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android_mt6572_jiabo/frameworks/av/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
FuQuan 3ff2876a30 first commit
2025-09-05 16:56:03 +08:00

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/*
* 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.
*
* This file was modified by Dolby Laboratories, Inc. The portions of the
* code that are surrounded by "DOLBY..." are copyrighted and
* licensed separately, as follows:
*
* (C) 2014-2016 Dolby Laboratories, Inc.
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#define LOG_TAG "APM_AudioPolicyManager"
//#define LOG_NDEBUG 0
//#define VERY_VERBOSE_LOGGING
#ifdef VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
#define AUDIO_POLICY_XML_CONFIG_FILE "/system/etc/audio_policy_configuration.xml"
#include <inttypes.h>
#include <math.h>
#include <AudioPolicyManagerInterface.h>
#include <AudioPolicyEngineInstance.h>
#include <cutils/properties.h>
#include <utils/Log.h>
#include <hardware/audio.h>
#include <hardware/audio_effect.h>
#include <media/AudioParameter.h>
#include <media/AudioPolicyHelper.h>
#include <soundtrigger/SoundTrigger.h>
#include "AudioPolicyManager.h"
#ifndef USE_XML_AUDIO_POLICY_CONF
#include <ConfigParsingUtils.h>
#include <StreamDescriptor.h>
#endif
#include <Serializer.h>
#include "TypeConverter.h"
#include <policy.h>
#ifdef DOLBY_ENABLE
#include "DolbyAudioPolicy_impl.h"
#endif // DOLBY_END
namespace android {
//FIXME: workaround for truncated touch sounds
// to be removed when the problem is handled by system UI
#define TOUCH_SOUND_FIXED_DELAY_MS 100
// ----------------------------------------------------------------------------
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
const char *device_name)
{
return setDeviceConnectionStateInt(device, state, device_address, device_name);
}
status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
const char *device_name)
{
ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
- device, state, device_address, device_name);
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
sp<DeviceDescriptor> devDesc =
mHwModules.getDeviceDescriptor(device, device_address, device_name);
// handle output devices
if (audio_is_output_device(device)) {
SortedVector <audio_io_handle_t> outputs;
ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
// save a copy of the opened output descriptors before any output is opened or closed
// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
mPreviousOutputs = mOutputs;
switch (state)
{
// handle output device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
ALOGW("setDeviceConnectionState() device already connected: %x", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() connecting device %x", device);
// register new device as available
index = mAvailableOutputDevices.add(devDesc);
if (index >= 0) {
sp<HwModule> module = mHwModules.getModuleForDevice(device);
if (module == 0) {
ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
device);
mAvailableOutputDevices.remove(devDesc);
return INVALID_OPERATION;
}
mAvailableOutputDevices[index]->attach(module);
} else {
return NO_MEMORY;
}
if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
mAvailableOutputDevices.remove(devDesc);
return INVALID_OPERATION;
}
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
// outputs should never be empty here
ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
"checkOutputsForDevice() returned no outputs but status OK");
ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
outputs.size());
// Send connect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
} break;
// handle output device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
ALOGW("setDeviceConnectionState() device not connected: %x", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
// Send Disconnect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
// remove device from available output devices
mAvailableOutputDevices.remove(devDesc);
checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
} break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
// checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
// output is suspended before any tracks are moved to it
checkA2dpSuspend();
checkOutputForAllStrategies();
// outputs must be closed after checkOutputForAllStrategies() is executed
if (!outputs.isEmpty()) {
for (size_t i = 0; i < outputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
// close unused outputs after device disconnection or direct outputs that have been
// opened by checkOutputsForDevice() to query dynamic parameters
if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
(desc->mDirectOpenCount == 0))) {
closeOutput(outputs[i]);
}
}
// check again after closing A2DP output to reset mA2dpSuspended if needed
checkA2dpSuspend();
}
updateDevicesAndOutputs();
#ifdef DOLBY_ENABLE // DOLBY_UDC
// Before closing the opened outputs, update endpoint property with device capabilities
audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AUDIO_STREAM_MUSIC), true);
mDolbyAudioPolicy.setEndpointSystemProperty(audioOutputDevice, mHwModules);
#endif // DOLBY_END
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevice);
}
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
// do not force device change on duplicated output because if device is 0, it will
// also force a device 0 for the two outputs it is duplicated to which may override
// a valid device selection on those outputs.
bool force = !desc->isDuplicated()
&& (!device_distinguishes_on_address(device)
// always force when disconnecting (a non-duplicated device)
|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
setOutputDevice(desc, newDevice, force, 0);
}
}
if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
cleanUpForDevice(devDesc);
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is output device
// handle input devices
if (audio_is_input_device(device)) {
SortedVector <audio_io_handle_t> inputs;
ssize_t index = mAvailableInputDevices.indexOf(devDesc);
switch (state)
{
// handle input device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
ALOGW("setDeviceConnectionState() device already connected: %d", device);
return INVALID_OPERATION;
}
sp<HwModule> module = mHwModules.getModuleForDevice(device);
if (module == NULL) {
ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
device);
return INVALID_OPERATION;
}
if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) {
return INVALID_OPERATION;
}
index = mAvailableInputDevices.add(devDesc);
if (index >= 0) {
mAvailableInputDevices[index]->attach(module);
} else {
return NO_MEMORY;
}
// Set connect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
} break;
// handle input device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
ALOGW("setDeviceConnectionState() device not connected: %d", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
// Set Disconnect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress);
mAvailableInputDevices.remove(devDesc);
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
} break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
closeAllInputs();
// As the input device list can impact the output device selection, update
// getDeviceForStrategy() cache
updateDevicesAndOutputs();
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevice);
}
if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
cleanUpForDevice(devDesc);
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is input device
ALOGW("setDeviceConnectionState() invalid device: %x", device);
return BAD_VALUE;
}
audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
const char *device_address)
{
sp<DeviceDescriptor> devDesc =
mHwModules.getDeviceDescriptor(device, device_address, "",
(strlen(device_address) != 0)/*matchAddress*/);
if (devDesc == 0) {
ALOGW("getDeviceConnectionState() undeclared device, type %08x, address: %s",
device, device_address);
return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
DeviceVector *deviceVector;
if (audio_is_output_device(device)) {
deviceVector = &mAvailableOutputDevices;
} else if (audio_is_input_device(device)) {
deviceVector = &mAvailableInputDevices;
} else {
ALOGW("getDeviceConnectionState() invalid device type %08x", device);
return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
return (deviceVector->getDevice(device, String8(device_address)) != 0) ?
AUDIO_POLICY_DEVICE_STATE_AVAILABLE : AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
uint32_t AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, uint32_t delayMs)
{
bool createTxPatch = false;
status_t status;
audio_patch_handle_t afPatchHandle;
DeviceVector deviceList;
uint32_t muteWaitMs = 0;
if(!hasPrimaryOutput() || mPrimaryOutput->device() == AUDIO_DEVICE_OUT_STUB) {
return muteWaitMs;
}
audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice);
// release existing RX patch if any
if (mCallRxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
mCallRxPatch.clear();
}
// release TX patch if any
if (mCallTxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
mCallTxPatch.clear();
}
// If the RX device is on the primary HW module, then use legacy routing method for voice calls
// via setOutputDevice() on primary output.
// Otherwise, create two audio patches for TX and RX path.
if (availablePrimaryOutputDevices() & rxDevice) {
muteWaitMs = setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs);
// If the TX device is also on the primary HW module, setOutputDevice() will take care
// of it due to legacy implementation. If not, create a patch.
if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN)
== AUDIO_DEVICE_NONE) {
createTxPatch = true;
}
} else { // create RX path audio patch
struct audio_patch patch;
patch.num_sources = 1;
patch.num_sinks = 1;
deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice);
ALOG_ASSERT(!deviceList.isEmpty(),
"updateCallRouting() selected device not in output device list");
sp<DeviceDescriptor> rxSinkDeviceDesc = deviceList.itemAt(0);
deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX);
ALOG_ASSERT(!deviceList.isEmpty(),
"updateCallRouting() no telephony RX device");
sp<DeviceDescriptor> rxSourceDeviceDesc = deviceList.itemAt(0);
rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
// request to reuse existing output stream if one is already opened to reach the RX device
SortedVector<audio_io_handle_t> outputs =
getOutputsForDevice(rxDevice, mOutputs);
audio_io_handle_t output = selectOutput(outputs,
AUDIO_OUTPUT_FLAG_NONE,
AUDIO_FORMAT_INVALID);
if (output != AUDIO_IO_HANDLE_NONE) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ALOG_ASSERT(!outputDesc->isDuplicated(),
"updateCallRouting() RX device output is duplicated");
outputDesc->toAudioPortConfig(&patch.sources[1]);
patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH;
patch.num_sources = 2;
}
afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, delayMs);
ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch",
status);
if (status == NO_ERROR) {
mCallRxPatch = new AudioPatch(&patch, mUidCached);
mCallRxPatch->mAfPatchHandle = afPatchHandle;
mCallRxPatch->mUid = mUidCached;
}
createTxPatch = true;
}
if (createTxPatch) { // create TX path audio patch
struct audio_patch patch;
patch.num_sources = 1;
patch.num_sinks = 1;
deviceList = mAvailableInputDevices.getDevicesFromType(txDevice);
ALOG_ASSERT(!deviceList.isEmpty(),
"updateCallRouting() selected device not in input device list");
sp<DeviceDescriptor> txSourceDeviceDesc = deviceList.itemAt(0);
txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX);
ALOG_ASSERT(!deviceList.isEmpty(),
"updateCallRouting() no telephony TX device");
sp<DeviceDescriptor> txSinkDeviceDesc = deviceList.itemAt(0);
txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
SortedVector<audio_io_handle_t> outputs =
getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs);
audio_io_handle_t output = selectOutput(outputs,
AUDIO_OUTPUT_FLAG_NONE,
AUDIO_FORMAT_INVALID);
// request to reuse existing output stream if one is already opened to reach the TX
// path output device
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ALOG_ASSERT(!outputDesc->isDuplicated(),
"updateCallRouting() RX device output is duplicated");
outputDesc->toAudioPortConfig(&patch.sources[1]);
patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH;
patch.num_sources = 2;
}
// terminate active capture if on the same HW module as the call TX source device
// FIXME: would be better to refine to only inputs whose profile connects to the
// call TX device but this information is not in the audio patch and logic here must be
// symmetric to the one in startInput()
audio_io_handle_t activeInput = mInputs.getActiveInput();
if (activeInput != 0) {
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
if (activeDesc->getModuleHandle() == txSourceDeviceDesc->getModuleHandle()) {
//FIXME: consider all active sessions
AudioSessionCollection activeSessions = activeDesc->getActiveAudioSessions();
audio_session_t activeSession = activeSessions.keyAt(0);
stopInput(activeInput, activeSession);
releaseInput(activeInput, activeSession);
}
}
afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, delayMs);
ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch",
status);
if (status == NO_ERROR) {
mCallTxPatch = new AudioPatch(&patch, mUidCached);
mCallTxPatch->mAfPatchHandle = afPatchHandle;
mCallTxPatch->mUid = mUidCached;
}
}
return muteWaitMs;
}
void AudioPolicyManager::setPhoneState(audio_mode_t state)
{
ALOGV("setPhoneState() state %d", state);
// store previous phone state for management of sonification strategy below
int oldState = mEngine->getPhoneState();
if (mEngine->setPhoneState(state) != NO_ERROR) {
ALOGW("setPhoneState() invalid or same state %d", state);
return;
}
/// Opens: can these line be executed after the switch of volume curves???
// if leaving call state, handle special case of active streams
// pertaining to sonification strategy see handleIncallSonification()
if (isStateInCall(oldState)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
handleIncallSonification((audio_stream_type_t)stream, false, true);
}
// force reevaluating accessibility routing when call stops
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();
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*/));
}
}
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);
}
}
// if entering in call state, handle special case of active streams
// pertaining to sonification strategy see handleIncallSonification()
if (isStateInCall(state)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
handleIncallSonification((audio_stream_type_t)stream, true, true);
}
// force reevaluating accessibility routing when call starts
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
if (state == AUDIO_MODE_RINGTONE &&
isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
mLimitRingtoneVolume = true;
} else {
mLimitRingtoneVolume = false;
}
}
audio_mode_t AudioPolicyManager::getPhoneState() {
return mEngine->getPhoneState();
}
void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config)
{
ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
if (mEngine->setForceUse(usage, config) != NO_ERROR) {
ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
return;
}
bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
(usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
(usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
// check for device and output changes triggered by new force usage
checkA2dpSuspend();
checkOutputForAllStrategies();
updateDevicesAndOutputs();
//FIXME: workaround for truncated touch sounds
// to be removed when the problem is handled by system UI
uint32_t delayMs = 0;
uint32_t waitMs = 0;
if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) {
delayMs = TOUCH_SOUND_FIXED_DELAY_MS;
}
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
waitMs = updateCallRouting(newDevice, delayMs);
}
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
waitMs = setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE),
delayMs);
}
if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
applyStreamVolumes(outputDesc, newDevice, waitMs, 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);
}
}
}
void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
{
ALOGV("setSystemProperty() property %s, value %s", property, value);
}
// Find a direct output profile compatible with the parameters passed, even if the input flags do
// not explicitly request a direct output
sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput(
audio_devices_t device,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags)
{
// only retain flags that will drive the direct output profile selection
// if explicitly requested
static const uint32_t kRelevantFlags =
(AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
AUDIO_OUTPUT_FLAG_VOIP_RX);
flags =
(audio_output_flags_t)((flags & kRelevantFlags) | AUDIO_OUTPUT_FLAG_DIRECT);
sp<IOProfile> profile;
for (size_t i = 0; i < mHwModules.size(); i++) {
if (mHwModules[i]->mHandle == 0) {
continue;
}
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
sp<IOProfile> curProfile = mHwModules[i]->mOutputProfiles[j];
if (!curProfile->isCompatibleProfile(device, String8(""),
samplingRate, NULL /*updatedSamplingRate*/,
format, NULL /*updatedFormat*/,
channelMask, NULL /*updatedChannelMask*/,
flags)) {
continue;
}
// reject profiles not corresponding to a device currently available
if ((mAvailableOutputDevices.types() & curProfile->getSupportedDevicesType()) == 0) {
continue;
}
// if several profiles are compatible, give priority to one with offload capability
if (profile != 0 && ((curProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0)) {
continue;
}
profile = curProfile;
if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
break;
}
}
}
return profile;
}
audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
routing_strategy strategy = getStrategy(stream);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
device, stream, samplingRate, format, channelMask, flags);
return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE,
stream, samplingRate,format, channelMask,
flags, offloadInfo);
}
status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
audio_port_handle_t selectedDeviceId,
const audio_offload_info_t *offloadInfo)
{
audio_attributes_t attributes;
if (attr != NULL) {
if (!isValidAttributes(attr)) {
ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
attr->usage, attr->content_type, attr->flags,
attr->tags);
return BAD_VALUE;
}
attributes = *attr;
} else {
if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) {
ALOGE("getOutputForAttr(): invalid stream type");
return BAD_VALUE;
}
stream_type_to_audio_attributes(*stream, &attributes);
}
sp<SwAudioOutputDescriptor> desc;
if (mPolicyMixes.getOutputForAttr(attributes, uid, desc) == NO_ERROR) {
ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr");
if (!audio_has_proportional_frames(format)) {
return BAD_VALUE;
}
*stream = streamTypefromAttributesInt(&attributes);
*output = desc->mIoHandle;
ALOGV("getOutputForAttr() returns output %d", *output);
return NO_ERROR;
}
if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
return BAD_VALUE;
}
ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x"
" session %d selectedDeviceId %d",
attributes.usage, attributes.content_type, attributes.tags, attributes.flags,
session, selectedDeviceId);
*stream = streamTypefromAttributesInt(&attributes);
// Explicit routing?
sp<DeviceDescriptor> deviceDesc;
for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
if (mAvailableOutputDevices[i]->getId() == selectedDeviceId) {
deviceDesc = mAvailableOutputDevices[i];
break;
}
}
mOutputRoutes.addRoute(session, *stream, SessionRoute::SOURCE_TYPE_NA, deviceDesc, uid);
routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
}
ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x",
device, samplingRate, format, channelMask, flags);
*output = getOutputForDevice(device, session, *stream,
samplingRate, format, channelMask,
flags, offloadInfo);
if (*output == AUDIO_IO_HANDLE_NONE) {
mOutputRoutes.removeRoute(session);
return INVALID_OPERATION;
}
return NO_ERROR;
}
audio_io_handle_t AudioPolicyManager::getOutputForDevice(
audio_devices_t device,
audio_session_t session __unused,
audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
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
// 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", false /* default_value */)) {
// use DEEP_BUFFER as default output for music stream type
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 or MasterMono is enabled.
// This prevents 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;
}
audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
audio_output_flags_t flags,
audio_format_t format)
{
// select one output among several that provide a path to a particular device or set of
// devices (the list was previously build by getOutputsForDevice()).
// The priority is as follows:
// 1: the output with the highest number of requested policy flags
// 2: the output with the bit depth the closest to the requested one
// 3: the primary output
// 4: 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 outputForFlags = 0;
audio_io_handle_t outputForPrimary = 0;
audio_io_handle_t outputForFormat = 0;
audio_format_t bestFormat = AUDIO_FORMAT_INVALID;
audio_format_t bestFormatForFlags = AUDIO_FORMAT_INVALID;
for (size_t i = 0; i < outputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
if (!outputDesc->isDuplicated()) {
// if a valid format is specified, skip output if not compatible
if (format != AUDIO_FORMAT_INVALID) {
if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (!audio_formats_match(format, outputDesc->mFormat)) {
continue;
}
} else if (!audio_is_linear_pcm(format)) {
continue;
}
if (AudioPort::isBetterFormatMatch(
outputDesc->mFormat, bestFormat, format)) {
outputForFormat = outputs[i];
bestFormat = outputDesc->mFormat;
}
}
int commonFlags = popcount(outputDesc->mProfile->getFlags() & flags);
if (commonFlags >= maxCommonFlags) {
if (commonFlags == maxCommonFlags) {
if (AudioPort::isBetterFormatMatch(
outputDesc->mFormat, bestFormatForFlags, format)) {
outputForFlags = outputs[i];
bestFormatForFlags = outputDesc->mFormat;
}
} else {
outputForFlags = outputs[i];
maxCommonFlags = commonFlags;
bestFormatForFlags = outputDesc->mFormat;
}
ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
}
if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
outputForPrimary = outputs[i];
}
}
}
if (outputForFlags != 0) {
return outputForFlags;
}
if (outputForFormat != 0) {
return outputForFormat;
}
if (outputForPrimary != 0) {
return outputForPrimary;
}
return outputs[0];
}
status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
audio_session_t session)
{
ALOGV("startOutput() output %d, stream %d, session %d",
output, stream, session);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
ALOGW("startOutput() unknown output %d", output);
return BAD_VALUE;
}
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
// Routing?
mOutputRoutes.incRouteActivity(session);
audio_devices_t newDevice;
AudioMix *policyMix = NULL;
const char *address = NULL;
if (outputDesc->mPolicyMix != NULL) {
policyMix = outputDesc->mPolicyMix;
address = policyMix->mDeviceAddress.string();
if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
newDevice = policyMix->mDeviceType;
} else {
newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
}
} else if (mOutputRoutes.hasRouteChanged(session)) {
newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
checkStrategyRoute(getStrategy(stream), output);
} else {
newDevice = AUDIO_DEVICE_NONE;
}
uint32_t delayMs = 0;
status_t status = startSource(outputDesc, stream, newDevice, address, &delayMs);
if (status != NO_ERROR) {
mOutputRoutes.decRouteActivity(session);
return status;
}
// Automatically enable the remote submix input when output is started on a re routing mix
// of type MIX_TYPE_RECORDERS
if (audio_is_remote_submix_device(newDevice) && policyMix != NULL &&
policyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address,
"remote-submix");
}
if (delayMs != 0) {
usleep(delayMs * 1000);
}
#ifdef DOLBY_ENABLE
// DOLBY_UDC
// It is observed that in some use-cases where multiple outputs are present eg. bluetooth and headphone,
// the output for particular stream type is decided in this routine. Hence we must call
// getDeviceForStrategy in order to get the current active output for this stream type and update
// the dolby system property.
if (stream == AUDIO_STREAM_MUSIC)
{
audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AUDIO_STREAM_MUSIC), true);
mDolbyAudioPolicy.setEndpointSystemProperty(audioOutputDevice, mHwModules);
}
// DOLBY_DAP_MOVE_EFFECT
// Note: The global effect can't be taken away from the deep-buffered output (source output) if there're still
// music playing on the deep-buffered output.
mDolbyAudioPolicy.moveOutput(stream, outputDesc, &mOutputs, mpClientInterface, output);
#endif //DOLBY_END
return status;
}
status_t AudioPolicyManager::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;
*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);
}
// force device change if the output is inactive and no audio patch is already present.
// check active before incrementing usage count
bool force = !outputDesc->isActive() &&
(outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE);
// 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
// - has a current device selection that differs from selected device.
// - supports currently selected device
// - has an active audio patch
// 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 &&
desc->supportedDevices() & device &&
desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) {
force = true;
}
// wait for audio on other active outputs to be presented when starting
// a notification so that audio focus effect can propagate, or that a mute/unmute
// event occurred for beacon
uint32_t latency = desc->latency();
if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
waitMs = latency;
}
}
}
uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force, 0, NULL, address);
// 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,
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);
}
if (waitMs > muteWaitMs) {
*delayMs = waitMs - muteWaitMs;
}
}
return NO_ERROR;
}
status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
audio_session_t session)
{
ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
ALOGW("stopOutput() unknown output %d", output);
return BAD_VALUE;
}
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
if (outputDesc->mRefCount[stream] == 1) {
// Automatically disable the remote submix input when output is stopped on a
// re routing mix of type MIX_TYPE_RECORDERS
if (audio_is_remote_submix_device(outputDesc->mDevice) &&
outputDesc->mPolicyMix != NULL &&
outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
outputDesc->mPolicyMix->mDeviceAddress,
"remote-submix");
}
}
// Routing?
bool forceDeviceUpdate = false;
if (outputDesc->mRefCount[stream] > 0) {
int activityCount = mOutputRoutes.decRouteActivity(session);
forceDeviceUpdate = (mOutputRoutes.hasRoute(session) && (activityCount == 0));
if (forceDeviceUpdate) {
checkStrategyRoute(getStrategy(stream), AUDIO_IO_HANDLE_NONE);
}
}
return stopSource(outputDesc, stream, forceDeviceUpdate);
}
status_t AudioPolicyManager::stopSource(sp<AudioOutputDescriptor> outputDesc,
audio_stream_type_t stream,
bool forceDeviceUpdate)
{
// always handle stream stop, check which stream type is stopping
handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
// handle special case for sonification while in call
if (isInCall()) {
handleIncallSonification(stream, false, false);
}
if (outputDesc->mRefCount[stream] > 0) {
// decrement usage count of this stream on the output
outputDesc->changeRefCount(stream, -1);
// store time at which the stream was stopped - see isStreamActive()
if (outputDesc->mRefCount[stream] == 0 || forceDeviceUpdate) {
outputDesc->mStopTime[stream] = systemTime();
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
uint32_t delayMs = outputDesc->latency()*2;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc &&
desc->isActive() &&
outputDesc->sharesHwModuleWith(desc) &&
(newDevice != desc->device())) {
audio_devices_t newDevice2 = getNewOutputDevice(desc, false /*fromCache*/);
bool force = desc->device() != newDevice2;
setOutputDevice(desc,
newDevice2,
force,
delayMs);
// re-apply device specific volume if not done by setOutputDevice()
if (!force) {
applyStreamVolumes(desc, newDevice2, 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;
}
}
void AudioPolicyManager::releaseOutput(audio_io_handle_t output,
audio_stream_type_t stream __unused,
audio_session_t session __unused)
{
ALOGV("releaseOutput() %d", output);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
ALOGW("releaseOutput() releasing unknown output %d", output);
return;
}
#ifdef AUDIO_POLICY_TEST
int testIndex = testOutputIndex(output);
if (testIndex != 0) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
if (outputDesc->isActive()) {
mpClientInterface->closeOutput(output);
removeOutput(output);
mTestOutputs[testIndex] = 0;
}
return;
}
#endif //AUDIO_POLICY_TEST
// Routing
mOutputRoutes.removeRoute(session);
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(index);
if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (desc->mDirectOpenCount <= 0) {
ALOGW("releaseOutput() invalid open count %d for output %d",
desc->mDirectOpenCount, output);
return;
}
if (--desc->mDirectOpenCount == 0) {
closeOutput(output);
// If effects where present on the output, audioflinger moved them to the primary
// output by default: move them back to the appropriate output.
audio_io_handle_t dstOutput = getOutputForEffect();
if (hasPrimaryOutput() && dstOutput != mPrimaryOutput->mIoHandle) {
#ifdef DOLBY_ENABLE
status_t status = mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
mPrimaryOutput->mIoHandle, 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__, mPrimaryOutput->mIoHandle, dstOutput);
}
#else // DOLBY_END
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
mPrimaryOutput->mIoHandle, dstOutput);
#endif // LINE_ADDED_BY_DOLBY
}
mpClientInterface->onAudioPortListUpdate();
}
}
}
status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_input_flags_t flags,
audio_port_handle_t selectedDeviceId,
input_type_t *inputType)
{
ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x,"
"session %d, flags %#x",
attr->source, samplingRate, format, channelMask, session, flags);
*input = AUDIO_IO_HANDLE_NONE;
*inputType = API_INPUT_INVALID;
audio_devices_t device;
// handle legacy remote submix case where the address was not always specified
String8 address = String8("");
audio_source_t inputSource = attr->source;
audio_source_t halInputSource;
AudioMix *policyMix = NULL;
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
halInputSource = inputSource;
// Explicit routing?
sp<DeviceDescriptor> deviceDesc;
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
if (mAvailableInputDevices[i]->getId() == selectedDeviceId) {
deviceDesc = mAvailableInputDevices[i];
break;
}
}
mInputRoutes.addRoute(session, SessionRoute::STREAM_TYPE_NA, inputSource, deviceDesc, uid);
if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
status_t ret = mPolicyMixes.getInputMixForAttr(*attr, &policyMix);
if (ret != NO_ERROR) {
return ret;
}
*inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
address = String8(attr->tags + strlen("addr="));
} else {
device = getDeviceAndMixForInputSource(inputSource, &policyMix);
if (device == AUDIO_DEVICE_NONE) {
ALOGW("getInputForAttr() could not find device for source %d", inputSource);
return BAD_VALUE;
}
if (policyMix != NULL) {
address = policyMix->mDeviceAddress;
if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
// there is an external policy, but this input is attached to a mix of recorders,
// meaning it receives audio injected into the framework, so the recorder doesn't
// know about it and is therefore considered "legacy"
*inputType = API_INPUT_LEGACY;
} else {
// recording a mix of players defined by an external policy, we're rerouting for
// an external policy
*inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
}
} else if (audio_is_remote_submix_device(device)) {
address = String8("0");
*inputType = API_INPUT_MIX_CAPTURE;
} else if (device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
*inputType = API_INPUT_TELEPHONY_RX;
} else {
*inputType = API_INPUT_LEGACY;
}
}
*input = getInputForDevice(device, address, session, uid, inputSource,
samplingRate, format, channelMask, flags,
policyMix);
if (*input == AUDIO_IO_HANDLE_NONE) {
mInputRoutes.removeRoute(session);
return INVALID_OPERATION;
}
ALOGV("getInputForAttr() returns input type = %d", *inputType);
return NO_ERROR;
}
audio_io_handle_t AudioPolicyManager::getInputForDevice(audio_devices_t device,
String8 address,
audio_session_t session,
uid_t uid,
audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_input_flags_t flags,
AudioMix *policyMix)
{
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
audio_source_t halInputSource = inputSource;
bool isSoundTrigger = false;
if (inputSource == AUDIO_SOURCE_HOTWORD) {
ssize_t index = mSoundTriggerSessions.indexOfKey(session);
if (index >= 0) {
input = mSoundTriggerSessions.valueFor(session);
isSoundTrigger = true;
flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
ALOGV("SoundTrigger capture on session %d input %d", session, input);
} else {
halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
}
}
// find a compatible input profile (not necessarily identical in parameters)
sp<IOProfile> profile;
// samplingRate and flags may be updated by getInputProfile
uint32_t profileSamplingRate = (samplingRate == 0) ? SAMPLE_RATE_HZ_DEFAULT : samplingRate;
audio_format_t profileFormat = format;
audio_channel_mask_t profileChannelMask = channelMask;
audio_input_flags_t profileFlags = flags;
for (;;) {
profile = getInputProfile(device, address,
profileSamplingRate, profileFormat, profileChannelMask,
profileFlags);
if (profile != 0) {
break; // success
} else if (profileFlags & AUDIO_INPUT_FLAG_RAW) {
profileFlags = (audio_input_flags_t) (profileFlags & ~AUDIO_INPUT_FLAG_RAW); // retry
} else if (profileFlags != AUDIO_INPUT_FLAG_NONE) {
profileFlags = AUDIO_INPUT_FLAG_NONE; // retry
} else { // fail
ALOGW("getInputForDevice() could not find profile for device 0x%X,"
"samplingRate %u, format %#x, channelMask 0x%X, flags %#x",
device, samplingRate, format, channelMask, flags);
return input;
}
}
// Pick input sampling rate if not specified by client
if (samplingRate == 0) {
samplingRate = profileSamplingRate;
}
if (profile->getModuleHandle() == 0) {
ALOGE("getInputForAttr(): HW module %s not opened", profile->getModuleName());
return input;
}
sp<AudioSession> audioSession = new AudioSession(session,
inputSource,
format,
samplingRate,
channelMask,
flags,
uid,
isSoundTrigger,
policyMix, mpClientInterface);
// TODO enable input reuse
#if 0
// reuse an open input if possible
for (size_t i = 0; i < mInputs.size(); i++) {
sp<AudioInputDescriptor> desc = mInputs.valueAt(i);
// reuse input if it shares the same profile and same sound trigger attribute
if (profile == desc->mProfile &&
isSoundTrigger == desc->isSoundTrigger()) {
sp<AudioSession> as = desc->getAudioSession(session);
if (as != 0) {
// do not allow unmatching properties on same session
if (as->matches(audioSession)) {
as->changeOpenCount(1);
} else {
ALOGW("getInputForDevice() record with different attributes"
" exists for session %d", session);
return input;
}
} else {
desc->addAudioSession(session, audioSession);
}
ALOGV("getInputForDevice() reusing input %d", mInputs.keyAt(i));
return mInputs.keyAt(i);
}
}
#endif
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = profileSamplingRate;
config.channel_mask = profileChannelMask;
config.format = profileFormat;
status_t status = mpClientInterface->openInput(profile->getModuleHandle(),
&input,
&config,
&device,
address,
halInputSource,
profileFlags);
// only accept input with the exact requested set of parameters
if (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE ||
(profileSamplingRate != config.sample_rate) ||
!audio_formats_match(profileFormat, config.format) ||
(profileChannelMask != config.channel_mask)) {
ALOGW("getInputForAttr() failed opening input: samplingRate %d"
", format %d, channelMask %x",
samplingRate, format, channelMask);
if (input != AUDIO_IO_HANDLE_NONE) {
mpClientInterface->closeInput(input);
}
return AUDIO_IO_HANDLE_NONE;
}
sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
inputDesc->mSamplingRate = profileSamplingRate;
inputDesc->mFormat = profileFormat;
inputDesc->mChannelMask = profileChannelMask;
inputDesc->mDevice = device;
inputDesc->mPolicyMix = policyMix;
inputDesc->addAudioSession(session, audioSession);
addInput(input, inputDesc);
mpClientInterface->onAudioPortListUpdate();
return input;
}
status_t AudioPolicyManager::startInput(audio_io_handle_t input,
audio_session_t session)
{
ALOGV("startInput() input %d", input);
ssize_t index = mInputs.indexOfKey(input);
if (index < 0) {
ALOGW("startInput() unknown input %d", input);
return BAD_VALUE;
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
if (audioSession == 0) {
ALOGW("startInput() unknown session %d on input %d", session, input);
return BAD_VALUE;
}
// virtual input devices are compatible with other input devices
if (!is_virtual_input_device(inputDesc->mDevice)) {
// for a non-virtual input device, check if there is another (non-virtual) active input
audio_io_handle_t activeInput = mInputs.getActiveInput();
if (activeInput != 0 && activeInput != input) {
// If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
// otherwise the active input continues and the new input cannot be started.
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
if ((activeDesc->inputSource() == AUDIO_SOURCE_HOTWORD) &&
!activeDesc->hasPreemptedSession(session)) {
ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
//FIXME: consider all active sessions
AudioSessionCollection activeSessions = activeDesc->getActiveAudioSessions();
audio_session_t activeSession = activeSessions.keyAt(0);
SortedVector<audio_session_t> sessions =
activeDesc->getPreemptedSessions();
sessions.add(activeSession);
inputDesc->setPreemptedSessions(sessions);
stopInput(activeInput, activeSession);
releaseInput(activeInput, activeSession);
} else {
ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
return INVALID_OPERATION;
}
}
// Do not allow capture if an active voice call is using a software patch and
// the call TX source device is on the same HW module.
// FIXME: would be better to refine to only inputs whose profile connects to the
// call TX device but this information is not in the audio patch
if (mCallTxPatch != 0 &&
inputDesc->getModuleHandle() == mCallTxPatch->mPatch.sources[0].ext.device.hw_module) {
return INVALID_OPERATION;
}
}
// Routing?
mInputRoutes.incRouteActivity(session);
if (!inputDesc->isActive() || mInputRoutes.hasRouteChanged(session)) {
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
&& ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mDeviceAddress,
MIX_STATE_MIXING);
}
// indicate active capture to sound trigger service if starting capture from a mic on
// primary HW module
audio_devices_t device = getNewInputDevice(input);
audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
SoundTrigger::setCaptureState(true);
}
setInputDevice(input, device, true /* force */);
// automatically enable the remote submix output when input is started if not
// used by a policy mix of type MIX_TYPE_RECORDERS
// For remote submix (a virtual device), we open only one input per capture request.
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
String8 address = String8("");
if (inputDesc->mPolicyMix == NULL) {
address = String8("0");
} else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
address = inputDesc->mPolicyMix->mDeviceAddress;
}
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address, "remote-submix");
}
}
}
ALOGV("AudioPolicyManager::startInput() input source = %d", audioSession->inputSource());
audioSession->changeActiveCount(1);
return NO_ERROR;
}
status_t AudioPolicyManager::stopInput(audio_io_handle_t input,
audio_session_t session)
{
ALOGV("stopInput() input %d", input);
ssize_t index = mInputs.indexOfKey(input);
if (index < 0) {
ALOGW("stopInput() unknown input %d", input);
return BAD_VALUE;
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
if (index < 0) {
ALOGW("stopInput() unknown session %d on input %d", session, input);
return BAD_VALUE;
}
if (audioSession->activeCount() == 0) {
ALOGW("stopInput() input %d already stopped", input);
return INVALID_OPERATION;
}
audioSession->changeActiveCount(-1);
// Routing?
mInputRoutes.decRouteActivity(session);
if (!inputDesc->isActive()) {
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
&& ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mDeviceAddress,
MIX_STATE_IDLE);
}
// automatically disable the remote submix output when input is stopped if not
// used by a policy mix of type MIX_TYPE_RECORDERS
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
String8 address = String8("");
if (inputDesc->mPolicyMix == NULL) {
address = String8("0");
} else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
address = inputDesc->mPolicyMix->mDeviceAddress;
}
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address, "remote-submix");
}
}
audio_devices_t device = inputDesc->mDevice;
resetInputDevice(input);
// indicate inactive capture to sound trigger service if stopping capture from a mic on
// primary HW module
audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
SoundTrigger::setCaptureState(false);
}
inputDesc->clearPreemptedSessions();
}
return NO_ERROR;
}
void AudioPolicyManager::releaseInput(audio_io_handle_t input,
audio_session_t session)
{
ALOGV("releaseInput() %d", input);
ssize_t index = mInputs.indexOfKey(input);
if (index < 0) {
ALOGW("releaseInput() releasing unknown input %d", input);
return;
}
// Routing
mInputRoutes.removeRoute(session);
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
ALOG_ASSERT(inputDesc != 0);
sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
if (index < 0) {
ALOGW("releaseInput() unknown session %d on input %d", session, input);
return;
}
if (audioSession->openCount() == 0) {
ALOGW("releaseInput() invalid open count %d on session %d",
audioSession->openCount(), session);
return;
}
if (audioSession->changeOpenCount(-1) == 0) {
inputDesc->removeAudioSession(session);
}
if (inputDesc->getOpenRefCount() > 0) {
ALOGV("releaseInput() exit > 0");
return;
}
closeInput(input);
mpClientInterface->onAudioPortListUpdate();
ALOGV("releaseInput() exit");
}
void AudioPolicyManager::closeAllInputs() {
bool patchRemoved = false;
for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index);
ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (patch_index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
(void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
mAudioPatches.removeItemsAt(patch_index);
patchRemoved = true;
}
mpClientInterface->closeInput(mInputs.keyAt(input_index));
}
mInputs.clear();
SoundTrigger::setCaptureState(false);
nextAudioPortGeneration();
if (patchRemoved) {
mpClientInterface->onAudioPatchListUpdate();
}
}
void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax)
{
ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
mVolumeCurves->initStreamVolume(stream, indexMin, indexMax);
// initialize other private stream volumes which follow this one
for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) {
continue;
}
mVolumeCurves->initStreamVolume((audio_stream_type_t)curStream, indexMin, indexMax);
}
}
status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
int index,
audio_devices_t device)
{
if ((index < mVolumeCurves->getVolumeIndexMin(stream)) ||
(index > mVolumeCurves->getVolumeIndexMax(stream))) {
return BAD_VALUE;
}
if (!audio_is_output_device(device)) {
return BAD_VALUE;
}
// Force max volume if stream cannot be muted
if (!mVolumeCurves->canBeMuted(stream)) index = mVolumeCurves->getVolumeIndexMax(stream);
ALOGV("setStreamVolumeIndex() stream %d, device %08x, index %d",
stream, device, index);
// update other private stream volumes which follow this one
for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) {
continue;
}
mVolumeCurves->addCurrentVolumeIndex((audio_stream_type_t)curStream, device, index);
}
// update volume on all outputs and streams matching the following:
// - The requested stream (or a stream matching for volume control) is active on the output
// - The device (or devices) selected by the strategy corresponding to this stream includes
// the requested device
// - For non default requested device, currently selected device on the output is either the
// requested device or one of the devices selected by the strategy
// - For default requested device (AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME), apply volume only if
// no specific device volume value exists for currently selected device.
status_t status = NO_ERROR;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
audio_devices_t curDevice = Volume::getDeviceForVolume(desc->device());
for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) {
continue;
}
if (!(desc->isStreamActive((audio_stream_type_t)curStream) ||
(isInCall() && (curStream == AUDIO_STREAM_VOICE_CALL)))) {
continue;
}
routing_strategy curStrategy = getStrategy((audio_stream_type_t)curStream);
audio_devices_t curStreamDevice = getDeviceForStrategy(curStrategy, false /*fromCache*/);
if ((device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) &&
((curStreamDevice & device) == 0)) {
continue;
}
bool applyVolume;
if (device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
curStreamDevice |= device;
applyVolume = (curDevice & curStreamDevice) != 0;
} else {
applyVolume = !mVolumeCurves->hasVolumeIndexForDevice(
stream, Volume::getDeviceForVolume(curStreamDevice));
}
if (applyVolume) {
//FIXME: workaround for truncated touch sounds
// delayed volume change for system stream to be removed when the problem is
// handled by system UI
status_t volStatus =
checkAndSetVolume((audio_stream_type_t)curStream, index, desc, curDevice,
(stream == AUDIO_STREAM_SYSTEM) ? TOUCH_SOUND_FIXED_DELAY_MS : 0);
if (volStatus != NO_ERROR) {
status = volStatus;
}
}
}
}
return status;
}
status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
int *index,
audio_devices_t device)
{
if (index == NULL) {
return BAD_VALUE;
}
if (!audio_is_output_device(device)) {
return BAD_VALUE;
}
// if device is AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME, return volume for device corresponding to
// the strategy the stream belongs to.
if (device == AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
}
device = Volume::getDeviceForVolume(device);
*index = mVolumeCurves->getVolumeIndex(stream, device);
ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
return NO_ERROR;
}
audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
const SortedVector<audio_io_handle_t>& outputs)
{
// select one output among several suitable for global effects.
// The priority is as follows:
// 1: An offloaded output. If the effect ends up not being offloadable,
// AudioFlinger will invalidate the track and the offloaded output
// will be closed causing the effect to be moved to a PCM output.
// 2: A deep buffer output
// 3: the first output in the list
if (outputs.size() == 0) {
return 0;
}
audio_io_handle_t outputOffloaded = 0;
audio_io_handle_t outputDeepBuffer = 0;
audio_io_handle_t outputDirectPcm = 0;
for (size_t i = 0; i < outputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
outputOffloaded = outputs[i];
}
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) != 0) {
outputDirectPcm = 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 (outputDirectPcm != 0) {
return outputDirectPcm;
}
if (outputDeepBuffer != 0) {
return outputDeepBuffer;
}
return outputs[0];
}
audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
{
// apply simple rule where global effects are attached to the same output as MUSIC streams
routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
audio_io_handle_t output = selectOutputForEffects(dstOutputs);
ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
return output;
}
status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
audio_io_handle_t io,
uint32_t strategy,
int session,
int id)
{
ssize_t index = mOutputs.indexOfKey(io);
if (index < 0) {
index = mInputs.indexOfKey(io);
if (index < 0) {
ALOGW("registerEffect() unknown io %d", io);
return INVALID_OPERATION;
}
}
#ifdef DOLBY_ENABLE
return mDolbyAudioPolicy.registerEffect(desc, io, strategy, session, id, &mEffects);
#else // DOLBY_END
return mEffects.registerEffect(desc, io, strategy, session, id);
#endif // LINE_ADDED_BY_DOLBY
}
bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
{
bool active = false;
for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT && !active; curStream++) {
if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) {
continue;
}
active = mOutputs.isStreamActive((audio_stream_type_t)curStream, inPastMs);
}
return active;
}
bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const
{
return mOutputs.isStreamActiveRemotely(stream, inPastMs);
}
bool AudioPolicyManager::isSourceActive(audio_source_t source) const
{
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
if (inputDescriptor->isSourceActive(source)) {
return true;
}
}
return false;
}
// Register a list of custom mixes with their attributes and format.
// When a mix is registered, corresponding input and output profiles are
// added to the remote submix hw module. The profile contains only the
// parameters (sampling rate, format...) specified by the mix.
// The corresponding input remote submix device is also connected.
//
// When a remote submix device is connected, the address is checked to select the
// appropriate profile and the corresponding input or output stream is opened.
//
// When capture starts, getInputForAttr() will:
// - 1 look for a mix matching the address passed in attribtutes tags if any
// - 2 if none found, getDeviceForInputSource() will:
// - 2.1 look for a mix matching the attributes source
// - 2.2 if none found, default to device selection by policy rules
// At this time, the corresponding output remote submix device is also connected
// and active playback use cases can be transferred to this mix if needed when reconnecting
// after AudioTracks are invalidated
//
// When playback starts, getOutputForAttr() will:
// - 1 look for a mix matching the address passed in attribtutes tags if any
// - 2 if none found, look for a mix matching the attributes usage
// - 3 if none found, default to device and output selection by policy rules.
status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes)
{
ALOGV("registerPolicyMixes() %zu mix(es)", mixes.size());
status_t res = NO_ERROR;
sp<HwModule> rSubmixModule;
// examine each mix's route type
for (size_t i = 0; i < mixes.size(); i++) {
// we only support MIX_ROUTE_FLAG_LOOP_BACK or MIX_ROUTE_FLAG_RENDER, not the combination
if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_ALL) == MIX_ROUTE_FLAG_ALL) {
res = INVALID_OPERATION;
break;
}
if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
// Loop back through "remote submix"
if (rSubmixModule == 0) {
for (size_t j = 0; i < mHwModules.size(); j++) {
if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[j]->mName) == 0
&& mHwModules[j]->mHandle != 0) {
rSubmixModule = mHwModules[j];
break;
}
}
}
ALOGV("registerPolicyMixes() mix %zu of %zu is LOOP_BACK", i, mixes.size());
if (rSubmixModule == 0) {
ALOGE(" Unable to find audio module for submix, aborting mix %zu registration", i);
res = INVALID_OPERATION;
break;
}
String8 address = mixes[i].mDeviceAddress;
if (mPolicyMixes.registerMix(address, mixes[i], 0 /*output desc*/) != NO_ERROR) {
ALOGE(" Error registering mix %zu for address %s", i, address.string());
res = INVALID_OPERATION;
break;
}
audio_config_t outputConfig = mixes[i].mFormat;
audio_config_t inputConfig = mixes[i].mFormat;
// NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
// stereo and let audio flinger do the channel conversion if needed.
outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
rSubmixModule->addOutputProfile(address, &outputConfig,
AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
rSubmixModule->addInputProfile(address, &inputConfig,
AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.string(), "remote-submix");
} else {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.string(), "remote-submix");
}
} else if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
String8 address = mixes[i].mDeviceAddress;
audio_devices_t device = mixes[i].mDeviceType;
ALOGV(" registerPolicyMixes() mix %zu of %zu is RENDER, dev=0x%X addr=%s",
i, mixes.size(), device, address.string());
bool foundOutput = false;
for (size_t j = 0 ; j < mOutputs.size() ; j++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(j);
sp<AudioPatch> patch = mAudioPatches.valueFor(desc->getPatchHandle());
if ((patch != 0) && (patch->mPatch.num_sinks != 0)
&& (patch->mPatch.sinks[0].type == AUDIO_PORT_TYPE_DEVICE)
&& (patch->mPatch.sinks[0].ext.device.type == device)
&& (strncmp(patch->mPatch.sinks[0].ext.device.address, address.string(),
AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
if (mPolicyMixes.registerMix(address, mixes[i], desc) != NO_ERROR) {
res = INVALID_OPERATION;
} else {
foundOutput = true;
}
break;
}
}
if (res != NO_ERROR) {
ALOGE(" Error registering mix %zu for device 0x%X addr %s",
i, device, address.string());
res = INVALID_OPERATION;
break;
} else if (!foundOutput) {
ALOGE(" Output not found for mix %zu for device 0x%X addr %s",
i, device, address.string());
res = INVALID_OPERATION;
break;
}
}
}
if (res != NO_ERROR) {
unregisterPolicyMixes(mixes);
}
return res;
}
status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes)
{
ALOGV("unregisterPolicyMixes() num mixes %zu", mixes.size());
status_t res = NO_ERROR;
sp<HwModule> rSubmixModule;
// examine each mix's route type
for (size_t i = 0; i < mixes.size(); i++) {
if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
if (rSubmixModule == 0) {
for (size_t j = 0; i < mHwModules.size(); j++) {
if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[j]->mName) == 0
&& mHwModules[j]->mHandle != 0) {
rSubmixModule = mHwModules[j];
break;
}
}
}
if (rSubmixModule == 0) {
res = INVALID_OPERATION;
continue;
}
String8 address = mixes[i].mDeviceAddress;
if (mPolicyMixes.unregisterMix(address) != NO_ERROR) {
res = INVALID_OPERATION;
continue;
}
if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) ==
AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.string(), "remote-submix");
}
if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.string(), "remote-submix");
}
rSubmixModule->removeOutputProfile(address);
rSubmixModule->removeInputProfile(address);
} if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
if (mPolicyMixes.unregisterMix(mixes[i].mDeviceAddress) != NO_ERROR) {
res = INVALID_OPERATION;
continue;
}
}
}
return res;
}
status_t AudioPolicyManager::dump(int fd)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
result.append(buffer);
snprintf(buffer, SIZE, " Primary Output: %d\n",
hasPrimaryOutput() ? mPrimaryOutput->mIoHandle : AUDIO_IO_HANDLE_NONE);
result.append(buffer);
snprintf(buffer, SIZE, " Phone state: %d\n", mEngine->getPhoneState());
result.append(buffer);
snprintf(buffer, SIZE, " Force use for communications %d\n",
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION));
result.append(buffer);
snprintf(buffer, SIZE, " Force use for media %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA));
result.append(buffer);
snprintf(buffer, SIZE, " Force use for record %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD));
result.append(buffer);
snprintf(buffer, SIZE, " Force use for dock %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK));
result.append(buffer);
snprintf(buffer, SIZE, " Force use for system %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM));
result.append(buffer);
snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n",
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO));
result.append(buffer);
snprintf(buffer, SIZE, " Force use for encoded surround output %d\n",
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND));
result.append(buffer);
snprintf(buffer, SIZE, " TTS output %s\n", mTtsOutputAvailable ? "available" : "not available");
result.append(buffer);
snprintf(buffer, SIZE, " Master mono: %s\n", mMasterMono ? "on" : "off");
result.append(buffer);
write(fd, result.string(), result.size());
mAvailableOutputDevices.dump(fd, String8("Available output"));
mAvailableInputDevices.dump(fd, String8("Available input"));
mHwModules.dump(fd);
mOutputs.dump(fd);
mInputs.dump(fd);
mVolumeCurves->dump(fd);
mEffects.dump(fd);
mAudioPatches.dump(fd);
return NO_ERROR;
}
// This function checks for the parameters which can be offloaded.
// This can be enhanced depending on the capability of the DSP and policy
// of the system.
bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
{
ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
" BitRate=%u, duration=%" PRId64 " us, has_video=%d",
offloadInfo.sample_rate, offloadInfo.channel_mask,
offloadInfo.format,
offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
offloadInfo.has_video);
if (mMasterMono) {
return false; // no offloading if mono is set.
}
// 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
const bool allowOffloadWithVideo =
property_get_bool("audio.offload.video", false /* default_value */);
if (offloadInfo.has_video && !allowOffloadWithVideo) {
ALOGV("isOffloadSupported: has_video == true, returning false");
return false;
}
//If duration is less than minimum value defined in property, return false
if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
return false;
}
} else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
return false;
}
// Do not allow offloading if one non offloadable effect is enabled. This prevents from
// creating an offloaded track and tearing it down immediately after start when audioflinger
// detects there is an active non offloadable effect.
// FIXME: We should check the audio session here but we do not have it in this context.
// This may prevent offloading in rare situations where effects are left active by apps
// in the background.
if (mEffects.isNonOffloadableEffectEnabled()) {
return false;
}
// See if there is a profile to support this.
// AUDIO_DEVICE_NONE
sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
offloadInfo.sample_rate,
offloadInfo.format,
offloadInfo.channel_mask,
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
return (profile != 0);
}
status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role,
audio_port_type_t type,
unsigned int *num_ports,
struct audio_port *ports,
unsigned int *generation)
{
if (num_ports == NULL || (*num_ports != 0 && ports == NULL) ||
generation == NULL) {
return BAD_VALUE;
}
ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports);
if (ports == NULL) {
*num_ports = 0;
}
size_t portsWritten = 0;
size_t portsMax = *num_ports;
*num_ports = 0;
if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) {
// do not report devices with type AUDIO_DEVICE_IN_STUB or AUDIO_DEVICE_OUT_STUB
// as they are used by stub HALs by convention
if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
if (mAvailableOutputDevices[i]->type() == AUDIO_DEVICE_OUT_STUB) {
continue;
}
if (portsWritten < portsMax) {
mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]);
}
(*num_ports)++;
}
}
if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_STUB) {
continue;
}
if (portsWritten < portsMax) {
mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]);
}
(*num_ports)++;
}
}
}
if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) {
if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) {
mInputs[i]->toAudioPort(&ports[portsWritten++]);
}
*num_ports += mInputs.size();
}
if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
size_t numOutputs = 0;
for (size_t i = 0; i < mOutputs.size(); i++) {
if (!mOutputs[i]->isDuplicated()) {
numOutputs++;
if (portsWritten < portsMax) {
mOutputs[i]->toAudioPort(&ports[portsWritten++]);
}
}
}
*num_ports += numOutputs;
}
}
*generation = curAudioPortGeneration();
ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports);
return NO_ERROR;
}
status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused)
{
return NO_ERROR;
}
status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
audio_patch_handle_t *handle,
uid_t uid)
{
ALOGV("createAudioPatch()");
if (handle == NULL || patch == NULL) {
return BAD_VALUE;
}
ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
return BAD_VALUE;
}
// only one source per audio patch supported for now
if (patch->num_sources > 1) {
return INVALID_OPERATION;
}
if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) {
return INVALID_OPERATION;
}
for (size_t i = 0; i < patch->num_sinks; i++) {
if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) {
return INVALID_OPERATION;
}
}
sp<AudioPatch> patchDesc;
ssize_t index = mAudioPatches.indexOfKey(*handle);
ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
patch->sources[0].role,
patch->sources[0].type);
#if LOG_NDEBUG == 0
for (size_t i = 0; i < patch->num_sinks; i++) {
ALOGV("createAudioPatch sink %zu: id %d role %d type %d", i, patch->sinks[i].id,
patch->sinks[i].role,
patch->sinks[i].type);
}
#endif
if (index >= 0) {
patchDesc = mAudioPatches.valueAt(index);
ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
mUidCached, patchDesc->mUid, uid);
if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
return INVALID_OPERATION;
}
} else {
*handle = AUDIO_PATCH_HANDLE_NONE;
}
if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
if (outputDesc == NULL) {
ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
return BAD_VALUE;
}
ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
outputDesc->mIoHandle);
if (patchDesc != 0) {
if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
patchDesc->mPatch.sources[0].id, patch->sources[0].id);
return BAD_VALUE;
}
}
DeviceVector devices;
for (size_t i = 0; i < patch->num_sinks; i++) {
// Only support mix to devices connection
// TODO add support for mix to mix connection
if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
ALOGV("createAudioPatch() source mix but sink is not a device");
return INVALID_OPERATION;
}
sp<DeviceDescriptor> devDesc =
mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
if (devDesc == 0) {
ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id);
return BAD_VALUE;
}
if (!outputDesc->mProfile->isCompatibleProfile(devDesc->type(),
devDesc->mAddress,
patch->sources[0].sample_rate,
NULL, // updatedSamplingRate
patch->sources[0].format,
NULL, // updatedFormat
patch->sources[0].channel_mask,
NULL, // updatedChannelMask
AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
ALOGV("createAudioPatch() profile not supported for device %08x",
devDesc->type());
return INVALID_OPERATION;
}
devices.add(devDesc);
}
if (devices.size() == 0) {
return INVALID_OPERATION;
}
// TODO: reconfigure output format and channels here
ALOGV("createAudioPatch() setting device %08x on output %d",
devices.types(), outputDesc->mIoHandle);
setOutputDevice(outputDesc, devices.types(), true, 0, handle);
index = mAudioPatches.indexOfKey(*handle);
if (index >= 0) {
if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
}
patchDesc = mAudioPatches.valueAt(index);
patchDesc->mUid = uid;
ALOGV("createAudioPatch() success");
} else {
ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
return INVALID_OPERATION;
}
} else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
// input device to input mix connection
// only one sink supported when connecting an input device to a mix
if (patch->num_sinks > 1) {
return INVALID_OPERATION;
}
sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id);
if (inputDesc == NULL) {
return BAD_VALUE;
}
if (patchDesc != 0) {
if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) {
return BAD_VALUE;
}
}
sp<DeviceDescriptor> devDesc =
mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
if (devDesc == 0) {
return BAD_VALUE;
}
if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(),
devDesc->mAddress,
patch->sinks[0].sample_rate,
NULL, /*updatedSampleRate*/
patch->sinks[0].format,
NULL, /*updatedFormat*/
patch->sinks[0].channel_mask,
NULL, /*updatedChannelMask*/
// FIXME for the parameter type,
// and the NONE
(audio_output_flags_t)
AUDIO_INPUT_FLAG_NONE)) {
return INVALID_OPERATION;
}
// TODO: reconfigure output format and channels here
ALOGV("createAudioPatch() setting device %08x on output %d",
devDesc->type(), inputDesc->mIoHandle);
setInputDevice(inputDesc->mIoHandle, devDesc->type(), true, handle);
index = mAudioPatches.indexOfKey(*handle);
if (index >= 0) {
if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
}
patchDesc = mAudioPatches.valueAt(index);
patchDesc->mUid = uid;
ALOGV("createAudioPatch() success");
} else {
ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
return INVALID_OPERATION;
}
} else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
// device to device connection
if (patchDesc != 0) {
if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
return BAD_VALUE;
}
}
sp<DeviceDescriptor> srcDeviceDesc =
mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
if (srcDeviceDesc == 0) {
return BAD_VALUE;
}
//update source and sink with our own data as the data passed in the patch may
// be incomplete.
struct audio_patch newPatch = *patch;
srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
for (size_t i = 0; i < patch->num_sinks; i++) {
if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
ALOGV("createAudioPatch() source device but one sink is not a device");
return INVALID_OPERATION;
}
sp<DeviceDescriptor> sinkDeviceDesc =
mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
if (sinkDeviceDesc == 0) {
return BAD_VALUE;
}
sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
// create a software bridge in PatchPanel if:
// - source and sink devices are on differnt HW modules OR
// - audio HAL version is < 3.0
if ((srcDeviceDesc->getModuleHandle() != sinkDeviceDesc->getModuleHandle()) ||
(srcDeviceDesc->mModule->getHalVersion() < AUDIO_DEVICE_API_VERSION_3_0)) {
// support only one sink device for now to simplify output selection logic
if (patch->num_sinks > 1) {
return INVALID_OPERATION;
}
SortedVector<audio_io_handle_t> outputs =
getOutputsForDevice(sinkDeviceDesc->type(), mOutputs);
// if the sink device is reachable via an opened output stream, request to go via
// this output stream by adding a second source to the patch description
audio_io_handle_t output = selectOutput(outputs,
AUDIO_OUTPUT_FLAG_NONE,
AUDIO_FORMAT_INVALID);
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc->isDuplicated()) {
return INVALID_OPERATION;
}
outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]);
newPatch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH;
newPatch.num_sources = 2;
}
}
}
// TODO: check from routing capabilities in config file and other conflicting patches
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (index >= 0) {
afPatchHandle = patchDesc->mAfPatchHandle;
}
status_t status = mpClientInterface->createAudioPatch(&newPatch,
&afPatchHandle,
0);
ALOGV("createAudioPatch() patch panel returned %d patchHandle %d",
status, afPatchHandle);
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch(&newPatch, uid);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = newPatch;
}
patchDesc->mAfPatchHandle = afPatchHandle;
*handle = patchDesc->mHandle;
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
} else {
ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
status);
return INVALID_OPERATION;
}
} else {
return BAD_VALUE;
}
} else {
return BAD_VALUE;
}
return NO_ERROR;
}
status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle,
uid_t uid)
{
ALOGV("releaseAudioPatch() patch %d", handle);
ssize_t index = mAudioPatches.indexOfKey(handle);
if (index < 0) {
return BAD_VALUE;
}
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
mUidCached, patchDesc->mUid, uid);
if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
return INVALID_OPERATION;
}
struct audio_patch *patch = &patchDesc->mPatch;
patchDesc->mUid = mUidCached;
if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
if (outputDesc == NULL) {
ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
return BAD_VALUE;
}
setOutputDevice(outputDesc,
getNewOutputDevice(outputDesc, true /*fromCache*/),
true,
0,
NULL);
} else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id);
if (inputDesc == NULL) {
ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
return BAD_VALUE;
}
setInputDevice(inputDesc->mIoHandle,
getNewInputDevice(inputDesc->mIoHandle),
true,
NULL);
} else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d",
status, patchDesc->mAfPatchHandle);
removeAudioPatch(patchDesc->mHandle);
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
} else {
return BAD_VALUE;
}
} else {
return BAD_VALUE;
}
return NO_ERROR;
}
status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches,
struct audio_patch *patches,
unsigned int *generation)
{
if (generation == NULL) {
return BAD_VALUE;
}
*generation = curAudioPortGeneration();
return mAudioPatches.listAudioPatches(num_patches, patches);
}
status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config)
{
ALOGV("setAudioPortConfig()");
if (config == NULL) {
return BAD_VALUE;
}
ALOGV("setAudioPortConfig() on port handle %d", config->id);
// Only support gain configuration for now
if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) {
return INVALID_OPERATION;
}
sp<AudioPortConfig> audioPortConfig;
if (config->type == AUDIO_PORT_TYPE_MIX) {
if (config->role == AUDIO_PORT_ROLE_SOURCE) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id);
if (outputDesc == NULL) {
return BAD_VALUE;
}
ALOG_ASSERT(!outputDesc->isDuplicated(),
"setAudioPortConfig() called on duplicated output %d",
outputDesc->mIoHandle);
audioPortConfig = outputDesc;
} else if (config->role == AUDIO_PORT_ROLE_SINK) {
sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(config->id);
if (inputDesc == NULL) {
return BAD_VALUE;
}
audioPortConfig = inputDesc;
} else {
return BAD_VALUE;
}
} else if (config->type == AUDIO_PORT_TYPE_DEVICE) {
sp<DeviceDescriptor> deviceDesc;
if (config->role == AUDIO_PORT_ROLE_SOURCE) {
deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id);
} else if (config->role == AUDIO_PORT_ROLE_SINK) {
deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id);
} else {
return BAD_VALUE;
}
if (deviceDesc == NULL) {
return BAD_VALUE;
}
audioPortConfig = deviceDesc;
} else {
return BAD_VALUE;
}
struct audio_port_config backupConfig;
status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig);
if (status == NO_ERROR) {
struct audio_port_config newConfig;
audioPortConfig->toAudioPortConfig(&newConfig, config);
status = mpClientInterface->setAudioPortConfig(&newConfig, 0);
}
if (status != NO_ERROR) {
audioPortConfig->applyAudioPortConfig(&backupConfig);
}
return status;
}
void AudioPolicyManager::releaseResourcesForUid(uid_t uid)
{
clearAudioSources(uid);
clearAudioPatches(uid);
clearSessionRoutes(uid);
}
void AudioPolicyManager::clearAudioPatches(uid_t uid)
{
for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
if (patchDesc->mUid == uid) {
releaseAudioPatch(mAudioPatches.keyAt(i), uid);
}
}
}
void AudioPolicyManager::checkStrategyRoute(routing_strategy strategy,
audio_io_handle_t ouptutToSkip)
{
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
for (size_t j = 0; j < mOutputs.size(); j++) {
if (mOutputs.keyAt(j) == ouptutToSkip) {
continue;
}
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(j);
if (!isStrategyActive(outputDesc, (routing_strategy)strategy)) {
continue;
}
// If the default device for this strategy is on another output mix,
// invalidate all tracks in this strategy to force re connection.
// Otherwise select new device on the output mix.
if (outputs.indexOf(mOutputs.keyAt(j)) < 0) {
for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
if (getStrategy((audio_stream_type_t)stream) == strategy) {
mpClientInterface->invalidateStream((audio_stream_type_t)stream);
}
}
} else {
audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
setOutputDevice(outputDesc, newDevice, false);
}
}
}
void AudioPolicyManager::clearSessionRoutes(uid_t uid)
{
// remove output routes associated with this uid
SortedVector<routing_strategy> affectedStrategies;
for (ssize_t i = (ssize_t)mOutputRoutes.size() - 1; i >= 0; i--) {
sp<SessionRoute> route = mOutputRoutes.valueAt(i);
if (route->mUid == uid) {
mOutputRoutes.removeItemsAt(i);
if (route->mDeviceDescriptor != 0) {
affectedStrategies.add(getStrategy(route->mStreamType));
}
}
}
// reroute outputs if necessary
for (size_t i = 0; i < affectedStrategies.size(); i++) {
checkStrategyRoute(affectedStrategies[i], AUDIO_IO_HANDLE_NONE);
}
// remove input routes associated with this uid
SortedVector<audio_source_t> affectedSources;
for (ssize_t i = (ssize_t)mInputRoutes.size() - 1; i >= 0; i--) {
sp<SessionRoute> route = mInputRoutes.valueAt(i);
if (route->mUid == uid) {
mInputRoutes.removeItemsAt(i);
if (route->mDeviceDescriptor != 0) {
affectedSources.add(route->mSource);
}
}
}
// reroute inputs if necessary
SortedVector<audio_io_handle_t> inputsToClose;
for (size_t i = 0; i < mInputs.size(); i++) {
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(i);
if (affectedSources.indexOf(inputDesc->inputSource()) >= 0) {
inputsToClose.add(inputDesc->mIoHandle);
}
}
for (size_t i = 0; i < inputsToClose.size(); i++) {
closeInput(inputsToClose[i]);
}
}
void AudioPolicyManager::clearAudioSources(uid_t uid)
{
for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--) {
sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
if (sourceDesc->mUid == uid) {
stopAudioSource(mAudioSources.keyAt(i));
}
}
}
status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session,
audio_io_handle_t *ioHandle,
audio_devices_t *device)
{
*session = (audio_session_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
*ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
*device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD);
return mSoundTriggerSessions.acquireSession(*session, *ioHandle);
}
status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
audio_io_handle_t *handle,
uid_t uid)
{
ALOGV("%s source %p attributes %p handle %p", __FUNCTION__, source, attributes, handle);
if (source == NULL || attributes == NULL || handle == NULL) {
return BAD_VALUE;
}
*handle = AUDIO_IO_HANDLE_NONE;
if (source->role != AUDIO_PORT_ROLE_SOURCE ||
source->type != AUDIO_PORT_TYPE_DEVICE) {
ALOGV("%s INVALID_OPERATION source->role %d source->type %d", __FUNCTION__, source->role, source->type);
return INVALID_OPERATION;
}
sp<DeviceDescriptor> srcDeviceDesc =
mAvailableInputDevices.getDevice(source->ext.device.type,
String8(source->ext.device.address));
if (srcDeviceDesc == 0) {
ALOGV("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type);
return BAD_VALUE;
}
sp<AudioSourceDescriptor> sourceDesc =
new AudioSourceDescriptor(srcDeviceDesc, attributes, uid);
struct audio_patch dummyPatch;
sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
sourceDesc->mPatchDesc = patchDesc;
status_t status = connectAudioSource(sourceDesc);
if (status == NO_ERROR) {
mAudioSources.add(sourceDesc->getHandle(), sourceDesc);
*handle = sourceDesc->getHandle();
}
return status;
}
status_t AudioPolicyManager::connectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc)
{
ALOGV("%s handle %d", __FUNCTION__, sourceDesc->getHandle());
// make sure we only have one patch per source.
disconnectAudioSource(sourceDesc);
routing_strategy strategy = (routing_strategy) getStrategyForAttr(&sourceDesc->mAttributes);
audio_stream_type_t stream = streamTypefromAttributesInt(&sourceDesc->mAttributes);
sp<DeviceDescriptor> srcDeviceDesc = sourceDesc->mDevice;
audio_devices_t sinkDevice = getDeviceForStrategy(strategy, true);
sp<DeviceDescriptor> sinkDeviceDesc =
mAvailableOutputDevices.getDevice(sinkDevice, String8(""));
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
struct audio_patch *patch = &sourceDesc->mPatchDesc->mPatch;
if (srcDeviceDesc->getAudioPort()->mModule->getHandle() ==
sinkDeviceDesc->getAudioPort()->mModule->getHandle() &&
srcDeviceDesc->getAudioPort()->mModule->getHalVersion() >= AUDIO_DEVICE_API_VERSION_3_0 &&
srcDeviceDesc->getAudioPort()->mGains.size() > 0) {
ALOGV("%s AUDIO_DEVICE_API_VERSION_3_0", __FUNCTION__);
// create patch between src device and output device
// create Hwoutput and add to mHwOutputs
} else {
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(sinkDevice, mOutputs);
audio_io_handle_t output =
selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID);
if (output == AUDIO_IO_HANDLE_NONE) {
ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevice);
return INVALID_OPERATION;
}
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc->isDuplicated()) {
ALOGV("%s output for device %08x is duplicated", __FUNCTION__, sinkDevice);
return INVALID_OPERATION;
}
// create a special patch with no sink and two sources:
// - the second source indicates to PatchPanel through which output mix this patch should
// be connected as well as the stream type for volume control
// - the sink is defined by whatever output device is currently selected for the output
// though which this patch is routed.
patch->num_sinks = 0;
patch->num_sources = 2;
srcDeviceDesc->toAudioPortConfig(&patch->sources[0], NULL);
outputDesc->toAudioPortConfig(&patch->sources[1], NULL);
patch->sources[1].ext.mix.usecase.stream = stream;
status_t status = mpClientInterface->createAudioPatch(patch,
&afPatchHandle,
0);
ALOGV("%s patch panel returned %d patchHandle %d", __FUNCTION__,
status, afPatchHandle);
if (status != NO_ERROR) {
ALOGW("%s patch panel could not connect device patch, error %d",
__FUNCTION__, status);
return INVALID_OPERATION;
}
uint32_t delayMs = 0;
status = startSource(outputDesc, stream, sinkDevice, NULL, &delayMs);
if (status != NO_ERROR) {
mpClientInterface->releaseAudioPatch(sourceDesc->mPatchDesc->mAfPatchHandle, 0);
return status;
}
sourceDesc->mSwOutput = outputDesc;
if (delayMs != 0) {
usleep(delayMs * 1000);
}
}
sourceDesc->mPatchDesc->mAfPatchHandle = afPatchHandle;
addAudioPatch(sourceDesc->mPatchDesc->mHandle, sourceDesc->mPatchDesc);
return NO_ERROR;
}
status_t AudioPolicyManager::stopAudioSource(audio_io_handle_t handle __unused)
{
sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueFor(handle);
ALOGV("%s handle %d", __FUNCTION__, handle);
if (sourceDesc == 0) {
ALOGW("%s unknown source for handle %d", __FUNCTION__, handle);
return BAD_VALUE;
}
status_t status = disconnectAudioSource(sourceDesc);
mAudioSources.removeItem(handle);
return status;
}
status_t AudioPolicyManager::setMasterMono(bool mono)
{
if (mMasterMono == mono) {
return NO_ERROR;
}
mMasterMono = mono;
// if enabling mono we close all offloaded devices, which will invalidate the
// corresponding AudioTrack. The AudioTrack client/MediaPlayer is responsible
// for recreating the new AudioTrack as non-offloaded PCM.
//
// If disabling mono, we leave all tracks as is: we don't know which clients
// and tracks are able to be recreated as offloaded. The next "song" should
// play back offloaded.
if (mMasterMono) {
Vector<audio_io_handle_t> offloaded;
for (size_t i = 0; i < mOutputs.size(); ++i) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
offloaded.push(desc->mIoHandle);
}
}
for (size_t i = 0; i < offloaded.size(); ++i) {
closeOutput(offloaded[i]);
}
}
// update master mono for all remaining outputs
for (size_t i = 0; i < mOutputs.size(); ++i) {
updateMono(mOutputs.keyAt(i));
}
return NO_ERROR;
}
status_t AudioPolicyManager::getMasterMono(bool *mono)
{
*mono = mMasterMono;
return NO_ERROR;
}
status_t AudioPolicyManager::disconnectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc)
{
ALOGV("%s handle %d", __FUNCTION__, sourceDesc->getHandle());
sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->mPatchDesc->mHandle);
if (patchDesc == 0) {
ALOGW("%s source has no patch with handle %d", __FUNCTION__,
sourceDesc->mPatchDesc->mHandle);
return BAD_VALUE;
}
removeAudioPatch(sourceDesc->mPatchDesc->mHandle);
audio_stream_type_t stream = streamTypefromAttributesInt(&sourceDesc->mAttributes);
sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->mSwOutput.promote();
if (swOutputDesc != 0) {
stopSource(swOutputDesc, stream, false);
mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
} else {
sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->mHwOutput.promote();
if (hwOutputDesc != 0) {
// release patch between src device and output device
// close Hwoutput and remove from mHwOutputs
} else {
ALOGW("%s source has neither SW nor HW output", __FUNCTION__);
}
}
return NO_ERROR;
}
sp<AudioSourceDescriptor> AudioPolicyManager::getSourceForStrategyOnOutput(
audio_io_handle_t output, routing_strategy strategy)
{
sp<AudioSourceDescriptor> source;
for (size_t i = 0; i < mAudioSources.size(); i++) {
sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
routing_strategy sourceStrategy =
(routing_strategy) getStrategyForAttr(&sourceDesc->mAttributes);
sp<SwAudioOutputDescriptor> outputDesc = sourceDesc->mSwOutput.promote();
if (sourceStrategy == strategy && outputDesc != 0 && outputDesc->mIoHandle == output) {
source = sourceDesc;
break;
}
}
return source;
}
// ----------------------------------------------------------------------------
// AudioPolicyManager
// ----------------------------------------------------------------------------
uint32_t AudioPolicyManager::nextAudioPortGeneration()
{
return android_atomic_inc(&mAudioPortGeneration);
}
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
:
#ifdef AUDIO_POLICY_TEST
Thread(false),
#endif //AUDIO_POLICY_TEST
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
mA2dpSuspended(false),
mAudioPortGeneration(1),
mBeaconMuteRefCount(0),
mBeaconPlayingRefCount(0),
mBeaconMuted(false),
mTtsOutputAvailable(false),
mMasterMono(false)
{
mUidCached = getuid();
mpClientInterface = clientInterface;
// TODO: remove when legacy conf file is removed. true on devices that use DRC on the
// DEVICE_CATEGORY_SPEAKER path to boost soft sounds, used to adjust volume curves accordingly.
// Note: remove also speaker_drc_enabled from global configuration of XML config file.
bool speakerDrcEnabled = false;
#ifdef USE_XML_AUDIO_POLICY_CONF
mVolumeCurves = new VolumeCurvesCollection();
AudioPolicyConfig config(mHwModules, mAvailableOutputDevices, mAvailableInputDevices,
mDefaultOutputDevice, speakerDrcEnabled,
static_cast<VolumeCurvesCollection *>(mVolumeCurves));
PolicySerializer serializer;
if (serializer.deserialize(AUDIO_POLICY_XML_CONFIG_FILE, config) != NO_ERROR) {
#else
mVolumeCurves = new StreamDescriptorCollection();
AudioPolicyConfig config(mHwModules, mAvailableOutputDevices, mAvailableInputDevices,
mDefaultOutputDevice, speakerDrcEnabled);
if ((ConfigParsingUtils::loadConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE, config) != NO_ERROR) &&
(ConfigParsingUtils::loadConfig(AUDIO_POLICY_CONFIG_FILE, config) != NO_ERROR)) {
#endif
ALOGE("could not load audio policy configuration file, setting defaults");
config.setDefault();
}
// must be done after reading the policy (since conditionned by Speaker Drc Enabling)
mVolumeCurves->initializeVolumeCurves(speakerDrcEnabled);
// Once policy config has been parsed, retrieve an instance of the engine and initialize it.
audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance();
if (!engineInstance) {
ALOGE("%s: Could not get an instance of policy engine", __FUNCTION__);
return;
}
// Retrieve the Policy Manager Interface
mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>();
if (mEngine == NULL) {
ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__);
return;
}
mEngine->setObserver(this);
status_t status = mEngine->initCheck();
(void) status;
ALOG_ASSERT(status == NO_ERROR, "Policy engine not initialized(err=%d)", status);
// mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
// open all output streams needed to access attached devices
audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
for (size_t i = 0; i < mHwModules.size(); i++) {
mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->getName());
if (mHwModules[i]->mHandle == 0) {
ALOGW("could not open HW module %s", mHwModules[i]->getName());
continue;
}
// open all output streams needed to access attached devices
// except for direct output streams that are only opened when they are actually
// required by an app.
// This also validates mAvailableOutputDevices list
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
if (!outProfile->hasSupportedDevices()) {
ALOGW("Output profile contains no device on module %s", mHwModules[i]->getName());
continue;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_TTS) != 0) {
mTtsOutputAvailable = true;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
continue;
}
audio_devices_t profileType = outProfile->getSupportedDevicesType();
if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) {
profileType = mDefaultOutputDevice->type();
} else {
// chose first device present in profile's SupportedDevices also part of
// outputDeviceTypes
profileType = outProfile->getSupportedDeviceForType(outputDeviceTypes);
}
if ((profileType & outputDeviceTypes) == 0) {
continue;
}
sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
const DeviceVector &devicesForType = supportedDevices.getDevicesFromType(profileType);
String8 address = devicesForType.size() > 0 ? devicesForType.itemAt(0)->mAddress
: String8("");
outputDesc->mDevice = profileType;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = outputDesc->mSamplingRate;
config.channel_mask = outputDesc->mChannelMask;
config.format = outputDesc->mFormat;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openOutput(outProfile->getModuleHandle(),
&output,
&config,
&outputDesc->mDevice,
address,
&outputDesc->mLatency,
outputDesc->mFlags);
if (status != NO_ERROR) {
ALOGW("Cannot open output stream for device %08x on hw module %s",
outputDesc->mDevice,
mHwModules[i]->getName());
} else {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
for (size_t k = 0; k < supportedDevices.size(); k++) {
ssize_t index = mAvailableOutputDevices.indexOf(supportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
mAvailableOutputDevices[index]->attach(mHwModules[i]);
}
}
if (mPrimaryOutput == 0 &&
outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = outputDesc;
}
addOutput(output, outputDesc);
setOutputDevice(outputDesc,
outputDesc->mDevice,
true,
0,
NULL,
address.string());
}
}
// open input streams needed to access attached devices to validate
// mAvailableInputDevices list
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
{
const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
if (!inProfile->hasSupportedDevices()) {
ALOGW("Input profile contains no device on module %s", mHwModules[i]->getName());
continue;
}
// chose first device present in profile's SupportedDevices also part of
// inputDeviceTypes
audio_devices_t profileType = inProfile->getSupportedDeviceForType(inputDeviceTypes);
if ((profileType & inputDeviceTypes) == 0) {
continue;
}
sp<AudioInputDescriptor> inputDesc =
new AudioInputDescriptor(inProfile);
inputDesc->mDevice = profileType;
// find the address
DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType);
// the inputs vector must be of size 1, but we don't want to crash here
String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress
: String8("");
ALOGV(" for input device 0x%x using address %s", profileType, address.string());
ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = inputDesc->mSamplingRate;
config.channel_mask = inputDesc->mChannelMask;
config.format = inputDesc->mFormat;
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openInput(inProfile->getModuleHandle(),
&input,
&config,
&inputDesc->mDevice,
address,
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE);
if (status == NO_ERROR) {
const DeviceVector &supportedDevices = inProfile->getSupportedDevices();
for (size_t k = 0; k < supportedDevices.size(); k++) {
ssize_t index = mAvailableInputDevices.indexOf(supportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if (index >= 0) {
sp<DeviceDescriptor> devDesc = mAvailableInputDevices[index];
if (!devDesc->isAttached()) {
devDesc->attach(mHwModules[i]);
devDesc->importAudioPort(inProfile);
}
}
}
mpClientInterface->closeInput(input);
} else {
ALOGW("Cannot open input stream for device %08x on hw module %s",
inputDesc->mDevice,
mHwModules[i]->getName());
}
}
}
// make sure all attached devices have been allocated a unique ID
for (size_t i = 0; i < mAvailableOutputDevices.size();) {
if (!mAvailableOutputDevices[i]->isAttached()) {
ALOGW("Output device %08x unreachable", mAvailableOutputDevices[i]->type());
mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
continue;
}
// The device is now validated and can be appended to the available devices of the engine
mEngine->setDeviceConnectionState(mAvailableOutputDevices[i],
AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
i++;
}
for (size_t i = 0; i < mAvailableInputDevices.size();) {
if (!mAvailableInputDevices[i]->isAttached()) {
ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type());
mAvailableInputDevices.remove(mAvailableInputDevices[i]);
continue;
}
// The device is now validated and can be appended to the available devices of the engine
mEngine->setDeviceConnectionState(mAvailableInputDevices[i],
AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
i++;
}
// make sure default device is reachable
if (mDefaultOutputDevice == 0 || mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type());
}
ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
updateDevicesAndOutputs();
#ifdef AUDIO_POLICY_TEST
if (mPrimaryOutput != 0) {
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"), 0);
mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, outputCmd.toString());
mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
mTestSamplingRate = 44100;
mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
mTestLatencyMs = 0;
mCurOutput = 0;
mDirectOutput = false;
for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
mTestOutputs[i] = 0;
}
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "AudioPolicyManagerTest");
run(buffer, ANDROID_PRIORITY_AUDIO);
}
#endif //AUDIO_POLICY_TEST
}
AudioPolicyManager::~AudioPolicyManager()
{
#ifdef AUDIO_POLICY_TEST
exit();
#endif //AUDIO_POLICY_TEST
for (size_t i = 0; i < mOutputs.size(); i++) {
mpClientInterface->closeOutput(mOutputs.keyAt(i));
}
for (size_t i = 0; i < mInputs.size(); i++) {
mpClientInterface->closeInput(mInputs.keyAt(i));
}
mAvailableOutputDevices.clear();
mAvailableInputDevices.clear();
mOutputs.clear();
mInputs.clear();
mHwModules.clear();
}
status_t AudioPolicyManager::initCheck()
{
return hasPrimaryOutput() ? NO_ERROR : NO_INIT;
}
#ifdef AUDIO_POLICY_TEST
bool AudioPolicyManager::threadLoop()
{
ALOGV("entering threadLoop()");
while (!exitPending())
{
String8 command;
int valueInt;
String8 value;
Mutex::Autolock _l(mLock);
mWaitWorkCV.waitRelative(mLock, milliseconds(50));
command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
AudioParameter param = AudioParameter(command);
if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
valueInt != 0) {
ALOGV("Test command %s received", command.string());
String8 target;
if (param.get(String8("target"), target) != NO_ERROR) {
target = "Manager";
}
if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
param.remove(String8("test_cmd_policy_output"));
mCurOutput = valueInt;
}
if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_direct"));
if (value == "false") {
mDirectOutput = false;
} else if (value == "true") {
mDirectOutput = true;
}
}
if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
param.remove(String8("test_cmd_policy_input"));
mTestInput = valueInt;
}
if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_format"));
int format = AUDIO_FORMAT_INVALID;
if (value == "PCM 16 bits") {
format = AUDIO_FORMAT_PCM_16_BIT;
} else if (value == "PCM 8 bits") {
format = AUDIO_FORMAT_PCM_8_BIT;
} else if (value == "Compressed MP3") {
format = AUDIO_FORMAT_MP3;
}
if (format != AUDIO_FORMAT_INVALID) {
if (target == "Manager") {
mTestFormat = format;
} else if (mTestOutputs[mCurOutput] != 0) {
AudioParameter outputParam = AudioParameter();
outputParam.addInt(String8("format"), format);
mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
}
}
}
if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_channels"));
int channels = 0;
if (value == "Channels Stereo") {
channels = AUDIO_CHANNEL_OUT_STEREO;
} else if (value == "Channels Mono") {
channels = AUDIO_CHANNEL_OUT_MONO;
}
if (channels != 0) {
if (target == "Manager") {
mTestChannels = channels;
} else if (mTestOutputs[mCurOutput] != 0) {
AudioParameter outputParam = AudioParameter();
outputParam.addInt(String8("channels"), channels);
mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
}
}
}
if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
param.remove(String8("test_cmd_policy_sampleRate"));
if (valueInt >= 0 && valueInt <= 96000) {
int samplingRate = valueInt;
if (target == "Manager") {
mTestSamplingRate = samplingRate;
} else if (mTestOutputs[mCurOutput] != 0) {
AudioParameter outputParam = AudioParameter();
outputParam.addInt(String8("sampling_rate"), samplingRate);
mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
}
}
}
if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_reopen"));
mpClientInterface->closeOutput(mpClientInterface->closeOutput(mPrimaryOutput););
audio_module_handle_t moduleHandle = mPrimaryOutput->getModuleHandle();
removeOutput(mPrimaryOutput->mIoHandle);
sp<SwAudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL,
mpClientInterface);
outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = outputDesc->mSamplingRate;
config.channel_mask = outputDesc->mChannelMask;
config.format = outputDesc->mFormat;
audio_io_handle_t handle;
status_t status = mpClientInterface->openOutput(moduleHandle,
&handle,
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
if (status != NO_ERROR) {
ALOGE("Failed to reopen hardware output stream, "
"samplingRate: %d, format %d, channels %d",
outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
} else {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
mPrimaryOutput = outputDesc;
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"), 0);
mpClientInterface->setParameters(handle, outputCmd.toString());
addOutput(handle, outputDesc);
}
}
mpClientInterface->setParameters(0, String8("test_cmd_policy="));
}
}
return false;
}
void AudioPolicyManager::exit()
{
{
AutoMutex _l(mLock);
requestExit();
mWaitWorkCV.signal();
}
requestExitAndWait();
}
int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
{
for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
if (output == mTestOutputs[i]) return i;
}
return 0;
}
#endif //AUDIO_POLICY_TEST
// ---
void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<SwAudioOutputDescriptor> outputDesc)
{
outputDesc->setIoHandle(output);
mOutputs.add(output, outputDesc);
updateMono(output); // update mono status when adding to output list
nextAudioPortGeneration();
}
void AudioPolicyManager::removeOutput(audio_io_handle_t output)
{
mOutputs.removeItem(output);
}
void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc)
{
inputDesc->setIoHandle(input);
mInputs.add(input, inputDesc);
nextAudioPortGeneration();
}
void AudioPolicyManager::findIoHandlesByAddress(sp<SwAudioOutputDescriptor> desc /*in*/,
const audio_devices_t device /*in*/,
const String8 address /*in*/,
SortedVector<audio_io_handle_t>& outputs /*out*/) {
sp<DeviceDescriptor> devDesc =
desc->mProfile->getSupportedDeviceByAddress(device, address);
if (devDesc != 0) {
ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s",
desc->mIoHandle, address.string());
outputs.add(desc->mIoHandle);
}
}
status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
audio_policy_dev_state_t state,
SortedVector<audio_io_handle_t>& outputs,
const String8 address)
{
audio_devices_t device = devDesc->type();
sp<SwAudioOutputDescriptor> desc;
if (audio_device_is_digital(device)) {
// erase all current sample rates, formats and channel masks
devDesc->clearAudioProfiles();
}
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
// first list already open outputs that can be routed to this device
for (size_t i = 0; i < mOutputs.size(); i++) {
desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && (desc->supportedDevices() & device)) {
if (!device_distinguishes_on_address(device)) {
ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
outputs.add(mOutputs.keyAt(i));
} else {
ALOGV(" checking address match due to device 0x%x", device);
findIoHandlesByAddress(desc, device, address, outputs);
}
}
}
// then look for output profiles that can be routed to this device
SortedVector< sp<IOProfile> > profiles;
for (size_t i = 0; i < mHwModules.size(); i++)
{
if (mHwModules[i]->mHandle == 0) {
continue;
}
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
if (profile->supportDevice(device)) {
if (!device_distinguishes_on_address(device) ||
profile->supportDeviceAddress(address)) {
profiles.add(profile);
ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
}
}
}
}
ALOGV(" found %zu profiles, %zu outputs", profiles.size(), outputs.size());
if (profiles.isEmpty() && outputs.isEmpty()) {
ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
return BAD_VALUE;
}
// open outputs for matching profiles if needed. Direct outputs are also opened to
// query for dynamic parameters and will be closed later by setDeviceConnectionState()
for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
sp<IOProfile> profile = profiles[profile_index];
// nothing to do if one output is already opened for this profile
size_t j;
for (j = 0; j < outputs.size(); j++) {
desc = mOutputs.valueFor(outputs.itemAt(j));
if (!desc->isDuplicated() && desc->mProfile == profile) {
// matching profile: save the sample rates, format and channel masks supported
// by the profile in our device descriptor
if (audio_device_is_digital(device)) {
devDesc->importAudioPort(profile);
}
break;
}
}
if (j != outputs.size()) {
continue;
}
ALOGV("opening output for device %08x with params %s profile %p",
device, address.string(), profile.get());
desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
desc->mDevice = device;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = desc->mSamplingRate;
config.channel_mask = desc->mChannelMask;
config.format = desc->mFormat;
config.offload_info.sample_rate = desc->mSamplingRate;
config.offload_info.channel_mask = desc->mChannelMask;
config.offload_info.format = desc->mFormat;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openOutput(profile->getModuleHandle(),
&output,
&config,
&desc->mDevice,
address,
&desc->mLatency,
desc->mFlags);
if (status == NO_ERROR) {
desc->mSamplingRate = config.sample_rate;
desc->mChannelMask = config.channel_mask;
desc->mFormat = config.format;
// Here is where the out_set_parameters() for card & device gets called
if (!address.isEmpty()) {
char *param = audio_device_address_to_parameter(device, address);
mpClientInterface->setParameters(output, String8(param));
free(param);
}
updateAudioProfiles(device, output, profile->getAudioProfiles());
if (!profile->hasValidAudioProfile()) {
ALOGW("checkOutputsForDevice() missing param");
mpClientInterface->closeOutput(output);
output = AUDIO_IO_HANDLE_NONE;
} else if (profile->hasDynamicAudioProfile()) {
mpClientInterface->closeOutput(output);
output = AUDIO_IO_HANDLE_NONE;
profile->pickAudioProfile(config.sample_rate, config.channel_mask, config.format);
config.offload_info.sample_rate = config.sample_rate;
config.offload_info.channel_mask = config.channel_mask;
config.offload_info.format = config.format;
status = mpClientInterface->openOutput(profile->getModuleHandle(),
&output,
&config,
&desc->mDevice,
address,
&desc->mLatency,
desc->mFlags);
if (status == NO_ERROR) {
desc->mSamplingRate = config.sample_rate;
desc->mChannelMask = config.channel_mask;
desc->mFormat = config.format;
} else {
output = AUDIO_IO_HANDLE_NONE;
}
}
if (output != AUDIO_IO_HANDLE_NONE) {
addOutput(output, desc);
if (device_distinguishes_on_address(device) && address != "0") {
sp<AudioPolicyMix> policyMix;
if (mPolicyMixes.getAudioPolicyMix(address, policyMix) != NO_ERROR) {
ALOGE("checkOutputsForDevice() cannot find policy for address %s",
address.string());
}
policyMix->setOutput(desc);
desc->mPolicyMix = policyMix->getMix();
} else if (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
hasPrimaryOutput()) {
// no duplicated output for direct outputs and
// outputs used by dynamic policy mixes
audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
// set initial stream volume for device
applyStreamVolumes(desc, 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->mIoHandle);
if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) {
// add duplicated output descriptor
sp<SwAudioOutputDescriptor> dupOutputDesc =
new SwAudioOutputDescriptor(NULL, mpClientInterface);
dupOutputDesc->mOutput1 = mPrimaryOutput;
dupOutputDesc->mOutput2 = desc;
dupOutputDesc->mSamplingRate = desc->mSamplingRate;
dupOutputDesc->mFormat = desc->mFormat;
dupOutputDesc->mChannelMask = desc->mChannelMask;
dupOutputDesc->mLatency = desc->mLatency;
addOutput(duplicatedOutput, dupOutputDesc);
applyStreamVolumes(dupOutputDesc, device, 0, true);
} else {
ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
mPrimaryOutput->mIoHandle, output);
mpClientInterface->closeOutput(output);
removeOutput(output);
nextAudioPortGeneration();
output = AUDIO_IO_HANDLE_NONE;
}
}
}
} else {
output = AUDIO_IO_HANDLE_NONE;
}
if (output == AUDIO_IO_HANDLE_NONE) {
ALOGW("checkOutputsForDevice() could not open output for device %x", device);
profiles.removeAt(profile_index);
profile_index--;
} else {
outputs.add(output);
// Load digital format info only for digital devices
if (audio_device_is_digital(device)) {
devDesc->importAudioPort(profile);
}
if (device_distinguishes_on_address(device)) {
ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
device, address.string());
setOutputDevice(desc, device, true/*force*/, 0/*delay*/,
NULL/*patch handle*/, address.string());
}
ALOGV("checkOutputsForDevice(): adding output %d", output);
}
}
if (profiles.isEmpty()) {
ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
return BAD_VALUE;
}
} else { // Disconnect
// check if one opened output is not needed any more after disconnecting one device
for (size_t i = 0; i < mOutputs.size(); i++) {
desc = mOutputs.valueAt(i);
if (!desc->isDuplicated()) {
// exact match on device
if (device_distinguishes_on_address(device) &&
(desc->supportedDevices() == device)) {
findIoHandlesByAddress(desc, device, address, outputs);
} else if (!(desc->supportedDevices() & mAvailableOutputDevices.types())) {
ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
mOutputs.keyAt(i));
outputs.add(mOutputs.keyAt(i));
}
}
}
// Clear any profiles associated with the disconnected device.
for (size_t i = 0; i < mHwModules.size(); i++)
{
if (mHwModules[i]->mHandle == 0) {
continue;
}
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
if (profile->supportDevice(device)) {
ALOGV("checkOutputsForDevice(): "
"clearing direct output profile %zu on module %zu", j, i);
profile->clearAudioProfiles();
}
}
}
}
return NO_ERROR;
}
status_t AudioPolicyManager::checkInputsForDevice(const sp<DeviceDescriptor> devDesc,
audio_policy_dev_state_t state,
SortedVector<audio_io_handle_t>& inputs,
const String8 address)
{
audio_devices_t device = devDesc->type();
sp<AudioInputDescriptor> desc;
if (audio_device_is_digital(device)) {
// erase all current sample rates, formats and channel masks
devDesc->clearAudioProfiles();
}
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
// first list already open inputs that can be routed to this device
for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
if (desc->mProfile->supportDevice(device)) {
ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
inputs.add(mInputs.keyAt(input_index));
}
}
// then look for input profiles that can be routed to this device
SortedVector< sp<IOProfile> > profiles;
for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
{
if (mHwModules[module_idx]->mHandle == 0) {
continue;
}
for (size_t profile_index = 0;
profile_index < mHwModules[module_idx]->mInputProfiles.size();
profile_index++)
{
sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index];
if (profile->supportDevice(device)) {
if (!device_distinguishes_on_address(device) ||
profile->supportDeviceAddress(address)) {
profiles.add(profile);
ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
profile_index, module_idx);
}
}
}
}
if (profiles.isEmpty() && inputs.isEmpty()) {
ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
return BAD_VALUE;
}
// open inputs for matching profiles if needed. Direct inputs are also opened to
// query for dynamic parameters and will be closed later by setDeviceConnectionState()
for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
sp<IOProfile> profile = profiles[profile_index];
// nothing to do if one input is already opened for this profile
size_t input_index;
for (input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
if (desc->mProfile == profile) {
if (audio_device_is_digital(device)) {
devDesc->importAudioPort(profile);
}
break;
}
}
if (input_index != mInputs.size()) {
continue;
}
ALOGV("opening input for device 0x%X with params %s", device, address.string());
desc = new AudioInputDescriptor(profile);
desc->mDevice = device;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = desc->mSamplingRate;
config.channel_mask = desc->mChannelMask;
config.format = desc->mFormat;
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openInput(profile->getModuleHandle(),
&input,
&config,
&desc->mDevice,
address,
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE /*FIXME*/);
if (status == NO_ERROR) {
desc->mSamplingRate = config.sample_rate;
desc->mChannelMask = config.channel_mask;
desc->mFormat = config.format;
if (!address.isEmpty()) {
char *param = audio_device_address_to_parameter(device, address);
mpClientInterface->setParameters(input, String8(param));
free(param);
}
updateAudioProfiles(device, input, profile->getAudioProfiles());
if (!profile->hasValidAudioProfile()) {
ALOGW("checkInputsForDevice() direct input missing param");
mpClientInterface->closeInput(input);
input = AUDIO_IO_HANDLE_NONE;
}
if (input != 0) {
addInput(input, desc);
}
} // endif input != 0
if (input == AUDIO_IO_HANDLE_NONE) {
ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
profiles.removeAt(profile_index);
profile_index--;
} else {
inputs.add(input);
if (audio_device_is_digital(device)) {
devDesc->importAudioPort(profile);
}
ALOGV("checkInputsForDevice(): adding input %d", input);
}
} // end scan profiles
if (profiles.isEmpty()) {
ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
return BAD_VALUE;
}
} else {
// Disconnect
// check if one opened input is not needed any more after disconnecting one device
for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
if (!(desc->mProfile->supportDevice(mAvailableInputDevices.types()))) {
ALOGV("checkInputsForDevice(): disconnecting adding input %d",
mInputs.keyAt(input_index));
inputs.add(mInputs.keyAt(input_index));
}
}
// Clear any profiles associated with the disconnected device.
for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) {
if (mHwModules[module_index]->mHandle == 0) {
continue;
}
for (size_t profile_index = 0;
profile_index < mHwModules[module_index]->mInputProfiles.size();
profile_index++) {
sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
if (profile->supportDevice(device)) {
ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
profile_index, module_index);
profile->clearAudioProfiles();
}
}
}
} // end disconnect
return NO_ERROR;
}
void AudioPolicyManager::closeOutput(audio_io_handle_t output)
{
ALOGV("closeOutput(%d)", output);
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc == NULL) {
ALOGW("closeOutput() unknown output %d", output);
return;
}
mPolicyMixes.closeOutput(outputDesc);
// look for duplicated outputs connected to the output being removed.
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
if (dupOutputDesc->isDuplicated() &&
(dupOutputDesc->mOutput1 == outputDesc ||
dupOutputDesc->mOutput2 == outputDesc)) {
sp<AudioOutputDescriptor> outputDesc2;
if (dupOutputDesc->mOutput1 == outputDesc) {
outputDesc2 = dupOutputDesc->mOutput2;
} else {
outputDesc2 = dupOutputDesc->mOutput1;
}
// As all active tracks on duplicated output will be deleted,
// and as they were also referenced on the other output, the reference
// count for their stream type must be adjusted accordingly on
// the other output.
for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
int refCount = dupOutputDesc->mRefCount[j];
outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
}
audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
mpClientInterface->closeOutput(duplicatedOutput);
removeOutput(duplicatedOutput);
}
}
nextAudioPortGeneration();
ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
(void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
mAudioPatches.removeItemsAt(index);
mpClientInterface->onAudioPatchListUpdate();
}
AudioParameter param;
param.add(String8("closing"), String8("true"));
mpClientInterface->setParameters(output, param.toString());
mpClientInterface->closeOutput(output);
removeOutput(output);
mPreviousOutputs = mOutputs;
}
void AudioPolicyManager::closeInput(audio_io_handle_t input)
{
ALOGV("closeInput(%d)", input);
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
if (inputDesc == NULL) {
ALOGW("closeInput() unknown input %d", input);
return;
}
nextAudioPortGeneration();
ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
(void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
mAudioPatches.removeItemsAt(index);
mpClientInterface->onAudioPatchListUpdate();
}
mpClientInterface->closeInput(input);
mInputs.removeItem(input);
}
SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(
audio_devices_t device,
SwAudioOutputCollection openOutputs)
{
SortedVector<audio_io_handle_t> outputs;
ALOGVV("getOutputsForDevice() device %04x", device);
for (size_t i = 0; i < openOutputs.size(); i++) {
ALOGVV("output %d isDuplicated=%d device=%04x",
i, openOutputs.valueAt(i)->isDuplicated(),
openOutputs.valueAt(i)->supportedDevices());
if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
outputs.add(openOutputs.keyAt(i));
}
}
return outputs;
}
bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
SortedVector<audio_io_handle_t>& outputs2)
{
if (outputs1.size() != outputs2.size()) {
return false;
}
for (size_t i = 0; i < outputs1.size(); i++) {
if (outputs1[i] != outputs2[i]) {
return false;
}
}
return true;
}
void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
{
audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mOutputs);
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
// also take into account external policy-related changes: add all outputs which are
// associated with policies in the "before" and "after" output vectors
ALOGVV("checkOutputForStrategy(): policy related outputs");
for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
if (desc != 0 && desc->mPolicyMix != NULL) {
srcOutputs.add(desc->mIoHandle);
ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
}
}
for (size_t i = 0 ; i < mOutputs.size() ; i++) {
const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != 0 && desc->mPolicyMix != NULL) {
dstOutputs.add(desc->mIoHandle);
ALOGVV(" new outputs: adding %d", desc->mIoHandle);
}
}
if (!vectorsEqual(srcOutputs,dstOutputs)) {
ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
strategy, srcOutputs[0], dstOutputs[0]);
// mute strategy while moving tracks from one output to another
for (size_t i = 0; i < srcOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
if (isStrategyActive(desc, strategy)) {
setStrategyMute(strategy, true, desc);
setStrategyMute(strategy, false, desc, MUTE_TIME_MS, newDevice);
}
sp<AudioSourceDescriptor> source =
getSourceForStrategyOnOutput(srcOutputs[i], strategy);
if (source != 0){
connectAudioSource(source);
}
}
// Move effects associated to this strategy from previous output to new output
if (strategy == STRATEGY_MEDIA) {
audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
SortedVector<audio_io_handle_t> moved;
for (size_t i = 0; i < mEffects.size(); i++) {
sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
effectDesc->mIo != fxOutput) {
if (moved.indexOf(effectDesc->mIo) < 0) {
ALOGV("checkOutputForStrategy() moving effect %d to output %d",
mEffects.keyAt(i), fxOutput);
#ifdef DOLBY_ENABLE
status_t status = mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo,
fxOutput);
if (status != NO_ERROR) {
ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__, effectDesc->mIo, fxOutput);
continue;
}
#else // DOLBY_END
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo,
fxOutput);
#endif // LINE_ADDED_BY_DOLBY
moved.add(effectDesc->mIo);
}
effectDesc->mIo = fxOutput;
}
}
}
// Move tracks associated to this strategy from previous output to new output
for (int i = 0; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
if (getStrategy((audio_stream_type_t)i) == strategy) {
mpClientInterface->invalidateStream((audio_stream_type_t)i);
}
}
}
}
void AudioPolicyManager::checkOutputForAllStrategies()
{
if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
checkOutputForStrategy(STRATEGY_PHONE);
if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
checkOutputForStrategy(STRATEGY_SONIFICATION);
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
checkOutputForStrategy(STRATEGY_ACCESSIBILITY);
checkOutputForStrategy(STRATEGY_MEDIA);
checkOutputForStrategy(STRATEGY_DTMF);
checkOutputForStrategy(STRATEGY_REROUTING);
}
void AudioPolicyManager::checkA2dpSuspend()
{
audio_io_handle_t a2dpOutput = mOutputs.getA2dpOutput();
if (a2dpOutput == 0 || mOutputs.isA2dpOnPrimary()) {
mA2dpSuspended = false;
return;
}
bool isScoConnected =
((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET &
~AUDIO_DEVICE_BIT_IN) != 0) ||
((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0);
// suspend A2DP output if:
// (NOT already suspended) &&
// ((SCO device is connected &&
// (forced usage for communication || for record is SCO))) ||
// (phone state is ringing || in call)
//
// restore A2DP output if:
// (Already suspended) &&
// ((SCO device is NOT connected ||
// (forced usage NOT for communication && NOT for record is SCO))) &&
// (phone state is NOT ringing && NOT in call)
//
if (mA2dpSuspended) {
if ((!isScoConnected ||
((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) != AUDIO_POLICY_FORCE_BT_SCO) &&
(mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) != AUDIO_POLICY_FORCE_BT_SCO))) &&
((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) &&
(mEngine->getPhoneState() != AUDIO_MODE_RINGTONE))) {
mpClientInterface->restoreOutput(a2dpOutput);
mA2dpSuspended = false;
}
} else {
if ((isScoConnected &&
((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == AUDIO_POLICY_FORCE_BT_SCO) ||
(mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO))) ||
((mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) ||
(mEngine->getPhoneState() == AUDIO_MODE_RINGTONE))) {
mpClientInterface->suspendOutput(a2dpOutput);
mA2dpSuspended = true;
}
}
}
audio_devices_t AudioPolicyManager::getNewOutputDevice(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 accessibility is active on the output:
// use device for strategy accessibility
// 6: the strategy "respectful" sonification is active on the output:
// use device for strategy "respectful" sonification
// 7: the strategy media is active on the output:
// use device for strategy media
// 8: the strategy DTMF is active on the output:
// use device for strategy DTMF
// 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output:
// use device for strategy t-t-s
if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) &&
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
} else if (isInCall() ||
isStrategyActive(outputDesc, STRATEGY_PHONE)) {
device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) {
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)) {
device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) {
device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)) {
device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) {
device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) {
device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) {
device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
}
ALOGV("getNewOutputDevice() selected device %x", device);
return device;
}
audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
{
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
if (patchDesc->mUid != mUidCached) {
ALOGV("getNewInputDevice() device %08x forced by patch %d",
inputDesc->mDevice, inputDesc->getPatchHandle());
return inputDesc->mDevice;
}
}
audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->inputSource());
return device;
}
bool AudioPolicyManager::streamsMatchForvolume(audio_stream_type_t stream1,
audio_stream_type_t stream2) {
return ((stream1 == stream2) ||
((stream1 == AUDIO_STREAM_ACCESSIBILITY) && (stream2 == AUDIO_STREAM_MUSIC)) ||
((stream1 == AUDIO_STREAM_MUSIC) && (stream2 == AUDIO_STREAM_ACCESSIBILITY)));
}
uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
return (uint32_t)getStrategy(stream);
}
audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
// By checking the range of stream before calling getStrategy, we avoid
// getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
// and then return STRATEGY_MEDIA, but we want to return the empty set.
if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) {
return AUDIO_DEVICE_NONE;
}
audio_devices_t devices = AUDIO_DEVICE_NONE;
for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) {
continue;
}
routing_strategy curStrategy = getStrategy((audio_stream_type_t)curStream);
audio_devices_t curDevices =
getDeviceForStrategy((routing_strategy)curStrategy, false /*fromCache*/);
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(curDevices, mOutputs);
for (size_t i = 0; i < outputs.size(); i++) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
if (outputDesc->isStreamActive((audio_stream_type_t)curStream)) {
curDevices |= outputDesc->device();
}
}
devices |= curDevices;
}
/*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
and doesn't really need to.*/
if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
devices |= AUDIO_DEVICE_OUT_SPEAKER;
devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
}
return devices;
}
routing_strategy AudioPolicyManager::getStrategy(audio_stream_type_t stream) const
{
ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH");
return mEngine->getStrategyForStream(stream);
}
uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
// flags to strategy mapping
if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
}
if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
}
// usage to strategy mapping
return static_cast<uint32_t>(mEngine->getStrategyForUsage(attr->usage));
}
void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
switch(stream) {
case AUDIO_STREAM_MUSIC:
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
updateDevicesAndOutputs();
break;
default:
break;
}
}
uint32_t AudioPolicyManager::handleEventForBeacon(int event) {
// skip beacon mute management if a dedicated TTS output is available
if (mTtsOutputAvailable) {
return 0;
}
switch(event) {
case STARTING_OUTPUT:
mBeaconMuteRefCount++;
break;
case STOPPING_OUTPUT:
if (mBeaconMuteRefCount > 0) {
mBeaconMuteRefCount--;
}
break;
case STARTING_BEACON:
mBeaconPlayingRefCount++;
break;
case STOPPING_BEACON:
if (mBeaconPlayingRefCount > 0) {
mBeaconPlayingRefCount--;
}
break;
}
if (mBeaconMuteRefCount > 0) {
// any playback causes beacon to be muted
return setBeaconMute(true);
} else {
// no other playback: unmute when beacon starts playing, mute when it stops
return setBeaconMute(mBeaconPlayingRefCount == 0);
}
}
uint32_t AudioPolicyManager::setBeaconMute(bool mute) {
ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d",
mute, mBeaconMuteRefCount, mBeaconPlayingRefCount);
// keep track of muted state to avoid repeating mute/unmute operations
if (mBeaconMuted != mute) {
// mute/unmute AUDIO_STREAM_TTS on all outputs
ALOGV("\t muting %d", mute);
uint32_t maxLatency = 0;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
setStreamMute(AUDIO_STREAM_TTS, mute/*on*/,
desc,
0 /*delay*/, AUDIO_DEVICE_NONE);
const uint32_t latency = desc->latency() * 2;
if (latency > maxLatency) {
maxLatency = latency;
}
}
mBeaconMuted = mute;
return maxLatency;
}
return 0;
}
audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
bool fromCache)
{
// Routing
// see if we have an explicit route
// scan the whole RouteMap, for each entry, convert the stream type to a strategy
// (getStrategy(stream)).
// if the strategy from the stream type in the RouteMap is the same as the argument above,
// and activity count is non-zero
// the device = the device from the descriptor in the RouteMap, and exit.
for (size_t routeIndex = 0; routeIndex < mOutputRoutes.size(); routeIndex++) {
sp<SessionRoute> route = mOutputRoutes.valueAt(routeIndex);
routing_strategy routeStrategy = getStrategy(route->mStreamType);
if ((routeStrategy == strategy) && route->isActive()) {
return route->mDeviceDescriptor->type();
}
}
if (fromCache) {
ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
strategy, mDeviceForStrategy[strategy]);
return mDeviceForStrategy[strategy];
}
return mEngine->getDeviceForStrategy(strategy);
}
void AudioPolicyManager::updateDevicesAndOutputs()
{
for (int i = 0; i < NUM_STRATEGIES; i++) {
mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
}
mPreviousOutputs = mOutputs;
}
uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
audio_devices_t prevDevice,
uint32_t delayMs)
{
// mute/unmute strategies using an incompatible device combination
// if muting, wait for the audio in pcm buffer to be drained before proceeding
// if unmuting, unmute only after the specified delay
if (outputDesc->isDuplicated()) {
return 0;
}
uint32_t muteWaitMs = 0;
audio_devices_t device = outputDesc->device();
bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
for (size_t i = 0; i < NUM_STRATEGIES; i++) {
audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
curDevice = curDevice & outputDesc->supportedDevices();
bool mute = shouldMute && (curDevice & device) && (curDevice != device);
bool doMute = false;
if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
doMute = true;
outputDesc->mStrategyMutedByDevice[i] = true;
} else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
doMute = true;
outputDesc->mStrategyMutedByDevice[i] = false;
}
if (doMute) {
for (size_t j = 0; j < mOutputs.size(); j++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
// skip output if it does not share any device with current output
if ((desc->supportedDevices() & outputDesc->supportedDevices())
== AUDIO_DEVICE_NONE) {
continue;
}
ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x)",
mute ? "muting" : "unmuting", i, curDevice);
setStrategyMute((routing_strategy)i, mute, desc, mute ? 0 : delayMs);
if (isStrategyActive(desc, (routing_strategy)i)) {
if (mute) {
// FIXME: should not need to double latency if volume could be applied
// immediately by the audioflinger mixer. We must account for the delay
// between now and the next time the audioflinger thread for this output
// will process a buffer (which corresponds to one buffer size,
// usually 1/2 or 1/4 of the latency).
if (muteWaitMs < desc->latency() * 2) {
muteWaitMs = desc->latency() * 2;
}
}
}
}
}
}
// temporary mute output if device selection changes to avoid volume bursts due to
// different per device volumes
if (outputDesc->isActive() && (device != prevDevice)) {
uint32_t tempMuteWaitMs = outputDesc->latency() * 2;
// temporary mute duration is conservatively set to 4 times the reported latency
uint32_t tempMuteDurationMs = outputDesc->latency() * 4;
if (muteWaitMs < tempMuteWaitMs) {
muteWaitMs = tempMuteWaitMs;
}
for (size_t i = 0; i < NUM_STRATEGIES; i++) {
if (isStrategyActive(outputDesc, (routing_strategy)i)) {
// make sure that we do not start the temporary mute period too early in case of
// delayed device change
setStrategyMute((routing_strategy)i, true, outputDesc, delayMs);
setStrategyMute((routing_strategy)i, false, outputDesc,
delayMs + tempMuteDurationMs, device);
}
}
}
// wait for the PCM output buffers to empty before proceeding with the rest of the command
if (muteWaitMs > delayMs) {
muteWaitMs -= delayMs;
usleep(muteWaitMs * 1000);
return muteWaitMs;
}
return 0;
}
uint32_t AudioPolicyManager::setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device,
bool force,
int delayMs,
audio_patch_handle_t *patchHandle,
const char* address)
{
ALOGV("setOutputDevice() device %04x delayMs %d", device, delayMs);
AudioParameter param;
uint32_t muteWaitMs;
if (outputDesc->isDuplicated()) {
muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs);
muteWaitMs += setOutputDevice(outputDesc->subOutput2(), 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->supportedDevices()) == 0)) {
return 0;
}
// filter devices according to output selected
device = (audio_devices_t)(device & outputDesc->supportedDevices());
audio_devices_t prevDevice = outputDesc->mDevice;
ALOGV("setOutputDevice() prevDevice 0x%04x", prevDevice);
if (device != AUDIO_DEVICE_NONE) {
outputDesc->mDevice = device;
}
muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
// Do not change the routing if:
// the requested device is AUDIO_DEVICE_NONE
// OR the requested device is the same as current device
// AND force is not specified
// AND the output is connected by a valid audio patch.
// Doing this check here allows the caller to call setOutputDevice() without conditions
if ((device == AUDIO_DEVICE_NONE || device == prevDevice) &&
!force &&
outputDesc->getPatchHandle() != 0) {
ALOGV("setOutputDevice() setting same device 0x%04x or null device", device);
return muteWaitMs;
}
ALOGV("setOutputDevice() changing device");
// do the routing
if (device == AUDIO_DEVICE_NONE) {
resetOutputDevice(outputDesc, delayMs, NULL);
} else {
DeviceVector deviceList;
if ((address == NULL) || (strlen(address) == 0)) {
deviceList = mAvailableOutputDevices.getDevicesFromType(device);
} else {
deviceList = mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address));
}
if (!deviceList.isEmpty()) {
struct audio_patch patch;
outputDesc->toAudioPortConfig(&patch.sources[0]);
patch.num_sources = 1;
patch.num_sinks = 0;
for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
patch.num_sinks++;
}
ssize_t index;
if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
}
sp< AudioPatch> patchDesc;
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (index >= 0) {
patchDesc = mAudioPatches.valueAt(index);
afPatchHandle = patchDesc->mAfPatchHandle;
}
status_t status = mpClientInterface->createAudioPatch(&patch,
&afPatchHandle,
delayMs);
ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
"num_sources %d num_sinks %d",
status, afPatchHandle, patch.num_sources, patch.num_sinks);
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch(&patch, mUidCached);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = patch;
}
patchDesc->mAfPatchHandle = afPatchHandle;
if (patchHandle) {
*patchHandle = patchDesc->mHandle;
}
outputDesc->setPatchHandle(patchDesc->mHandle);
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
}
}
// inform all input as well
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
if (!is_virtual_input_device(inputDescriptor->mDevice)) {
AudioParameter inputCmd = AudioParameter();
ALOGV("%s: inform input %d of device:%d", __func__,
inputDescriptor->mIoHandle, device);
inputCmd.addInt(String8(AudioParameter::keyRouting),device);
mpClientInterface->setParameters(inputDescriptor->mIoHandle,
inputCmd.toString(),
delayMs);
}
}
}
// update stream volumes according to new device
applyStreamVolumes(outputDesc, device, delayMs);
return muteWaitMs;
}
status_t AudioPolicyManager::resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
int delayMs,
audio_patch_handle_t *patchHandle)
{
ssize_t index;
if (patchHandle) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
}
if (index < 0) {
return INVALID_OPERATION;
}
sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs);
ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
outputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE);
removeAudioPatch(patchDesc->mHandle);
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
return status;
}
status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
audio_devices_t device,
bool force,
audio_patch_handle_t *patchHandle)
{
status_t status = NO_ERROR;
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
inputDesc->mDevice = device;
DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
if (!deviceList.isEmpty()) {
struct audio_patch patch;
inputDesc->toAudioPortConfig(&patch.sinks[0]);
// AUDIO_SOURCE_HOTWORD is for internal use only:
// handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
!inputDesc->isSoundTrigger()) {
patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
}
patch.num_sinks = 1;
//only one input device for now
deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
patch.num_sources = 1;
ssize_t index;
if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
}
sp< AudioPatch> patchDesc;
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (index >= 0) {
patchDesc = mAudioPatches.valueAt(index);
afPatchHandle = patchDesc->mAfPatchHandle;
}
status_t status = mpClientInterface->createAudioPatch(&patch,
&afPatchHandle,
0);
ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
status, afPatchHandle);
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch(&patch, mUidCached);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = patch;
}
patchDesc->mAfPatchHandle = afPatchHandle;
if (patchHandle) {
*patchHandle = patchDesc->mHandle;
}
inputDesc->setPatchHandle(patchDesc->mHandle);
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
}
}
}
return status;
}
status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input,
audio_patch_handle_t *patchHandle)
{
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
ssize_t index;
if (patchHandle) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
}
if (index < 0) {
return INVALID_OPERATION;
}
sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
inputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE);
removeAudioPatch(patchDesc->mHandle);
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
return status;
}
sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
String8 address,
uint32_t& samplingRate,
audio_format_t& format,
audio_channel_mask_t& channelMask,
audio_input_flags_t flags)
{
// Choose an input profile based on the requested capture parameters: select the first available
// profile supporting all requested parameters.
//
// TODO: perhaps isCompatibleProfile should return a "matching" score so we can return
// the best matching profile, not the first one.
for (size_t i = 0; i < mHwModules.size(); i++)
{
if (mHwModules[i]->mHandle == 0) {
continue;
}
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
{
sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
// profile->log();
if (profile->isCompatibleProfile(device, address, samplingRate,
&samplingRate /*updatedSamplingRate*/,
format,
&format /*updatedFormat*/,
channelMask,
&channelMask /*updatedChannelMask*/,
(audio_output_flags_t) flags,
true)) {
return profile;
}
}
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
{
sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
// profile->log();
if (profile->isCompatibleProfile(device, address, samplingRate,
&samplingRate /*updatedSamplingRate*/,
format,
&format /*updatedFormat*/,
channelMask,
&channelMask /*updatedChannelMask*/,
(audio_output_flags_t) flags,
false)) {
return profile;
}
}
}
return NULL;
}
audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource,
AudioMix **policyMix)
{
audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
audio_devices_t selectedDeviceFromMix =
mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix);
if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) {
return selectedDeviceFromMix;
}
return getDeviceForInputSource(inputSource);
}
audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
{
for (size_t routeIndex = 0; routeIndex < mInputRoutes.size(); routeIndex++) {
sp<SessionRoute> route = mInputRoutes.valueAt(routeIndex);
if (inputSource == route->mSource && route->isActive()) {
return route->mDeviceDescriptor->type();
}
}
return mEngine->getDeviceForInputSource(inputSource);
}
float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
int index,
audio_devices_t device)
{
float volumeDB = mVolumeCurves->volIndexToDb(stream, Volume::getDeviceCategory(device), index);
// handle the case of accessibility active while a ringtone is playing: if the ringtone is much
// louder than the accessibility prompt, the prompt cannot be heard, thus masking the touch
// exploration of the dialer UI. In this situation, bring the accessibility volume closer to
// the ringtone volume
if ((stream == AUDIO_STREAM_ACCESSIBILITY)
&& (AUDIO_MODE_RINGTONE == mEngine->getPhoneState())
&& isStreamActive(AUDIO_STREAM_RING, 0)) {
const float ringVolumeDB = computeVolume(AUDIO_STREAM_RING, index, device);
return ringVolumeDB - 4 > volumeDB ? ringVolumeDB - 4 : volumeDB;
}
// 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 notifications volume by 6dB
// - attenuate ring tones volume by 6dB unless music is not playing and
// speaker is part of the select devices
// - if music is playing, always limit the volume to current music volume,
// with a minimum threshold at -36dB so that notification is always perceived.
const routing_strategy stream_strategy = getStrategy(stream);
if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
AUDIO_DEVICE_OUT_WIRED_HEADSET |
AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
((stream_strategy == STRATEGY_SONIFICATION)
|| (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
|| (stream == AUDIO_STREAM_SYSTEM)
|| ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
(mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) &&
mVolumeCurves->canBeMuted(stream)) {
// when the phone is ringing we must consider that music could have been paused just before
// by the music application and behave as if music was active if the last music track was
// just stopped
if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
mLimitRingtoneVolume) {
volumeDB += SONIFICATION_HEADSET_VOLUME_FACTOR_DB;
audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
float musicVolDB = computeVolume(AUDIO_STREAM_MUSIC,
mVolumeCurves->getVolumeIndex(AUDIO_STREAM_MUSIC,
musicDevice),
musicDevice);
float minVolDB = (musicVolDB > SONIFICATION_HEADSET_VOLUME_MIN_DB) ?
musicVolDB : SONIFICATION_HEADSET_VOLUME_MIN_DB;
if (volumeDB > minVolDB) {
volumeDB = minVolDB;
ALOGV("computeVolume limiting volume to %f musicVol %f", minVolDB, musicVolDB);
}
if (device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES)) {
// on A2DP, also ensure notification volume is not too low compared to media when
// intended to be played
if ((volumeDB > -96.0f) &&
(musicVolDB - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB > volumeDB)) {
ALOGV("computeVolume increasing volume for stream=%d device=0x%X from %f to %f",
stream, device,
volumeDB, musicVolDB - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB);
volumeDB = musicVolDB - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB;
}
}
} else if ((Volume::getDeviceForVolume(device) != AUDIO_DEVICE_OUT_SPEAKER) ||
stream_strategy != STRATEGY_SONIFICATION) {
volumeDB += SONIFICATION_HEADSET_VOLUME_FACTOR_DB;
}
}
return volumeDB;
}
status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
int index,
const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device,
int delayMs,
bool force)
{
// 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) {
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
mLastVoiceVolume = voiceVolume;
}
}
return NO_ERROR;
}
void AudioPolicyManager::applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device,
int delayMs,
bool force)
{
ALOGVV("applyStreamVolumes() for device %08x", device);
for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
checkAndSetVolume((audio_stream_type_t)stream,
mVolumeCurves->getVolumeIndex((audio_stream_type_t)stream, device),
outputDesc,
device,
delayMs,
force);
}
}
void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs,
audio_devices_t device)
{
ALOGVV("setStrategyMute() strategy %d, mute %d, output ID %d",
strategy, on, outputDesc->getId());
for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
if (getStrategy((audio_stream_type_t)stream) == strategy) {
setStreamMute((audio_stream_type_t)stream, on, outputDesc, delayMs, device);
}
}
}
void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs,
audio_devices_t device)
{
if (device == AUDIO_DEVICE_NONE) {
device = outputDesc->device();
}
ALOGVV("setStreamMute() stream %d, mute %d, mMuteCount %d device %04x",
stream, on, outputDesc->mMuteCount[stream], device);
if (on) {
if (outputDesc->mMuteCount[stream] == 0) {
if (mVolumeCurves->canBeMuted(stream) &&
((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
(mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) {
checkAndSetVolume(stream, 0, outputDesc, 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,
mVolumeCurves->getVolumeIndex(stream, device),
outputDesc,
device,
delayMs);
}
}
}
void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
bool starting, bool stateChange)
{
if(!hasPrimaryOutput()) {
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 = mPrimaryOutput;
ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
stream, starting, outputDesc->mDevice, stateChange);
if (outputDesc->mRefCount[stream]) {
int muteCount = 1;
if (stateChange) {
muteCount = outputDesc->mRefCount[stream];
}
if (audio_is_low_visibility(stream)) {
ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
for (int i = 0; i < muteCount; i++) {
setStreamMute(stream, starting, mPrimaryOutput);
}
} else {
ALOGV("handleIncallSonification() high visibility");
if (outputDesc->device() &
getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
for (int i = 0; i < muteCount; i++) {
setStreamMute(stream, starting, mPrimaryOutput);
}
}
if (starting) {
mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
AUDIO_STREAM_VOICE_CALL);
} else {
mpClientInterface->stopTone();
}
}
}
}
}
audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
{
// flags to stream type mapping
if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
return AUDIO_STREAM_ENFORCED_AUDIBLE;
}
if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
return AUDIO_STREAM_BLUETOOTH_SCO;
}
if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
return AUDIO_STREAM_TTS;
}
// usage to stream type mapping
switch (attr->usage) {
case AUDIO_USAGE_MEDIA:
case AUDIO_USAGE_GAME:
case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
return AUDIO_STREAM_MUSIC;
case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
return AUDIO_STREAM_ACCESSIBILITY;
case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
return AUDIO_STREAM_SYSTEM;
case AUDIO_USAGE_VOICE_COMMUNICATION:
return AUDIO_STREAM_VOICE_CALL;
case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
return AUDIO_STREAM_DTMF;
case AUDIO_USAGE_ALARM:
return AUDIO_STREAM_ALARM;
case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
return AUDIO_STREAM_RING;
case AUDIO_USAGE_NOTIFICATION:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
case AUDIO_USAGE_NOTIFICATION_EVENT:
return AUDIO_STREAM_NOTIFICATION;
case AUDIO_USAGE_UNKNOWN:
default:
return AUDIO_STREAM_MUSIC;
}
}
bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa)
{
// has flags that map to a strategy?
if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) {
return true;
}
// has known usage?
switch (paa->usage) {
case AUDIO_USAGE_UNKNOWN:
case AUDIO_USAGE_MEDIA:
case AUDIO_USAGE_VOICE_COMMUNICATION:
case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
case AUDIO_USAGE_ALARM:
case AUDIO_USAGE_NOTIFICATION:
case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
case AUDIO_USAGE_NOTIFICATION_EVENT:
case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
case AUDIO_USAGE_GAME:
case AUDIO_USAGE_VIRTUAL_SOURCE:
break;
default:
return false;
}
return true;
}
bool AudioPolicyManager::isStrategyActive(const sp<AudioOutputDescriptor> outputDesc,
routing_strategy strategy, uint32_t inPastMs,
nsecs_t sysTime) const
{
if ((sysTime == 0) && (inPastMs != 0)) {
sysTime = systemTime();
}
for (int i = 0; i < (int)AUDIO_STREAM_FOR_POLICY_CNT; i++) {
if (((getStrategy((audio_stream_type_t)i) == strategy) ||
(NUM_STRATEGIES == strategy)) &&
outputDesc->isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
return true;
}
}
return false;
}
audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
{
return mEngine->getForceUse(usage);
}
bool AudioPolicyManager::isInCall()
{
return isStateInCall(mEngine->getPhoneState());
}
bool AudioPolicyManager::isStateInCall(int state)
{
return is_state_in_call(state);
}
void AudioPolicyManager::cleanUpForDevice(const sp<DeviceDescriptor>& deviceDesc)
{
for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--) {
sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
if (sourceDesc->mDevice->equals(deviceDesc)) {
ALOGV("%s releasing audio source %d", __FUNCTION__, sourceDesc->getHandle());
stopAudioSource(sourceDesc->getHandle());
}
}
for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
bool release = false;
for (size_t j = 0; j < patchDesc->mPatch.num_sources && !release; j++) {
const struct audio_port_config *source = &patchDesc->mPatch.sources[j];
if (source->type == AUDIO_PORT_TYPE_DEVICE &&
source->ext.device.type == deviceDesc->type()) {
release = true;
}
}
for (size_t j = 0; j < patchDesc->mPatch.num_sinks && !release; j++) {
const struct audio_port_config *sink = &patchDesc->mPatch.sinks[j];
if (sink->type == AUDIO_PORT_TYPE_DEVICE &&
sink->ext.device.type == deviceDesc->type()) {
release = true;
}
}
if (release) {
ALOGV("%s releasing patch %u", __FUNCTION__, patchDesc->mHandle);
releaseAudioPatch(patchDesc->mHandle, patchDesc->mUid);
}
}
}
// Modify the list of surround sound formats supported.
void AudioPolicyManager::filterSurroundFormats(FormatVector *formatsPtr) {
FormatVector &formats = *formatsPtr;
// TODO Set this based on Config properties.
const bool alwaysForceAC3 = true;
audio_policy_forced_cfg_t forceUse = mEngine->getForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND);
ALOGD("%s: forced use = %d", __FUNCTION__, forceUse);
// Analyze original support for various formats.
bool supportsAC3 = false;
bool supportsOtherSurround = false;
bool supportsIEC61937 = false;
for (size_t formatIndex = 0; formatIndex < formats.size(); formatIndex++) {
audio_format_t format = formats[formatIndex];
switch (format) {
case AUDIO_FORMAT_AC3:
supportsAC3 = true;
break;
case AUDIO_FORMAT_E_AC3:
case AUDIO_FORMAT_DTS:
case AUDIO_FORMAT_DTS_HD:
supportsOtherSurround = true;
break;
case AUDIO_FORMAT_IEC61937:
supportsIEC61937 = true;
break;
default:
break;
}
}
// Modify formats based on surround preferences.
// If NEVER, remove support for surround formats.
if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) {
if (supportsAC3 || supportsOtherSurround || supportsIEC61937) {
// Remove surround sound related formats.
for (size_t formatIndex = 0; formatIndex < formats.size(); ) {
audio_format_t format = formats[formatIndex];
switch(format) {
case AUDIO_FORMAT_AC3:
case AUDIO_FORMAT_E_AC3:
case AUDIO_FORMAT_DTS:
case AUDIO_FORMAT_DTS_HD:
case AUDIO_FORMAT_IEC61937:
formats.removeAt(formatIndex);
break;
default:
formatIndex++; // keep it
break;
}
}
supportsAC3 = false;
supportsOtherSurround = false;
supportsIEC61937 = false;
}
} else { // AUTO or ALWAYS
// Most TVs support AC3 even if they do not report it in the EDID.
if ((alwaysForceAC3 || (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS))
&& !supportsAC3) {
formats.add(AUDIO_FORMAT_AC3);
supportsAC3 = true;
}
// If ALWAYS, add support for raw surround formats if all are missing.
// This assumes that if any of these formats are reported by the HAL
// then the report is valid and should not be modified.
if ((forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS)
&& !supportsOtherSurround) {
formats.add(AUDIO_FORMAT_E_AC3);
formats.add(AUDIO_FORMAT_DTS);
formats.add(AUDIO_FORMAT_DTS_HD);
supportsOtherSurround = true;
}
// Add support for IEC61937 if any raw surround supported.
// The HAL could do this but add it here, just in case.
if ((supportsAC3 || supportsOtherSurround) && !supportsIEC61937) {
formats.add(AUDIO_FORMAT_IEC61937);
supportsIEC61937 = true;
}
}
}
// Modify the list of channel masks supported.
void AudioPolicyManager::filterSurroundChannelMasks(ChannelsVector *channelMasksPtr) {
ChannelsVector &channelMasks = *channelMasksPtr;
audio_policy_forced_cfg_t forceUse = mEngine->getForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND);
// If NEVER, then remove support for channelMasks > stereo.
if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) {
for (size_t maskIndex = 0; maskIndex < channelMasks.size(); ) {
audio_channel_mask_t channelMask = channelMasks[maskIndex];
if (channelMask & ~AUDIO_CHANNEL_OUT_STEREO) {
ALOGI("%s: force NEVER, so remove channelMask 0x%08x", __FUNCTION__, channelMask);
channelMasks.removeAt(maskIndex);
} else {
maskIndex++;
}
}
// If ALWAYS, then make sure we at least support 5.1
} else if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS) {
bool supports5dot1 = false;
// Are there any channel masks that can be considered "surround"?
for (size_t maskIndex = 0; maskIndex < channelMasks.size(); maskIndex++) {
audio_channel_mask_t channelMask = channelMasks[maskIndex];
if ((channelMask & AUDIO_CHANNEL_OUT_5POINT1) == AUDIO_CHANNEL_OUT_5POINT1) {
supports5dot1 = true;
break;
}
}
// If not then add 5.1 support.
if (!supports5dot1) {
channelMasks.add(AUDIO_CHANNEL_OUT_5POINT1);
ALOGI("%s: force ALWAYS, so adding channelMask for 5.1 surround", __FUNCTION__);
}
}
}
void AudioPolicyManager::updateAudioProfiles(audio_devices_t device,
audio_io_handle_t ioHandle,
AudioProfileVector &profiles)
{
String8 reply;
// Format MUST be checked first to update the list of AudioProfile
if (profiles.hasDynamicFormat()) {
reply = mpClientInterface->getParameters(ioHandle,
String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
ALOGV("%s: supported formats %s", __FUNCTION__, reply.string());
AudioParameter repliedParameters(reply);
if (repliedParameters.get(
String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS), reply) != NO_ERROR) {
ALOGE("%s: failed to retrieve format, bailing out", __FUNCTION__);
return;
}
FormatVector formats = formatsFromString(reply.string());
if (device == AUDIO_DEVICE_OUT_HDMI) {
filterSurroundFormats(&formats);
}
profiles.setFormats(formats);
}
const FormatVector &supportedFormats = profiles.getSupportedFormats();
for (size_t formatIndex = 0; formatIndex < supportedFormats.size(); formatIndex++) {
audio_format_t format = supportedFormats[formatIndex];
ChannelsVector channelMasks;
SampleRateVector samplingRates;
AudioParameter requestedParameters;
requestedParameters.addInt(String8(AUDIO_PARAMETER_STREAM_FORMAT), format);
if (profiles.hasDynamicRateFor(format)) {
reply = mpClientInterface->getParameters(ioHandle,
requestedParameters.toString() + ";" +
AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES);
ALOGV("%s: supported sampling rates %s", __FUNCTION__, reply.string());
AudioParameter repliedParameters(reply);
if (repliedParameters.get(
String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES), reply) == NO_ERROR) {
samplingRates = samplingRatesFromString(reply.string());
}
}
if (profiles.hasDynamicChannelsFor(format)) {
reply = mpClientInterface->getParameters(ioHandle,
requestedParameters.toString() + ";" +
AUDIO_PARAMETER_STREAM_SUP_CHANNELS);
ALOGV("%s: supported channel masks %s", __FUNCTION__, reply.string());
AudioParameter repliedParameters(reply);
if (repliedParameters.get(
String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS), reply) == NO_ERROR) {
channelMasks = channelMasksFromString(reply.string());
if (device == AUDIO_DEVICE_OUT_HDMI) {
filterSurroundChannelMasks(&channelMasks);
}
}
}
profiles.addProfileFromHal(new AudioProfile(format, channelMasks, samplingRates));
}
}
}; // namespace android
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