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Diffstat (limited to 'services/audiopolicy/managerdefault/AudioPolicyManager.cpp')
| -rw-r--r-- | services/audiopolicy/managerdefault/AudioPolicyManager.cpp | 4561 |
1 files changed, 4561 insertions, 0 deletions
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp new file mode 100644 index 0000000..797a2b4 --- /dev/null +++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp @@ -0,0 +1,4561 @@ +/* + * 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 "APM::AudioPolicyManager" +//#define LOG_NDEBUG 0 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +#include <inttypes.h> +#include <math.h> + +#include <AudioPolicyManagerInterface.h> +#include <AudioPolicyEngineInstance.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" +#include <ConfigParsingUtils.h> +#include <policy.h> + +namespace android { + +// ---------------------------------------------------------------------------- +// AudioPolicyInterface implementation +// ---------------------------------------------------------------------------- + +status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name) +{ + return setDeviceConnectionStateInt(device, state, device_address, device_name); +} + +status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name) +{ + ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s", +- device, state, device_address, device_name); + + // 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 = + mHwModules.getDeviceDescriptor(device, device_address, device_name); + + // 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 = mHwModules.getModuleForDevice(device); + if (module == 0) { + ALOGD("setDeviceConnectionState() could not find HW module for device %08x", + device); + mAvailableOutputDevices.remove(devDesc); + return INVALID_OPERATION; + } + mAvailableOutputDevices[index]->attach(module); + } else { + return NO_MEMORY; + } + + if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) { + mAvailableOutputDevices.remove(devDesc); + return INVALID_OPERATION; + } + // Propagate device availability to Engine + mEngine->setDeviceConnectionState(devDesc, state); + + // 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()); + + // Send 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); + + // Send 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); + + // Propagate device availability to Engine + mEngine->setDeviceConnectionState(devDesc, state); + } 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 (mEngine->getPhoneState() == 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 ((mEngine->getPhoneState() != 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() + && (!device_distinguishes_on_address(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 = mHwModules.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]->attach(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()); + + // Propagate device availability to Engine + mEngine->setDeviceConnectionState(devDesc, state); + } 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); + + // Propagate device availability to Engine + mEngine->setDeviceConnectionState(devDesc, state); + } break; + + default: + ALOGE("setDeviceConnectionState() invalid state: %x", state); + return BAD_VALUE; + } + + closeAllInputs(); + + if (mEngine->getPhoneState() == 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 = mHwModules.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; + } + return deviceVector->getDeviceConnectionState(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(&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(&patch, mUidCached); + mCallTxPatch->mAfPatchHandle = afPatchHandle; + mCallTxPatch->mUid = mUidCached; + } + } +} + +void AudioPolicyManager::setPhoneState(audio_mode_t state) +{ + ALOGV("setPhoneState() state %d", state); + // store previous phone state for management of sonification strategy below + int oldState = mEngine->getPhoneState(); + + if (mEngine->setPhoneState(state) != NO_ERROR) { + ALOGW("setPhoneState() invalid or same state %d", state); + return; + } + /// Opens: can these line be executed after the switch of volume curves??? + // if 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); + } + + /** + * Switching to or from incall state or switching between telephony and VoIP lead to force + * routing command. + */ + bool force = ((is_state_in_call(oldState) != is_state_in_call(state)) + || (is_state_in_call(state) && (state != oldState))); + + // 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 ((isStrategyActive(desc, STRATEGY_MEDIA, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime) || + isStrategyActive(desc, 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; + } +} + +audio_mode_t AudioPolicyManager::getPhoneState() { + return mEngine->getPhoneState(); +} + +void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) +{ + ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState()); + + if (mEngine->setForceUse(usage, config) != NO_ERROR) { + ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage); + return; + } + bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) || + (usage == AUDIO_POLICY_FORCE_FOR_DOCK) || + (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM); + + // check for device and output changes triggered by new force usage + checkA2dpSuspend(); + checkOutputForAllStrategies(); + updateDevicesAndOutputs(); + if (mEngine->getPhoneState() == 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 ((mEngine->getPhoneState() != 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 = mInputs.getActiveInput(); + if (activeInput != 0) { + setInputDevice(activeInput, getNewInputDevice(activeInput)); + } + +} + +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<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); + } + sp<AudioOutputDescriptor> desc; + if (mPolicyMixes.getOutputForAttr(attributes, desc) == NO_ERROR) { + ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr"); + if (!audio_is_linear_pcm(format)) { + return BAD_VALUE; + } + *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) || + !mEffects.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; + } + // 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 (mOutputs.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, + "remote-submix"); + } + + // 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, + "remote-submix"); + } + + 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); + removeOutput(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) { + status_t ret = mPolicyMixes.getInputMixForAttr(*attr, policyMix); + if (ret != NO_ERROR) { + return ret; + } + *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; + device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; + address = String8(attr->tags + strlen("addr=")); + } 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 (!is_virtual_input_device(inputDesc->mDevice)) { + + // for a non-virtual input device, check if there is another (non-virtual) active input + audio_io_handle_t activeInput = mInputs.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 (mInputs.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, "remote-submix"); + } + } + } + + 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, "remote-submix"); + } + } + + resetInputDevice(input); + + if (mInputs.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); + mEngine->initStreamVolume(stream, indexMin, indexMax); + //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now + if (stream == AUDIO_STREAM_MUSIC) { + mEngine->initStreamVolume(AUDIO_STREAM_ACCESSIBILITY, indexMin, indexMax); + } +} + +status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, + int index, + audio_devices_t device) +{ + + if ((index < mStreams[stream].getVolumeIndexMin()) || + (index > mStreams[stream].getVolumeIndexMax())) { + return BAD_VALUE; + } + if (!audio_is_output_device(device)) { + return BAD_VALUE; + } + + // Force max volume if stream cannot be muted + if (!mStreams.canBeMuted(stream)) index = mStreams[stream].getVolumeIndexMax(); + + 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.clearCurrentVolumeIndex(stream); + } + mStreams.addCurrentVolumeIndex(stream, 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.addCurrentVolumeIndex(AUDIO_STREAM_ACCESSIBILITY, 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 = Volume::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 = Volume::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; + } + } + return mEffects.registerEffect(desc, io, strategy, session, id); +} + +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; + + if (mPolicyMixes.registerMix(address, mixes[i]) != NO_ERROR) { + 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); + + if (mixes[i].mMixType == MIX_TYPE_PLAYERS) { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + address.string(), "remote-submix"); + } else { + setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + address.string(), "remote-submix"); + } + } + 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; + + if (mPolicyMixes.unregisterMix(address) != NO_ERROR) { + continue; + } + + 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(), "remote-submix"); + } + + 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(), "remote-submix"); + } + 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", mEngine->getPhoneState()); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for communications %d\n", + mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for media %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA)); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for record %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD)); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for dock %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK)); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for system %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM)); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n", + mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO)); + result.append(buffer); + write(fd, result.string(), result.size()); + + mAvailableOutputDevices.dump(fd, String8("output")); + mAvailableInputDevices.dump(fd, String8("input")); + mHwModules.dump(fd); + mOutputs.dump(fd); + mInputs.dump(fd); + mStreams.dump(fd); + mEffects.dump(fd); + mAudioPatches.dump(fd); + + 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 (mEffects.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; +} + +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 = mOutputs.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->type(), + 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->type()); + 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 = mInputs.getInputFromId(patch->sinks[0].id); + if (inputDesc == NULL) { + return BAD_VALUE; + } + if (patchDesc != 0) { + if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) { + return BAD_VALUE; + } + } + sp<DeviceDescriptor> devDesc = + mAvailableInputDevices.getDeviceFromId(patch->sources[0].id); + if (devDesc == 0) { + return BAD_VALUE; + } + + if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(), + 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->type(), inputDesc->mIoHandle); + setInputDevice(inputDesc->mIoHandle, devDesc->type(), 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->type(), 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(&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 = mOutputs.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 = mInputs.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 (generation == NULL) { + return BAD_VALUE; + } + *generation = curAudioPortGeneration(); + return mAudioPatches.listAudioPatches(num_patches, patches); +} + +status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config) +{ + ALOGV("setAudioPortConfig()"); + + if (config == NULL) { + return BAD_VALUE; + } + ALOGV("setAudioPortConfig() on port handle %d", config->id); + // Only support gain configuration for now + if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) { + return INVALID_OPERATION; + } + + sp<AudioPortConfig> audioPortConfig; + if (config->type == AUDIO_PORT_TYPE_MIX) { + if (config->role == AUDIO_PORT_ROLE_SOURCE) { + sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id); + if (outputDesc == NULL) { + return BAD_VALUE; + } + ALOG_ASSERT(!outputDesc->isDuplicated(), + "setAudioPortConfig() called on duplicated output %d", + outputDesc->mIoHandle); + audioPortConfig = outputDesc; + } else if (config->role == AUDIO_PORT_ROLE_SINK) { + sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(config->id); + if (inputDesc == NULL) { + return BAD_VALUE; + } + audioPortConfig = inputDesc; + } else { + return BAD_VALUE; + } + } else if (config->type == AUDIO_PORT_TYPE_DEVICE) { + sp<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); + + return mSoundTriggerSessions.acquireSession(*session, *ioHandle); +} + +// ---------------------------------------------------------------------------- +// AudioPolicyManager +// ---------------------------------------------------------------------------- +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), + mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), + mA2dpSuspended(false), + mSpeakerDrcEnabled(false), + mAudioPortGeneration(1), + mBeaconMuteRefCount(0), + mBeaconPlayingRefCount(0), + mBeaconMuted(false) +{ + audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance(); + if (!engineInstance) { + ALOGE("%s: Could not get an instance of policy engine", __FUNCTION__); + return; + } + // Retrieve the Policy Manager Interface + mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>(); + if (mEngine == NULL) { + ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__); + return; + } + mEngine->setObserver(this); + status_t status = mEngine->initCheck(); + ALOG_ASSERT(status == NO_ERROR, "Policy engine not initialized(err=%d)", status); + + mUidCached = getuid(); + mpClientInterface = clientInterface; + + mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER); + if (ConfigParsingUtils::loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE, + mHwModules, mAvailableInputDevices, mAvailableOutputDevices, + mDefaultOutputDevice, mSpeakerDrcEnabled) != NO_ERROR) { + if (ConfigParsingUtils::loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE, + mHwModules, mAvailableInputDevices, mAvailableOutputDevices, + mDefaultOutputDevice, mSpeakerDrcEnabled) != 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 (since conditionned by Speaker Drc Enabling) + mEngine->initializeVolumeCurves(mSpeakerDrcEnabled); + + // 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->type()) != AUDIO_DEVICE_NONE) { + profileType = mDefaultOutputDevice->type(); + } 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]->type(); + 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]->type(); + 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]->isAttached()) { + mAvailableOutputDevices[index]->attach(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]->type(); + 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]->type(); + 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]->isAttached()) { + mAvailableInputDevices[index]->attach(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]->isAttached()) { + ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->type()); + mAvailableOutputDevices.remove(mAvailableOutputDevices[i]); + continue; + } + // The device is now validated and can be appended to the available devices of the engine + mEngine->setDeviceConnectionState(mAvailableOutputDevices[i], + AUDIO_POLICY_DEVICE_STATE_AVAILABLE); + i++; + } + for (size_t i = 0; i < mAvailableInputDevices.size();) { + if (!mAvailableInputDevices[i]->isAttached()) { + ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type()); + mAvailableInputDevices.remove(mAvailableInputDevices[i]); + continue; + } + // The device is now validated and can be appended to the available devices of the engine + mEngine->setDeviceConnectionState(mAvailableInputDevices[i], + AUDIO_POLICY_DEVICE_STATE_AVAILABLE); + i++; + } + // make sure default device is reachable + if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) { + ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type()); + } + + 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; + + removeOutput(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->setIoHandle(output); + mOutputs.add(output, outputDesc); + nextAudioPortGeneration(); +} + +void AudioPolicyManager::removeOutput(audio_io_handle_t output) +{ + mOutputs.removeItem(output); +} + +void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc) +{ + inputDesc->setIoHandle(input); + 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->type(); + 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 (!device_distinguishes_on_address(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 (!device_distinguishes_on_address(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 (device_distinguishes_on_address(device) && address != "0") { + sp<AudioPolicyMix> policyMix; + if (mPolicyMixes.getAudioPolicyMix(address, policyMix) != NO_ERROR) { + ALOGE("checkOutputsForDevice() cannot find policy for address %s", + address.string()); + } + policyMix->setOutput(desc); + desc->mPolicyMix = &(policyMix->getMix()); + + } 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); + removeOutput(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 (device_distinguishes_on_address(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 (device_distinguishes_on_address(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 (!device_distinguishes_on_address(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; + } + mPolicyMixes.closeOutput(outputDesc); + + // 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); + removeOutput(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); + removeOutput(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, + AudioOutputCollection 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 (isStrategyActive(desc, 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 (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) + checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); + checkOutputForStrategy(STRATEGY_PHONE); + if (mEngine->getForceUse(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); +} + +void AudioPolicyManager::checkA2dpSuspend() +{ + audio_io_handle_t a2dpOutput = mOutputs.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 || + ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) != AUDIO_POLICY_FORCE_BT_SCO) && + (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) != AUDIO_POLICY_FORCE_BT_SCO))) && + ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) && + (mEngine->getPhoneState() != AUDIO_MODE_RINGTONE))) { + + mpClientInterface->restoreOutput(a2dpOutput); + mA2dpSuspended = false; + } + } else { + if ((isScoConnected && + ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == AUDIO_POLICY_FORCE_BT_SCO) || + (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO))) || + ((mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) || + (mEngine->getPhoneState() == 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 (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) && + mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { + device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); + } else if (isInCall() || + isStrategyActive(outputDesc, STRATEGY_PHONE)) { + device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); + } else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) { + device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); + } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); + } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); + } else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) { + device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache); + } else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) { + device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); + } else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) { + device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); + } else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) { + device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache); + } else if (isStrategyActive(outputDesc, 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; + 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 (isStrategyActive(outputDesc, 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; +} + +routing_strategy AudioPolicyManager::getStrategy(audio_stream_type_t stream) const +{ + ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH"); + return mEngine->getStrategyForStream(stream); +} + +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 + return static_cast<uint32_t>(mEngine->getStrategyForUsage(attr->usage)); +} + +void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) { + switch(stream) { + case AUDIO_STREAM_MUSIC: + checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); + updateDevicesAndOutputs(); + break; + default: + break; + } +} + +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) +{ + if (fromCache) { + ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", + strategy, mDeviceForStrategy[strategy]); + return mDeviceForStrategy[strategy]; + } + return mEngine->getDeviceForStrategy(strategy); +} + +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 (isStrategyActive(desc, (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 (isStrategyActive(outputDesc, (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(&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 (!is_virtual_input_device(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(&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<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; + audio_devices_t selectedDeviceFromMix = + mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix); + + if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) { + return selectedDeviceFromMix; + } + return getDeviceForInputSource(inputSource); +} + +audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource) +{ + return mEngine->getDeviceForInputSource(inputSource); +} + +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); + + if (device == AUDIO_DEVICE_NONE) { + device = outputDesc->device(); + } + volume = mEngine->volIndexToAmpl(Volume::getDeviceCategory(device), stream, 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) && + (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) && + mStreams.canBeMuted(stream)) { + 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; + } + audio_policy_forced_cfg_t forceUseForComm = + mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION); + // do not change in call volume if bluetooth is connected and vice versa + if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) || + (stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) { + ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", + stream, forceUseForComm); + 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].getVolumeIndexMax(); + } 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) +{ + const 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.canBeMuted() && + ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) || + (mEngine->getForceUse(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(); + } + } + } + } +} + +void AudioPolicyManager::defaultAudioPolicyConfig(void) +{ + sp<HwModule> module; + sp<IOProfile> profile; + sp<DeviceDescriptor> defaultInputDevice = + new DeviceDescriptor(String8("builtin-mic"), 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; +} + +bool AudioPolicyManager::isStrategyActive(const sp<AudioOutputDescriptor> outputDesc, + 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)) && + outputDesc->isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) +{ + return mEngine->getForceUse(usage); +} + +bool AudioPolicyManager::isInCall() +{ + return isStateInCall(mEngine->getPhoneState()); +} + +bool AudioPolicyManager::isStateInCall(int state) +{ + return is_state_in_call(state); +} + +}; // namespace android |