1414 lines
64 KiB
C++
1414 lines
64 KiB
C++
/*
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* Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.
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* Not a contribution.
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*
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* Copyright (C) 2009 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*
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* This file was modified by Dolby Laboratories, Inc. The portions of the
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* code that are surrounded by "DOLBY..." are copyrighted and
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* licensed separately, as follows:
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*
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* (C) 2015 Dolby Laboratories, Inc.
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#define LOG_TAG "AudioPolicyManagerCustom"
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//#define LOG_NDEBUG 0
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//#define VERY_VERBOSE_LOGGING
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#ifdef VERY_VERBOSE_LOGGING
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#define ALOGVV ALOGV
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#else
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#define ALOGVV(a...) do { } while(0)
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#endif
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#define MIN(a, b) ((a) < (b) ? (a) : (b))
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// A device mask for all audio output devices that are considered "remote" when evaluating
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// active output devices in isStreamActiveRemotely()
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#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
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// A device mask for all audio input and output devices where matching inputs/outputs on device
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// type alone is not enough: the address must match too
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#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
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AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
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// Following delay should be used if the calculated routing delay from all active
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// input streams is higher than this value
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#define MAX_VOICE_CALL_START_DELAY_MS 100
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#include <inttypes.h>
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#include <math.h>
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#include <cutils/properties.h>
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#include <utils/Log.h>
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#include <hardware/audio.h>
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#include <hardware/audio_effect.h>
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#include <media/AudioParameter.h>
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#include <soundtrigger/SoundTrigger.h>
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#include "AudioPolicyManager.h"
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#include <policy.h>
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#ifdef DOLBY_ENABLE
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#include "DolbyAudioPolicy_impl.h"
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#endif // DOLBY_END
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namespace android {
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// ----------------------------------------------------------------------------
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// AudioPolicyInterface implementation
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// ----------------------------------------------------------------------------
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extern "C" AudioPolicyInterface* createAudioPolicyManager(
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AudioPolicyClientInterface *clientInterface)
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{
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return new AudioPolicyManagerCustom(clientInterface);
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}
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extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
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{
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delete interface;
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}
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status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t device,
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audio_policy_dev_state_t state,
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const char *device_address,
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const char *device_name)
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{
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ALOGD("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
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device, state, device_address, device_name);
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// connect/disconnect only 1 device at a time
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if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
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sp<DeviceDescriptor> devDesc =
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mHwModules.getDeviceDescriptor(device, device_address, device_name);
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// handle output devices
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if (audio_is_output_device(device)) {
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SortedVector <audio_io_handle_t> outputs;
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ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
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// save a copy of the opened output descriptors before any output is opened or closed
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// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
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mPreviousOutputs = mOutputs;
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switch (state)
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{
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// handle output device connection
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case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
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if (index >= 0) {
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#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
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if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
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if (!strncmp(device_address, "hdmi_spkr", 9)) {
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mHdmiAudioDisabled = false;
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} else {
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mHdmiAudioEvent = true;
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}
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}
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#endif
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ALOGW("setDeviceConnectionState() device already connected: %x", device);
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return INVALID_OPERATION;
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}
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ALOGV("setDeviceConnectionState() connecting device %x", device);
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// register new device as available
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index = mAvailableOutputDevices.add(devDesc);
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#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
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if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
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if (!strncmp(device_address, "hdmi_spkr", 9)) {
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mHdmiAudioDisabled = false;
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} else {
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mHdmiAudioEvent = true;
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}
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if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
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mAvailableOutputDevices.remove(devDesc);
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ALOGW("HDMI sink not connected, do not route audio to HDMI out");
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return INVALID_OPERATION;
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}
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}
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#endif
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if (index >= 0) {
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sp<HwModule> module = mHwModules.getModuleForDevice(device);
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if (module == 0) {
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ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
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device);
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mAvailableOutputDevices.remove(devDesc);
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return INVALID_OPERATION;
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}
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mAvailableOutputDevices[index]->attach(module);
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} else {
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return NO_MEMORY;
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}
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if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
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mAvailableOutputDevices.remove(devDesc);
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return INVALID_OPERATION;
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}
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// Propagate device availability to Engine
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mEngine->setDeviceConnectionState(devDesc, state);
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// outputs should never be empty here
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ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
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"checkOutputsForDevice() returned no outputs but status OK");
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ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
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outputs.size());
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// Send connect to HALs
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AudioParameter param = AudioParameter(devDesc->mAddress);
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param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
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mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
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} break;
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// handle output device disconnection
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case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
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if (index < 0) {
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#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
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if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
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if (!strncmp(device_address, "hdmi_spkr", 9)) {
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mHdmiAudioDisabled = true;
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} else {
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mHdmiAudioEvent = false;
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}
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}
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#endif
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ALOGW("setDeviceConnectionState() device not connected: %x", device);
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return INVALID_OPERATION;
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}
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ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
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// Send Disconnect to HALs
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AudioParameter param = AudioParameter(devDesc->mAddress);
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param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
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mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
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// remove device from available output devices
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mAvailableOutputDevices.remove(devDesc);
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#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
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if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
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if (!strncmp(device_address, "hdmi_spkr", 9)) {
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mHdmiAudioDisabled = true;
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} else {
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mHdmiAudioEvent = false;
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}
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}
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#endif
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checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
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// Propagate device availability to Engine
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mEngine->setDeviceConnectionState(devDesc, state);
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} break;
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default:
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ALOGE("setDeviceConnectionState() invalid state: %x", state);
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return BAD_VALUE;
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}
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// checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
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// output is suspended before any tracks are moved to it
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checkA2dpSuspend();
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checkOutputForAllStrategies();
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// outputs must be closed after checkOutputForAllStrategies() is executed
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if (!outputs.isEmpty()) {
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for (size_t i = 0; i < outputs.size(); i++) {
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sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
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// close unused outputs after device disconnection or direct outputs that have been
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// opened by checkOutputsForDevice() to query dynamic parameters
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if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
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(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
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(desc->mDirectOpenCount == 0))) {
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closeOutput(outputs[i]);
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}
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}
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// check again after closing A2DP output to reset mA2dpSuspended if needed
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checkA2dpSuspend();
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}
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#ifdef FM_POWER_OPT
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// handle FM device connection state to trigger FM AFE loopback
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if (device == AUDIO_DEVICE_OUT_FM && hasPrimaryOutput()) {
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audio_devices_t newDevice = AUDIO_DEVICE_NONE;
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if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
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mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, 1);
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newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false)|AUDIO_DEVICE_OUT_FM);
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mFMIsActive = true;
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mPrimaryOutput->mDevice = newDevice & ~AUDIO_DEVICE_OUT_FM;
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} else {
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newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false));
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mFMIsActive = false;
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mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, -1);
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}
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AudioParameter param = AudioParameter();
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param.addInt(String8("handle_fm"), (int)newDevice);
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mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString());
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}
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#endif /* FM_POWER_OPT end */
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updateDevicesAndOutputs();
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#ifdef DOLBY_ENABLE
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// Before closing the opened outputs, update endpoint property with device capabilities
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audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AUDIO_STREAM_MUSIC), true);
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mDolbyAudioPolicy.setEndpointSystemProperty(audioOutputDevice, mHwModules);
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#endif // DOLBY_END
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if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
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audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
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updateCallRouting(newDevice);
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}
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for (size_t i = 0; i < mOutputs.size(); i++) {
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sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
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if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
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audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
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// do not force device change on duplicated output because if device is 0, it will
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// also force a device 0 for the two outputs it is duplicated to which may override
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// a valid device selection on those outputs.
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bool force = !desc->isDuplicated()
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&& (!device_distinguishes_on_address(device)
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// always force when disconnecting (a non-duplicated device)
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|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
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setOutputDevice(desc, newDevice, force, 0);
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}
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}
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if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
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cleanUpForDevice(devDesc);
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}
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mpClientInterface->onAudioPortListUpdate();
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return NO_ERROR;
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} // end if is output device
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// handle input devices
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if (audio_is_input_device(device)) {
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SortedVector <audio_io_handle_t> inputs;
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ssize_t index = mAvailableInputDevices.indexOf(devDesc);
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switch (state)
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{
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// handle input device connection
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case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
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if (index >= 0) {
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ALOGW("setDeviceConnectionState() device already connected: %d", device);
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return INVALID_OPERATION;
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}
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sp<HwModule> module = mHwModules.getModuleForDevice(device);
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if (module == NULL) {
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ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
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device);
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return INVALID_OPERATION;
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}
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if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) {
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return INVALID_OPERATION;
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}
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index = mAvailableInputDevices.add(devDesc);
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if (index >= 0) {
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mAvailableInputDevices[index]->attach(module);
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} else {
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return NO_MEMORY;
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}
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// Set connect to HALs
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AudioParameter param = AudioParameter(devDesc->mAddress);
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param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
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mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
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// Propagate device availability to Engine
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mEngine->setDeviceConnectionState(devDesc, state);
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} break;
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// handle input device disconnection
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case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
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if (index < 0) {
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ALOGW("setDeviceConnectionState() device not connected: %d", device);
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return INVALID_OPERATION;
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}
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ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
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// Set Disconnect to HALs
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AudioParameter param = AudioParameter(devDesc->mAddress);
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param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
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mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
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checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress);
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mAvailableInputDevices.remove(devDesc);
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// Propagate device availability to Engine
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mEngine->setDeviceConnectionState(devDesc, state);
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} break;
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default:
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ALOGE("setDeviceConnectionState() invalid state: %x", state);
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return BAD_VALUE;
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}
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closeAllInputs();
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if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
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audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
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updateCallRouting(newDevice);
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}
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if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
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cleanUpForDevice(devDesc);
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}
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mpClientInterface->onAudioPortListUpdate();
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return NO_ERROR;
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} // end if is input device
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ALOGW("setDeviceConnectionState() invalid device: %x", device);
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return BAD_VALUE;
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}
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// This function checks for the parameters which can be offloaded.
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// This can be enhanced depending on the capability of the DSP and policy
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// of the system.
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bool AudioPolicyManagerCustom::isOffloadSupported(const audio_offload_info_t& offloadInfo)
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{
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ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
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" BitRate=%u, duration=%" PRId64 " us, has_video=%d",
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offloadInfo.sample_rate, offloadInfo.channel_mask,
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offloadInfo.format,
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offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
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offloadInfo.has_video);
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if (mMasterMono) {
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return false; // no offloading if mono is set.
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}
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// Check if offload has been disabled
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char propValue[PROPERTY_VALUE_MAX];
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if (property_get("audio.offload.disable", propValue, "0")) {
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if (atoi(propValue) != 0) {
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ALOGV("offload disabled by audio.offload.disable=%s", propValue );
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return false;
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}
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}
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// Check if stream type is music, then only allow offload as of now.
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if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
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{
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ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
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return false;
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}
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//check if it's multi-channel AAC (includes sub formats) and FLAC format
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if ((popcount(offloadInfo.channel_mask) > 2) &&
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(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC)||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
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ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
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return false;
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}
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//TODO: enable audio offloading with video when ready
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const bool allowOffloadWithVideo =
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property_get_bool("audio.offload.video", false /* default_value */);
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if (offloadInfo.has_video && !allowOffloadWithVideo) {
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ALOGV("isOffloadSupported: has_video == true, returning false");
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return false;
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}
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//If duration is less than minimum value defined in property, return false
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if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
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if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
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ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
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return false;
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}
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} else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
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ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
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//duration checks only valid for MP3/AAC/ formats,
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//do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
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if ((offloadInfo.format == AUDIO_FORMAT_MP3) ||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS) ||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
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((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE))
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return false;
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}
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// Do not allow offloading if one non offloadable effect is enabled. This prevents from
|
|
// creating an offloaded track and tearing it down immediately after start when audioflinger
|
|
// detects there is an active non offloadable effect.
|
|
// FIXME: We should check the audio session here but we do not have it in this context.
|
|
// This may prevent offloading in rare situations where effects are left active by apps
|
|
// in the background.
|
|
if (mEffects.isNonOffloadableEffectEnabled()) {
|
|
return false;
|
|
}
|
|
// Check for soundcard status
|
|
String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
|
|
String8("SND_CARD_STATUS"));
|
|
AudioParameter result = AudioParameter(valueStr);
|
|
int isonline = 0;
|
|
if ((result.getInt(String8("SND_CARD_STATUS"), isonline) == NO_ERROR)
|
|
&& !isonline) {
|
|
ALOGD("copl: soundcard is offline rejecting offload request");
|
|
return false;
|
|
}
|
|
// See if there is a profile to support this.
|
|
// AUDIO_DEVICE_NONE
|
|
sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
|
|
offloadInfo.sample_rate,
|
|
offloadInfo.format,
|
|
offloadInfo.channel_mask,
|
|
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
|
|
ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
|
|
return (profile != 0);
|
|
}
|
|
audio_devices_t AudioPolicyManagerCustom::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
|
|
bool fromCache)
|
|
{
|
|
audio_devices_t device = AUDIO_DEVICE_NONE;
|
|
|
|
ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
|
|
if (index >= 0) {
|
|
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
|
|
if (patchDesc->mUid != mUidCached) {
|
|
ALOGV("getNewOutputDevice() device %08x forced by patch %d",
|
|
outputDesc->device(), outputDesc->getPatchHandle());
|
|
return outputDesc->device();
|
|
}
|
|
}
|
|
|
|
// check the following by order of priority to request a routing change if necessary:
|
|
// 1: the strategy enforced audible is active and enforced on the output:
|
|
// use device for strategy enforced audible
|
|
// 2: we are in call or the strategy phone is active on the output:
|
|
// use device for strategy phone
|
|
// 3: the strategy for enforced audible is active but not enforced on the output:
|
|
// use the device for strategy enforced audible
|
|
// 4: the strategy sonification is active on the output:
|
|
// use device for strategy sonification
|
|
// 5: the strategy "respectful" sonification is active on the output:
|
|
// use device for strategy "respectful" sonification
|
|
// 6: the strategy accessibility is active on the output:
|
|
// use device for strategy accessibility
|
|
// 7: the strategy media is active on the output:
|
|
// use device for strategy media
|
|
// 8: the strategy DTMF is active on the output:
|
|
// use device for strategy DTMF
|
|
// 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output:
|
|
// use device for strategy t-t-s
|
|
if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) &&
|
|
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
|
|
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
|
|
} else if (isInCall() ||
|
|
isStrategyActive(outputDesc, STRATEGY_PHONE)||
|
|
isStrategyActive(mPrimaryOutput, STRATEGY_PHONE)) {
|
|
device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
|
|
} else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) {
|
|
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
|
|
} else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)||
|
|
(isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION)
|
|
&& (!isStrategyActive(mPrimaryOutput,STRATEGY_MEDIA)))) {
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
|
|
} else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)||
|
|
(isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION_RESPECTFUL)
|
|
&& (!isStrategyActive(mPrimaryOutput, STRATEGY_MEDIA)))) {
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
|
|
} else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) {
|
|
device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
|
|
} else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) {
|
|
device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
|
|
} else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) {
|
|
device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
|
|
} else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
|
|
device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
|
|
} else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) {
|
|
device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
|
|
}
|
|
|
|
ALOGV("getNewOutputDevice() selected device %x", device);
|
|
return device;
|
|
}
|
|
void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)
|
|
{
|
|
ALOGD("setPhoneState() state %d", state);
|
|
// store previous phone state for management of sonification strategy below
|
|
audio_devices_t newDevice = AUDIO_DEVICE_NONE;
|
|
int oldState = mEngine->getPhoneState();
|
|
|
|
if (mEngine->setPhoneState(state) != NO_ERROR) {
|
|
ALOGW("setPhoneState() invalid or same state %d", state);
|
|
return;
|
|
}
|
|
/// Opens: can these line be executed after the switch of volume curves???
|
|
// if leaving call state, handle special case of active streams
|
|
// pertaining to sonification strategy see handleIncallSonification()
|
|
if (isStateInCall(oldState)) {
|
|
ALOGV("setPhoneState() in call state management: new state is %d", state);
|
|
for (size_t j = 0; j < mOutputs.size(); j++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(j);
|
|
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
|
|
if (stream == AUDIO_STREAM_PATCH) {
|
|
continue;
|
|
}
|
|
|
|
handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput);
|
|
}
|
|
}
|
|
|
|
// force reevaluating accessibility routing when call starts
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
|
|
}
|
|
|
|
/**
|
|
* Switching to or from incall state or switching between telephony and VoIP lead to force
|
|
* routing command.
|
|
*/
|
|
bool force = ((is_state_in_call(oldState) != is_state_in_call(state))
|
|
|| (is_state_in_call(state) && (state != oldState)));
|
|
|
|
// check for device and output changes triggered by new phone state
|
|
checkA2dpSuspend();
|
|
checkOutputForAllStrategies();
|
|
updateDevicesAndOutputs();
|
|
|
|
sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
|
|
|
|
int delayMs = 0;
|
|
if (isStateInCall(state)) {
|
|
nsecs_t sysTime = systemTime();
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
// mute media and sonification strategies and delay device switch by the largest
|
|
// latency of any output where either strategy is active.
|
|
// This avoid sending the ring tone or music tail into the earpiece or headset.
|
|
if ((isStrategyActive(desc, STRATEGY_MEDIA,
|
|
SONIFICATION_HEADSET_MUSIC_DELAY,
|
|
sysTime) ||
|
|
isStrategyActive(desc, STRATEGY_SONIFICATION,
|
|
SONIFICATION_HEADSET_MUSIC_DELAY,
|
|
sysTime)) &&
|
|
(delayMs < (int)desc->latency()*2)) {
|
|
delayMs = desc->latency()*2;
|
|
}
|
|
setStrategyMute(STRATEGY_MEDIA, true, desc);
|
|
setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS,
|
|
getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
|
|
setStrategyMute(STRATEGY_SONIFICATION, true, desc);
|
|
setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS,
|
|
getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
|
|
}
|
|
ALOGV("Setting the delay from %dms to %dms", delayMs,
|
|
MIN(delayMs, MAX_VOICE_CALL_START_DELAY_MS));
|
|
delayMs = MIN(delayMs, MAX_VOICE_CALL_START_DELAY_MS);
|
|
}
|
|
|
|
if (hasPrimaryOutput()) {
|
|
// Note that despite the fact that getNewOutputDevice() is called on the primary output,
|
|
// the device returned is not necessarily reachable via this output
|
|
audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
|
|
// force routing command to audio hardware when ending call
|
|
// even if no device change is needed
|
|
if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
|
|
rxDevice = mPrimaryOutput->device();
|
|
}
|
|
|
|
if (state == AUDIO_MODE_IN_CALL) {
|
|
updateCallRouting(rxDevice, delayMs);
|
|
} else if (oldState == AUDIO_MODE_IN_CALL) {
|
|
if (mCallRxPatch != 0) {
|
|
mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
|
|
mCallRxPatch.clear();
|
|
}
|
|
if (mCallTxPatch != 0) {
|
|
mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
|
|
mCallTxPatch.clear();
|
|
}
|
|
setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
|
|
} else {
|
|
setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
|
|
}
|
|
}
|
|
//update device for all non-primary outputs
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
audio_io_handle_t output = mOutputs.keyAt(i);
|
|
if (output != mPrimaryOutput->mIoHandle) {
|
|
newDevice = getNewOutputDevice(mOutputs.valueFor(output), false /*fromCache*/);
|
|
setOutputDevice(mOutputs.valueFor(output), newDevice, (newDevice != AUDIO_DEVICE_NONE));
|
|
}
|
|
}
|
|
// 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 (size_t j = 0; j < mOutputs.size(); j++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(j);
|
|
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
|
|
if (stream == AUDIO_STREAM_PATCH) {
|
|
continue;
|
|
}
|
|
handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput);
|
|
}
|
|
}
|
|
}
|
|
|
|
// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
|
|
if (state == AUDIO_MODE_RINGTONE &&
|
|
isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
|
|
mLimitRingtoneVolume = true;
|
|
} else {
|
|
mLimitRingtoneVolume = false;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
|
|
audio_policy_forced_cfg_t config)
|
|
{
|
|
ALOGD("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
|
|
|
|
if (mEngine->setForceUse(usage, config) != NO_ERROR) {
|
|
ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
|
|
return;
|
|
}
|
|
bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
|
|
(usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
|
|
(usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
|
|
|
|
// check for device and output changes triggered by new force usage
|
|
checkA2dpSuspend();
|
|
checkOutputForAllStrategies();
|
|
updateDevicesAndOutputs();
|
|
|
|
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
|
|
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
|
|
updateCallRouting(newDevice);
|
|
}
|
|
// Use reverse loop to make sure any low latency usecases (generally tones)
|
|
// are not routed before non LL usecases (generally music).
|
|
// We can safely assume that LL output would always have lower index,
|
|
// and use this work-around to avoid routing of output with music stream
|
|
// from the context of short lived LL output.
|
|
// Note: in case output's share backend(HAL sharing is implicit) all outputs
|
|
// gets routing update while processing first output itself.
|
|
for (size_t i = mOutputs.size(); i > 0; i--) {
|
|
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
|
|
audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
|
|
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || outputDesc != mPrimaryOutput) {
|
|
setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE));
|
|
}
|
|
if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
|
|
applyStreamVolumes(outputDesc, newDevice, 0, true);
|
|
}
|
|
}
|
|
|
|
audio_io_handle_t activeInput = mInputs.getActiveInput();
|
|
if (activeInput != 0) {
|
|
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
|
|
audio_devices_t newDevice = getNewInputDevice(activeInput);
|
|
// Force new input selection if the new device can not be reached via current input
|
|
if (activeDesc->mProfile->getSupportedDevices().types() & (newDevice & ~AUDIO_DEVICE_BIT_IN)) {
|
|
setInputDevice(activeInput, newDevice);
|
|
} else {
|
|
closeInput(activeInput);
|
|
}
|
|
}
|
|
|
|
}
|
|
|
|
status_t AudioPolicyManagerCustom::stopSource(sp<AudioOutputDescriptor> outputDesc,
|
|
audio_stream_type_t stream,
|
|
bool forceDeviceUpdate)
|
|
{
|
|
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
|
|
ALOGW("stopSource() invalid stream %d", stream);
|
|
return INVALID_OPERATION;
|
|
}
|
|
// always handle stream stop, check which stream type is stopping
|
|
handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
|
|
|
|
// handle special case for sonification while in call
|
|
if (isInCall() && (outputDesc->mRefCount[stream] == 1)) {
|
|
if (outputDesc->isDuplicated()) {
|
|
handleIncallSonification(stream, false, false, outputDesc->subOutput1()->mIoHandle);
|
|
handleIncallSonification(stream, false, false, outputDesc->subOutput2()->mIoHandle);
|
|
}
|
|
handleIncallSonification(stream, false, false, outputDesc->mIoHandle);
|
|
}
|
|
|
|
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 || forceDeviceUpdate) {
|
|
outputDesc->mStopTime[stream] = systemTime();
|
|
audio_devices_t prevDevice = outputDesc->device();
|
|
audio_devices_t newDevice = getNewOutputDevice(outputDesc, 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(outputDesc, newDevice, false, outputDesc->latency()*2);
|
|
|
|
// force restoring the device selection on other active outputs if it differs from the
|
|
// one being selected for this output
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(i);
|
|
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
if (desc != outputDesc &&
|
|
desc->isActive() &&
|
|
outputDesc->sharesHwModuleWith(desc) &&
|
|
(newDevice != desc->device())) {
|
|
audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/);
|
|
bool force = desc->device() != dev;
|
|
uint32_t delayMs;
|
|
if (dev == prevDevice) {
|
|
delayMs = 0;
|
|
} else {
|
|
delayMs = outputDesc->latency()*2;
|
|
}
|
|
setOutputDevice(desc,
|
|
dev,
|
|
force,
|
|
delayMs);
|
|
// re-apply device specific volume if not done by setOutputDevice()
|
|
if (!force) {
|
|
applyStreamVolumes(desc, dev, delayMs);
|
|
}
|
|
}
|
|
}
|
|
// 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");
|
|
return INVALID_OPERATION;
|
|
}
|
|
}
|
|
status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputDesc,
|
|
audio_stream_type_t stream,
|
|
audio_devices_t device,
|
|
const char *address,
|
|
uint32_t *delayMs)
|
|
{
|
|
// cannot start playback of STREAM_TTS if any other output is being used
|
|
uint32_t beaconMuteLatency = 0;
|
|
|
|
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
|
|
ALOGW("startSource() invalid stream %d", stream);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
*delayMs = 0;
|
|
if (stream == AUDIO_STREAM_TTS) {
|
|
ALOGV("\t found BEACON stream");
|
|
if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
|
|
return INVALID_OPERATION;
|
|
} else {
|
|
beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
|
|
}
|
|
} else {
|
|
// some playback other than beacon starts
|
|
beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
|
|
}
|
|
|
|
// check active before incrementing usage count
|
|
bool force = !outputDesc->isActive();
|
|
|
|
// 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 || device != AUDIO_DEVICE_NONE) {
|
|
// starting an output being rerouted?
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
device = getNewOutputDevice(outputDesc, false /*fromCache*/);
|
|
}
|
|
routing_strategy strategy = getStrategy(stream);
|
|
bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
|
|
(strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
|
|
(beaconMuteLatency > 0);
|
|
uint32_t waitMs = beaconMuteLatency;
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
if (desc != outputDesc) {
|
|
// force a device change if any other output is managed by the same hw
|
|
// module and has a current device selection that differs from selected device.
|
|
// In this case, the audio HAL must receive the new device selection so that it can
|
|
// change the device currently selected by the other active output.
|
|
if (outputDesc->sharesHwModuleWith(desc) &&
|
|
desc->device() != device) {
|
|
force = true;
|
|
}
|
|
// wait for audio on other active outputs to be presented when starting
|
|
// a notification so that audio focus effect can propagate, or that a mute/unmute
|
|
// event occurred for beacon
|
|
uint32_t latency = desc->latency();
|
|
if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
|
|
waitMs = latency;
|
|
}
|
|
}
|
|
}
|
|
uint32_t muteWaitMs;
|
|
muteWaitMs = setOutputDevice(outputDesc, device, force, 0, NULL, address);
|
|
|
|
// handle special case for sonification while in call
|
|
if (isInCall()) {
|
|
handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
|
|
}
|
|
|
|
// apply volume rules for current stream and device if necessary
|
|
checkAndSetVolume(stream,
|
|
mVolumeCurves->getVolumeIndex(stream, device),
|
|
outputDesc,
|
|
device);
|
|
|
|
// update the outputs if starting an output with a stream that can affect notification
|
|
// routing
|
|
handleNotificationRoutingForStream(stream);
|
|
|
|
// force reevaluating accessibility routing when ringtone or alarm starts
|
|
if (strategy == STRATEGY_SONIFICATION) {
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
|
|
}
|
|
}
|
|
else {
|
|
// handle special case for sonification while in call
|
|
if (isInCall()) {
|
|
handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
|
|
}
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
void AudioPolicyManagerCustom::handleIncallSonification(audio_stream_type_t stream,
|
|
bool starting, bool stateChange,
|
|
audio_io_handle_t output)
|
|
{
|
|
if(!hasPrimaryOutput()) {
|
|
return;
|
|
}
|
|
// no action needed for AUDIO_STREAM_PATCH stream type, it's for internal flinger tracks
|
|
if (stream == AUDIO_STREAM_PATCH) {
|
|
return;
|
|
}
|
|
// if the stream pertains to sonification strategy and we are in call we must
|
|
// mute the stream if it is low visibility. If it is high visibility, we must play a tone
|
|
// in the device used for phone strategy and play the tone if the selected device does not
|
|
// interfere with the device used for phone strategy
|
|
// if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
|
|
// many times as there are active tracks on the output
|
|
const routing_strategy stream_strategy = getStrategy(stream);
|
|
if ((stream_strategy == STRATEGY_SONIFICATION) ||
|
|
((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
|
|
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
|
|
ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
|
|
stream, starting, outputDesc->mDevice, stateChange);
|
|
if (outputDesc->mRefCount[stream]) {
|
|
int muteCount = 1;
|
|
if (stateChange) {
|
|
muteCount = outputDesc->mRefCount[stream];
|
|
}
|
|
if (audio_is_low_visibility(stream)) {
|
|
ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
|
|
for (int i = 0; i < muteCount; i++) {
|
|
setStreamMute(stream, starting, outputDesc);
|
|
}
|
|
} 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, outputDesc);
|
|
}
|
|
}
|
|
if (starting) {
|
|
mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
|
|
AUDIO_STREAM_VOICE_CALL);
|
|
} else {
|
|
mpClientInterface->stopTone();
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
void AudioPolicyManagerCustom::handleNotificationRoutingForStream(audio_stream_type_t stream) {
|
|
switch(stream) {
|
|
case AUDIO_STREAM_MUSIC:
|
|
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
|
|
updateDevicesAndOutputs();
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream,
|
|
int index,
|
|
const sp<AudioOutputDescriptor>& outputDesc,
|
|
audio_devices_t device,
|
|
int delayMs, bool force)
|
|
{
|
|
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
|
|
ALOGW("checkAndSetVolume() invalid stream %d", stream);
|
|
return INVALID_OPERATION;
|
|
}
|
|
// do not change actual stream volume if the stream is muted
|
|
if (outputDesc->mMuteCount[stream] != 0) {
|
|
ALOGVV("checkAndSetVolume() stream %d muted count %d",
|
|
stream, outputDesc->mMuteCount[stream]);
|
|
return NO_ERROR;
|
|
}
|
|
audio_policy_forced_cfg_t forceUseForComm =
|
|
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION);
|
|
// do not change in call volume if bluetooth is connected and vice versa
|
|
if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
|
|
(stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) {
|
|
ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
|
|
stream, forceUseForComm);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
device = outputDesc->device();
|
|
}
|
|
|
|
float volumeDb = computeVolume(stream, index, device);
|
|
if (outputDesc->isFixedVolume(device)) {
|
|
volumeDb = 0.0f;
|
|
}
|
|
|
|
outputDesc->setVolume(volumeDb, stream, device, delayMs, force);
|
|
|
|
if (stream == AUDIO_STREAM_VOICE_CALL ||
|
|
stream == AUDIO_STREAM_BLUETOOTH_SCO) {
|
|
float voiceVolume;
|
|
// Force voice volume to max for bluetooth SCO as volume is managed by the headset
|
|
if (stream == AUDIO_STREAM_VOICE_CALL) {
|
|
voiceVolume = (float)index/(float)mVolumeCurves->getVolumeIndexMax(stream);
|
|
} else {
|
|
voiceVolume = 1.0;
|
|
}
|
|
|
|
if (voiceVolume != mLastVoiceVolume && ((outputDesc == mPrimaryOutput) ||
|
|
isDirectOutput(outputDesc->mIoHandle) || device & AUDIO_DEVICE_OUT_ALL_USB)) {
|
|
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
|
|
mLastVoiceVolume = voiceVolume;
|
|
}
|
|
#ifdef FM_POWER_OPT
|
|
} else if (stream == AUDIO_STREAM_MUSIC && hasPrimaryOutput() &&
|
|
outputDesc == mPrimaryOutput && mFMIsActive) {
|
|
/* Avoid unnecessary set_parameter calls as it puts the primary
|
|
outputs FastMixer in HOT_IDLE leading to breaks in audio */
|
|
if (volumeDb != mPrevFMVolumeDb) {
|
|
mPrevFMVolumeDb = volumeDb;
|
|
AudioParameter param = AudioParameter();
|
|
param.addFloat(String8("fm_volume"), Volume::DbToAmpl(volumeDb));
|
|
//Double delayMs to avoid sound burst while device switch.
|
|
mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString(), delayMs*2);
|
|
}
|
|
#endif /* FM_POWER_OPT end */
|
|
}
|
|
|
|
return NO_ERROR;
|
|
}
|
|
bool AudioPolicyManagerCustom::isDirectOutput(audio_io_handle_t output) {
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(i);
|
|
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice(
|
|
audio_devices_t device,
|
|
audio_session_t session __unused,
|
|
audio_stream_type_t stream,
|
|
uint32_t samplingRate,
|
|
audio_format_t format,
|
|
audio_channel_mask_t channelMask,
|
|
audio_output_flags_t flags,
|
|
const audio_offload_info_t *offloadInfo)
|
|
{
|
|
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
|
|
status_t status;
|
|
|
|
#ifdef AUDIO_POLICY_TEST
|
|
if (mCurOutput != 0) {
|
|
ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
|
|
mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
|
|
|
|
if (mTestOutputs[mCurOutput] == 0) {
|
|
ALOGV("getOutput() opening test output");
|
|
sp<AudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(NULL,
|
|
mpClientInterface);
|
|
outputDesc->mDevice = mTestDevice;
|
|
outputDesc->mLatency = mTestLatencyMs;
|
|
outputDesc->mFlags =
|
|
(audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
|
|
outputDesc->mRefCount[stream] = 0;
|
|
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
|
|
config.sample_rate = mTestSamplingRate;
|
|
config.channel_mask = mTestChannels;
|
|
config.format = mTestFormat;
|
|
if (offloadInfo != NULL) {
|
|
config.offload_info = *offloadInfo;
|
|
}
|
|
status = mpClientInterface->openOutput(0,
|
|
&mTestOutputs[mCurOutput],
|
|
&config,
|
|
&outputDesc->mDevice,
|
|
String8(""),
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
if (status == NO_ERROR) {
|
|
outputDesc->mSamplingRate = config.sample_rate;
|
|
outputDesc->mFormat = config.format;
|
|
outputDesc->mChannelMask = config.channel_mask;
|
|
AudioParameter outputCmd = AudioParameter();
|
|
outputCmd.addInt(String8("set_id"),mCurOutput);
|
|
mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
|
|
addOutput(mTestOutputs[mCurOutput], outputDesc);
|
|
}
|
|
}
|
|
return mTestOutputs[mCurOutput];
|
|
}
|
|
#endif //AUDIO_POLICY_TEST
|
|
if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
|
|
(stream != AUDIO_STREAM_MUSIC)) {
|
|
// compress should not be used for non-music streams
|
|
ALOGE("Offloading only allowed with music stream");
|
|
return 0;
|
|
}
|
|
|
|
if ((stream == AUDIO_STREAM_VOICE_CALL) &&
|
|
(channelMask == 1) &&
|
|
(samplingRate == 8000 || samplingRate == 16000)) {
|
|
// Allow Voip direct output only if:
|
|
// audio mode is MODE_IN_COMMUNCATION; AND
|
|
// voip output is not opened already; AND
|
|
// requested sample rate matches with that of voip input stream (if opened already)
|
|
int value = 0;
|
|
uint32_t mode = 0, voipOutCount = 1, voipSampleRate = 1;
|
|
String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
|
|
String8("audio_mode"));
|
|
AudioParameter result = AudioParameter(valueStr);
|
|
if (result.getInt(String8("audio_mode"), value) == NO_ERROR) {
|
|
mode = value;
|
|
}
|
|
|
|
valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
|
|
String8("voip_out_stream_count"));
|
|
result = AudioParameter(valueStr);
|
|
if (result.getInt(String8("voip_out_stream_count"), value) == NO_ERROR) {
|
|
voipOutCount = value;
|
|
}
|
|
|
|
valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
|
|
String8("voip_sample_rate"));
|
|
result = AudioParameter(valueStr);
|
|
if (result.getInt(String8("voip_sample_rate"), value) == NO_ERROR) {
|
|
voipSampleRate = value;
|
|
}
|
|
|
|
if ((mode == AUDIO_MODE_IN_COMMUNICATION) && (voipOutCount == 0) &&
|
|
((voipSampleRate == 0) || (voipSampleRate == samplingRate))) {
|
|
if (audio_is_linear_pcm(format)) {
|
|
char propValue[PROPERTY_VALUE_MAX] = {0};
|
|
property_get("use.voice.path.for.pcm.voip", propValue, "0");
|
|
bool voipPcmSysPropEnabled = !strncmp("true", propValue, sizeof("true"));
|
|
if (voipPcmSysPropEnabled && (format == AUDIO_FORMAT_PCM_16_BIT)) {
|
|
flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
|
|
AUDIO_OUTPUT_FLAG_DIRECT);
|
|
ALOGD("Set VoIP and Direct output flags for PCM format");
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
|
|
/*
|
|
* WFD audio routes back to target speaker when starting a ringtone playback.
|
|
* This is because primary output is reused for ringtone, so output device is
|
|
* updated based on SONIFICATION strategy for both ringtone and music playback.
|
|
* The same issue is not seen on remoted_submix HAL based WFD audio because
|
|
* primary output is not reused and a new output is created for ringtone playback.
|
|
* Issue is fixed by updating output flag to AUDIO_OUTPUT_FLAG_FAST when there is
|
|
* a non-music stream playback on WFD, so primary output is not reused for ringtone.
|
|
*/
|
|
audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
|
|
if ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY)
|
|
&& (stream != AUDIO_STREAM_MUSIC)) {
|
|
ALOGD("WFD audio: use OUTPUT_FLAG_FAST for non music stream. flags:%x", flags );
|
|
//For voip paths
|
|
if(flags & AUDIO_OUTPUT_FLAG_DIRECT)
|
|
flags = AUDIO_OUTPUT_FLAG_DIRECT;
|
|
else //route every thing else to ULL path
|
|
flags = AUDIO_OUTPUT_FLAG_FAST;
|
|
}
|
|
#endif
|
|
// open a direct output if required by specified parameters
|
|
//force direct flag if offload flag is set: offloading implies a direct output stream
|
|
// and all common behaviors are driven by checking only the direct flag
|
|
// this should normally be set appropriately in the policy configuration file
|
|
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
|
|
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
|
|
}
|
|
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
|
|
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
|
|
}
|
|
// only allow deep buffering for music stream type
|
|
if (stream != AUDIO_STREAM_MUSIC) {
|
|
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
|
|
} else if (/* stream == AUDIO_STREAM_MUSIC && */
|
|
flags == AUDIO_OUTPUT_FLAG_NONE &&
|
|
property_get_bool("audio.deep_buffer.media", true/* default_value */)) {
|
|
flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
|
|
}
|
|
|
|
if (stream == AUDIO_STREAM_TTS) {
|
|
flags = AUDIO_OUTPUT_FLAG_TTS;
|
|
}
|
|
|
|
sp<IOProfile> profile;
|
|
|
|
// skip direct output selection if the request can obviously be attached to a mixed output
|
|
// and not explicitly requested
|
|
if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
|
|
audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX &&
|
|
audio_channel_count_from_out_mask(channelMask) <= 2) {
|
|
goto non_direct_output;
|
|
}
|
|
|
|
// Do not allow offloading if one non offloadable effect is enabled. This prevents from
|
|
// creating an offloaded track and tearing it down immediately after start when audioflinger
|
|
// detects there is an active non offloadable effect.
|
|
// FIXME: We should check the audio session here but we do not have it in this context.
|
|
// This may prevent offloading in rare situations where effects are left active by apps
|
|
// in the background.
|
|
|
|
if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
|
|
!(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {
|
|
profile = getProfileForDirectOutput(device,
|
|
samplingRate,
|
|
format,
|
|
channelMask,
|
|
(audio_output_flags_t)flags);
|
|
}
|
|
|
|
if (profile != 0) {
|
|
sp<SwAudioOutputDescriptor> outputDesc = NULL;
|
|
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<SwAudioOutputDescriptor> 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) &&
|
|
audio_formats_match(format, outputDesc->mFormat) &&
|
|
(channelMask == outputDesc->mChannelMask)) {
|
|
outputDesc->mDirectOpenCount++;
|
|
ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
|
|
return mOutputs.keyAt(i);
|
|
}
|
|
}
|
|
}
|
|
// close direct output if currently open and configured with different parameters
|
|
if (outputDesc != NULL) {
|
|
closeOutput(outputDesc->mIoHandle);
|
|
}
|
|
|
|
// if the selected profile is offloaded and no offload info was specified,
|
|
// create a default one
|
|
audio_offload_info_t defaultOffloadInfo = AUDIO_INFO_INITIALIZER;
|
|
if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) {
|
|
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
|
|
defaultOffloadInfo.sample_rate = samplingRate;
|
|
defaultOffloadInfo.channel_mask = channelMask;
|
|
defaultOffloadInfo.format = format;
|
|
defaultOffloadInfo.stream_type = stream;
|
|
defaultOffloadInfo.bit_rate = 0;
|
|
defaultOffloadInfo.duration_us = -1;
|
|
defaultOffloadInfo.has_video = true; // conservative
|
|
defaultOffloadInfo.is_streaming = true; // likely
|
|
offloadInfo = &defaultOffloadInfo;
|
|
}
|
|
|
|
outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface);
|
|
outputDesc->mDevice = device;
|
|
outputDesc->mLatency = 0;
|
|
outputDesc->mFlags = (audio_output_flags_t)(outputDesc->mFlags | flags);
|
|
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
|
|
config.sample_rate = samplingRate;
|
|
config.channel_mask = channelMask;
|
|
config.format = format;
|
|
if (offloadInfo != NULL) {
|
|
config.offload_info = *offloadInfo;
|
|
}
|
|
status = mpClientInterface->openOutput(profile->getModuleHandle(),
|
|
&output,
|
|
&config,
|
|
&outputDesc->mDevice,
|
|
String8(""),
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
|
|
// only accept an output with the requested parameters
|
|
if (status != NO_ERROR ||
|
|
(samplingRate != 0 && samplingRate != config.sample_rate) ||
|
|
(format != AUDIO_FORMAT_DEFAULT && !audio_formats_match(format, config.format)) ||
|
|
(channelMask != 0 && channelMask != config.channel_mask)) {
|
|
ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
|
|
"format %d %d, channelMask %04x %04x", output, samplingRate,
|
|
outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
|
|
outputDesc->mChannelMask);
|
|
if (output != AUDIO_IO_HANDLE_NONE) {
|
|
mpClientInterface->closeOutput(output);
|
|
}
|
|
// fall back to mixer output if possible when the direct output could not be open
|
|
if (audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX) {
|
|
goto non_direct_output;
|
|
}
|
|
return AUDIO_IO_HANDLE_NONE;
|
|
}
|
|
outputDesc->mSamplingRate = config.sample_rate;
|
|
outputDesc->mChannelMask = config.channel_mask;
|
|
outputDesc->mFormat = config.format;
|
|
outputDesc->mRefCount[stream] = 0;
|
|
outputDesc->mStopTime[stream] = 0;
|
|
outputDesc->mDirectOpenCount = 1;
|
|
|
|
audio_io_handle_t srcOutput = getOutputForEffect();
|
|
addOutput(output, outputDesc);
|
|
audio_io_handle_t dstOutput = getOutputForEffect();
|
|
if (dstOutput == output) {
|
|
#ifdef DOLBY_ENABLE
|
|
status_t status = mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
|
|
if (status == NO_ERROR) {
|
|
for (size_t i = 0; i < mEffects.size(); i++) {
|
|
sp<EffectDescriptor> desc = mEffects.valueAt(i);
|
|
if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX) {
|
|
// update the mIo member of EffectDescriptor for the global effect
|
|
ALOGV("%s updating mIo", __FUNCTION__);
|
|
desc->mIo = dstOutput;
|
|
}
|
|
}
|
|
} else {
|
|
ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__, srcOutput, dstOutput);
|
|
}
|
|
#else // DOLBY_END
|
|
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
|
|
#endif // LINE_ADDED_BY_DOLBY
|
|
}
|
|
mPreviousOutputs = mOutputs;
|
|
ALOGV("getOutput() returns new direct output %d", output);
|
|
mpClientInterface->onAudioPortListUpdate();
|
|
return output;
|
|
}
|
|
|
|
non_direct_output:
|
|
|
|
// A request for HW A/V sync cannot fallback to a mixed output because time
|
|
// stamps are embedded in audio data
|
|
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
|
|
return AUDIO_IO_HANDLE_NONE;
|
|
}
|
|
|
|
// ignoring channel mask due to downmix capability in mixer
|
|
|
|
// open a non direct output
|
|
|
|
// for non direct outputs, only PCM is supported
|
|
if (audio_is_linear_pcm(format)) {
|
|
// get which output is suitable for the specified stream. The actual
|
|
// routing change will happen when startOutput() will be called
|
|
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
|
|
|
|
// at this stage we should ignore the DIRECT flag as no direct output could be found earlier
|
|
flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
|
|
output = selectOutput(outputs, flags, format);
|
|
}
|
|
ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
|
|
"format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
|
|
|
|
ALOGV(" getOutputForDevice() returns output %d", output);
|
|
|
|
return output;
|
|
}
|
|
|
|
AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
|
|
: AudioPolicyManager(clientInterface),
|
|
mHdmiAudioDisabled(false),
|
|
mHdmiAudioEvent(false),
|
|
mPrevPhoneState(0),
|
|
mPrevFMVolumeDb(0.0f),
|
|
mFMIsActive(false)
|
|
{
|
|
|
|
//TODO: Check the new logic to parse policy conf and update the below code
|
|
// Need this when SSR encoding is enabled
|
|
char ssr_enabled[PROPERTY_VALUE_MAX] = {0};
|
|
bool prop_ssr_enabled = false;
|
|
|
|
if (property_get("ro.qc.sdk.audio.ssr", ssr_enabled, NULL)) {
|
|
prop_ssr_enabled = atoi(ssr_enabled) || !strncmp("true", ssr_enabled, 4);
|
|
}
|
|
|
|
for (size_t i = 0; i < mHwModules.size(); i++) {
|
|
ALOGV("Hw module %zu", i);
|
|
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
|
|
const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
|
|
AudioProfileVector profiles = inProfile->getAudioProfiles();
|
|
for (size_t k = 0; k < profiles.size(); k++){
|
|
ChannelsVector channels = profiles[k]->getChannels();
|
|
for (size_t x = 0; x < channels.size(); x++) {
|
|
audio_channel_mask_t channelMask = channels[x];
|
|
ALOGV("Channel Mask %x size %zu", channelMask,
|
|
channels.size());
|
|
if (AUDIO_CHANNEL_IN_5POINT1 == channelMask) {
|
|
if (!prop_ssr_enabled) {
|
|
ALOGI("removing AUDIO_CHANNEL_IN_5POINT1 from"
|
|
" input profile as SSR(surround sound record)"
|
|
" is not supported on this chipset variant");
|
|
channels.removeItemsAt(x, 1);
|
|
ALOGV("Channel Mask size now %zu",
|
|
channels.size());
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
#ifdef RECORD_PLAY_CONCURRENCY
|
|
mIsInputRequestOnProgress = false;
|
|
#endif
|
|
|
|
|
|
#ifdef VOICE_CONCURRENCY
|
|
mFallBackflag = getFallBackPath();
|
|
#endif
|
|
}
|
|
|
|
}
|