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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.
*/
#define LOG_TAG "APM_AudioPolicyManager"
// Need to keep the log statements even in production builds
// to enable VERBOSE logging dynamically.
// You can enable VERBOSE logging as follows:
// adb shell setprop log.tag.APM_AudioPolicyManager V
#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_PATH_MAX_LENGTH 128
#define AUDIO_POLICY_XML_CONFIG_FILE_NAME "audio_policy_configuration.xml"
#define AUDIO_POLICY_A2DP_OFFLOAD_DISABLED_XML_CONFIG_FILE_NAME \
"audio_policy_configuration_a2dp_offload_disabled.xml"
#define AUDIO_POLICY_BLUETOOTH_LEGACY_HAL_XML_CONFIG_FILE_NAME \
"audio_policy_configuration_bluetooth_legacy_hal.xml"
#include <algorithm>
#include <inttypes.h>
#include <math.h>
#include <set>
#include <unordered_set>
#include <vector>
#include <AudioPolicyManagerInterface.h>
#include <AudioPolicyEngineInstance.h>
#include <cutils/properties.h>
#include <utils/Log.h>
#include <media/AudioParameter.h>
#include <private/android_filesystem_config.h>
#include <soundtrigger/SoundTrigger.h>
#include <system/audio.h>
#include <audio_policy_conf.h>
#include "AudioPolicyManager.h"
#include <Serializer.h>
#include "TypeConverter.h"
#include <policy.h>
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
// Largest difference in dB on earpiece in call between the voice volume and another
// media / notification / system volume.
constexpr float IN_CALL_EARPIECE_HEADROOM_DB = 3.f;
// Compressed formats for MSD module, ordered from most preferred to least preferred.
static const std::vector<audio_format_t> compressedFormatsOrder = {{
AUDIO_FORMAT_MAT_2_1, AUDIO_FORMAT_MAT_2_0, AUDIO_FORMAT_E_AC3,
AUDIO_FORMAT_AC3, AUDIO_FORMAT_PCM_16_BIT }};
// Channel masks for MSD module, 3D > 2D > 1D ordering (most preferred to least preferred).
static const std::vector<audio_channel_mask_t> surroundChannelMasksOrder = {{
AUDIO_CHANNEL_OUT_3POINT1POINT2, AUDIO_CHANNEL_OUT_3POINT0POINT2,
AUDIO_CHANNEL_OUT_2POINT1POINT2, AUDIO_CHANNEL_OUT_2POINT0POINT2,
AUDIO_CHANNEL_OUT_5POINT1, AUDIO_CHANNEL_OUT_STEREO }};
// ----------------------------------------------------------------------------
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
const char *device_name,
audio_format_t encodedFormat)
{
status_t status = setDeviceConnectionStateInt(device, state, device_address,
device_name, encodedFormat);
nextAudioPortGeneration();
return status;
}
void AudioPolicyManager::broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
audio_policy_dev_state_t state)
{
AudioParameter param(device->address());
const String8 key(state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE ?
AudioParameter::keyStreamConnect : AudioParameter::keyStreamDisconnect);
param.addInt(key, device->type());
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
}
status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t deviceType,
audio_policy_dev_state_t state,
const char *device_address,
const char *device_name,
audio_format_t encodedFormat)
{
ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s format 0x%X",
deviceType, state, device_address, device_name, encodedFormat);
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(deviceType) && !audio_is_input_device(deviceType)) return BAD_VALUE;
sp<DeviceDescriptor> device =
mHwModules.getDeviceDescriptor(deviceType, device_address, device_name, encodedFormat,
state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
if (device == 0) {
return INVALID_OPERATION;
}
// handle output devices
if (audio_is_output_device(deviceType)) {
SortedVector <audio_io_handle_t> outputs;
ssize_t index = mAvailableOutputDevices.indexOf(device);
// 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("%s() device already connected: %s", __func__, device->toString().c_str());
return INVALID_OPERATION;
}
ALOGV("%s() connecting device %s format %x",
__func__, device->toString().c_str(), encodedFormat);
// register new device as available
if (mAvailableOutputDevices.add(device) < 0) {
return NO_MEMORY;
}
// Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
// parameters on newly connected devices (instead of opening the outputs...)
broadcastDeviceConnectionState(device, state);
if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
mAvailableOutputDevices.remove(device);
mHwModules.cleanUpForDevice(device);
broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
return INVALID_OPERATION;
}
// outputs should never be empty here
ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
"checkOutputsForDevice() returned no outputs but status OK");
ALOGV("%s() checkOutputsForDevice() returned %zu outputs", __func__, outputs.size());
} break;
// handle output device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
ALOGW("%s() device not connected: %s", __func__, device->toString().c_str());
return INVALID_OPERATION;
}
ALOGV("%s() disconnecting output device %s", __func__, device->toString().c_str());
// Send Disconnect to HALs
broadcastDeviceConnectionState(device, state);
// remove device from available output devices
mAvailableOutputDevices.remove(device);
mOutputs.clearSessionRoutesForDevice(device);
checkOutputsForDevice(device, state, outputs);
// Reset active device codec
device->setEncodedFormat(AUDIO_FORMAT_DEFAULT);
} break;
default:
ALOGE("%s() invalid state: %x", __func__, state);
return BAD_VALUE;
}
// Propagate device availability to Engine
setEngineDeviceConnectionState(device, state);
// No need to evaluate playback routing when connecting a remote submix
// output device used by a dynamic policy of type recorder as no
// playback use case is affected.
bool doCheckForDeviceAndOutputChanges = true;
if (device->type() == AUDIO_DEVICE_OUT_REMOTE_SUBMIX
&& strncmp(device_address, "0", AUDIO_DEVICE_MAX_ADDRESS_LEN) != 0) {
for (audio_io_handle_t output : outputs) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
sp<AudioPolicyMix> policyMix = desc->mPolicyMix.promote();
if (policyMix != nullptr
&& policyMix->mMixType == MIX_TYPE_RECORDERS
&& strncmp(device_address,
policyMix->mDeviceAddress.string(),
AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
doCheckForDeviceAndOutputChanges = false;
break;
}
}
}
auto checkCloseOutputs = [&]() {
// outputs must be closed after checkOutputForAllStrategies() is executed
if (!outputs.isEmpty()) {
for (audio_io_handle_t output : outputs) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
// 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(output);
}
}
// check A2DP again after closing A2DP output to reset mA2dpSuspended if needed
return true;
}
return false;
};
if (doCheckForDeviceAndOutputChanges) {
checkForDeviceAndOutputChanges(checkCloseOutputs);
} else {
checkCloseOutputs();
}
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevices);
}
const DeviceVector msdOutDevices = getMsdAudioOutDevices();
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
DeviceVector newDevices = getNewOutputDevices(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 = (msdOutDevices.isEmpty() || msdOutDevices != desc->devices())
&& !desc->isDuplicated()
&& (!device_distinguishes_on_address(deviceType)
// always force when disconnecting (a non-duplicated device)
|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
setOutputDevices(desc, newDevices, force, 0);
}
}
if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
cleanUpForDevice(device);
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is output device
// handle input devices
if (audio_is_input_device(deviceType)) {
ssize_t index = mAvailableInputDevices.indexOf(device);
switch (state)
{
// handle input device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
ALOGW("%s() device already connected: %s", __func__, device->toString().c_str());
return INVALID_OPERATION;
}
if (mAvailableInputDevices.add(device) < 0) {
return NO_MEMORY;
}
// Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
// parameters on newly connected devices (instead of opening the inputs...)
broadcastDeviceConnectionState(device, state);
if (checkInputsForDevice(device, state) != NO_ERROR) {
mAvailableInputDevices.remove(device);
broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
mHwModules.cleanUpForDevice(device);
return INVALID_OPERATION;
}
} break;
// handle input device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
ALOGW("%s() device not connected: %s", __func__, device->toString().c_str());
return INVALID_OPERATION;
}
ALOGV("%s() disconnecting input device %s", __func__, device->toString().c_str());
// Set Disconnect to HALs
broadcastDeviceConnectionState(device, state);
mAvailableInputDevices.remove(device);
checkInputsForDevice(device, state);
} break;
default:
ALOGE("%s() invalid state: %x", __func__, state);
return BAD_VALUE;
}
// Propagate device availability to Engine
setEngineDeviceConnectionState(device, state);
checkCloseInputs();
// As the input device list can impact the output device selection, update
// getDeviceForStrategy() cache
updateDevicesAndOutputs();
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevices);
}
if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
cleanUpForDevice(device);
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is input device
ALOGW("%s() invalid device: %s", __func__, device->toString().c_str());
return BAD_VALUE;
}
void AudioPolicyManager::setEngineDeviceConnectionState(const sp<DeviceDescriptor> device,
audio_policy_dev_state_t state) {
// the Engine does not have to know about remote submix devices used by dynamic audio policies
if (audio_is_remote_submix_device(device->type()) && device->address() != "0") {
return;
}
mEngine->setDeviceConnectionState(device, state);
}
audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
const char *device_address)
{
sp<DeviceDescriptor> devDesc =
mHwModules.getDeviceDescriptor(device, device_address, "", AUDIO_FORMAT_DEFAULT,
false /* allowToCreate */,
(strlen(device_address) != 0)/*matchAddress*/);
if (devDesc == 0) {
ALOGV("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("%s() invalid device type %08x", __func__, device);
return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
return (deviceVector->getDevice(
device, String8(device_address), AUDIO_FORMAT_DEFAULT) != 0) ?
AUDIO_POLICY_DEVICE_STATE_AVAILABLE : AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
status_t AudioPolicyManager::handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
const char *device_name,
audio_format_t encodedFormat)
{
status_t status;
String8 reply;
AudioParameter param;
int isReconfigA2dpSupported = 0;
ALOGV("handleDeviceConfigChange(() device: 0x%X, address %s name %s encodedFormat: 0x%X",
device, device_address, device_name, encodedFormat);
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
// Check if the device is currently connected
DeviceVector deviceList = mAvailableOutputDevices.getDevicesFromTypeMask(device);
if (deviceList.empty()) {
// Nothing to do: device is not connected
return NO_ERROR;
}
sp<DeviceDescriptor> devDesc = deviceList.itemAt(0);
// For offloaded A2DP, Hw modules may have the capability to
// configure codecs.
// Handle two specific cases by sending a set parameter to
// configure A2DP codecs. No need to toggle device state.
// Case 1: A2DP active device switches from primary to primary
// module
// Case 2: A2DP device config changes on primary module.
if (device & AUDIO_DEVICE_OUT_ALL_A2DP) {
sp<HwModule> module = mHwModules.getModuleForDeviceTypes(device, encodedFormat);
audio_module_handle_t primaryHandle = mPrimaryOutput->getModuleHandle();
if (availablePrimaryOutputDevices().contains(devDesc) &&
(module != 0 && module->getHandle() == primaryHandle)) {
reply = mpClientInterface->getParameters(
AUDIO_IO_HANDLE_NONE,
String8(AudioParameter::keyReconfigA2dpSupported));
AudioParameter repliedParameters(reply);
repliedParameters.getInt(
String8(AudioParameter::keyReconfigA2dpSupported), isReconfigA2dpSupported);
if (isReconfigA2dpSupported) {
const String8 key(AudioParameter::keyReconfigA2dp);
param.add(key, String8("true"));
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
devDesc->setEncodedFormat(encodedFormat);
return NO_ERROR;
}
}
}
// Toggle the device state: UNAVAILABLE -> AVAILABLE
// This will force reading again the device configuration
status = setDeviceConnectionState(device,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
device_address, device_name,
devDesc->getEncodedFormat());
if (status != NO_ERROR) {
ALOGW("handleDeviceConfigChange() error disabling connection state: %d",
status);
return status;
}
status = setDeviceConnectionState(device,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
device_address, device_name, encodedFormat);
if (status != NO_ERROR) {
ALOGW("handleDeviceConfigChange() error enabling connection state: %d",
status);
return status;
}
return NO_ERROR;
}
status_t AudioPolicyManager::getHwOffloadEncodingFormatsSupportedForA2DP(
std::vector<audio_format_t> *formats)
{
ALOGV("getHwOffloadEncodingFormatsSupportedForA2DP()");
status_t status = NO_ERROR;
std::unordered_set<audio_format_t> formatSet;
sp<HwModule> primaryModule =
mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY);
DeviceVector declaredDevices = primaryModule->getDeclaredDevices().getDevicesFromTypeMask(
AUDIO_DEVICE_OUT_ALL_A2DP);
for (const auto& device : declaredDevices) {
formatSet.insert(device->encodedFormats().begin(), device->encodedFormats().end());
}
formats->assign(formatSet.begin(), formatSet.end());
return status;
}
uint32_t AudioPolicyManager::updateCallRouting(const DeviceVector &rxDevices, uint32_t delayMs)
{
bool createTxPatch = false;
bool createRxPatch = false;
uint32_t muteWaitMs = 0;
if(!hasPrimaryOutput() || mPrimaryOutput->devices().types() == AUDIO_DEVICE_OUT_STUB) {
return muteWaitMs;
}
ALOG_ASSERT(!rxDevices.isEmpty(), "updateCallRouting() no selected output device");
audio_attributes_t attr = { .source = AUDIO_SOURCE_VOICE_COMMUNICATION };
auto txSourceDevice = mEngine->getInputDeviceForAttributes(attr);
ALOG_ASSERT(txSourceDevice != 0, "updateCallRouting() input selected device not available");
ALOGV("updateCallRouting device rxDevice %s txDevice %s",
rxDevices.itemAt(0)->toString().c_str(), txSourceDevice->toString().c_str());
// 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();
}
auto telephonyRxModule =
mHwModules.getModuleForDeviceTypes(AUDIO_DEVICE_IN_TELEPHONY_RX, AUDIO_FORMAT_DEFAULT);
auto telephonyTxModule =
mHwModules.getModuleForDeviceTypes(AUDIO_DEVICE_OUT_TELEPHONY_TX, AUDIO_FORMAT_DEFAULT);
// retrieve Rx Source and Tx Sink device descriptors
sp<DeviceDescriptor> rxSourceDevice =
mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_TELEPHONY_RX,
String8(),
AUDIO_FORMAT_DEFAULT);
sp<DeviceDescriptor> txSinkDevice =
mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX,
String8(),
AUDIO_FORMAT_DEFAULT);
// RX and TX Telephony device are declared by Primary Audio HAL
if (isPrimaryModule(telephonyRxModule) && isPrimaryModule(telephonyTxModule) &&
(telephonyRxModule->getHalVersionMajor() >= 3)) {
if (rxSourceDevice == 0 || txSinkDevice == 0) {
// RX / TX Telephony device(s) is(are) not currently available
ALOGE("updateCallRouting() no telephony Tx and/or RX device");
return muteWaitMs;
}
// do not create a patch (aka Sw Bridging) if Primary HW module has declared supporting a
// route between telephony RX to Sink device and Source device to telephony TX
const auto &primaryModule = telephonyRxModule;
createRxPatch = !primaryModule->supportsPatch(rxSourceDevice, rxDevices.itemAt(0));
createTxPatch = !primaryModule->supportsPatch(txSourceDevice, txSinkDevice);
} else {
// 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.
createRxPatch = !(availablePrimaryOutputDevices().contains(rxDevices.itemAt(0))) &&
(rxSourceDevice != 0);
// 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.
createTxPatch = !(availablePrimaryModuleInputDevices().contains(txSourceDevice)) &&
(txSinkDevice != 0);
}
// Use legacy routing method for voice calls via setOutputDevice() on primary output.
// Otherwise, create two audio patches for TX and RX path.
if (!createRxPatch) {
muteWaitMs = setOutputDevices(mPrimaryOutput, rxDevices, true, delayMs);
} else { // create RX path audio patch
mCallRxPatch = createTelephonyPatch(true /*isRx*/, rxDevices.itemAt(0), delayMs);
// If the TX device is on the primary HW module but RX device is
// on other HW module, SinkMetaData of telephony input should handle it
// assuming the device uses audio HAL V5.0 and above
}
if (createTxPatch) { // create TX path audio patch
mCallTxPatch = createTelephonyPatch(false /*isRx*/, txSourceDevice, delayMs);
}
return muteWaitMs;
}
sp<AudioPatch> AudioPolicyManager::createTelephonyPatch(
bool isRx, const sp<DeviceDescriptor> &device, uint32_t delayMs) {
PatchBuilder patchBuilder;
if (device == nullptr) {
return nullptr;
}
if (isRx) {
patchBuilder.addSink(device).
addSource(mAvailableInputDevices.getDevice(
AUDIO_DEVICE_IN_TELEPHONY_RX, String8(), AUDIO_FORMAT_DEFAULT));
} else {
patchBuilder.addSource(device).
addSink(mAvailableOutputDevices.getDevice(
AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT));
}
// @TODO: still ignoring the address, or not dealing platform with mutliple telephonydevices
const sp<DeviceDescriptor> outputDevice = isRx ?
device : mAvailableOutputDevices.getDevice(
AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT);
SortedVector<audio_io_handle_t> outputs =
getOutputsForDevices(DeviceVector(outputDevice), mOutputs);
const audio_io_handle_t output = selectOutput(outputs);
// request to reuse existing output stream if one is already opened to reach the target device
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ALOG_ASSERT(!outputDesc->isDuplicated(), "%s() %s device output %d is duplicated", __func__,
outputDevice->toString().c_str(), output);
patchBuilder.addSource(outputDesc, { .stream = AUDIO_STREAM_PATCH });
}
if (!isRx) {
// 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()
for (const auto& activeDesc : mInputs.getActiveInputs()) {
if (activeDesc->hasSameHwModuleAs(device)) {
closeActiveClients(activeDesc);
}
}
}
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
status_t status = mpClientInterface->createAudioPatch(
patchBuilder.patch(), &afPatchHandle, delayMs);
ALOGW_IF(status != NO_ERROR,
"%s() error %d creating %s audio patch", __func__, status, isRx ? "RX" : "TX");
sp<AudioPatch> audioPatch;
if (status == NO_ERROR) {
audioPatch = new AudioPatch(patchBuilder.patch(), mUidCached);
audioPatch->mAfPatchHandle = afPatchHandle;
audioPatch->mUid = mUidCached;
}
return audioPatch;
}
sp<DeviceDescriptor> AudioPolicyManager::findDevice(
const DeviceVector& devices, audio_devices_t device) const {
DeviceVector deviceList = devices.getDevicesFromTypeMask(device);
ALOG_ASSERT(!deviceList.isEmpty(),
"%s() selected device type %#x is not in devices list", __func__, device);
return deviceList.itemAt(0);
}
audio_devices_t AudioPolicyManager::getModuleDeviceTypes(
const DeviceVector& devices, const char *moduleId) const {
sp<HwModule> mod = mHwModules.getModuleFromName(moduleId);
return mod != 0 ? devices.getDeviceTypesFromHwModule(mod->getHandle()) : AUDIO_DEVICE_NONE;
}
bool AudioPolicyManager::isDeviceOfModule(
const sp<DeviceDescriptor>& devDesc, const char *moduleId) const {
sp<HwModule> module = mHwModules.getModuleFromName(moduleId);
if (module != 0) {
return mAvailableOutputDevices.getDevicesFromHwModule(module->getHandle())
.indexOf(devDesc) != NAME_NOT_FOUND
|| mAvailableInputDevices.getDevicesFromHwModule(module->getHandle())
.indexOf(devDesc) != NAME_NOT_FOUND;
}
return false;
}
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 (isStateInCall(oldState)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
// 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
checkForDeviceAndOutputChanges();
int delayMs = 0;
if (isStateInCall(state)) {
nsecs_t sysTime = systemTime();
auto musicStrategy = streamToStrategy(AUDIO_STREAM_MUSIC);
auto sonificationStrategy = streamToStrategy(AUDIO_STREAM_ALARM);
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 ((desc->isStrategyActive(musicStrategy, SONIFICATION_HEADSET_MUSIC_DELAY, sysTime) ||
desc->isStrategyActive(sonificationStrategy, SONIFICATION_HEADSET_MUSIC_DELAY,
sysTime)) &&
(delayMs < (int)desc->latency()*2)) {
delayMs = desc->latency()*2;
}
setStrategyMute(musicStrategy, true, desc);
setStrategyMute(musicStrategy, false, desc, MUTE_TIME_MS,
mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA),
nullptr, true /*fromCache*/).types());
setStrategyMute(sonificationStrategy, true, desc);
setStrategyMute(sonificationStrategy, false, desc, MUTE_TIME_MS,
mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_ALARM),
nullptr, true /*fromCache*/).types());
}
}
if (hasPrimaryOutput()) {
// Note that despite the fact that getNewOutputDevices() is called on the primary output,
// the device returned is not necessarily reachable via this output
DeviceVector rxDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
// force routing command to audio hardware when ending call
// even if no device change is needed
if (isStateInCall(oldState) && rxDevices.isEmpty()) {
rxDevices = mPrimaryOutput->devices();
}
if (state == AUDIO_MODE_IN_CALL) {
updateCallRouting(rxDevices, 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();
}
setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
} else {
setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
}
}
// reevaluate routing on all outputs in case tracks have been started during the call
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
if (state != AUDIO_MODE_IN_CALL || desc != mPrimaryOutput) {
setOutputDevices(desc, newDevices, !newDevices.isEmpty(), 0 /*delayMs*/);
}
}
if (isStateInCall(state)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
// 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
mLimitRingtoneVolume = (state == AUDIO_MODE_RINGTONE &&
isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY));
}
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 (config == mEngine->getForceUse(usage)) {
return;
}
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
checkForDeviceAndOutputChanges();
// force client reconnection to reevaluate flag AUDIO_FLAG_AUDIBILITY_ENFORCED
if (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM) {
mpClientInterface->invalidateStream(AUDIO_STREAM_SYSTEM);
mpClientInterface->invalidateStream(AUDIO_STREAM_ENFORCED_AUDIBLE);
}
//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()) {
DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
waitMs = updateCallRouting(newDevices, delayMs);
}
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
// As done in setDeviceConnectionState, we could also fix default device issue by
// preventing the force re-routing in case of default dev that distinguishes on address.
// Let's give back to engine full device choice decision however.
waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), delayMs);
}
if (forceVolumeReeval && !newDevices.isEmpty()) {
applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
}
}
for (const auto& activeDesc : mInputs.getActiveInputs()) {
auto newDevice = getNewInputDevice(activeDesc);
// Force new input selection if the new device can not be reached via current input
if (activeDesc->mProfile->getSupportedDevices().contains(newDevice)) {
setInputDevice(activeDesc->mIoHandle, newDevice);
} else {
closeInput(activeDesc->mIoHandle);
}
}
}
void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
{
ALOGV("setSystemProperty() property %s, value %s", property, value);
}
// Find an output profile compatible with the parameters passed. When "directOnly" is set, restrict
// search to profiles for direct outputs.
sp<IOProfile> AudioPolicyManager::getProfileForOutput(
const DeviceVector& devices,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
bool directOnly)
{
if (directOnly) {
// 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 (const auto& hwModule : mHwModules) {
for (const auto& curProfile : hwModule->getOutputProfiles()) {
if (!curProfile->isCompatibleProfile(devices,
samplingRate, NULL /*updatedSamplingRate*/,
format, NULL /*updatedFormat*/,
channelMask, NULL /*updatedChannelMask*/,
flags)) {
continue;
}
// reject profiles not corresponding to a device currently available
if (!mAvailableOutputDevices.containsAtLeastOne(curProfile->getSupportedDevices())) {
continue;
}
// reject profiles if connected device does not support codec
if (!curProfile->deviceSupportsEncodedFormats(devices.types())) {
continue;
}
if (!directOnly) return curProfile;
// when searching for direct outputs, 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)
{
DeviceVector devices = mEngine->getOutputDevicesForStream(stream, false /*fromCache*/);
// Note that related method getOutputForAttr() uses getOutputForDevice() not selectOutput().
// We use selectOutput() here since we don't have the desired AudioTrack sample rate,
// format, flags, etc. This may result in some discrepancy for functions that utilize
// getOutput() solely on audio_stream_type such as AudioSystem::getOutputFrameCount()
// and AudioSystem::getOutputSamplingRate().
SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
const audio_io_handle_t output = selectOutput(outputs);
ALOGV("getOutput() stream %d selected devices %s, output %d", stream,
devices.toString().c_str(), output);
return output;
}
status_t AudioPolicyManager::getAudioAttributes(audio_attributes_t *dstAttr,
const audio_attributes_t *srcAttr,
audio_stream_type_t srcStream)
{
if (srcAttr != NULL) {
if (!isValidAttributes(srcAttr)) {
ALOGE("%s invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
__func__,
srcAttr->usage, srcAttr->content_type, srcAttr->flags,
srcAttr->tags);
return BAD_VALUE;
}
*dstAttr = *srcAttr;
} else {
if (srcStream < AUDIO_STREAM_MIN || srcStream >= AUDIO_STREAM_PUBLIC_CNT) {
ALOGE("%s: invalid stream type", __func__);
return BAD_VALUE;
}
*dstAttr = mEngine->getAttributesForStreamType(srcStream);
}
// Only honor audibility enforced when required. The client will be
// forced to reconnect if the forced usage changes.
if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
dstAttr->flags &= ~AUDIO_FLAG_AUDIBILITY_ENFORCED;
}
return NO_ERROR;
}
status_t AudioPolicyManager::getOutputForAttrInt(
audio_attributes_t *resultAttr,
audio_io_handle_t *output,
audio_session_t session,
const audio_attributes_t *attr,
audio_stream_type_t *stream,
uid_t uid,
const audio_config_t *config,
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId,
bool *isRequestedDeviceForExclusiveUse,
std::vector<sp<SwAudioOutputDescriptor>> *secondaryDescs)
{
DeviceVector outputDevices;
const audio_port_handle_t requestedPortId = *selectedDeviceId;
DeviceVector msdDevices = getMsdAudioOutDevices();
const sp<DeviceDescriptor> requestedDevice =
mAvailableOutputDevices.getDeviceFromId(requestedPortId);
status_t status = getAudioAttributes(resultAttr, attr, *stream);
if (status != NO_ERROR) {
return status;
}
if (auto it = mAllowedCapturePolicies.find(uid); it != end(mAllowedCapturePolicies)) {
resultAttr->flags |= it->second;
}
*stream = mEngine->getStreamTypeForAttributes(*resultAttr);
ALOGV("%s() attributes=%s stream=%s session %d selectedDeviceId %d", __func__,
toString(*resultAttr).c_str(), toString(*stream).c_str(), session, requestedPortId);
// The primary output is the explicit routing (eg. setPreferredDevice) if specified,
// otherwise, fallback to the dynamic policies, if none match, query the engine.
// Secondary outputs are always found by dynamic policies as the engine do not support them
sp<SwAudioOutputDescriptor> policyDesc;
status = mPolicyMixes.getOutputForAttr(*resultAttr, uid, *flags, policyDesc, secondaryDescs);
if (status != OK) {
return status;
}
// Explicit routing is higher priority then any dynamic policy primary output
bool usePrimaryOutputFromPolicyMixes = requestedDevice == nullptr && policyDesc != nullptr;
// FIXME: in case of RENDER policy, the output capabilities should be checked
if ((usePrimaryOutputFromPolicyMixes || !secondaryDescs->empty())
&& !audio_is_linear_pcm(config->format)) {
ALOGD("%s: rejecting request as dynamic audio policy only support pcm", __func__);
return BAD_VALUE;
}
if (usePrimaryOutputFromPolicyMixes) {
*output = policyDesc->mIoHandle;
sp<AudioPolicyMix> mix = policyDesc->mPolicyMix.promote();
sp<DeviceDescriptor> deviceDesc =
mAvailableOutputDevices.getDevice(mix->mDeviceType,
mix->mDeviceAddress,
AUDIO_FORMAT_DEFAULT);
*selectedDeviceId = deviceDesc != 0 ? deviceDesc->getId() : AUDIO_PORT_HANDLE_NONE;
ALOGV("getOutputForAttr() returns output %d", *output);
return NO_ERROR;
}
// Virtual sources must always be dynamicaly or explicitly routed
if (resultAttr->usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
return BAD_VALUE;
}
// explicit routing managed by getDeviceForStrategy in APM is now handled by engine
// in order to let the choice of the order to future vendor engine
outputDevices = mEngine->getOutputDevicesForAttributes(*resultAttr, requestedDevice, false);
if ((resultAttr->flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
*flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
}
// Set incall music only if device was explicitly set, and fallback to the device which is
// chosen by the engine if not.
// FIXME: provide a more generic approach which is not device specific and move this back
// to getOutputForDevice.
// TODO: Remove check of AUDIO_STREAM_MUSIC once migration is completed on the app side.
if (outputDevices.types() == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
(*stream == AUDIO_STREAM_MUSIC || resultAttr->usage == AUDIO_USAGE_VOICE_COMMUNICATION) &&
audio_is_linear_pcm(config->format) &&
isInCall()) {
if (requestedPortId != AUDIO_PORT_HANDLE_NONE) {
*flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
*isRequestedDeviceForExclusiveUse = true;
}
}
ALOGV("%s() device %s, sampling rate %d, format %#x, channel mask %#x, flags %#x stream %s",
__func__, outputDevices.toString().c_str(), config->sample_rate, config->format,
config->channel_mask, *flags, toString(*stream).c_str());
*output = AUDIO_IO_HANDLE_NONE;
if (!msdDevices.isEmpty()) {
*output = getOutputForDevices(msdDevices, session, *stream, config, flags);
sp<DeviceDescriptor> device = outputDevices.isEmpty() ? nullptr : outputDevices.itemAt(0);
if (*output != AUDIO_IO_HANDLE_NONE && setMsdPatch(device) == NO_ERROR) {
ALOGV("%s() Using MSD devices %s instead of devices %s",
__func__, msdDevices.toString().c_str(), outputDevices.toString().c_str());
outputDevices = msdDevices;
} else {
*output = AUDIO_IO_HANDLE_NONE;
}
}
if (*output == AUDIO_IO_HANDLE_NONE) {
*output = getOutputForDevices(outputDevices, session, *stream, config,
flags, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
}
if (*output == AUDIO_IO_HANDLE_NONE) {
return INVALID_OPERATION;
}
*selectedDeviceId = getFirstDeviceId(outputDevices);
ALOGV("%s returns output %d selectedDeviceId %d", __func__, *output, *selectedDeviceId);
return NO_ERROR;
}
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,
const audio_config_t *config,
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs)
{
// The supplied portId must be AUDIO_PORT_HANDLE_NONE
if (*portId != AUDIO_PORT_HANDLE_NONE) {
return INVALID_OPERATION;
}
const audio_port_handle_t requestedPortId = *selectedDeviceId;
audio_attributes_t resultAttr;
bool isRequestedDeviceForExclusiveUse = false;
std::vector<sp<SwAudioOutputDescriptor>> secondaryOutputDescs;
const sp<DeviceDescriptor> requestedDevice =
mAvailableOutputDevices.getDeviceFromId(requestedPortId);
// Prevent from storing invalid requested device id in clients
const audio_port_handle_t sanitizedRequestedPortId =
requestedDevice != nullptr ? requestedPortId : AUDIO_PORT_HANDLE_NONE;
*selectedDeviceId = sanitizedRequestedPortId;
status_t status = getOutputForAttrInt(&resultAttr, output, session, attr, stream, uid,
config, flags, selectedDeviceId, &isRequestedDeviceForExclusiveUse,
&secondaryOutputDescs);
if (status != NO_ERROR) {
return status;
}
std::vector<wp<SwAudioOutputDescriptor>> weakSecondaryOutputDescs;
for (auto& secondaryDesc : secondaryOutputDescs) {
secondaryOutputs->push_back(secondaryDesc->mIoHandle);
weakSecondaryOutputDescs.push_back(secondaryDesc);
}
audio_config_base_t clientConfig = {.sample_rate = config->sample_rate,
.format = config->format,
.channel_mask = config->channel_mask };
*portId = AudioPort::getNextUniqueId();
sp<TrackClientDescriptor> clientDesc =
new TrackClientDescriptor(*portId, uid, session, resultAttr, clientConfig,
sanitizedRequestedPortId, *stream,
mEngine->getProductStrategyForAttributes(resultAttr),
toVolumeSource(resultAttr),
*flags, isRequestedDeviceForExclusiveUse,
std::move(weakSecondaryOutputDescs));
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(*output);
outputDesc->addClient(clientDesc);
ALOGV("%s() returns output %d requestedPortId %d selectedDeviceId %d for port ID %d", __func__,
*output, requestedPortId, *selectedDeviceId, *portId);
return NO_ERROR;
}
audio_io_handle_t AudioPolicyManager::getOutputForDevices(
const DeviceVector &devices,
audio_session_t session,
audio_stream_type_t stream,
const audio_config_t *config,
audio_output_flags_t *flags,
bool forceMutingHaptic)
{
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status;
// Discard haptic channel mask when forcing muting haptic channels.
audio_channel_mask_t channelMask = forceMutingHaptic
? (config->channel_mask & ~AUDIO_CHANNEL_HAPTIC_ALL) : config->channel_mask;
// 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;
} else if (stream == AUDIO_STREAM_VOICE_CALL &&
audio_is_linear_pcm(config->format) &&
(*flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) == 0) {
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
AUDIO_OUTPUT_FLAG_DIRECT);
ALOGV("Set VoIP and Direct output flags for PCM format");
}
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(config->format) && config->sample_rate <= 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 = getProfileForOutput(devices,
config->sample_rate,
config->format,
channelMask,
(audio_output_flags_t)*flags,
true /* directOnly */);
}
if (profile != 0) {
// exclusive outputs for MMAP and Offload are enforced by different session ids.
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
// reuse direct output if currently open by the same client
// and configured with same parameters
if ((config->sample_rate == desc->mSamplingRate) &&
(config->format == desc->mFormat) &&
(channelMask == desc->mChannelMask) &&
(session == desc->mDirectClientSession)) {
desc->mDirectOpenCount++;
ALOGI("%s reusing direct output %d for session %d", __func__,
mOutputs.keyAt(i), session);
return mOutputs.keyAt(i);
}
}
}
if (!profile->canOpenNewIo()) {
goto non_direct_output;
}
sp<SwAudioOutputDescriptor> outputDesc =
new SwAudioOutputDescriptor(profile, mpClientInterface);
String8 address = getFirstDeviceAddress(devices);
// MSD patch may be using the only output stream that can service this request. Release
// MSD patch to prioritize this request over any active output on MSD.
AudioPatchCollection msdPatches = getMsdPatches();
for (size_t i = 0; i < msdPatches.size(); i++) {
const auto& patch = msdPatches[i];
for (size_t j = 0; j < patch->mPatch.num_sinks; ++j) {
const struct audio_port_config *sink = &patch->mPatch.sinks[j];
if (sink->type == AUDIO_PORT_TYPE_DEVICE &&
(sink->ext.device.type & devices.types()) != AUDIO_DEVICE_NONE &&
(address.isEmpty() || strncmp(sink->ext.device.address, address.string(),
AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
releaseAudioPatch(patch->mHandle, mUidCached);
break;
}
}
}
status = outputDesc->open(config, devices, stream, *flags, &output);
// only accept an output with the requested parameters
if (status != NO_ERROR ||
(config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
(config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->mFormat) ||
(channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
ALOGV("%s failed opening direct output: output %d sample rate %d %d,"
"format %d %d, channel mask %04x %04x", __func__, output, config->sample_rate,
outputDesc->mSamplingRate, config->format, outputDesc->mFormat,
channelMask, outputDesc->mChannelMask);
if (output != AUDIO_IO_HANDLE_NONE) {
outputDesc->close();
}
// fall back to mixer output if possible when the direct output could not be open
if (audio_is_linear_pcm(config->format) &&
config->sample_rate <= SAMPLE_RATE_HZ_MAX) {
goto non_direct_output;
}
return AUDIO_IO_HANDLE_NONE;
}
outputDesc->mDirectOpenCount = 1;
outputDesc->mDirectClientSession = session;
addOutput(output, outputDesc);
mPreviousOutputs = mOutputs;
ALOGV("%s returns new direct output %d", __func__, 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 | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)) != 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(config->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 = getOutputsForDevices(devices, 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, config->format, channelMask, config->sample_rate);
}
ALOGW_IF((output == 0), "getOutputForDevices() could not find output for stream %d, "
"sampling rate %d, format %#x, channels %#x, flags %#x",
stream, config->sample_rate, config->format, channelMask, *flags);
return output;
}
sp<DeviceDescriptor> AudioPolicyManager::getMsdAudioInDevice() const {
auto msdInDevices = mHwModules.getAvailableDevicesFromModuleName(AUDIO_HARDWARE_MODULE_ID_MSD,
mAvailableInputDevices);
return msdInDevices.isEmpty()? nullptr : msdInDevices.itemAt(0);
}
DeviceVector AudioPolicyManager::getMsdAudioOutDevices() const {
return mHwModules.getAvailableDevicesFromModuleName(AUDIO_HARDWARE_MODULE_ID_MSD,
mAvailableOutputDevices);
}
const AudioPatchCollection AudioPolicyManager::getMsdPatches() const {
AudioPatchCollection msdPatches;
sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
if (msdModule != 0) {
for (size_t i = 0; i < mAudioPatches.size(); ++i) {
sp<AudioPatch> patch = mAudioPatches.valueAt(i);
for (size_t j = 0; j < patch->mPatch.num_sources; ++j) {
const struct audio_port_config *source = &patch->mPatch.sources[j];
if (source->type == AUDIO_PORT_TYPE_DEVICE &&
source->ext.device.hw_module == msdModule->getHandle()) {
msdPatches.addAudioPatch(patch->mHandle, patch);
}
}
}
}
return msdPatches;
}
status_t AudioPolicyManager::getBestMsdAudioProfileFor(const sp<DeviceDescriptor> &outputDevice,
bool hwAvSync, audio_port_config *sourceConfig, audio_port_config *sinkConfig) const
{
sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
if (msdModule == nullptr) {
ALOGE("%s() unable to get MSD module", __func__);
return NO_INIT;
}
sp<HwModule> deviceModule = mHwModules.getModuleForDevice(outputDevice, AUDIO_FORMAT_DEFAULT);
if (deviceModule == nullptr) {
ALOGE("%s() unable to get module for %s", __func__, outputDevice->toString().c_str());
return NO_INIT;
}
const InputProfileCollection &inputProfiles = msdModule->getInputProfiles();
if (inputProfiles.isEmpty()) {
ALOGE("%s() no input profiles for MSD module", __func__);
return NO_INIT;
}
const OutputProfileCollection &outputProfiles = deviceModule->getOutputProfiles();
if (outputProfiles.isEmpty()) {
ALOGE("%s() no output profiles for device %s", __func__, outputDevice->toString().c_str());
return NO_INIT;
}
AudioProfileVector msdProfiles;
// Each IOProfile represents a MixPort from audio_policy_configuration.xml
for (const auto &inProfile : inputProfiles) {
if (hwAvSync == ((inProfile->getFlags() & AUDIO_INPUT_FLAG_HW_AV_SYNC) != 0)) {
msdProfiles.appendVector(inProfile->getAudioProfiles());
}
}
AudioProfileVector deviceProfiles;
for (const auto &outProfile : outputProfiles) {
if (hwAvSync == ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0)) {
deviceProfiles.appendVector(outProfile->getAudioProfiles());
}
}
struct audio_config_base bestSinkConfig;
status_t result = msdProfiles.findBestMatchingOutputConfig(deviceProfiles,
compressedFormatsOrder, surroundChannelMasksOrder, true /*preferHigherSamplingRates*/,
&bestSinkConfig);
if (result != NO_ERROR) {
ALOGD("%s() no matching profiles found for device: %s, hwAvSync: %d",
__func__, outputDevice->toString().c_str(), hwAvSync);
return result;
}
sinkConfig->sample_rate = bestSinkConfig.sample_rate;
sinkConfig->channel_mask = bestSinkConfig.channel_mask;
sinkConfig->format = bestSinkConfig.format;
// For encoded streams force direct flag to prevent downstream mixing.
sinkConfig->flags.output = static_cast<audio_output_flags_t>(
sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_DIRECT);
sourceConfig->sample_rate = bestSinkConfig.sample_rate;
// Specify exact channel mask to prevent guessing by bit count in PatchPanel.
sourceConfig->channel_mask = audio_channel_mask_out_to_in(bestSinkConfig.channel_mask);
sourceConfig->format = bestSinkConfig.format;
// Copy input stream directly without any processing (e.g. resampling).
sourceConfig->flags.input = static_cast<audio_input_flags_t>(
sourceConfig->flags.input | AUDIO_INPUT_FLAG_DIRECT);
if (hwAvSync) {
sinkConfig->flags.output = static_cast<audio_output_flags_t>(
sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
sourceConfig->flags.input = static_cast<audio_input_flags_t>(
sourceConfig->flags.input | AUDIO_INPUT_FLAG_HW_AV_SYNC);
}
const unsigned int config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE |
AUDIO_PORT_CONFIG_CHANNEL_MASK | AUDIO_PORT_CONFIG_FORMAT | AUDIO_PORT_CONFIG_FLAGS;
sinkConfig->config_mask |= config_mask;
sourceConfig->config_mask |= config_mask;
return NO_ERROR;
}
PatchBuilder AudioPolicyManager::buildMsdPatch(const sp<DeviceDescriptor> &outputDevice) const
{
PatchBuilder patchBuilder;
patchBuilder.addSource(getMsdAudioInDevice()).addSink(outputDevice);
audio_port_config sourceConfig = patchBuilder.patch()->sources[0];
audio_port_config sinkConfig = patchBuilder.patch()->sinks[0];
// TODO: Figure out whether MSD module has HW_AV_SYNC flag set in the AP config file.
// For now, we just forcefully try with HwAvSync first.
status_t res = getBestMsdAudioProfileFor(outputDevice, true /*hwAvSync*/,
&sourceConfig, &sinkConfig) == NO_ERROR ? NO_ERROR :
getBestMsdAudioProfileFor(
outputDevice, false /*hwAvSync*/, &sourceConfig, &sinkConfig);
if (res == NO_ERROR) {
// Found a matching profile for encoded audio. Re-create PatchBuilder with this config.
return (PatchBuilder()).addSource(sourceConfig).addSink(sinkConfig);
}
ALOGV("%s() no matching profile found. Fall through to default PCM patch"
" supporting PCM format conversion.", __func__);
return patchBuilder;
}
status_t AudioPolicyManager::setMsdPatch(const sp<DeviceDescriptor> &outputDevice) {
sp<DeviceDescriptor> device = outputDevice;
if (device == nullptr) {
// Use media strategy for unspecified output device. This should only
// occur on checkForDeviceAndOutputChanges(). Device connection events may
// therefore invalidate explicit routing requests.
DeviceVector devices = mEngine->getOutputDevicesForAttributes(
attributes_initializer(AUDIO_USAGE_MEDIA), nullptr, false /*fromCache*/);
LOG_ALWAYS_FATAL_IF(devices.isEmpty(), "no outpudevice to set Msd Patch");
device = devices.itemAt(0);
}
ALOGV("%s() for device %s", __func__, device->toString().c_str());
PatchBuilder patchBuilder = buildMsdPatch(device);
const struct audio_patch* patch = patchBuilder.patch();
const AudioPatchCollection msdPatches = getMsdPatches();
if (!msdPatches.isEmpty()) {
LOG_ALWAYS_FATAL_IF(msdPatches.size() > 1,
"The current MSD prototype only supports one output patch");
sp<AudioPatch> currentPatch = msdPatches.valueAt(0);
if (audio_patches_are_equal(¤tPatch->mPatch, patch)) {
return NO_ERROR;
}
releaseAudioPatch(currentPatch->mHandle, mUidCached);
}
status_t status = installPatch(__func__, -1 /*index*/, nullptr /*patchHandle*/,
patch, 0 /*delayMs*/, mUidCached, nullptr /*patchDescPtr*/);
ALOGE_IF(status != NO_ERROR, "%s() error %d creating MSD audio patch", __func__, status);
ALOGI_IF(status == NO_ERROR, "%s() Patch created from MSD_IN to "
"device:%s (format:%#x channels:%#x samplerate:%d)", __func__,
device->toString().c_str(), patch->sources[0].format,
patch->sources[0].channel_mask, patch->sources[0].sample_rate);
return status;
}
audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
audio_output_flags_t flags,
audio_format_t format,
audio_channel_mask_t channelMask,
uint32_t samplingRate)
{
LOG_ALWAYS_FATAL_IF(!(format == AUDIO_FORMAT_INVALID || audio_is_linear_pcm(format)),
"%s called with format %#x", __func__, format);
// Flags disqualifying an output: the match must happen before calling selectOutput()
static const audio_output_flags_t kExcludedFlags = (audio_output_flags_t)
(AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT);
// Flags expressing a functional request: must be honored in priority over
// other criteria
static const audio_output_flags_t kFunctionalFlags = (audio_output_flags_t)
(AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_INCALL_MUSIC |
AUDIO_OUTPUT_FLAG_TTS | AUDIO_OUTPUT_FLAG_DIRECT_PCM);
// Flags expressing a performance request: have lower priority than serving
// requested sampling rate or channel mask
static const audio_output_flags_t kPerformanceFlags = (audio_output_flags_t)
(AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER |
AUDIO_OUTPUT_FLAG_RAW | AUDIO_OUTPUT_FLAG_SYNC);
const audio_output_flags_t functionalFlags =
(audio_output_flags_t)(flags & kFunctionalFlags);
const audio_output_flags_t performanceFlags =
(audio_output_flags_t)(flags & kPerformanceFlags);
audio_io_handle_t bestOutput = (outputs.size() == 0) ? AUDIO_IO_HANDLE_NONE : outputs[0];
// select one output among several that provide a path to a particular device or set of
// devices (the list was previously build by getOutputsForDevices()).
// The priority is as follows:
// 1: the output supporting haptic playback when requesting haptic playback
// 2: the output with the highest number of requested functional flags
// 3: the output supporting the exact channel mask
// 4: the output with a higher channel count than requested
// 5: the output with a higher sampling rate than requested
// 6: the output with the highest number of requested performance flags
// 7: the output with the bit depth the closest to the requested one
// 8: the primary output
// 9: the first output in the list
// matching criteria values in priority order for best matching output so far
std::vector<uint32_t> bestMatchCriteria(8, 0);
const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
const uint32_t hapticChannelCount = audio_channel_count_from_out_mask(
channelMask & AUDIO_CHANNEL_HAPTIC_ALL);
for (audio_io_handle_t output : outputs) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
// matching criteria values in priority order for current output
std::vector<uint32_t> currentMatchCriteria(8, 0);
if (outputDesc->isDuplicated()) {
continue;
}
if ((kExcludedFlags & outputDesc->mFlags) != 0) {
continue;
}
// If haptic channel is specified, use the haptic output if present.
// When using haptic output, same audio format and sample rate are required.
const uint32_t outputHapticChannelCount = audio_channel_count_from_out_mask(
outputDesc->mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
if ((hapticChannelCount == 0) != (outputHapticChannelCount == 0)) {
continue;
}
if (outputHapticChannelCount >= hapticChannelCount
&& format == outputDesc->mFormat
&& samplingRate == outputDesc->mSamplingRate) {
currentMatchCriteria[0] = outputHapticChannelCount;
}
// functional flags match
currentMatchCriteria[1] = popcount(outputDesc->mFlags & functionalFlags);
// channel mask and channel count match
uint32_t outputChannelCount = audio_channel_count_from_out_mask(outputDesc->mChannelMask);
if (channelMask != AUDIO_CHANNEL_NONE && channelCount > 2 &&
channelCount <= outputChannelCount) {
if ((audio_channel_mask_get_representation(channelMask) ==
audio_channel_mask_get_representation(outputDesc->mChannelMask)) &&
((channelMask & outputDesc->mChannelMask) == channelMask)) {
currentMatchCriteria[2] = outputChannelCount;
}
currentMatchCriteria[3] = outputChannelCount;
}
// sampling rate match
if (samplingRate > SAMPLE_RATE_HZ_DEFAULT &&
samplingRate <= outputDesc->mSamplingRate) {
currentMatchCriteria[4] = outputDesc->mSamplingRate;
}
// performance flags match
currentMatchCriteria[5] = popcount(outputDesc->mFlags & performanceFlags);
// format match
if (format != AUDIO_FORMAT_INVALID) {
currentMatchCriteria[6] =
AudioPort::kFormatDistanceMax -
AudioPort::formatDistance(format, outputDesc->mFormat);
}
// primary output match
currentMatchCriteria[7] = outputDesc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY;
// compare match criteria by priority then value
if (std::lexicographical_compare(bestMatchCriteria.begin(), bestMatchCriteria.end(),
currentMatchCriteria.begin(), currentMatchCriteria.end())) {
bestMatchCriteria = currentMatchCriteria;
bestOutput = output;
std::stringstream result;
std::copy(bestMatchCriteria.begin(), bestMatchCriteria.end(),
std::ostream_iterator<int>(result, " "));
ALOGV("%s new bestOutput %d criteria %s",
__func__, bestOutput, result.str().c_str());
}
}
return bestOutput;
}
status_t AudioPolicyManager::startOutput(audio_port_handle_t portId)
{
ALOGV("%s portId %d", __FUNCTION__, portId);
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputForClient(portId);
if (outputDesc == 0) {
ALOGW("startOutput() no output for client %d", portId);
return BAD_VALUE;
}
sp<TrackClientDescriptor> client = outputDesc->getClient(portId);
ALOGV("startOutput() output %d, stream %d, session %d",
outputDesc->mIoHandle, client->stream(), client->session());
status_t status = outputDesc->start();
if (status != NO_ERROR) {
return status;
}
uint32_t delayMs;
status = startSource(outputDesc, client, &delayMs);
if (status != NO_ERROR) {
outputDesc->stop();
return status;
}
if (delayMs != 0) {
usleep(delayMs * 1000);
}
return status;
}
status_t AudioPolicyManager::startSource(const sp<SwAudioOutputDescriptor>& outputDesc,
const sp<TrackClientDescriptor>& client,
uint32_t *delayMs)
{
// cannot start playback of STREAM_TTS if any other output is being used
uint32_t beaconMuteLatency = 0;
*delayMs = 0;
audio_stream_type_t stream = client->stream();
auto clientVolSrc = client->volumeSource();
auto clientStrategy = client->strategy();
auto clientAttr = client->attributes();
if (stream == AUDIO_STREAM_TTS) {
ALOGV("\t found BEACON stream");
if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive(
toVolumeSource(AUDIO_STREAM_TTS) /*sourceToIgnore*/)) {
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);
DeviceVector devices;
sp<AudioPolicyMix> policyMix = outputDesc->mPolicyMix.promote();
const char *address = NULL;
if (policyMix != NULL) {
audio_devices_t newDeviceType;
address = policyMix->mDeviceAddress.string();
if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
newDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
} else {
newDeviceType = policyMix->mDeviceType;
}
sp device = mAvailableOutputDevices.getDevice(newDeviceType, String8(address),
AUDIO_FORMAT_DEFAULT);
ALOG_ASSERT(device, "%s: no device found t=%u, a=%s", __func__, newDeviceType, address);
devices.add(device);
}
// requiresMuteCheck is false when we can bypass mute strategy.
// It covers a common case when there is no materially active audio
// and muting would result in unnecessary delay and dropped audio.
const uint32_t outputLatencyMs = outputDesc->latency();
bool requiresMuteCheck = outputDesc->isActive(outputLatencyMs * 2); // account for drain
// 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->setClientActive(client, true);
if (client->hasPreferredDevice(true)) {
if (outputDesc->clientsList(true /*activeOnly*/).size() == 1 &&
client->isPreferredDeviceForExclusiveUse()) {
// Preferred device may be exclusive, use only if no other active clients on this output
devices = DeviceVector(
mAvailableOutputDevices.getDeviceFromId(client->preferredDeviceId()));
} else {
devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
}
if (devices != outputDesc->devices()) {
checkStrategyRoute(clientStrategy, outputDesc->mIoHandle);
}
}
if (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_MEDIA))) {
selectOutputForMusicEffects();
}
if (outputDesc->getActivityCount(clientVolSrc) == 1 || !devices.isEmpty()) {
// starting an output being rerouted?
if (devices.isEmpty()) {
devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
}
bool shouldWait =
(followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM)) ||
followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_NOTIFICATION)) ||
(beaconMuteLatency > 0));
uint32_t waitMs = beaconMuteLatency;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc) {
// An output has a shared device if
// - managed by the same hw module
// - supports the currently selected device
const bool sharedDevice = outputDesc->sharesHwModuleWith(desc)
&& (!desc->filterSupportedDevices(devices).isEmpty());
// force a device change if any other output is:
// - managed by the same hw module
// - supports currently selected device
// - has a current device selection that differs from 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 output.
if (sharedDevice &&
desc->devices() != devices &&
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
const uint32_t latencyMs = desc->latency();
const bool isActive = desc->isActive(latencyMs * 2); // account for drain
if (shouldWait && isActive && (waitMs < latencyMs)) {
waitMs = latencyMs;
}
// Require mute check if another output is on a shared device
// and currently active to have proper drain and avoid pops.
// Note restoring AudioTracks onto this output needs to invoke
// a volume ramp if there is no mute.
requiresMuteCheck |= sharedDevice && isActive;
}
}
const uint32_t muteWaitMs =
setOutputDevices(outputDesc, devices, force, 0, NULL, requiresMuteCheck);
// apply volume rules for current stream and device if necessary
auto &curves = getVolumeCurves(client->attributes());
checkAndSetVolume(curves, client->volumeSource(),
curves.getVolumeIndex(outputDesc->devices().types()),
outputDesc,
outputDesc->devices().types());
// 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 (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM))) {
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
if (waitMs > muteWaitMs) {
*delayMs = waitMs - muteWaitMs;
}
// FIXME: A device change (muteWaitMs > 0) likely introduces a volume change.
// A volume change enacted by APM with 0 delay is not synchronous, as it goes
// via AudioCommandThread to AudioFlinger. Hence it is possible that the volume
// change occurs after the MixerThread starts and causes a stream volume
// glitch.
//
// We do not introduce additional delay here.
}
if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE &&
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), true, outputDesc);
}
// 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(devices.types()) && policyMix != NULL &&
policyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address,
"remote-submix",
AUDIO_FORMAT_DEFAULT);
}
return NO_ERROR;
}
status_t AudioPolicyManager::stopOutput(audio_port_handle_t portId)
{
ALOGV("%s portId %d", __FUNCTION__, portId);
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputForClient(portId);
if (outputDesc == 0) {
ALOGW("stopOutput() no output for client %d", portId);
return BAD_VALUE;
}
sp<TrackClientDescriptor> client = outputDesc->getClient(portId);
ALOGV("stopOutput() output %d, stream %d, session %d",
outputDesc->mIoHandle, client->stream(), client->session());
status_t status = stopSource(outputDesc, client);
if (status == NO_ERROR ) {
outputDesc->stop();
}
return status;
}
status_t AudioPolicyManager::stopSource(const sp<SwAudioOutputDescriptor>& outputDesc,
const sp<TrackClientDescriptor>& client)
{
// always handle stream stop, check which stream type is stopping
audio_stream_type_t stream = client->stream();
auto clientVolSrc = client->volumeSource();
handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
if (outputDesc->getActivityCount(clientVolSrc) > 0) {
if (outputDesc->getActivityCount(clientVolSrc) == 1) {
// Automatically disable the remote submix input when output is stopped on a
// re routing mix of type MIX_TYPE_RECORDERS
sp<AudioPolicyMix> policyMix = outputDesc->mPolicyMix.promote();
if (audio_is_remote_submix_device(outputDesc->devices().types()) &&
policyMix != NULL &&
policyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
policyMix->mDeviceAddress,
"remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
bool forceDeviceUpdate = false;
if (client->hasPreferredDevice(true)) {
checkStrategyRoute(client->strategy(), AUDIO_IO_HANDLE_NONE);
forceDeviceUpdate = true;
}
// decrement usage count of this stream on the output
outputDesc->setClientActive(client, false);
// store time at which the stream was stopped - see isStreamActive()
if (outputDesc->getActivityCount(clientVolSrc) == 0 || forceDeviceUpdate) {
outputDesc->setStopTime(client, systemTime());
DeviceVector newDevices = getNewOutputDevices(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 ...)
setOutputDevices(outputDesc, newDevices, 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<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc &&
desc->isActive() &&
outputDesc->sharesHwModuleWith(desc) &&
(newDevices != desc->devices())) {
DeviceVector newDevices2 = getNewOutputDevices(desc, false /*fromCache*/);
bool force = desc->devices() != newDevices2;
setOutputDevices(desc, newDevices2, force, delayMs);
// re-apply device specific volume if not done by setOutputDevice()
if (!force) {
applyStreamVolumes(desc, newDevices2.types(), delayMs);
}
}
}
// update the outputs if stopping one with a stream that can affect notification routing
handleNotificationRoutingForStream(stream);
}
if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE &&
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), false, outputDesc);
}
if (followsSameRouting(client->attributes(), attributes_initializer(AUDIO_USAGE_MEDIA))) {
selectOutputForMusicEffects();
}
return NO_ERROR;
} else {
ALOGW("stopOutput() refcount is already 0");
return INVALID_OPERATION;
}
}
void AudioPolicyManager::releaseOutput(audio_port_handle_t portId)
{
ALOGV("%s portId %d", __FUNCTION__, portId);
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputForClient(portId);
if (outputDesc == 0) {
// If an output descriptor is closed due to a device routing change,
// then there are race conditions with releaseOutput from tracks
// that may be destroyed (with no PlaybackThread) or a PlaybackThread
// destroyed shortly thereafter.
//
// Here we just log a warning, instead of a fatal error.
ALOGW("releaseOutput() no output for client %d", portId);
return;
}
ALOGV("releaseOutput() %d", outputDesc->mIoHandle);
if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (outputDesc->mDirectOpenCount <= 0) {
ALOGW("releaseOutput() invalid open count %d for output %d",
outputDesc->mDirectOpenCount, outputDesc->mIoHandle);
return;
}
if (--outputDesc->mDirectOpenCount == 0) {
closeOutput(outputDesc->mIoHandle);
mpClientInterface->onAudioPortListUpdate();
}
}
// stopOutput() needs to be successfully called before releaseOutput()
// otherwise there may be inaccurate stream reference counts.
// This is checked in outputDesc->removeClient below.
outputDesc->removeClient(portId);
}
status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_unique_id_t riid,
audio_session_t session,
uid_t uid,
const audio_config_base_t *config,
audio_input_flags_t flags,
audio_port_handle_t *selectedDeviceId,
input_type_t *inputType,
audio_port_handle_t *portId)
{
ALOGV("%s() source %d, sampling rate %d, format %#x, channel mask %#x, session %d, "
"flags %#x attributes=%s", __func__, attr->source, config->sample_rate,
config->format, config->channel_mask, session, flags, toString(*attr).c_str());
status_t status = NO_ERROR;
audio_source_t halInputSource;
audio_attributes_t attributes = *attr;
sp<AudioPolicyMix> policyMix;
sp<DeviceDescriptor> device;
sp<AudioInputDescriptor> inputDesc;
sp<RecordClientDescriptor> clientDesc;
audio_port_handle_t requestedDeviceId = *selectedDeviceId;
bool isSoundTrigger;
// The supplied portId must be AUDIO_PORT_HANDLE_NONE
if (*portId != AUDIO_PORT_HANDLE_NONE) {
return INVALID_OPERATION;
}
if (attr->source == AUDIO_SOURCE_DEFAULT) {
attributes.source = AUDIO_SOURCE_MIC;
}
// Explicit routing?
sp<DeviceDescriptor> explicitRoutingDevice =
mAvailableInputDevices.getDeviceFromId(*selectedDeviceId);
// special case for mmap capture: if an input IO handle is specified, we reuse this input if
// possible
if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) == AUDIO_INPUT_FLAG_MMAP_NOIRQ &&
*input != AUDIO_IO_HANDLE_NONE) {
ssize_t index = mInputs.indexOfKey(*input);
if (index < 0) {
ALOGW("getInputForAttr() unknown MMAP input %d", *input);
status = BAD_VALUE;
goto error;
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
RecordClientVector clients = inputDesc->getClientsForSession(session);
if (clients.size() == 0) {
ALOGW("getInputForAttr() unknown session %d on input %d", session, *input);
status = BAD_VALUE;
goto error;
}
// For MMAP mode, the first call to getInputForAttr() is made on behalf of audioflinger.
// The second call is for the first active client and sets the UID. Any further call
// corresponds to a new client and is only permitted from the same UID.
// If the first UID is silenced, allow a new UID connection and replace with new UID
if (clients.size() > 1) {
for (const auto& client : clients) {
// The client map is ordered by key values (portId) and portIds are allocated
// incrementaly. So the first client in this list is the one opened by audio flinger
// when the mmap stream is created and should be ignored as it does not correspond
// to an actual client
if (client == *clients.cbegin()) {
continue;
}
if (uid != client->uid() && !client->isSilenced()) {
ALOGW("getInputForAttr() bad uid %d for client %d uid %d",
uid, client->portId(), client->uid());
status = INVALID_OPERATION;
goto error;
}
}
}
*inputType = API_INPUT_LEGACY;
device = inputDesc->getDevice();
ALOGI("%s reusing MMAP input %d for session %d", __FUNCTION__, *input, session);
goto exit;
}
*input = AUDIO_IO_HANDLE_NONE;
*inputType = API_INPUT_INVALID;
halInputSource = attributes.source;
if (attributes.source == AUDIO_SOURCE_REMOTE_SUBMIX &&
strncmp(attributes.tags, "addr=", strlen("addr=")) == 0) {
status = mPolicyMixes.getInputMixForAttr(attributes, &policyMix);
if (status != NO_ERROR) {
ALOGW("%s could not find input mix for attr %s",
__func__, toString(attributes).c_str());
goto error;
}
device = mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
String8(attr->tags + strlen("addr=")),
AUDIO_FORMAT_DEFAULT);
if (device == nullptr) {
ALOGW("%s could not find in Remote Submix device for source %d, tags %s",
__func__, attributes.source, attributes.tags);
status = BAD_VALUE;
goto error;
}
if (is_mix_loopback_render(policyMix->mRouteFlags)) {
*inputType = API_INPUT_MIX_PUBLIC_CAPTURE_PLAYBACK;
} else {
*inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
}
} else {
if (explicitRoutingDevice != nullptr) {
device = explicitRoutingDevice;
} else {
// Prevent from storing invalid requested device id in clients
requestedDeviceId = AUDIO_PORT_HANDLE_NONE;
device = mEngine->getInputDeviceForAttributes(attributes, &policyMix);
}
if (device == nullptr) {
ALOGW("getInputForAttr() could not find device for source %d", attributes.source);
status = BAD_VALUE;
goto error;
}
if (policyMix) {
ALOG_ASSERT(policyMix->mMixType == MIX_TYPE_RECORDERS, "Invalid Mix Type");
// 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 if (audio_is_remote_submix_device(device->type())) {
*inputType = API_INPUT_MIX_CAPTURE;
} else if (device->type() == AUDIO_DEVICE_IN_TELEPHONY_RX) {
*inputType = API_INPUT_TELEPHONY_RX;
} else {
*inputType = API_INPUT_LEGACY;
}
}
*input = getInputForDevice(device, session, attributes, config, flags, policyMix);
if (*input == AUDIO_IO_HANDLE_NONE) {
status = INVALID_OPERATION;
goto error;
}
exit:
*selectedDeviceId = mAvailableInputDevices.contains(device) ?
device->getId() : AUDIO_PORT_HANDLE_NONE;
isSoundTrigger = attributes.source == AUDIO_SOURCE_HOTWORD &&
mSoundTriggerSessions.indexOfKey(session) >= 0;
*portId = AudioPort::getNextUniqueId();
clientDesc = new RecordClientDescriptor(*portId, riid, uid, session, attributes, *config,
requestedDeviceId, attributes.source, flags,
isSoundTrigger);
inputDesc = mInputs.valueFor(*input);
inputDesc->addClient(clientDesc);
ALOGV("getInputForAttr() returns input %d type %d selectedDeviceId %d for port ID %d",
*input, *inputType, *selectedDeviceId, *portId);
return NO_ERROR;
error:
return status;
}
audio_io_handle_t AudioPolicyManager::getInputForDevice(const sp<DeviceDescriptor> &device,
audio_session_t session,
const audio_attributes_t &attributes,
const audio_config_base_t *config,
audio_input_flags_t flags,
const sp<AudioPolicyMix> &policyMix)
{
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
audio_source_t halInputSource = attributes.source;
bool isSoundTrigger = false;
if (attributes.source == 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;
}
} else if (attributes.source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
audio_is_linear_pcm(config->format)) {
flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_VOIP_TX);
}
// find a compatible input profile (not necessarily identical in parameters)
sp<IOProfile> profile;
// sampling rate and flags may be updated by getInputProfile
uint32_t profileSamplingRate = (config->sample_rate == 0) ?
SAMPLE_RATE_HZ_DEFAULT : config->sample_rate;
audio_format_t profileFormat;
audio_channel_mask_t profileChannelMask = config->channel_mask;
audio_input_flags_t profileFlags = flags;
for (;;) {
profileFormat = config->format; // reset each time through loop, in case it is updated
profile = getInputProfile(device, 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("%s could not find profile for device %s, sampling rate %u, format %#x, "
"channel mask 0x%X, flags %#x", __func__, device->toString().c_str(),
config->sample_rate, config->format, config->channel_mask, flags);
return input;
}
}
// Pick input sampling rate if not specified by client
uint32_t samplingRate = config->sample_rate;
if (samplingRate == 0) {
samplingRate = profileSamplingRate;
}
if (profile->getModuleHandle() == 0) {
ALOGE("getInputForAttr(): HW module %s not opened", profile->getModuleName());
return input;
}
if (!profile->canOpenNewIo()) {
for (size_t i = 0; i < mInputs.size(); ) {
sp <AudioInputDescriptor> desc = mInputs.valueAt(i);
if (desc->mProfile != profile) {
i++;
continue;
}
// if sound trigger, reuse input if used by other sound trigger on same session
// else
// reuse input if active client app is not in IDLE state
//
RecordClientVector clients = desc->clientsList();
bool doClose = false;
for (const auto& client : clients) {
if (isSoundTrigger != client->isSoundTrigger()) {
continue;
}
if (client->isSoundTrigger()) {
if (session == client->session()) {
return desc->mIoHandle;
}
continue;
}
if (client->active() && client->appState() != APP_STATE_IDLE) {
return desc->mIoHandle;
}
doClose = true;
}
if (doClose) {
closeInput(desc->mIoHandle);
} else {
i++;
}
}
}
sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile, mpClientInterface);
audio_config_t lConfig = AUDIO_CONFIG_INITIALIZER;
lConfig.sample_rate = profileSamplingRate;
lConfig.channel_mask = profileChannelMask;
lConfig.format = profileFormat;
status_t status = inputDesc->open(&lConfig, device, halInputSource, profileFlags, &input);
// only accept input with the exact requested set of parameters
if (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE ||
(profileSamplingRate != lConfig.sample_rate) ||
!audio_formats_match(profileFormat, lConfig.format) ||
(profileChannelMask != lConfig.channel_mask)) {
ALOGW("getInputForAttr() failed opening input: sampling rate %d"
", format %#x, channel mask %#x",
profileSamplingRate, profileFormat, profileChannelMask);
if (input != AUDIO_IO_HANDLE_NONE) {
inputDesc->close();
}
return AUDIO_IO_HANDLE_NONE;
}
inputDesc->mPolicyMix = policyMix;
addInput(input, inputDesc);
mpClientInterface->onAudioPortListUpdate();
return input;
}
status_t AudioPolicyManager::startInput(audio_port_handle_t portId)
{
ALOGV("%s portId %d", __FUNCTION__, portId);
sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
if (inputDesc == 0) {
ALOGW("%s no input for client %d", __FUNCTION__, portId);
return BAD_VALUE;
}
audio_io_handle_t input = inputDesc->mIoHandle;
sp<RecordClientDescriptor> client = inputDesc->getClient(portId);
if (client->active()) {
ALOGW("%s input %d client %d already started", __FUNCTION__, input, client->portId());
return INVALID_OPERATION;
}
audio_session_t session = client->session();
ALOGV("%s input:%d, session:%d)", __FUNCTION__, input, session);
Vector<sp<AudioInputDescriptor>> activeInputs = mInputs.getActiveInputs();
status_t status = inputDesc->start();
if (status != NO_ERROR) {
return status;
}
// increment activity count before calling getNewInputDevice() below as only active sessions
// are considered for device selection
inputDesc->setClientActive(client, true);
// indicate active capture to sound trigger service if starting capture from a mic on
// primary HW module
sp<DeviceDescriptor> device = getNewInputDevice(inputDesc);
setInputDevice(input, device, true /* force */);
if (inputDesc->activeCount() == 1) {
sp<AudioPolicyMix> policyMix = inputDesc->mPolicyMix.promote();
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((policyMix != NULL)
&& ((policyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
mpClientInterface->onDynamicPolicyMixStateUpdate(policyMix->mDeviceAddress,
MIX_STATE_MIXING);
}
DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
if (primaryInputDevices.contains(device) &&
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1) {
SoundTrigger::setCaptureState(true);
}
// 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->getDeviceType())) {
String8 address = String8("");
if (policyMix == NULL) {
address = String8("0");
} else if (policyMix->mMixType == MIX_TYPE_PLAYERS) {
address = policyMix->mDeviceAddress;
}
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address, "remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
}
ALOGV("%s input %d source = %d exit", __FUNCTION__, input, client->source());
return NO_ERROR;
}
status_t AudioPolicyManager::stopInput(audio_port_handle_t portId)
{
ALOGV("%s portId %d", __FUNCTION__, portId);
sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
if (inputDesc == 0) {
ALOGW("%s no input for client %d", __FUNCTION__, portId);
return BAD_VALUE;
}
audio_io_handle_t input = inputDesc->mIoHandle;
sp<RecordClientDescriptor> client = inputDesc->getClient(portId);
if (!client->active()) {
ALOGW("%s input %d client %d already stopped", __FUNCTION__, input, client->portId());
return INVALID_OPERATION;
}
inputDesc->setClientActive(client, false);
inputDesc->stop();
if (inputDesc->isActive()) {
setInputDevice(input, getNewInputDevice(inputDesc), false /* force */);
} else {
sp<AudioPolicyMix> policyMix = inputDesc->mPolicyMix.promote();
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((policyMix != NULL)
&& ((policyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
mpClientInterface->onDynamicPolicyMixStateUpdate(policyMix->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->getDeviceType())) {
String8 address = String8("");
if (policyMix == NULL) {
address = String8("0");
} else if (policyMix->mMixType == MIX_TYPE_PLAYERS) {
address = policyMix->mDeviceAddress;
}
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address, "remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
resetInputDevice(input);
// indicate inactive capture to sound trigger service if stopping capture from a mic on
// primary HW module
DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
if (primaryInputDevices.contains(inputDesc->getDevice()) &&
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
SoundTrigger::setCaptureState(false);
}
inputDesc->clearPreemptedSessions();
}
return NO_ERROR;
}
void AudioPolicyManager::releaseInput(audio_port_handle_t portId)
{
ALOGV("%s portId %d", __FUNCTION__, portId);
sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
if (inputDesc == 0) {
ALOGW("%s no input for client %d", __FUNCTION__, portId);
return;
}
sp<RecordClientDescriptor> client = inputDesc->getClient(portId);
audio_io_handle_t input = inputDesc->mIoHandle;
ALOGV("%s %d", __FUNCTION__, input);
inputDesc->removeClient(portId);
if (inputDesc->getClientCount() > 0) {
ALOGV("%s(%d) %zu clients remaining", __func__, portId, inputDesc->getClientCount());
return;
}
closeInput(input);
mpClientInterface->onAudioPortListUpdate();
ALOGV("%s exit", __FUNCTION__);
}
void AudioPolicyManager::closeActiveClients(const sp<AudioInputDescriptor>& input)
{
RecordClientVector clients = input->clientsList(true);
for (const auto& client : clients) {
closeClient(client->portId());
}
}
void AudioPolicyManager::closeClient(audio_port_handle_t portId)
{
stopInput(portId);
releaseInput(portId);
}
void AudioPolicyManager::checkCloseInputs() {
// After connecting or disconnecting an input device, close input if:
// - it has no client (was just opened to check profile) OR
// - none of its supported devices are connected anymore OR
// - one of its clients cannot be routed to one of its supported
// devices anymore. Otherwise update device selection
std::vector<audio_io_handle_t> inputsToClose;
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> input = mInputs.valueAt(i);
if (input->clientsList().size() == 0
|| !mAvailableInputDevices.containsAtLeastOne(input->supportedDevices())
|| (input->getAudioPort()->getFlags() & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
inputsToClose.push_back(mInputs.keyAt(i));
} else {
bool close = false;
for (const auto& client : input->clientsList()) {
sp<DeviceDescriptor> device =
mEngine->getInputDeviceForAttributes(client->attributes());
if (!input->supportedDevices().contains(device)) {
close = true;
break;
}
}
if (close) {
inputsToClose.push_back(mInputs.keyAt(i));
} else {
setInputDevice(input->mIoHandle, getNewInputDevice(input));
}
}
}
for (const audio_io_handle_t handle : inputsToClose) {
ALOGV("%s closing input %d", __func__, handle);
closeInput(handle);
}
}
void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax)
{
ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
if (indexMin < 0 || indexMax < 0) {
ALOGE("%s for stream %d: invalid min %d or max %d", __func__, stream , indexMin, indexMax);
return;
}
getVolumeCurves(stream).initVolume(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;
}
getVolumeCurves((audio_stream_type_t)curStream).initVolume(indexMin, indexMax);
}
}
status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
int index,
audio_devices_t device)
{
auto attributes = mEngine->getAttributesForStreamType(stream);
ALOGV("%s: stream %s attributes=%s", __func__,
toString(stream).c_str(), toString(attributes).c_str());
return setVolumeIndexForAttributes(attributes, index, device);
}
status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
int *index,
audio_devices_t device)
{
// if device is AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME, return volume for device selected for this
// stream by the engine.
if (device == AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
device = mEngine->getOutputDevicesForStream(stream, true /*fromCache*/).types();
}
return getVolumeIndex(getVolumeCurves(stream), *index, device);
}
status_t AudioPolicyManager::setVolumeIndexForAttributes(const audio_attributes_t &attributes,
int index,
audio_devices_t device)
{
// Get Volume group matching the Audio Attributes
auto group = mEngine->getVolumeGroupForAttributes(attributes);
if (group == VOLUME_GROUP_NONE) {
ALOGD("%s: no group matching with %s", __FUNCTION__, toString(attributes).c_str());
return BAD_VALUE;
}
ALOGV("%s: group %d matching with %s", __FUNCTION__, group, toString(attributes).c_str());
status_t status = NO_ERROR;
IVolumeCurves &curves = getVolumeCurves(attributes);
VolumeSource vs = toVolumeSource(group);
product_strategy_t strategy = mEngine->getProductStrategyForAttributes(attributes);
status = setVolumeCurveIndex(index, device, curves);
if (status != NO_ERROR) {
ALOGE("%s failed to set curve index for group %d device 0x%X", __func__, group, device);
return status;
}
audio_devices_t curSrcDevice;
auto curCurvAttrs = curves.getAttributes();
if (!curCurvAttrs.empty() && curCurvAttrs.front() != defaultAttr) {
auto attr = curCurvAttrs.front();
curSrcDevice = mEngine->getOutputDevicesForAttributes(attr, nullptr, false).types();
} else if (!curves.getStreamTypes().empty()) {
auto stream = curves.getStreamTypes().front();
curSrcDevice = mEngine->getOutputDevicesForStream(stream, false).types();
} else {
ALOGE("%s: Invalid src %d: no valid attributes nor stream",__func__, vs);
return BAD_VALUE;
}
curSrcDevice = Volume::getDeviceForVolume(curSrcDevice);
// 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 engine for 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 engine for this stream
// - For default requested device (AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME), apply volume only if
// no specific device volume value exists for currently selected device.
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
audio_devices_t curDevice = desc->devices().types();
if (curDevice & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
curDevice |= AUDIO_DEVICE_OUT_SPEAKER;
curDevice &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
}
// Inter / intra volume group priority management: Loop on strategies arranged by priority
// If a higher priority strategy is active, and the output is routed to a device with a
// HW Gain management, do not change the volume
bool applyVolume = false;
if (desc->useHwGain()) {
if (!(desc->isActive(toVolumeSource(group)) || isInCall())) {
continue;
}
for (const auto &productStrategy : mEngine->getOrderedProductStrategies()) {
auto activeClients = desc->clientsList(true /*activeOnly*/, productStrategy,
false /*preferredDevice*/);
if (activeClients.empty()) {
continue;
}
bool isPreempted = false;
bool isHigherPriority = productStrategy < strategy;
for (const auto &client : activeClients) {
if (isHigherPriority && (client->volumeSource() != vs)) {
ALOGV("%s: Strategy=%d (\nrequester:\n"
" group %d, volumeGroup=%d attributes=%s)\n"
" higher priority source active:\n"
" volumeGroup=%d attributes=%s) \n"
" on output %zu, bailing out", __func__, productStrategy,
group, group, toString(attributes).c_str(),
client->volumeSource(), toString(client->attributes()).c_str(), i);
applyVolume = false;
isPreempted = true;
break;
}
// However, continue for loop to ensure no higher prio clients running on output
if (client->volumeSource() == vs) {
applyVolume = true;
}
}
if (isPreempted || applyVolume) {
break;
}
}
if (!applyVolume) {
continue; // next output
}
status_t volStatus = checkAndSetVolume(curves, vs, index, desc, curDevice,
(vs == toVolumeSource(AUDIO_STREAM_SYSTEM)?
TOUCH_SOUND_FIXED_DELAY_MS : 0));
if (volStatus != NO_ERROR) {
status = volStatus;
}
continue;
}
if (!(desc->isActive(vs) || isInCall())) {
continue;
}
if ((device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) && ((curDevice & device) == 0)) {
continue;
}
if (device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
curSrcDevice |= device;
applyVolume = (Volume::getDeviceForVolume(curDevice) & curSrcDevice) != 0;
} else {
applyVolume = !curves.hasVolumeIndexForDevice(curSrcDevice);
}
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(
curves, vs, index, desc, curDevice,
((vs == toVolumeSource(AUDIO_STREAM_SYSTEM))?
TOUCH_SOUND_FIXED_DELAY_MS : 0));
if (volStatus != NO_ERROR) {
status = volStatus;
}
}
}
mpClientInterface->onAudioVolumeGroupChanged(group, 0 /*flags*/);
return status;
}
status_t AudioPolicyManager::setVolumeCurveIndex(int index,
audio_devices_t device,
IVolumeCurves &volumeCurves)
{
// VOICE_CALL stream has minVolumeIndex > 0 but can be muted directly by an
// app that has MODIFY_PHONE_STATE permission.
bool hasVoice = hasVoiceStream(volumeCurves.getStreamTypes());
if (((index < volumeCurves.getVolumeIndexMin()) && !(hasVoice && index == 0)) ||
(index > volumeCurves.getVolumeIndexMax())) {
ALOGD("%s: wrong index %d min=%d max=%d", __FUNCTION__, index,
volumeCurves.getVolumeIndexMin(), volumeCurves.getVolumeIndexMax());
return BAD_VALUE;
}
if (!audio_is_output_device(device)) {
return BAD_VALUE;
}
// Force max volume if stream cannot be muted
if (!volumeCurves.canBeMuted()) index = volumeCurves.getVolumeIndexMax();
ALOGV("%s device %08x, index %d", __FUNCTION__ , device, index);
volumeCurves.addCurrentVolumeIndex(device, index);
return NO_ERROR;
}
status_t AudioPolicyManager::getVolumeIndexForAttributes(const audio_attributes_t &attr,
int &index,
audio_devices_t device)
{
// if device is AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME, return volume for device selected for this
// stream by the engine.
if (device == AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
device = mEngine->getOutputDevicesForAttributes(attr, nullptr, true /*fromCache*/).types();
}
return getVolumeIndex(getVolumeCurves(attr), index, device);
}
status_t AudioPolicyManager::getVolumeIndex(const IVolumeCurves &curves,
int &index,
audio_devices_t device) const
{
if (!audio_is_output_device(device)) {
return BAD_VALUE;
}
device = Volume::getDeviceForVolume(device);
index = curves.getVolumeIndex(device);
ALOGV("%s: device %08x index %d", __FUNCTION__, device, index);
return NO_ERROR;
}
status_t AudioPolicyManager::getMinVolumeIndexForAttributes(const audio_attributes_t &attr,
int &index)
{
index = getVolumeCurves(attr).getVolumeIndexMin();
return NO_ERROR;
}
status_t AudioPolicyManager::getMaxVolumeIndexForAttributes(const audio_attributes_t &attr,
int &index)
{
index = getVolumeCurves(attr).getVolumeIndexMax();
return NO_ERROR;
}
audio_io_handle_t AudioPolicyManager::selectOutputForMusicEffects()
{
// 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 primary output
// 4: the first output in the list
DeviceVector devices = mEngine->getOutputDevicesForAttributes(
attributes_initializer(AUDIO_USAGE_MEDIA), nullptr, false /*fromCache*/);
SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
if (outputs.size() == 0) {
return AUDIO_IO_HANDLE_NONE;
}
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
bool activeOnly = true;
while (output == AUDIO_IO_HANDLE_NONE) {
audio_io_handle_t outputOffloaded = AUDIO_IO_HANDLE_NONE;
audio_io_handle_t outputDeepBuffer = AUDIO_IO_HANDLE_NONE;
audio_io_handle_t outputPrimary = AUDIO_IO_HANDLE_NONE;
for (audio_io_handle_t output : outputs) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
if (activeOnly && !desc->isActive(toVolumeSource(AUDIO_STREAM_MUSIC))) {
continue;
}
ALOGV("selectOutputForMusicEffects activeOnly %d output %d flags 0x%08x",
activeOnly, output, desc->mFlags);
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
outputOffloaded = output;
}
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
outputDeepBuffer = output;
}
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) != 0) {
outputPrimary = output;
}
}
if (outputOffloaded != AUDIO_IO_HANDLE_NONE) {
output = outputOffloaded;
} else if (outputDeepBuffer != AUDIO_IO_HANDLE_NONE) {
output = outputDeepBuffer;
} else if (outputPrimary != AUDIO_IO_HANDLE_NONE) {
output = outputPrimary;
} else {
output = outputs[0];
}
activeOnly = false;
}
if (output != mMusicEffectOutput) {
mEffects.moveEffects(AUDIO_SESSION_OUTPUT_MIX, mMusicEffectOutput, output);
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mMusicEffectOutput, output);
mMusicEffectOutput = output;
}
ALOGV("selectOutputForMusicEffects selected output %d", output);
return output;
}
audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc __unused)
{
return selectOutputForMusicEffects();
}
status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
audio_io_handle_t io,
uint32_t strategy,
int session,
int id)
{
ssize_t index = mOutputs.indexOfKey(io);
if (index < 0) {
index = mInputs.indexOfKey(io);
if (index < 0) {
ALOGW("registerEffect() unknown io %d", io);
return INVALID_OPERATION;
}
}
return mEffects.registerEffect(desc, io, session, id,
(strategy == streamToStrategy(AUDIO_STREAM_MUSIC) ||
strategy == PRODUCT_STRATEGY_NONE));
}
status_t AudioPolicyManager::unregisterEffect(int id)
{
if (mEffects.getEffect(id) == nullptr) {
return INVALID_OPERATION;
}
if (mEffects.isEffectEnabled(id)) {
ALOGW("%s effect %d enabled", __FUNCTION__, id);
setEffectEnabled(id, false);
}
return mEffects.unregisterEffect(id);
}
void AudioPolicyManager::cleanUpEffectsForIo(audio_io_handle_t io)
{
EffectDescriptorCollection effects = mEffects.getEffectsForIo(io);
for (size_t i = 0; i < effects.size(); i++) {
ALOGW("%s removing stale effect %s, id %d on closed IO %d",
__func__, effects.valueAt(i)->mDesc.name, effects.keyAt(i), io);
unregisterEffect(effects.keyAt(i));
}
}
status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled)
{
sp<EffectDescriptor> effect = mEffects.getEffect(id);
if (effect == nullptr) {
return INVALID_OPERATION;
}
status_t status = mEffects.setEffectEnabled(id, enabled);
if (status == NO_ERROR) {
mInputs.trackEffectEnabled(effect, enabled);
}
return status;
}
status_t AudioPolicyManager::moveEffectsToIo(const std::vector<int>& ids, audio_io_handle_t io)
{
mEffects.moveEffects(ids, io);
return NO_ERROR;
}
bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
{
return mOutputs.isActive(toVolumeSource(stream), inPastMs);
}
bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const
{
return mOutputs.isActiveRemotely(toVolumeSource(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(const 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++) {
AudioMix mix = mixes[i];
// Only capture of playback is allowed in LOOP_BACK & RENDER mode
if (is_mix_loopback_render(mix.mRouteFlags) && mix.mMixType != MIX_TYPE_PLAYERS) {
ALOGE("Unsupported Policy Mix %zu of %zu: "
"Only capture of playback is allowed in LOOP_BACK & RENDER mode",
i, mixes.size());
res = INVALID_OPERATION;
break;
}
// LOOP_BACK and LOOP_BACK | RENDER have the same remote submix backend and are handled
// in the same way.
if ((mix.mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
ALOGV("registerPolicyMixes() mix %zu of %zu is LOOP_BACK %d", i, mixes.size(),
mix.mRouteFlags);
if (rSubmixModule == 0) {
rSubmixModule = mHwModules.getModuleFromName(
AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX);
if (rSubmixModule == 0) {
ALOGE("Unable to find audio module for submix, aborting mix %zu registration",
i);
res = INVALID_OPERATION;
break;
}
}
String8 address = mix.mDeviceAddress;
audio_devices_t deviceTypeToMakeAvailable;
if (mix.mMixType == MIX_TYPE_PLAYERS) {
mix.mDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
deviceTypeToMakeAvailable = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
} else {
mix.mDeviceType = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
deviceTypeToMakeAvailable = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
}
if (mPolicyMixes.registerMix(mix, 0 /*output desc*/) != NO_ERROR) {
ALOGE("Error registering mix %zu for address %s", i, address.string());
res = INVALID_OPERATION;
break;
}
audio_config_t outputConfig = mix.mFormat;
audio_config_t inputConfig = mix.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 ((res = setDeviceConnectionStateInt(deviceTypeToMakeAvailable,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.string(), "remote-submix", AUDIO_FORMAT_DEFAULT)) != NO_ERROR) {
ALOGE("Failed to set remote submix device available, type %u, address %s",
mix.mDeviceType, address.string());
break;
}
} else if ((mix.mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
String8 address = mix.mDeviceAddress;
audio_devices_t type = mix.mDeviceType;
ALOGV(" registerPolicyMixes() mix %zu of %zu is RENDER, dev=0x%X addr=%s",
i, mixes.size(), type, address.string());
sp<DeviceDescriptor> device = mHwModules.getDeviceDescriptor(
mix.mDeviceType, mix.mDeviceAddress,
String8(), AUDIO_FORMAT_DEFAULT);
if (device == nullptr) {
res = INVALID_OPERATION;
break;
}
bool foundOutput = false;
for (size_t j = 0 ; j < mOutputs.size() ; j++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(j);
if (desc->supportedDevices().contains(device)) {
if (mPolicyMixes.registerMix(mix, desc) != NO_ERROR) {
ALOGE("Could not register mix RENDER, dev=0x%X addr=%s", type,
address.string());
res = INVALID_OPERATION;
} else {
foundOutput = true;
}
break;
}
}
if (res != NO_ERROR) {
ALOGE(" Error registering mix %zu for device 0x%X addr %s",
i, type, address.string());
res = INVALID_OPERATION;
break;
} else if (!foundOutput) {
ALOGE(" Output not found for mix %zu for device 0x%X addr %s",
i, type, 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 (const auto& mix : mixes) {
if ((mix.mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
if (rSubmixModule == 0) {
rSubmixModule = mHwModules.getModuleFromName(
AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX);
if (rSubmixModule == 0) {
res = INVALID_OPERATION;
continue;
}
}
String8 address = mix.mDeviceAddress;
if (mPolicyMixes.unregisterMix(mix) != NO_ERROR) {
res = INVALID_OPERATION;
continue;
}
for (auto device : {AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_DEVICE_OUT_REMOTE_SUBMIX}) {
if (getDeviceConnectionState(device, address.string()) ==
AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
res = setDeviceConnectionStateInt(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.string(), "remote-submix",
AUDIO_FORMAT_DEFAULT);
if (res != OK) {
ALOGE("Error making RemoteSubmix device unavailable for mix "
"with type %d, address %s", device, address.string());
}
}
}
rSubmixModule->removeOutputProfile(address);
rSubmixModule->removeInputProfile(address);
} else if ((mix.mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
if (mPolicyMixes.unregisterMix(mix) != NO_ERROR) {
res = INVALID_OPERATION;
continue;
}
}
}
return res;
}
void AudioPolicyManager::dumpManualSurroundFormats(String8 *dst) const
{
size_t i = 0;
constexpr size_t audioFormatPrefixLen = sizeof("AUDIO_FORMAT_");
for (const auto& fmt : mManualSurroundFormats) {
if (i++ != 0) dst->append(", ");
std::string sfmt;
FormatConverter::toString(fmt, sfmt);
dst->append(sfmt.size() >= audioFormatPrefixLen ?
sfmt.c_str() + audioFormatPrefixLen - 1 : sfmt.c_str());
}
}
status_t AudioPolicyManager::setUidDeviceAffinities(uid_t uid,
const Vector<AudioDeviceTypeAddr>& devices) {
ALOGV("%s() uid=%d num devices %zu", __FUNCTION__, uid, devices.size());
// uid/device affinity is only for output devices
for (size_t i = 0; i < devices.size(); i++) {
if (!audio_is_output_device(devices[i].mType)) {
ALOGE("setUidDeviceAffinities() device=%08x is NOT an output device",
devices[i].mType);
return BAD_VALUE;
}
}
status_t res = mPolicyMixes.setUidDeviceAffinities(uid, devices);
if (res == NO_ERROR) {
// reevaluate outputs for all given devices
for (size_t i = 0; i < devices.size(); i++) {
sp<DeviceDescriptor> devDesc = mHwModules.getDeviceDescriptor(
devices[i].mType, devices[i].mAddress, String8(),
AUDIO_FORMAT_DEFAULT);
SortedVector<audio_io_handle_t> outputs;
if (checkOutputsForDevice(devDesc, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
outputs) != NO_ERROR) {
ALOGE("setUidDeviceAffinities() error in checkOutputsForDevice for device=%08x"
" addr=%s", devices[i].mType, devices[i].mAddress.string());
return INVALID_OPERATION;
}
}
}
return res;
}
status_t AudioPolicyManager::removeUidDeviceAffinities(uid_t uid) {
ALOGV("%s() uid=%d", __FUNCTION__, uid);
status_t res = mPolicyMixes.removeUidDeviceAffinities(uid);
if (res != NO_ERROR) {
ALOGE("%s() Could not remove all device affinities fo uid = %d",
__FUNCTION__, uid);
return INVALID_OPERATION;
}
return res;
}
void AudioPolicyManager::dump(String8 *dst) const
{
dst->appendFormat("\nAudioPolicyManager Dump: %p\n", this);
dst->appendFormat(" Primary Output: %d\n",
hasPrimaryOutput() ? mPrimaryOutput->mIoHandle : AUDIO_IO_HANDLE_NONE);
std::string stateLiteral;
AudioModeConverter::toString(mEngine->getPhoneState(), stateLiteral);
dst->appendFormat(" Phone state: %s\n", stateLiteral.c_str());
const char* forceUses[AUDIO_POLICY_FORCE_USE_CNT] = {
"communications", "media", "record", "dock", "system",
"HDMI system audio", "encoded surround output", "vibrate ringing" };
for (audio_policy_force_use_t i = AUDIO_POLICY_FORCE_FOR_COMMUNICATION;
i < AUDIO_POLICY_FORCE_USE_CNT; i = (audio_policy_force_use_t)((int)i + 1)) {
audio_policy_forced_cfg_t forceUseValue = mEngine->getForceUse(i);
dst->appendFormat(" Force use for %s: %d", forceUses[i], forceUseValue);
if (i == AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND &&
forceUseValue == AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL) {
dst->append(" (MANUAL: ");
dumpManualSurroundFormats(dst);
dst->append(")");
}
dst->append("\n");
}
dst->appendFormat(" TTS output %savailable\n", mTtsOutputAvailable ? "" : "not ");
dst->appendFormat(" Master mono: %s\n", mMasterMono ? "on" : "off");
dst->appendFormat(" Config source: %s\n", mConfig.getSource().c_str()); // getConfig not const
mAvailableOutputDevices.dump(dst, String8("Available output"));
mAvailableInputDevices.dump(dst, String8("Available input"));
mHwModulesAll.dump(dst);
mOutputs.dump(dst);
mInputs.dump(dst);
mEffects.dump(dst);
mAudioPatches.dump(dst);
mPolicyMixes.dump(dst);
mAudioSources.dump(dst);
dst->appendFormat(" AllowedCapturePolicies:\n");
for (auto& policy : mAllowedCapturePolicies) {
dst->appendFormat(" - uid=%d flag_mask=%#x\n", policy.first, policy.second);
}
dst->appendFormat("\nPolicy Engine dump:\n");
mEngine->dump(dst);
}
status_t AudioPolicyManager::dump(int fd)
{
String8 result;
dump(&result);
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioPolicyManager::setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t capturePolicy)
{
mAllowedCapturePolicies[uid] = capturePolicy;
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
if (property_get_bool("audio.offload.disable", false /* default_value */)) {
ALOGV("offload disabled by audio.offload.disable");
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
const int min_duration_secs = property_get_int32(
"audio.offload.min.duration.secs", -1 /* default_value */);
if (min_duration_secs >= 0) {
if (offloadInfo.duration_us < min_duration_secs * 1000000LL) {
ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%d)",
min_duration_secs);
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 = getProfileForOutput(DeviceVector() /*ignore device */,
offloadInfo.sample_rate,
offloadInfo.format,
offloadInfo.channel_mask,
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,
true /* directOnly */);
ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
return (profile != 0);
}
bool AudioPolicyManager::isDirectOutputSupported(const audio_config_base_t& config,
const audio_attributes_t& attributes) {
audio_output_flags_t output_flags = AUDIO_OUTPUT_FLAG_NONE;
audio_flags_to_audio_output_flags(attributes.flags, &output_flags);
sp<IOProfile> profile = getProfileForOutput(DeviceVector() /*ignore device */,
config.sample_rate,
config.format,
config.channel_mask,
output_flags,
true /* directOnly */);
ALOGV("%s() profile %sfound with name: %s, "
"sample rate: %u, format: 0x%x, channel_mask: 0x%x, output flags: 0x%x",
__FUNCTION__, profile != 0 ? "" : "NOT ",
(profile != 0 ? profile->getTagName().string() : "null"),
config.sample_rate, config.format, config.channel_mask, output_flags);
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 (const auto& dev : mAvailableOutputDevices) {
if (dev->type() == AUDIO_DEVICE_OUT_STUB) {
continue;
}
if (portsWritten < portsMax) {
dev->toAudioPort(&ports[portsWritten++]);
}
(*num_ports)++;
}
}
if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
for (const auto& dev : mAvailableInputDevices) {
if (dev->type() == AUDIO_DEVICE_IN_STUB) {
continue;
}
if (portsWritten < portsMax) {
dev->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)
{
if (port == nullptr || port->id == AUDIO_PORT_HANDLE_NONE) {
return BAD_VALUE;
}
sp<DeviceDescriptor> dev = mAvailableOutputDevices.getDeviceFromId(port->id);
if (dev != 0) {
dev->toAudioPort(port);
return NO_ERROR;
}
dev = mAvailableInputDevices.getDeviceFromId(port->id);
if (dev != 0) {
dev->toAudioPort(port);
return NO_ERROR;
}
sp<SwAudioOutputDescriptor> out = mOutputs.getOutputFromId(port->id);
if (out != 0) {
out->toAudioPort(port);
return NO_ERROR;
}
sp<AudioInputDescriptor> in = mInputs.getInputFromId(port->id);
if (in != 0) {
in->toAudioPort(port);
return NO_ERROR;
}
return BAD_VALUE;
}
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 (!audio_patch_is_valid(patch)) {
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(DeviceVector(devDesc),
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);
setOutputDevices(outputDesc, devices, 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> device =
mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
if (device == 0) {
return BAD_VALUE;
}
if (!inputDesc->mProfile->isCompatibleProfile(DeviceVector(device),
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("%s() setting device %s on output %d", __func__,
device->toString().c_str(), inputDesc->mIoHandle);
setInputDevice(inputDesc->mIoHandle, device, 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> srcDevice =
mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
if (srcDevice == 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;
srcDevice->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> sinkDevice =
mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
if (sinkDevice == 0) {
return BAD_VALUE;
}
sinkDevice->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
// create a software bridge in PatchPanel if:
// - source and sink devices are on different HW modules OR
// - audio HAL version is < 3.0
// - audio HAL version is >= 3.0 but no route has been declared between devices
if (!srcDevice->hasSameHwModuleAs(sinkDevice) ||
(srcDevice->getModuleVersionMajor() < 3) ||
!srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice)) {
// 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 =
getOutputsForDevices(DeviceVector(sinkDevice), 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
const audio_io_handle_t output = selectOutput(outputs);
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
status_t status = installPatch(__func__, index, handle, &newPatch, 0, uid, &patchDesc);
if (status != NO_ERROR) {
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;
}
setOutputDevices(outputDesc,
getNewOutputDevices(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),
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(product_strategy_t ps, audio_io_handle_t ouptutToSkip)
{
// Take the first attributes following the product strategy as it is used to retrieve the routed
// device. All attributes wihin a strategy follows the same "routing strategy"
auto attributes = mEngine->getAllAttributesForProductStrategy(ps).front();
DeviceVector devices = mEngine->getOutputDevicesForAttributes(attributes, nullptr, false);
SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
for (size_t j = 0; j < mOutputs.size(); j++) {
if (mOutputs.keyAt(j) == ouptutToSkip) {
continue;
}
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(j);
if (!outputDesc->isStrategyActive(ps)) {
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 (auto stream : mEngine->getStreamTypesForProductStrategy(ps)) {
mpClientInterface->invalidateStream(stream);
}
} else {
setOutputDevices(
outputDesc, getNewOutputDevices(outputDesc, false /*fromCache*/), false);
}
}
}
void AudioPolicyManager::clearSessionRoutes(uid_t uid)
{
// remove output routes associated with this uid
std::vector<product_strategy_t> affectedStrategies;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
for (const auto& client : outputDesc->getClientIterable()) {
if (client->hasPreferredDevice() && client->uid() == uid) {
client->setPreferredDeviceId(AUDIO_PORT_HANDLE_NONE);
auto clientStrategy = client->strategy();
if (std::find(begin(affectedStrategies), end(affectedStrategies), clientStrategy) !=
end(affectedStrategies)) {
continue;
}
affectedStrategies.push_back(client->strategy());
}
}
}
// reroute outputs if necessary
for (const auto& strategy : affectedStrategies) {
checkStrategyRoute(strategy, AUDIO_IO_HANDLE_NONE);
}
// remove input routes associated with this uid
SortedVector<audio_source_t> affectedSources;
for (size_t i = 0; i < mInputs.size(); i++) {
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(i);
for (const auto& client : inputDesc->getClientIterable()) {
if (client->hasPreferredDevice() && client->uid() == uid) {
client->setPreferredDeviceId(AUDIO_PORT_HANDLE_NONE);
affectedSources.add(client->source());
}
}
}
// 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->source()) >= 0) {
inputsToClose.add(inputDesc->mIoHandle);
}
}
for (const auto& input : inputsToClose) {
closeInput(input);
}
}
void AudioPolicyManager::clearAudioSources(uid_t uid)
{
for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--) {
sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
if (sourceDesc->uid() == 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);
audio_attributes_t attr = { .source = AUDIO_SOURCE_HOTWORD };
*device = mEngine->getInputDeviceForAttributes(attr)->type();
return mSoundTriggerSessions.acquireSession(*session, *ioHandle);
}
status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
audio_port_handle_t *portId,
uid_t uid)
{
ALOGV("%s", __FUNCTION__);
*portId = AUDIO_PORT_HANDLE_NONE;
if (source == NULL || attributes == NULL || portId == NULL) {
ALOGW("%s invalid argument: source %p attributes %p handle %p",
__FUNCTION__, source, attributes, portId);
return BAD_VALUE;
}
if (source->role != AUDIO_PORT_ROLE_SOURCE ||
source->type != AUDIO_PORT_TYPE_DEVICE) {
ALOGW("%s INVALID_OPERATION source->role %d source->type %d",
__FUNCTION__, source->role, source->type);
return INVALID_OPERATION;
}
sp<DeviceDescriptor> srcDevice =
mAvailableInputDevices.getDevice(source->ext.device.type,
String8(source->ext.device.address),
AUDIO_FORMAT_DEFAULT);
if (srcDevice == 0) {
ALOGW("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type);
return BAD_VALUE;
}
*portId = AudioPort::getNextUniqueId();
struct audio_patch dummyPatch = {};
sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
sp<SourceClientDescriptor> sourceDesc =
new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDevice,
mEngine->getStreamTypeForAttributes(*attributes),
mEngine->getProductStrategyForAttributes(*attributes),
toVolumeSource(*attributes));
status_t status = connectAudioSource(sourceDesc);
if (status == NO_ERROR) {
mAudioSources.add(*portId, sourceDesc);
}
return status;
}
status_t AudioPolicyManager::connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
{
ALOGV("%s handle %d", __FUNCTION__, sourceDesc->portId());
// make sure we only have one patch per source.
disconnectAudioSource(sourceDesc);
audio_attributes_t attributes = sourceDesc->attributes();
audio_stream_type_t stream = sourceDesc->stream();
sp<DeviceDescriptor> srcDevice = sourceDesc->srcDevice();
DeviceVector sinkDevices =
mEngine->getOutputDevicesForAttributes(attributes, nullptr, true);
ALOG_ASSERT(!sinkDevices.isEmpty(), "connectAudioSource(): no device found for attributes");
sp<DeviceDescriptor> sinkDevice = sinkDevices.itemAt(0);
ALOG_ASSERT(mAvailableOutputDevices.contains(sinkDevice), "%s: Device %s not available",
__FUNCTION__, sinkDevice->toString().c_str());
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (srcDevice->hasSameHwModuleAs(sinkDevice) &&
srcDevice->getModuleVersionMajor() >= 3 &&
sinkDevice->getModule()->supportsPatch(srcDevice, sinkDevice) &&
srcDevice->getAudioPort()->mGains.size() > 0) {
ALOGV("%s Device to Device route supported by >=3.0 HAL", __FUNCTION__);
// TODO: may explicitly specify whether we should use HW or SW patch
// create patch between src device and output device
// create Hwoutput and add to mHwOutputs
} else {
audio_attributes_t resultAttr;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = sourceDesc->config().sample_rate;
config.channel_mask = sourceDesc->config().channel_mask;
config.format = sourceDesc->config().format;
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
bool isRequestedDeviceForExclusiveUse = false;
std::vector<sp<SwAudioOutputDescriptor>> secondaryOutputs;
getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE,
&attributes, &stream, sourceDesc->uid(), &config, &flags,
&selectedDeviceId, &isRequestedDeviceForExclusiveUse,
&secondaryOutputs);
if (output == AUDIO_IO_HANDLE_NONE) {
ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevices.types());
return INVALID_OPERATION;
}
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc->isDuplicated()) {
ALOGV("%s output for device %08x is duplicated", __FUNCTION__, sinkDevices.types());
return INVALID_OPERATION;
}
status_t status = outputDesc->start();
if (status != NO_ERROR) {
return status;
}
// 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.
PatchBuilder patchBuilder;
patchBuilder.addSource(srcDevice).addSource(outputDesc, { .stream = stream });
status = mpClientInterface->createAudioPatch(patchBuilder.patch(),
&afPatchHandle,
0);
ALOGV("%s patch panel returned %d patchHandle %d", __FUNCTION__,
status, afPatchHandle);
sourceDesc->patchDesc()->mPatch = *patchBuilder.patch();
if (status != NO_ERROR) {
ALOGW("%s patch panel could not connect device patch, error %d",
__FUNCTION__, status);
return INVALID_OPERATION;
}
if (outputDesc->getClient(sourceDesc->portId()) != nullptr) {
ALOGW("%s source portId has already been attached to outputDesc", __func__);
return INVALID_OPERATION;
}
outputDesc->addClient(sourceDesc);
uint32_t delayMs = 0;
status = startSource(outputDesc, sourceDesc, &delayMs);
if (status != NO_ERROR) {
mpClientInterface->releaseAudioPatch(sourceDesc->patchDesc()->mAfPatchHandle, 0);
outputDesc->removeClient(sourceDesc->portId());
outputDesc->stop();
return status;
}
sourceDesc->setSwOutput(outputDesc);
if (delayMs != 0) {
usleep(delayMs * 1000);
}
}
sourceDesc->patchDesc()->mAfPatchHandle = afPatchHandle;
addAudioPatch(sourceDesc->patchDesc()->mHandle, sourceDesc->patchDesc());
return NO_ERROR;
}
status_t AudioPolicyManager::stopAudioSource(audio_port_handle_t portId)
{
sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueFor(portId);
ALOGV("%s port ID %d", __FUNCTION__, portId);
if (sourceDesc == 0) {
ALOGW("%s unknown source for port ID %d", __FUNCTION__, portId);
return BAD_VALUE;
}
status_t status = disconnectAudioSource(sourceDesc);
mAudioSources.removeItem(portId);
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 (const auto& handle : offloaded) {
closeOutput(handle);
}
}
// 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;
}
float AudioPolicyManager::getStreamVolumeDB(
audio_stream_type_t stream, int index, audio_devices_t device)
{
return computeVolume(getVolumeCurves(stream), toVolumeSource(stream), index, device);
}
status_t AudioPolicyManager::getSurroundFormats(unsigned int *numSurroundFormats,
audio_format_t *surroundFormats,
bool *surroundFormatsEnabled,
bool reported)
{
if (numSurroundFormats == NULL || (*numSurroundFormats != 0 &&
(surroundFormats == NULL || surroundFormatsEnabled == NULL))) {
return BAD_VALUE;
}
ALOGV("%s() numSurroundFormats %d surroundFormats %p surroundFormatsEnabled %p reported %d",
__func__, *numSurroundFormats, surroundFormats, surroundFormatsEnabled, reported);
size_t formatsWritten = 0;
size_t formatsMax = *numSurroundFormats;
std::unordered_set<audio_format_t> formats; // Uses primary surround formats only
if (reported) {
// Return formats from all device profiles that have already been resolved by
// checkOutputsForDevice().
for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
sp<DeviceDescriptor> device = mAvailableOutputDevices[i];
FormatVector supportedFormats =
device->getAudioPort()->getAudioProfiles().getSupportedFormats();
for (size_t j = 0; j < supportedFormats.size(); j++) {
if (mConfig.getSurroundFormats().count(supportedFormats[j]) != 0) {
formats.insert(supportedFormats[j]);
} else {
for (const auto& pair : mConfig.getSurroundFormats()) {
if (pair.second.count(supportedFormats[j]) != 0) {
formats.insert(pair.first);
break;
}
}
}
}
}
} else {
for (const auto& pair : mConfig.getSurroundFormats()) {
formats.insert(pair.first);
}
}
*numSurroundFormats = formats.size();
audio_policy_forced_cfg_t forceUse = mEngine->getForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND);
for (const auto& format: formats) {
if (formatsWritten < formatsMax) {
surroundFormats[formatsWritten] = format;
bool formatEnabled = true;
switch (forceUse) {
case AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL:
formatEnabled = mManualSurroundFormats.count(format) != 0;
break;
case AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER:
formatEnabled = false;
break;
default: // AUTO or ALWAYS => true
break;
}
surroundFormatsEnabled[formatsWritten++] = formatEnabled;
}
}
return NO_ERROR;
}
status_t AudioPolicyManager::setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled)
{
ALOGV("%s() format 0x%X enabled %d", __func__, audioFormat, enabled);
const auto& formatIter = mConfig.getSurroundFormats().find(audioFormat);
if (formatIter == mConfig.getSurroundFormats().end()) {
ALOGW("%s() format 0x%X is not a known surround format", __func__, audioFormat);
return BAD_VALUE;
}
if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND) !=
AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL) {
ALOGW("%s() not in manual mode for surround sound format selection", __func__);
return INVALID_OPERATION;
}
if ((mManualSurroundFormats.count(audioFormat) != 0) == enabled) {
return NO_ERROR;
}
std::unordered_set<audio_format_t> surroundFormatsBackup(mManualSurroundFormats);
if (enabled) {
mManualSurroundFormats.insert(audioFormat);
for (const auto& subFormat : formatIter->second) {
mManualSurroundFormats.insert(subFormat);
}
} else {
mManualSurroundFormats.erase(audioFormat);
for (const auto& subFormat : formatIter->second) {
mManualSurroundFormats.erase(subFormat);
}
}
sp<SwAudioOutputDescriptor> outputDesc;
bool profileUpdated = false;
DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(
AUDIO_DEVICE_OUT_HDMI);
for (size_t i = 0; i < hdmiOutputDevices.size(); i++) {
// Simulate reconnection to update enabled surround sound formats.
String8 address = hdmiOutputDevices[i]->address();
String8 name = hdmiOutputDevices[i]->getName();
status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.c_str(),
name.c_str(),
AUDIO_FORMAT_DEFAULT);
if (status != NO_ERROR) {
continue;
}
status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.c_str(),
name.c_str(),
AUDIO_FORMAT_DEFAULT);
profileUpdated |= (status == NO_ERROR);
}
// FIXME: Why doing this for input HDMI devices if we don't augment their reported formats?
DeviceVector hdmiInputDevices = mAvailableInputDevices.getDevicesFromTypeMask(
AUDIO_DEVICE_IN_HDMI);
for (size_t i = 0; i < hdmiInputDevices.size(); i++) {
// Simulate reconnection to update enabled surround sound formats.
String8 address = hdmiInputDevices[i]->address();
String8 name = hdmiInputDevices[i]->getName();
status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.c_str(),
name.c_str(),
AUDIO_FORMAT_DEFAULT);
if (status != NO_ERROR) {
continue;
}
status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.c_str(),
name.c_str(),
AUDIO_FORMAT_DEFAULT);
profileUpdated |= (status == NO_ERROR);
}
if (!profileUpdated) {
ALOGW("%s() no audio profiles updated, undoing surround formats change", __func__);
mManualSurroundFormats = std::move(surroundFormatsBackup);
}
return profileUpdated ? NO_ERROR : INVALID_OPERATION;
}
void AudioPolicyManager::setAppState(uid_t uid, app_state_t state)
{
ALOGV("%s(uid:%d, state:%d)", __func__, uid, state);
for (size_t i = 0; i < mInputs.size(); i++) {
mInputs.valueAt(i)->setAppState(uid, state);
}
}
bool AudioPolicyManager::isHapticPlaybackSupported()
{
for (const auto& hwModule : mHwModules) {
const OutputProfileCollection &outputProfiles = hwModule->getOutputProfiles();
for (const auto &outProfile : outputProfiles) {
struct audio_port audioPort;
outProfile->toAudioPort(&audioPort);
for (size_t i = 0; i < audioPort.num_channel_masks; i++) {
if (audioPort.channel_masks[i] & AUDIO_CHANNEL_HAPTIC_ALL) {
return true;
}
}
}
}
return false;
}
status_t AudioPolicyManager::disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
{
ALOGV("%s port Id %d", __FUNCTION__, sourceDesc->portId());
sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->patchDesc()->mHandle);
if (patchDesc == 0) {
ALOGW("%s source has no patch with handle %d", __FUNCTION__,
sourceDesc->patchDesc()->mHandle);
return BAD_VALUE;
}
removeAudioPatch(sourceDesc->patchDesc()->mHandle);
sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->swOutput().promote();
if (swOutputDesc != 0) {
status_t status = stopSource(swOutputDesc, sourceDesc);
if (status == NO_ERROR) {
swOutputDesc->stop();
}
swOutputDesc->removeClient(sourceDesc->portId());
mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
} else {
sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->hwOutput().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<SourceClientDescriptor> AudioPolicyManager::getSourceForAttributesOnOutput(
audio_io_handle_t output, const audio_attributes_t &attr)
{
sp<SourceClientDescriptor> source;
for (size_t i = 0; i < mAudioSources.size(); i++) {
sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
sp<SwAudioOutputDescriptor> outputDesc = sourceDesc->swOutput().promote();
if (followsSameRouting(attr, sourceDesc->attributes()) &&
outputDesc != 0 && outputDesc->mIoHandle == output) {
source = sourceDesc;
break;
}
}
return source;
}
// ----------------------------------------------------------------------------
// AudioPolicyManager
// ----------------------------------------------------------------------------
uint32_t AudioPolicyManager::nextAudioPortGeneration()
{
return mAudioPortGeneration++;
}
// Treblized audio policy xml config will be located in /odm/etc or /vendor/etc.
static const char *kConfigLocationList[] =
{"/odm/etc", "/vendor/etc", "/system/etc"};
static const int kConfigLocationListSize =
(sizeof(kConfigLocationList) / sizeof(kConfigLocationList[0]));
static status_t deserializeAudioPolicyXmlConfig(AudioPolicyConfig &config) {
char audioPolicyXmlConfigFile[AUDIO_POLICY_XML_CONFIG_FILE_PATH_MAX_LENGTH];
std::vector<const char*> fileNames;
status_t ret;
if (property_get_bool("ro.bluetooth.a2dp_offload.supported", false)) {
if (property_get_bool("persist.bluetooth.bluetooth_audio_hal.disabled", false) &&
property_get_bool("persist.bluetooth.a2dp_offload.disabled", false)) {
// Both BluetoothAudio@2.0 and BluetoothA2dp@1.0 (Offlaod) are disabled, and uses
// the legacy hardware module for A2DP and hearing aid.
fileNames.push_back(AUDIO_POLICY_BLUETOOTH_LEGACY_HAL_XML_CONFIG_FILE_NAME);
} else if (property_get_bool("persist.bluetooth.a2dp_offload.disabled", false)) {
// A2DP offload supported but disabled: try to use special XML file
fileNames.push_back(AUDIO_POLICY_A2DP_OFFLOAD_DISABLED_XML_CONFIG_FILE_NAME);
}
} else if (property_get_bool("persist.bluetooth.bluetooth_audio_hal.disabled", false)) {
fileNames.push_back(AUDIO_POLICY_BLUETOOTH_LEGACY_HAL_XML_CONFIG_FILE_NAME);
}
fileNames.push_back(AUDIO_POLICY_XML_CONFIG_FILE_NAME);
for (const char* fileName : fileNames) {
for (int i = 0; i < kConfigLocationListSize; i++) {
snprintf(audioPolicyXmlConfigFile, sizeof(audioPolicyXmlConfigFile),
"%s/%s", kConfigLocationList[i], fileName);
ret = deserializeAudioPolicyFile(audioPolicyXmlConfigFile, &config);
if (ret == NO_ERROR) {
config.setSource(audioPolicyXmlConfigFile);
return ret;
}
}
}
return ret;
}
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface,
bool /*forTesting*/)
:
mUidCached(AID_AUDIOSERVER), // no need to call getuid(), there's only one of us running.
mpClientInterface(clientInterface),
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
mA2dpSuspended(false),
mConfig(mHwModulesAll, mAvailableOutputDevices, mAvailableInputDevices, mDefaultOutputDevice),
mAudioPortGeneration(1),
mBeaconMuteRefCount(0),
mBeaconPlayingRefCount(0),
mBeaconMuted(false),
mTtsOutputAvailable(false),
mMasterMono(false),
mMusicEffectOutput(AUDIO_IO_HANDLE_NONE)
{
}
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
: AudioPolicyManager(clientInterface, false /*forTesting*/)
{
loadConfig();
initialize();
}
// This check is to catch any legacy platform updating to Q without having
// switched to XML since its deprecation on O.
// TODO: after Q release, remove this check and flag as XML is now the only
// option and all legacy platform should have transitioned to XML.
#ifndef USE_XML_AUDIO_POLICY_CONF
#error Audio policy no longer supports legacy .conf configuration format
#endif
void AudioPolicyManager::loadConfig() {
if (deserializeAudioPolicyXmlConfig(getConfig()) != NO_ERROR) {
ALOGE("could not load audio policy configuration file, setting defaults");
getConfig().setDefault();
}
}
status_t AudioPolicyManager::initialize() {
// 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 NO_INIT;
}
// Retrieve the Policy Manager Interface
mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>();
if (mEngine == NULL) {
ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__);
return NO_INIT;
}
mEngine->setObserver(this);
status_t status = mEngine->initCheck();
if (status != NO_ERROR) {
LOG_FATAL("Policy engine not initialized(err=%d)", status);
return status;
}
// mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
// open all output streams needed to access attached devices
for (const auto& hwModule : mHwModulesAll) {
hwModule->setHandle(mpClientInterface->loadHwModule(hwModule->getName()));
if (hwModule->getHandle() == AUDIO_MODULE_HANDLE_NONE) {
ALOGW("could not open HW module %s", hwModule->getName());
continue;
}
mHwModules.push_back(hwModule);
// 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 (const auto& outProfile : hwModule->getOutputProfiles()) {
if (!outProfile->canOpenNewIo()) {
ALOGE("Invalid Output profile max open count %u for profile %s",
outProfile->maxOpenCount, outProfile->getTagName().c_str());
continue;
}
if (!outProfile->hasSupportedDevices()) {
ALOGW("Output profile contains no device on module %s", hwModule->getName());
continue;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_TTS) != 0) {
mTtsOutputAvailable = true;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
continue;
}
const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
DeviceVector availProfileDevices = supportedDevices.filter(mAvailableOutputDevices);
sp<DeviceDescriptor> supportedDevice = 0;
if (supportedDevices.contains(mDefaultOutputDevice)) {
supportedDevice = mDefaultOutputDevice;
} else {
// choose first device present in profile's SupportedDevices also part of
// mAvailableOutputDevices.
if (availProfileDevices.isEmpty()) {
continue;
}
supportedDevice = availProfileDevices.itemAt(0);
}
if (!mAvailableOutputDevices.contains(supportedDevice)) {
continue;
}
sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = outputDesc->open(nullptr, DeviceVector(supportedDevice),
AUDIO_STREAM_DEFAULT,
AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
ALOGW("Cannot open output stream for devices %s on hw module %s",
supportedDevice->toString().c_str(), hwModule->getName());
continue;
}
for (const auto &device : availProfileDevices) {
// give a valid ID to an attached device once confirmed it is reachable
if (!device->isAttached()) {
device->attach(hwModule);
}
}
if (mPrimaryOutput == 0 &&
outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = outputDesc;
}
addOutput(output, outputDesc);
setOutputDevices(outputDesc,
DeviceVector(supportedDevice),
true,
0,
NULL);
}
// open input streams needed to access attached devices to validate
// mAvailableInputDevices list
for (const auto& inProfile : hwModule->getInputProfiles()) {
if (!inProfile->canOpenNewIo()) {
ALOGE("Invalid Input profile max open count %u for profile %s",
inProfile->maxOpenCount, inProfile->getTagName().c_str());
continue;
}
if (!inProfile->hasSupportedDevices()) {
ALOGW("Input profile contains no device on module %s", hwModule->getName());
continue;
}
// chose first device present in profile's SupportedDevices also part of
// available input devices
const DeviceVector &supportedDevices = inProfile->getSupportedDevices();
DeviceVector availProfileDevices = supportedDevices.filter(mAvailableInputDevices);
if (availProfileDevices.isEmpty()) {
ALOGE("%s: Input device list is empty!", __FUNCTION__);
continue;
}
sp<AudioInputDescriptor> inputDesc =
new AudioInputDescriptor(inProfile, mpClientInterface);
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = inputDesc->open(nullptr,
availProfileDevices.itemAt(0),
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE,
&input);
if (status != NO_ERROR) {
ALOGW("Cannot open input stream for device %s on hw module %s",
availProfileDevices.toString().c_str(),
hwModule->getName());
continue;
}
for (const auto &device : availProfileDevices) {
// give a valid ID to an attached device once confirmed it is reachable
if (!device->isAttached()) {
device->attach(hwModule);
device->importAudioPort(inProfile, true);
}
}
inputDesc->close();
}
}
// make sure all attached devices have been allocated a unique ID
auto checkAndSetAvailable = [this](auto& devices) {
for (size_t i = 0; i < devices.size();) {
const auto &device = devices[i];
if (!device->isAttached()) {
ALOGW("device %s is unreachable", device->toString().c_str());
devices.remove(device);
continue;
}
// Device is now validated and can be appended to the available devices of the engine
setEngineDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
i++;
}
};
checkAndSetAvailable(mAvailableOutputDevices);
checkAndSetAvailable(mAvailableInputDevices);
// make sure default device is reachable
if (mDefaultOutputDevice == 0 || !mAvailableOutputDevices.contains(mDefaultOutputDevice)) {
ALOGE_IF(mDefaultOutputDevice != 0, "Default device %s is unreachable",
mDefaultOutputDevice->toString().c_str());
status = NO_INIT;
}
// If microphones address is empty, set it according to device type
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
if (mAvailableInputDevices[i]->address().isEmpty()) {
if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BUILTIN_MIC) {
mAvailableInputDevices[i]->setAddress(String8(AUDIO_BOTTOM_MICROPHONE_ADDRESS));
} else if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BACK_MIC) {
mAvailableInputDevices[i]->setAddress(String8(AUDIO_BACK_MICROPHONE_ADDRESS));
}
}
}
if (mPrimaryOutput == 0) {
ALOGE("Failed to open primary output");
status = NO_INIT;
}
// Silence ALOGV statements
property_set("log.tag." LOG_TAG, "D");
updateDevicesAndOutputs();
return status;
}
AudioPolicyManager::~AudioPolicyManager()
{
for (size_t i = 0; i < mOutputs.size(); i++) {
mOutputs.valueAt(i)->close();
}
for (size_t i = 0; i < mInputs.size(); i++) {
mInputs.valueAt(i)->close();
}
mAvailableOutputDevices.clear();
mAvailableInputDevices.clear();
mOutputs.clear();
mInputs.clear();
mHwModules.clear();
mHwModulesAll.clear();
mManualSurroundFormats.clear();
}
status_t AudioPolicyManager::initCheck()
{
return hasPrimaryOutput() ? NO_ERROR : NO_INIT;
}
// ---
void AudioPolicyManager::addOutput(audio_io_handle_t output,
const sp<SwAudioOutputDescriptor>& outputDesc)
{
mOutputs.add(output, outputDesc);
applyStreamVolumes(outputDesc, AUDIO_DEVICE_NONE, 0 /* delayMs */, true /* force */);
updateMono(output); // update mono status when adding to output list
selectOutputForMusicEffects();
nextAudioPortGeneration();
}
void AudioPolicyManager::removeOutput(audio_io_handle_t output)
{
mOutputs.removeItem(output);
selectOutputForMusicEffects();
}
void AudioPolicyManager::addInput(audio_io_handle_t input,
const sp<AudioInputDescriptor>& inputDesc)
{
mInputs.add(input, inputDesc);
nextAudioPortGeneration();
}
status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor>& device,
audio_policy_dev_state_t state,
SortedVector<audio_io_handle_t>& outputs)
{
audio_devices_t deviceType = device->type();
const String8 &address = device->address();
sp<SwAudioOutputDescriptor> desc;
if (audio_device_is_digital(deviceType)) {
// erase all current sample rates, formats and channel masks
device->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->supportsDevice(device)
&& desc->deviceSupportsEncodedFormats(deviceType)) {
ALOGV("checkOutputsForDevice(): adding opened output %d on device %s",
mOutputs.keyAt(i), device->toString().c_str());
outputs.add(mOutputs.keyAt(i));
}
}
// then look for output profiles that can be routed to this device
SortedVector< sp<IOProfile> > profiles;
for (const auto& hwModule : mHwModules) {
for (size_t j = 0; j < hwModule->getOutputProfiles().size(); j++) {
sp<IOProfile> profile = hwModule->getOutputProfiles()[j];
if (profile->supportsDevice(device)) {
profiles.add(profile);
ALOGV("checkOutputsForDevice(): adding profile %zu from module %s",
j, hwModule->getName());
}
}
}
ALOGV(" found %zu profiles, %zu outputs", profiles.size(), outputs.size());
if (profiles.isEmpty() && outputs.isEmpty()) {
ALOGW("checkOutputsForDevice(): No output available for device %04x", deviceType);
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(deviceType)) {
device->importAudioPort(profile);
}
break;
}
}
if (j != outputs.size()) {
continue;
}
if (!profile->canOpenNewIo()) {
ALOGW("Max Output number %u already opened for this profile %s",
profile->maxOpenCount, profile->getTagName().c_str());
continue;
}
ALOGV("opening output for device %08x with params %s profile %p name %s",
deviceType, address.string(), profile.get(), profile->getName().string());
desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = desc->open(nullptr, DeviceVector(device),
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status == NO_ERROR) {
// Here is where the out_set_parameters() for card & device gets called
if (!address.isEmpty()) {
char *param = audio_device_address_to_parameter(deviceType, address);
mpClientInterface->setParameters(output, String8(param));
free(param);
}
updateAudioProfiles(device, output, profile->getAudioProfiles());
if (!profile->hasValidAudioProfile()) {
ALOGW("checkOutputsForDevice() missing param");
desc->close();
output = AUDIO_IO_HANDLE_NONE;
} else if (profile->hasDynamicAudioProfile()) {
desc->close();
output = AUDIO_IO_HANDLE_NONE;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
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_t status = desc->open(&config, DeviceVector(device),
AUDIO_STREAM_DEFAULT,
AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
output = AUDIO_IO_HANDLE_NONE;
}
}
if (output != AUDIO_IO_HANDLE_NONE) {
addOutput(output, desc);
if (device_distinguishes_on_address(deviceType) && address != "0") {
sp<AudioPolicyMix> policyMix;
if (mPolicyMixes.getAudioPolicyMix(deviceType, address, policyMix)
== NO_ERROR) {
policyMix->setOutput(desc);
desc->mPolicyMix = policyMix;
} else {
ALOGW("checkOutputsForDevice() cannot find policy for address %s",
address.string());
}
} 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;
//TODO: configure audio effect output stage here
// open a duplicating output thread for the new output and the primary output
sp<SwAudioOutputDescriptor> dupOutputDesc =
new SwAudioOutputDescriptor(NULL, mpClientInterface);
status_t status = dupOutputDesc->openDuplicating(mPrimaryOutput, desc,
&duplicatedOutput);
if (status == NO_ERROR) {
// add duplicated output descriptor
addOutput(duplicatedOutput, dupOutputDesc);
} else {
ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
mPrimaryOutput->mIoHandle, output);
desc->close();
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", deviceType);
profiles.removeAt(profile_index);
profile_index--;
} else {
outputs.add(output);
// Load digital format info only for digital devices
if (audio_device_is_digital(deviceType)) {
device->importAudioPort(profile);
}
if (device_distinguishes_on_address(deviceType)) {
ALOGV("checkOutputsForDevice(): setOutputDevices %s",
device->toString().c_str());
setOutputDevices(desc, DeviceVector(device), true/*force*/, 0/*delay*/,
NULL/*patch handle*/);
}
ALOGV("checkOutputsForDevice(): adding output %d", output);
}
}
if (profiles.isEmpty()) {
ALOGW("checkOutputsForDevice(): No output available for device %04x", deviceType);
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(deviceType) && desc->supportsDevice(device)
&& desc->deviceSupportsEncodedFormats(deviceType)) {
outputs.add(mOutputs.keyAt(i));
} else if (!mAvailableOutputDevices.containsAtLeastOne(desc->supportedDevices())) {
ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
mOutputs.keyAt(i));
outputs.add(mOutputs.keyAt(i));
}
}
}
// Clear any profiles associated with the disconnected device.
for (const auto& hwModule : mHwModules) {
for (size_t j = 0; j < hwModule->getOutputProfiles().size(); j++) {
sp<IOProfile> profile = hwModule->getOutputProfiles()[j];
if (profile->supportsDevice(device)) {
ALOGV("checkOutputsForDevice(): "
"clearing direct output profile %zu on module %s",
j, hwModule->getName());
profile->clearAudioProfiles();
}
}
}
}
return NO_ERROR;
}
status_t AudioPolicyManager::checkInputsForDevice(const sp<DeviceDescriptor>& device,
audio_policy_dev_state_t state)
{
sp<AudioInputDescriptor> desc;
if (audio_device_is_digital(device->type())) {
// erase all current sample rates, formats and channel masks
device->clearAudioProfiles();
}
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
// look for input profiles that can be routed to this device
SortedVector< sp<IOProfile> > profiles;
for (const auto& hwModule : mHwModules) {
for (size_t profile_index = 0;
profile_index < hwModule->getInputProfiles().size();
profile_index++) {
sp<IOProfile> profile = hwModule->getInputProfiles()[profile_index];
if (profile->supportsDevice(device)) {
profiles.add(profile);
ALOGV("checkInputsForDevice(): adding profile %zu from module %s",
profile_index, hwModule->getName());
}
}
}
if (profiles.isEmpty()) {
ALOGW("%s: No input profile available for device %s",
__func__, device->toString().c_str());
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->type())) {
device->importAudioPort(profile);
}
break;
}
}
if (input_index != mInputs.size()) {
continue;
}
if (!profile->canOpenNewIo()) {
ALOGW("Max Input number %u already opened for this profile %s",
profile->maxOpenCount, profile->getTagName().c_str());
continue;
}
desc = new AudioInputDescriptor(profile, mpClientInterface);
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = desc->open(nullptr,
device,
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE,
&input);
if (status == NO_ERROR) {
const String8& address = device->address();
if (!address.isEmpty()) {
char *param = audio_device_address_to_parameter(device->type(), address);
mpClientInterface->setParameters(input, String8(param));
free(param);
}
updateAudioProfiles(device, input, profile->getAudioProfiles());
if (!profile->hasValidAudioProfile()) {
ALOGW("checkInputsForDevice() direct input missing param");
desc->close();
input = AUDIO_IO_HANDLE_NONE;
}
if (input != AUDIO_IO_HANDLE_NONE) {
addInput(input, desc);
}
} // endif input != 0
if (input == AUDIO_IO_HANDLE_NONE) {
ALOGW("%s could not open input for device %s", __func__,
device->toString().c_str());
profiles.removeAt(profile_index);
profile_index--;
} else {
if (audio_device_is_digital(device->type())) {
device->importAudioPort(profile);
}
ALOGV("checkInputsForDevice(): adding input %d", input);
}
} // end scan profiles
if (profiles.isEmpty()) {
ALOGW("%s: No input available for device %s", __func__, device->toString().c_str());
return BAD_VALUE;
}
} else {
// Disconnect
// Clear any profiles associated with the disconnected device.
for (const auto& hwModule : mHwModules) {
for (size_t profile_index = 0;
profile_index < hwModule->getInputProfiles().size();
profile_index++) {
sp<IOProfile> profile = hwModule->getInputProfiles()[profile_index];
if (profile->supportsDevice(device)) {
ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %s",
profile_index, hwModule->getName());
profile->clearAudioProfiles();
}
}
}
} // end disconnect
return NO_ERROR;
}
void AudioPolicyManager::closeOutput(audio_io_handle_t output)
{
ALOGV("closeOutput(%d)", output);
sp<SwAudioOutputDescriptor> closingOutput = mOutputs.valueFor(output);
if (closingOutput == NULL) {
ALOGW("closeOutput() unknown output %d", output);
return;
}
const bool closingOutputWasActive = closingOutput->isActive();
mPolicyMixes.closeOutput(closingOutput);
// look for duplicated outputs connected to the output being removed.
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> dupOutput = mOutputs.valueAt(i);
if (dupOutput->isDuplicated() &&
(dupOutput->mOutput1 == closingOutput || dupOutput->mOutput2 == closingOutput)) {
sp<SwAudioOutputDescriptor> remainingOutput =
dupOutput->mOutput1 == closingOutput ? dupOutput->mOutput2 : dupOutput->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.
const bool wasActive = remainingOutput->isActive();
// Note: no-op on the closing output where all clients has already been set inactive
dupOutput->setAllClientsInactive();
// stop() will be a no op if the output is still active but is needed in case all
// active streams refcounts where cleared above
if (wasActive) {
remainingOutput->stop();
}
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(closingOutput->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
(void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
mAudioPatches.removeItemsAt(index);
mpClientInterface->onAudioPatchListUpdate();
}
if (closingOutputWasActive) {
closingOutput->stop();
}
closingOutput->close();
removeOutput(output);
mPreviousOutputs = mOutputs;
// MSD patches may have been released to support a non-MSD direct output. Reset MSD patch if
// no direct outputs are open.
if (!getMsdAudioOutDevices().isEmpty()) {
bool directOutputOpen = false;
for (size_t i = 0; i < mOutputs.size(); i++) {
if (mOutputs[i]->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
directOutputOpen = true;
break;
}
}
if (!directOutputOpen) {
ALOGV("no direct outputs open, reset MSD patch");
setMsdPatch();
}
}
cleanUpEffectsForIo(output);
}
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();
sp<DeviceDescriptor> device = inputDesc->getDevice();
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();
}
inputDesc->close();
mInputs.removeItem(input);
DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
if (primaryInputDevices.contains(device) &&
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
SoundTrigger::setCaptureState(false);
}
cleanUpEffectsForIo(input);
}
SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevices(
const DeviceVector &devices,
const SwAudioOutputCollection& openOutputs)
{
SortedVector<audio_io_handle_t> outputs;
ALOGVV("%s() devices %s", __func__, devices.toString().c_str());
for (size_t i = 0; i < openOutputs.size(); i++) {
ALOGVV("output %zu isDuplicated=%d device=%s",
i, openOutputs.valueAt(i)->isDuplicated(),
openOutputs.valueAt(i)->supportedDevices().toString().c_str());
if (openOutputs.valueAt(i)->supportsAllDevices(devices)
&& openOutputs.valueAt(i)->deviceSupportsEncodedFormats(devices.types())) {
ALOGVV("%s() found output %d", __func__, openOutputs.keyAt(i));
outputs.add(openOutputs.keyAt(i));
}
}
return outputs;
}
void AudioPolicyManager::checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked)
{
// checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
// output is suspended before any tracks are moved to it
checkA2dpSuspend();
checkOutputForAllStrategies();
checkSecondaryOutputs();
if (onOutputsChecked != nullptr && onOutputsChecked()) checkA2dpSuspend();
updateDevicesAndOutputs();
if (mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD) != 0) {
setMsdPatch();
}
}
bool AudioPolicyManager::followsSameRouting(const audio_attributes_t &lAttr,
const audio_attributes_t &rAttr) const
{
return mEngine->getProductStrategyForAttributes(lAttr) ==
mEngine->getProductStrategyForAttributes(rAttr);
}
void AudioPolicyManager::checkOutputForAttributes(const audio_attributes_t &attr)
{
auto psId = mEngine->getProductStrategyForAttributes(attr);
DeviceVector oldDevices = mEngine->getOutputDevicesForAttributes(attr, 0, true /*fromCache*/);
DeviceVector newDevices = mEngine->getOutputDevicesForAttributes(attr, 0, false /*fromCache*/);
SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevices(oldDevices, mPreviousOutputs);
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevices(newDevices, 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("%s(): policy related outputs", __func__);
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 (srcOutputs != dstOutputs) {
// get maximum latency of all source outputs to determine the minimum mute time guaranteeing
// audio from invalidated tracks will be rendered when unmuting
uint32_t maxLatency = 0;
for (audio_io_handle_t srcOut : srcOutputs) {
sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueFor(srcOut);
if (desc != 0 && maxLatency < desc->latency()) {
maxLatency = desc->latency();
}
}
ALOGV_IF(!(srcOutputs.isEmpty() || dstOutputs.isEmpty()),
"%s: strategy %d, moving from output %s to output %s", __func__, psId,
std::to_string(srcOutputs[0]).c_str(),
std::to_string(dstOutputs[0]).c_str());
// mute strategy while moving tracks from one output to another
for (audio_io_handle_t srcOut : srcOutputs) {
sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueFor(srcOut);
if (desc != 0 && desc->isStrategyActive(psId)) {
setStrategyMute(psId, true, desc);
setStrategyMute(psId, false, desc, maxLatency * LATENCY_MUTE_FACTOR,
newDevices.types());
}
sp<SourceClientDescriptor> source = getSourceForAttributesOnOutput(srcOut, attr);
if (source != 0){
connectAudioSource(source);
}
}
// Move effects associated to this stream from previous output to new output
if (followsSameRouting(attr, attributes_initializer(AUDIO_USAGE_MEDIA))) {
selectOutputForMusicEffects();
}
// Move tracks associated to this stream (and linked) from previous output to new output
for (auto stream : mEngine->getStreamTypesForProductStrategy(psId)) {
mpClientInterface->invalidateStream(stream);
}
}
}
void AudioPolicyManager::checkOutputForAllStrategies()
{
for (const auto &strategy : mEngine->getOrderedProductStrategies()) {
auto attributes = mEngine->getAllAttributesForProductStrategy(strategy).front();
checkOutputForAttributes(attributes);
}
}
void AudioPolicyManager::checkSecondaryOutputs() {
std::set<audio_stream_type_t> streamsToInvalidate;
for (size_t i = 0; i < mOutputs.size(); i++) {
const sp<SwAudioOutputDescriptor>& outputDescriptor = mOutputs[i];
for (const sp<TrackClientDescriptor>& client : outputDescriptor->getClientIterable()) {
sp<SwAudioOutputDescriptor> desc;
std::vector<sp<SwAudioOutputDescriptor>> secondaryDescs;
status_t status = mPolicyMixes.getOutputForAttr(client->attributes(), client->uid(),
client->flags(), desc, &secondaryDescs);
if (status != OK ||
!std::equal(client->getSecondaryOutputs().begin(),
client->getSecondaryOutputs().end(),
secondaryDescs.begin(), secondaryDescs.end())) {
streamsToInvalidate.insert(client->stream());
}
}
}
for (audio_stream_type_t stream : streamsToInvalidate) {
ALOGD("%s Invalidate stream %d due to secondary output change", __func__, stream);
mpClientInterface->invalidateStream(stream);
}
}
void AudioPolicyManager::checkA2dpSuspend()
{
audio_io_handle_t a2dpOutput = mOutputs.getA2dpOutput();
if (a2dpOutput == 0 || mOutputs.isA2dpOffloadedOnPrimary()) {
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);
// if suspended, restore A2DP output if:
// ((SCO device is NOT connected) ||
// ((forced usage communication is NOT SCO) && (forced usage for record is NOT SCO) &&
// (phone state is NOT in call) && (phone state is NOT ringing)))
//
// if not suspended, suspend A2DP output if:
// (SCO device is connected) &&
// ((forced usage for communication is SCO) || (forced usage for record is SCO) ||
// ((phone state is in call) || (phone state is ringing)))
//
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;
}
}
}
DeviceVector AudioPolicyManager::getNewOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
bool fromCache)
{
DeviceVector devices;
ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
if (patchDesc->mUid != mUidCached) {
ALOGV("%s device %s forced by patch %d", __func__,
outputDesc->devices().toString().c_str(), outputDesc->getPatchHandle());
return outputDesc->devices();
}
}
// Honor explicit routing requests only if no client using default routing is active on this
// input: a specific app can not force routing for other apps by setting a preferred device.
bool active; // unused
sp<DeviceDescriptor> device =
findPreferredDevice(outputDesc, PRODUCT_STRATEGY_NONE, active, mAvailableOutputDevices);
if (device != nullptr) {
return DeviceVector(device);
}
// Legacy Engine cannot take care of bus devices and mix, so we need to handle the conflict
// of setForceUse / Default Bus device here
device = mPolicyMixes.getDeviceAndMixForOutput(outputDesc, mAvailableOutputDevices);
if (device != nullptr) {
return DeviceVector(device);
}
for (const auto &productStrategy : mEngine->getOrderedProductStrategies()) {
StreamTypeVector streams = mEngine->getStreamTypesForProductStrategy(productStrategy);
auto attr = mEngine->getAllAttributesForProductStrategy(productStrategy).front();
if ((hasVoiceStream(streams) &&
(isInCall() || mOutputs.isStrategyActiveOnSameModule(productStrategy, outputDesc))) ||
((hasStream(streams, AUDIO_STREAM_ALARM) || hasStream(streams, AUDIO_STREAM_ENFORCED_AUDIBLE)) &&
mOutputs.isStrategyActiveOnSameModule(productStrategy, outputDesc)) ||
outputDesc->isStrategyActive(productStrategy)) {
// Retrieval of devices for voice DL is done on primary output profile, cannot
// check the route (would force modifying configuration file for this profile)
devices = mEngine->getOutputDevicesForAttributes(attr, nullptr, fromCache);
break;
}
}
ALOGV("%s selected devices %s", __func__, devices.toString().c_str());
return devices;
}
sp<DeviceDescriptor> AudioPolicyManager::getNewInputDevice(
const sp<AudioInputDescriptor>& inputDesc)
{
sp<DeviceDescriptor> device;
ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
if (patchDesc->mUid != mUidCached) {
ALOGV("getNewInputDevice() device %s forced by patch %d",
inputDesc->getDevice()->toString().c_str(), inputDesc->getPatchHandle());
return inputDesc->getDevice();
}
}
// Honor explicit routing requests only if no client using default routing is active on this
// input: a specific app can not force routing for other apps by setting a preferred device.
bool active;
device = findPreferredDevice(inputDesc, AUDIO_SOURCE_DEFAULT, active, mAvailableInputDevices);
if (device != nullptr) {
return device;
}
// If we are not in call and no client is active on this input, this methods returns
// a null sp<>, causing the patch on the input stream to be released.
audio_attributes_t attributes = inputDesc->getHighestPriorityAttributes();
if (attributes.source == AUDIO_SOURCE_DEFAULT && isInCall()) {
attributes.source = AUDIO_SOURCE_VOICE_COMMUNICATION;
}
if (attributes.source != AUDIO_SOURCE_DEFAULT) {
device = mEngine->getInputDeviceForAttributes(attributes);
}
return device;
}
bool AudioPolicyManager::streamsMatchForvolume(audio_stream_type_t stream1,
audio_stream_type_t stream2) {
return (stream1 == stream2);
}
audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
// By checking the range of stream before calling getStrategy, we avoid
// getOutputDevicesForStream's behavior for invalid streams.
// engine's getOutputDevicesForStream would fallback on its default behavior (most probably
// device for music stream), but we want to return the empty set.
if (stream < AUDIO_STREAM_MIN || stream >= AUDIO_STREAM_PUBLIC_CNT) {
return AUDIO_DEVICE_NONE;
}
DeviceVector activeDevices;
DeviceVector devices;
for (audio_stream_type_t curStream = AUDIO_STREAM_MIN; curStream < AUDIO_STREAM_PUBLIC_CNT;
curStream = (audio_stream_type_t) (curStream + 1)) {
if (!streamsMatchForvolume(stream, curStream)) {
continue;
}
DeviceVector curDevices = mEngine->getOutputDevicesForStream(curStream, false/*fromCache*/);
devices.merge(curDevices);
for (audio_io_handle_t output : getOutputsForDevices(curDevices, mOutputs)) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc->isActive(toVolumeSource(curStream))) {
activeDevices.merge(outputDesc->devices());
}
}
}
// Favor devices selected on active streams if any to report correct device in case of
// explicit device selection
if (!activeDevices.isEmpty()) {
devices = activeDevices;
}
/*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
and doesn't really need to.*/
DeviceVector speakerSafeDevices = devices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER_SAFE);
if (!speakerSafeDevices.isEmpty()) {
devices.merge(mAvailableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER));
devices.remove(speakerSafeDevices);
}
return devices.types();
}
void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
switch(stream) {
case AUDIO_STREAM_MUSIC:
checkOutputForAttributes(attributes_initializer(AUDIO_USAGE_NOTIFICATION));
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;
auto ttsVolumeSource = toVolumeSource(AUDIO_STREAM_TTS);
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
setVolumeSourceMute(ttsVolumeSource, 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;
}
void AudioPolicyManager::updateDevicesAndOutputs()
{
mEngine->updateDeviceSelectionCache();
mPreviousOutputs = mOutputs;
}
uint32_t AudioPolicyManager::checkDeviceMuteStrategies(const sp<AudioOutputDescriptor>& outputDesc,
const DeviceVector &prevDevices,
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;
DeviceVector devices = outputDesc->devices();
bool shouldMute = outputDesc->isActive() && (devices.size() >= 2);
auto productStrategies = mEngine->getOrderedProductStrategies();
for (const auto &productStrategy : productStrategies) {
auto attributes = mEngine->getAllAttributesForProductStrategy(productStrategy).front();
DeviceVector curDevices =
mEngine->getOutputDevicesForAttributes(attributes, nullptr, false/*fromCache*/);
curDevices = curDevices.filter(outputDesc->supportedDevices());
bool mute = shouldMute && curDevices.containsAtLeastOne(devices) && curDevices != devices;
bool doMute = false;
if (mute && !outputDesc->isStrategyMutedByDevice(productStrategy)) {
doMute = true;
outputDesc->setStrategyMutedByDevice(productStrategy, true);
} else if (!mute && outputDesc->isStrategyMutedByDevice(productStrategy)) {
doMute = true;
outputDesc->setStrategyMutedByDevice(productStrategy, 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().containsAtLeastOne(outputDesc->supportedDevices())) {
continue;
}
ALOGVV("%s() %s (curDevice %s)", __func__,
mute ? "muting" : "unmuting", curDevices.toString().c_str());
setStrategyMute(productStrategy, mute, desc, mute ? 0 : delayMs);
if (desc->isStrategyActive(productStrategy)) {
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() && (devices != prevDevices)) {
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 (const auto &activeVs : outputDesc->getActiveVolumeSources()) {
// make sure that we do not start the temporary mute period too early in case of
// delayed device change
setVolumeSourceMute(activeVs, true, outputDesc, delayMs);
setVolumeSourceMute(activeVs, false, outputDesc, delayMs + tempMuteDurationMs,
devices.types());
}
}
// 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::setOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
const DeviceVector &devices,
bool force,
int delayMs,
audio_patch_handle_t *patchHandle,
bool requiresMuteCheck)
{
ALOGV("%s device %s delayMs %d", __func__, devices.toString().c_str(), delayMs);
uint32_t muteWaitMs;
if (outputDesc->isDuplicated()) {
muteWaitMs = setOutputDevices(outputDesc->subOutput1(), devices, force, delayMs,
nullptr /* patchHandle */, requiresMuteCheck);
muteWaitMs += setOutputDevices(outputDesc->subOutput2(), devices, force, delayMs,
nullptr /* patchHandle */, requiresMuteCheck);
return muteWaitMs;
}
// filter devices according to output selected
DeviceVector filteredDevices = outputDesc->filterSupportedDevices(devices);
DeviceVector prevDevices = outputDesc->devices();
// no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
// output profile or if new device is not supported AND previous device(s) is(are) still
// available (otherwise reset device must be done on the output)
if (!devices.isEmpty() && filteredDevices.isEmpty() &&
!mAvailableOutputDevices.filter(prevDevices).empty()) {
ALOGV("%s: unsupported device %s for output", __func__, devices.toString().c_str());
return 0;
}
ALOGV("setOutputDevices() prevDevice %s", prevDevices.toString().c_str());
if (!filteredDevices.isEmpty()) {
outputDesc->setDevices(filteredDevices);
}
// if the outputs are not materially active, there is no need to mute.
if (requiresMuteCheck) {
muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevices, delayMs);
} else {
ALOGV("%s: suppressing checkDeviceMuteStrategies", __func__);
muteWaitMs = 0;
}
// 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 setOutputDevices() without conditions
if ((filteredDevices.isEmpty() || filteredDevices == prevDevices) &&
!force && outputDesc->getPatchHandle() != 0) {
ALOGV("%s setting same device %s or null device, force=%d, patch handle=%d", __func__,
filteredDevices.toString().c_str(), force, outputDesc->getPatchHandle());
return muteWaitMs;
}
ALOGV("%s changing device to %s", __func__, filteredDevices.toString().c_str());
// do the routing
if (filteredDevices.isEmpty()) {
resetOutputDevice(outputDesc, delayMs, NULL);
} else {
PatchBuilder patchBuilder;
patchBuilder.addSource(outputDesc);
ALOG_ASSERT(filteredDevices.size() <= AUDIO_PATCH_PORTS_MAX, "Too many sink ports");
for (const auto &filteredDevice : filteredDevices) {
patchBuilder.addSink(filteredDevice);
}
installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), delayMs);
}
// update stream volumes according to new device
applyStreamVolumes(outputDesc, filteredDevices.types(), 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,
const sp<DeviceDescriptor> &device,
bool force,
audio_patch_handle_t *patchHandle)
{
status_t status = NO_ERROR;
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
if ((device != nullptr) && ((device != inputDesc->getDevice()) || force)) {
inputDesc->setDevice(device);
if (mAvailableInputDevices.contains(device)) {
PatchBuilder patchBuilder;
patchBuilder.addSink(inputDesc,
// AUDIO_SOURCE_HOTWORD is for internal use only:
// handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
[inputDesc](const PatchBuilder::mix_usecase_t& usecase) {
auto result = usecase;
if (result.source == AUDIO_SOURCE_HOTWORD && !inputDesc->isSoundTrigger()) {
result.source = AUDIO_SOURCE_VOICE_RECOGNITION;
}
return result; }).
//only one input device for now
addSource(device);
status = installPatch(__func__, patchHandle, inputDesc.get(), patchBuilder.patch(), 0);
}
}
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(const sp<DeviceDescriptor> &device,
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.
sp<IOProfile> firstInexact;
uint32_t updatedSamplingRate = 0;
audio_format_t updatedFormat = AUDIO_FORMAT_INVALID;
audio_channel_mask_t updatedChannelMask = AUDIO_CHANNEL_INVALID;
for (const auto& hwModule : mHwModules) {
for (const auto& profile : hwModule->getInputProfiles()) {
// profile->log();
//updatedFormat = format;
if (profile->isCompatibleProfile(DeviceVector(device), samplingRate,
&samplingRate /*updatedSamplingRate*/,
format,
&format, /*updatedFormat*/
channelMask,
&channelMask /*updatedChannelMask*/,
// FIXME ugly cast
(audio_output_flags_t) flags,
true /*exactMatchRequiredForInputFlags*/)) {
return profile;
}
if (firstInexact == nullptr && profile->isCompatibleProfile(DeviceVector(device),
samplingRate,
&updatedSamplingRate,
format,
&updatedFormat,
channelMask,
&updatedChannelMask,
// FIXME ugly cast
(audio_output_flags_t) flags,
false /*exactMatchRequiredForInputFlags*/)) {
firstInexact = profile;
}
}
}
if (firstInexact != nullptr) {
samplingRate = updatedSamplingRate;
format = updatedFormat;
channelMask = updatedChannelMask;
return firstInexact;
}
return NULL;
}
float AudioPolicyManager::computeVolume(IVolumeCurves &curves,
VolumeSource volumeSource,
int index,
audio_devices_t device)
{
float volumeDb = curves.volIndexToDb(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
const auto callVolumeSrc = toVolumeSource(AUDIO_STREAM_VOICE_CALL);
const auto ringVolumeSrc = toVolumeSource(AUDIO_STREAM_RING);
const auto musicVolumeSrc = toVolumeSource(AUDIO_STREAM_MUSIC);
const auto alarmVolumeSrc = toVolumeSource(AUDIO_STREAM_ALARM);
if (volumeSource == toVolumeSource(AUDIO_STREAM_ACCESSIBILITY)
&& (AUDIO_MODE_RINGTONE == mEngine->getPhoneState()) &&
mOutputs.isActive(ringVolumeSrc, 0)) {
auto &ringCurves = getVolumeCurves(AUDIO_STREAM_RING);
const float ringVolumeDb = computeVolume(ringCurves, ringVolumeSrc, index, device);
return ringVolumeDb - 4 > volumeDb ? ringVolumeDb - 4 : volumeDb;
}
// in-call: always cap volume by voice volume + some low headroom
if ((volumeSource != callVolumeSrc && (isInCall() ||
mOutputs.isActiveLocally(callVolumeSrc))) &&
(volumeSource == toVolumeSource(AUDIO_STREAM_SYSTEM) ||
volumeSource == ringVolumeSrc || volumeSource == musicVolumeSrc ||
volumeSource == alarmVolumeSrc ||
volumeSource == toVolumeSource(AUDIO_STREAM_NOTIFICATION) ||
volumeSource == toVolumeSource(AUDIO_STREAM_ENFORCED_AUDIBLE) ||
volumeSource == toVolumeSource(AUDIO_STREAM_DTMF) ||
volumeSource == toVolumeSource(AUDIO_STREAM_ACCESSIBILITY))) {
auto &voiceCurves = getVolumeCurves(callVolumeSrc);
int voiceVolumeIndex = voiceCurves.getVolumeIndex(device);
const float maxVoiceVolDb =
computeVolume(voiceCurves, callVolumeSrc, voiceVolumeIndex, device)
+ IN_CALL_EARPIECE_HEADROOM_DB;
// FIXME: Workaround for call screening applications until a proper audio mode is defined
// to support this scenario : Exempt the RING stream from the audio cap if the audio was
// programmatically muted.
// VOICE_CALL stream has minVolumeIndex > 0 : Users cannot set the volume of voice calls to
// 0. We don't want to cap volume when the system has programmatically muted the voice call
// stream. See setVolumeCurveIndex() for more information.
bool exemptFromCapping = (volumeSource == ringVolumeSrc) && (voiceVolumeIndex == 0);
ALOGV_IF(exemptFromCapping, "%s volume source %d at vol=%f not capped", __func__,
volumeSource, volumeDb);
if ((volumeDb > maxVoiceVolDb) && !exemptFromCapping) {
ALOGV("%s volume source %d at vol=%f overriden by volume group %d at vol=%f", __func__,
volumeSource, volumeDb, callVolumeSrc, maxVoiceVolDb);
volumeDb = maxVoiceVolDb;
}
}
// 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.
if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
AUDIO_DEVICE_OUT_USB_HEADSET | AUDIO_DEVICE_OUT_HEARING_AID)) &&
((volumeSource == alarmVolumeSrc ||
volumeSource == ringVolumeSrc) ||
(volumeSource == toVolumeSource(AUDIO_STREAM_NOTIFICATION)) ||
(volumeSource == toVolumeSource(AUDIO_STREAM_SYSTEM)) ||
((volumeSource == toVolumeSource(AUDIO_STREAM_ENFORCED_AUDIBLE)) &&
(mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) &&
curves.canBeMuted()) {
// 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 =
mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA),
nullptr, true /*fromCache*/).types();
auto &musicCurves = getVolumeCurves(AUDIO_STREAM_MUSIC);
float musicVolDb = computeVolume(musicCurves, musicVolumeSrc,
musicCurves.getVolumeIndex(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("%s increasing volume for volume source=%d device=0x%X from %f to %f",
__func__, volumeSource, 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) ||
(!(volumeSource == alarmVolumeSrc || volumeSource == ringVolumeSrc))) {
volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB;
}
}
return volumeDb;
}
int AudioPolicyManager::rescaleVolumeIndex(int srcIndex,
VolumeSource fromVolumeSource,
VolumeSource toVolumeSource)
{
if (fromVolumeSource == toVolumeSource) {
return srcIndex;
}
auto &srcCurves = getVolumeCurves(fromVolumeSource);
auto &dstCurves = getVolumeCurves(toVolumeSource);
float minSrc = (float)srcCurves.getVolumeIndexMin();
float maxSrc = (float)srcCurves.getVolumeIndexMax();
float minDst = (float)dstCurves.getVolumeIndexMin();
float maxDst = (float)dstCurves.getVolumeIndexMax();
// preserve mute request or correct range
if (srcIndex < minSrc) {
if (srcIndex == 0) {
return 0;
}
srcIndex = minSrc;
} else if (srcIndex > maxSrc) {
srcIndex = maxSrc;
}
return (int)(minDst + ((srcIndex - minSrc) * (maxDst - minDst)) / (maxSrc - minSrc));
}
status_t AudioPolicyManager::checkAndSetVolume(IVolumeCurves &curves,
VolumeSource volumeSource,
int index,
const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device,
int delayMs,
bool force)
{
// do not change actual attributes volume if the attributes is muted
if (outputDesc->isMuted(volumeSource)) {
ALOGVV("%s: volume source %d muted count %d active=%d", __func__, volumeSource,
outputDesc->getMuteCount(volumeSource), outputDesc->isActive(volumeSource));
return NO_ERROR;
}
VolumeSource callVolSrc = toVolumeSource(AUDIO_STREAM_VOICE_CALL);
VolumeSource btScoVolSrc = toVolumeSource(AUDIO_STREAM_BLUETOOTH_SCO);
bool isVoiceVolSrc = callVolSrc == volumeSource;
bool isBtScoVolSrc = btScoVolSrc == volumeSource;
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 sco and call follow same curves, bypass forceUseForComm
if ((callVolSrc != btScoVolSrc) &&
((isVoiceVolSrc && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
(isBtScoVolSrc && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO))) {
ALOGV("%s cannot set volume group %d volume with force use = %d for comm", __func__,
volumeSource, forceUseForComm);
return INVALID_OPERATION;
}
if (device == AUDIO_DEVICE_NONE) {
device = outputDesc->devices().types();
}
float volumeDb = computeVolume(curves, volumeSource, index, device);
if (outputDesc->isFixedVolume(device) ||
// Force VoIP volume to max for bluetooth SCO
((isVoiceVolSrc || isBtScoVolSrc) && (device & AUDIO_DEVICE_OUT_ALL_SCO) != 0)) {
volumeDb = 0.0f;
}
outputDesc->setVolume(volumeDb, volumeSource, curves.getStreamTypes(), device, delayMs, force);
if (isVoiceVolSrc || isBtScoVolSrc) {
float voiceVolume;
// Force voice volume to max or mute for Bluetooth SCO as other attenuations are managed by the headset
if (isVoiceVolSrc) {
voiceVolume = (float)index/(float)curves.getVolumeIndexMax();
} else {
voiceVolume = index == 0 ? 0.0 : 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 (const auto &volumeGroup : mEngine->getVolumeGroups()) {
auto &curves = getVolumeCurves(toVolumeSource(volumeGroup));
checkAndSetVolume(curves, toVolumeSource(volumeGroup),
curves.getVolumeIndex(device), outputDesc, device, delayMs, force);
}
}
void AudioPolicyManager::setStrategyMute(product_strategy_t strategy,
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs,
audio_devices_t device)
{
std::vector<VolumeSource> sourcesToMute;
for (auto attributes: mEngine->getAllAttributesForProductStrategy(strategy)) {
ALOGVV("%s() attributes %s, mute %d, output ID %d", __func__,
toString(attributes).c_str(), on, outputDesc->getId());
VolumeSource source = toVolumeSource(attributes);
if (std::find(begin(sourcesToMute), end(sourcesToMute), source) == end(sourcesToMute)) {
sourcesToMute.push_back(source);
}
}
for (auto source : sourcesToMute) {
setVolumeSourceMute(source, on, outputDesc, delayMs, device);
}
}
void AudioPolicyManager::setVolumeSourceMute(VolumeSource volumeSource,
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs,
audio_devices_t device)
{
if (device == AUDIO_DEVICE_NONE) {
device = outputDesc->devices().types();
}
auto &curves = getVolumeCurves(volumeSource);
if (on) {
if (!outputDesc->isMuted(volumeSource)) {
if (curves.canBeMuted() &&
(volumeSource != toVolumeSource(AUDIO_STREAM_ENFORCED_AUDIBLE) ||
(mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) ==
AUDIO_POLICY_FORCE_NONE))) {
checkAndSetVolume(curves, volumeSource, 0, outputDesc, device, delayMs);
}
}
// increment mMuteCount after calling checkAndSetVolume() so that volume change is not
// ignored
outputDesc->incMuteCount(volumeSource);
} else {
if (!outputDesc->isMuted(volumeSource)) {
ALOGV("%s unmuting non muted attributes!", __func__);
return;
}
if (outputDesc->decMuteCount(volumeSource) == 0) {
checkAndSetVolume(curves, volumeSource,
curves.getVolumeIndex(device),
outputDesc,
device,
delayMs);
}
}
}
bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa)
{
// has flags that map to a stream type?
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:
case AUDIO_USAGE_ASSISTANT:
break;
default:
return false;
}
return true;
}
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<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
if (sourceDesc->srcDevice()->equals(deviceDesc)) {
ALOGV("%s releasing audio source %d", __FUNCTION__, sourceDesc->portId());
stopAudioSource(sourceDesc->portId());
}
}
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);
}
}
mInputs.clearSessionRoutesForDevice(deviceDesc);
mHwModules.cleanUpForDevice(deviceDesc);
}
void AudioPolicyManager::modifySurroundFormats(
const sp<DeviceDescriptor>& devDesc, FormatVector *formatsPtr) {
std::unordered_set<audio_format_t> enforcedSurround(
devDesc->encodedFormats().begin(), devDesc->encodedFormats().end());
std::unordered_set<audio_format_t> allSurround; // A flat set of all known surround formats
for (const auto& pair : mConfig.getSurroundFormats()) {
allSurround.insert(pair.first);
for (const auto& subformat : pair.second) allSurround.insert(subformat);
}
audio_policy_forced_cfg_t forceUse = mEngine->getForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND);
ALOGD("%s: forced use = %d", __FUNCTION__, forceUse);
// This is the resulting set of formats depending on the surround mode:
// 'all surround' = allSurround
// 'enforced surround' = enforcedSurround [may include IEC69137 which isn't raw surround fmt]
// 'non-surround' = not in 'all surround' and not in 'enforced surround'
// 'manual surround' = mManualSurroundFormats
// AUTO: formats v 'enforced surround'
// ALWAYS: formats v 'all surround' v 'enforced surround'
// NEVER: formats ^ 'non-surround'
// MANUAL: formats ^ ('non-surround' v 'manual surround' v (IEC69137 ^ 'enforced surround'))
std::unordered_set<audio_format_t> formatSet;
if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL
|| forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) {
// formatSet is (formats ^ 'non-surround')
for (auto formatIter = formatsPtr->begin(); formatIter != formatsPtr->end(); ++formatIter) {
if (allSurround.count(*formatIter) == 0 && enforcedSurround.count(*formatIter) == 0) {
formatSet.insert(*formatIter);
}
}
} else {
formatSet.insert(formatsPtr->begin(), formatsPtr->end());
}
formatsPtr->clear(); // Re-filled from the formatSet at the end.
if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL) {
formatSet.insert(mManualSurroundFormats.begin(), mManualSurroundFormats.end());
// Enable IEC61937 when in MANUAL mode if it's enforced for this device.
if (enforcedSurround.count(AUDIO_FORMAT_IEC61937) != 0) {
formatSet.insert(AUDIO_FORMAT_IEC61937);
}
} else if (forceUse != AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) { // AUTO or ALWAYS
if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS) {
formatSet.insert(allSurround.begin(), allSurround.end());
}
formatSet.insert(enforcedSurround.begin(), enforcedSurround.end());
}
for (const auto& format : formatSet) {
formatsPtr->push(format);
}
}
void AudioPolicyManager::modifySurroundChannelMasks(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 or MANUAL, then make sure we at least support 5.1
} else if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS
|| forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_MANUAL) {
bool supports5dot1 = false;
// Are there any channel masks that can be considered "surround"?
for (audio_channel_mask_t channelMask : channelMasks) {
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 MANUAL or ALWAYS, so adding channelMask for 5.1 surround", __func__);
}
}
}
void AudioPolicyManager::updateAudioProfiles(const sp<DeviceDescriptor>& devDesc,
audio_io_handle_t ioHandle,
AudioProfileVector &profiles)
{
String8 reply;
audio_devices_t device = devDesc->type();
// Format MUST be checked first to update the list of AudioProfile
if (profiles.hasDynamicFormat()) {
reply = mpClientInterface->getParameters(
ioHandle, String8(AudioParameter::keyStreamSupportedFormats));
ALOGV("%s: supported formats %d, %s", __FUNCTION__, ioHandle, reply.string());
AudioParameter repliedParameters(reply);
if (repliedParameters.get(
String8(AudioParameter::keyStreamSupportedFormats), reply) != NO_ERROR) {
ALOGE("%s: failed to retrieve format, bailing out", __FUNCTION__);
return;
}
FormatVector formats = formatsFromString(reply.string());
if (device == AUDIO_DEVICE_OUT_HDMI
|| isDeviceOfModule(devDesc, AUDIO_HARDWARE_MODULE_ID_MSD)) {
modifySurroundFormats(devDesc, &formats);
}
profiles.setFormats(formats);
}
for (audio_format_t format : profiles.getSupportedFormats()) {
ChannelsVector channelMasks;
SampleRateVector samplingRates;
AudioParameter requestedParameters;
requestedParameters.addInt(String8(AudioParameter::keyFormat), format);
if (profiles.hasDynamicRateFor(format)) {
reply = mpClientInterface->getParameters(
ioHandle,
requestedParameters.toString() + ";" +
AudioParameter::keyStreamSupportedSamplingRates);
ALOGV("%s: supported sampling rates %s", __FUNCTION__, reply.string());
AudioParameter repliedParameters(reply);
if (repliedParameters.get(
String8(AudioParameter::keyStreamSupportedSamplingRates), reply) == NO_ERROR) {
samplingRates = samplingRatesFromString(reply.string());
}
}
if (profiles.hasDynamicChannelsFor(format)) {
reply = mpClientInterface->getParameters(ioHandle,
requestedParameters.toString() + ";" +
AudioParameter::keyStreamSupportedChannels);
ALOGV("%s: supported channel masks %s", __FUNCTION__, reply.string());
AudioParameter repliedParameters(reply);
if (repliedParameters.get(
String8(AudioParameter::keyStreamSupportedChannels), reply) == NO_ERROR) {
channelMasks = channelMasksFromString(reply.string());
if (device == AUDIO_DEVICE_OUT_HDMI
|| isDeviceOfModule(devDesc, AUDIO_HARDWARE_MODULE_ID_MSD)) {
modifySurroundChannelMasks(&channelMasks);
}
}
}
profiles.addProfileFromHal(new AudioProfile(format, channelMasks, samplingRates));
}
}
status_t AudioPolicyManager::installPatch(const char *caller,
audio_patch_handle_t *patchHandle,
AudioIODescriptorInterface *ioDescriptor,
const struct audio_patch *patch,
int delayMs)
{
ssize_t index = mAudioPatches.indexOfKey(
patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE ?
*patchHandle : ioDescriptor->getPatchHandle());
sp<AudioPatch> patchDesc;
status_t status = installPatch(
caller, index, patchHandle, patch, delayMs, mUidCached, &patchDesc);
if (status == NO_ERROR) {
ioDescriptor->setPatchHandle(patchDesc->mHandle);
}
return status;
}
status_t AudioPolicyManager::installPatch(const char *caller,
ssize_t index,
audio_patch_handle_t *patchHandle,
const struct audio_patch *patch,
int delayMs,
uid_t uid,
sp<AudioPatch> *patchDescPtr)
{
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("%s() AF::createAudioPatch returned %d patchHandle %d num_sources %d num_sinks %d",
caller, status, afPatchHandle, patch->num_sources, patch->num_sinks);
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch(patch, uid);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = *patch;
}
patchDesc->mAfPatchHandle = afPatchHandle;
if (patchHandle) {
*patchHandle = patchDesc->mHandle;
}
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
}
if (patchDescPtr) *patchDescPtr = patchDesc;
return status;
}
} // namespace android