From e552edb33fb5873179ae0a46d9579d1103eb13c6 Mon Sep 17 00:00:00 2001 From: Eric Laurent Date: Mon, 10 Mar 2014 17:42:56 -0700 Subject: audio policy: import audio policy manager from hardware legacy Import AudioPolicyManagerBase from hardware/libhardware_legacy to prepare move from android_audio_legacy name space. Change-Id: I5d6682ccd2bfdeefbf2f6f81a557480a76aaf4fc --- services/audiopolicy/AudioPolicyInterface.h | 261 ++ services/audiopolicy/AudioPolicyManagerBase.cpp | 4091 +++++++++++++++++++++++ services/audiopolicy/AudioPolicyManagerBase.h | 587 ++++ 3 files changed, 4939 insertions(+) create mode 100644 services/audiopolicy/AudioPolicyInterface.h create mode 100644 services/audiopolicy/AudioPolicyManagerBase.cpp create mode 100644 services/audiopolicy/AudioPolicyManagerBase.h (limited to 'services/audiopolicy') diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h new file mode 100644 index 0000000..da03ee3 --- /dev/null +++ b/services/audiopolicy/AudioPolicyInterface.h @@ -0,0 +1,261 @@ +/* + * 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. + */ + +#ifndef ANDROID_AUDIOPOLICYINTERFACE_H +#define ANDROID_AUDIOPOLICYINTERFACE_H + +#include +#include +#include + +#include +#include + +namespace android_audio_legacy { + using android::Vector; + using android::String8; + using android::ToneGenerator; + +// ---------------------------------------------------------------------------- + +// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces +// between the platform specific audio policy manager and Android generic audio policy manager. +// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class. +// This implementation makes use of the AudioPolicyClientInterface to control the activity and +// configuration of audio input and output streams. +// +// The platform specific audio policy manager is in charge of the audio routing and volume control +// policies for a given platform. +// The main roles of this module are: +// - keep track of current system state (removable device connections, phone state, user requests...). +// System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface. +// - process getOutput() queries received when AudioTrack objects are created: Those queries +// return a handler on an output that has been selected, configured and opened by the audio policy manager and that +// must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method. +// When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide +// to close or reconfigure the output depending on other streams using this output and current system state. +// - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs. +// - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value +// applicable to each output as a function of platform specific settings and current output route (destination device). It +// also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries). +// +// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so) +// and is linked with libaudioflinger.so + + +// Audio Policy Manager Interface +class AudioPolicyInterface +{ + +public: + virtual ~AudioPolicyInterface() {} + // + // configuration functions + // + + // indicate a change in device connection status + virtual status_t setDeviceConnectionState(audio_devices_t device, + AudioSystem::device_connection_state state, + const char *device_address) = 0; + // retrieve a device connection status + virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device, + const char *device_address) = 0; + // indicate a change in phone state. Valid phones states are defined by AudioSystem::audio_mode + virtual void setPhoneState(int state) = 0; + // force using a specific device category for the specified usage + virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0; + // retrieve current device category forced for a given usage + virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0; + // set a system property (e.g. camera sound always audible) + virtual void setSystemProperty(const char* property, const char* value) = 0; + // check proper initialization + virtual status_t initCheck() = 0; + + // + // Audio routing query functions + // + + // request an output appropriate for playback of the supplied stream type and parameters + virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::output_flags flags, + const audio_offload_info_t *offloadInfo) = 0; + // indicates to the audio policy manager that the output starts being used by corresponding stream. + virtual status_t startOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0) = 0; + // indicates to the audio policy manager that the output stops being used by corresponding stream. + virtual status_t stopOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0) = 0; + // releases the output. + virtual void releaseOutput(audio_io_handle_t output) = 0; + + // request an input appropriate for record from the supplied device with supplied parameters. + virtual audio_io_handle_t getInput(int inputSource, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::audio_in_acoustics acoustics) = 0; + // indicates to the audio policy manager that the input starts being used. + virtual status_t startInput(audio_io_handle_t input) = 0; + // indicates to the audio policy manager that the input stops being used. + virtual status_t stopInput(audio_io_handle_t input) = 0; + // releases the input. + virtual void releaseInput(audio_io_handle_t input) = 0; + + // + // volume control functions + // + + // initialises stream volume conversion parameters by specifying volume index range. + virtual void initStreamVolume(AudioSystem::stream_type stream, + int indexMin, + int indexMax) = 0; + + // sets the new stream volume at a level corresponding to the supplied index for the + // supplied device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means + // setting volume for all devices + virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, + int index, + audio_devices_t device) = 0; + + // retrieve current volume index for the specified stream and the + // specified device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means + // querying the volume of the active device. + virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, + int *index, + audio_devices_t device) = 0; + + // return the strategy corresponding to a given stream type + virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream) = 0; + + // return the enabled output devices for the given stream type + virtual audio_devices_t getDevicesForStream(AudioSystem::stream_type stream) = 0; + + // Audio effect management + virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc) = 0; + virtual status_t registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) = 0; + virtual status_t unregisterEffect(int id) = 0; + virtual status_t setEffectEnabled(int id, bool enabled) = 0; + + virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const = 0; + virtual bool isStreamActiveRemotely(int stream, uint32_t inPastMs = 0) const = 0; + virtual bool isSourceActive(audio_source_t source) const = 0; + + //dump state + virtual status_t dump(int fd) = 0; + + virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo) = 0; +}; + + +// Audio Policy client Interface +class AudioPolicyClientInterface +{ +public: + virtual ~AudioPolicyClientInterface() {} + + // + // Audio HW module functions + // + + // loads a HW module. + virtual audio_module_handle_t loadHwModule(const char *name) = 0; + + // + // Audio output Control functions + // + + // opens an audio output with the requested parameters. The parameter values can indicate to use the default values + // in case the audio policy manager has no specific requirements for the output being opened. + // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream. + // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly. + virtual audio_io_handle_t openOutput(audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask, + uint32_t *pLatencyMs, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo = NULL) = 0; + // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by + // a special mixer thread in the AudioFlinger. + virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0; + // closes the output stream + virtual status_t closeOutput(audio_io_handle_t output) = 0; + // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in + // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded. + virtual status_t suspendOutput(audio_io_handle_t output) = 0; + // restores a suspended output. + virtual status_t restoreOutput(audio_io_handle_t output) = 0; + + // + // Audio input Control functions + // + + // opens an audio input + virtual audio_io_handle_t openInput(audio_module_handle_t module, + audio_devices_t *pDevices, + uint32_t *pSamplingRate, + audio_format_t *pFormat, + audio_channel_mask_t *pChannelMask) = 0; + // closes an audio input + virtual status_t closeInput(audio_io_handle_t input) = 0; + // + // misc control functions + // + + // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes + // for each output (destination device) it is attached to. + virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0; + + // invalidate a stream type, causing a reroute to an unspecified new output + virtual status_t invalidateStream(AudioSystem::stream_type stream) = 0; + + // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface. + virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0) = 0; + // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager. + virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0; + + // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing + // over a telephony device during a phone call. + virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream) = 0; + virtual status_t stopTone() = 0; + + // set down link audio volume. + virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0; + + // move effect to the specified output + virtual status_t moveEffects(int session, + audio_io_handle_t srcOutput, + audio_io_handle_t dstOutput) = 0; + +}; + +extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface); +extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface); + + +}; // namespace android + +#endif // ANDROID_AUDIOPOLICYINTERFACE_H diff --git a/services/audiopolicy/AudioPolicyManagerBase.cpp b/services/audiopolicy/AudioPolicyManagerBase.cpp new file mode 100644 index 0000000..6f58cf7 --- /dev/null +++ b/services/audiopolicy/AudioPolicyManagerBase.cpp @@ -0,0 +1,4091 @@ +/* + * 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 "AudioPolicyManagerBase" +//#define LOG_NDEBUG 0 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +// A device mask for all audio input devices that are considered "virtual" when evaluating +// active inputs in getActiveInput() +#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX +// A device mask for all audio output devices that are considered "remote" when evaluating +// active output devices in isStreamActiveRemotely() +#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX + +#include +#include +#include +#include +#include +#include +#include + +namespace android_audio_legacy { + +// ---------------------------------------------------------------------------- +// AudioPolicyInterface implementation +// ---------------------------------------------------------------------------- + + +status_t AudioPolicyManagerBase::setDeviceConnectionState(audio_devices_t device, + AudioSystem::device_connection_state state, + const char *device_address) +{ + SortedVector outputs; + + ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address); + + // connect/disconnect only 1 device at a time + if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; + + if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) { + ALOGE("setDeviceConnectionState() invalid address: %s", device_address); + return BAD_VALUE; + } + + // handle output devices + if (audio_is_output_device(device)) { + + if (!mHasA2dp && audio_is_a2dp_device(device)) { + ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device); + return BAD_VALUE; + } + if (!mHasUsb && audio_is_usb_device(device)) { + ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device); + return BAD_VALUE; + } + if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) { + ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device); + return BAD_VALUE; + } + + // 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; + String8 paramStr; + switch (state) + { + // handle output device connection + case AudioSystem::DEVICE_STATE_AVAILABLE: + if (mAvailableOutputDevices & device) { + ALOGW("setDeviceConnectionState() device already connected: %x", device); + return INVALID_OPERATION; + } + ALOGV("setDeviceConnectionState() connecting device %x", device); + + if (mHasA2dp && audio_is_a2dp_device(device)) { + // handle A2DP device connection + AudioParameter param; + param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address)); + paramStr = param.toString(); + } else if (mHasUsb && audio_is_usb_device(device)) { + // handle USB device connection + paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN); + } + + if (checkOutputsForDevice(device, state, outputs, paramStr) != NO_ERROR) { + return INVALID_OPERATION; + } + ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs", + outputs.size()); + // register new device as available + mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device); + + if (mHasA2dp && audio_is_a2dp_device(device)) { + // handle A2DP device connection + mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); + mA2dpSuspended = false; + } else if (audio_is_bluetooth_sco_device(device)) { + // handle SCO device connection + mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN); + } else if (mHasUsb && audio_is_usb_device(device)) { + // handle USB device connection + mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN); + } + + break; + // handle output device disconnection + case AudioSystem::DEVICE_STATE_UNAVAILABLE: { + if (!(mAvailableOutputDevices & device)) { + ALOGW("setDeviceConnectionState() device not connected: %x", device); + return INVALID_OPERATION; + } + + ALOGV("setDeviceConnectionState() disconnecting device %x", device); + // remove device from available output devices + mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device); + + checkOutputsForDevice(device, state, outputs, paramStr); + if (mHasA2dp && audio_is_a2dp_device(device)) { + // handle A2DP device disconnection + mA2dpDeviceAddress = ""; + mA2dpSuspended = false; + } else if (audio_is_bluetooth_sco_device(device)) { + // handle SCO device disconnection + mScoDeviceAddress = ""; + } else if (mHasUsb && audio_is_usb_device(device)) { + // handle USB device disconnection + mUsbCardAndDevice = ""; + } + // not currently handling multiple simultaneous submixes: ignoring remote submix + // case and address + } break; + + default: + ALOGE("setDeviceConnectionState() invalid state: %x", state); + return BAD_VALUE; + } + + checkA2dpSuspend(); + checkOutputForAllStrategies(); + // outputs must be closed after checkOutputForAllStrategies() is executed + if (!outputs.isEmpty()) { + for (size_t i = 0; i < outputs.size(); i++) { + 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 == AudioSystem::DEVICE_STATE_UNAVAILABLE) || + (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) && + (desc->mDirectOpenCount == 0))) { + closeOutput(outputs[i]); + } + } + } + + updateDevicesAndOutputs(); + for (size_t i = 0; i < mOutputs.size(); i++) { + // 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. + setOutputDevice(mOutputs.keyAt(i), + getNewDevice(mOutputs.keyAt(i), true /*fromCache*/), + !mOutputs.valueAt(i)->isDuplicated(), + 0); + } + + if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO || + device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET || + device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + } else { + return NO_ERROR; + } + } + // handle input devices + if (audio_is_input_device(device)) { + + switch (state) + { + // handle input device connection + case AudioSystem::DEVICE_STATE_AVAILABLE: { + if (mAvailableInputDevices & device) { + ALOGW("setDeviceConnectionState() device already connected: %d", device); + return INVALID_OPERATION; + } + mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN); + } + break; + + // handle input device disconnection + case AudioSystem::DEVICE_STATE_UNAVAILABLE: { + if (!(mAvailableInputDevices & device)) { + ALOGW("setDeviceConnectionState() device not connected: %d", device); + return INVALID_OPERATION; + } + mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device); + } break; + + default: + ALOGE("setDeviceConnectionState() invalid state: %x", state); + return BAD_VALUE; + } + + audio_io_handle_t activeInput = getActiveInput(); + if (activeInput != 0) { + AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); + audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); + if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { + ALOGV("setDeviceConnectionState() changing device from %x to %x for input %d", + inputDesc->mDevice, newDevice, activeInput); + inputDesc->mDevice = newDevice; + AudioParameter param = AudioParameter(); + param.addInt(String8(AudioParameter::keyRouting), (int)newDevice); + mpClientInterface->setParameters(activeInput, param.toString()); + } + } + + return NO_ERROR; + } + + ALOGW("setDeviceConnectionState() invalid device: %x", device); + return BAD_VALUE; +} + +AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(audio_devices_t device, + const char *device_address) +{ + AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE; + String8 address = String8(device_address); + if (audio_is_output_device(device)) { + if (device & mAvailableOutputDevices) { + if (audio_is_a2dp_device(device) && + (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) { + return state; + } + if (audio_is_bluetooth_sco_device(device) && + address != "" && mScoDeviceAddress != address) { + return state; + } + if (audio_is_usb_device(device) && + (!mHasUsb || (address != "" && mUsbCardAndDevice != address))) { + ALOGE("getDeviceConnectionState() invalid device: %x", device); + return state; + } + if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) { + return state; + } + state = AudioSystem::DEVICE_STATE_AVAILABLE; + } + } else if (audio_is_input_device(device)) { + if (device & mAvailableInputDevices) { + state = AudioSystem::DEVICE_STATE_AVAILABLE; + } + } + + return state; +} + +void AudioPolicyManagerBase::setPhoneState(int state) +{ + ALOGV("setPhoneState() state %d", state); + audio_devices_t newDevice = AUDIO_DEVICE_NONE; + if (state < 0 || state >= AudioSystem::NUM_MODES) { + ALOGW("setPhoneState() invalid state %d", state); + return; + } + + if (state == mPhoneState ) { + ALOGW("setPhoneState() setting same state %d", state); + return; + } + + // if leaving call state, handle special case of active streams + // pertaining to sonification strategy see handleIncallSonification() + if (isInCall()) { + ALOGV("setPhoneState() in call state management: new state is %d", state); + for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { + handleIncallSonification(stream, false, true); + } + } + + // store previous phone state for management of sonification strategy below + int oldState = mPhoneState; + mPhoneState = state; + bool force = false; + + // are we entering or starting a call + if (!isStateInCall(oldState) && isStateInCall(state)) { + ALOGV(" Entering call in setPhoneState()"); + // force routing command to audio hardware when starting a call + // even if no device change is needed + force = true; + for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { + mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = + sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]; + } + } else if (isStateInCall(oldState) && !isStateInCall(state)) { + ALOGV(" Exiting call in setPhoneState()"); + // force routing command to audio hardware when exiting a call + // even if no device change is needed + force = true; + for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { + mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = + sVolumeProfiles[AUDIO_STREAM_DTMF][j]; + } + } else if (isStateInCall(state) && (state != oldState)) { + ALOGV(" Switching between telephony and VoIP in setPhoneState()"); + // force routing command to audio hardware when switching between telephony and VoIP + // even if no device change is needed + force = true; + } + + // check for device and output changes triggered by new phone state + newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/); + checkA2dpSuspend(); + checkOutputForAllStrategies(); + updateDevicesAndOutputs(); + + AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); + + // force routing command to audio hardware when ending call + // even if no device change is needed + if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) { + newDevice = hwOutputDesc->device(); + } + + int delayMs = 0; + if (isStateInCall(state)) { + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueAt(i); + // mute media and sonification strategies and delay device switch by the largest + // latency of any output where either strategy is active. + // This avoid sending the ring tone or music tail into the earpiece or headset. + if ((desc->isStrategyActive(STRATEGY_MEDIA, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime) || + desc->isStrategyActive(STRATEGY_SONIFICATION, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime)) && + (delayMs < (int)desc->mLatency*2)) { + delayMs = desc->mLatency*2; + } + setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); + setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, + getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); + setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); + setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, + getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); + } + } + + // change routing is necessary + setOutputDevice(mPrimaryOutput, newDevice, force, delayMs); + + // 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 < AudioSystem::NUM_STREAM_TYPES; stream++) { + handleIncallSonification(stream, true, true); + } + } + + // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE + if (state == AudioSystem::MODE_RINGTONE && + isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) { + mLimitRingtoneVolume = true; + } else { + mLimitRingtoneVolume = false; + } +} + +void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) +{ + ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState); + + bool forceVolumeReeval = false; + switch(usage) { + case AudioSystem::FOR_COMMUNICATION: + if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO && + config != AudioSystem::FORCE_NONE) { + ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); + return; + } + forceVolumeReeval = true; + mForceUse[usage] = config; + break; + case AudioSystem::FOR_MEDIA: + if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP && + config != AudioSystem::FORCE_WIRED_ACCESSORY && + config != AudioSystem::FORCE_ANALOG_DOCK && + config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE && + config != AudioSystem::FORCE_NO_BT_A2DP) { + ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config); + return; + } + mForceUse[usage] = config; + break; + case AudioSystem::FOR_RECORD: + if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY && + config != AudioSystem::FORCE_NONE) { + ALOGW("setForceUse() invalid config %d for FOR_RECORD", config); + return; + } + mForceUse[usage] = config; + break; + case AudioSystem::FOR_DOCK: + if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK && + config != AudioSystem::FORCE_BT_DESK_DOCK && + config != AudioSystem::FORCE_WIRED_ACCESSORY && + config != AudioSystem::FORCE_ANALOG_DOCK && + config != AudioSystem::FORCE_DIGITAL_DOCK) { + ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); + } + forceVolumeReeval = true; + mForceUse[usage] = config; + break; + case AudioSystem::FOR_SYSTEM: + if (config != AudioSystem::FORCE_NONE && + config != AudioSystem::FORCE_SYSTEM_ENFORCED) { + ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); + } + forceVolumeReeval = true; + mForceUse[usage] = config; + break; + default: + ALOGW("setForceUse() invalid usage %d", usage); + break; + } + + // check for device and output changes triggered by new force usage + checkA2dpSuspend(); + checkOutputForAllStrategies(); + updateDevicesAndOutputs(); + for (size_t i = 0; i < mOutputs.size(); i++) { + audio_io_handle_t output = mOutputs.keyAt(i); + audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/); + setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); + if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { + applyStreamVolumes(output, newDevice, 0, true); + } + } + + audio_io_handle_t activeInput = getActiveInput(); + if (activeInput != 0) { + AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput); + audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); + if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { + ALOGV("setForceUse() changing device from %x to %x for input %d", + inputDesc->mDevice, newDevice, activeInput); + inputDesc->mDevice = newDevice; + AudioParameter param = AudioParameter(); + param.addInt(String8(AudioParameter::keyRouting), (int)newDevice); + mpClientInterface->setParameters(activeInput, param.toString()); + } + } + +} + +AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage) +{ + return mForceUse[usage]; +} + +void AudioPolicyManagerBase::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 +AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::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++) { + IOProfile *profile = mHwModules[i]->mOutputProfiles[j]; + if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { + if (profile->isCompatibleProfile(device, samplingRate, format, + channelMask, + AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) { + if (mAvailableOutputDevices & profile->mSupportedDevices) { + return mHwModules[i]->mOutputProfiles[j]; + } + } + } else { + if (profile->isCompatibleProfile(device, samplingRate, format, + channelMask, + AUDIO_OUTPUT_FLAG_DIRECT)) { + if (mAvailableOutputDevices & profile->mSupportedDevices) { + return mHwModules[i]->mOutputProfiles[j]; + } + } + } + } + } + return 0; +} + +audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::output_flags flags, + const audio_offload_info_t *offloadInfo) +{ + audio_io_handle_t output = 0; + uint32_t latency = 0; + routing_strategy strategy = getStrategy((AudioSystem::stream_type)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); + +#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"); + AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL); + outputDesc->mDevice = mTestDevice; + outputDesc->mSamplingRate = mTestSamplingRate; + outputDesc->mFormat = mTestFormat; + outputDesc->mChannelMask = mTestChannels; + outputDesc->mLatency = mTestLatencyMs; + outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0); + outputDesc->mRefCount[stream] = 0; + mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice, + &outputDesc->mSamplingRate, + &outputDesc->mFormat, + &outputDesc->mChannelMask, + &outputDesc->mLatency, + outputDesc->mFlags, + offloadInfo); + if (mTestOutputs[mCurOutput]) { + 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 = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT); + } + + // 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. + IOProfile *profile = NULL; + if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || + !isNonOffloadableEffectEnabled()) { + profile = getProfileForDirectOutput(device, + samplingRate, + format, + channelMask, + (audio_output_flags_t)flags); + } + + if (profile != NULL) { + AudioOutputDescriptor *outputDesc = NULL; + + for (size_t i = 0; i < mOutputs.size(); i++) { + 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->mId); + } + outputDesc = new AudioOutputDescriptor(profile); + outputDesc->mDevice = device; + outputDesc->mSamplingRate = samplingRate; + outputDesc->mFormat = format; + outputDesc->mChannelMask = channelMask; + outputDesc->mLatency = 0; + outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); + outputDesc->mRefCount[stream] = 0; + outputDesc->mStopTime[stream] = 0; + outputDesc->mDirectOpenCount = 1; + output = mpClientInterface->openOutput(profile->mModule->mHandle, + &outputDesc->mDevice, + &outputDesc->mSamplingRate, + &outputDesc->mFormat, + &outputDesc->mChannelMask, + &outputDesc->mLatency, + outputDesc->mFlags, + offloadInfo); + + // only accept an output with the requested parameters + if (output == 0 || + (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) || + (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) || + (channelMask != 0 && channelMask != outputDesc->mChannelMask)) { + 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 != 0) { + mpClientInterface->closeOutput(output); + } + delete outputDesc; + return 0; + } + 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); + return output; + } + + // ignoring channel mask due to downmix capability in mixer + + // open a non direct output + + // for non direct outputs, only PCM is supported + if (audio_is_linear_pcm(format)) { + // get which output is suitable for the specified stream. The actual + // routing change will happen when startOutput() will be called + SortedVector outputs = getOutputsForDevice(device, mOutputs); + + output = selectOutput(outputs, flags); + } + 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 AudioPolicyManagerBase::selectOutput(const SortedVector& outputs, + AudioSystem::output_flags flags) +{ + // 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++) { + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]); + if (!outputDesc->isDuplicated()) { + int commonFlags = (int)AudioSystem::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 AudioPolicyManagerBase::startOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int 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; + } + + 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) { + audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/); + routing_strategy strategy = getStrategy(stream); + bool shouldWait = (strategy == STRATEGY_SONIFICATION) || + (strategy == STRATEGY_SONIFICATION_RESPECTFUL); + uint32_t waitMs = 0; + bool force = false; + for (size_t i = 0; i < mOutputs.size(); i++) { + 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. + 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); + if (waitMs > muteWaitMs) { + usleep((waitMs - muteWaitMs) * 2 * 1000); + } + } + return NO_ERROR; +} + + +status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int 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; + } + + AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); + + // 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) { + outputDesc->mStopTime[stream] = systemTime(); + audio_devices_t newDevice = getNewDevice(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); + AudioOutputDescriptor *desc = mOutputs.valueAt(i); + if (curOutput != output && + desc->isActive() && + outputDesc->sharesHwModuleWith(desc) && + (newDevice != desc->device())) { + setOutputDevice(curOutput, + getNewDevice(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 AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output) +{ + 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) { + AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index); + if (outputDesc->isActive()) { + mpClientInterface->closeOutput(output); + delete mOutputs.valueAt(index); + mOutputs.removeItem(output); + mTestOutputs[testIndex] = 0; + } + return; + } +#endif //AUDIO_POLICY_TEST + + AudioOutputDescriptor *desc = mOutputs.valueAt(index); + if (desc->mFlags & AudioSystem::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); + } + } + } +} + + +audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::audio_in_acoustics acoustics) +{ + audio_io_handle_t input = 0; + audio_devices_t device = getDeviceForInputSource(inputSource); + + ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x", + inputSource, samplingRate, format, channelMask, acoustics); + + if (device == AUDIO_DEVICE_NONE) { + ALOGW("getInput() could not find device for inputSource %d", inputSource); + return 0; + } + + // 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; + } + + IOProfile *profile = getInputProfile(device, + samplingRate, + format, + channelMask); + if (profile == NULL) { + ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d, " + "channelMask %04x", + device, samplingRate, format, channelMask); + return 0; + } + + if (profile->mModule->mHandle == 0) { + ALOGE("getInput(): HW module %s not opened", profile->mModule->mName); + return 0; + } + + AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile); + + inputDesc->mInputSource = inputSource; + inputDesc->mDevice = device; + inputDesc->mSamplingRate = samplingRate; + inputDesc->mFormat = format; + inputDesc->mChannelMask = channelMask; + inputDesc->mRefCount = 0; + input = mpClientInterface->openInput(profile->mModule->mHandle, + &inputDesc->mDevice, + &inputDesc->mSamplingRate, + &inputDesc->mFormat, + &inputDesc->mChannelMask); + + // only accept input with the exact requested set of parameters + if (input == 0 || + (samplingRate != inputDesc->mSamplingRate) || + (format != inputDesc->mFormat) || + (channelMask != inputDesc->mChannelMask)) { + ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask %x", + samplingRate, format, channelMask); + if (input != 0) { + mpClientInterface->closeInput(input); + } + delete inputDesc; + return 0; + } + mInputs.add(input, inputDesc); + return input; +} + +status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input) +{ + ALOGV("startInput() input %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("startInput() unknown input %d", input); + return BAD_VALUE; + } + AudioInputDescriptor *inputDesc = mInputs.valueAt(index); + +#ifdef AUDIO_POLICY_TEST + if (mTestInput == 0) +#endif //AUDIO_POLICY_TEST + { + // refuse 2 active AudioRecord clients at the same time except if the active input + // uses AUDIO_SOURCE_HOTWORD in which case it is closed. + audio_io_handle_t activeInput = getActiveInput(); + if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) { + AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput); + if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) { + ALOGW("startInput() preempting already started low-priority input %d", activeInput); + stopInput(activeInput); + releaseInput(activeInput); + } else { + ALOGW("startInput() input %d failed: other input already started", input); + return INVALID_OPERATION; + } + } + } + + audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource); + if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) { + inputDesc->mDevice = newDevice; + } + + // automatically enable the remote submix output when input is started + if (audio_is_remote_submix_device(inputDesc->mDevice)) { + setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AudioSystem::DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS); + } + + AudioParameter param = AudioParameter(); + param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice); + + int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ? + AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource; + + param.addInt(String8(AudioParameter::keyInputSource), aliasSource); + ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource); + + mpClientInterface->setParameters(input, param.toString()); + + inputDesc->mRefCount = 1; + return NO_ERROR; +} + +status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input) +{ + ALOGV("stopInput() input %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("stopInput() unknown input %d", input); + return BAD_VALUE; + } + AudioInputDescriptor *inputDesc = mInputs.valueAt(index); + + if (inputDesc->mRefCount == 0) { + ALOGW("stopInput() input %d already stopped", input); + return INVALID_OPERATION; + } else { + // automatically disable the remote submix output when input is stopped + if (audio_is_remote_submix_device(inputDesc->mDevice)) { + setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, + AudioSystem::DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS); + } + + AudioParameter param = AudioParameter(); + param.addInt(String8(AudioParameter::keyRouting), 0); + mpClientInterface->setParameters(input, param.toString()); + inputDesc->mRefCount = 0; + return NO_ERROR; + } +} + +void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input) +{ + ALOGV("releaseInput() %d", input); + ssize_t index = mInputs.indexOfKey(input); + if (index < 0) { + ALOGW("releaseInput() releasing unknown input %d", input); + return; + } + mpClientInterface->closeInput(input); + delete mInputs.valueAt(index); + mInputs.removeItem(input); + ALOGV("releaseInput() exit"); +} + +void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream, + int indexMin, + int indexMax) +{ + ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); + if (indexMin < 0 || indexMin >= indexMax) { + ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax); + return; + } + mStreams[stream].mIndexMin = indexMin; + mStreams[stream].mIndexMax = indexMax; +} + +status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream, + int index, + audio_devices_t device) +{ + + if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { + return BAD_VALUE; + } + if (!audio_is_output_device(device)) { + return BAD_VALUE; + } + + // Force max volume if stream cannot be muted + if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; + + ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", + stream, device, index); + + // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and + // clear all device specific values + if (device == AUDIO_DEVICE_OUT_DEFAULT) { + mStreams[stream].mIndexCur.clear(); + } + mStreams[stream].mIndexCur.add(device, index); + + // compute and apply stream volume on all outputs according to connected device + status_t status = NO_ERROR; + for (size_t i = 0; i < mOutputs.size(); i++) { + audio_devices_t curDevice = + getDeviceForVolume(mOutputs.valueAt(i)->device()); + if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) { + status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); + if (volStatus != NO_ERROR) { + status = volStatus; + } + } + } + return status; +} + +status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, + int *index, + audio_devices_t device) +{ + if (index == NULL) { + return BAD_VALUE; + } + if (!audio_is_output_device(device)) { + return BAD_VALUE; + } + // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to + // the strategy the stream belongs to. + if (device == AUDIO_DEVICE_OUT_DEFAULT) { + device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); + } + device = getDeviceForVolume(device); + + *index = mStreams[stream].getVolumeIndex(device); + ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); + return NO_ERROR; +} + +audio_io_handle_t AudioPolicyManagerBase::selectOutputForEffects( + const SortedVector& outputs) +{ + // select one output among several suitable for global effects. + // The priority is as follows: + // 1: An offloaded output. If the effect ends up not being offloadable, + // AudioFlinger will invalidate the track and the offloaded output + // will be closed causing the effect to be moved to a PCM output. + // 2: A deep buffer output + // 3: the first output in the list + + if (outputs.size() == 0) { + return 0; + } + + audio_io_handle_t outputOffloaded = 0; + audio_io_handle_t outputDeepBuffer = 0; + + for (size_t i = 0; i < outputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]); + ALOGV("selectOutputForEffects outputs[%d] 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 AudioPolicyManagerBase::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(AudioSystem::MUSIC); + audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); + SortedVector dstOutputs = getOutputsForDevice(device, mOutputs); + + audio_io_handle_t output = selectOutputForEffects(dstOutputs); + ALOGV("getOutputForEffect() got output %d for fx %s flags %x", + output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags); + + return output; +} + +status_t AudioPolicyManagerBase::registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) +{ + ssize_t index = mOutputs.indexOfKey(io); + if (index < 0) { + index = mInputs.indexOfKey(io); + if (index < 0) { + ALOGW("registerEffect() unknown io %d", io); + return INVALID_OPERATION; + } + } + + if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { + ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", + desc->name, desc->memoryUsage); + return INVALID_OPERATION; + } + mTotalEffectsMemory += desc->memoryUsage; + ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", + desc->name, io, strategy, session, id); + ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); + + EffectDescriptor *pDesc = new EffectDescriptor(); + memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t)); + pDesc->mIo = io; + pDesc->mStrategy = (routing_strategy)strategy; + pDesc->mSession = session; + pDesc->mEnabled = false; + + mEffects.add(id, pDesc); + + return NO_ERROR; +} + +status_t AudioPolicyManagerBase::unregisterEffect(int id) +{ + ssize_t index = mEffects.indexOfKey(id); + if (index < 0) { + ALOGW("unregisterEffect() unknown effect ID %d", id); + return INVALID_OPERATION; + } + + EffectDescriptor *pDesc = mEffects.valueAt(index); + + setEffectEnabled(pDesc, false); + + if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) { + ALOGW("unregisterEffect() memory %d too big for total %d", + pDesc->mDesc.memoryUsage, mTotalEffectsMemory); + pDesc->mDesc.memoryUsage = mTotalEffectsMemory; + } + mTotalEffectsMemory -= pDesc->mDesc.memoryUsage; + ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d", + pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory); + + mEffects.removeItem(id); + delete pDesc; + + return NO_ERROR; +} + +status_t AudioPolicyManagerBase::setEffectEnabled(int id, bool enabled) +{ + ssize_t index = mEffects.indexOfKey(id); + if (index < 0) { + ALOGW("unregisterEffect() unknown effect ID %d", id); + return INVALID_OPERATION; + } + + return setEffectEnabled(mEffects.valueAt(index), enabled); +} + +status_t AudioPolicyManagerBase::setEffectEnabled(EffectDescriptor *pDesc, bool enabled) +{ + if (enabled == pDesc->mEnabled) { + ALOGV("setEffectEnabled(%s) effect already %s", + enabled?"true":"false", enabled?"enabled":"disabled"); + return INVALID_OPERATION; + } + + if (enabled) { + if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { + ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS", + pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10); + return INVALID_OPERATION; + } + mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad; + ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); + } else { + if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) { + ALOGW("setEffectEnabled(false) CPU load %d too high for total %d", + pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); + pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; + } + mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad; + ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); + } + pDesc->mEnabled = enabled; + return NO_ERROR; +} + +bool AudioPolicyManagerBase::isNonOffloadableEffectEnabled() +{ + for (size_t i = 0; i < mEffects.size(); i++) { + const EffectDescriptor * const pDesc = mEffects.valueAt(i); + if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) && + ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) { + ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d", + pDesc->mDesc.name, pDesc->mSession); + return true; + } + } + return false; +} + +bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const +{ + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); + if (outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool AudioPolicyManagerBase::isStreamActiveRemotely(int stream, uint32_t inPastMs) const +{ + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < mOutputs.size(); i++) { + const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); + if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && + outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool AudioPolicyManagerBase::isSourceActive(audio_source_t source) const +{ + for (size_t i = 0; i < mInputs.size(); i++) { + const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i); + if ((inputDescriptor->mInputSource == (int)source || + (source == (audio_source_t)AUDIO_SOURCE_VOICE_RECOGNITION && + inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) + && (inputDescriptor->mRefCount > 0)) { + return true; + } + } + return false; +} + + +status_t AudioPolicyManagerBase::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, " A2DP device address: %s\n", mA2dpDeviceAddress.string()); + result.append(buffer); + snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string()); + result.append(buffer); + snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbCardAndDevice.string()); + result.append(buffer); + snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices); + result.append(buffer); + snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices); + result.append(buffer); + snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]); + result.append(buffer); + snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AudioSystem::FOR_SYSTEM]); + result.append(buffer); + write(fd, result.string(), result.size()); + + + snprintf(buffer, SIZE, "\nHW Modules dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mHwModules.size(); i++) { + snprintf(buffer, SIZE, "- HW Module %d:\n", i + 1); + write(fd, buffer, strlen(buffer)); + mHwModules[i]->dump(fd); + } + + snprintf(buffer, SIZE, "\nOutputs dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mOutputs.size(); i++) { + snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i)); + write(fd, buffer, strlen(buffer)); + mOutputs.valueAt(i)->dump(fd); + } + + snprintf(buffer, SIZE, "\nInputs dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mInputs.size(); i++) { + snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i)); + write(fd, buffer, strlen(buffer)); + mInputs.valueAt(i)->dump(fd); + } + + snprintf(buffer, SIZE, "\nStreams dump:\n"); + write(fd, buffer, strlen(buffer)); + snprintf(buffer, SIZE, + " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { + snprintf(buffer, SIZE, " %02d ", i); + write(fd, buffer, strlen(buffer)); + mStreams[i].dump(fd); + } + + snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n", + (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory); + write(fd, buffer, strlen(buffer)); + + snprintf(buffer, SIZE, "Registered effects:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < mEffects.size(); i++) { + snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i)); + write(fd, buffer, strlen(buffer)); + mEffects.valueAt(i)->dump(fd); + } + + + 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 AudioPolicyManagerBase::isOffloadSupported(const audio_offload_info_t& offloadInfo) +{ + ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d," + " BitRate=%u, duration=%lld us, has_video=%d", + offloadInfo.sample_rate, offloadInfo.channel_mask, + offloadInfo.format, + offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us, + offloadInfo.has_video); + + // Check if offload has been disabled + char propValue[PROPERTY_VALUE_MAX]; + if (property_get("audio.offload.disable", propValue, "0")) { + if (atoi(propValue) != 0) { + ALOGV("offload disabled by audio.offload.disable=%s", propValue ); + return false; + } + } + + // Check if stream type is music, then only allow offload as of now. + if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC) + { + ALOGV("isOffloadSupported: stream_type != MUSIC, returning false"); + return false; + } + + //TODO: enable audio offloading with video when ready + if (offloadInfo.has_video) + { + ALOGV("isOffloadSupported: has_video == true, returning false"); + return false; + } + + //If duration is less than minimum value defined in property, return false + if (property_get("audio.offload.min.duration.secs", propValue, NULL)) { + if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) { + ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue); + return false; + } + } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) { + ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS); + return false; + } + + // Do not allow offloading if one non offloadable effect is enabled. This prevents from + // creating an offloaded track and tearing it down immediately after start when audioflinger + // detects there is an active non offloadable effect. + // FIXME: We should check the audio session here but we do not have it in this context. + // This may prevent offloading in rare situations where effects are left active by apps + // in the background. + if (isNonOffloadableEffectEnabled()) { + return false; + } + + // See if there is a profile to support this. + // AUDIO_DEVICE_NONE + 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 != NULL ? "" : "NOT "); + return (profile != NULL); +} + +// ---------------------------------------------------------------------------- +// AudioPolicyManagerBase +// ---------------------------------------------------------------------------- + +AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface) + : +#ifdef AUDIO_POLICY_TEST + Thread(false), +#endif //AUDIO_POLICY_TEST + mPrimaryOutput((audio_io_handle_t)0), + mAvailableOutputDevices(AUDIO_DEVICE_NONE), + mPhoneState(AudioSystem::MODE_NORMAL), + mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), + mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), + mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false), + mSpeakerDrcEnabled(false) +{ + mpClientInterface = clientInterface; + + for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) { + mForceUse[i] = AudioSystem::FORCE_NONE; + } + + mA2dpDeviceAddress = String8(""); + mScoDeviceAddress = String8(""); + mUsbCardAndDevice = String8(""); + + if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) { + if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) { + ALOGE("could not load audio policy configuration file, setting defaults"); + defaultAudioPolicyConfig(); + } + } + + // must be done after reading the policy + initializeVolumeCurves(); + + // open all output streams needed to access attached devices + 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. + for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) + { + const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j]; + + if ((outProfile->mSupportedDevices & mAttachedOutputDevices) && + ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) { + AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile); + outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice & + outProfile->mSupportedDevices); + audio_io_handle_t output = mpClientInterface->openOutput( + outProfile->mModule->mHandle, + &outputDesc->mDevice, + &outputDesc->mSamplingRate, + &outputDesc->mFormat, + &outputDesc->mChannelMask, + &outputDesc->mLatency, + outputDesc->mFlags); + if (output == 0) { + delete outputDesc; + } else { + mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | + (outProfile->mSupportedDevices & mAttachedOutputDevices)); + if (mPrimaryOutput == 0 && + outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { + mPrimaryOutput = output; + } + addOutput(output, outputDesc); + setOutputDevice(output, + (audio_devices_t)(mDefaultOutputDevice & + outProfile->mSupportedDevices), + true); + } + } + } + } + + ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices), + "Not output found for attached devices %08x", + (mAttachedOutputDevices & ~mAvailableOutputDevices)); + + 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 = AudioSystem::PCM_16_BIT; + mTestChannels = AudioSystem::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 +} + +AudioPolicyManagerBase::~AudioPolicyManagerBase() +{ +#ifdef AUDIO_POLICY_TEST + exit(); +#endif //AUDIO_POLICY_TEST + for (size_t i = 0; i < mOutputs.size(); i++) { + mpClientInterface->closeOutput(mOutputs.keyAt(i)); + delete mOutputs.valueAt(i); + } + for (size_t i = 0; i < mInputs.size(); i++) { + mpClientInterface->closeInput(mInputs.keyAt(i)); + delete mInputs.valueAt(i); + } + for (size_t i = 0; i < mHwModules.size(); i++) { + delete mHwModules[i]; + } +} + +status_t AudioPolicyManagerBase::initCheck() +{ + return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR; +} + +#ifdef AUDIO_POLICY_TEST +bool AudioPolicyManagerBase::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 = AudioSystem::INVALID_FORMAT; + if (value == "PCM 16 bits") { + format = AudioSystem::PCM_16_BIT; + } else if (value == "PCM 8 bits") { + format = AudioSystem::PCM_8_BIT; + } else if (value == "Compressed MP3") { + format = AudioSystem::MP3; + } + if (format != AudioSystem::INVALID_FORMAT) { + 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 = AudioSystem::CHANNEL_OUT_STEREO; + } else if (value == "Channels Mono") { + channels = AudioSystem::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")); + + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput); + mpClientInterface->closeOutput(mPrimaryOutput); + + audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; + + delete mOutputs.valueFor(mPrimaryOutput); + mOutputs.removeItem(mPrimaryOutput); + + AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL); + outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; + mPrimaryOutput = mpClientInterface->openOutput(moduleHandle, + &outputDesc->mDevice, + &outputDesc->mSamplingRate, + &outputDesc->mFormat, + &outputDesc->mChannelMask, + &outputDesc->mLatency, + outputDesc->mFlags); + if (mPrimaryOutput == 0) { + ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d", + outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask); + } else { + 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 AudioPolicyManagerBase::exit() +{ + { + AutoMutex _l(mLock); + requestExit(); + mWaitWorkCV.signal(); + } + requestExitAndWait(); +} + +int AudioPolicyManagerBase::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 AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc) +{ + outputDesc->mId = id; + mOutputs.add(id, outputDesc); +} + + +status_t AudioPolicyManagerBase::checkOutputsForDevice(audio_devices_t device, + AudioSystem::device_connection_state state, + SortedVector& outputs, + const String8 paramStr) +{ + AudioOutputDescriptor *desc; + + if (state == AudioSystem::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 & device)) { + ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); + outputs.add(mOutputs.keyAt(i)); + } + } + // then look for output profiles that can be routed to this device + SortedVector 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++) + { + if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) { + ALOGV("checkOutputsForDevice(): adding profile %d from module %d", j, i); + profiles.add(mHwModules[i]->mOutputProfiles[j]); + } + } + } + + 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++) { + IOProfile *profile = profiles[profile_index]; + + // nothing to do if one output is already opened for this profile + size_t j; + for (j = 0; j < mOutputs.size(); j++) { + desc = mOutputs.valueAt(j); + if (!desc->isDuplicated() && desc->mProfile == profile) { + break; + } + } + if (j != mOutputs.size()) { + continue; + } + + ALOGV("opening output for device %08x with params %s", device, paramStr.string()); + desc = new AudioOutputDescriptor(profile); + desc->mDevice = device; + audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER; + offloadInfo.sample_rate = desc->mSamplingRate; + offloadInfo.format = desc->mFormat; + offloadInfo.channel_mask = desc->mChannelMask; + + audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle, + &desc->mDevice, + &desc->mSamplingRate, + &desc->mFormat, + &desc->mChannelMask, + &desc->mLatency, + desc->mFlags, + &offloadInfo); + if (output != 0) { + if (!paramStr.isEmpty()) { + mpClientInterface->setParameters(output, paramStr); + } + + if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { + String8 reply; + char *value; + if (profile->mSamplingRates[0] == 0) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)); + ALOGV("checkOutputsForDevice() direct output sup sampling rates %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + loadSamplingRates(value + 1, profile); + } + } + if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS)); + ALOGV("checkOutputsForDevice() direct output sup formats %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + loadFormats(value + 1, profile); + } + } + if (profile->mChannelMasks[0] == 0) { + reply = mpClientInterface->getParameters(output, + String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS)); + ALOGV("checkOutputsForDevice() direct output sup channel masks %s", + reply.string()); + value = strpbrk((char *)reply.string(), "="); + if (value != NULL) { + loadOutChannels(value + 1, profile); + } + } + if (((profile->mSamplingRates[0] == 0) && + (profile->mSamplingRates.size() < 2)) || + ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) && + (profile->mFormats.size() < 2)) || + ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) && + (profile->mChannelMasks.size() < 2))) { + ALOGW("checkOutputsForDevice() direct output missing param"); + mpClientInterface->closeOutput(output); + output = 0; + } else { + addOutput(output, desc); + } + } else { + audio_io_handle_t duplicatedOutput = 0; + // add output descriptor + addOutput(output, desc); + // 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 != 0) { + // add duplicated output descriptor + AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL); + dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); + dupOutputDesc->mOutput2 = mOutputs.valueFor(output); + dupOutputDesc->mSamplingRate = desc->mSamplingRate; + dupOutputDesc->mFormat = desc->mFormat; + dupOutputDesc->mChannelMask = desc->mChannelMask; + dupOutputDesc->mLatency = desc->mLatency; + addOutput(duplicatedOutput, dupOutputDesc); + applyStreamVolumes(duplicatedOutput, device, 0, true); + } else { + ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", + mPrimaryOutput, output); + mpClientInterface->closeOutput(output); + mOutputs.removeItem(output); + output = 0; + } + } + } + if (output == 0) { + ALOGW("checkOutputsForDevice() could not open output for device %x", device); + delete desc; + profiles.removeAt(profile_index); + profile_index--; + } else { + outputs.add(output); + ALOGV("checkOutputsForDevice(): adding output %d", output); + } + } + + if (profiles.isEmpty()) { + ALOGW("checkOutputsForDevice(): No output available for device %04x", device); + return BAD_VALUE; + } + } else { + // 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() && + !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) { + ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); + outputs.add(mOutputs.keyAt(i)); + } + } + 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++) + { + IOProfile *profile = mHwModules[i]->mOutputProfiles[j]; + if ((profile->mSupportedDevices & device) && + (profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { + ALOGV("checkOutputsForDevice(): clearing direct output profile %d on module %d", + 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; +} + +void AudioPolicyManagerBase::closeOutput(audio_io_handle_t output) +{ + ALOGV("closeOutput(%d)", output); + + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + if (outputDesc == NULL) { + ALOGW("closeOutput() unknown output %d", output); + return; + } + + // look for duplicated outputs connected to the output being removed. + for (size_t i = 0; i < mOutputs.size(); i++) { + AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i); + if (dupOutputDesc->isDuplicated() && + (dupOutputDesc->mOutput1 == outputDesc || + dupOutputDesc->mOutput2 == outputDesc)) { + 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 < (int)AudioSystem::NUM_STREAM_TYPES; j++) { + int refCount = dupOutputDesc->mRefCount[j]; + outputDesc2->changeRefCount((AudioSystem::stream_type)j,-refCount); + } + audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i); + ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); + + mpClientInterface->closeOutput(duplicatedOutput); + delete mOutputs.valueFor(duplicatedOutput); + mOutputs.removeItem(duplicatedOutput); + } + } + + AudioParameter param; + param.add(String8("closing"), String8("true")); + mpClientInterface->setParameters(output, param.toString()); + + mpClientInterface->closeOutput(output); + delete outputDesc; + mOutputs.removeItem(output); + mPreviousOutputs = mOutputs; +} + +SortedVector AudioPolicyManagerBase::getOutputsForDevice(audio_devices_t device, + DefaultKeyedVector openOutputs) +{ + SortedVector outputs; + + ALOGVV("getOutputsForDevice() device %04x", device); + for (size_t i = 0; i < openOutputs.size(); i++) { + ALOGVV("output %d isDuplicated=%d device=%04x", + i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices()); + if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) { + ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i)); + outputs.add(openOutputs.keyAt(i)); + } + } + return outputs; +} + +bool AudioPolicyManagerBase::vectorsEqual(SortedVector& outputs1, + SortedVector& outputs2) +{ + if (outputs1.size() != outputs2.size()) { + return false; + } + for (size_t i = 0; i < outputs1.size(); i++) { + if (outputs1[i] != outputs2[i]) { + return false; + } + } + return true; +} + +void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy) +{ + audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/); + audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/); + SortedVector srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs); + SortedVector dstOutputs = getOutputsForDevice(newDevice, mOutputs); + + if (!vectorsEqual(srcOutputs,dstOutputs)) { + ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d", + strategy, srcOutputs[0], dstOutputs[0]); + // mute strategy while moving tracks from one output to another + for (size_t i = 0; i < srcOutputs.size(); i++) { + AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]); + if (desc->isStrategyActive(strategy)) { + setStrategyMute(strategy, true, srcOutputs[i]); + setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); + } + } + + // Move effects associated to this strategy from previous output to new output + if (strategy == STRATEGY_MEDIA) { + audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs); + SortedVector moved; + for (size_t i = 0; i < mEffects.size(); i++) { + EffectDescriptor *desc = mEffects.valueAt(i); + if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX && + desc->mIo != fxOutput) { + if (moved.indexOf(desc->mIo) < 0) { + ALOGV("checkOutputForStrategy() moving effect %d to output %d", + mEffects.keyAt(i), fxOutput); + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo, + fxOutput); + moved.add(desc->mIo); + } + desc->mIo = fxOutput; + } + } + } + // Move tracks associated to this strategy from previous output to new output + for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) { + if (getStrategy((AudioSystem::stream_type)i) == strategy) { + mpClientInterface->invalidateStream((AudioSystem::stream_type)i); + } + } + } +} + +void AudioPolicyManagerBase::checkOutputForAllStrategies() +{ + checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); + checkOutputForStrategy(STRATEGY_PHONE); + checkOutputForStrategy(STRATEGY_SONIFICATION); + checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); + checkOutputForStrategy(STRATEGY_MEDIA); + checkOutputForStrategy(STRATEGY_DTMF); +} + +audio_io_handle_t AudioPolicyManagerBase::getA2dpOutput() +{ + if (!mHasA2dp) { + return 0; + } + + for (size_t i = 0; i < mOutputs.size(); i++) { + AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i); + if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { + return mOutputs.keyAt(i); + } + } + + return 0; +} + +void AudioPolicyManagerBase::checkA2dpSuspend() +{ + if (!mHasA2dp) { + return; + } + audio_io_handle_t a2dpOutput = getA2dpOutput(); + if (a2dpOutput == 0) { + return; + } + + // 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 (((mScoDeviceAddress == "") || + ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) && + (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) && + ((mPhoneState != AudioSystem::MODE_IN_CALL) && + (mPhoneState != AudioSystem::MODE_RINGTONE))) { + + mpClientInterface->restoreOutput(a2dpOutput); + mA2dpSuspended = false; + } + } else { + if (((mScoDeviceAddress != "") && + ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || + (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) || + ((mPhoneState == AudioSystem::MODE_IN_CALL) || + (mPhoneState == AudioSystem::MODE_RINGTONE))) { + + mpClientInterface->suspendOutput(a2dpOutput); + mA2dpSuspended = true; + } + } +} + +audio_devices_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache) +{ + audio_devices_t device = AUDIO_DEVICE_NONE; + + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + // check the following by order of priority to request a routing change if necessary: + // 1: the strategy enforced audible is active 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 sonification is active on the output: + // use device for strategy sonification + // 4: the strategy "respectful" sonification is active on the output: + // use device for strategy "respectful" sonification + // 5: the strategy media is active on the output: + // use device for strategy media + // 6: the strategy DTMF is active on the output: + // use device for strategy DTMF + if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) { + device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); + } else if (isInCall() || + outputDesc->isStrategyActive(STRATEGY_PHONE)) { + device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) { + device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); + } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) { + device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); + } + + ALOGV("getNewDevice() selected device %x", device); + return device; +} + +uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) { + return (uint32_t)getStrategy(stream); +} + +audio_devices_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) { + audio_devices_t devices; + // 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 < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) { + devices = AUDIO_DEVICE_NONE; + } else { + AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream); + devices = getDeviceForStrategy(strategy, true /*fromCache*/); + } + return devices; +} + +AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy( + AudioSystem::stream_type stream) { + // stream to strategy mapping + switch (stream) { + case AudioSystem::VOICE_CALL: + case AudioSystem::BLUETOOTH_SCO: + return STRATEGY_PHONE; + case AudioSystem::RING: + case AudioSystem::ALARM: + return STRATEGY_SONIFICATION; + case AudioSystem::NOTIFICATION: + return STRATEGY_SONIFICATION_RESPECTFUL; + case AudioSystem::DTMF: + return STRATEGY_DTMF; + default: + ALOGE("unknown stream type"); + case AudioSystem::SYSTEM: + // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs + // while key clicks are played produces a poor result + case AudioSystem::TTS: + case AudioSystem::MUSIC: + return STRATEGY_MEDIA; + case AudioSystem::ENFORCED_AUDIBLE: + return STRATEGY_ENFORCED_AUDIBLE; + } +} + +void AudioPolicyManagerBase::handleNotificationRoutingForStream(AudioSystem::stream_type stream) { + switch(stream) { + case AudioSystem::MUSIC: + checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); + updateDevicesAndOutputs(); + break; + default: + break; + } +} + +audio_devices_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, + bool fromCache) +{ + uint32_t device = AUDIO_DEVICE_NONE; + + if (fromCache) { + ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", + strategy, mDeviceForStrategy[strategy]); + return mDeviceForStrategy[strategy]; + } + + switch (strategy) { + + case STRATEGY_SONIFICATION_RESPECTFUL: + if (isInCall()) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + } else if (isStreamActiveRemotely(AudioSystem::MUSIC, + SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { + // while media is playing on a remote device, use the the sonification behavior. + // Note that we test this usecase before testing if media is playing because + // the isStreamActive() method only informs about the activity of a stream, not + // if it's for local playback. Note also that we use the same delay between both tests + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + } else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { + // while media is playing (or has recently played), use the same device + device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); + } else { + // when media is not playing anymore, fall back on the sonification behavior + device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); + } + + break; + + case STRATEGY_DTMF: + if (!isInCall()) { + // when off call, DTMF strategy follows the same rules as MEDIA strategy + device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); + break; + } + // when in call, DTMF and PHONE strategies follow the same rules + // FALL THROUGH + + case STRATEGY_PHONE: + // for phone strategy, we first consider the forced use and then the available devices by order + // of priority + switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) { + case AudioSystem::FORCE_BT_SCO: + if (!isInCall() || strategy != STRATEGY_DTMF) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; + if (device) break; + } + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO; + if (device) break; + // if SCO device is requested but no SCO device is available, fall back to default case + // FALL THROUGH + + default: // FORCE_NONE + // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP + if (mHasA2dp && !isInCall() && + (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0) && !mA2dpSuspended) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; + if (device) break; + } + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET; + if (device) break; + if (mPhoneState != AudioSystem::MODE_IN_CALL) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + if (device) break; + } + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE; + if (device) break; + device = mDefaultOutputDevice; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); + } + break; + + case AudioSystem::FORCE_SPEAKER: + // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to + // A2DP speaker when forcing to speaker output + if (mHasA2dp && !isInCall() && + (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0) && !mA2dpSuspended) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; + if (device) break; + } + if (mPhoneState != AudioSystem::MODE_IN_CALL) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; + if (device) break; + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + if (device) break; + } + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; + if (device) break; + device = mDefaultOutputDevice; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); + } + break; + } + break; + + case STRATEGY_SONIFICATION: + + // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by + // handleIncallSonification(). + if (isInCall()) { + device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); + break; + } + // FALL THROUGH + + case STRATEGY_ENFORCED_AUDIBLE: + // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION + // except: + // - when in call where it doesn't default to STRATEGY_PHONE behavior + // - in countries where not enforced in which case it follows STRATEGY_MEDIA + + if ((strategy == STRATEGY_SONIFICATION) || + (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) { + device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); + } + } + // The second device used for sonification is the same as the device used by media strategy + // FALL THROUGH + + case STRATEGY_MEDIA: { + uint32_t device2 = AUDIO_DEVICE_NONE; + if (strategy != STRATEGY_SONIFICATION) { + // no sonification on remote submix (e.g. WFD) + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; + } + if ((device2 == AUDIO_DEVICE_NONE) && + mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) && + (getA2dpOutput() != 0) && !mA2dpSuspended) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; + } + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + } + if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { + // no sonification on aux digital (e.g. HDMI) + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL; + } + if ((device2 == AUDIO_DEVICE_NONE) && + (mForceUse[AudioSystem::FOR_DOCK] == AudioSystem::FORCE_ANALOG_DOCK)) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER; + } + + // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or + // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise + device |= device2; + if (device) break; + device = mDefaultOutputDevice; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); + } + } break; + + default: + ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); + break; + } + + ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); + return device; +} + +void AudioPolicyManagerBase::updateDevicesAndOutputs() +{ + for (int i = 0; i < NUM_STRATEGIES; i++) { + mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); + } + mPreviousOutputs = mOutputs; +} + +uint32_t AudioPolicyManagerBase::checkDeviceMuteStrategies(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() && (AudioSystem::popCount(device) >= 2); + // temporary mute output if device selection changes to avoid volume bursts due to + // different per device volumes + bool tempMute = outputDesc->isActive() && (device != prevDevice); + + for (size_t i = 0; i < NUM_STRATEGIES; i++) { + audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); + 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 || tempMute) { + for (size_t j = 0; j < mOutputs.size(); j++) { + AudioOutputDescriptor *desc = mOutputs.valueAt(j); + // skip output if it does not share any device with current output + if ((desc->supportedDevices() & outputDesc->supportedDevices()) + == AUDIO_DEVICE_NONE) { + continue; + } + audio_io_handle_t curOutput = mOutputs.keyAt(j); + ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", + mute ? "muting" : "unmuting", i, curDevice, curOutput); + setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs); + if (desc->isStrategyActive((routing_strategy)i)) { + // do tempMute only for current output + if (tempMute && (desc == outputDesc)) { + setStrategyMute((routing_strategy)i, true, curOutput); + setStrategyMute((routing_strategy)i, false, curOutput, + desc->latency() * 2, device); + } + if ((tempMute && (desc == outputDesc)) || mute) { + if (muteWaitMs < desc->latency()) { + muteWaitMs = desc->latency(); + } + } + } + } + } + } + + // 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). + muteWaitMs *= 2; + // 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 AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, + audio_devices_t device, + bool force, + int delayMs) +{ + ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + AudioParameter param; + uint32_t muteWaitMs; + + if (outputDesc->isDuplicated()) { + muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs); + muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, 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) == 0)) { + return 0; + } + + // filter devices according to output selected + device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices); + + 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 + // - the requested device is the same as current device and force is not specified. + // Doing this check here allows the caller to call setOutputDevice() without conditions + if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) { + ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output); + return muteWaitMs; + } + + ALOGV("setOutputDevice() changing device"); + // do the routing + param.addInt(String8(AudioParameter::keyRouting), (int)device); + mpClientInterface->setParameters(output, param.toString(), delayMs); + + // update stream volumes according to new device + applyStreamVolumes(output, device, delayMs); + + return muteWaitMs; +} + +AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getInputProfile(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask) +{ + // 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++) + { + IOProfile *profile = mHwModules[i]->mInputProfiles[j]; + if (profile->isCompatibleProfile(device, samplingRate, format, + channelMask, AUDIO_OUTPUT_FLAG_NONE)) { + return profile; + } + } + } + return NULL; +} + +audio_devices_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource) +{ + uint32_t device = AUDIO_DEVICE_NONE; + + switch (inputSource) { + case AUDIO_SOURCE_VOICE_UPLINK: + if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) { + device = AUDIO_DEVICE_IN_VOICE_CALL; + break; + } + // FALL THROUGH + + case AUDIO_SOURCE_DEFAULT: + case AUDIO_SOURCE_MIC: + case AUDIO_SOURCE_VOICE_RECOGNITION: + case AUDIO_SOURCE_HOTWORD: + case AUDIO_SOURCE_VOICE_COMMUNICATION: + if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO && + mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + case AUDIO_SOURCE_CAMCORDER: + if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) { + device = AUDIO_DEVICE_IN_BACK_MIC; + } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + case AUDIO_SOURCE_VOICE_DOWNLINK: + case AUDIO_SOURCE_VOICE_CALL: + if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) { + device = AUDIO_DEVICE_IN_VOICE_CALL; + } + break; + case AUDIO_SOURCE_REMOTE_SUBMIX: + if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { + device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; + } + break; + default: + ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); + break; + } + ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); + return device; +} + +bool AudioPolicyManagerBase::isVirtualInputDevice(audio_devices_t device) +{ + if ((device & AUDIO_DEVICE_BIT_IN) != 0) { + device &= ~AUDIO_DEVICE_BIT_IN; + if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) + return true; + } + return false; +} + +audio_io_handle_t AudioPolicyManagerBase::getActiveInput(bool ignoreVirtualInputs) +{ + for (size_t i = 0; i < mInputs.size(); i++) { + const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i); + if ((input_descriptor->mRefCount > 0) + && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) { + return mInputs.keyAt(i); + } + } + return 0; +} + + +audio_devices_t AudioPolicyManagerBase::getDeviceForVolume(audio_devices_t device) +{ + if (device == AUDIO_DEVICE_NONE) { + // this happens when forcing a route update and no track is active on an output. + // In this case the returned category is not important. + device = AUDIO_DEVICE_OUT_SPEAKER; + } else if (AudioSystem::popCount(device) > 1) { + // Multiple device selection is either: + // - speaker + one other device: give priority to speaker in this case. + // - one A2DP device + another device: happens with duplicated output. In this case + // retain the device on the A2DP output as the other must not correspond to an active + // selection if not the speaker. + if (device & AUDIO_DEVICE_OUT_SPEAKER) { + device = AUDIO_DEVICE_OUT_SPEAKER; + } else { + device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP); + } + } + + ALOGW_IF(AudioSystem::popCount(device) != 1, + "getDeviceForVolume() invalid device combination: %08x", + device); + + return device; +} + +AudioPolicyManagerBase::device_category AudioPolicyManagerBase::getDeviceCategory(audio_devices_t device) +{ + switch(getDeviceForVolume(device)) { + case AUDIO_DEVICE_OUT_EARPIECE: + return DEVICE_CATEGORY_EARPIECE; + case AUDIO_DEVICE_OUT_WIRED_HEADSET: + case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES: + return DEVICE_CATEGORY_HEADSET; + case AUDIO_DEVICE_OUT_SPEAKER: + case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: + case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER: + case AUDIO_DEVICE_OUT_AUX_DIGITAL: + case AUDIO_DEVICE_OUT_USB_ACCESSORY: + case AUDIO_DEVICE_OUT_USB_DEVICE: + case AUDIO_DEVICE_OUT_REMOTE_SUBMIX: + default: + return DEVICE_CATEGORY_SPEAKER; + } +} + +float AudioPolicyManagerBase::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, + int indexInUi) +{ + device_category deviceCategory = getDeviceCategory(device); + const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory]; + + // the volume index in the UI is relative to the min and max volume indices for this stream type + int nbSteps = 1 + curve[VOLMAX].mIndex - + curve[VOLMIN].mIndex; + int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) / + (streamDesc.mIndexMax - streamDesc.mIndexMin); + + // find what part of the curve this index volume belongs to, or if it's out of bounds + int segment = 0; + if (volIdx < curve[VOLMIN].mIndex) { // out of bounds + return 0.0f; + } else if (volIdx < curve[VOLKNEE1].mIndex) { + segment = 0; + } else if (volIdx < curve[VOLKNEE2].mIndex) { + segment = 1; + } else if (volIdx <= curve[VOLMAX].mIndex) { + segment = 2; + } else { // out of bounds + return 1.0f; + } + + // linear interpolation in the attenuation table in dB + float decibels = curve[segment].mDBAttenuation + + ((float)(volIdx - curve[segment].mIndex)) * + ( (curve[segment+1].mDBAttenuation - + curve[segment].mDBAttenuation) / + ((float)(curve[segment+1].mIndex - + curve[segment].mIndex)) ); + + float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) + + ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f", + curve[segment].mIndex, volIdx, + curve[segment+1].mIndex, + curve[segment].mDBAttenuation, + decibels, + curve[segment+1].mDBAttenuation, + amplification); + + return amplification; +} + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = { + {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f} +}; + +// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks +// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets. +// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java). +// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset. + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = { + {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + AudioPolicyManagerBase::sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = { + {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f} +}; + +const AudioPolicyManagerBase::VolumeCurvePoint + *AudioPolicyManagerBase::sVolumeProfiles[AUDIO_STREAM_CNT] + [AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = { + { // AUDIO_STREAM_VOICE_CALL + sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_SYSTEM + sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_RING + sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_MUSIC + sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_ALARM + sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_NOTIFICATION + sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_BLUETOOTH_SCO + sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_ENFORCED_AUDIBLE + sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_DTMF + sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, + { // AUDIO_STREAM_TTS + sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET + sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER + sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE + }, +}; + +void AudioPolicyManagerBase::initializeVolumeCurves() +{ + for (int i = 0; i < AUDIO_STREAM_CNT; i++) { + for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) { + mStreams[i].mVolumeCurve[j] = + sVolumeProfiles[i][j]; + } + } + + // Check availability of DRC on speaker path: if available, override some of the speaker curves + if (mSpeakerDrcEnabled) { + mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = + sDefaultSystemVolumeCurveDrc; + mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = + sSpeakerSonificationVolumeCurveDrc; + mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = + sSpeakerSonificationVolumeCurveDrc; + mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = + sSpeakerSonificationVolumeCurveDrc; + } +} + +float AudioPolicyManagerBase::computeVolume(int stream, + int index, + audio_io_handle_t output, + audio_devices_t device) +{ + float volume = 1.0; + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + StreamDescriptor &streamDesc = mStreams[stream]; + + if (device == AUDIO_DEVICE_NONE) { + device = outputDesc->device(); + } + + // if volume is not 0 (not muted), force media volume to max on digital output + if (stream == AudioSystem::MUSIC && + index != mStreams[stream].mIndexMin && + (device == AUDIO_DEVICE_OUT_AUX_DIGITAL || + device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET || + device == AUDIO_DEVICE_OUT_USB_ACCESSORY || + device == AUDIO_DEVICE_OUT_USB_DEVICE)) { + return 1.0; + } + + volume = volIndexToAmpl(device, streamDesc, index); + + // if a headset is connected, apply the following rules to ring tones and notifications + // to avoid sound level bursts in user's ears: + // - always attenuate ring tones and notifications volume by 6dB + // - if music is playing, always limit the volume to current music volume, + // with a minimum threshold at -36dB so that notification is always perceived. + const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)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 == AudioSystem::SYSTEM) + || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && + (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) && + streamDesc.mCanBeMuted) { + volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; + // when the phone is ringing we must consider that music could have been paused just before + // by the music application and behave as if music was active if the last music track was + // just stopped + if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || + mLimitRingtoneVolume) { + audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); + float musicVol = computeVolume(AudioSystem::MUSIC, + mStreams[AudioSystem::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 AudioPolicyManagerBase::checkAndSetVolume(int stream, + int index, + audio_io_handle_t output, + audio_devices_t device, + int delayMs, + bool force) +{ + + // do not change actual stream volume if the stream is muted + if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { + ALOGVV("checkAndSetVolume() stream %d muted count %d", + stream, mOutputs.valueFor(output)->mMuteCount[stream]); + return NO_ERROR; + } + + // do not change in call volume if bluetooth is connected and vice versa + if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) || + (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) { + ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", + stream, mForceUse[AudioSystem::FOR_COMMUNICATION]); + return INVALID_OPERATION; + } + + float volume = computeVolume(stream, index, output, device); + // 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 == AudioSystem::BLUETOOTH_SCO) { + mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs); + } + mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs); + } + + if (stream == AudioSystem::VOICE_CALL || + stream == AudioSystem::BLUETOOTH_SCO) { + float voiceVolume; + // Force voice volume to max for bluetooth SCO as volume is managed by the headset + if (stream == AudioSystem::VOICE_CALL) { + voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; + } else { + voiceVolume = 1.0; + } + + if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) { + mpClientInterface->setVoiceVolume(voiceVolume, delayMs); + mLastVoiceVolume = voiceVolume; + } + } + + return NO_ERROR; +} + +void AudioPolicyManagerBase::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 < AudioSystem::NUM_STREAM_TYPES; stream++) { + checkAndSetVolume(stream, + mStreams[stream].getVolumeIndex(device), + output, + device, + delayMs, + force); + } +} + +void AudioPolicyManagerBase::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 < AudioSystem::NUM_STREAM_TYPES; stream++) { + if (getStrategy((AudioSystem::stream_type)stream) == strategy) { + setStreamMute(stream, on, output, delayMs, device); + } + } +} + +void AudioPolicyManagerBase::setStreamMute(int stream, + bool on, + audio_io_handle_t output, + int delayMs, + audio_devices_t device) +{ + StreamDescriptor &streamDesc = mStreams[stream]; + AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output); + if (device == AUDIO_DEVICE_NONE) { + device = outputDesc->device(); + } + + ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", + stream, on, output, outputDesc->mMuteCount[stream], device); + + if (on) { + if (outputDesc->mMuteCount[stream] == 0) { + if (streamDesc.mCanBeMuted && + ((stream != AudioSystem::ENFORCED_AUDIBLE) || + (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::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 AudioPolicyManagerBase::handleIncallSonification(int 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((AudioSystem::stream_type)stream); + if ((stream_strategy == STRATEGY_SONIFICATION) || + ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { + 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 (AudioSystem::isLowVisibility((AudioSystem::stream_type)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(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL); + } else { + mpClientInterface->stopTone(); + } + } + } + } +} + +bool AudioPolicyManagerBase::isInCall() +{ + return isStateInCall(mPhoneState); +} + +bool AudioPolicyManagerBase::isStateInCall(int state) { + return ((state == AudioSystem::MODE_IN_CALL) || + (state == AudioSystem::MODE_IN_COMMUNICATION)); +} + +uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad() +{ + return MAX_EFFECTS_CPU_LOAD; +} + +uint32_t AudioPolicyManagerBase::getMaxEffectsMemory() +{ + return MAX_EFFECTS_MEMORY; +} + +// --- AudioOutputDescriptor class implementation + +AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor( + const IOProfile *profile) + : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), + mChannelMask(0), mLatency(0), + mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), + mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0) +{ + // clear usage count for all stream types + for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { + mRefCount[i] = 0; + mCurVolume[i] = -1.0; + mMuteCount[i] = 0; + mStopTime[i] = 0; + } + for (int i = 0; i < NUM_STRATEGIES; i++) { + mStrategyMutedByDevice[i] = false; + } + if (profile != NULL) { + mSamplingRate = profile->mSamplingRates[0]; + mFormat = profile->mFormats[0]; + mChannelMask = profile->mChannelMasks[0]; + mFlags = profile->mFlags; + } +} + +audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::device() const +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); + } else { + return mDevice; + } +} + +uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::latency() +{ + if (isDuplicated()) { + return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; + } else { + return mLatency; + } +} + +bool AudioPolicyManagerBase::AudioOutputDescriptor::sharesHwModuleWith( + const AudioOutputDescriptor *outputDesc) +{ + if (isDuplicated()) { + return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); + } else if (outputDesc->isDuplicated()){ + return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2); + } else { + return (mProfile->mModule == outputDesc->mProfile->mModule); + } +} + +void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta) +{ + // forward usage count change to attached outputs + if (isDuplicated()) { + mOutput1->changeRefCount(stream, delta); + mOutput2->changeRefCount(stream, delta); + } + if ((delta + (int)mRefCount[stream]) < 0) { + ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]); + mRefCount[stream] = 0; + return; + } + mRefCount[stream] += delta; + ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]); +} + +audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::supportedDevices() +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); + } else { + return mProfile->mSupportedDevices ; + } +} + +bool AudioPolicyManagerBase::AudioOutputDescriptor::isActive(uint32_t inPastMs) const +{ + return isStrategyActive(NUM_STRATEGIES, inPastMs); +} + +bool AudioPolicyManagerBase::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy, + uint32_t inPastMs, + nsecs_t sysTime) const +{ + if ((sysTime == 0) && (inPastMs != 0)) { + sysTime = systemTime(); + } + for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { + if (((getStrategy((AudioSystem::stream_type)i) == strategy) || + (NUM_STRATEGIES == strategy)) && + isStreamActive((AudioSystem::stream_type)i, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool AudioPolicyManagerBase::AudioOutputDescriptor::isStreamActive(AudioSystem::stream_type stream, + uint32_t inPastMs, + nsecs_t sysTime) const +{ + if (mRefCount[stream] != 0) { + return true; + } + if (inPastMs == 0) { + return false; + } + if (sysTime == 0) { + sysTime = systemTime(); + } + if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) { + return true; + } + return false; +} + + +status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); + result.append(buffer); + snprintf(buffer, SIZE, " Format: %08x\n", mFormat); + result.append(buffer); + snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); + result.append(buffer); + snprintf(buffer, SIZE, " Latency: %d\n", mLatency); + result.append(buffer); + snprintf(buffer, SIZE, " Flags %08x\n", mFlags); + result.append(buffer); + snprintf(buffer, SIZE, " Devices %08x\n", device()); + result.append(buffer); + snprintf(buffer, SIZE, " Stream volume refCount muteCount\n"); + result.append(buffer); + for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) { + snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n", i, mCurVolume[i], mRefCount[i], mMuteCount[i]); + result.append(buffer); + } + write(fd, result.string(), result.size()); + + return NO_ERROR; +} + +// --- AudioInputDescriptor class implementation + +AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor(const IOProfile *profile) + : mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0), + mDevice(AUDIO_DEVICE_NONE), mRefCount(0), + mInputSource(0), mProfile(profile) +{ +} + +status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate); + result.append(buffer); + snprintf(buffer, SIZE, " Format: %d\n", mFormat); + result.append(buffer); + snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask); + result.append(buffer); + snprintf(buffer, SIZE, " Devices %08x\n", mDevice); + result.append(buffer); + snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount); + result.append(buffer); + write(fd, result.string(), result.size()); + + return NO_ERROR; +} + +// --- StreamDescriptor class implementation + +AudioPolicyManagerBase::StreamDescriptor::StreamDescriptor() + : mIndexMin(0), mIndexMax(1), mCanBeMuted(true) +{ + mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0); +} + +int AudioPolicyManagerBase::StreamDescriptor::getVolumeIndex(audio_devices_t device) +{ + device = AudioPolicyManagerBase::getDeviceForVolume(device); + // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT + if (mIndexCur.indexOfKey(device) < 0) { + device = AUDIO_DEVICE_OUT_DEFAULT; + } + return mIndexCur.valueFor(device); +} + +void AudioPolicyManagerBase::StreamDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "%s %02d %02d ", + mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax); + result.append(buffer); + for (size_t i = 0; i < mIndexCur.size(); i++) { + snprintf(buffer, SIZE, "%04x : %02d, ", + mIndexCur.keyAt(i), + mIndexCur.valueAt(i)); + result.append(buffer); + } + result.append("\n"); + + write(fd, result.string(), result.size()); +} + +// --- EffectDescriptor class implementation + +status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " I/O: %d\n", mIo); + result.append(buffer); + snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy); + result.append(buffer); + snprintf(buffer, SIZE, " Session: %d\n", mSession); + result.append(buffer); + snprintf(buffer, SIZE, " Name: %s\n", mDesc.name); + result.append(buffer); + snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled"); + result.append(buffer); + write(fd, result.string(), result.size()); + + return NO_ERROR; +} + +// --- IOProfile class implementation + +AudioPolicyManagerBase::HwModule::HwModule(const char *name) + : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0) +{ +} + +AudioPolicyManagerBase::HwModule::~HwModule() +{ + for (size_t i = 0; i < mOutputProfiles.size(); i++) { + delete mOutputProfiles[i]; + } + for (size_t i = 0; i < mInputProfiles.size(); i++) { + delete mInputProfiles[i]; + } + free((void *)mName); +} + +void AudioPolicyManagerBase::HwModule::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " - name: %s\n", mName); + result.append(buffer); + snprintf(buffer, SIZE, " - handle: %d\n", mHandle); + result.append(buffer); + write(fd, result.string(), result.size()); + if (mOutputProfiles.size()) { + write(fd, " - outputs:\n", strlen(" - outputs:\n")); + for (size_t i = 0; i < mOutputProfiles.size(); i++) { + snprintf(buffer, SIZE, " output %d:\n", i); + write(fd, buffer, strlen(buffer)); + mOutputProfiles[i]->dump(fd); + } + } + if (mInputProfiles.size()) { + write(fd, " - inputs:\n", strlen(" - inputs:\n")); + for (size_t i = 0; i < mInputProfiles.size(); i++) { + snprintf(buffer, SIZE, " input %d:\n", i); + write(fd, buffer, strlen(buffer)); + mInputProfiles[i]->dump(fd); + } + } +} + +AudioPolicyManagerBase::IOProfile::IOProfile(HwModule *module) + : mFlags((audio_output_flags_t)0), mModule(module) +{ +} + +AudioPolicyManagerBase::IOProfile::~IOProfile() +{ +} + +// checks if the IO profile is compatible with specified parameters. +// Sampling rate, format and channel mask must be specified in order to +// get a valid a match +bool AudioPolicyManagerBase::IOProfile::isCompatibleProfile(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags) const +{ + if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) { + return false; + } + + if ((mSupportedDevices & device) != device) { + return false; + } + if ((mFlags & flags) != flags) { + return false; + } + size_t i; + for (i = 0; i < mSamplingRates.size(); i++) + { + if (mSamplingRates[i] == samplingRate) { + break; + } + } + if (i == mSamplingRates.size()) { + return false; + } + for (i = 0; i < mFormats.size(); i++) + { + if (mFormats[i] == format) { + break; + } + } + if (i == mFormats.size()) { + return false; + } + for (i = 0; i < mChannelMasks.size(); i++) + { + if (mChannelMasks[i] == channelMask) { + break; + } + } + if (i == mChannelMasks.size()) { + return false; + } + return true; +} + +void AudioPolicyManagerBase::IOProfile::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " - sampling rates: "); + result.append(buffer); + for (size_t i = 0; i < mSamplingRates.size(); i++) { + snprintf(buffer, SIZE, "%d", mSamplingRates[i]); + result.append(buffer); + result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", "); + } + + snprintf(buffer, SIZE, " - channel masks: "); + result.append(buffer); + for (size_t i = 0; i < mChannelMasks.size(); i++) { + snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]); + result.append(buffer); + result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", "); + } + + snprintf(buffer, SIZE, " - formats: "); + result.append(buffer); + for (size_t i = 0; i < mFormats.size(); i++) { + snprintf(buffer, SIZE, "0x%08x", mFormats[i]); + result.append(buffer); + result.append(i == (mFormats.size() - 1) ? "\n" : ", "); + } + + snprintf(buffer, SIZE, " - devices: 0x%04x\n", mSupportedDevices); + result.append(buffer); + snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags); + result.append(buffer); + + write(fd, result.string(), result.size()); +} + +// --- audio_policy.conf file parsing + +struct StringToEnum { + const char *name; + uint32_t value; +}; + +#define STRING_TO_ENUM(string) { #string, string } +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) + +const struct StringToEnum sDeviceNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB), + STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL), + STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL), + STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC), + STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX), + STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET), + STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY), +}; + +const struct StringToEnum sFlagNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD), + STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING), +}; + +const struct StringToEnum sFormatNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT), + STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED), + STRING_TO_ENUM(AUDIO_FORMAT_MP3), + STRING_TO_ENUM(AUDIO_FORMAT_AAC), + STRING_TO_ENUM(AUDIO_FORMAT_VORBIS), +}; + +const struct StringToEnum sOutChannelsNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), + STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), +}; + +const struct StringToEnum sInChannelsNameToEnumTable[] = { + STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO), + STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO), + STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK), +}; + + +uint32_t AudioPolicyManagerBase::stringToEnum(const struct StringToEnum *table, + size_t size, + const char *name) +{ + for (size_t i = 0; i < size; i++) { + if (strcmp(table[i].name, name) == 0) { + ALOGV("stringToEnum() found %s", table[i].name); + return table[i].value; + } + } + return 0; +} + +bool AudioPolicyManagerBase::stringToBool(const char *value) +{ + return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0)); +} + +audio_output_flags_t AudioPolicyManagerBase::parseFlagNames(char *name) +{ + uint32_t flag = 0; + + // it is OK to cast name to non const here as we are not going to use it after + // strtok() modifies it + char *flagName = strtok(name, "|"); + while (flagName != NULL) { + if (strlen(flagName) != 0) { + flag |= stringToEnum(sFlagNameToEnumTable, + ARRAY_SIZE(sFlagNameToEnumTable), + flagName); + } + flagName = strtok(NULL, "|"); + } + //force direct flag if offload flag is set: offloading implies a direct output stream + // and all common behaviors are driven by checking only the direct flag + // this should normally be set appropriately in the policy configuration file + if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { + flag |= AUDIO_OUTPUT_FLAG_DIRECT; + } + + return (audio_output_flags_t)flag; +} + +audio_devices_t AudioPolicyManagerBase::parseDeviceNames(char *name) +{ + uint32_t device = 0; + + char *devName = strtok(name, "|"); + while (devName != NULL) { + if (strlen(devName) != 0) { + device |= stringToEnum(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + devName); + } + devName = strtok(NULL, "|"); + } + return device; +} + +void AudioPolicyManagerBase::loadSamplingRates(char *name, IOProfile *profile) +{ + char *str = strtok(name, "|"); + + // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling + // rates should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + profile->mSamplingRates.add(0); + return; + } + + while (str != NULL) { + uint32_t rate = atoi(str); + if (rate != 0) { + ALOGV("loadSamplingRates() adding rate %d", rate); + profile->mSamplingRates.add(rate); + } + str = strtok(NULL, "|"); + } + return; +} + +void AudioPolicyManagerBase::loadFormats(char *name, IOProfile *profile) +{ + char *str = strtok(name, "|"); + + // by convention, "0' in the first entry in mFormats indicates the supported formats + // should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + profile->mFormats.add(AUDIO_FORMAT_DEFAULT); + return; + } + + while (str != NULL) { + audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable, + ARRAY_SIZE(sFormatNameToEnumTable), + str); + if (format != AUDIO_FORMAT_DEFAULT) { + profile->mFormats.add(format); + } + str = strtok(NULL, "|"); + } + return; +} + +void AudioPolicyManagerBase::loadInChannels(char *name, IOProfile *profile) +{ + const char *str = strtok(name, "|"); + + ALOGV("loadInChannels() %s", name); + + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + profile->mChannelMasks.add(0); + return; + } + + while (str != NULL) { + audio_channel_mask_t channelMask = + (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable, + ARRAY_SIZE(sInChannelsNameToEnumTable), + str); + if (channelMask != 0) { + ALOGV("loadInChannels() adding channelMask %04x", channelMask); + profile->mChannelMasks.add(channelMask); + } + str = strtok(NULL, "|"); + } + return; +} + +void AudioPolicyManagerBase::loadOutChannels(char *name, IOProfile *profile) +{ + const char *str = strtok(name, "|"); + + ALOGV("loadOutChannels() %s", name); + + // by convention, "0' in the first entry in mChannelMasks indicates the supported channel + // masks should be read from the output stream after it is opened for the first time + if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { + profile->mChannelMasks.add(0); + return; + } + + while (str != NULL) { + audio_channel_mask_t channelMask = + (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable, + ARRAY_SIZE(sOutChannelsNameToEnumTable), + str); + if (channelMask != 0) { + profile->mChannelMasks.add(channelMask); + } + str = strtok(NULL, "|"); + } + return; +} + +status_t AudioPolicyManagerBase::loadInput(cnode *root, HwModule *module) +{ + cnode *node = root->first_child; + + IOProfile *profile = new IOProfile(module); + + while (node) { + if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { + loadSamplingRates((char *)node->value, profile); + } else if (strcmp(node->name, FORMATS_TAG) == 0) { + loadFormats((char *)node->value, profile); + } else if (strcmp(node->name, CHANNELS_TAG) == 0) { + loadInChannels((char *)node->value, profile); + } else if (strcmp(node->name, DEVICES_TAG) == 0) { + profile->mSupportedDevices = parseDeviceNames((char *)node->value); + } + node = node->next; + } + ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE, + "loadInput() invalid supported devices"); + ALOGW_IF(profile->mChannelMasks.size() == 0, + "loadInput() invalid supported channel masks"); + ALOGW_IF(profile->mSamplingRates.size() == 0, + "loadInput() invalid supported sampling rates"); + ALOGW_IF(profile->mFormats.size() == 0, + "loadInput() invalid supported formats"); + if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) && + (profile->mChannelMasks.size() != 0) && + (profile->mSamplingRates.size() != 0) && + (profile->mFormats.size() != 0)) { + + ALOGV("loadInput() adding input mSupportedDevices %04x", profile->mSupportedDevices); + + module->mInputProfiles.add(profile); + return NO_ERROR; + } else { + delete profile; + return BAD_VALUE; + } +} + +status_t AudioPolicyManagerBase::loadOutput(cnode *root, HwModule *module) +{ + cnode *node = root->first_child; + + IOProfile *profile = new IOProfile(module); + + while (node) { + if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) { + loadSamplingRates((char *)node->value, profile); + } else if (strcmp(node->name, FORMATS_TAG) == 0) { + loadFormats((char *)node->value, profile); + } else if (strcmp(node->name, CHANNELS_TAG) == 0) { + loadOutChannels((char *)node->value, profile); + } else if (strcmp(node->name, DEVICES_TAG) == 0) { + profile->mSupportedDevices = parseDeviceNames((char *)node->value); + } else if (strcmp(node->name, FLAGS_TAG) == 0) { + profile->mFlags = parseFlagNames((char *)node->value); + } + node = node->next; + } + ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE, + "loadOutput() invalid supported devices"); + ALOGW_IF(profile->mChannelMasks.size() == 0, + "loadOutput() invalid supported channel masks"); + ALOGW_IF(profile->mSamplingRates.size() == 0, + "loadOutput() invalid supported sampling rates"); + ALOGW_IF(profile->mFormats.size() == 0, + "loadOutput() invalid supported formats"); + if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) && + (profile->mChannelMasks.size() != 0) && + (profile->mSamplingRates.size() != 0) && + (profile->mFormats.size() != 0)) { + + ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x", + profile->mSupportedDevices, profile->mFlags); + + module->mOutputProfiles.add(profile); + return NO_ERROR; + } else { + delete profile; + return BAD_VALUE; + } +} + +void AudioPolicyManagerBase::loadHwModule(cnode *root) +{ + cnode *node = config_find(root, OUTPUTS_TAG); + status_t status = NAME_NOT_FOUND; + + HwModule *module = new HwModule(root->name); + + if (node != NULL) { + if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) { + mHasA2dp = true; + } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) { + mHasUsb = true; + } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) { + mHasRemoteSubmix = true; + } + + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading output %s", node->name); + status_t tmpStatus = loadOutput(node, module); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + node = config_find(root, INPUTS_TAG); + if (node != NULL) { + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading input %s", node->name); + status_t tmpStatus = loadInput(node, module); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + if (status == NO_ERROR) { + mHwModules.add(module); + } else { + delete module; + } +} + +void AudioPolicyManagerBase::loadHwModules(cnode *root) +{ + cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); + if (node == NULL) { + return; + } + + node = node->first_child; + while (node) { + ALOGV("loadHwModules() loading module %s", node->name); + loadHwModule(node); + node = node->next; + } +} + +void AudioPolicyManagerBase::loadGlobalConfig(cnode *root) +{ + cnode *node = config_find(root, GLOBAL_CONFIG_TAG); + if (node == NULL) { + return; + } + node = node->first_child; + while (node) { + if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) { + mAttachedOutputDevices = parseDeviceNames((char *)node->value); + ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE, + "loadGlobalConfig() no attached output devices"); + ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices); + } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) { + mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + (char *)node->value); + ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE, + "loadGlobalConfig() default device not specified"); + ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice); + } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) { + mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN; + ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices); + } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) { + mSpeakerDrcEnabled = stringToBool((char *)node->value); + ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled); + } + node = node->next; + } +} + +status_t AudioPolicyManagerBase::loadAudioPolicyConfig(const char *path) +{ + cnode *root; + char *data; + + data = (char *)load_file(path, NULL); + if (data == NULL) { + return -ENODEV; + } + root = config_node("", ""); + config_load(root, data); + + loadGlobalConfig(root); + loadHwModules(root); + + config_free(root); + free(root); + free(data); + + ALOGI("loadAudioPolicyConfig() loaded %s\n", path); + + return NO_ERROR; +} + +void AudioPolicyManagerBase::defaultAudioPolicyConfig(void) +{ + HwModule *module; + IOProfile *profile; + + mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER; + mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER; + mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN; + + module = new HwModule("primary"); + + profile = new IOProfile(module); + profile->mSamplingRates.add(44100); + profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); + profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); + profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER; + profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; + module->mOutputProfiles.add(profile); + + profile = new IOProfile(module); + profile->mSamplingRates.add(8000); + profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); + profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); + profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC; + module->mInputProfiles.add(profile); + + mHwModules.add(module); +} + +}; // namespace android diff --git a/services/audiopolicy/AudioPolicyManagerBase.h b/services/audiopolicy/AudioPolicyManagerBase.h new file mode 100644 index 0000000..1ff409e --- /dev/null +++ b/services/audiopolicy/AudioPolicyManagerBase.h @@ -0,0 +1,587 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + + +#include +#include +#include +#include +#include +#include +#include +#include +#include + + +namespace android_audio_legacy { + using android::KeyedVector; + using android::DefaultKeyedVector; + using android::SortedVector; + +// ---------------------------------------------------------------------------- + +#define MAX_DEVICE_ADDRESS_LEN 20 +// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB +#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 +// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB +#define SONIFICATION_HEADSET_VOLUME_MIN 0.016 +// Time in milliseconds during which we consider that music is still active after a music +// track was stopped - see computeVolume() +#define SONIFICATION_HEADSET_MUSIC_DELAY 5000 +// Time in milliseconds after media stopped playing during which we consider that the +// sonification should be as unobtrusive as during the time media was playing. +#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000 +// Time in milliseconds during witch some streams are muted while the audio path +// is switched +#define MUTE_TIME_MS 2000 + +#define NUM_TEST_OUTPUTS 5 + +#define NUM_VOL_CURVE_KNEES 2 + +// Default minimum length allowed for offloading a compressed track +// Can be overridden by the audio.offload.min.duration.secs property +#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60 + +// ---------------------------------------------------------------------------- +// AudioPolicyManagerBase implements audio policy manager behavior common to all platforms. +// Each platform must implement an AudioPolicyManager class derived from AudioPolicyManagerBase +// and override methods for which the platform specific behavior differs from the implementation +// in AudioPolicyManagerBase. Even if no specific behavior is required, the AudioPolicyManager +// class must be implemented as well as the class factory function createAudioPolicyManager() +// and provided in a shared library libaudiopolicy.so. +// ---------------------------------------------------------------------------- + +class AudioPolicyManagerBase: public AudioPolicyInterface +#ifdef AUDIO_POLICY_TEST + , public Thread +#endif //AUDIO_POLICY_TEST +{ + +public: + AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface); + virtual ~AudioPolicyManagerBase(); + + // AudioPolicyInterface + virtual status_t setDeviceConnectionState(audio_devices_t device, + AudioSystem::device_connection_state state, + const char *device_address); + virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device, + const char *device_address); + virtual void setPhoneState(int state); + virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config); + virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage); + virtual void setSystemProperty(const char* property, const char* value); + virtual status_t initCheck(); + virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::output_flags flags, + const audio_offload_info_t *offloadInfo); + virtual status_t startOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0); + virtual status_t stopOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0); + virtual void releaseOutput(audio_io_handle_t output); + virtual audio_io_handle_t getInput(int inputSource, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + AudioSystem::audio_in_acoustics acoustics); + + // indicates to the audio policy manager that the input starts being used. + virtual status_t startInput(audio_io_handle_t input); + + // indicates to the audio policy manager that the input stops being used. + virtual status_t stopInput(audio_io_handle_t input); + virtual void releaseInput(audio_io_handle_t input); + virtual void initStreamVolume(AudioSystem::stream_type stream, + int indexMin, + int indexMax); + virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, + int index, + audio_devices_t device); + virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, + int *index, + audio_devices_t device); + + // return the strategy corresponding to a given stream type + virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream); + + // return the enabled output devices for the given stream type + virtual audio_devices_t getDevicesForStream(AudioSystem::stream_type stream); + + virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL); + virtual status_t registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id); + virtual status_t unregisterEffect(int id); + virtual status_t setEffectEnabled(int id, bool enabled); + + virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const; + // return whether a stream is playing remotely, override to change the definition of + // local/remote playback, used for instance by notification manager to not make + // media players lose audio focus when not playing locally + virtual bool isStreamActiveRemotely(int stream, uint32_t inPastMs = 0) const; + virtual bool isSourceActive(audio_source_t source) const; + + virtual status_t dump(int fd); + + virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo); + +protected: + + enum routing_strategy { + STRATEGY_MEDIA, + STRATEGY_PHONE, + STRATEGY_SONIFICATION, + STRATEGY_SONIFICATION_RESPECTFUL, + STRATEGY_DTMF, + STRATEGY_ENFORCED_AUDIBLE, + NUM_STRATEGIES + }; + + // 4 points to define the volume attenuation curve, each characterized by the volume + // index (from 0 to 100) at which they apply, and the attenuation in dB at that index. + // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl() + + enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4}; + + class VolumeCurvePoint + { + public: + int mIndex; + float mDBAttenuation; + }; + + // device categories used for volume curve management. + enum device_category { + DEVICE_CATEGORY_HEADSET, + DEVICE_CATEGORY_SPEAKER, + DEVICE_CATEGORY_EARPIECE, + DEVICE_CATEGORY_CNT + }; + + class IOProfile; + + class HwModule { + public: + HwModule(const char *name); + ~HwModule(); + + void dump(int fd); + + const char *const mName; // base name of the audio HW module (primary, a2dp ...) + audio_module_handle_t mHandle; + Vector mOutputProfiles; // output profiles exposed by this module + Vector mInputProfiles; // input profiles exposed by this module + }; + + // the IOProfile class describes the capabilities of an output or input stream. + // It is currently assumed that all combination of listed parameters are supported. + // It is used by the policy manager to determine if an output or input is suitable for + // a given use case, open/close it accordingly and connect/disconnect audio tracks + // to/from it. + class IOProfile + { + public: + IOProfile(HwModule *module); + ~IOProfile(); + + bool isCompatibleProfile(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags) const; + + void dump(int fd); + + // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats + // indicates the supported parameters should be read from the output stream + // after it is opened for the first time + Vector mSamplingRates; // supported sampling rates + Vector mChannelMasks; // supported channel masks + Vector mFormats; // supported audio formats + audio_devices_t mSupportedDevices; // supported devices (devices this output can be + // routed to) + audio_output_flags_t mFlags; // attribute flags (e.g primary output, + // direct output...). For outputs only. + HwModule *mModule; // audio HW module exposing this I/O stream + }; + + // default volume curve + static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + // default volume curve for media strategy + static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + // volume curve for media strategy on speakers + static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + // volume curve for sonification strategy on speakers + static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT]; + // default volume curves per stream and device category. See initializeVolumeCurves() + static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT]; + + // descriptor for audio outputs. Used to maintain current configuration of each opened audio output + // and keep track of the usage of this output by each audio stream type. + class AudioOutputDescriptor + { + public: + AudioOutputDescriptor(const IOProfile *profile); + + status_t dump(int fd); + + audio_devices_t device() const; + void changeRefCount(AudioSystem::stream_type stream, int delta); + + bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } + audio_devices_t supportedDevices(); + uint32_t latency(); + bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc); + bool isActive(uint32_t inPastMs = 0) const; + bool isStreamActive(AudioSystem::stream_type stream, + uint32_t inPastMs = 0, + nsecs_t sysTime = 0) const; + bool isStrategyActive(routing_strategy strategy, + uint32_t inPastMs = 0, + nsecs_t sysTime = 0) const; + + audio_io_handle_t mId; // output handle + uint32_t mSamplingRate; // + audio_format_t mFormat; // + audio_channel_mask_t mChannelMask; // output configuration + uint32_t mLatency; // + audio_output_flags_t mFlags; // + audio_devices_t mDevice; // current device this output is routed to + uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output + nsecs_t mStopTime[AudioSystem::NUM_STREAM_TYPES]; + AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output + AudioOutputDescriptor *mOutput2; // used by duplicated outputs: second output + float mCurVolume[AudioSystem::NUM_STREAM_TYPES]; // current stream volume + int mMuteCount[AudioSystem::NUM_STREAM_TYPES]; // mute request counter + const IOProfile *mProfile; // I/O profile this output derives from + bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible + // device selection. See checkDeviceMuteStrategies() + uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) + }; + + // descriptor for audio inputs. Used to maintain current configuration of each opened audio input + // and keep track of the usage of this input. + class AudioInputDescriptor + { + public: + AudioInputDescriptor(const IOProfile *profile); + + status_t dump(int fd); + + uint32_t mSamplingRate; // + audio_format_t mFormat; // input configuration + audio_channel_mask_t mChannelMask; // + audio_devices_t mDevice; // current device this input is routed to + uint32_t mRefCount; // number of AudioRecord clients using this output + int mInputSource; // input source selected by application (mediarecorder.h) + const IOProfile *mProfile; // I/O profile this output derives from + }; + + // stream descriptor used for volume control + class StreamDescriptor + { + public: + StreamDescriptor(); + + int getVolumeIndex(audio_devices_t device); + void dump(int fd); + + int mIndexMin; // min volume index + int mIndexMax; // max volume index + KeyedVector mIndexCur; // current volume index per device + bool mCanBeMuted; // true is the stream can be muted + + const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT]; + }; + + // stream descriptor used for volume control + class EffectDescriptor + { + public: + + status_t dump(int fd); + + int mIo; // io the effect is attached to + routing_strategy mStrategy; // routing strategy the effect is associated to + int mSession; // audio session the effect is on + effect_descriptor_t mDesc; // effect descriptor + bool mEnabled; // enabled state: CPU load being used or not + }; + + void addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc); + + // return the strategy corresponding to a given stream type + static routing_strategy getStrategy(AudioSystem::stream_type stream); + + // return appropriate device for streams handled by the specified strategy according to current + // phone state, connected devices... + // if fromCache is true, the device is returned from mDeviceForStrategy[], + // otherwise it is determine by current state + // (device connected,phone state, force use, a2dp output...) + // This allows to: + // 1 speed up process when the state is stable (when starting or stopping an output) + // 2 access to either current device selection (fromCache == true) or + // "future" device selection (fromCache == false) when called from a context + // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND + // before updateDevicesAndOutputs() is called. + virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, + bool fromCache); + + // change the route of the specified output. Returns the number of ms we have slept to + // allow new routing to take effect in certain cases. + uint32_t setOutputDevice(audio_io_handle_t output, + audio_devices_t device, + bool force = false, + int delayMs = 0); + + // select input device corresponding to requested audio source + virtual audio_devices_t getDeviceForInputSource(int inputSource); + + // return io handle of active input or 0 if no input is active + // Only considers inputs from physical devices (e.g. main mic, headset mic) when + // ignoreVirtualInputs is true. + audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true); + + // initialize volume curves for each strategy and device category + void initializeVolumeCurves(); + + // compute the actual volume for a given stream according to the requested index and a particular + // device + virtual float computeVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device); + + // check that volume change is permitted, compute and send new volume to audio hardware + status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); + + // apply all stream volumes to the specified output and device + void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); + + // Mute or unmute all streams handled by the specified strategy on the specified output + void setStrategyMute(routing_strategy strategy, + bool on, + audio_io_handle_t output, + int delayMs = 0, + audio_devices_t device = (audio_devices_t)0); + + // Mute or unmute the stream on the specified output + void setStreamMute(int stream, + bool on, + audio_io_handle_t output, + int delayMs = 0, + audio_devices_t device = (audio_devices_t)0); + + // handle special cases for sonification strategy while in call: mute streams or replace by + // a special tone in the device used for communication + void handleIncallSonification(int stream, bool starting, bool stateChange); + + // true if device is in a telephony or VoIP call + virtual bool isInCall(); + + // true if given state represents a device in a telephony or VoIP call + virtual bool isStateInCall(int state); + + // when a device is connected, checks if an open output can be routed + // to this device. If none is open, tries to open one of the available outputs. + // Returns an output suitable to this device or 0. + // when a device is disconnected, checks if an output is not used any more and + // returns its handle if any. + // transfers the audio tracks and effects from one output thread to another accordingly. + status_t checkOutputsForDevice(audio_devices_t device, + AudioSystem::device_connection_state state, + SortedVector& outputs, + const String8 paramStr); + + // close an output and its companion duplicating output. + void closeOutput(audio_io_handle_t output); + + // checks and if necessary changes outputs used for all strategies. + // must be called every time a condition that affects the output choice for a given strategy + // changes: connected device, phone state, force use... + // Must be called before updateDevicesAndOutputs() + void checkOutputForStrategy(routing_strategy strategy); + + // Same as checkOutputForStrategy() but for a all strategies in order of priority + void checkOutputForAllStrategies(); + + // manages A2DP output suspend/restore according to phone state and BT SCO usage + void checkA2dpSuspend(); + + // returns the A2DP output handle if it is open or 0 otherwise + audio_io_handle_t getA2dpOutput(); + + // selects the most appropriate device on output for current state + // must be called every time a condition that affects the device choice for a given output is + // changed: connected device, phone state, force use, output start, output stop.. + // see getDeviceForStrategy() for the use of fromCache parameter + + audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache); + // updates cache of device used by all strategies (mDeviceForStrategy[]) + // must be called every time a condition that affects the device choice for a given strategy is + // changed: connected device, phone state, force use... + // cached values are used by getDeviceForStrategy() if parameter fromCache is true. + // Must be called after checkOutputForAllStrategies() + + void updateDevicesAndOutputs(); + + virtual uint32_t getMaxEffectsCpuLoad(); + virtual uint32_t getMaxEffectsMemory(); +#ifdef AUDIO_POLICY_TEST + virtual bool threadLoop(); + void exit(); + int testOutputIndex(audio_io_handle_t output); +#endif //AUDIO_POLICY_TEST + + status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled); + + // returns the category the device belongs to with regard to volume curve management + static device_category getDeviceCategory(audio_devices_t device); + + // extract one device relevant for volume control from multiple device selection + static audio_devices_t getDeviceForVolume(audio_devices_t device); + + SortedVector getOutputsForDevice(audio_devices_t device, + DefaultKeyedVector openOutputs); + bool vectorsEqual(SortedVector& outputs1, + SortedVector& outputs2); + + // 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 + // Returns the number of ms waited + uint32_t checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc, + audio_devices_t prevDevice, + uint32_t delayMs); + + audio_io_handle_t selectOutput(const SortedVector& outputs, + AudioSystem::output_flags flags); + IOProfile *getInputProfile(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask); + IOProfile *getProfileForDirectOutput(audio_devices_t device, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags); + + audio_io_handle_t selectOutputForEffects(const SortedVector& outputs); + + bool isNonOffloadableEffectEnabled(); + + // + // Audio policy configuration file parsing (audio_policy.conf) + // + static uint32_t stringToEnum(const struct StringToEnum *table, + size_t size, + const char *name); + static bool stringToBool(const char *value); + static audio_output_flags_t parseFlagNames(char *name); + static audio_devices_t parseDeviceNames(char *name); + void loadSamplingRates(char *name, IOProfile *profile); + void loadFormats(char *name, IOProfile *profile); + void loadOutChannels(char *name, IOProfile *profile); + void loadInChannels(char *name, IOProfile *profile); + status_t loadOutput(cnode *root, HwModule *module); + status_t loadInput(cnode *root, HwModule *module); + void loadHwModule(cnode *root); + void loadHwModules(cnode *root); + void loadGlobalConfig(cnode *root); + status_t loadAudioPolicyConfig(const char *path); + void defaultAudioPolicyConfig(void); + + + AudioPolicyClientInterface *mpClientInterface; // audio policy client interface + audio_io_handle_t mPrimaryOutput; // primary output handle + // list of descriptors for outputs currently opened + DefaultKeyedVector mOutputs; + // copy of mOutputs before setDeviceConnectionState() opens new outputs + // reset to mOutputs when updateDevicesAndOutputs() is called. + DefaultKeyedVector mPreviousOutputs; + DefaultKeyedVector mInputs; // list of input descriptors + audio_devices_t mAvailableOutputDevices; // bit field of all available output devices + audio_devices_t mAvailableInputDevices; // bit field of all available input devices + // without AUDIO_DEVICE_BIT_IN to allow direct bit + // field comparisons + int mPhoneState; // current phone state + AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE]; // current forced use configuration + + StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES]; // stream descriptors for volume control + String8 mA2dpDeviceAddress; // A2DP device MAC address + String8 mScoDeviceAddress; // SCO device MAC address + String8 mUsbCardAndDevice; // USB audio ALSA card and device numbers: + // card=;device=<> + bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected + audio_devices_t mDeviceForStrategy[NUM_STRATEGIES]; + float mLastVoiceVolume; // last voice volume value sent to audio HAL + + // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units + static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000; + // Maximum memory allocated to audio effects in KB + static const uint32_t MAX_EFFECTS_MEMORY = 512; + uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects + uint32_t mTotalEffectsMemory; // current memory used by effects + KeyedVector mEffects; // list of registered audio effects + bool mA2dpSuspended; // true if A2DP output is suspended + bool mHasA2dp; // true on platforms with support for bluetooth A2DP + bool mHasUsb; // true on platforms with support for USB audio + bool mHasRemoteSubmix; // true on platforms with support for remote presentation of a submix + audio_devices_t mAttachedOutputDevices; // output devices always available on the platform + audio_devices_t mDefaultOutputDevice; // output device selected by default at boot time + // (must be in mAttachedOutputDevices) + bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path + // to boost soft sounds, used to adjust volume curves accordingly + + Vector mHwModules; + +#ifdef AUDIO_POLICY_TEST + Mutex mLock; + Condition mWaitWorkCV; + + int mCurOutput; + bool mDirectOutput; + audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS]; + int mTestInput; + uint32_t mTestDevice; + uint32_t mTestSamplingRate; + uint32_t mTestFormat; + uint32_t mTestChannels; + uint32_t mTestLatencyMs; +#endif //AUDIO_POLICY_TEST + +private: + static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, + int indexInUi); + // updates device caching and output for streams that can influence the + // routing of notifications + void handleNotificationRoutingForStream(AudioSystem::stream_type stream); + static bool isVirtualInputDevice(audio_devices_t device); +}; + +}; -- cgit v1.1