| /* |
| * 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 <inttypes.h> |
| #include <math.h> |
| |
| #include <cutils/properties.h> |
| #include <utils/Log.h> |
| #include <hardware/audio.h> |
| #include <hardware/audio_effect.h> |
| #include <media/AudioParameter.h> |
| #include <media/AudioPolicyHelper.h> |
| #include <soundtrigger/SoundTrigger.h> |
| #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) |
| { |
| return setDeviceConnectionStateInt(device, state, device_address); |
| } |
| |
| status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, |
| audio_policy_dev_state_t state, |
| const char *device_address) |
| { |
| ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", |
| device, state, device_address != NULL ? device_address : ""); |
| |
| // connect/disconnect only 1 device at a time |
| if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; |
| |
| sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address); |
| |
| // handle output devices |
| if (audio_is_output_device(device)) { |
| SortedVector <audio_io_handle_t> outputs; |
| |
| 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<HwModule> 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, devDesc->mAddress) != 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()); |
| |
| |
| // Set connect to HALs |
| AudioParameter param = AudioParameter(devDesc->mAddress); |
| param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); |
| mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); |
| |
| } 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(devDesc->mAddress); |
| 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, devDesc->mAddress); |
| } 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<AudioOutputDescriptor> 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 <audio_io_handle_t> inputs; |
| |
| 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<HwModule> 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, devDesc->mAddress) != NO_ERROR) { |
| return INVALID_OPERATION; |
| } |
| |
| index = mAvailableInputDevices.add(devDesc); |
| if (index >= 0) { |
| mAvailableInputDevices[index]->mId = nextUniqueId(); |
| mAvailableInputDevices[index]->mModule = module; |
| } else { |
| return NO_MEMORY; |
| } |
| |
| // Set connect to HALs |
| AudioParameter param = AudioParameter(devDesc->mAddress); |
| param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); |
| mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); |
| |
| } 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(devDesc->mAddress); |
| param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); |
| mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); |
| |
| checkInputsForDevice(device, state, inputs, devDesc->mAddress); |
| 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) |
| { |
| sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address); |
| 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; |
| } |
| |
| ssize_t index = deviceVector->indexOf(devDesc); |
| if (index >= 0) { |
| return AUDIO_POLICY_DEVICE_STATE_AVAILABLE; |
| } else { |
| return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; |
| } |
| } |
| |
| sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor( |
| const audio_devices_t device, |
| 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"); |
| } |
| |
| for (size_t i = 0; i < mHwModules.size(); i++) { |
| if (mHwModules[i]->mHandle == 0) { |
| continue; |
| } |
| DeviceVector deviceList = |
| mHwModules[i]->mDeclaredDevices.getDevicesFromTypeAddr(device, address); |
| if (!deviceList.isEmpty()) { |
| return deviceList.itemAt(0); |
| } |
| deviceList = mHwModules[i]->mDeclaredDevices.getDevicesFromType(device); |
| if (!deviceList.isEmpty()) { |
| return deviceList.itemAt(0); |
| } |
| } |
| |
| sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device); |
| devDesc->mAddress = address; |
| return devDesc; |
| } |
| |
| 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 = getDeviceAndMixForInputSource(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<DeviceDescriptor> rxSinkDeviceDesc = deviceList.itemAt(0); |
| deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX); |
| ALOG_ASSERT(!deviceList.isEmpty(), |
| "updateCallRouting() no telephony RX device"); |
| sp<DeviceDescriptor> 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<audio_io_handle_t> 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<AudioOutputDescriptor> 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<DeviceDescriptor> 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<DeviceDescriptor> txSinkDeviceDesc = deviceList.itemAt(0); |
| txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]); |
| |
| SortedVector<audio_io_handle_t> 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<AudioOutputDescriptor> 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); |
| } |
| |
| // force reevaluating accessibility routing when call starts |
| mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); |
| } |
| |
| // 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<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); |
| |
| int delayMs = 0; |
| if (isStateInCall(state)) { |
| nsecs_t sysTime = systemTime(); |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<AudioOutputDescriptor> 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::IOProfile> 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<IOProfile> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<IOProfile> 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<AudioOutputDescriptor> outputDesc = NULL; |
| |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| sp<AudioOutputDescriptor> 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); |
| } |
| // fall back to mixer output if possible when the direct output could not be open |
| if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) { |
| goto non_direct_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<audio_io_handle_t> 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<audio_io_handle_t>& 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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) { |
| setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, |
| AUDIO_POLICY_DEVICE_STATE_AVAILABLE, |
| outputDesc->mPolicyMix->mRegistrationId); |
| } |
| |
| // force reevaluating accessibility routing when ringtone or alarm starts |
| if (strategy == STRATEGY_SONIFICATION) { |
| mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); |
| } |
| |
| 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<AudioOutputDescriptor> 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) { |
| setDeviceConnectionStateInt(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<AudioOutputDescriptor> 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<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); |
| if (outputDesc->isActive()) { |
| mpClientInterface->closeOutput(output); |
| mOutputs.removeItem(output); |
| mTestOutputs[testIndex] = 0; |
| } |
| return; |
| } |
| #endif //AUDIO_POLICY_TEST |
| |
| sp<AudioOutputDescriptor> 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, |
| input_type_t *inputType) |
| { |
| 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; |
| *inputType = API_INPUT_INVALID; |
| 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 inputSource = attr->source; |
| audio_source_t halInputSource; |
| AudioMix *policyMix = NULL; |
| |
| if (inputSource == AUDIO_SOURCE_DEFAULT) { |
| inputSource = AUDIO_SOURCE_MIC; |
| } |
| halInputSource = inputSource; |
| |
| if (inputSource == 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; |
| *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; |
| } else { |
| device = getDeviceAndMixForInputSource(inputSource, &policyMix); |
| if (device == AUDIO_DEVICE_NONE) { |
| ALOGW("getInputForAttr() could not find device for source %d", inputSource); |
| return BAD_VALUE; |
| } |
| if (policyMix != NULL) { |
| address = policyMix->mRegistrationId; |
| if (policyMix->mMixType == MIX_TYPE_RECORDERS) { |
| // there is an external policy, but this input is attached to a mix of recorders, |
| // meaning it receives audio injected into the framework, so the recorder doesn't |
| // know about it and is therefore considered "legacy" |
| *inputType = API_INPUT_LEGACY; |
| } else { |
| // recording a mix of players defined by an external policy, we're rerouting for |
| // an external policy |
| *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; |
| } |
| } else if (audio_is_remote_submix_device(device)) { |
| address = String8("0"); |
| *inputType = API_INPUT_MIX_CAPTURE; |
| } else { |
| *inputType = API_INPUT_LEGACY; |
| } |
| // adapt channel selection to input source |
| switch (inputSource) { |
| 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 (inputSource == 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<IOProfile> 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<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile); |
| inputDesc->mInputSource = inputSource; |
| 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; |
| |
| ALOGV("getInputForAttr() returns input type = %d", inputType); |
| |
| 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<AudioInputDescriptor> 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<AudioInputDescriptor> 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 != "") { |
| setDeviceConnectionStateInt(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<AudioInputDescriptor> 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 != "") { |
| setDeviceConnectionStateInt(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<AudioInputDescriptor> 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<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index); |
| ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); |
| if (patch_index >= 0) { |
| sp<AudioPatch> 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<audio_io_handle_t>& 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<AudioOutputDescriptor> 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<audio_io_handle_t> 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<EffectDescriptor> 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<EffectDescriptor> 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<EffectDescriptor>& 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<EffectDescriptor> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i); |
| if (inputDescriptor->mRefCount == 0) { |
| continue; |
| } |
| if (inputDescriptor->mInputSource == (int)source) { |
| return true; |
| } |
| // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it |
| // corresponds to an active capture triggered by a hardware hotword recognition |
| if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) && |
| (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) { |
| // FIXME: we should not assume that the first session is the active one and keep |
| // activity count per session. Same in startInput(). |
| ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0)); |
| if (index >= 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<AudioMix> mixes) |
| { |
| sp<HwModule> 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<AudioPolicyMix> policyMix = new AudioPolicyMix(); |
| policyMix->mMix = mixes[i]; |
| mPolicyMixes.add(address, policyMix); |
| if (mixes[i].mMixType == MIX_TYPE_PLAYERS) { |
| setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, |
| AUDIO_POLICY_DEVICE_STATE_AVAILABLE, |
| address.string()); |
| } else { |
| setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, |
| AUDIO_POLICY_DEVICE_STATE_AVAILABLE, |
| address.string()); |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes) |
| { |
| sp<HwModule> 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) |
| { |
| setDeviceConnectionStateInt(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) |
| { |
| setDeviceConnectionStateInt(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<IOProfile> 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::AudioOutputDescriptor> AudioPolicyManager::getOutputFromId( |
| audio_port_handle_t id) const |
| { |
| sp<AudioOutputDescriptor> outputDesc = NULL; |
| for (size_t i = 0; i < mOutputs.size(); i++) { |
| outputDesc = mOutputs.valueAt(i); |
| if (outputDesc->mId == id) { |
| break; |
| } |
| } |
| return outputDesc; |
| } |
| |
| sp<AudioPolicyManager::AudioInputDescriptor> AudioPolicyManager::getInputFromId( |
| audio_port_handle_t id) const |
| { |
| sp<AudioInputDescriptor> inputDesc = NULL; |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| inputDesc = mInputs.valueAt(i); |
| if (inputDesc->mId == id) { |
| break; |
| } |
| } |
| return inputDesc; |
| } |
| |
| sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleForDevice( |
| audio_devices_t device) const |
| { |
| sp <HwModule> 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::HwModule> AudioPolicyManager::getModuleFromName(const char *name) const |
| { |
| sp <HwModule> 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<AudioOutputDescriptor> 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<AudioPatch> 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<AudioOutputDescriptor> 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<DeviceDescriptor> 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<AudioInputDescriptor> 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<DeviceDescriptor> 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<DeviceDescriptor> 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<DeviceDescriptor> 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<audio_io_handle_t> 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<AudioOutputDescriptor> 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<AudioPatch> 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<AudioOutputDescriptor> 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<AudioInputDescriptor> 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> audioPortConfig; |
| if (config->type == AUDIO_PORT_TYPE_MIX) { |
| if (config->role == AUDIO_PORT_ROLE_SOURCE) { |
| sp<AudioOutputDescriptor> 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<AudioInputDescriptor> 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<DeviceDescriptor> 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<AudioPatch> 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 = getDeviceAndMixForInputSource(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<AudioPatch>& 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<IOProfile> 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<AudioOutputDescriptor> 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<IOProfile> 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<AudioInputDescriptor> 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<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput); |
| mpClientInterface->closeOutput(mPrimaryOutput); |
| |
| audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; |
| |
| mOutputs.removeItem(mPrimaryOutput); |
| |
| sp<AudioOutputDescriptor> 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<AudioOutputDescriptor> outputDesc) |
| { |
| outputDesc->mIoHandle = output; |
| outputDesc->mId = nextUniqueId(); |
| mOutputs.add(output, outputDesc); |
| nextAudioPortGeneration(); |
| } |
| |
| void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc) |
| { |
| inputDesc->mIoHandle = input; |
| inputDesc->mId = nextUniqueId(); |
| mInputs.add(input, inputDesc); |
| nextAudioPortGeneration(); |
| } |
| |
| void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/, |
| const audio_devices_t device /*in*/, |
| const String8 address /*in*/, |
| SortedVector<audio_io_handle_t>& outputs /*out*/) { |
| sp<DeviceDescriptor> devDesc = |
| desc->mProfile->mSupportedDevices.getDevice(device, address); |
| if (devDesc != 0) { |
| ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s", |
| desc->mIoHandle, address.string()); |
| outputs.add(desc->mIoHandle); |
| } |
| } |
| |
| status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> devDesc, |
| audio_policy_dev_state_t state, |
| SortedVector<audio_io_handle_t>& outputs, |
| const String8 address) |
| { |
| audio_devices_t device = devDesc->mDeviceType; |
| sp<AudioOutputDescriptor> 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, device, address, outputs); |
| } |
| } |
| } |
| // then look for output profiles that can be routed to this device |
| SortedVector< sp<IOProfile> > 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<IOProfile> 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<IOProfile> 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<AudioOutputDescriptor> 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, device, 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<IOProfile> 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<audio_io_handle_t>& inputs, |
| const String8 address) |
| { |
| sp<AudioInputDescriptor> 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<IOProfile> > 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<IOProfile> 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<IOProfile> 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<IOProfile> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i); |
| if (dupOutputDesc->isDuplicated() && |
| (dupOutputDesc->mOutput1 == outputDesc || |
| dupOutputDesc->mOutput2 == outputDesc)) { |
| sp<AudioOutputDescriptor> 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<AudioPatch> 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<AudioInputDescriptor> 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<AudioPatch> patchDesc = mAudioPatches.valueAt(index); |
| status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0); |
| mAudioPatches.removeItemsAt(index); |
| mpClientInterface->onAudioPatchListUpdate(); |
| } |
| |
| mpClientInterface->closeInput(input); |
| mInputs.removeItem(input); |
| } |
| |
| SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device, |
| DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs) |
| { |
| SortedVector<audio_io_handle_t> 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<audio_io_handle_t>& outputs1, |
| SortedVector<audio_io_handle_t>& 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<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); |
| SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs); |
| |
| // also take into account external policy-related changes: add all outputs which are |
| // associated with policies in the "before" and "after" output vectors |
| ALOGVV("checkOutputForStrategy(): policy related outputs"); |
| for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) { |
| const sp<AudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i); |
| if (desc != 0 && desc->mPolicyMix != NULL) { |
| srcOutputs.add(desc->mIoHandle); |
| ALOGVV(" previous outputs: adding %d", desc->mIoHandle); |
| } |
| } |
| for (size_t i = 0 ; i < mOutputs.size() ; i++) { |
| const sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); |
| if (desc != 0 && desc->mPolicyMix != NULL) { |
| dstOutputs.add(desc->mIoHandle); |
| ALOGVV(" new outputs: adding %d", desc->mIoHandle); |
| } |
| } |
| |
| 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<AudioOutputDescriptor> 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<audio_io_handle_t> moved; |
| for (size_t i = 0; i < mEffects.size(); i++) { |
| sp<EffectDescriptor> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); |
| |
| ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); |
| if (index >= 0) { |
| sp<AudioPatch> 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<AudioInputDescriptor> inputDesc = mInputs.valueFor(input); |
| |
| ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle); |
| if (index >= 0) { |
| sp<AudioPatch> 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 = getDeviceAndMixForInputSource(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<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs); |
| for (size_t i = 0; i < outputs.size(); i++) { |
| sp<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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 = |
| getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); |
| sp<AudioOutputDescriptor> 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 (!isInCall()) { |
| 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<AudioInputDescriptor> 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<AudioOutputDescriptor> 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<AudioInputDescriptor> 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<AudioInputDescriptor> 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::IOProfile> 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<IOProfile> 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::getDeviceAndMixForInputSource(audio_source_t inputSource, |
| AudioMix **policyMix) |
| { |
| 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; |
| } |
| } |
| } |
| |
| return getDeviceForInputSource(inputSource); |
| } |
| |
| audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource) |
| { |
| uint32_t device = AUDIO_DEVICE_NONE; |
| audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & |
| ~AUDIO_DEVICE_BIT_IN; |
| |
| 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 ((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_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<AudioInputDescriptor> 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<AudioInputDescriptor> desc = mInputs.valueAt(i); |
| if (desc->mRefCount > 0) { |
| 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<AudioOutputDescriptor> 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<IOProfile>& 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<AudioOutputDescriptor> 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<IOProfile>& 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<IOProfile> 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<IOProfile> 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<DeviceDescriptor> 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<IOProfile> 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<DeviceDescriptor> 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<IOProfile> 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<DeviceDescriptor> 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<HwModule>& 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<AudioPort> 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<AudioGain> 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<AudioGain> 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> 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<HwModule>& 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) |
| { |
| } |
| |
| bool AudioPolicyManager::DeviceDescriptor::equals(const sp<DeviceDescriptor>& 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::DeviceDescriptor::loadGains(cnode *root) |
| { |
| AudioPort::loadGains(root); |
| if (mGains.size() > 0) { |
| mGains[0]->getDefaultConfig(&mGain); |
| } |
| } |
| |
| |
| 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<DeviceDescriptor>& 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<DeviceDescriptor>& 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<DeviceDescriptor>& 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<DeviceDescriptor> 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<DeviceDescriptor> deviceDesc = |
| declaredDevices.getDeviceFromName(String8(devName)); |
| if (deviceDesc != 0) { |
| add(deviceDesc); |
| } |
| } |
| } |
| devName = strtok(NULL, "|"); |
| } |
| } |
| |
| sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDevice( |
| audio_devices_t type, String8 address) const |
| { |
| sp<DeviceDescriptor> 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::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromId( |
| audio_port_handle_t id) const |
| { |
| sp<DeviceDescriptor> 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; |
| bool isOutput = audio_is_output_devices(type); |
| type &= ~AUDIO_DEVICE_BIT_IN; |
| for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) { |
| bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType); |
| audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN; |
| if ((isOutput == curIsOutput) && ((type & curType) != 0)) { |
| devices.add(itemAt(i)); |
| type &= ~curType; |
| 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::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromName( |
| const String8& name) const |
| { |
| sp<DeviceDescriptor> 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<HwModule> 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<HwModule>& 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<HwModule> module; |
| sp<IOProfile> profile; |
| sp<DeviceDescriptor> 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; |
| } |
| if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { |
| return AUDIO_STREAM_TTS; |
| } |
| |
| // 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 |
