPatches.size(); *generation = curAudioPortGeneration(); ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches); return NO_ERROR; } 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 = 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 = 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::clearAudioPatches(uid_t uid) { for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) { sp patchDesc = mAudioPatches.valueAt(i); if (patchDesc->mUid == uid) { releaseAudioPatch(mAudioPatches.keyAt(i), uid); } } } status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session, audio_io_handle_t *ioHandle, audio_devices_t *device) { *session = (audio_session_t)mpClientInterface->newAudioUniqueId(); *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(); *device = getDeviceForInputSource(AUDIO_SOURCE_HOTWORD); mSoundTriggerSessions.add(*session, *ioHandle); return NO_ERROR; } status_t AudioPolicyManager::releaseSoundTriggerSession(audio_session_t session) { ssize_t index = mSoundTriggerSessions.indexOfKey(session); if (index < 0) { ALOGW("acquireSoundTriggerSession() session %d not registered", session); return BAD_VALUE; } mSoundTriggerSessions.removeItem(session); return NO_ERROR; } status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle, const sp& patch) { ssize_t index = mAudioPatches.indexOfKey(handle); if (index >= 0) { ALOGW("addAudioPatch() patch %d already in", handle); return ALREADY_EXISTS; } mAudioPatches.add(handle, patch); ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d" "sink handle %d", handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks, patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id); return NO_ERROR; } status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle) { ssize_t index = mAudioPatches.indexOfKey(handle); if (index < 0) { ALOGW("removeAudioPatch() patch %d not in", handle); return ALREADY_EXISTS; } ALOGV("removeAudioPatch() handle %d af handle %d", handle, mAudioPatches.valueAt(index)->mAfPatchHandle); mAudioPatches.removeItemsAt(index); return NO_ERROR; } // ---------------------------------------------------------------------------- // AudioPolicyManager // ---------------------------------------------------------------------------- uint32_t AudioPolicyManager::nextUniqueId() { return android_atomic_inc(&mNextUniqueId); } uint32_t AudioPolicyManager::nextAudioPortGeneration() { return android_atomic_inc(&mAudioPortGeneration); } AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface) : #ifdef AUDIO_POLICY_TEST Thread(false), #endif //AUDIO_POLICY_TEST mPrimaryOutput((audio_io_handle_t)0), mPhoneState(AUDIO_MODE_NORMAL), mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), mA2dpSuspended(false), mSpeakerDrcEnabled(false), mNextUniqueId(1), mAudioPortGeneration(1), mBeaconMuteRefCount(0), mBeaconPlayingRefCount(0), mBeaconMuted(false) { mUidCached = getuid(); mpClientInterface = clientInterface; for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) { mForceUse[i] = AUDIO_POLICY_FORCE_NONE; } mDefaultOutputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_OUT_SPEAKER); if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) { if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) { ALOGE("could not load audio policy configuration file, setting defaults"); defaultAudioPolicyConfig(); } } // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices // must be done after reading the policy initializeVolumeCurves(); // open all output streams needed to access attached devices audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types(); audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; for (size_t i = 0; i < mHwModules.size(); i++) { mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName); if (mHwModules[i]->mHandle == 0) { ALOGW("could not open HW module %s", mHwModules[i]->mName); continue; } // open all output streams needed to access attached devices // except for direct output streams that are only opened when they are actually // required by an app. // This also validates mAvailableOutputDevices list for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { const sp outProfile = mHwModules[i]->mOutputProfiles[j]; if (outProfile->mSupportedDevices.isEmpty()) { ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName); continue; } if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { continue; } audio_devices_t profileType = outProfile->mSupportedDevices.types(); if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) { profileType = mDefaultOutputDevice->mDeviceType; } else { // chose first device present in mSupportedDevices also part of // outputDeviceTypes for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { profileType = outProfile->mSupportedDevices[k]->mDeviceType; if ((profileType & outputDeviceTypes) != 0) { break; } } } if ((profileType & outputDeviceTypes) == 0) { continue; } sp outputDesc = new AudioOutputDescriptor(outProfile); outputDesc->mDevice = profileType; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = outputDesc->mSamplingRate; config.channel_mask = outputDesc->mChannelMask; config.format = outputDesc->mFormat; audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle, &output, &config, &outputDesc->mDevice, String8(""), &outputDesc->mLatency, outputDesc->mFlags); if (status != NO_ERROR) { ALOGW("Cannot open output stream for device %08x on hw module %s", outputDesc->mDevice, mHwModules[i]->mName); } else { outputDesc->mSamplingRate = config.sample_rate; outputDesc->mChannelMask = config.channel_mask; outputDesc->mFormat = config.format; for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType; ssize_t index = mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]); // give a valid ID to an attached device once confirmed it is reachable if ((index >= 0) && (mAvailableOutputDevices[index]->mId == 0)) { mAvailableOutputDevices[index]->mId = nextUniqueId(); mAvailableOutputDevices[index]->mModule = mHwModules[i]; } } if (mPrimaryOutput == 0 && outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { mPrimaryOutput = output; } addOutput(output, outputDesc); setOutputDevice(output, outputDesc->mDevice, true); } } // open input streams needed to access attached devices to validate // mAvailableInputDevices list for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { const sp inProfile = mHwModules[i]->mInputProfiles[j]; if (inProfile->mSupportedDevices.isEmpty()) { ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName); continue; } // chose first device present in mSupportedDevices also part of // inputDeviceTypes audio_devices_t profileType = AUDIO_DEVICE_NONE; for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { profileType = inProfile->mSupportedDevices[k]->mDeviceType; if (profileType & inputDeviceTypes) { break; } } if ((profileType & inputDeviceTypes) == 0) { continue; } sp inputDesc = new AudioInputDescriptor(inProfile); inputDesc->mInputSource = AUDIO_SOURCE_MIC; inputDesc->mDevice = profileType; // find the address DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType); // the inputs vector must be of size 1, but we don't want to crash here String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress : String8(""); ALOGV(" for input device 0x%x using address %s", profileType, address.string()); ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!"); audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = inputDesc->mSamplingRate; config.channel_mask = inputDesc->mChannelMask; config.format = inputDesc->mFormat; audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle, &input, &config, &inputDesc->mDevice, address, AUDIO_SOURCE_MIC, AUDIO_INPUT_FLAG_NONE); if (status == NO_ERROR) { for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType; ssize_t index = mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]); // give a valid ID to an attached device once confirmed it is reachable if ((index >= 0) && (mAvailableInputDevices[index]->mId == 0)) { mAvailableInputDevices[index]->mId = nextUniqueId(); mAvailableInputDevices[index]->mModule = mHwModules[i]; } } mpClientInterface->closeInput(input); } else { ALOGW("Cannot open input stream for device %08x on hw module %s", inputDesc->mDevice, mHwModules[i]->mName); } } } // make sure all attached devices have been allocated a unique ID for (size_t i = 0; i < mAvailableOutputDevices.size();) { if (mAvailableOutputDevices[i]->mId == 0) { ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType); mAvailableOutputDevices.remove(mAvailableOutputDevices[i]); continue; } i++; } for (size_t i = 0; i < mAvailableInputDevices.size();) { if (mAvailableInputDevices[i]->mId == 0) { ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType); mAvailableInputDevices.remove(mAvailableInputDevices[i]); continue; } i++; } // make sure default device is reachable if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) { ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType); } ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output"); updateDevicesAndOutputs(); #ifdef AUDIO_POLICY_TEST if (mPrimaryOutput != 0) { AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"), 0); mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); mTestDevice = AUDIO_DEVICE_OUT_SPEAKER; mTestSamplingRate = 44100; mTestFormat = AUDIO_FORMAT_PCM_16_BIT; mTestChannels = AUDIO_CHANNEL_OUT_STEREO; mTestLatencyMs = 0; mCurOutput = 0; mDirectOutput = false; for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { mTestOutputs[i] = 0; } const size_t SIZE = 256; char buffer[SIZE]; snprintf(buffer, SIZE, "AudioPolicyManagerTest"); run(buffer, ANDROID_PRIORITY_AUDIO); } #endif //AUDIO_POLICY_TEST } AudioPolicyManager::~AudioPolicyManager() { #ifdef AUDIO_POLICY_TEST exit(); #endif //AUDIO_POLICY_TEST for (size_t i = 0; i < mOutputs.size(); i++) { mpClientInterface->closeOutput(mOutputs.keyAt(i)); } for (size_t i = 0; i < mInputs.size(); i++) { mpClientInterface->closeInput(mInputs.keyAt(i)); } mAvailableOutputDevices.clear(); mAvailableInputDevices.clear(); mOutputs.clear(); mInputs.clear(); mHwModules.clear(); } status_t AudioPolicyManager::initCheck() { return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR; } #ifdef AUDIO_POLICY_TEST bool AudioPolicyManager::threadLoop() { ALOGV("entering threadLoop()"); while (!exitPending()) { String8 command; int valueInt; String8 value; Mutex::Autolock _l(mLock); mWaitWorkCV.waitRelative(mLock, milliseconds(50)); command = mpClientInterface->getParameters(0, String8("test_cmd_policy")); AudioParameter param = AudioParameter(command); if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR && valueInt != 0) { ALOGV("Test command %s received", command.string()); String8 target; if (param.get(String8("target"), target) != NO_ERROR) { target = "Manager"; } if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) { param.remove(String8("test_cmd_policy_output")); mCurOutput = valueInt; } if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_direct")); if (value == "false") { mDirectOutput = false; } else if (value == "true") { mDirectOutput = true; } } if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) { param.remove(String8("test_cmd_policy_input")); mTestInput = valueInt; } if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_format")); int format = AUDIO_FORMAT_INVALID; if (value == "PCM 16 bits") { format = AUDIO_FORMAT_PCM_16_BIT; } else if (value == "PCM 8 bits") { format = AUDIO_FORMAT_PCM_8_BIT; } else if (value == "Compressed MP3") { format = AUDIO_FORMAT_MP3; } if (format != AUDIO_FORMAT_INVALID) { if (target == "Manager") { mTestFormat = format; } else if (mTestOutputs[mCurOutput] != 0) { AudioParameter outputParam = AudioParameter(); outputParam.addInt(String8("format"), format); mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); } } } if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_channels")); int channels = 0; if (value == "Channels Stereo") { channels = AUDIO_CHANNEL_OUT_STEREO; } else if (value == "Channels Mono") { channels = AUDIO_CHANNEL_OUT_MONO; } if (channels != 0) { if (target == "Manager") { mTestChannels = channels; } else if (mTestOutputs[mCurOutput] != 0) { AudioParameter outputParam = AudioParameter(); outputParam.addInt(String8("channels"), channels); mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); } } } if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) { param.remove(String8("test_cmd_policy_sampleRate")); if (valueInt >= 0 && valueInt <= 96000) { int samplingRate = valueInt; if (target == "Manager") { mTestSamplingRate = samplingRate; } else if (mTestOutputs[mCurOutput] != 0) { AudioParameter outputParam = AudioParameter(); outputParam.addInt(String8("sampling_rate"), samplingRate); mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString()); } } } if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_reopen")); sp outputDesc = mOutputs.valueFor(mPrimaryOutput); mpClientInterface->closeOutput(mPrimaryOutput); audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; mOutputs.removeItem(mPrimaryOutput); sp outputDesc = new AudioOutputDescriptor(NULL); outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = outputDesc->mSamplingRate; config.channel_mask = outputDesc->mChannelMask; config.format = outputDesc->mFormat; status_t status = mpClientInterface->openOutput(moduleHandle, &mPrimaryOutput, &config, &outputDesc->mDevice, String8(""), &outputDesc->mLatency, outputDesc->mFlags); if (status != NO_ERROR) { ALOGE("Failed to reopen hardware output stream, " "samplingRate: %d, format %d, channels %d", outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask); } else { outputDesc->mSamplingRate = config.sample_rate; outputDesc->mChannelMask = config.channel_mask; outputDesc->mFormat = config.format; AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"), 0); mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); addOutput(mPrimaryOutput, outputDesc); } } mpClientInterface->setParameters(0, String8("test_cmd_policy=")); } } return false; } void AudioPolicyManager::exit() { { AutoMutex _l(mLock); requestExit(); mWaitWorkCV.signal(); } requestExitAndWait(); } int AudioPolicyManager::testOutputIndex(audio_io_handle_t output) { for (int i = 0; i < NUM_TEST_OUTPUTS; i++) { if (output == mTestOutputs[i]) return i; } return 0; } #endif //AUDIO_POLICY_TEST // --- void AudioPolicyManager::addOutput(audio_io_handle_t output, sp outputDesc) { outputDesc->mIoHandle = output; outputDesc->mId = nextUniqueId(); mOutputs.add(output, outputDesc); nextAudioPortGeneration(); } void AudioPolicyManager::addInput(audio_io_handle_t input, sp inputDesc) { inputDesc->mIoHandle = input; inputDesc->mId = nextUniqueId(); mInputs.add(input, inputDesc); nextAudioPortGeneration(); } void AudioPolicyManager::findIoHandlesByAddress(sp desc /*in*/, const String8 address /*in*/, SortedVector& outputs /*out*/) { // look for a match on the given address on the addresses of the outputs: // find the address by finding the patch that maps to this output ssize_t patchIdx = mAudioPatches.indexOfKey(desc->mPatchHandle); //ALOGV(" inspecting output %d (patch %d) for supported device=0x%x", // outputIdx, patchIdx, desc->mProfile->mSupportedDevices.types()); if (patchIdx >= 0) { const sp patchDesc = mAudioPatches.valueAt(patchIdx); const int numSinks = patchDesc->mPatch.num_sinks; for (ssize_t j=0; j < numSinks; j++) { if (patchDesc->mPatch.sinks[j].type == AUDIO_PORT_TYPE_DEVICE) { const char* patchAddr = patchDesc->mPatch.sinks[j].ext.device.address; if (strncmp(patchAddr, address.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) { ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s", desc->mIoHandle, patchDesc->mPatch.sinks[j].ext.device.address); outputs.add(desc->mIoHandle); break; } } } } } status_t AudioPolicyManager::checkOutputsForDevice(const sp devDesc, audio_policy_dev_state_t state, SortedVector& outputs, const String8 address) { audio_devices_t device = devDesc->mDeviceType; sp desc; // erase all current sample rates, formats and channel masks devDesc->clearCapabilities(); if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { // first 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->mProfile->mSupportedDevices.types() & device)) { if (!deviceDistinguishesOnAddress(device)) { ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); } else { ALOGV(" checking address match due to device 0x%x", device); findIoHandlesByAddress(desc, address, outputs); } } } // then look for output profiles that can be routed to this device SortedVector< sp > profiles; for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { sp profile = mHwModules[i]->mOutputProfiles[j]; if (profile->mSupportedDevices.types() & device) { if (!deviceDistinguishesOnAddress(device) || address == profile->mSupportedDevices[0]->mAddress) { profiles.add(profile); ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i); } } } } ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size()); if (profiles.isEmpty() && outputs.isEmpty()) { ALOGW("checkOutputsForDevice(): No output available for device %04x", device); 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 devDesc->importAudioPort(profile); break; } } if (j != outputs.size()) { continue; } ALOGV("opening output for device %08x with params %s profile %p", device, address.string(), profile.get()); desc = new AudioOutputDescriptor(profile); desc->mDevice = device; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = desc->mSamplingRate; config.channel_mask = desc->mChannelMask; config.format = desc->mFormat; config.offload_info.sample_rate = desc->mSamplingRate; config.offload_info.channel_mask = desc->mChannelMask; config.offload_info.format = desc->mFormat; audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; status_t status = mpClientInterface->openOutput(profile->mModule->mHandle, &output, &config, &desc->mDevice, address, &desc->mLatency, desc->mFlags); if (status == NO_ERROR) { desc->mSamplingRate = config.sample_rate; desc->mChannelMask = config.channel_mask; desc->mFormat = config.format; // Here is where the out_set_parameters() for card & device gets called if (!address.isEmpty()) { char *param = audio_device_address_to_parameter(device, address); mpClientInterface->setParameters(output, String8(param)); free(param); } // Here is where we step through and resolve any "dynamic" fields String8 reply; char *value; if (profile->mSamplingRates[0] == 0) { reply = mpClientInterface->getParameters(output, String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); ALOGV("checkOutputsForDevice() supported sampling rates %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadSamplingRates(value + 1); } } if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { reply = mpClientInterface->getParameters(output, String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); ALOGV("checkOutputsForDevice() supported formats %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadFormats(value + 1); } } if (profile->mChannelMasks[0] == 0) { reply = mpClientInterface->getParameters(output, String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); ALOGV("checkOutputsForDevice() supported channel masks %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadOutChannels(value + 1); } } if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) || ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) && (profile->mFormats.size() < 2)) || ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) { ALOGW("checkOutputsForDevice() missing param"); mpClientInterface->closeOutput(output); output = AUDIO_IO_HANDLE_NONE; } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 || profile->mChannelMasks[0] == 0) { mpClientInterface->closeOutput(output); config.sample_rate = profile->pickSamplingRate(); config.channel_mask = profile->pickChannelMask(); config.format = profile->pickFormat(); config.offload_info.sample_rate = config.sample_rate; config.offload_info.channel_mask = config.channel_mask; config.offload_info.format = config.format; status = mpClientInterface->openOutput(profile->mModule->mHandle, &output, &config, &desc->mDevice, address, &desc->mLatency, desc->mFlags); if (status == NO_ERROR) { desc->mSamplingRate = config.sample_rate; desc->mChannelMask = config.channel_mask; desc->mFormat = config.format; } else { output = AUDIO_IO_HANDLE_NONE; } } if (output != AUDIO_IO_HANDLE_NONE) { addOutput(output, desc); if (deviceDistinguishesOnAddress(device) && address != "0") { ssize_t index = mPolicyMixes.indexOfKey(address); if (index >= 0) { mPolicyMixes[index]->mOutput = desc; desc->mPolicyMix = &mPolicyMixes[index]->mMix; } else { ALOGE("checkOutputsForDevice() cannot find policy for address %s", address.string()); } } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) { // no duplicated output for direct outputs and // outputs used by dynamic policy mixes audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE; // set initial stream volume for device applyStreamVolumes(output, device, 0, true); //TODO: configure audio effect output stage here // open a duplicating output thread for the new output and the primary output duplicatedOutput = mpClientInterface->openDuplicateOutput(output, mPrimaryOutput); if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) { // add duplicated output descriptor sp dupOutputDesc = new AudioOutputDescriptor(NULL); dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); dupOutputDesc->mOutput2 = mOutputs.valueFor(output); dupOutputDesc->mSamplingRate = desc->mSamplingRate; dupOutputDesc->mFormat = desc->mFormat; dupOutputDesc->mChannelMask = desc->mChannelMask; dupOutputDesc->mLatency = desc->mLatency; addOutput(duplicatedOutput, dupOutputDesc); applyStreamVolumes(duplicatedOutput, device, 0, true); } else { ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", mPrimaryOutput, output); mpClientInterface->closeOutput(output); mOutputs.removeItem(output); nextAudioPortGeneration(); output = AUDIO_IO_HANDLE_NONE; } } } } else { output = AUDIO_IO_HANDLE_NONE; } if (output == AUDIO_IO_HANDLE_NONE) { ALOGW("checkOutputsForDevice() could not open output for device %x", device); profiles.removeAt(profile_index); profile_index--; } else { outputs.add(output); devDesc->importAudioPort(profile); if (deviceDistinguishesOnAddress(device)) { ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)", device, address.string()); setOutputDevice(output, device, true/*force*/, 0/*delay*/, NULL/*patch handle*/, address.string()); } ALOGV("checkOutputsForDevice(): adding output %d", output); } } if (profiles.isEmpty()) { ALOGW("checkOutputsForDevice(): No output available for device %04x", device); 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 (deviceDistinguishesOnAddress(device) && (desc->mProfile->mSupportedDevices.types() == device)) { findIoHandlesByAddress(desc, address, outputs); } else if (!(desc->mProfile->mSupportedDevices.types() & mAvailableOutputDevices.types())) { ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); } } } // Clear any profiles associated with the disconnected device. for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) { sp profile = mHwModules[i]->mOutputProfiles[j]; if (profile->mSupportedDevices.types() & device) { ALOGV("checkOutputsForDevice(): " "clearing direct output profile %zu on module %zu", j, i); if (profile->mSamplingRates[0] == 0) { profile->mSamplingRates.clear(); profile->mSamplingRates.add(0); } if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { profile->mFormats.clear(); profile->mFormats.add(AUDIO_FORMAT_DEFAULT); } if (profile->mChannelMasks[0] == 0) { profile->mChannelMasks.clear(); profile->mChannelMasks.add(0); } } } } } return NO_ERROR; } status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device, audio_policy_dev_state_t state, SortedVector& inputs, const String8 address) { sp desc; if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { // first list already open inputs that can be routed to this device for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { desc = mInputs.valueAt(input_index); if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index)); inputs.add(mInputs.keyAt(input_index)); } } // then look for input profiles that can be routed to this device SortedVector< sp > profiles; for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++) { if (mHwModules[module_idx]->mHandle == 0) { continue; } for (size_t profile_index = 0; profile_index < mHwModules[module_idx]->mInputProfiles.size(); profile_index++) { sp profile = mHwModules[module_idx]->mInputProfiles[profile_index]; if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { if (!deviceDistinguishesOnAddress(device) || address == profile->mSupportedDevices[0]->mAddress) { profiles.add(profile); ALOGV("checkInputsForDevice(): adding profile %zu from module %zu", profile_index, module_idx); } } } } if (profiles.isEmpty() && inputs.isEmpty()) { ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); 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) { break; } } if (input_index != mInputs.size()) { continue; } ALOGV("opening input for device 0x%X with params %s", device, address.string()); desc = new AudioInputDescriptor(profile); desc->mDevice = device; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = desc->mSamplingRate; config.channel_mask = desc->mChannelMask; config.format = desc->mFormat; audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; status_t status = mpClientInterface->openInput(profile->mModule->mHandle, &input, &config, &desc->mDevice, address, AUDIO_SOURCE_MIC, AUDIO_INPUT_FLAG_NONE /*FIXME*/); if (status == NO_ERROR) { desc->mSamplingRate = config.sample_rate; desc->mChannelMask = config.channel_mask; desc->mFormat = config.format; if (!address.isEmpty()) { char *param = audio_device_address_to_parameter(device, address); mpClientInterface->setParameters(input, String8(param)); free(param); } // Here is where we step through and resolve any "dynamic" fields String8 reply; char *value; if (profile->mSamplingRates[0] == 0) { reply = mpClientInterface->getParameters(input, String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); ALOGV("checkInputsForDevice() direct input sup sampling rates %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadSamplingRates(value + 1); } } if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { reply = mpClientInterface->getParameters(input, String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadFormats(value + 1); } } if (profile->mChannelMasks[0] == 0) { reply = mpClientInterface->getParameters(input, String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); ALOGV("checkInputsForDevice() direct input sup channel masks %s", reply.string()); value = strpbrk((char *)reply.string(), "="); if (value != NULL) { profile->loadInChannels(value + 1); } } if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) || ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) || ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) { ALOGW("checkInputsForDevice() direct input missing param"); mpClientInterface->closeInput(input); input = AUDIO_IO_HANDLE_NONE; } if (input != 0) { addInput(input, desc); } } // endif input != 0 if (input == AUDIO_IO_HANDLE_NONE) { ALOGW("checkInputsForDevice() could not open input for device 0x%X", device); profiles.removeAt(profile_index); profile_index--; } else { inputs.add(input); ALOGV("checkInputsForDevice(): adding input %d", input); } } // end scan profiles if (profiles.isEmpty()) { ALOGW("checkInputsForDevice(): No input available for device 0x%X", device); return BAD_VALUE; } } else { // Disconnect // check if one opened input is not needed any more after disconnecting one device for (size_t input_index = 0; input_index < mInputs.size(); input_index++) { desc = mInputs.valueAt(input_index); if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN)) { ALOGV("checkInputsForDevice(): disconnecting adding input %d", mInputs.keyAt(input_index)); inputs.add(mInputs.keyAt(input_index)); } } // Clear any profiles associated with the disconnected device. for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) { if (mHwModules[module_index]->mHandle == 0) { continue; } for (size_t profile_index = 0; profile_index < mHwModules[module_index]->mInputProfiles.size(); profile_index++) { sp profile = mHwModules[module_index]->mInputProfiles[profile_index]; if (profile->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) { ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu", profile_index, module_index); if (profile->mSamplingRates[0] == 0) { profile->mSamplingRates.clear(); profile->mSamplingRates.add(0); } if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { profile->mFormats.clear(); profile->mFormats.add(AUDIO_FORMAT_DEFAULT); } if (profile->mChannelMasks[0] == 0) { profile->mChannelMasks.clear(); profile->mChannelMasks.add(0); } } } } } // end disconnect return NO_ERROR; } void AudioPolicyManager::closeOutput(audio_io_handle_t output) { ALOGV("closeOutput(%d)", output); sp outputDesc = mOutputs.valueFor(output); if (outputDesc == NULL) { ALOGW("closeOutput() unknown output %d", output); return; } for (size_t i = 0; i < mPolicyMixes.size(); i++) { if (mPolicyMixes[i]->mOutput == outputDesc) { mPolicyMixes[i]->mOutput.clear(); } } // look for duplicated outputs connected to the output being removed. for (size_t i = 0; i < mOutputs.size(); i++) { sp dupOutputDesc = mOutputs.valueAt(i); if (dupOutputDesc->isDuplicated() && (dupOutputDesc->mOutput1 == outputDesc || dupOutputDesc->mOutput2 == outputDesc)) { sp outputDesc2; if (dupOutputDesc->mOutput1 == outputDesc) { outputDesc2 = dupOutputDesc->mOutput2; } else { outputDesc2 = dupOutputDesc->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. for (int j = 0; j < AUDIO_STREAM_CNT; j++) { int refCount = dupOutputDesc->mRefCount[j]; outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount); } audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); mpClientInterface->closeOutput(duplicatedOutput); mOutputs.removeItem(duplicatedOutput); } } nextAudioPortGeneration(); ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); mAudioPatches.removeItemsAt(index); mpClientInterface->onAudioPatchListUpdate(); } AudioParameter param; param.add(String8("closing"), String8("true")); mpClientInterface->setParameters(output, param.toString()); mpClientInterface->closeOutput(output); mOutputs.removeItem(output); mPreviousOutputs = mOutputs; } 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(); ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); mAudioPatches.removeItemsAt(index); mpClientInterface->onAudioPatchListUpdate(); } mpClientInterface->closeInput(input); mInputs.removeItem(input); } SortedVector AudioPolicyManager::getOutputsForDevice(audio_devices_t device, DefaultKeyedVector > openOutputs) { SortedVector outputs; ALOGVV("getOutputsForDevice() device %04x", device); for (size_t i = 0; i < openOutputs.size(); i++) { ALOGVV("output %d isDuplicated=%d device=%04x", i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices()); if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) { ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i)); outputs.add(openOutputs.keyAt(i)); } } return outputs; } bool AudioPolicyManager::vectorsEqual(SortedVector& outputs1, SortedVector& outputs2) { if (outputs1.size() != outputs2.size()) { return false; } for (size_t i = 0; i < outputs1.size(); i++) { if (outputs1[i] != outputs2[i]) { return false; } } return true; } void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) { audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/); audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/); SortedVector srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); SortedVector dstOutputs = getOutputsForDevice(newDevice, mOutputs); if (!vectorsEqual(srcOutputs,dstOutputs)) { ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d", strategy, srcOutputs[0], dstOutputs[0]); // mute strategy while moving tracks from one output to another for (size_t i = 0; i < srcOutputs.size(); i++) { sp desc = mOutputs.valueFor(srcOutputs[i]); if (desc->isStrategyActive(strategy)) { setStrategyMute(strategy, true, srcOutputs[i]); setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); } } // Move effects associated to this strategy from previous output to new output if (strategy == STRATEGY_MEDIA) { audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs); SortedVector moved; for (size_t i = 0; i < mEffects.size(); i++) { sp effectDesc = mEffects.valueAt(i); if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX && effectDesc->mIo != fxOutput) { if (moved.indexOf(effectDesc->mIo) < 0) { ALOGV("checkOutputForStrategy() moving effect %d to output %d", mEffects.keyAt(i), fxOutput); mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo, fxOutput); moved.add(effectDesc->mIo); } effectDesc->mIo = fxOutput; } } } // Move tracks associated to this strategy from previous output to new output for (int i = 0; i < AUDIO_STREAM_CNT; i++) { if (i == AUDIO_STREAM_PATCH) { continue; } if (getStrategy((audio_stream_type_t)i) == strategy) { mpClientInterface->invalidateStream((audio_stream_type_t)i); } } } } void AudioPolicyManager::checkOutputForAllStrategies() { if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); checkOutputForStrategy(STRATEGY_PHONE); if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); checkOutputForStrategy(STRATEGY_SONIFICATION); checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); checkOutputForStrategy(STRATEGY_ACCESSIBILITY); checkOutputForStrategy(STRATEGY_MEDIA); checkOutputForStrategy(STRATEGY_DTMF); checkOutputForStrategy(STRATEGY_REROUTING); } audio_io_handle_t AudioPolicyManager::getA2dpOutput() { for (size_t i = 0; i < mOutputs.size(); i++) { sp outputDesc = mOutputs.valueAt(i); if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { return mOutputs.keyAt(i); } } return 0; } void AudioPolicyManager::checkA2dpSuspend() { audio_io_handle_t a2dpOutput = getA2dpOutput(); if (a2dpOutput == 0) { mA2dpSuspended = false; return; } bool isScoConnected = ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET & ~AUDIO_DEVICE_BIT_IN) != 0) || ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0); // suspend A2DP output if: // (NOT already suspended) && // ((SCO device is connected && // (forced usage for communication || for record is SCO))) || // (phone state is ringing || in call) // // restore A2DP output if: // (Already suspended) && // ((SCO device is NOT connected || // (forced usage NOT for communication && NOT for record is SCO))) && // (phone state is NOT ringing && NOT in call) // if (mA2dpSuspended) { if ((!isScoConnected || ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) && (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) && ((mPhoneState != AUDIO_MODE_IN_CALL) && (mPhoneState != AUDIO_MODE_RINGTONE))) { mpClientInterface->restoreOutput(a2dpOutput); mA2dpSuspended = false; } } else { if ((isScoConnected && ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) || ((mPhoneState == AUDIO_MODE_IN_CALL) || (mPhoneState == AUDIO_MODE_RINGTONE))) { mpClientInterface->suspendOutput(a2dpOutput); mA2dpSuspended = true; } } } audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache) { audio_devices_t device = AUDIO_DEVICE_NONE; sp outputDesc = mOutputs.valueFor(output); ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); if (patchDesc->mUid != mUidCached) { ALOGV("getNewOutputDevice() device %08x forced by patch %d", outputDesc->device(), outputDesc->mPatchHandle); return outputDesc->device(); } } // check the following by order of priority to request a routing change if necessary: // 1: the strategy enforced audible is active and enforced on the output: // use device for strategy enforced audible // 2: we are in call or the strategy phone is active on the output: // use device for strategy phone // 3: the strategy for enforced audible is active but not enforced on the output: // use the device for strategy enforced audible // 4: the strategy sonification is active on the output: // use device for strategy sonification // 5: the strategy "respectful" sonification is active on the output: // use device for strategy "respectful" sonification // 6: the strategy accessibility is active on the output: // use device for strategy accessibility // 7: the strategy media is active on the output: // use device for strategy media // 8: the strategy DTMF is active on the output: // use device for strategy DTMF // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output: // use device for strategy t-t-s if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE) && mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); } else if (isInCall() || outputDesc->isStrategyActive(STRATEGY_PHONE)) { device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); } else if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) { device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) { device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) { device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); } else if (outputDesc->isStrategyActive(STRATEGY_ACCESSIBILITY)) { device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache); } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) { device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) { device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); } else if (outputDesc->isStrategyActive(STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) { device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache); } else if (outputDesc->isStrategyActive(STRATEGY_REROUTING)) { device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache); } ALOGV("getNewOutputDevice() selected device %x", device); return device; } audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input) { sp inputDesc = mInputs.valueFor(input); ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); if (index >= 0) { sp patchDesc = mAudioPatches.valueAt(index); if (patchDesc->mUid != mUidCached) { ALOGV("getNewInputDevice() device %08x forced by patch %d", inputDesc->mDevice, inputDesc->mPatchHandle); return inputDesc->mDevice; } } audio_devices_t device = getDeviceForInputSource(inputDesc->mInputSource); ALOGV("getNewInputDevice() selected device %x", device); return device; } uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) { return (uint32_t)getStrategy(stream); } audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) { // By checking the range of stream before calling getStrategy, we avoid // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE // and then return STRATEGY_MEDIA, but we want to return the empty set. if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) { return AUDIO_DEVICE_NONE; } audio_devices_t devices; AudioPolicyManager::routing_strategy strategy = getStrategy(stream); devices = getDeviceForStrategy(strategy, true /*fromCache*/); SortedVector outputs = getOutputsForDevice(devices, mOutputs); for (size_t i = 0; i < outputs.size(); i++) { sp outputDesc = mOutputs.valueFor(outputs[i]); if (outputDesc->isStrategyActive(strategy)) { devices = outputDesc->device(); break; } } /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it and doesn't really need to.*/ if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) { devices |= AUDIO_DEVICE_OUT_SPEAKER; devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE; } return devices; } AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy( audio_stream_type_t stream) { ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH"); // stream to strategy mapping switch (stream) { case AUDIO_STREAM_VOICE_CALL: case AUDIO_STREAM_BLUETOOTH_SCO: return STRATEGY_PHONE; case AUDIO_STREAM_RING: case AUDIO_STREAM_ALARM: return STRATEGY_SONIFICATION; case AUDIO_STREAM_NOTIFICATION: return STRATEGY_SONIFICATION_RESPECTFUL; case AUDIO_STREAM_DTMF: return STRATEGY_DTMF; default: ALOGE("unknown stream type %d", stream); case AUDIO_STREAM_SYSTEM: // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs // while key clicks are played produces a poor result case AUDIO_STREAM_MUSIC: return STRATEGY_MEDIA; case AUDIO_STREAM_ENFORCED_AUDIBLE: return STRATEGY_ENFORCED_AUDIBLE; case AUDIO_STREAM_TTS: return STRATEGY_TRANSMITTED_THROUGH_SPEAKER; case AUDIO_STREAM_ACCESSIBILITY: return STRATEGY_ACCESSIBILITY; case AUDIO_STREAM_REROUTING: return STRATEGY_REROUTING; } } uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) { // flags to strategy mapping if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER; } if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { return (uint32_t) STRATEGY_ENFORCED_AUDIBLE; } // usage to strategy mapping switch (attr->usage) { case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: if (isStreamActive(AUDIO_STREAM_RING) || isStreamActive(AUDIO_STREAM_ALARM)) { return (uint32_t) STRATEGY_SONIFICATION; } if (isInCall()) { return (uint32_t) STRATEGY_PHONE; } return (uint32_t) STRATEGY_ACCESSIBILITY; case AUDIO_USAGE_MEDIA: case AUDIO_USAGE_GAME: case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: case AUDIO_USAGE_ASSISTANCE_SONIFICATION: return (uint32_t) STRATEGY_MEDIA; case AUDIO_USAGE_VOICE_COMMUNICATION: return (uint32_t) STRATEGY_PHONE; case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: return (uint32_t) STRATEGY_DTMF; case AUDIO_USAGE_ALARM: case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: return (uint32_t) STRATEGY_SONIFICATION; case AUDIO_USAGE_NOTIFICATION: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: case AUDIO_USAGE_NOTIFICATION_EVENT: return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL; case AUDIO_USAGE_UNKNOWN: default: return (uint32_t) STRATEGY_MEDIA; } } void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) { switch(stream) { case AUDIO_STREAM_MUSIC: checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); updateDevicesAndOutputs(); break; default: break; } } bool AudioPolicyManager::isAnyOutputActive(audio_stream_type_t streamToIgnore) { for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) { if (s == (size_t) streamToIgnore) { continue; } for (size_t i = 0; i < mOutputs.size(); i++) { const sp outputDesc = mOutputs.valueAt(i); if (outputDesc->mRefCount[s] != 0) { return true; } } } return false; } uint32_t AudioPolicyManager::handleEventForBeacon(int event) { 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; for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); setStreamMute(AUDIO_STREAM_TTS, mute/*on*/, desc->mIoHandle, 0 /*delay*/, AUDIO_DEVICE_NONE); const uint32_t latency = desc->latency() * 2; if (latency > maxLatency) { maxLatency = latency; } } mBeaconMuted = mute; return maxLatency; } return 0; } audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy, bool fromCache) { uint32_t device = AUDIO_DEVICE_NONE; if (fromCache) { ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]); return mDeviceForStrategy[strategy]; } audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types(); switch (strategy) { case STRATEGY_TRANSMITTED_THROUGH_SPEAKER: device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; if (!device) { ALOGE("getDeviceForStrategy() no device found for "\ "STRATEGY_TRANSMITTED_THROUGH_SPEAKER"); } break; case STRATEGY_SONIFICATION_RESPECTFUL: if (isInCall()) { device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { // while media is playing on a remote device, use the the sonification behavior. // Note that we test this usecase before testing if media is playing because // the isStreamActive() method only informs about the activity of a stream, not // if it's for local playback. Note also that we use the same delay between both tests device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); //user "safe" speaker if available instead of normal speaker to avoid triggering //other acoustic safety mechanisms for notification if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { // while media is playing (or has recently played), use the same device device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); } else { // when media is not playing anymore, fall back on the sonification behavior device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); //user "safe" speaker if available instead of normal speaker to avoid triggering //other acoustic safety mechanisms for notification if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; } break; case STRATEGY_DTMF: if (!isInCall()) { // when off call, DTMF strategy follows the same rules as MEDIA strategy device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); break; } // when in call, DTMF and PHONE strategies follow the same rules // FALL THROUGH case STRATEGY_PHONE: // Force use of only devices on primary output if: // - in call AND // - cannot route from voice call RX OR // - audio HAL version is < 3.0 and TX device is on the primary HW module if (mPhoneState == AUDIO_MODE_IN_CALL) { audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); sp hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); if (((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) || (((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) && (hwOutputDesc->getAudioPort()->mModule->mHalVersion < AUDIO_DEVICE_API_VERSION_3_0))) { availableOutputDeviceTypes = availablePrimaryOutputDevices(); } } // for phone strategy, we first consider the forced use and then the available devices by order // of priority switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { case AUDIO_POLICY_FORCE_BT_SCO: if (!isInCall() || strategy != STRATEGY_DTMF) { device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; if (device) break; } device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO; if (device) break; // if SCO device is requested but no SCO device is available, fall back to default case // FALL THROUGH default: // FORCE_NONE // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP if (!isInCall() && (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && (getA2dpOutput() != 0)) { device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; if (device) break; } device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; if (device) break; if (mPhoneState != AUDIO_MODE_IN_CALL) { device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; if (device) break; } device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE; if (device) break; device = mDefaultOutputDevice->mDeviceType; if (device == AUDIO_DEVICE_NONE) { ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); } break; case AUDIO_POLICY_FORCE_SPEAKER: // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to // A2DP speaker when forcing to speaker output if (!isInCall() && (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && (getA2dpOutput() != 0)) { device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; if (device) break; } if (mPhoneState != AUDIO_MODE_IN_CALL) { device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; if (device) break; } device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; if (device) break; device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; if (device) break; device = mDefaultOutputDevice->mDeviceType; if (device == AUDIO_DEVICE_NONE) { ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); } break; } break; case STRATEGY_SONIFICATION: // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by // handleIncallSonification(). if (isInCall()) { device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); break; } // FALL THROUGH case STRATEGY_ENFORCED_AUDIBLE: // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION // except: // - when in call where it doesn't default to STRATEGY_PHONE behavior // - in countries where not enforced in which case it follows STRATEGY_MEDIA if ((strategy == STRATEGY_SONIFICATION) || (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) { device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; if (device == AUDIO_DEVICE_NONE) { ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); } } // The second device used for sonification is the same as the device used by media strategy // FALL THROUGH // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now case STRATEGY_ACCESSIBILITY: if (strategy == STRATEGY_ACCESSIBILITY) { // do not route accessibility prompts to a digital output currently configured with a // compressed format as they would likely not be mixed and dropped. for (size_t i = 0; i < mOutputs.size(); i++) { sp desc = mOutputs.valueAt(i); audio_devices_t devices = desc->device() & (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC); if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) && devices != AUDIO_DEVICE_NONE) { availableOutputDeviceTypes = availableOutputDeviceTypes & ~devices; } } } // FALL THROUGH case STRATEGY_REROUTING: case STRATEGY_MEDIA: { uint32_t device2 = AUDIO_DEVICE_NONE; if (strategy != STRATEGY_SONIFICATION) { // no sonification on remote submix (e.g. WFD) if (mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; } } if ((device2 == AUDIO_DEVICE_NONE) && (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && (getA2dpOutput() != 0)) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; if (device2 == AUDIO_DEVICE_NONE) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; } if (device2 == AUDIO_DEVICE_NONE) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; } } if ((device2 == AUDIO_DEVICE_NONE) && (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; } if (device2 == AUDIO_DEVICE_NONE) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; } if ((device2 == AUDIO_DEVICE_NONE)) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; } if (device2 == AUDIO_DEVICE_NONE) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; } if (device2 == AUDIO_DEVICE_NONE) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; } if (device2 == AUDIO_DEVICE_NONE) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; } if (device2 == AUDIO_DEVICE_NONE) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; } if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { // no sonification on aux digital (e.g. HDMI) device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; } if ((device2 == AUDIO_DEVICE_NONE) && (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; } if (device2 == AUDIO_DEVICE_NONE) { device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; } int device3 = AUDIO_DEVICE_NONE; if (strategy == STRATEGY_MEDIA) { // ARC, SPDIF and AUX_LINE can co-exist with others. device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC; device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF); device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE); } device2 |= device3; // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise device |= device2; // If hdmi system audio mode is on, remove speaker out of output list. if ((strategy == STRATEGY_MEDIA) && (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] == AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) { device &= ~AUDIO_DEVICE_OUT_SPEAKER; } if (device) break; device = mDefaultOutputDevice->mDeviceType; if (device == AUDIO_DEVICE_NONE) { ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); } } break; default: ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); break; } ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); return device; } void AudioPolicyManager::updateDevicesAndOutputs() { for (int i = 0; i < NUM_STRATEGIES; i++) { mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); } mPreviousOutputs = mOutputs; } uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp outputDesc, audio_devices_t prevDevice, 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; audio_devices_t device = outputDesc->device(); bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2); for (size_t i = 0; i < NUM_STRATEGIES; i++) { audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types(); bool mute = shouldMute && (curDevice & device) && (curDevice != device); bool doMute = false; if (mute && !outputDesc->mStrategyMutedByDevice[i]) { doMute = true; outputDesc->mStrategyMutedByDevice[i] = true; } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){ doMute = true; outputDesc->mStrategyMutedByDevice[i] = 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() & outputDesc->supportedDevices()) == AUDIO_DEVICE_NONE) { continue; } audio_io_handle_t curOutput = mOutputs.keyAt(j); ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", mute ? "muting" : "unmuting", i, curDevice, curOutput); setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs); if (desc->isStrategyActive((routing_strategy)i)) { 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() && (device != prevDevice)) { if (muteWaitMs < outputDesc->latency() * 2) { muteWaitMs = outputDesc->latency() * 2; } for (size_t i = 0; i < NUM_STRATEGIES; i++) { if (outputDesc->isStrategyActive((routing_strategy)i)) { setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle); // do tempMute unmute after twice the mute wait time setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle, muteWaitMs *2, device); } } } // 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::setOutputDevice(audio_io_handle_t output, audio_devices_t device, bool force, int delayMs, audio_patch_handle_t *patchHandle, const char* address) { ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); sp outputDesc = mOutputs.valueFor(output); AudioParameter param; uint32_t muteWaitMs; if (outputDesc->isDuplicated()) { muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs); muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs); return muteWaitMs; } // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current // output profile if ((device != AUDIO_DEVICE_NONE) && ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) { return 0; } // filter devices according to output selected device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types()); audio_devices_t prevDevice = outputDesc->mDevice; ALOGV("setOutputDevice() prevDevice %04x", prevDevice); if (device != AUDIO_DEVICE_NONE) { outputDesc->mDevice = device; } muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs); // 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 setOutputDevice() without conditions if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force && outputDesc->mPatchHandle != 0) { ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output); return muteWaitMs; } ALOGV("setOutputDevice() changing device"); // do the routing if (device == AUDIO_DEVICE_NONE) { resetOutputDevice(output, delayMs, NULL); } else { DeviceVector deviceList = (address == NULL) ? mAvailableOutputDevices.getDevicesFromType(device) : mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address)); if (!deviceList.isEmpty()) { struct audio_patch patch; outputDesc->toAudioPortConfig(&patch.sources[0]); patch.num_sources = 1; patch.num_sinks = 0; for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) { deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]); patch.num_sinks++; } ssize_t index; if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); } sp< AudioPatch> patchDesc; audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; if (index >= 0) { patchDesc = mAudioPatches.valueAt(index); afPatchHandle = patchDesc->mAfPatchHandle; } status_t status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, delayMs); ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d" "num_sources %d num_sinks %d", status, afPatchHandle, patch.num_sources, patch.num_sinks); if (status == NO_ERROR) { if (index < 0) { patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), &patch, mUidCached); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = patch; } patchDesc->mAfPatchHandle = afPatchHandle; patchDesc->mUid = mUidCached; if (patchHandle) { *patchHandle = patchDesc->mHandle; } outputDesc->mPatchHandle = patchDesc->mHandle; nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } } // inform all input as well for (size_t i = 0; i < mInputs.size(); i++) { const sp inputDescriptor = mInputs.valueAt(i); if (!isVirtualInputDevice(inputDescriptor->mDevice)) { AudioParameter inputCmd = AudioParameter(); ALOGV("%s: inform input %d of device:%d", __func__, inputDescriptor->mIoHandle, device); inputCmd.addInt(String8(AudioParameter::keyRouting),device); mpClientInterface->setParameters(inputDescriptor->mIoHandle, inputCmd.toString(), delayMs); } } } // update stream volumes according to new device applyStreamVolumes(output, device, delayMs); return muteWaitMs; } status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output, int delayMs, audio_patch_handle_t *patchHandle) { sp outputDesc = mOutputs.valueFor(output); ssize_t index; if (patchHandle) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); } if (index < 0) { return INVALID_OPERATION; } sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs); ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status); outputDesc->mPatchHandle = 0; removeAudioPatch(patchDesc->mHandle); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); return status; } status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input, audio_devices_t device, bool force, audio_patch_handle_t *patchHandle) { status_t status = NO_ERROR; sp inputDesc = mInputs.valueFor(input); if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) { inputDesc->mDevice = device; DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device); if (!deviceList.isEmpty()) { struct audio_patch patch; inputDesc->toAudioPortConfig(&patch.sinks[0]); // AUDIO_SOURCE_HOTWORD is for internal use only: // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD && !inputDesc->mIsSoundTrigger) { patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION; } patch.num_sinks = 1; //only one input device for now deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]); patch.num_sources = 1; ssize_t index; if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) { index = mAudioPatches.indexOfKey(*patchHandle); } else { index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); } sp< AudioPatch> patchDesc; audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE; if (index >= 0) { patchDesc = mAudioPatches.valueAt(index); afPatchHandle = patchDesc->mAfPatchHandle; } status_t status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0); ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d", status, afPatchHandle); if (status == NO_ERROR) { if (index < 0) { patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), &patch, mUidCached); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = patch; } patchDesc->mAfPatchHandle = afPatchHandle; patchDesc->mUid = mUidCached; if (patchHandle) { *patchHandle = patchDesc->mHandle; } inputDesc->mPatchHandle = patchDesc->mHandle; nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); } } } 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->mPatchHandle); } if (index < 0) { return INVALID_OPERATION; } sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index); status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); ALOGV("resetInputDevice() releaseAudioPatch returned %d", status); inputDesc->mPatchHandle = 0; removeAudioPatch(patchDesc->mHandle); nextAudioPortGeneration(); mpClientInterface->onAudioPatchListUpdate(); return status; } sp AudioPolicyManager::getInputProfile(audio_devices_t device, String8 address, 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. for (size_t i = 0; i < mHwModules.size(); i++) { if (mHwModules[i]->mHandle == 0) { continue; } for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { sp profile = mHwModules[i]->mInputProfiles[j]; // profile->log(); if (profile->isCompatibleProfile(device, address, samplingRate, &samplingRate /*updatedSamplingRate*/, format, channelMask, (audio_output_flags_t) flags)) { return profile; } } } return NULL; } audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource, AudioMix **policyMix) { uint32_t device = AUDIO_DEVICE_NONE; audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; for (size_t i = 0; i < mPolicyMixes.size(); i++) { if (mPolicyMixes[i]->mMix.mMixType != MIX_TYPE_RECORDERS) { continue; } for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) { if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule && mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource == inputSource) || (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule && mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource != inputSource)) { if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { if (policyMix != NULL) { *policyMix = &mPolicyMixes[i]->mMix; } return AUDIO_DEVICE_IN_REMOTE_SUBMIX; } break; } } } switch (inputSource) { case AUDIO_SOURCE_VOICE_UPLINK: if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { device = AUDIO_DEVICE_IN_VOICE_CALL; break; } break; case AUDIO_SOURCE_DEFAULT: case AUDIO_SOURCE_MIC: if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) { device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { device = AUDIO_DEVICE_IN_WIRED_HEADSET; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { device = AUDIO_DEVICE_IN_USB_DEVICE; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { device = AUDIO_DEVICE_IN_BUILTIN_MIC; } break; case AUDIO_SOURCE_VOICE_COMMUNICATION: // Allow only use of devices on primary input if in call and HAL does not support routing // to voice call path. if ((mPhoneState == AUDIO_MODE_IN_CALL) && (mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) { availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN; } switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { case AUDIO_POLICY_FORCE_BT_SCO: // if SCO device is requested but no SCO device is available, fall back to default case if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; break; } // FALL THROUGH default: // FORCE_NONE if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { device = AUDIO_DEVICE_IN_WIRED_HEADSET; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { device = AUDIO_DEVICE_IN_USB_DEVICE; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { device = AUDIO_DEVICE_IN_BUILTIN_MIC; } break; case AUDIO_POLICY_FORCE_SPEAKER: if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { device = AUDIO_DEVICE_IN_BACK_MIC; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { device = AUDIO_DEVICE_IN_BUILTIN_MIC; } break; } break; case AUDIO_SOURCE_VOICE_RECOGNITION: case AUDIO_SOURCE_HOTWORD: if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO && availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { device = AUDIO_DEVICE_IN_WIRED_HEADSET; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { device = AUDIO_DEVICE_IN_USB_DEVICE; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { device = AUDIO_DEVICE_IN_BUILTIN_MIC; } break; case AUDIO_SOURCE_CAMCORDER: if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { device = AUDIO_DEVICE_IN_BACK_MIC; } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { device = AUDIO_DEVICE_IN_BUILTIN_MIC; } break; case AUDIO_SOURCE_VOICE_DOWNLINK: case AUDIO_SOURCE_VOICE_CALL: if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { device = AUDIO_DEVICE_IN_VOICE_CALL; } break; case AUDIO_SOURCE_REMOTE_SUBMIX: if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; } break; case AUDIO_SOURCE_FM_TUNER: if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) { device = AUDIO_DEVICE_IN_FM_TUNER; } break; default: ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); break; } ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); return device; } bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device) { if ((device & AUDIO_DEVICE_BIT_IN) != 0) { device &= ~AUDIO_DEVICE_BIT_IN; if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) return true; } return false; } bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) { return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0); } audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs) { for (size_t i = 0; i < mInputs.size(); i++) { const sp input_descriptor = mInputs.valueAt(i); if ((input_descriptor->mRefCount > 0) && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) { return mInputs.keyAt(i); } } return 0; } uint32_t AudioPolicyManager::activeInputsCount() const { uint32_t count = 0; for (size_t i = 0; i < mInputs.size(); i++) { const sp desc = mInputs.valueAt(i); if (desc->mRefCount > 0) { return count++; } } return count; } audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device) { if (device == AUDIO_DEVICE_NONE) { // this happens when forcing a route update and no track is active on an output. // In this case the returned category is not important. device = AUDIO_DEVICE_OUT_SPEAKER; } else if (popcount(device) > 1) { // Multiple device selection is either: // - speaker + one other device: give priority to speaker in this case. // - one A2DP device + another device: happens with duplicated output. In this case // retain the device on the A2DP output as the other must not correspond to an active // selection if not the speaker. // - HDMI-CEC system audio mode only output: give priority to available item in order. if (device & AUDIO_DEVICE_OUT_SPEAKER) { device = AUDIO_DEVICE_OUT_SPEAKER; } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) { device = AUDIO_DEVICE_OUT_HDMI_ARC; } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) { device = AUDIO_DEVICE_OUT_AUX_LINE; } else if (device & AUDIO_DEVICE_OUT_SPDIF) { device = AUDIO_DEVICE_OUT_SPDIF; } else { device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP); } } /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/ if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE) device = AUDIO_DEVICE_OUT_SPEAKER; ALOGW_IF(popcount(device) != 1, "getDeviceForVolume() invalid device combination: %08x", device); return device; } AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device) { switch(getDeviceForVolume(device)) { case AUDIO_DEVICE_OUT_EARPIECE: return DEVICE_CATEGORY_EARPIECE; case AUDIO_DEVICE_OUT_WIRED_HEADSET: case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: case AUDIO_DEVICE_OUT_BLUETOOTH_SCO: case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET: case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP: case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES: return DEVICE_CATEGORY_HEADSET; case AUDIO_DEVICE_OUT_LINE: case AUDIO_DEVICE_OUT_AUX_DIGITAL: /*USB? Remote submix?*/ return DEVICE_CATEGORY_EXT_MEDIA; case AUDIO_DEVICE_OUT_SPEAKER: case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: case AUDIO_DEVICE_OUT_USB_ACCESSORY: case AUDIO_DEVICE_OUT_USB_DEVICE: case AUDIO_DEVICE_OUT_REMOTE_SUBMIX: default: return DEVICE_CATEGORY_SPEAKER; } } /* static */ float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, int indexInUi) { device_category deviceCategory = getDeviceCategory(device); const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory]; // the volume index in the UI is relative to the min and max volume indices for this stream type int nbSteps = 1 + curve[VOLMAX].mIndex - curve[VOLMIN].mIndex; int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin); // find what part of the curve this index volume belongs to, or if it's out of bounds int segment = 0; if (volIdx < curve[VOLMIN].mIndex) { // out of bounds return 0.0f; } else if (volIdx < curve[VOLKNEE1].mIndex) { segment = 0; } else if (volIdx < curve[VOLKNEE2].mIndex) { segment = 1; } else if (volIdx <= curve[VOLMAX].mIndex) { segment = 2; } else { // out of bounds return 1.0f; } // linear interpolation in the attenuation table in dB float decibels = curve[segment].mDBAttenuation + ((float)(volIdx - curve[segment].mIndex)) * ( (curve[segment+1].mDBAttenuation - curve[segment].mDBAttenuation) / ((float)(curve[segment+1].mIndex - curve[segment].mIndex)) ); float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", curve[segment].mIndex, volIdx, curve[segment+1].mIndex, curve[segment].mDBAttenuation, decibels, curve[segment+1].mDBAttenuation, amplification); return amplification; } const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = { {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = { {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT] = { {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = { {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT] = { {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = { {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = { {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f} }; // AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks // AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets. // AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java). // The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset. const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = { {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = { {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = { {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = { {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = { {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sLinearVolumeCurve[AudioPolicyManager::VOLCNT] = { {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sSilentVolumeCurve[AudioPolicyManager::VOLCNT] = { {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f} }; const AudioPolicyManager::VolumeCurvePoint AudioPolicyManager::sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT] = { {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f} }; const AudioPolicyManager::VolumeCurvePoint *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT] [AudioPolicyManager::DEVICE_CATEGORY_CNT] = { { // AUDIO_STREAM_VOICE_CALL sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_SYSTEM sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_RING sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_MUSIC sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_ALARM sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_NOTIFICATION sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_BLUETOOTH_SCO sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_ENFORCED_AUDIBLE sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_DTMF sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_TTS // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_ACCESSIBILITY sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_REROUTING sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, { // AUDIO_STREAM_PATCH sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA }, }; void AudioPolicyManager::initializeVolumeCurves() { for (int i = 0; i < AUDIO_STREAM_CNT; i++) { for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { mStreams[i].mVolumeCurve[j] = sVolumeProfiles[i][j]; } } // Check availability of DRC on speaker path: if available, override some of the speaker curves if (mSpeakerDrcEnabled) { mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sDefaultSystemVolumeCurveDrc; mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sSpeakerSonificationVolumeCurveDrc; mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sSpeakerSonificationVolumeCurveDrc; mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sSpeakerSonificationVolumeCurveDrc; mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sSpeakerMediaVolumeCurveDrc; mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sSpeakerMediaVolumeCurveDrc; } } float AudioPolicyManager::computeVolume(audio_stream_type_t stream, int index, audio_io_handle_t output, audio_devices_t device) { float volume = 1.0; sp outputDesc = mOutputs.valueFor(output); StreamDescriptor &streamDesc = mStreams[stream]; if (device == AUDIO_DEVICE_NONE) { device = outputDesc->device(); } volume = volIndexToAmpl(device, streamDesc, index); // 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 ring tones and notifications volume by 6dB // - if music is playing, always limit the volume to current music volume, // with a minimum threshold at -36dB so that notification is always perceived. const routing_strategy stream_strategy = getStrategy(stream); if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) && ((stream_strategy == STRATEGY_SONIFICATION) || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) || (stream == AUDIO_STREAM_SYSTEM) || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) && streamDesc.mCanBeMuted) { volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; // 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) { audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); float musicVol = computeVolume(AUDIO_STREAM_MUSIC, mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice), output, musicDevice); float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN; if (volume > minVol) { volume = minVol; ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); } } } return volume; } status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs, bool force) { // do not change actual stream volume if the stream is muted if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { ALOGVV("checkAndSetVolume() stream %d muted count %d", stream, mOutputs.valueFor(output)->mMuteCount[stream]); return NO_ERROR; } // do not change in call volume if bluetooth is connected and vice versa if ((stream == AUDIO_STREAM_VOICE_CALL && mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || (stream == AUDIO_STREAM_BLUETOOTH_SCO && mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) { ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); return INVALID_OPERATION; } float volume = computeVolume(stream, index, output, device); // unit gain if rerouting to external policy if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { ssize_t index = mOutputs.indexOfKey(output); if (index >= 0) { sp outputDesc = mOutputs.valueAt(index); if (outputDesc->mPolicyMix != NULL) { ALOGV("max gain when rerouting for output=%d", output); volume = 1.0f; } } } // We actually change the volume if: // - the float value returned by computeVolume() changed // - the force flag is set if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || force) { mOutputs.valueFor(output)->mCurVolume[stream] = volume; ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is // enabled if (stream == AUDIO_STREAM_BLUETOOTH_SCO) { mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs); } mpClientInterface->setStreamVolume(stream, volume, output, delayMs); } if (stream == AUDIO_STREAM_VOICE_CALL || stream == AUDIO_STREAM_BLUETOOTH_SCO) { float voiceVolume; // Force voice volume to max for bluetooth SCO as volume is managed by the headset if (stream == AUDIO_STREAM_VOICE_CALL) { voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; } else { voiceVolume = 1.0; } if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) { mpClientInterface->setVoiceVolume(voiceVolume, delayMs); mLastVoiceVolume = voiceVolume; } } return NO_ERROR; } void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs, bool force) { ALOGVV("applyStreamVolumes() for output %d and device %x", output, device); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } checkAndSetVolume((audio_stream_type_t)stream, mStreams[stream].getVolumeIndex(device), output, device, delayMs, force); } } void AudioPolicyManager::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs, audio_devices_t device) { ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } if (getStrategy((audio_stream_type_t)stream) == strategy) { setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device); } } } void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, bool on, audio_io_handle_t output, int delayMs, audio_devices_t device) { StreamDescriptor &streamDesc = mStreams[stream]; sp outputDesc = mOutputs.valueFor(output); if (device == AUDIO_DEVICE_NONE) { device = outputDesc->device(); } ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", stream, on, output, outputDesc->mMuteCount[stream], device); if (on) { if (outputDesc->mMuteCount[stream] == 0) { if (streamDesc.mCanBeMuted && ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) || (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) { checkAndSetVolume(stream, 0, output, device, delayMs); } } // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored outputDesc->mMuteCount[stream]++; } else { if (outputDesc->mMuteCount[stream] == 0) { ALOGV("setStreamMute() unmuting non muted stream!"); return; } if (--outputDesc->mMuteCount[stream] == 0) { checkAndSetVolume(stream, streamDesc.getVolumeIndex(device), output, device, delayMs); } } } void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange) { // if the stream pertains to sonification strategy and we are in call we must // mute the stream if it is low visibility. If it is high visibility, we must play a tone // in the device used for phone strategy and play the tone if the selected device does not // interfere with the device used for phone strategy // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as // many times as there are active tracks on the output const routing_strategy stream_strategy = getStrategy(stream); if ((stream_strategy == STRATEGY_SONIFICATION) || ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { sp outputDesc = mOutputs.valueFor(mPrimaryOutput); ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", stream, starting, outputDesc->mDevice, stateChange); if (outputDesc->mRefCount[stream]) { int muteCount = 1; if (stateChange) { muteCount = outputDesc->mRefCount[stream]; } if (audio_is_low_visibility(stream)) { ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount); for (int i = 0; i < muteCount; i++) { setStreamMute(stream, starting, mPrimaryOutput); } } else { ALOGV("handleIncallSonification() high visibility"); if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) { ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount); for (int i = 0; i < muteCount; i++) { setStreamMute(stream, starting, mPrimaryOutput); } } if (starting) { mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION, AUDIO_STREAM_VOICE_CALL); } else { mpClientInterface->stopTone(); } } } } } bool AudioPolicyManager::isInCall() { return isStateInCall(mPhoneState); } bool AudioPolicyManager::isStateInCall(int state) { return ((state == AUDIO_MODE_IN_CALL) || (state == AUDIO_MODE_IN_COMMUNICATION)); } uint32_t AudioPolicyManager::getMaxEffectsCpuLoad() { return MAX_EFFECTS_CPU_LOAD; } uint32_t AudioPolicyManager::getMaxEffectsMemory() { return MAX_EFFECTS_MEMORY; } // --- AudioOutputDescriptor class implementation AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor( const sp& profile) : mId(0), mIoHandle(0), mLatency(0), mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0) { // clear usage count for all stream types for (int i = 0; i < AUDIO_STREAM_CNT; i++) { mRefCount[i] = 0; mCurVolume[i] = -1.0; mMuteCount[i] = 0; mStopTime[i] = 0; } for (int i = 0; i < NUM_STRATEGIES; i++) { mStrategyMutedByDevice[i] = false; } if (profile != NULL) { mFlags = (audio_output_flags_t)profile->mFlags; mSamplingRate = profile->pickSamplingRate(); mFormat = profile->pickFormat(); mChannelMask = profile->pickChannelMask(); if (profile->mGains.size() > 0) { profile->mGains[0]->getDefaultConfig(&mGain); } } } audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const { if (isDuplicated()) { return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); } else { return mDevice; } } uint32_t AudioPolicyManager::AudioOutputDescriptor::latency() { if (isDuplicated()) { return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; } else { return mLatency; } } bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith( const sp outputDesc) { if (isDuplicated()) { return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); } else if (outputDesc->isDuplicated()){ return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2); } else { return (mProfile->mModule == outputDesc->mProfile->mModule); } } void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream, int delta) { // forward usage count change to attached outputs if (isDuplicated()) { mOutput1->changeRefCount(stream, delta); mOutput2->changeRefCount(stream, delta); } if ((delta + (int)mRefCount[stream]) < 0) { ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]); mRefCount[stream] = 0; return; } mRefCount[stream] += delta; ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); } audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices() { if (isDuplicated()) { return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); } else { return mProfile->mSupportedDevices.types() ; } } bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const { return isStrategyActive(NUM_STRATEGIES, inPastMs); } bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy, uint32_t inPastMs, nsecs_t sysTime) const { if ((sysTime == 0) && (inPastMs != 0)) { sysTime = systemTime(); } for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { if (i == AUDIO_STREAM_PATCH) { continue; } if (((getStrategy((audio_stream_type_t)i) == strategy) || (NUM_STRATEGIES == strategy)) && isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) { return true; } } return false; } bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs, nsecs_t sysTime) const { if (mRefCount[stream] != 0) { return true; } if (inPastMs == 0) { return false; } if (sysTime == 0) { sysTime = systemTime(); } if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) { return true; } return false; } void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig( struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig) const { ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle); dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN; if (srcConfig != NULL) { dstConfig->config_mask |= srcConfig->config_mask; } AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); dstConfig->id = mId; dstConfig->role = AUDIO_PORT_ROLE_SOURCE; dstConfig->type = AUDIO_PORT_TYPE_MIX; dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle; dstConfig->ext.mix.handle = mIoHandle; dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; } void AudioPolicyManager::AudioOutputDescriptor::toAudioPort( struct audio_port *port) const { ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle); mProfile->toAudioPort(port); port->id = mId; toAudioPortConfig(&port->active_config); port->ext.mix.hw_module = mProfile->mModule->mHandle; port->ext.mix.handle = mIoHandle; port->ext.mix.latency_class = mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL; } status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, " ID: %d\n", mId); result.append(buffer); snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); result.append(buffer); snprintf(buffer, SIZE, " Format: %08x\n", mFormat); result.append(buffer); snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); result.append(buffer); snprintf(buffer, SIZE, " Latency: %d\n", mLatency); result.append(buffer); snprintf(buffer, SIZE, " Flags %08x\n", mFlags); result.append(buffer); snprintf(buffer, SIZE, " Devices %08x\n", device()); result.append(buffer); snprintf(buffer, SIZE, " Stream volume refCount muteCount\n"); result.append(buffer); for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n", i, mCurVolume[i], mRefCount[i], mMuteCount[i]); result.append(buffer); } write(fd, result.string(), result.size()); return NO_ERROR; } // --- AudioInputDescriptor class implementation AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp& profile) : mId(0), mIoHandle(0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0), mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false) { if (profile != NULL) { mSamplingRate = profile->pickSamplingRate(); mFormat = profile->pickFormat(); mChannelMask = profile->pickChannelMask(); if (profile->mGains.size() > 0) { profile->mGains[0]->getDefaultConfig(&mGain); } } } void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig( struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig) const { ALOG_ASSERT(mProfile != 0, "toAudioPortConfig() called on input with null profile %d", mIoHandle); dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN; if (srcConfig != NULL) { dstConfig->config_mask |= srcConfig->config_mask; } AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); dstConfig->id = mId; dstConfig->role = AUDIO_PORT_ROLE_SINK; dstConfig->type = AUDIO_PORT_TYPE_MIX; dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle; dstConfig->ext.mix.handle = mIoHandle; dstConfig->ext.mix.usecase.source = mInputSource; } void AudioPolicyManager::AudioInputDescriptor::toAudioPort( struct audio_port *port) const { ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle); mProfile->toAudioPort(port); port->id = mId; toAudioPortConfig(&port->active_config); port->ext.mix.hw_module = mProfile->mModule->mHandle; port->ext.mix.handle = mIoHandle; port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL; } status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, " ID: %d\n", mId); result.append(buffer); snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); result.append(buffer); snprintf(buffer, SIZE, " Format: %d\n", mFormat); result.append(buffer); snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); result.append(buffer); snprintf(buffer, SIZE, " Devices %08x\n", mDevice); result.append(buffer); snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount); result.append(buffer); snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount); result.append(buffer); write(fd, result.string(), result.size()); return NO_ERROR; } // --- StreamDescriptor class implementation AudioPolicyManager::StreamDescriptor::StreamDescriptor() : mIndexMin(0), mIndexMax(1), mCanBeMuted(true) { mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0); } int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device) { device = AudioPolicyManager::getDeviceForVolume(device); // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT if (mIndexCur.indexOfKey(device) < 0) { device = AUDIO_DEVICE_OUT_DEFAULT; } return mIndexCur.valueFor(device); } void AudioPolicyManager::StreamDescriptor::dump(int fd) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "%s %02d %02d ", mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax); result.append(buffer); for (size_t i = 0; i < mIndexCur.size(); i++) { snprintf(buffer, SIZE, "%04x : %02d, ", mIndexCur.keyAt(i), mIndexCur.valueAt(i)); result.append(buffer); } result.append("\n"); write(fd, result.string(), result.size()); } // --- EffectDescriptor class implementation status_t AudioPolicyManager::EffectDescriptor::dump(int fd) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, " I/O: %d\n", mIo); result.append(buffer); snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy); result.append(buffer); snprintf(buffer, SIZE, " Session: %d\n", mSession); result.append(buffer); snprintf(buffer, SIZE, " Name: %s\n", mDesc.name); result.append(buffer); snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled"); result.append(buffer); write(fd, result.string(), result.size()); return NO_ERROR; } // --- HwModule class implementation AudioPolicyManager::HwModule::HwModule(const char *name) : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0) { } AudioPolicyManager::HwModule::~HwModule() { for (size_t i = 0; i < mOutputProfiles.size(); i++) { mOutputProfiles[i]->mSupportedDevices.clear(); } for (size_t i = 0; i < mInputProfiles.size(); i++) { mInputProfiles[i]->mSupportedDevices.clear(); } free((void *)mName); } status_t AudioPolicyManager::HwModule::loadInput(cnode *root) { cnode *node = root->first_child; sp profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this); while (node) { if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { profile->loadSamplingRates((char *)node->value); } else if (strcmp(node->name, FORMATS_TAG) == 0) { profile->loadFormats((char *)node->value); } else if (strcmp(node->name, CHANNELS_TAG) == 0) { profile->loadInChannels((char *)node->value); } else if (strcmp(node->name, DEVICES_TAG) == 0) { profile->mSupportedDevices.loadDevicesFromName((char *)node->value, mDeclaredDevices); } else if (strcmp(node->name, FLAGS_TAG) == 0) { profile->mFlags = parseInputFlagNames((char *)node->value); } else if (strcmp(node->name, GAINS_TAG) == 0) { profile->loadGains(node); } node = node->next; } ALOGW_IF(profile->mSupportedDevices.isEmpty(), "loadInput() invalid supported devices"); ALOGW_IF(profile->mChannelMasks.size() == 0, "loadInput() invalid supported channel masks"); ALOGW_IF(profile->mSamplingRates.size() == 0, "loadInput() invalid supported sampling rates"); ALOGW_IF(profile->mFormats.size() == 0, "loadInput() invalid supported formats"); if (!profile->mSupportedDevices.isEmpty() && (profile->mChannelMasks.size() != 0) && (profile->mSamplingRates.size() != 0) && (profile->mFormats.size() != 0)) { ALOGV("loadInput() adding input Supported Devices %04x", profile->mSupportedDevices.types()); mInputProfiles.add(profile); return NO_ERROR; } else { return BAD_VALUE; } } status_t AudioPolicyManager::HwModule::loadOutput(cnode *root) { cnode *node = root->first_child; sp profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this); while (node) { if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { profile->loadSamplingRates((char *)node->value); } else if (strcmp(node->name, FORMATS_TAG) == 0) { profile->loadFormats((char *)node->value); } else if (strcmp(node->name, CHANNELS_TAG) == 0) { profile->loadOutChannels((char *)node->value); } else if (strcmp(node->name, DEVICES_TAG) == 0) { profile->mSupportedDevices.loadDevicesFromName((char *)node->value, mDeclaredDevices); } else if (strcmp(node->name, FLAGS_TAG) == 0) { profile->mFlags = parseOutputFlagNames((char *)node->value); } else if (strcmp(node->name, GAINS_TAG) == 0) { profile->loadGains(node); } node = node->next; } ALOGW_IF(profile->mSupportedDevices.isEmpty(), "loadOutput() invalid supported devices"); ALOGW_IF(profile->mChannelMasks.size() == 0, "loadOutput() invalid supported channel masks"); ALOGW_IF(profile->mSamplingRates.size() == 0, "loadOutput() invalid supported sampling rates"); ALOGW_IF(profile->mFormats.size() == 0, "loadOutput() invalid supported formats"); if (!profile->mSupportedDevices.isEmpty() && (profile->mChannelMasks.size() != 0) && (profile->mSamplingRates.size() != 0) && (profile->mFormats.size() != 0)) { ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x", profile->mSupportedDevices.types(), profile->mFlags); mOutputProfiles.add(profile); return NO_ERROR; } else { return BAD_VALUE; } } status_t AudioPolicyManager::HwModule::loadDevice(cnode *root) { cnode *node = root->first_child; audio_devices_t type = AUDIO_DEVICE_NONE; while (node) { if (strcmp(node->name, DEVICE_TYPE) == 0) { type = parseDeviceNames((char *)node->value); break; } node = node->next; } if (type == AUDIO_DEVICE_NONE || (!audio_is_input_device(type) && !audio_is_output_device(type))) { ALOGW("loadDevice() bad type %08x", type); return BAD_VALUE; } sp deviceDesc = new DeviceDescriptor(String8(root->name), type); deviceDesc->mModule = this; node = root->first_child; while (node) { if (strcmp(node->name, DEVICE_ADDRESS) == 0) { deviceDesc->mAddress = String8((char *)node->value); } else if (strcmp(node->name, CHANNELS_TAG) == 0) { if (audio_is_input_device(type)) { deviceDesc->loadInChannels((char *)node->value); } else { deviceDesc->loadOutChannels((char *)node->value); } } else if (strcmp(node->name, GAINS_TAG) == 0) { deviceDesc->loadGains(node); } node = node->next; } ALOGV("loadDevice() adding device name %s type %08x address %s", deviceDesc->mName.string(), type, deviceDesc->mAddress.string()); mDeclaredDevices.add(deviceDesc); return NO_ERROR; } status_t AudioPolicyManager::HwModule::addOutputProfile(String8 name, const audio_config_t *config, audio_devices_t device, String8 address) { sp profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this); profile->mSamplingRates.add(config->sample_rate); profile->mChannelMasks.add(config->channel_mask); profile->mFormats.add(config->format); sp devDesc = new DeviceDescriptor(String8(""), device); devDesc->mAddress = address; profile->mSupportedDevices.add(devDesc); mOutputProfiles.add(profile); return NO_ERROR; } status_t AudioPolicyManager::HwModule::removeOutputProfile(String8 name) { for (size_t i = 0; i < mOutputProfiles.size(); i++) { if (mOutputProfiles[i]->mName == name) { mOutputProfiles.removeAt(i); break; } } return NO_ERROR; } status_t AudioPolicyManager::HwModule::addInputProfile(String8 name, const audio_config_t *config, audio_devices_t device, String8 address) { sp profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this); profile->mSamplingRates.add(config->sample_rate); profile->mChannelMasks.add(config->channel_mask); profile->mFormats.add(config->format); sp devDesc = new DeviceDescriptor(String8(""), device); devDesc->mAddress = address; profile->mSupportedDevices.add(devDesc); ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask); mInputProfiles.add(profile); return NO_ERROR; } status_t AudioPolicyManager::HwModule::removeInputProfile(String8 name) { for (size_t i = 0; i < mInputProfiles.size(); i++) { if (mInputProfiles[i]->mName == name) { mInputProfiles.removeAt(i); break; } } return NO_ERROR; } void AudioPolicyManager::HwModule::dump(int fd) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, " - name: %s\n", mName); result.append(buffer); snprintf(buffer, SIZE, " - handle: %d\n", mHandle); result.append(buffer); snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF); result.append(buffer); write(fd, result.string(), result.size()); if (mOutputProfiles.size()) { write(fd, " - outputs:\n", strlen(" - outputs:\n")); for (size_t i = 0; i < mOutputProfiles.size(); i++) { snprintf(buffer, SIZE, " output %zu:\n", i); write(fd, buffer, strlen(buffer)); mOutputProfiles[i]->dump(fd); } } if (mInputProfiles.size()) { write(fd, " - inputs:\n", strlen(" - inputs:\n")); for (size_t i = 0; i < mInputProfiles.size(); i++) { snprintf(buffer, SIZE, " input %zu:\n", i); write(fd, buffer, strlen(buffer)); mInputProfiles[i]->dump(fd); } } if (mDeclaredDevices.size()) { write(fd, " - devices:\n", strlen(" - devices:\n")); for (size_t i = 0; i < mDeclaredDevices.size(); i++) { mDeclaredDevices[i]->dump(fd, 4, i); } } } // --- AudioPort class implementation AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type, audio_port_role_t role, const sp& module) : mName(name), mType(type), mRole(role), mModule(module), mFlags(0) { mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) || ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK)); } void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const { port->role = mRole; port->type = mType; unsigned int i; for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) { if (mSamplingRates[i] != 0) { port->sample_rates[i] = mSamplingRates[i]; } } port->num_sample_rates = i; for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) { if (mChannelMasks[i] != 0) { port->channel_masks[i] = mChannelMasks[i]; } } port->num_channel_masks = i; for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) { if (mFormats[i] != 0) { port->formats[i] = mFormats[i]; } } port->num_formats = i; ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size()); for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) { port->gains[i] = mGains[i]->mGain; } port->num_gains = i; } void AudioPolicyManager::AudioPort::importAudioPort(const sp port) { for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) { const uint32_t rate = port->mSamplingRates.itemAt(k); if (rate != 0) { // skip "dynamic" rates bool hasRate = false; for (size_t l = 0 ; l < mSamplingRates.size() ; l++) { if (rate == mSamplingRates.itemAt(l)) { hasRate = true; break; } } if (!hasRate) { // never import a sampling rate twice mSamplingRates.add(rate); } } } for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) { const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k); if (mask != 0) { // skip "dynamic" masks bool hasMask = false; for (size_t l = 0 ; l < mChannelMasks.size() ; l++) { if (mask == mChannelMasks.itemAt(l)) { hasMask = true; break; } } if (!hasMask) { // never import a channel mask twice mChannelMasks.add(mask); } } } for (size_t k = 0 ; k < port->mFormats.size() ; k++) { const audio_format_t format = port->mFormats.itemAt(k); if (format != 0) { // skip "dynamic" formats bool hasFormat = false; for (size_t l = 0 ; l < mFormats.size() ; l++) { if (format == mFormats.itemAt(l)) { hasFormat = true; break; } } if (!hasFormat) { // never import a channel mask twice mFormats.add(format); } } } for (size_t k = 0 ; k < port->mGains.size() ; k++) { sp gain = port->mGains.itemAt(k); if (gain != 0) { bool hasGain = false; for (size_t l = 0 ; l < mGains.size() ; l++) { if (gain == mGains.itemAt(l)) { hasGain = true; break; } } if (!hasGain) { // never import a gain twice mGains.add(gain); } } } } void AudioPolicyManager::AudioPort::clearCapabilities() { mChannelMasks.clear(); mFormats.clear(); mSamplingRates.clear(); mGains.clear(); } void AudioPolicyManager::AudioPort::loadSamplingRates(char *name) { char *str = strtok(name, "|"); // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling // rates should be read from the output stream after it is opened for the first time if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { mSamplingRates.add(0); return; } while (str != NULL) { uint32_t rate = atoi(str); if (rate != 0) { ALOGV("loadSamplingRates() adding rate %d", rate); mSamplingRates.add(rate); } str = strtok(NULL, "|"); } } void AudioPolicyManager::AudioPort::loadFormats(char *name) { char *str = strtok(name, "|"); // by convention, "0' in the first entry in mFormats indicates the supported formats // should be read from the output stream after it is opened for the first time if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { mFormats.add(AUDIO_FORMAT_DEFAULT); return; } while (str != NULL) { audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable, ARRAY_SIZE(sFormatNameToEnumTable), str); if (format != AUDIO_FORMAT_DEFAULT) { mFormats.add(format); } str = strtok(NULL, "|"); } } void AudioPolicyManager::AudioPort::loadInChannels(char *name) { const char *str = strtok(name, "|"); ALOGV("loadInChannels() %s", name); if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { mChannelMasks.add(0); return; } while (str != NULL) { audio_channel_mask_t channelMask = (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable, ARRAY_SIZE(sInChannelsNameToEnumTable), str); if (channelMask != 0) { ALOGV("loadInChannels() adding channelMask %04x", channelMask); mChannelMasks.add(channelMask); } str = strtok(NULL, "|"); } } void AudioPolicyManager::AudioPort::loadOutChannels(char *name) { const char *str = strtok(name, "|"); ALOGV("loadOutChannels() %s", name); // by convention, "0' in the first entry in mChannelMasks indicates the supported channel // masks should be read from the output stream after it is opened for the first time if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { mChannelMasks.add(0); return; } while (str != NULL) { audio_channel_mask_t channelMask = (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable, ARRAY_SIZE(sOutChannelsNameToEnumTable), str); if (channelMask != 0) { mChannelMasks.add(channelMask); } str = strtok(NULL, "|"); } return; } audio_gain_mode_t AudioPolicyManager::AudioPort::loadGainMode(char *name) { const char *str = strtok(name, "|"); ALOGV("loadGainMode() %s", name); audio_gain_mode_t mode = 0; while (str != NULL) { mode |= (audio_gain_mode_t)stringToEnum(sGainModeNameToEnumTable, ARRAY_SIZE(sGainModeNameToEnumTable), str); str = strtok(NULL, "|"); } return mode; } void AudioPolicyManager::AudioPort::loadGain(cnode *root, int index) { cnode *node = root->first_child; sp gain = new AudioGain(index, mUseInChannelMask); while (node) { if (strcmp(node->name, GAIN_MODE) == 0) { gain->mGain.mode = loadGainMode((char *)node->value); } else if (strcmp(node->name, GAIN_CHANNELS) == 0) { if (mUseInChannelMask) { gain->mGain.channel_mask = (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable, ARRAY_SIZE(sInChannelsNameToEnumTable), (char *)node->value); } else { gain->mGain.channel_mask = (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable, ARRAY_SIZE(sOutChannelsNameToEnumTable), (char *)node->value); } } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) { gain->mGain.min_value = atoi((char *)node->value); } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) { gain->mGain.max_value = atoi((char *)node->value); } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) { gain->mGain.default_value = atoi((char *)node->value); } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) { gain->mGain.step_value = atoi((char *)node->value); } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) { gain->mGain.min_ramp_ms = atoi((char *)node->value); } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) { gain->mGain.max_ramp_ms = atoi((char *)node->value); } node = node->next; } ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d", gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value); if (gain->mGain.mode == 0) { return; } mGains.add(gain); } void AudioPolicyManager::AudioPort::loadGains(cnode *root) { cnode *node = root->first_child; int index = 0; while (node) { ALOGV("loadGains() loading gain %s", node->name); loadGain(node, index++); node = node->next; } } status_t AudioPolicyManager::AudioPort::checkExactSamplingRate(uint32_t samplingRate) const { if (mSamplingRates.isEmpty()) { return NO_ERROR; } for (size_t i = 0; i < mSamplingRates.size(); i ++) { if (mSamplingRates[i] == samplingRate) { return NO_ERROR; } } return BAD_VALUE; } status_t AudioPolicyManager::AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate, uint32_t *updatedSamplingRate) const { if (mSamplingRates.isEmpty()) { return NO_ERROR; } // Search for the closest supported sampling rate that is above (preferred) // or below (acceptable) the desired sampling rate, within a permitted ratio. // The sampling rates do not need to be sorted in ascending order. ssize_t maxBelow = -1; ssize_t minAbove = -1; uint32_t candidate; for (size_t i = 0; i < mSamplingRates.size(); i++) { candidate = mSamplingRates[i]; if (candidate == samplingRate) { if (updatedSamplingRate != NULL) { *updatedSamplingRate = candidate; } return NO_ERROR; } // candidate < desired if (candidate < samplingRate) { if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) { maxBelow = i; } // candidate > desired } else { if (minAbove < 0 || candidate < mSamplingRates[minAbove]) { minAbove = i; } } } // This uses hard-coded knowledge about AudioFlinger resampling ratios. // TODO Move these assumptions out. static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur // due to approximation by an int32_t of the // phase increments // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum. if (minAbove >= 0) { candidate = mSamplingRates[minAbove]; if (candidate / kMaxDownSampleRatio <= samplingRate) { if (updatedSamplingRate != NULL) { *updatedSamplingRate = candidate; } return NO_ERROR; } } // But if we have to up-sample from a lower sampling rate, that's OK. if (maxBelow >= 0) { candidate = mSamplingRates[maxBelow]; if (candidate * kMaxUpSampleRatio >= samplingRate) { if (updatedSamplingRate != NULL) { *updatedSamplingRate = candidate; } return NO_ERROR; } } // leave updatedSamplingRate unmodified return BAD_VALUE; } status_t AudioPolicyManager::AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const { if (mChannelMasks.isEmpty()) { return NO_ERROR; } for (size_t i = 0; i < mChannelMasks.size(); i++) { if (mChannelMasks[i] == channelMask) { return NO_ERROR; } } return BAD_VALUE; } status_t AudioPolicyManager::AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask) const { if (mChannelMasks.isEmpty()) { return NO_ERROR; } const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK; for (size_t i = 0; i < mChannelMasks.size(); i ++) { // FIXME Does not handle multi-channel automatic conversions yet audio_channel_mask_t supported = mChannelMasks[i]; if (supported == channelMask) { return NO_ERROR; } if (isRecordThread) { // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix. // FIXME Abstract this out to a table. if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO) && channelMask == AUDIO_CHANNEL_IN_MONO) || (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK || channelMask == AUDIO_CHANNEL_IN_STEREO))) { return NO_ERROR; } } } return BAD_VALUE; } status_t AudioPolicyManager::AudioPort::checkFormat(audio_format_t format) const { if (mFormats.isEmpty()) { return NO_ERROR; } for (size_t i = 0; i < mFormats.size(); i ++) { if (mFormats[i] == format) { return NO_ERROR; } } return BAD_VALUE; } uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const { // special case for uninitialized dynamic profile if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) { return 0; } // For direct outputs, pick minimum sampling rate: this helps ensuring that the // channel count / sampling rate combination chosen will be supported by the connected // sink if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { uint32_t samplingRate = UINT_MAX; for (size_t i = 0; i < mSamplingRates.size(); i ++) { if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) { samplingRate = mSamplingRates[i]; } } return (samplingRate == UINT_MAX) ? 0 : samplingRate; } uint32_t samplingRate = 0; uint32_t maxRate = MAX_MIXER_SAMPLING_RATE; // For mixed output and inputs, use max mixer sampling rates. Do not // limit sampling rate otherwise if (mType != AUDIO_PORT_TYPE_MIX) { maxRate = UINT_MAX; } for (size_t i = 0; i < mSamplingRates.size(); i ++) { if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) { samplingRate = mSamplingRates[i]; } } return samplingRate; } audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const { // special case for uninitialized dynamic profile if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) { return AUDIO_CHANNEL_NONE; } audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE; // For direct outputs, pick minimum channel count: this helps ensuring that the // channel count / sampling rate combination chosen will be supported by the connected // sink if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { uint32_t channelCount = UINT_MAX; for (size_t i = 0; i < mChannelMasks.size(); i ++) { uint32_t cnlCount; if (mUseInChannelMask) { cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); } else { cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); } if ((cnlCount < channelCount) && (cnlCount > 0)) { channelMask = mChannelMasks[i]; channelCount = cnlCount; } } return channelMask; } uint32_t channelCount = 0; uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT; // For mixed output and inputs, use max mixer channel count. Do not // limit channel count otherwise if (mType != AUDIO_PORT_TYPE_MIX) { maxCount = UINT_MAX; } for (size_t i = 0; i < mChannelMasks.size(); i ++) { uint32_t cnlCount; if (mUseInChannelMask) { cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); } else { cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); } if ((cnlCount > channelCount) && (cnlCount <= maxCount)) { channelMask = mChannelMasks[i]; channelCount = cnlCount; } } return channelMask; } /* format in order of increasing preference */ const audio_format_t AudioPolicyManager::AudioPort::sPcmFormatCompareTable[] = { AUDIO_FORMAT_DEFAULT, AUDIO_FORMAT_PCM_16_BIT, AUDIO_FORMAT_PCM_8_24_BIT, AUDIO_FORMAT_PCM_24_BIT_PACKED, AUDIO_FORMAT_PCM_32_BIT, AUDIO_FORMAT_PCM_FLOAT, }; int AudioPolicyManager::AudioPort::compareFormats(audio_format_t format1, audio_format_t format2) { // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any // compressed format and better than any PCM format. This is by design of pickFormat() if (!audio_is_linear_pcm(format1)) { if (!audio_is_linear_pcm(format2)) { return 0; } return 1; } if (!audio_is_linear_pcm(format2)) { return -1; } int index1 = -1, index2 = -1; for (size_t i = 0; (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1)); i ++) { if (sPcmFormatCompareTable[i] == format1) { index1 = i; } if (sPcmFormatCompareTable[i] == format2) { index2 = i; } } // format1 not found => index1 < 0 => format2 > format1 // format2 not found => index2 < 0 => format2 < format1 return index1 - index2; } audio_format_t AudioPolicyManager::AudioPort::pickFormat() const { // special case for uninitialized dynamic profile if (mFormats.size() == 1 && mFormats[0] == 0) { return AUDIO_FORMAT_DEFAULT; } audio_format_t format = AUDIO_FORMAT_DEFAULT; audio_format_t bestFormat = AudioPolicyManager::AudioPort::sPcmFormatCompareTable[ ARRAY_SIZE(AudioPolicyManager::AudioPort::sPcmFormatCompareTable) - 1]; // For mixed output and inputs, use best mixer output format. Do not // limit format otherwise if ((mType != AUDIO_PORT_TYPE_MIX) || ((mRole == AUDIO_PORT_ROLE_SOURCE) && (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) { bestFormat = AUDIO_FORMAT_INVALID; } for (size_t i = 0; i < mFormats.size(); i ++) { if ((compareFormats(mFormats[i], format) > 0) && (compareFormats(mFormats[i], bestFormat) <= 0)) { format = mFormats[i]; } } return format; } status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig, int index) const { if (index < 0 || (size_t)index >= mGains.size()) { return BAD_VALUE; } return mGains[index]->checkConfig(gainConfig); } void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const { const size_t SIZE = 256; char buffer[SIZE]; String8 result; if (mName.size() != 0) { snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string()); result.append(buffer); } if (mSamplingRates.size() != 0) { snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, ""); result.append(buffer); for (size_t i = 0; i < mSamplingRates.size(); i++) { if (i == 0 && mSamplingRates[i] == 0) { snprintf(buffer, SIZE, "Dynamic"); } else { snprintf(buffer, SIZE, "%d", mSamplingRates[i]); } result.append(buffer); result.append(i == (mSamplingRates.size() - 1) ? "" : ", "); } result.append("\n"); } if (mChannelMasks.size() != 0) { snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, ""); result.append(buffer); for (size_t i = 0; i < mChannelMasks.size(); i++) { ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]); if (i == 0 && mChannelMasks[i] == 0) { snprintf(buffer, SIZE, "Dynamic"); } else { snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]); } result.append(buffer); result.append(i == (mChannelMasks.size() - 1) ? "" : ", "); } result.append("\n"); } if (mFormats.size() != 0) { snprintf(buffer, SIZE, "%*s- formats: ", spaces, ""); result.append(buffer); for (size_t i = 0; i < mFormats.size(); i++) { const char *formatStr = enumToString(sFormatNameToEnumTable, ARRAY_SIZE(sFormatNameToEnumTable), mFormats[i]); if (i == 0 && strcmp(formatStr, "") == 0) { snprintf(buffer, SIZE, "Dynamic"); } else { snprintf(buffer, SIZE, "%s", formatStr); } result.append(buffer); result.append(i == (mFormats.size() - 1) ? "" : ", "); } result.append("\n"); } write(fd, result.string(), result.size()); if (mGains.size() != 0) { snprintf(buffer, SIZE, "%*s- gains:\n", spaces, ""); write(fd, buffer, strlen(buffer) + 1); result.append(buffer); for (size_t i = 0; i < mGains.size(); i++) { mGains[i]->dump(fd, spaces + 2, i); } } } // --- AudioGain class implementation AudioPolicyManager::AudioGain::AudioGain(int index, bool useInChannelMask) { mIndex = index; mUseInChannelMask = useInChannelMask; memset(&mGain, 0, sizeof(struct audio_gain)); } void AudioPolicyManager::AudioGain::getDefaultConfig(struct audio_gain_config *config) { config->index = mIndex; config->mode = mGain.mode; config->channel_mask = mGain.channel_mask; if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) { config->values[0] = mGain.default_value; } else { uint32_t numValues; if (mUseInChannelMask) { numValues = audio_channel_count_from_in_mask(mGain.channel_mask); } else { numValues = audio_channel_count_from_out_mask(mGain.channel_mask); } for (size_t i = 0; i < numValues; i++) { config->values[i] = mGain.default_value; } } if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) { config->ramp_duration_ms = mGain.min_ramp_ms; } } status_t AudioPolicyManager::AudioGain::checkConfig(const struct audio_gain_config *config) { if ((config->mode & ~mGain.mode) != 0) { return BAD_VALUE; } if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) { if ((config->values[0] < mGain.min_value) || (config->values[0] > mGain.max_value)) { return BAD_VALUE; } } else { if ((config->channel_mask & ~mGain.channel_mask) != 0) { return BAD_VALUE; } uint32_t numValues; if (mUseInChannelMask) { numValues = audio_channel_count_from_in_mask(config->channel_mask); } else { numValues = audio_channel_count_from_out_mask(config->channel_mask); } for (size_t i = 0; i < numValues; i++) { if ((config->values[i] < mGain.min_value) || (config->values[i] > mGain.max_value)) { return BAD_VALUE; } } } if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) { if ((config->ramp_duration_ms < mGain.min_ramp_ms) || (config->ramp_duration_ms > mGain.max_ramp_ms)) { return BAD_VALUE; } } return NO_ERROR; } void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1); result.append(buffer); snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode); result.append(buffer); snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask); result.append(buffer); snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value); result.append(buffer); snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value); result.append(buffer); snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value); result.append(buffer); snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value); result.append(buffer); snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms); result.append(buffer); snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms); result.append(buffer); write(fd, result.string(), result.size()); } // --- AudioPortConfig class implementation AudioPolicyManager::AudioPortConfig::AudioPortConfig() { mSamplingRate = 0; mChannelMask = AUDIO_CHANNEL_NONE; mFormat = AUDIO_FORMAT_INVALID; mGain.index = -1; } status_t AudioPolicyManager::AudioPortConfig::applyAudioPortConfig( const struct audio_port_config *config, struct audio_port_config *backupConfig) { struct audio_port_config localBackupConfig; status_t status = NO_ERROR; localBackupConfig.config_mask = config->config_mask; toAudioPortConfig(&localBackupConfig); sp audioport = getAudioPort(); if (audioport == 0) { status = NO_INIT; goto exit; } if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { status = audioport->checkExactSamplingRate(config->sample_rate); if (status != NO_ERROR) { goto exit; } mSamplingRate = config->sample_rate; } if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { status = audioport->checkExactChannelMask(config->channel_mask); if (status != NO_ERROR) { goto exit; } mChannelMask = config->channel_mask; } if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) { status = audioport->checkFormat(config->format); if (status != NO_ERROR) { goto exit; } mFormat = config->format; } if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) { status = audioport->checkGain(&config->gain, config->gain.index); if (status != NO_ERROR) { goto exit; } mGain = config->gain; } exit: if (status != NO_ERROR) { applyAudioPortConfig(&localBackupConfig); } if (backupConfig != NULL) { *backupConfig = localBackupConfig; } return status; } void AudioPolicyManager::AudioPortConfig::toAudioPortConfig( struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig) const { if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { dstConfig->sample_rate = mSamplingRate; if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) { dstConfig->sample_rate = srcConfig->sample_rate; } } else { dstConfig->sample_rate = 0; } if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { dstConfig->channel_mask = mChannelMask; if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) { dstConfig->channel_mask = srcConfig->channel_mask; } } else { dstConfig->channel_mask = AUDIO_CHANNEL_NONE; } if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) { dstConfig->format = mFormat; if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) { dstConfig->format = srcConfig->format; } } else { dstConfig->format = AUDIO_FORMAT_INVALID; } if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) { dstConfig->gain = mGain; if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) { dstConfig->gain = srcConfig->gain; } } else { dstConfig->gain.index = -1; } if (dstConfig->gain.index != -1) { dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN; } else { dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN; } } // --- IOProfile class implementation AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role, const sp& module) : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module) { } AudioPolicyManager::IOProfile::~IOProfile() { } // checks if the IO profile is compatible with specified parameters. // Sampling rate, format and channel mask must be specified in order to // get a valid a match bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device, String8 address, uint32_t samplingRate, uint32_t *updatedSamplingRate, audio_format_t format, audio_channel_mask_t channelMask, uint32_t flags) const { const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE; const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK; ALOG_ASSERT(isPlaybackThread != isRecordThread); if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) { return false; } if (samplingRate == 0) { return false; } uint32_t myUpdatedSamplingRate = samplingRate; if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) { return false; } if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) != NO_ERROR) { return false; } if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) { return false; } if (isPlaybackThread && (!audio_is_output_channel(channelMask) || checkExactChannelMask(channelMask) != NO_ERROR)) { return false; } if (isRecordThread && (!audio_is_input_channel(channelMask) || checkCompatibleChannelMask(channelMask) != NO_ERROR)) { return false; } if (isPlaybackThread && (mFlags & flags) != flags) { return false; } // The only input flag that is allowed to be different is the fast flag. // An existing fast stream is compatible with a normal track request. // An existing normal stream is compatible with a fast track request, // but the fast request will be denied by AudioFlinger and converted to normal track. if (isRecordThread && ((mFlags ^ flags) & ~AUDIO_INPUT_FLAG_FAST)) { return false; } if (updatedSamplingRate != NULL) { *updatedSamplingRate = myUpdatedSamplingRate; } return true; } void AudioPolicyManager::IOProfile::dump(int fd) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; AudioPort::dump(fd, 4); snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags); result.append(buffer); snprintf(buffer, SIZE, " - devices:\n"); result.append(buffer); write(fd, result.string(), result.size()); for (size_t i = 0; i < mSupportedDevices.size(); i++) { mSupportedDevices[i]->dump(fd, 6, i); } } void AudioPolicyManager::IOProfile::log() { const size_t SIZE = 256; char buffer[SIZE]; String8 result; ALOGV(" - sampling rates: "); for (size_t i = 0; i < mSamplingRates.size(); i++) { ALOGV(" %d", mSamplingRates[i]); } ALOGV(" - channel masks: "); for (size_t i = 0; i < mChannelMasks.size(); i++) { ALOGV(" 0x%04x", mChannelMasks[i]); } ALOGV(" - formats: "); for (size_t i = 0; i < mFormats.size(); i++) { ALOGV(" 0x%08x", mFormats[i]); } ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types()); ALOGV(" - flags: 0x%04x\n", mFlags); } // --- DeviceDescriptor implementation AudioPolicyManager::DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) : AudioPort(name, AUDIO_PORT_TYPE_DEVICE, audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE, NULL), mDeviceType(type), mAddress(""), mId(0) { if (mGains.size() > 0) { mGains[0]->getDefaultConfig(&mGain); } } bool AudioPolicyManager::DeviceDescriptor::equals(const sp& other) const { // Devices are considered equal if they: // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE) // - have the same address or one device does not specify the address // - have the same channel mask or one device does not specify the channel mask return (mDeviceType == other->mDeviceType) && (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) && (mChannelMask == 0 || other->mChannelMask == 0 || mChannelMask == other->mChannelMask); } void AudioPolicyManager::DeviceVector::refreshTypes() { mDeviceTypes = AUDIO_DEVICE_NONE; for(size_t i = 0; i < size(); i++) { mDeviceTypes |= itemAt(i)->mDeviceType; } ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes); } ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp& item) const { for(size_t i = 0; i < size(); i++) { if (item->equals(itemAt(i))) { return i; } } return -1; } ssize_t AudioPolicyManager::DeviceVector::add(const sp& item) { ssize_t ret = indexOf(item); if (ret < 0) { ret = SortedVector::add(item); if (ret >= 0) { refreshTypes(); } } else { ALOGW("DeviceVector::add device %08x already in", item->mDeviceType); ret = -1; } return ret; } ssize_t AudioPolicyManager::DeviceVector::remove(const sp& item) { size_t i; ssize_t ret = indexOf(item); if (ret < 0) { ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType); } else { ret = SortedVector::removeAt(ret); if (ret >= 0) { refreshTypes(); } } return ret; } void AudioPolicyManager::DeviceVector::loadDevicesFromType(audio_devices_t types) { DeviceVector deviceList; uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types; types &= ~role_bit; while (types) { uint32_t i = 31 - __builtin_clz(types); uint32_t type = 1 << i; types &= ~type; add(new DeviceDescriptor(String8(""), type | role_bit)); } } void AudioPolicyManager::DeviceVector::loadDevicesFromName(char *name, const DeviceVector& declaredDevices) { char *devName = strtok(name, "|"); while (devName != NULL) { if (strlen(devName) != 0) { audio_devices_t type = stringToEnum(sDeviceNameToEnumTable, ARRAY_SIZE(sDeviceNameToEnumTable), devName); if (type != AUDIO_DEVICE_NONE) { sp dev = new DeviceDescriptor(String8(""), type); if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX || type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) { dev->mAddress = String8("0"); } add(dev); } else { sp deviceDesc = declaredDevices.getDeviceFromName(String8(devName)); if (deviceDesc != 0) { add(deviceDesc); } } } devName = strtok(NULL, "|"); } } sp AudioPolicyManager::DeviceVector::getDevice( audio_devices_t type, String8 address) const { sp device; for (size_t i = 0; i < size(); i++) { if (itemAt(i)->mDeviceType == type) { if (address == "" || itemAt(i)->mAddress == address) { device = itemAt(i); if (itemAt(i)->mAddress == address) { break; } } } } ALOGV("DeviceVector::getDevice() for type %08x address %s found %p", type, address.string(), device.get()); return device; } sp AudioPolicyManager::DeviceVector::getDeviceFromId( audio_port_handle_t id) const { sp device; for (size_t i = 0; i < size(); i++) { ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%zu)->mId %d", id, i, itemAt(i)->mId); if (itemAt(i)->mId == id) { device = itemAt(i); break; } } return device; } AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType( audio_devices_t type) const { DeviceVector devices; for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) { if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) { devices.add(itemAt(i)); type &= ~itemAt(i)->mDeviceType; ALOGV("DeviceVector::getDevicesFromType() for type %x found %p", itemAt(i)->mDeviceType, itemAt(i).get()); } } return devices; } AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromTypeAddr( audio_devices_t type, String8 address) const { DeviceVector devices; for (size_t i = 0; i < size(); i++) { if (itemAt(i)->mDeviceType == type) { if (itemAt(i)->mAddress == address) { devices.add(itemAt(i)); } } } return devices; } sp AudioPolicyManager::DeviceVector::getDeviceFromName( const String8& name) const { sp device; for (size_t i = 0; i < size(); i++) { if (itemAt(i)->mName == name) { device = itemAt(i); break; } } return device; } void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig( struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig) const { dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN; if (srcConfig != NULL) { dstConfig->config_mask |= srcConfig->config_mask; } AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig); dstConfig->id = mId; dstConfig->role = audio_is_output_device(mDeviceType) ? AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE; dstConfig->type = AUDIO_PORT_TYPE_DEVICE; dstConfig->ext.device.type = mDeviceType; dstConfig->ext.device.hw_module = mModule->mHandle; strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN); } void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const { ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType); AudioPort::toAudioPort(port); port->id = mId; toAudioPortConfig(&port->active_config); port->ext.device.type = mDeviceType; port->ext.device.hw_module = mModule->mHandle; strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN); } status_t AudioPolicyManager::DeviceDescriptor::dump(int fd, int spaces, int index) const { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1); result.append(buffer); if (mId != 0) { snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId); result.append(buffer); } snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "", enumToString(sDeviceNameToEnumTable, ARRAY_SIZE(sDeviceNameToEnumTable), mDeviceType)); result.append(buffer); if (mAddress.size() != 0) { snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string()); result.append(buffer); } write(fd, result.string(), result.size()); AudioPort::dump(fd, spaces); return NO_ERROR; } status_t AudioPolicyManager::AudioPatch::dump(int fd, int spaces, int index) const { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1); result.append(buffer); snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle); result.append(buffer); snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle); result.append(buffer); snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid); result.append(buffer); snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources); result.append(buffer); for (size_t i = 0; i < mPatch.num_sources; i++) { if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) { snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "", mPatch.sources[i].id, enumToString(sDeviceNameToEnumTable, ARRAY_SIZE(sDeviceNameToEnumTable), mPatch.sources[i].ext.device.type)); } else { snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "", mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle); } result.append(buffer); } snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks); result.append(buffer); for (size_t i = 0; i < mPatch.num_sinks; i++) { if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) { snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "", mPatch.sinks[i].id, enumToString(sDeviceNameToEnumTable, ARRAY_SIZE(sDeviceNameToEnumTable), mPatch.sinks[i].ext.device.type)); } else { snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "", mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle); } result.append(buffer); } write(fd, result.string(), result.size()); return NO_ERROR; } // --- audio_policy.conf file parsing uint32_t AudioPolicyManager::parseOutputFlagNames(char *name) { uint32_t flag = 0; // it is OK to cast name to non const here as we are not going to use it after // strtok() modifies it char *flagName = strtok(name, "|"); while (flagName != NULL) { if (strlen(flagName) != 0) { flag |= stringToEnum(sOutputFlagNameToEnumTable, ARRAY_SIZE(sOutputFlagNameToEnumTable), flagName); } flagName = strtok(NULL, "|"); } //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 ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { flag |= AUDIO_OUTPUT_FLAG_DIRECT; } return flag; } uint32_t AudioPolicyManager::parseInputFlagNames(char *name) { uint32_t flag = 0; // it is OK to cast name to non const here as we are not going to use it after // strtok() modifies it char *flagName = strtok(name, "|"); while (flagName != NULL) { if (strlen(flagName) != 0) { flag |= stringToEnum(sInputFlagNameToEnumTable, ARRAY_SIZE(sInputFlagNameToEnumTable), flagName); } flagName = strtok(NULL, "|"); } return flag; } audio_devices_t AudioPolicyManager::parseDeviceNames(char *name) { uint32_t device = 0; char *devName = strtok(name, "|"); while (devName != NULL) { if (strlen(devName) != 0) { device |= stringToEnum(sDeviceNameToEnumTable, ARRAY_SIZE(sDeviceNameToEnumTable), devName); } devName = strtok(NULL, "|"); } return device; } void AudioPolicyManager::loadHwModule(cnode *root) { status_t status = NAME_NOT_FOUND; cnode *node; sp module = new HwModule(root->name); node = config_find(root, DEVICES_TAG); if (node != NULL) { node = node->first_child; while (node) { ALOGV("loadHwModule() loading device %s", node->name); status_t tmpStatus = module->loadDevice(node); if (status == NAME_NOT_FOUND || status == NO_ERROR) { status = tmpStatus; } node = node->next; } } node = config_find(root, OUTPUTS_TAG); if (node != NULL) { node = node->first_child; while (node) { ALOGV("loadHwModule() loading output %s", node->name); status_t tmpStatus = module->loadOutput(node); if (status == NAME_NOT_FOUND || status == NO_ERROR) { status = tmpStatus; } node = node->next; } } node = config_find(root, INPUTS_TAG); if (node != NULL) { node = node->first_child; while (node) { ALOGV("loadHwModule() loading input %s", node->name); status_t tmpStatus = module->loadInput(node); if (status == NAME_NOT_FOUND || status == NO_ERROR) { status = tmpStatus; } node = node->next; } } loadGlobalConfig(root, module); if (status == NO_ERROR) { mHwModules.add(module); } } void AudioPolicyManager::loadHwModules(cnode *root) { cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); if (node == NULL) { return; } node = node->first_child; while (node) { ALOGV("loadHwModules() loading module %s", node->name); loadHwModule(node); node = node->next; } } void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp& module) { cnode *node = config_find(root, GLOBAL_CONFIG_TAG); if (node == NULL) { return; } DeviceVector declaredDevices; if (module != NULL) { declaredDevices = module->mDeclaredDevices; } node = node->first_child; while (node) { if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) { mAvailableOutputDevices.loadDevicesFromName((char *)node->value, declaredDevices); ALOGV("loadGlobalConfig() Attached Output Devices %08x", mAvailableOutputDevices.types()); } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) { audio_devices_t device = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable, ARRAY_SIZE(sDeviceNameToEnumTable), (char *)node->value); if (device != AUDIO_DEVICE_NONE) { mDefaultOutputDevice = new DeviceDescriptor(String8(""), device); } else { ALOGW("loadGlobalConfig() default device not specified"); } ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType); } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) { mAvailableInputDevices.loadDevicesFromName((char *)node->value, declaredDevices); ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types()); } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) { mSpeakerDrcEnabled = stringToBool((char *)node->value); ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled); } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) { uint32_t major, minor; sscanf((char *)node->value, "%u.%u", &major, &minor); module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor); ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u", module->mHalVersion, major, minor); } node = node->next; } } status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path) { cnode *root; char *data; data = (char *)load_file(path, NULL); if (data == NULL) { return -ENODEV; } root = config_node("", ""); config_load(root, data); loadHwModules(root); // legacy audio_policy.conf files have one global_configuration section loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)); config_free(root); free(root); free(data); ALOGI("loadAudioPolicyConfig() loaded %s\n", path); return NO_ERROR; } void AudioPolicyManager::defaultAudioPolicyConfig(void) { sp module; sp profile; sp defaultInputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_IN_BUILTIN_MIC); mAvailableOutputDevices.add(mDefaultOutputDevice); mAvailableInputDevices.add(defaultInputDevice); module = new HwModule("primary"); profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module); profile->mSamplingRates.add(44100); profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); profile->mSupportedDevices.add(mDefaultOutputDevice); profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; module->mOutputProfiles.add(profile); profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module); profile->mSamplingRates.add(8000); profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); profile->mSupportedDevices.add(defaultInputDevice); module->mInputProfiles.add(profile); mHwModules.add(module); } audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr) { // flags to stream type mapping if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { return AUDIO_STREAM_ENFORCED_AUDIBLE; } if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { return AUDIO_STREAM_BLUETOOTH_SCO; } // usage to stream type mapping switch (attr->usage) { case AUDIO_USAGE_MEDIA: case AUDIO_USAGE_GAME: case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: return AUDIO_STREAM_MUSIC; case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: if (isStreamActive(AUDIO_STREAM_ALARM)) { return AUDIO_STREAM_ALARM; } if (isStreamActive(AUDIO_STREAM_RING)) { return AUDIO_STREAM_RING; } if (isInCall()) { return AUDIO_STREAM_VOICE_CALL; } return AUDIO_STREAM_ACCESSIBILITY; case AUDIO_USAGE_ASSISTANCE_SONIFICATION: return AUDIO_STREAM_SYSTEM; case AUDIO_USAGE_VOICE_COMMUNICATION: return AUDIO_STREAM_VOICE_CALL; case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: return AUDIO_STREAM_DTMF; case AUDIO_USAGE_ALARM: return AUDIO_STREAM_ALARM; case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: return AUDIO_STREAM_RING; case AUDIO_USAGE_NOTIFICATION: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: case AUDIO_USAGE_NOTIFICATION_EVENT: return AUDIO_STREAM_NOTIFICATION; case AUDIO_USAGE_UNKNOWN: default: return AUDIO_STREAM_MUSIC; } } bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) { // has flags that map to a strategy? 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: break; default: return false; } return true; } }; // namespace android