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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 "AudioPolicyManager"
//#define LOG_NDEBUG 0
//#define VERY_VERBOSE_LOGGING
#ifdef VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
// A device mask for all audio input devices that are considered "virtual" when evaluating
// active inputs in getActiveInput()
#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER)
// A device mask for all audio output devices that are considered "remote" when evaluating
// active output devices in isStreamActiveRemotely()
#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
// A device mask for all audio input and output devices where matching inputs/outputs on device
// type alone is not enough: the address must match too
#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
#include <inttypes.h>
#include <math.h>
#include <cutils/properties.h>
#include <utils/Log.h>
#include <hardware/audio.h>
#include <hardware/audio_effect.h>
#include <media/AudioParameter.h>
#include <media/AudioPolicyHelper.h>
#include <soundtrigger/SoundTrigger.h>
#include "AudioPolicyManager.h"
#include "audio_policy_conf.h"
namespace android {
// ----------------------------------------------------------------------------
// Definitions for audio_policy.conf file parsing
// ----------------------------------------------------------------------------
struct StringToEnum {
const char *name;
uint32_t value;
};
#define STRING_TO_ENUM(string) { #string, string }
#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
const StringToEnum sDeviceNameToEnumTable[] = {
STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
};
const StringToEnum sOutputFlagNameToEnumTable[] = {
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
};
const StringToEnum sInputFlagNameToEnumTable[] = {
STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
};
const StringToEnum sFormatNameToEnumTable[] = {
STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
STRING_TO_ENUM(AUDIO_FORMAT_MP3),
STRING_TO_ENUM(AUDIO_FORMAT_AAC),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
STRING_TO_ENUM(AUDIO_FORMAT_AC3),
STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
};
const StringToEnum sOutChannelsNameToEnumTable[] = {
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
};
const StringToEnum sInChannelsNameToEnumTable[] = {
STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
};
const StringToEnum sGainModeNameToEnumTable[] = {
STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
};
uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table,
size_t size,
const char *name)
{
for (size_t i = 0; i < size; i++) {
if (strcmp(table[i].name, name) == 0) {
ALOGV("stringToEnum() found %s", table[i].name);
return table[i].value;
}
}
return 0;
}
const char *AudioPolicyManager::enumToString(const struct StringToEnum *table,
size_t size,
uint32_t value)
{
for (size_t i = 0; i < size; i++) {
if (table[i].value == value) {
return table[i].name;
}
}
return "";
}
bool AudioPolicyManager::stringToBool(const char *value)
{
return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
}
// ----------------------------------------------------------------------------
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address)
{
return setDeviceConnectionStateInt(device, state, device_address);
}
status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address)
{
ALOGV("setDeviceConnectionState() device: %x, state %d, address %s",
device, state, device_address != NULL ? device_address : "");
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address);
// handle output devices
if (audio_is_output_device(device)) {
SortedVector <audio_io_handle_t> outputs;
ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
// save a copy of the opened output descriptors before any output is opened or closed
// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
mPreviousOutputs = mOutputs;
switch (state)
{
// handle output device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
ALOGW("setDeviceConnectionState() device already connected: %x", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() connecting device %x", device);
// register new device as available
index = mAvailableOutputDevices.add(devDesc);
if (index >= 0) {
sp<HwModule> module = getModuleForDevice(device);
if (module == 0) {
ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
device);
mAvailableOutputDevices.remove(devDesc);
return INVALID_OPERATION;
}
mAvailableOutputDevices[index]->mId = nextUniqueId();
mAvailableOutputDevices[index]->mModule = module;
} else {
return NO_MEMORY;
}
if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
mAvailableOutputDevices.remove(devDesc);
return INVALID_OPERATION;
}
// outputs should never be empty here
ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
"checkOutputsForDevice() returned no outputs but status OK");
ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
outputs.size());
// Set connect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
} break;
// handle output device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
ALOGW("setDeviceConnectionState() device not connected: %x", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
// Set Disconnect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
// remove device from available output devices
mAvailableOutputDevices.remove(devDesc);
checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
} break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
// checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
// output is suspended before any tracks are moved to it
checkA2dpSuspend();
checkOutputForAllStrategies();
// outputs must be closed after checkOutputForAllStrategies() is executed
if (!outputs.isEmpty()) {
for (size_t i = 0; i < outputs.size(); i++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
// close unused outputs after device disconnection or direct outputs that have been
// opened by checkOutputsForDevice() to query dynamic parameters
if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
(desc->mDirectOpenCount == 0))) {
closeOutput(outputs[i]);
}
}
// check again after closing A2DP output to reset mA2dpSuspended if needed
checkA2dpSuspend();
}
updateDevicesAndOutputs();
if (mPhoneState == AUDIO_MODE_IN_CALL) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevice);
}
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_io_handle_t output = mOutputs.keyAt(i);
if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i),
true /*fromCache*/);
// do not force device change on duplicated output because if device is 0, it will
// also force a device 0 for the two outputs it is duplicated to which may override
// a valid device selection on those outputs.
bool force = !mOutputs.valueAt(i)->isDuplicated()
&& (!deviceDistinguishesOnAddress(device)
// always force when disconnecting (a non-duplicated device)
|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
setOutputDevice(output, newDevice, force, 0);
}
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is output device
// handle input devices
if (audio_is_input_device(device)) {
SortedVector <audio_io_handle_t> inputs;
ssize_t index = mAvailableInputDevices.indexOf(devDesc);
switch (state)
{
// handle input device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
ALOGW("setDeviceConnectionState() device already connected: %d", device);
return INVALID_OPERATION;
}
sp<HwModule> module = getModuleForDevice(device);
if (module == NULL) {
ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
device);
return INVALID_OPERATION;
}
if (checkInputsForDevice(device, state, inputs, devDesc->mAddress) != NO_ERROR) {
return INVALID_OPERATION;
}
index = mAvailableInputDevices.add(devDesc);
if (index >= 0) {
mAvailableInputDevices[index]->mId = nextUniqueId();
mAvailableInputDevices[index]->mModule = module;
} else {
return NO_MEMORY;
}
// Set connect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
} break;
// handle input device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
ALOGW("setDeviceConnectionState() device not connected: %d", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
// Set Disconnect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
checkInputsForDevice(device, state, inputs, devDesc->mAddress);
mAvailableInputDevices.remove(devDesc);
} break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
closeAllInputs();
if (mPhoneState == AUDIO_MODE_IN_CALL) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevice);
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is input device
ALOGW("setDeviceConnectionState() invalid device: %x", device);
return BAD_VALUE;
}
audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
const char *device_address)
{
sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address);
DeviceVector *deviceVector;
if (audio_is_output_device(device)) {
deviceVector = &mAvailableOutputDevices;
} else if (audio_is_input_device(device)) {
deviceVector = &mAvailableInputDevices;
} else {
ALOGW("getDeviceConnectionState() invalid device type %08x", device);
return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
ssize_t index = deviceVector->indexOf(devDesc);
if (index >= 0) {
return AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
} else {
return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
}
sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor(
const audio_devices_t device,
const char *device_address)
{
String8 address = (device_address == NULL) ? String8("") : String8(device_address);
// handle legacy remote submix case where the address was not always specified
if (deviceDistinguishesOnAddress(device) && (address.length() == 0)) {
address = String8("0");
}
for (size_t i = 0; i < mHwModules.size(); i++) {
if (mHwModules[i]->mHandle == 0) {
continue;
}
DeviceVector deviceList =
mHwModules[i]->mDeclaredDevices.getDevicesFromTypeAddr(device, address);
if (!deviceList.isEmpty()) {
return deviceList.itemAt(0);
}
deviceList = mHwModules[i]->mDeclaredDevices.getDevicesFromType(device);
if (!deviceList.isEmpty()) {
return deviceList.itemAt(0);
}
}
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
devDesc->mAddress = address;
return devDesc;
}
void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs)
{
bool createTxPatch = false;
struct audio_patch patch;
patch.num_sources = 1;
patch.num_sinks = 1;
status_t status;
audio_patch_handle_t afPatchHandle;
DeviceVector deviceList;
audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice);
// release existing RX patch if any
if (mCallRxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
mCallRxPatch.clear();
}
// release TX patch if any
if (mCallTxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
mCallTxPatch.clear();
}
// If the RX device is on the primary HW module, then use legacy routing method for voice calls
// via setOutputDevice() on primary output.
// Otherwise, create two audio patches for TX and RX path.
if (availablePrimaryOutputDevices() & rxDevice) {
setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs);
// If the TX device is also on the primary HW module, setOutputDevice() will take care
// of it due to legacy implementation. If not, create a patch.
if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN)
== AUDIO_DEVICE_NONE) {
createTxPatch = true;
}
} else {
// create RX path audio patch
deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice);
ALOG_ASSERT(!deviceList.isEmpty(),
"updateCallRouting() selected device not in output device list");
sp<DeviceDescriptor> rxSinkDeviceDesc = deviceList.itemAt(0);
deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX);
ALOG_ASSERT(!deviceList.isEmpty(),
"updateCallRouting() no telephony RX device");
sp<DeviceDescriptor> rxSourceDeviceDesc = deviceList.itemAt(0);
rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
// request to reuse existing output stream if one is already opened to reach the RX device
SortedVector<audio_io_handle_t> outputs =
getOutputsForDevice(rxDevice, mOutputs);
audio_io_handle_t output = selectOutput(outputs,
AUDIO_OUTPUT_FLAG_NONE,
AUDIO_FORMAT_INVALID);
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ALOG_ASSERT(!outputDesc->isDuplicated(),
"updateCallRouting() RX device output is duplicated");
outputDesc->toAudioPortConfig(&patch.sources[1]);
patch.num_sources = 2;
}
afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch",
status);
if (status == NO_ERROR) {
mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
&patch, mUidCached);
mCallRxPatch->mAfPatchHandle = afPatchHandle;
mCallRxPatch->mUid = mUidCached;
}
createTxPatch = true;
}
if (createTxPatch) {
struct audio_patch patch;
patch.num_sources = 1;
patch.num_sinks = 1;
deviceList = mAvailableInputDevices.getDevicesFromType(txDevice);
ALOG_ASSERT(!deviceList.isEmpty(),
"updateCallRouting() selected device not in input device list");
sp<DeviceDescriptor> txSourceDeviceDesc = deviceList.itemAt(0);
txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX);
ALOG_ASSERT(!deviceList.isEmpty(),
"updateCallRouting() no telephony TX device");
sp<DeviceDescriptor> txSinkDeviceDesc = deviceList.itemAt(0);
txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
SortedVector<audio_io_handle_t> outputs =
getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs);
audio_io_handle_t output = selectOutput(outputs,
AUDIO_OUTPUT_FLAG_NONE,
AUDIO_FORMAT_INVALID);
// request to reuse existing output stream if one is already opened to reach the TX
// path output device
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ALOG_ASSERT(!outputDesc->isDuplicated(),
"updateCallRouting() RX device output is duplicated");
outputDesc->toAudioPortConfig(&patch.sources[1]);
patch.num_sources = 2;
}
afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch",
status);
if (status == NO_ERROR) {
mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
&patch, mUidCached);
mCallTxPatch->mAfPatchHandle = afPatchHandle;
mCallTxPatch->mUid = mUidCached;
}
}
}
void AudioPolicyManager::setPhoneState(audio_mode_t state)
{
ALOGV("setPhoneState() state %d", state);
if (state < 0 || state >= AUDIO_MODE_CNT) {
ALOGW("setPhoneState() invalid state %d", state);
return;
}
if (state == mPhoneState ) {
ALOGW("setPhoneState() setting same state %d", state);
return;
}
// if leaving call state, handle special case of active streams
// pertaining to sonification strategy see handleIncallSonification()
if (isInCall()) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
if (stream == AUDIO_STREAM_PATCH) {
continue;
}
handleIncallSonification((audio_stream_type_t)stream, false, true);
}
// force reevaluating accessibility routing when call starts
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
// store previous phone state for management of sonification strategy below
int oldState = mPhoneState;
mPhoneState = state;
bool force = false;
// are we entering or starting a call
if (!isStateInCall(oldState) && isStateInCall(state)) {
ALOGV(" Entering call in setPhoneState()");
// force routing command to audio hardware when starting a call
// even if no device change is needed
force = true;
for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
}
} else if (isStateInCall(oldState) && !isStateInCall(state)) {
ALOGV(" Exiting call in setPhoneState()");
// force routing command to audio hardware when exiting a call
// even if no device change is needed
force = true;
for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
sVolumeProfiles[AUDIO_STREAM_DTMF][j];
}
} else if (isStateInCall(state) && (state != oldState)) {
ALOGV(" Switching between telephony and VoIP in setPhoneState()");
// force routing command to audio hardware when switching between telephony and VoIP
// even if no device change is needed
force = true;
}
// check for device and output changes triggered by new phone state
checkA2dpSuspend();
checkOutputForAllStrategies();
updateDevicesAndOutputs();
sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
int delayMs = 0;
if (isStateInCall(state)) {
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
// mute media and sonification strategies and delay device switch by the largest
// latency of any output where either strategy is active.
// This avoid sending the ring tone or music tail into the earpiece or headset.
if ((desc->isStrategyActive(STRATEGY_MEDIA,
SONIFICATION_HEADSET_MUSIC_DELAY,
sysTime) ||
desc->isStrategyActive(STRATEGY_SONIFICATION,
SONIFICATION_HEADSET_MUSIC_DELAY,
sysTime)) &&
(delayMs < (int)desc->mLatency*2)) {
delayMs = desc->mLatency*2;
}
setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
}
}
// Note that despite the fact that getNewOutputDevice() is called on the primary output,
// the device returned is not necessarily reachable via this output
audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
// force routing command to audio hardware when ending call
// even if no device change is needed
if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
rxDevice = hwOutputDesc->device();
}
if (state == AUDIO_MODE_IN_CALL) {
updateCallRouting(rxDevice, delayMs);
} else if (oldState == AUDIO_MODE_IN_CALL) {
if (mCallRxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
mCallRxPatch.clear();
}
if (mCallTxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
mCallTxPatch.clear();
}
setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
} else {
setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
}
// if entering in call state, handle special case of active streams
// pertaining to sonification strategy see handleIncallSonification()
if (isStateInCall(state)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
if (stream == AUDIO_STREAM_PATCH) {
continue;
}
handleIncallSonification((audio_stream_type_t)stream, true, true);
}
}
// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
if (state == AUDIO_MODE_RINGTONE &&
isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
mLimitRingtoneVolume = true;
} else {
mLimitRingtoneVolume = false;
}
}
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, mPhoneState);
bool forceVolumeReeval = false;
switch(usage) {
case AUDIO_POLICY_FORCE_FOR_COMMUNICATION:
if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO &&
config != AUDIO_POLICY_FORCE_NONE) {
ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
return;
}
forceVolumeReeval = true;
mForceUse[usage] = config;
break;
case AUDIO_POLICY_FORCE_FOR_MEDIA:
if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP &&
config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) {
ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
return;
}
mForceUse[usage] = config;
break;
case AUDIO_POLICY_FORCE_FOR_RECORD:
if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
config != AUDIO_POLICY_FORCE_NONE) {
ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
return;
}
mForceUse[usage] = config;
break;
case AUDIO_POLICY_FORCE_FOR_DOCK:
if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK &&
config != AUDIO_POLICY_FORCE_BT_DESK_DOCK &&
config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) {
ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
}
forceVolumeReeval = true;
mForceUse[usage] = config;
break;
case AUDIO_POLICY_FORCE_FOR_SYSTEM:
if (config != AUDIO_POLICY_FORCE_NONE &&
config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
}
forceVolumeReeval = true;
mForceUse[usage] = config;
break;
case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO:
if (config != AUDIO_POLICY_FORCE_NONE &&
config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) {
ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config);
}
mForceUse[usage] = config;
break;
default:
ALOGW("setForceUse() invalid usage %d", usage);
break;
}
// check for device and output changes triggered by new force usage
checkA2dpSuspend();
checkOutputForAllStrategies();
updateDevicesAndOutputs();
if (mPhoneState == AUDIO_MODE_IN_CALL) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
updateCallRouting(newDevice);
}
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_io_handle_t output = mOutputs.keyAt(i);
audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/);
if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
}
if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
applyStreamVolumes(output, newDevice, 0, true);
}
}
audio_io_handle_t activeInput = getActiveInput();
if (activeInput != 0) {
setInputDevice(activeInput, getNewInputDevice(activeInput));
}
}
audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
{
return mForceUse[usage];
}
void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
{
ALOGV("setSystemProperty() property %s, value %s", property, value);
}
// Find a direct output profile compatible with the parameters passed, even if the input flags do
// not explicitly request a direct output
sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getProfileForDirectOutput(
audio_devices_t device,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags)
{
for (size_t i = 0; i < mHwModules.size(); i++) {
if (mHwModules[i]->mHandle == 0) {
continue;
}
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
bool found = profile->isCompatibleProfile(device, String8(""), samplingRate,
NULL /*updatedSamplingRate*/, format, channelMask,
flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ?
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT);
if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) {
return profile;
}
}
}
return 0;
}
audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
routing_strategy strategy = getStrategy(stream);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
device, stream, samplingRate, format, channelMask, flags);
return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE,
stream, samplingRate,format, channelMask,
flags, offloadInfo);
}
status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
audio_attributes_t attributes;
if (attr != NULL) {
if (!isValidAttributes(attr)) {
ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
attr->usage, attr->content_type, attr->flags,
attr->tags);
return BAD_VALUE;
}
attributes = *attr;
} else {
if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) {
ALOGE("getOutputForAttr(): invalid stream type");
return BAD_VALUE;
}
stream_type_to_audio_attributes(*stream, &attributes);
}
for (size_t i = 0; i < mPolicyMixes.size(); i++) {
sp<AudioOutputDescriptor> desc;
if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) {
for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) ||
(RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) {
desc = mPolicyMixes[i]->mOutput;
break;
}
if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
strncmp(attributes.tags + strlen("addr="),
mPolicyMixes[i]->mMix.mRegistrationId.string(),
AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
desc = mPolicyMixes[i]->mOutput;
break;
}
}
} else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) {
if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE &&
strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
strncmp(attributes.tags + strlen("addr="),
mPolicyMixes[i]->mMix.mRegistrationId.string(),
AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
desc = mPolicyMixes[i]->mOutput;
}
}
if (desc != 0) {
if (!audio_is_linear_pcm(format)) {
return BAD_VALUE;
}
desc->mPolicyMix = &mPolicyMixes[i]->mMix;
*stream = streamTypefromAttributesInt(&attributes);
*output = desc->mIoHandle;
ALOGV("getOutputForAttr() returns output %d", *output);
return NO_ERROR;
}
}
if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
return BAD_VALUE;
}
ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x",
attributes.usage, attributes.content_type, attributes.tags, attributes.flags);
routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
}
ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x",
device, samplingRate, format, channelMask, flags);
*stream = streamTypefromAttributesInt(&attributes);
*output = getOutputForDevice(device, session, *stream,
samplingRate, format, channelMask,
flags, offloadInfo);
if (*output == AUDIO_IO_HANDLE_NONE) {
return INVALID_OPERATION;
}
return NO_ERROR;
}
audio_io_handle_t AudioPolicyManager::getOutputForDevice(
audio_devices_t device,
audio_session_t session __unused,
audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
uint32_t latency = 0;
status_t status;
#ifdef AUDIO_POLICY_TEST
if (mCurOutput != 0) {
ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
if (mTestOutputs[mCurOutput] == 0) {
ALOGV("getOutput() opening test output");
sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
outputDesc->mDevice = mTestDevice;
outputDesc->mLatency = mTestLatencyMs;
outputDesc->mFlags =
(audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
outputDesc->mRefCount[stream] = 0;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = mTestSamplingRate;
config.channel_mask = mTestChannels;
config.format = mTestFormat;
if (offloadInfo != NULL) {
config.offload_info = *offloadInfo;
}
status = mpClientInterface->openOutput(0,
&mTestOutputs[mCurOutput],
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
if (status == NO_ERROR) {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mFormat = config.format;
outputDesc->mChannelMask = config.channel_mask;
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"),mCurOutput);
mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
addOutput(mTestOutputs[mCurOutput], outputDesc);
}
}
return mTestOutputs[mCurOutput];
}
#endif //AUDIO_POLICY_TEST
// open a direct output if required by specified parameters
//force direct flag if offload flag is set: offloading implies a direct output stream
// and all common behaviors are driven by checking only the direct flag
// this should normally be set appropriately in the policy configuration file
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
// only allow deep buffering for music stream type
if (stream != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
sp<IOProfile> profile;
// skip direct output selection if the request can obviously be attached to a mixed output
// and not explicitly requested
if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE &&
audio_channel_count_from_out_mask(channelMask) <= 2) {
goto non_direct_output;
}
// Do not allow offloading if one non offloadable effect is enabled. This prevents from
// creating an offloaded track and tearing it down immediately after start when audioflinger
// detects there is an active non offloadable effect.
// FIXME: We should check the audio session here but we do not have it in this context.
// This may prevent offloading in rare situations where effects are left active by apps
// in the background.
if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
!isNonOffloadableEffectEnabled()) {
profile = getProfileForDirectOutput(device,
samplingRate,
format,
channelMask,
(audio_output_flags_t)flags);
}
if (profile != 0) {
sp<AudioOutputDescriptor> outputDesc = NULL;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
outputDesc = desc;
// reuse direct output if currently open and configured with same parameters
if ((samplingRate == outputDesc->mSamplingRate) &&
(format == outputDesc->mFormat) &&
(channelMask == outputDesc->mChannelMask)) {
outputDesc->mDirectOpenCount++;
ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
return mOutputs.keyAt(i);
}
}
}
// close direct output if currently open and configured with different parameters
if (outputDesc != NULL) {
closeOutput(outputDesc->mIoHandle);
}
outputDesc = new AudioOutputDescriptor(profile);
outputDesc->mDevice = device;
outputDesc->mLatency = 0;
outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = samplingRate;
config.channel_mask = channelMask;
config.format = format;
if (offloadInfo != NULL) {
config.offload_info = *offloadInfo;
}
status = mpClientInterface->openOutput(profile->mModule->mHandle,
&output,
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
// only accept an output with the requested parameters
if (status != NO_ERROR ||
(samplingRate != 0 && samplingRate != config.sample_rate) ||
(format != AUDIO_FORMAT_DEFAULT && format != config.format) ||
(channelMask != 0 && channelMask != config.channel_mask)) {
ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
"format %d %d, channelMask %04x %04x", output, samplingRate,
outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
outputDesc->mChannelMask);
if (output != AUDIO_IO_HANDLE_NONE) {
mpClientInterface->closeOutput(output);
}
// fall back to mixer output if possible when the direct output could not be open
if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
goto non_direct_output;
}
return AUDIO_IO_HANDLE_NONE;
}
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
outputDesc->mRefCount[stream] = 0;
outputDesc->mStopTime[stream] = 0;
outputDesc->mDirectOpenCount = 1;
audio_io_handle_t srcOutput = getOutputForEffect();
addOutput(output, outputDesc);
audio_io_handle_t dstOutput = getOutputForEffect();
if (dstOutput == output) {
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
}
mPreviousOutputs = mOutputs;
ALOGV("getOutput() returns new direct output %d", output);
mpClientInterface->onAudioPortListUpdate();
return output;
}
non_direct_output:
// ignoring channel mask due to downmix capability in mixer
// open a non direct output
// for non direct outputs, only PCM is supported
if (audio_is_linear_pcm(format)) {
// get which output is suitable for the specified stream. The actual
// routing change will happen when startOutput() will be called
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
// at this stage we should ignore the DIRECT flag as no direct output could be found earlier
flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
output = selectOutput(outputs, flags, format);
}
ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
"format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
ALOGV("getOutput() returns output %d", output);
return output;
}
audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
audio_output_flags_t flags,
audio_format_t format)
{
// select one output among several that provide a path to a particular device or set of
// devices (the list was previously build by getOutputsForDevice()).
// The priority is as follows:
// 1: the output with the highest number of requested policy flags
// 2: the primary output
// 3: the first output in the list
if (outputs.size() == 0) {
return 0;
}
if (outputs.size() == 1) {
return outputs[0];
}
int maxCommonFlags = 0;
audio_io_handle_t outputFlags = 0;
audio_io_handle_t outputPrimary = 0;
for (size_t i = 0; i < outputs.size(); i++) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
if (!outputDesc->isDuplicated()) {
// if a valid format is specified, skip output if not compatible
if (format != AUDIO_FORMAT_INVALID) {
if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (format != outputDesc->mFormat) {
continue;
}
} else if (!audio_is_linear_pcm(format)) {
continue;
}
}
int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
if (commonFlags > maxCommonFlags) {
outputFlags = outputs[i];
maxCommonFlags = commonFlags;
ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
}
if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
outputPrimary = outputs[i];
}
}
}
if (outputFlags != 0) {
return outputFlags;
}
if (outputPrimary != 0) {
return outputPrimary;
}
return outputs[0];
}
status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
audio_session_t session)
{
ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
ALOGW("startOutput() unknown output %d", output);
return BAD_VALUE;
}
// cannot start playback of STREAM_TTS if any other output is being used
uint32_t beaconMuteLatency = 0;
if (stream == AUDIO_STREAM_TTS) {
ALOGV("\t found BEACON stream");
if (isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
return INVALID_OPERATION;
} else {
beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
}
} else {
// some playback other than beacon starts
beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
}
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
// increment usage count for this stream on the requested output:
// NOTE that the usage count is the same for duplicated output and hardware output which is
// necessary for a correct control of hardware output routing by startOutput() and stopOutput()
outputDesc->changeRefCount(stream, 1);
if (outputDesc->mRefCount[stream] == 1) {
// starting an output being rerouted?
audio_devices_t newDevice;
if (outputDesc->mPolicyMix != NULL) {
newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
} else {
newDevice = getNewOutputDevice(output, false /*fromCache*/);
}
routing_strategy strategy = getStrategy(stream);
bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
(strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
(beaconMuteLatency > 0);
uint32_t waitMs = beaconMuteLatency;
bool force = false;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc) {
// force a device change if any other output is managed by the same hw
// module and has a current device selection that differs from selected device.
// In this case, the audio HAL must receive the new device selection so that it can
// change the device currently selected by the other active output.
if (outputDesc->sharesHwModuleWith(desc) &&
desc->device() != newDevice) {
force = true;
}
// wait for audio on other active outputs to be presented when starting
// a notification so that audio focus effect can propagate, or that a mute/unmute
// event occurred for beacon
uint32_t latency = desc->latency();
if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
waitMs = latency;
}
}
}
uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
// handle special case for sonification while in call
if (isInCall()) {
handleIncallSonification(stream, true, false);
}
// apply volume rules for current stream and device if necessary
checkAndSetVolume(stream,
mStreams[stream].getVolumeIndex(newDevice),
output,
newDevice);
// update the outputs if starting an output with a stream that can affect notification
// routing
handleNotificationRoutingForStream(stream);
// Automatically enable the remote submix input when output is started on a re routing mix
// of type MIX_TYPE_RECORDERS
if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL &&
outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
outputDesc->mPolicyMix->mRegistrationId);
}
// force reevaluating accessibility routing when ringtone or alarm starts
if (strategy == STRATEGY_SONIFICATION) {
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
if (waitMs > muteWaitMs) {
usleep((waitMs - muteWaitMs) * 2 * 1000);
}
}
return NO_ERROR;
}
status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
audio_session_t session)
{
ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
ALOGW("stopOutput() unknown output %d", output);
return BAD_VALUE;
}
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
// always handle stream stop, check which stream type is stopping
handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
// handle special case for sonification while in call
if (isInCall()) {
handleIncallSonification(stream, false, false);
}
if (outputDesc->mRefCount[stream] > 0) {
// decrement usage count of this stream on the output
outputDesc->changeRefCount(stream, -1);
// store time at which the stream was stopped - see isStreamActive()
if (outputDesc->mRefCount[stream] == 0) {
// Automatically disable the remote submix input when output is stopped on a
// re routing mix of type MIX_TYPE_RECORDERS
if (audio_is_remote_submix_device(outputDesc->mDevice) &&
outputDesc->mPolicyMix != NULL &&
outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
outputDesc->mPolicyMix->mRegistrationId);
}
outputDesc->mStopTime[stream] = systemTime();
audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
// delay the device switch by twice the latency because stopOutput() is executed when
// the track stop() command is received and at that time the audio track buffer can
// still contain data that needs to be drained. The latency only covers the audio HAL
// and kernel buffers. Also the latency does not always include additional delay in the
// audio path (audio DSP, CODEC ...)
setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
// force restoring the device selection on other active outputs if it differs from the
// one being selected for this output
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_io_handle_t curOutput = mOutputs.keyAt(i);
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (curOutput != output &&
desc->isActive() &&
outputDesc->sharesHwModuleWith(desc) &&
(newDevice != desc->device())) {
setOutputDevice(curOutput,
getNewOutputDevice(curOutput, false /*fromCache*/),
true,
outputDesc->mLatency*2);
}
}
// update the outputs if stopping one with a stream that can affect notification routing
handleNotificationRoutingForStream(stream);
}
return NO_ERROR;
} else {
ALOGW("stopOutput() refcount is already 0 for output %d", output);
return INVALID_OPERATION;
}
}
void AudioPolicyManager::releaseOutput(audio_io_handle_t output,
audio_stream_type_t stream __unused,
audio_session_t session __unused)
{
ALOGV("releaseOutput() %d", output);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
ALOGW("releaseOutput() releasing unknown output %d", output);
return;
}
#ifdef AUDIO_POLICY_TEST
int testIndex = testOutputIndex(output);
if (testIndex != 0) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
if (outputDesc->isActive()) {
mpClientInterface->closeOutput(output);
mOutputs.removeItem(output);
mTestOutputs[testIndex] = 0;
}
return;
}
#endif //AUDIO_POLICY_TEST
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index);
if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
if (desc->mDirectOpenCount <= 0) {
ALOGW("releaseOutput() invalid open count %d for output %d",
desc->mDirectOpenCount, output);
return;
}
if (--desc->mDirectOpenCount == 0) {
closeOutput(output);
// If effects where present on the output, audioflinger moved them to the primary
// output by default: move them back to the appropriate output.
audio_io_handle_t dstOutput = getOutputForEffect();
if (dstOutput != mPrimaryOutput) {
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
}
mpClientInterface->onAudioPortListUpdate();
}
}
}
status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_input_flags_t flags,
input_type_t *inputType)
{
ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x,"
"session %d, flags %#x",
attr->source, samplingRate, format, channelMask, session, flags);
*input = AUDIO_IO_HANDLE_NONE;
*inputType = API_INPUT_INVALID;
audio_devices_t device;
// handle legacy remote submix case where the address was not always specified
String8 address = String8("");
bool isSoundTrigger = false;
audio_source_t inputSource = attr->source;
audio_source_t halInputSource;
AudioMix *policyMix = NULL;
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
halInputSource = inputSource;
if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
address = String8(attr->tags + strlen("addr="));
ssize_t index = mPolicyMixes.indexOfKey(address);
if (index < 0) {
ALOGW("getInputForAttr() no policy for address %s", address.string());
return BAD_VALUE;
}
if (mPolicyMixes[index]->mMix.mMixType != MIX_TYPE_PLAYERS) {
ALOGW("getInputForAttr() bad policy mix type for address %s", address.string());
return BAD_VALUE;
}
policyMix = &mPolicyMixes[index]->mMix;
*inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
} else {
device = getDeviceAndMixForInputSource(inputSource, &policyMix);
if (device == AUDIO_DEVICE_NONE) {
ALOGW("getInputForAttr() could not find device for source %d", inputSource);
return BAD_VALUE;
}
if (policyMix != NULL) {
address = policyMix->mRegistrationId;
if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
// there is an external policy, but this input is attached to a mix of recorders,
// meaning it receives audio injected into the framework, so the recorder doesn't
// know about it and is therefore considered "legacy"
*inputType = API_INPUT_LEGACY;
} else {
// recording a mix of players defined by an external policy, we're rerouting for
// an external policy
*inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
}
} else if (audio_is_remote_submix_device(device)) {
address = String8("0");
*inputType = API_INPUT_MIX_CAPTURE;
} else {
*inputType = API_INPUT_LEGACY;
}
// adapt channel selection to input source
switch (inputSource) {
case AUDIO_SOURCE_VOICE_UPLINK:
channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
break;
case AUDIO_SOURCE_VOICE_DOWNLINK:
channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
break;
case AUDIO_SOURCE_VOICE_CALL:
channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
break;
default:
break;
}
if (inputSource == AUDIO_SOURCE_HOTWORD) {
ssize_t index = mSoundTriggerSessions.indexOfKey(session);
if (index >= 0) {
*input = mSoundTriggerSessions.valueFor(session);
isSoundTrigger = true;
flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
ALOGV("SoundTrigger capture on session %d input %d", session, *input);
} else {
halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
}
}
}
sp<IOProfile> profile = getInputProfile(device, address,
samplingRate, format, channelMask,
flags);
if (profile == 0) {
//retry without flags
audio_input_flags_t log_flags = flags;
flags = AUDIO_INPUT_FLAG_NONE;
profile = getInputProfile(device, address,
samplingRate, format, channelMask,
flags);
if (profile == 0) {
ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u,"
"format %#x, channelMask 0x%X, flags %#x",
device, samplingRate, format, channelMask, log_flags);
return BAD_VALUE;
}
}
if (profile->mModule->mHandle == 0) {
ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName);
return NO_INIT;
}
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = samplingRate;
config.channel_mask = channelMask;
config.format = format;
status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
input,
&config,
&device,
address,
halInputSource,
flags);
// only accept input with the exact requested set of parameters
if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE ||
(samplingRate != config.sample_rate) ||
(format != config.format) ||
(channelMask != config.channel_mask)) {
ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x",
samplingRate, format, channelMask);
if (*input != AUDIO_IO_HANDLE_NONE) {
mpClientInterface->closeInput(*input);
}
return BAD_VALUE;
}
sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
inputDesc->mInputSource = inputSource;
inputDesc->mRefCount = 0;
inputDesc->mOpenRefCount = 1;
inputDesc->mSamplingRate = samplingRate;
inputDesc->mFormat = format;
inputDesc->mChannelMask = channelMask;
inputDesc->mDevice = device;
inputDesc->mSessions.add(session);
inputDesc->mIsSoundTrigger = isSoundTrigger;
inputDesc->mPolicyMix = policyMix;
ALOGV("getInputForAttr() returns input type = %d", inputType);
addInput(*input, inputDesc);
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
}
status_t AudioPolicyManager::startInput(audio_io_handle_t input,
audio_session_t session)
{
ALOGV("startInput() input %d", input);
ssize_t index = mInputs.indexOfKey(input);
if (index < 0) {
ALOGW("startInput() unknown input %d", input);
return BAD_VALUE;
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
index = inputDesc->mSessions.indexOf(session);
if (index < 0) {
ALOGW("startInput() unknown session %d on input %d", session, input);
return BAD_VALUE;
}
// virtual input devices are compatible with other input devices
if (!isVirtualInputDevice(inputDesc->mDevice)) {
// for a non-virtual input device, check if there is another (non-virtual) active input
audio_io_handle_t activeInput = getActiveInput();
if (activeInput != 0 && activeInput != input) {
// If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
// otherwise the active input continues and the new input cannot be started.
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
stopInput(activeInput, activeDesc->mSessions.itemAt(0));
releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
} else {
ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
return INVALID_OPERATION;
}
}
}
if (inputDesc->mRefCount == 0) {
if (activeInputsCount() == 0) {
SoundTrigger::setCaptureState(true);
}
setInputDevice(input, getNewInputDevice(input), true /* force */);
// automatically enable the remote submix output when input is started if not
// used by a policy mix of type MIX_TYPE_RECORDERS
// For remote submix (a virtual device), we open only one input per capture request.
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
String8 address = String8("");
if (inputDesc->mPolicyMix == NULL) {
address = String8("0");
} else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
address = inputDesc->mPolicyMix->mRegistrationId;
}
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address);
}
}
}
ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
inputDesc->mRefCount++;
return NO_ERROR;
}
status_t AudioPolicyManager::stopInput(audio_io_handle_t input,
audio_session_t session)
{
ALOGV("stopInput() input %d", input);
ssize_t index = mInputs.indexOfKey(input);
if (index < 0) {
ALOGW("stopInput() unknown input %d", input);
return BAD_VALUE;
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
index = inputDesc->mSessions.indexOf(session);
if (index < 0) {
ALOGW("stopInput() unknown session %d on input %d", session, input);
return BAD_VALUE;
}
if (inputDesc->mRefCount == 0) {
ALOGW("stopInput() input %d already stopped", input);
return INVALID_OPERATION;
}
inputDesc->mRefCount--;
if (inputDesc->mRefCount == 0) {
// automatically disable the remote submix output when input is stopped if not
// used by a policy mix of type MIX_TYPE_RECORDERS
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
String8 address = String8("");
if (inputDesc->mPolicyMix == NULL) {
address = String8("0");
} else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
address = inputDesc->mPolicyMix->mRegistrationId;
}
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address);
}
}
resetInputDevice(input);
if (activeInputsCount() == 0) {
SoundTrigger::setCaptureState(false);
}
}
return NO_ERROR;
}
void AudioPolicyManager::releaseInput(audio_io_handle_t input,
audio_session_t session)
{
ALOGV("releaseInput() %d", input);
ssize_t index = mInputs.indexOfKey(input);
if (index < 0) {
ALOGW("releaseInput() releasing unknown input %d", input);
return;
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
ALOG_ASSERT(inputDesc != 0);
index = inputDesc->mSessions.indexOf(session);
if (index < 0) {
ALOGW("releaseInput() unknown session %d on input %d", session, input);
return;
}
inputDesc->mSessions.remove(session);
if (inputDesc->mOpenRefCount == 0) {
ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount);
return;
}
inputDesc->mOpenRefCount--;
if (inputDesc->mOpenRefCount > 0) {
ALOGV("releaseInput() exit > 0");
return;
}
closeInput(input);
mpClientInterface->onAudioPortListUpdate();
ALOGV("releaseInput() exit");
}
void AudioPolicyManager::closeAllInputs() {
bool patchRemoved = false;
for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index);
ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
if (patch_index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
mAudioPatches.removeItemsAt(patch_index);
patchRemoved = true;
}
mpClientInterface->closeInput(mInputs.keyAt(input_index));
}
mInputs.clear();
nextAudioPortGeneration();
if (patchRemoved) {
mpClientInterface->onAudioPatchListUpdate();
}
}
void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax)
{
ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
if (indexMin < 0 || indexMin >= indexMax) {
ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
return;
}
mStreams[stream].mIndexMin = indexMin;
mStreams[stream].mIndexMax = indexMax;
//FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
if (stream == AUDIO_STREAM_MUSIC) {
mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMin = indexMin;
mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMax = indexMax;
}
}
status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
int index,
audio_devices_t device)
{
if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
return BAD_VALUE;
}
if (!audio_is_output_device(device)) {
return BAD_VALUE;
}
// Force max volume if stream cannot be muted
if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
stream, device, index);
// if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
// clear all device specific values
if (device == AUDIO_DEVICE_OUT_DEFAULT) {
mStreams[stream].mIndexCur.clear();
}
mStreams[stream].mIndexCur.add(device, index);
// update volume on all outputs whose current device is also selected by the same
// strategy as the device specified by the caller
audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
//FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE;
if (stream == AUDIO_STREAM_MUSIC) {
mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexCur.add(device, index);
accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/);
}
if ((device != AUDIO_DEVICE_OUT_DEFAULT) &&
(device & (strategyDevice | accessibilityDevice)) == 0) {
return NO_ERROR;
}
status_t status = NO_ERROR;
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_devices_t curDevice =
getDeviceForVolume(mOutputs.valueAt(i)->device());
if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
if (volStatus != NO_ERROR) {
status = volStatus;
}
}
if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) {
status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY,
index, mOutputs.keyAt(i), curDevice);
}
}
return status;
}
status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
int *index,
audio_devices_t device)
{
if (index == NULL) {
return BAD_VALUE;
}
if (!audio_is_output_device(device)) {
return BAD_VALUE;
}
// if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
// the strategy the stream belongs to.
if (device == AUDIO_DEVICE_OUT_DEFAULT) {
device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
}
device = getDeviceForVolume(device);
*index = mStreams[stream].getVolumeIndex(device);
ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
return NO_ERROR;
}
audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
const SortedVector<audio_io_handle_t>& outputs)
{
// select one output among several suitable for global effects.
// The priority is as follows:
// 1: An offloaded output. If the effect ends up not being offloadable,
// AudioFlinger will invalidate the track and the offloaded output
// will be closed causing the effect to be moved to a PCM output.
// 2: A deep buffer output
// 3: the first output in the list
if (outputs.size() == 0) {
return 0;
}
audio_io_handle_t outputOffloaded = 0;
audio_io_handle_t outputDeepBuffer = 0;
for (size_t i = 0; i < outputs.size(); i++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
outputOffloaded = outputs[i];
}
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
outputDeepBuffer = outputs[i];
}
}
ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
outputOffloaded, outputDeepBuffer);
if (outputOffloaded != 0) {
return outputOffloaded;
}
if (outputDeepBuffer != 0) {
return outputDeepBuffer;
}
return outputs[0];
}
audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
{
// apply simple rule where global effects are attached to the same output as MUSIC streams
routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
audio_io_handle_t output = selectOutputForEffects(dstOutputs);
ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
return output;
}
status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
audio_io_handle_t io,
uint32_t strategy,
int session,
int id)
{
ssize_t index = mOutputs.indexOfKey(io);
if (index < 0) {
index = mInputs.indexOfKey(io);
if (index < 0) {
ALOGW("registerEffect() unknown io %d", io);
return INVALID_OPERATION;
}
}
if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
desc->name, desc->memoryUsage);
return INVALID_OPERATION;
}
mTotalEffectsMemory += desc->memoryUsage;
ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
desc->name, io, strategy, session, id);
ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
sp<EffectDescriptor> effectDesc = new EffectDescriptor();
memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t));
effectDesc->mIo = io;
effectDesc->mStrategy = (routing_strategy)strategy;
effectDesc->mSession = session;
effectDesc->mEnabled = false;
mEffects.add(id, effectDesc);
return NO_ERROR;
}
status_t AudioPolicyManager::unregisterEffect(int id)
{
ssize_t index = mEffects.indexOfKey(id);
if (index < 0) {
ALOGW("unregisterEffect() unknown effect ID %d", id);
return INVALID_OPERATION;
}
sp<EffectDescriptor> effectDesc = mEffects.valueAt(index);
setEffectEnabled(effectDesc, false);
if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) {
ALOGW("unregisterEffect() memory %d too big for total %d",
effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
effectDesc->mDesc.memoryUsage = mTotalEffectsMemory;
}
mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage;
ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
mEffects.removeItem(id);
return NO_ERROR;
}
status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled)
{
ssize_t index = mEffects.indexOfKey(id);
if (index < 0) {
ALOGW("unregisterEffect() unknown effect ID %d", id);
return INVALID_OPERATION;
}
return setEffectEnabled(mEffects.valueAt(index), enabled);
}
status_t AudioPolicyManager::setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled)
{
if (enabled == effectDesc->mEnabled) {
ALOGV("setEffectEnabled(%s) effect already %s",
enabled?"true":"false", enabled?"enabled":"disabled");
return INVALID_OPERATION;
}
if (enabled) {
if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10);
return INVALID_OPERATION;
}
mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad;
ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
} else {
if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) {
ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
}
mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad;
ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
}
effectDesc->mEnabled = enabled;
return NO_ERROR;
}
bool AudioPolicyManager::isNonOffloadableEffectEnabled()
{
for (size_t i = 0; i < mEffects.size(); i++) {
sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) &&
((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
effectDesc->mDesc.name, effectDesc->mSession);
return true;
}
}
return false;
}
bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
{
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < mOutputs.size(); i++) {
const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
return true;
}
}
return false;
}
bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream,
uint32_t inPastMs) const
{
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < mOutputs.size(); i++) {
const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
// do not consider re routing (when the output is going to a dynamic policy)
// as "remote playback"
if (outputDesc->mPolicyMix == NULL) {
return true;
}
}
}
return false;
}
bool AudioPolicyManager::isSourceActive(audio_source_t source) const
{
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
if (inputDescriptor->mRefCount == 0) {
continue;
}
if (inputDescriptor->mInputSource == (int)source) {
return true;
}
// AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it
// corresponds to an active capture triggered by a hardware hotword recognition
if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) &&
(inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) {
// FIXME: we should not assume that the first session is the active one and keep
// activity count per session. Same in startInput().
ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0));
if (index >= 0) {
return true;
}
}
}
return false;
}
// Register a list of custom mixes with their attributes and format.
// When a mix is registered, corresponding input and output profiles are
// added to the remote submix hw module. The profile contains only the
// parameters (sampling rate, format...) specified by the mix.
// The corresponding input remote submix device is also connected.
//
// When a remote submix device is connected, the address is checked to select the
// appropriate profile and the corresponding input or output stream is opened.
//
// When capture starts, getInputForAttr() will:
// - 1 look for a mix matching the address passed in attribtutes tags if any
// - 2 if none found, getDeviceForInputSource() will:
// - 2.1 look for a mix matching the attributes source
// - 2.2 if none found, default to device selection by policy rules
// At this time, the corresponding output remote submix device is also connected
// and active playback use cases can be transferred to this mix if needed when reconnecting
// after AudioTracks are invalidated
//
// When playback starts, getOutputForAttr() will:
// - 1 look for a mix matching the address passed in attribtutes tags if any
// - 2 if none found, look for a mix matching the attributes usage
// - 3 if none found, default to device and output selection by policy rules.
status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes)
{
sp<HwModule> module;
for (size_t i = 0; i < mHwModules.size(); i++) {
if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
mHwModules[i]->mHandle != 0) {
module = mHwModules[i];
break;
}
}
if (module == 0) {
return INVALID_OPERATION;
}
ALOGV("registerPolicyMixes() num mixes %d", mixes.size());
for (size_t i = 0; i < mixes.size(); i++) {
String8 address = mixes[i].mRegistrationId;
ssize_t index = mPolicyMixes.indexOfKey(address);
if (index >= 0) {
ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string());
continue;
}
audio_config_t outputConfig = mixes[i].mFormat;
audio_config_t inputConfig = mixes[i].mFormat;
// NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
// stereo and let audio flinger do the channel conversion if needed.
outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
module->addOutputProfile(address, &outputConfig,
AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
module->addInputProfile(address, &inputConfig,
AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
sp<AudioPolicyMix> policyMix = new AudioPolicyMix();
policyMix->mMix = mixes[i];
mPolicyMixes.add(address, policyMix);
if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.string());
} else {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.string());
}
}
return NO_ERROR;
}
status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes)
{
sp<HwModule> module;
for (size_t i = 0; i < mHwModules.size(); i++) {
if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
mHwModules[i]->mHandle != 0) {
module = mHwModules[i];
break;
}
}
if (module == 0) {
return INVALID_OPERATION;
}
ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size());
for (size_t i = 0; i < mixes.size(); i++) {
String8 address = mixes[i].mRegistrationId;
ssize_t index = mPolicyMixes.indexOfKey(address);
if (index < 0) {
ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string());
continue;
}
mPolicyMixes.removeItemsAt(index);
if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) ==
AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
{
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.string());
}
if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
{
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.string());
}
module->removeOutputProfile(address);
module->removeInputProfile(address);
}
return NO_ERROR;
}
status_t AudioPolicyManager::dump(int fd)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
result.append(buffer);
snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
result.append(buffer);
snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
result.append(buffer);
snprintf(buffer, SIZE, " Force use for communications %d\n",
mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
result.append(buffer);
snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]);
result.append(buffer);
snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]);
result.append(buffer);
snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]);
result.append(buffer);
snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]);
result.append(buffer);
snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n",
mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO]);
result.append(buffer);
snprintf(buffer, SIZE, " Available output devices:\n");
result.append(buffer);
write(fd, result.string(), result.size());
for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
mAvailableOutputDevices[i]->dump(fd, 2, i);
}
snprintf(buffer, SIZE, "\n Available input devices:\n");
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
mAvailableInputDevices[i]->dump(fd, 2, i);
}
snprintf(buffer, SIZE, "\nHW Modules dump:\n");
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mHwModules.size(); i++) {
snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1);
write(fd, buffer, strlen(buffer));
mHwModules[i]->dump(fd);
}
snprintf(buffer, SIZE, "\nOutputs dump:\n");
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mOutputs.size(); i++) {
snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
write(fd, buffer, strlen(buffer));
mOutputs.valueAt(i)->dump(fd);
}
snprintf(buffer, SIZE, "\nInputs dump:\n");
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mInputs.size(); i++) {
snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
write(fd, buffer, strlen(buffer));
mInputs.valueAt(i)->dump(fd);
}
snprintf(buffer, SIZE, "\nStreams dump:\n");
write(fd, buffer, strlen(buffer));
snprintf(buffer, SIZE,
" Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) {
snprintf(buffer, SIZE, " %02zu ", i);
write(fd, buffer, strlen(buffer));
mStreams[i].dump(fd);
}
snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
(float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
write(fd, buffer, strlen(buffer));
snprintf(buffer, SIZE, "Registered effects:\n");
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mEffects.size(); i++) {
snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
write(fd, buffer, strlen(buffer));
mEffects.valueAt(i)->dump(fd);
}
snprintf(buffer, SIZE, "\nAudio Patches:\n");
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mAudioPatches.size(); i++) {
mAudioPatches[i]->dump(fd, 2, i);
}
return NO_ERROR;
}
// This function checks for the parameters which can be offloaded.
// This can be enhanced depending on the capability of the DSP and policy
// of the system.
bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
{
ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
" BitRate=%u, duration=%" PRId64 " us, has_video=%d",
offloadInfo.sample_rate, offloadInfo.channel_mask,
offloadInfo.format,
offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
offloadInfo.has_video);
// Check if offload has been disabled
char propValue[PROPERTY_VALUE_MAX];
if (property_get("audio.offload.disable", propValue, "0")) {
if (atoi(propValue) != 0) {
ALOGV("offload disabled by audio.offload.disable=%s", propValue );
return false;
}
}
// Check if stream type is music, then only allow offload as of now.
if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
{
ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
return false;
}
//TODO: enable audio offloading with video when ready
if (offloadInfo.has_video)
{
ALOGV("isOffloadSupported: has_video == true, returning false");
return false;
}
//If duration is less than minimum value defined in property, return false
if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
return false;
}
} else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
return false;
}
// Do not allow offloading if one non offloadable effect is enabled. This prevents from
// creating an offloaded track and tearing it down immediately after start when audioflinger
// detects there is an active non offloadable effect.
// FIXME: We should check the audio session here but we do not have it in this context.
// This may prevent offloading in rare situations where effects are left active by apps
// in the background.
if (isNonOffloadableEffectEnabled()) {
return false;
}
// See if there is a profile to support this.
// AUDIO_DEVICE_NONE
sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
offloadInfo.sample_rate,
offloadInfo.format,
offloadInfo.channel_mask,
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
return (profile != 0);
}
status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role,
audio_port_type_t type,
unsigned int *num_ports,
struct audio_port *ports,
unsigned int *generation)
{
if (num_ports == NULL || (*num_ports != 0 && ports == NULL) ||
generation == NULL) {
return BAD_VALUE;
}
ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports);
if (ports == NULL) {
*num_ports = 0;
}
size_t portsWritten = 0;
size_t portsMax = *num_ports;
*num_ports = 0;
if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) {
if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
for (size_t i = 0;
i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) {
mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]);
}
*num_ports += mAvailableOutputDevices.size();
}
if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
for (size_t i = 0;
i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) {
mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]);
}
*num_ports += mAvailableInputDevices.size();
}
}
if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) {
if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) {
mInputs[i]->toAudioPort(&ports[portsWritten++]);
}
*num_ports += mInputs.size();
}
if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
size_t numOutputs = 0;
for (size_t i = 0; i < mOutputs.size(); i++) {
if (!mOutputs[i]->isDuplicated()) {
numOutputs++;
if (portsWritten < portsMax) {
mOutputs[i]->toAudioPort(&ports[portsWritten++]);
}
}
}
*num_ports += numOutputs;
}
}
*generation = curAudioPortGeneration();
ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports);
return NO_ERROR;
}
status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused)
{
return NO_ERROR;
}
sp<AudioPolicyManager::AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
audio_port_handle_t id) const
{
sp<AudioOutputDescriptor> outputDesc = NULL;
for (size_t i = 0; i < mOutputs.size(); i++) {
outputDesc = mOutputs.valueAt(i);
if (outputDesc->mId == id) {
break;
}
}
return outputDesc;
}
sp<AudioPolicyManager::AudioInputDescriptor> AudioPolicyManager::getInputFromId(
audio_port_handle_t id) const
{
sp<AudioInputDescriptor> inputDesc = NULL;
for (size_t i = 0; i < mInputs.size(); i++) {
inputDesc = mInputs.valueAt(i);
if (inputDesc->mId == id) {
break;
}
}
return inputDesc;
}
sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleForDevice(
audio_devices_t device) const
{
sp <HwModule> module;
for (size_t i = 0; i < mHwModules.size(); i++) {
if (mHwModules[i]->mHandle == 0) {
continue;
}
if (audio_is_output_device(device)) {
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) {
return mHwModules[i];
}
}
} else {
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() &
device & ~AUDIO_DEVICE_BIT_IN) {
return mHwModules[i];
}
}
}
}
return module;
}
sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
{
sp <HwModule> module;
for (size_t i = 0; i < mHwModules.size(); i++)
{
if (strcmp(mHwModules[i]->mName, name) == 0) {
return mHwModules[i];
}
}
return module;
}
audio_devices_t AudioPolicyManager::availablePrimaryOutputDevices()
{
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types();
return devices & mAvailableOutputDevices.types();
}