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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
// 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 <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)
{
String8 address = (device_address == NULL) ? String8("") : String8(device_address);
ALOGV("setDeviceConnectionState() device: %x, state %d, address %s",
device, state, address.string());
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
// handle output devices
if (audio_is_output_device(device)) {
SortedVector <audio_io_handle_t> outputs;
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
devDesc->mAddress = address;
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, address) != 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());
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(address);
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, address);
} 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;
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
devDesc->mAddress = address;
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, address) != NO_ERROR) {
return INVALID_OPERATION;
}
index = mAvailableInputDevices.add(devDesc);
if (index >= 0) {
mAvailableInputDevices[index]->mId = nextUniqueId();
mAvailableInputDevices[index]->mModule = module;
} else {
return NO_MEMORY;
}
} 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(address);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
checkInputsForDevice(device, state, inputs, address);
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)
{
audio_policy_dev_state_t state = AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
devDesc->mAddress = (device_address == NULL) ? String8("") : String8(device_address);
ssize_t index;
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;
}
index = deviceVector->indexOf(devDesc);
if (index >= 0) {
return AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
} else {
return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
}
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 = getDeviceForInputSource(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++) {
handleIncallSonification((audio_stream_type_t)stream, false, true);
}
}
// store previous phone state for management of sonification strategy below
int oldState = mPhoneState;
mPhoneState = state;
bool force = false;
// are we entering or starting a call
if (!isStateInCall(oldState) && isStateInCall(state)) {
ALOGV(" Entering call in setPhoneState()");
// force routing command to audio hardware when starting a call
// even if no device change is needed
force = true;
for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
}
} else if (isStateInCall(oldState) && !isStateInCall(state)) {
ALOGV(" Exiting call in setPhoneState()");
// force routing command to audio hardware when exiting a call
// even if no device change is needed
force = true;
for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
sVolumeProfiles[AUDIO_STREAM_DTMF][j];
}
} else if (isStateInCall(state) && (state != oldState)) {
ALOGV(" Switching between telephony and VoIP in setPhoneState()");
// force routing command to audio hardware when switching between telephony and VoIP
// even if no device change is needed
force = true;
}
// check for device and output changes triggered by new phone state
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++) {
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) {
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, 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, stream, samplingRate,format, channelMask, flags,
offloadInfo);
}
audio_io_handle_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
if (attr == NULL) {
ALOGE("getOutputForAttr() called with NULL audio attributes");
return 0;
}
ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x",
attr->usage, attr->content_type, attr->tags, attr->flags);
// TODO this is where filtering for custom policies (rerouting, dynamic sources) will go
routing_strategy strategy = (routing_strategy) getStrategyForAttr(attr);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
if ((attr->flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
}
ALOGV("getOutputForAttr() device %d, samplingRate %d, format %x, channelMask %x, flags %x",
device, samplingRate, format, channelMask, flags);
audio_stream_type_t stream = streamTypefromAttributesInt(attr);
return getOutputForDevice(device, stream, samplingRate, format, channelMask, flags,
offloadInfo);
}
audio_io_handle_t AudioPolicyManager::getOutputForDevice(
audio_devices_t device,
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);
}
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);
}
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,
int session)
{
ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
ALOGW("startOutput() unknown output %d", output);
return BAD_VALUE;
}
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) {
audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
routing_strategy strategy = getStrategy(stream);
bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
(strategy == STRATEGY_SONIFICATION_RESPECTFUL);
uint32_t waitMs = 0;
bool force = false;
for (size_t i = 0; i < mOutputs.size(); i++) {
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.
uint32_t latency = desc->latency();
if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
waitMs = latency;
}
}
}
uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
// handle special case for sonification while in call
if (isInCall()) {
handleIncallSonification(stream, true, false);
}
// apply volume rules for current stream and device if necessary
checkAndSetVolume(stream,
mStreams[stream].getVolumeIndex(newDevice),
output,
newDevice);
// update the outputs if starting an output with a stream that can affect notification
// routing
handleNotificationRoutingForStream(stream);
if (waitMs > muteWaitMs) {
usleep((waitMs - muteWaitMs) * 2 * 1000);
}
}
return NO_ERROR;
}
status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session)
{
ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
ssize_t index = mOutputs.indexOfKey(output);
if (index < 0) {
ALOGW("stopOutput() unknown output %d", output);
return BAD_VALUE;
}
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
// handle special case for sonification while in call
if (isInCall()) {
handleIncallSonification(stream, false, false);
}
if (outputDesc->mRefCount[stream] > 0) {
// decrement usage count of this stream on the output
outputDesc->changeRefCount(stream, -1);
// store time at which the stream was stopped - see isStreamActive()
if (outputDesc->mRefCount[stream] == 0) {
outputDesc->mStopTime[stream] = systemTime();
audio_devices_t newDevice = 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)
{
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();
}
}
}
audio_io_handle_t AudioPolicyManager::getInput(audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_session_t session,
audio_input_flags_t flags)
{
ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, session %d, "
"flags %#x",
inputSource, samplingRate, format, channelMask, session, flags);
audio_devices_t device = getDeviceForInputSource(inputSource);
if (device == AUDIO_DEVICE_NONE) {
ALOGW("getInput() could not find device for inputSource %d", inputSource);
return AUDIO_IO_HANDLE_NONE;
}
// 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;
}
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
bool isSoundTrigger = false;
audio_source_t halInputSource = inputSource;
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,
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,
samplingRate,
format,
channelMask,
flags);
if (profile == 0) {
ALOGW("getInput() could not find profile for device 0x%X, samplingRate %u, format %#x, "
"channelMask 0x%X, flags %#x",
device, samplingRate, format, channelMask, log_flags);
return AUDIO_IO_HANDLE_NONE;
}
}
if (profile->mModule->mHandle == 0) {
ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
return AUDIO_IO_HANDLE_NONE;
}
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,
String8(""),
halInputSource,
flags);
// only accept input with the exact requested set of parameters
if (status != NO_ERROR ||
(samplingRate != config.sample_rate) ||
(format != config.format) ||
(channelMask != config.channel_mask)) {
ALOGW("getInput() failed opening input: samplingRate %d, format %d, channelMask %x",
samplingRate, format, channelMask);
if (input != AUDIO_IO_HANDLE_NONE) {
mpClientInterface->closeInput(input);
}
return AUDIO_IO_HANDLE_NONE;
}
sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
inputDesc->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;
addInput(input, inputDesc);
mpClientInterface->onAudioPortListUpdate();
return input;
}
status_t AudioPolicyManager::startInput(audio_io_handle_t input,
audio_session_t session)
{
ALOGV("startInput() input %d", input);
ssize_t index = mInputs.indexOfKey(input);
if (index < 0) {
ALOGW("startInput() unknown input %d", input);
return BAD_VALUE;
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
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.
// For remote submix (a virtual device), we open only one input per capture request.
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_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 (audio_is_remote_submix_device(inputDesc->mDevice)) {
setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_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;
}
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);
// compute and apply stream volume on all outputs according to connected device
status_t status = NO_ERROR;
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_devices_t curDevice =
getDeviceForVolume(mOutputs.valueAt(i)->device());
if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) {
status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
if (volStatus != NO_ERROR) {
status = volStatus;
}
}
}
return status;
}
status_t 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)) {
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->mInputSource == (int)source ||
(source == AUDIO_SOURCE_VOICE_RECOGNITION &&
inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD))
&& (inputDescriptor->mRefCount > 0)) {
return true;
}
}
return false;
}
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();
}
audio_devices_t AudioPolicyManager::availablePrimaryInputDevices()
{
audio_module_handle_t primaryHandle =
mOutputs.valueFor(mPrimaryOutput)->mProfile->mModule->mHandle;
audio_devices_t devices = AUDIO_DEVICE_NONE;
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
if (mAvailableInputDevices[i]->mModule->mHandle == primaryHandle) {
devices |= mAvailableInputDevices[i]->mDeviceType;
}
}
return devices;
}
status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
audio_patch_handle_t *handle,
uid_t uid)
{
ALOGV("createAudioPatch()");
if (handle == NULL || patch == NULL) {
return BAD_VALUE;
}
ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
return BAD_VALUE;
}
// only one source per audio patch supported for now
if (patch->num_sources > 1) {
return INVALID_OPERATION;
}
if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) {
return INVALID_OPERATION;
}
for (size_t i = 0; i < patch->num_sinks; i++) {
if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) {
return INVALID_OPERATION;
}
}
sp<AudioPatch> patchDesc;
ssize_t index = mAudioPatches.indexOfKey(*handle);
ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
patch->sources[0].role,
patch->sources[0].type);
#if LOG_NDEBUG == 0
for (size_t i = 0; i < patch->num_sinks; i++) {
ALOGV("createAudioPatch sink %d: 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 = 0;
}
if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
if (outputDesc == NULL) {
ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
return BAD_VALUE;
}
ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
outputDesc->mIoHandle);
if (patchDesc != 0) {
if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
patchDesc->mPatch.sources[0].id, patch->sources[0].id);
return BAD_VALUE;
}
}
DeviceVector devices;
for (size_t i = 0; i < patch->num_sinks; i++) {
// Only support mix to devices connection
// TODO add support for mix to mix connection
if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
ALOGV("createAudioPatch() source mix but sink is not a device");
return INVALID_OPERATION;
}
sp<DeviceDescriptor> devDesc =
mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
if (devDesc == 0) {
ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id);
return BAD_VALUE;
}
if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
patch->sources[0].sample_rate,
NULL, // updatedSamplingRate
patch->sources[0].format,
patch->sources[0].channel_mask,
AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
ALOGV("createAudioPatch() profile not supported for device %08x",
devDesc->mDeviceType);
return INVALID_OPERATION;
}
devices.add(devDesc);
}
if (devices.size() == 0) {
return INVALID_OPERATION;
}
// TODO: reconfigure output format and channels here
ALOGV("createAudioPatch() setting device %08x on output %d",
devices.types(), outputDesc->mIoHandle);
setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle);
index = mAudioPatches.indexOfKey(*handle);
if (index >= 0) {
if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
}
patchDesc = mAudioPatches.valueAt(index);
patchDesc->mUid = uid;
ALOGV("createAudioPatch() success");
} else {
ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
return INVALID_OPERATION;
}
} else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
// input device to input mix connection
// only one sink supported when connecting an input device to a mix
if (patch->num_sinks > 1) {
return INVALID_OPERATION;
}
sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
if (inputDesc == NULL) {
return BAD_VALUE;
}
if (patchDesc != 0) {
if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) {
return BAD_VALUE;
}
}
sp<DeviceDescriptor> devDesc =
mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
if (devDesc == 0) {
return BAD_VALUE;
}
if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
patch->sinks[0].sample_rate,
NULL, /*updatedSampleRate*/
patch->sinks[0].format,
patch->sinks[0].channel_mask,
// FIXME for the parameter type,
// and the NONE
(audio_output_flags_t)
AUDIO_INPUT_FLAG_NONE)) {
return INVALID_OPERATION;
}
// TODO: reconfigure output format and channels here
ALOGV("createAudioPatch() setting device %08x on output %d",
devDesc->mDeviceType, inputDesc->mIoHandle);
setInputDevice(inputDesc->mIoHandle, devDesc->mDeviceType, true, handle);
index = mAudioPatches.indexOfKey(*handle);
if (index >= 0) {
if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
}
patchDesc = mAudioPatches.valueAt(index);
patchDesc->mUid = uid;
ALOGV("createAudioPatch() success");
} else {
ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
return INVALID_OPERATION;
}
} else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
// device to device connection
if (patchDesc != 0) {
if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
return BAD_VALUE;
}
}
sp<DeviceDescriptor> srcDeviceDesc =
mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
if (srcDeviceDesc == 0) {
return BAD_VALUE;
}
//update source and sink with our own data as the data passed in the patch may
// be incomplete.
struct audio_patch newPatch = *patch;
srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
for (size_t i = 0; i < patch->num_sinks; i++) {
if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
ALOGV("createAudioPatch() source device but one sink is not a device");
return INVALID_OPERATION;
}
sp<DeviceDescriptor> sinkDeviceDesc =
mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
if (sinkDeviceDesc == 0) {
return BAD_VALUE;
}
sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) {
// only one sink supported when connected devices across HW modules
if (patch->num_sinks > 1) {
return INVALID_OPERATION;
}
SortedVector<audio_io_handle_t> outputs =
getOutputsForDevice(sinkDeviceDesc->mDeviceType,
mOutputs);
// if the sink device is reachable via an opened output stream, request to go via
// this output stream by adding a second source to the patch description
audio_io_handle_t output = selectOutput(outputs,
AUDIO_OUTPUT_FLAG_NONE,
AUDIO_FORMAT_INVALID);
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc->isDuplicated()) {
return INVALID_OPERATION;
}
outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]);
newPatch.num_sources = 2;
}
}
}
// TODO: check from routing capabilities in config file and other conflicting patches
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (index >= 0) {
afPatchHandle = patchDesc->mAfPatchHandle;
}
status_t status = mpClientInterface->createAudioPatch(&newPatch,
&afPatchHandle,
0);
ALOGV("createAudioPatch() patch panel returned %d patchHandle %d",
status, afPatchHandle);
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
&newPatch, uid);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = newPatch;
}
patchDesc->mAfPatchHandle = afPatchHandle;
*handle = patchDesc->mHandle;
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
} else {
ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
status);
return INVALID_OPERATION;
}
} else {
return BAD_VALUE;
}
} else {
return BAD_VALUE;
}
return NO_ERROR;
}
status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle,
uid_t uid)
{
ALOGV("releaseAudioPatch() patch %d", handle);
ssize_t index = mAudioPatches.indexOfKey(handle);
if (index < 0) {
return BAD_VALUE;
}
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
mUidCached, patchDesc->mUid, uid);
if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
return INVALID_OPERATION;
}
struct audio_patch *patch = &patchDesc->mPatch;
patchDesc->mUid = mUidCached;
if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
if (outputDesc == NULL) {
ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
return BAD_VALUE;
}
setOutputDevice(outputDesc->mIoHandle,
getNewOutputDevice(outputDesc->mIoHandle, 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 = getInputFromId(patch->sinks[0].id);
if (inputDesc == NULL) {
ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
return BAD_VALUE;
}
setInputDevice(inputDesc->mIoHandle,
getNewInputDevice(inputDesc->mIoHandle),
true,
NULL);
} else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle;
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 (num_patches == NULL || (*num_patches != 0 && patches == NULL) ||
generation == NULL) {
return BAD_VALUE;
}
ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu",
*num_patches, patches, mAudioPatches.size());
if (patches == NULL) {
*num_patches = 0;
}
size_t patchesWritten = 0;
size_t patchesMax = *num_patches;
for (size_t i = 0;
i < mAudioPatches.size() && patchesWritten < patchesMax; i++) {
patches[patchesWritten] = mAudioPatches[i]->mPatch;
patches[patchesWritten++].id = mAudioPatches[i]->mHandle;
ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks);
}
*num_patches = mAudioPatches.size();
*generation = curAudioPortGeneration();
ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches);
return NO_ERROR;
}
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<AudioOutputDescriptor> outputDesc = 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 = 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::clearAudioPatches(uid_t uid)
{
for (ssize_t i = 0; i < (ssize_t)mAudioPatches.size(); i++) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
if (patchDesc->mUid == uid) {
// releaseAudioPatch() removes the patch from mAudioPatches
if (releaseAudioPatch(mAudioPatches.keyAt(i), uid) == NO_ERROR) {
i--;
}
}
}
}
status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session,
audio_io_handle_t *ioHandle,
audio_devices_t *device)
{
*session = (audio_session_t)mpClientInterface->newAudioUniqueId();
*ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId();
*device = getDeviceForInputSource(AUDIO_SOURCE_HOTWORD);
mSoundTriggerSessions.add(*session, *ioHandle);
return NO_ERROR;
}
status_t AudioPolicyManager::releaseSoundTriggerSession(audio_session_t session)
{
ssize_t index = mSoundTriggerSessions.indexOfKey(session);
if (index < 0) {
ALOGW("acquireSoundTriggerSession() session %d not registered", session);
return BAD_VALUE;
}
mSoundTriggerSessions.removeItem(session);
return NO_ERROR;
}
status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle,
const sp<AudioPatch>& patch)
{
ssize_t index = mAudioPatches.indexOfKey(handle);
if (index >= 0) {
ALOGW("addAudioPatch() patch %d already in", handle);
return ALREADY_EXISTS;
}
mAudioPatches.add(handle, patch);
ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d"
"sink handle %d",
handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks,
patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id);
return NO_ERROR;
}
status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle)
{
ssize_t index = mAudioPatches.indexOfKey(handle);
if (index < 0) {
ALOGW("removeAudioPatch() patch %d not in", handle);
return ALREADY_EXISTS;
}
ALOGV("removeAudioPatch() handle %d af handle %d", handle,
mAudioPatches.valueAt(index)->mAfPatchHandle);
mAudioPatches.removeItemsAt(index);
return NO_ERROR;
}
// ----------------------------------------------------------------------------
// AudioPolicyManager
// ----------------------------------------------------------------------------
uint32_t AudioPolicyManager::nextUniqueId()
{
return android_atomic_inc(&mNextUniqueId);
}
uint32_t AudioPolicyManager::nextAudioPortGeneration()
{
return android_atomic_inc(&mAudioPortGeneration);
}
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
:
#ifdef AUDIO_POLICY_TEST
Thread(false),
#endif //AUDIO_POLICY_TEST
mPrimaryOutput((audio_io_handle_t)0),
mPhoneState(AUDIO_MODE_NORMAL),
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
mA2dpSuspended(false),
mSpeakerDrcEnabled(false), mNextUniqueId(1),
mAudioPortGeneration(1)
{
mUidCached = getuid();
mpClientInterface = clientInterface;
for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
}
mDefaultOutputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_OUT_SPEAKER);
if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
ALOGE("could not load audio policy configuration file, setting defaults");
defaultAudioPolicyConfig();
}
}
// mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
// must be done after reading the policy
initializeVolumeCurves();
// open all output streams needed to access attached devices
audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
for (size_t i = 0; i < mHwModules.size(); i++) {
mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
if (mHwModules[i]->mHandle == 0) {
ALOGW("could not open HW module %s", mHwModules[i]->mName);
continue;
}
// open all output streams needed to access attached devices
// except for direct output streams that are only opened when they are actually
// required by an app.
// This also validates mAvailableOutputDevices list
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
if (outProfile->mSupportedDevices.isEmpty()) {
ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName);
continue;
}
if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
continue;
}
audio_devices_t profileType = outProfile->mSupportedDevices.types();
if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) {
profileType = mDefaultOutputDevice->mDeviceType;
} else {
// chose first device present in mSupportedDevices also part of
// outputDeviceTypes
for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
profileType = outProfile->mSupportedDevices[k]->mDeviceType;
if ((profileType & outputDeviceTypes) != 0) {
break;
}
}
}
if ((profileType & outputDeviceTypes) == 0) {
continue;
}
sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile);
outputDesc->mDevice = profileType;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = outputDesc->mSamplingRate;
config.channel_mask = outputDesc->mChannelMask;
config.format = outputDesc->mFormat;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle,
&output,
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
if (status != NO_ERROR) {
ALOGW("Cannot open output stream for device %08x on hw module %s",
outputDesc->mDevice,
mHwModules[i]->mName);
} else {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType;
ssize_t index =
mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if ((index >= 0) && (mAvailableOutputDevices[index]->mId == 0)) {
mAvailableOutputDevices[index]->mId = nextUniqueId();
mAvailableOutputDevices[index]->mModule = mHwModules[i];
}
}
if (mPrimaryOutput == 0 &&
outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = output;
}
addOutput(output, outputDesc);
setOutputDevice(output,
outputDesc->mDevice,
true);
}
}
// open input streams needed to access attached devices to validate
// mAvailableInputDevices list
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
{
const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
if (inProfile->mSupportedDevices.isEmpty()) {
ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName);
continue;
}
// chose first device present in mSupportedDevices also part of
// inputDeviceTypes
audio_devices_t profileType = AUDIO_DEVICE_NONE;
for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
profileType = inProfile->mSupportedDevices[k]->mDeviceType;
if (profileType & inputDeviceTypes) {
break;
}
}
if ((profileType & inputDeviceTypes) == 0) {
continue;
}
sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(inProfile);
inputDesc->mInputSource = AUDIO_SOURCE_MIC;
inputDesc->mDevice = profileType;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = inputDesc->mSamplingRate;
config.channel_mask = inputDesc->mChannelMask;
config.format = inputDesc->mFormat;
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle,
&input,
&config,
&inputDesc->mDevice,
String8(""),
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE);
if (status == NO_ERROR) {
for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType;
ssize_t index =
mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if ((index >= 0) && (mAvailableInputDevices[index]->mId == 0)) {
mAvailableInputDevices[index]->mId = nextUniqueId();
mAvailableInputDevices[index]->mModule = mHwModules[i];
}
}
mpClientInterface->closeInput(input);
} else {
ALOGW("Cannot open input stream for device %08x on hw module %s",
inputDesc->mDevice,
mHwModules[i]->mName);
}
}
}
// make sure all attached devices have been allocated a unique ID
for (size_t i = 0; i < mAvailableOutputDevices.size();) {
if (mAvailableOutputDevices[i]->mId == 0) {
ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
continue;
}
i++;
}
for (size_t i = 0; i < mAvailableInputDevices.size();) {
if (mAvailableInputDevices[i]->mId == 0) {
ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
mAvailableInputDevices.remove(mAvailableInputDevices[i]);
continue;
}
i++;
}
// make sure default device is reachable
if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType);
}
ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
updateDevicesAndOutputs();
#ifdef AUDIO_POLICY_TEST
if (mPrimaryOutput != 0) {
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"), 0);
mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
mTestSamplingRate = 44100;
mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
mTestLatencyMs = 0;
mCurOutput = 0;
mDirectOutput = false;
for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
mTestOutputs[i] = 0;
}
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "AudioPolicyManagerTest");
run(buffer, ANDROID_PRIORITY_AUDIO);
}
#endif //AUDIO_POLICY_TEST
}
AudioPolicyManager::~AudioPolicyManager()
{
#ifdef AUDIO_POLICY_TEST
exit();
#endif //AUDIO_POLICY_TEST
for (size_t i = 0; i < mOutputs.size(); i++) {
mpClientInterface->closeOutput(mOutputs.keyAt(i));
}
for (size_t i = 0; i < mInputs.size(); i++) {
mpClientInterface->closeInput(mInputs.keyAt(i));
}
mAvailableOutputDevices.clear();
mAvailableInputDevices.clear();
mOutputs.clear();
mInputs.clear();
mHwModules.clear();
}
status_t AudioPolicyManager::initCheck()
{
return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
}
#ifdef AUDIO_POLICY_TEST
bool AudioPolicyManager::threadLoop()
{
ALOGV("entering threadLoop()");
while (!exitPending())
{
String8 command;
int valueInt;
String8 value;
Mutex::Autolock _l(mLock);
mWaitWorkCV.waitRelative(mLock, milliseconds(50));
command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
AudioParameter param = AudioParameter(command);
if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
valueInt != 0) {
ALOGV("Test command %s received", command.string());
String8 target;
if (param.get(String8("target"), target) != NO_ERROR) {
target = "Manager";
}
if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
param.remove(String8("test_cmd_policy_output"));
mCurOutput = valueInt;
}
if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_direct"));
if (value == "false") {
mDirectOutput = false;
} else if (value == "true") {
mDirectOutput = true;
}
}
if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
param.remove(String8("test_cmd_policy_input"));
mTestInput = valueInt;
}
if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_format"));
int format = AUDIO_FORMAT_INVALID;
if (value == "PCM 16 bits") {
format = AUDIO_FORMAT_PCM_16_BIT;
} else if (value == "PCM 8 bits") {
format = AUDIO_FORMAT_PCM_8_BIT;
} else if (value == "Compressed MP3") {
format = AUDIO_FORMAT_MP3;
}
if (format != AUDIO_FORMAT_INVALID) {
if (target == "Manager") {
mTestFormat = format;
} else if (mTestOutputs[mCurOutput] != 0) {
AudioParameter outputParam = AudioParameter();
outputParam.addInt(String8("format"), format);
mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
}
}
}
if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_channels"));
int channels = 0;
if (value == "Channels Stereo") {
channels = AUDIO_CHANNEL_OUT_STEREO;
} else if (value == "Channels Mono") {
channels = AUDIO_CHANNEL_OUT_MONO;
}
if (channels != 0) {
if (target == "Manager") {
mTestChannels = channels;
} else if (mTestOutputs[mCurOutput] != 0) {
AudioParameter outputParam = AudioParameter();
outputParam.addInt(String8("channels"), channels);
mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
}
}
}
if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
param.remove(String8("test_cmd_policy_sampleRate"));
if (valueInt >= 0 && valueInt <= 96000) {
int samplingRate = valueInt;
if (target == "Manager") {
mTestSamplingRate = samplingRate;
} else if (mTestOutputs[mCurOutput] != 0) {
AudioParameter outputParam = AudioParameter();
outputParam.addInt(String8("sampling_rate"), samplingRate);
mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
}
}
}
if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
param.remove(String8("test_cmd_policy_reopen"));
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
mpClientInterface->closeOutput(mPrimaryOutput);
audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
mOutputs.removeItem(mPrimaryOutput);
sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = outputDesc->mSamplingRate;
config.channel_mask = outputDesc->mChannelMask;
config.format = outputDesc->mFormat;
status_t status = mpClientInterface->openOutput(moduleHandle,
&mPrimaryOutput,
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
if (status != NO_ERROR) {
ALOGE("Failed to reopen hardware output stream, "
"samplingRate: %d, format %d, channels %d",
outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
} else {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"), 0);
mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
addOutput(mPrimaryOutput, outputDesc);
}
}
mpClientInterface->setParameters(0, String8("test_cmd_policy="));
}
}
return false;
}
void AudioPolicyManager::exit()
{
{
AutoMutex _l(mLock);
requestExit();
mWaitWorkCV.signal();
}
requestExitAndWait();
}
int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
{
for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
if (output == mTestOutputs[i]) return i;
}
return 0;
}
#endif //AUDIO_POLICY_TEST
// ---
void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc)
{
outputDesc->mIoHandle = output;
outputDesc->mId = nextUniqueId();
mOutputs.add(output, outputDesc);
nextAudioPortGeneration();
}
void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc)
{
inputDesc->mIoHandle = input;
inputDesc->mId = nextUniqueId();
mInputs.add(input, inputDesc);
nextAudioPortGeneration();
}
void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
const String8 address /*in*/,
SortedVector<audio_io_handle_t>& outputs /*out*/) {
// look for a match on the given address on the addresses of the outputs:
// find the address by finding the patch that maps to this output
ssize_t patchIdx = mAudioPatches.indexOfKey(desc->mPatchHandle);
//ALOGV(" inspecting output %d (patch %d) for supported device=0x%x",
// outputIdx, patchIdx, desc->mProfile->mSupportedDevices.types());
if (patchIdx >= 0) {
const sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patchIdx);
const int numSinks = patchDesc->mPatch.num_sinks;
for (ssize_t j=0; j < numSinks; j++) {
if (patchDesc->mPatch.sinks[j].type == AUDIO_PORT_TYPE_DEVICE) {
const char* patchAddr =
patchDesc->mPatch.sinks[j].ext.device.address;
if (strncmp(patchAddr,
address.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s",
desc->mIoHandle, patchDesc->mPatch.sinks[j].ext.device.address);
outputs.add(desc->mIoHandle);
break;
}
}
}
}
}
status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
audio_policy_dev_state_t state,
SortedVector<audio_io_handle_t>& outputs,
const String8 address)
{
audio_devices_t device = devDesc->mDeviceType;
sp<AudioOutputDescriptor> desc;
// erase all current sample rates, formats and channel masks
devDesc->clearCapabilities();
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->mProfile->mSupportedDevices.types() & device)) {
if (!deviceDistinguishesOnAddress(device)) {
ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
outputs.add(mOutputs.keyAt(i));
} else {
ALOGV(" checking address match due to device 0x%x", device);
findIoHandlesByAddress(desc, address, outputs);
}
}
}
// then look for output profiles that can be routed to this device
SortedVector< sp<IOProfile> > profiles;
for (size_t i = 0; i < mHwModules.size(); i++)
{
if (mHwModules[i]->mHandle == 0) {
continue;
}
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) {
ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
profiles.add(mHwModules[i]->mOutputProfiles[j]);
}
}
}
ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size());
if (profiles.isEmpty() && outputs.isEmpty()) {
ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
return BAD_VALUE;
}
// open outputs for matching profiles if needed. Direct outputs are also opened to
// query for dynamic parameters and will be closed later by setDeviceConnectionState()
for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
sp<IOProfile> profile = profiles[profile_index];
// nothing to do if one output is already opened for this profile
size_t j;
for (j = 0; j < outputs.size(); j++) {
desc = mOutputs.valueFor(outputs.itemAt(j));
if (!desc->isDuplicated() && desc->mProfile == profile) {
// matching profile: save the sample rates, format and channel masks supported
// by the profile in our device descriptor
devDesc->importAudioPort(profile);
break;
}
}
if (j != outputs.size()) {
continue;
}
ALOGV("opening output for device %08x with params %s profile %p",
device, address.string(), profile.get());
desc = new AudioOutputDescriptor(profile);
desc->mDevice = device;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = desc->mSamplingRate;
config.channel_mask = desc->mChannelMask;
config.format = desc->mFormat;
config.offload_info.sample_rate = desc->mSamplingRate;
config.offload_info.channel_mask = desc->mChannelMask;
config.offload_info.format = desc->mFormat;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openOutput(profile->mModule->mHandle,
&output,
&config,
&desc->mDevice,
address,
&desc->mLatency,
desc->mFlags);
if (status == NO_ERROR) {
desc->mSamplingRate = config.sample_rate;
desc->mChannelMask = config.channel_mask;
desc->mFormat = config.format;
// Here is where the out_set_parameters() for card & device gets called
if (!address.isEmpty()) {
char *param = audio_device_address_to_parameter(device, address);
mpClientInterface->setParameters(output, String8(param));
free(param);
}
// Here is where we step through and resolve any "dynamic" fields
String8 reply;
char *value;
if (profile->mSamplingRates[0] == 0) {
reply = mpClientInterface->getParameters(output,
String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
ALOGV("checkOutputsForDevice() supported sampling rates %s",
reply.string());
value = strpbrk((char *)reply.string(), "=");
if (value != NULL) {
profile->loadSamplingRates(value + 1);
}
}
if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
reply = mpClientInterface->getParameters(output,
String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
ALOGV("checkOutputsForDevice() supported formats %s",
reply.string());
value = strpbrk((char *)reply.string(), "=");
if (value != NULL) {
profile->loadFormats(value + 1);
}
}
if (profile->mChannelMasks[0] == 0) {
reply = mpClientInterface->getParameters(output,
String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
ALOGV("checkOutputsForDevice() supported channel masks %s",
reply.string());
value = strpbrk((char *)reply.string(), "=");
if (value != NULL) {
profile->loadOutChannels(value + 1);
}
}
if (((profile->mSamplingRates[0] == 0) &&
(profile->mSamplingRates.size() < 2)) ||
((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
(profile->mFormats.size() < 2)) ||
((profile->mChannelMasks[0] == 0) &&
(profile->mChannelMasks.size() < 2))) {
ALOGW("checkOutputsForDevice() missing param");
mpClientInterface->closeOutput(output);
output = AUDIO_IO_HANDLE_NONE;
} else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 ||
profile->mChannelMasks[0] == 0) {
mpClientInterface->closeOutput(output);
config.sample_rate = profile->pickSamplingRate();
config.channel_mask = profile->pickChannelMask();
config.format = profile->pickFormat();
config.offload_info.sample_rate = config.sample_rate;
config.offload_info.channel_mask = config.channel_mask;
config.offload_info.format = config.format;
status = mpClientInterface->openOutput(profile->mModule->mHandle,
&output,
&config,
&desc->mDevice,
address,
&desc->mLatency,
desc->mFlags);
if (status == NO_ERROR) {
desc->mSamplingRate = config.sample_rate;
desc->mChannelMask = config.channel_mask;
desc->mFormat = config.format;
} else {
output = AUDIO_IO_HANDLE_NONE;
}
}
if (output != AUDIO_IO_HANDLE_NONE) {
addOutput(output, desc);
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
// set initial stream volume for device
applyStreamVolumes(output, device, 0, true);
//TODO: configure audio effect output stage here
// open a duplicating output thread for the new output and the primary output
duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
mPrimaryOutput);
if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) {
// add duplicated output descriptor
sp<AudioOutputDescriptor> dupOutputDesc =
new AudioOutputDescriptor(NULL);
dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
dupOutputDesc->mSamplingRate = desc->mSamplingRate;
dupOutputDesc->mFormat = desc->mFormat;
dupOutputDesc->mChannelMask = desc->mChannelMask;
dupOutputDesc->mLatency = desc->mLatency;
addOutput(duplicatedOutput, dupOutputDesc);
applyStreamVolumes(duplicatedOutput, device, 0, true);
} else {
ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
mPrimaryOutput, output);
mpClientInterface->closeOutput(output);
mOutputs.removeItem(output);
nextAudioPortGeneration();
output = AUDIO_IO_HANDLE_NONE;
}
}
}
} else {
output = AUDIO_IO_HANDLE_NONE;
}
if (output == AUDIO_IO_HANDLE_NONE) {
ALOGW("checkOutputsForDevice() could not open output for device %x", device);
profiles.removeAt(profile_index);
profile_index--;
} else {
outputs.add(output);
devDesc->importAudioPort(profile);
if (deviceDistinguishesOnAddress(device)) {
ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
device, address.string());
setOutputDevice(output, device, true/*force*/, 0/*delay*/,
NULL/*patch handle*/, address.string());
}
ALOGV("checkOutputsForDevice(): adding output %d", output);
}
}
if (profiles.isEmpty()) {
ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
return BAD_VALUE;
}
} else { // Disconnect
// check if one opened output is not needed any more after disconnecting one device
for (size_t i = 0; i < mOutputs.size(); i++) {
desc = mOutputs.valueAt(i);
if (!desc->isDuplicated()) {
if (!(desc->mProfile->mSupportedDevices.types()
& mAvailableOutputDevices.types())) {
ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
mOutputs.keyAt(i));
outputs.add(mOutputs.keyAt(i));
} else if (deviceDistinguishesOnAddress(device) &&
// exact match on device
(desc->mProfile->mSupportedDevices.types() == device)) {
findIoHandlesByAddress(desc, address, outputs);
}
}
}
// Clear any profiles associated with the disconnected device.
for (size_t i = 0; i < mHwModules.size(); i++)
{
if (mHwModules[i]->mHandle == 0) {
continue;
}
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
if (profile->mSupportedDevices.types() & device) {
ALOGV("checkOutputsForDevice(): "
"clearing direct output profile %zu on module %zu", j, i);
if (profile->mSamplingRates[0] == 0) {
profile->mSamplingRates.clear();
profile->mSamplingRates.add(0);
}
if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
profile->mFormats.clear();
profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
}
if (profile->mChannelMasks[0] == 0) {
profile->mChannelMasks.clear();
profile->mChannelMasks.add(0);
}
}
}
}
}
return NO_ERROR;
}
status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device,
audio_policy_dev_state_t state,
SortedVector<audio_io_handle_t>& inputs,
const String8 address)
{
sp<AudioInputDescriptor> desc;
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
// first list already open inputs that can be routed to this device
for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
inputs.add(mInputs.keyAt(input_index));
}
}
// then look for input profiles that can be routed to this device
SortedVector< sp<IOProfile> > profiles;
for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
{
if (mHwModules[module_idx]->mHandle == 0) {
continue;
}
for (size_t profile_index = 0;
profile_index < mHwModules[module_idx]->mInputProfiles.size();
profile_index++)
{
if (mHwModules[module_idx]->mInputProfiles[profile_index]->mSupportedDevices.types()
& (device & ~AUDIO_DEVICE_BIT_IN)) {
ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
profile_index, module_idx);
profiles.add(mHwModules[module_idx]->mInputProfiles[profile_index]);
}
}
}
if (profiles.isEmpty() && inputs.isEmpty()) {
ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
return BAD_VALUE;
}
// open inputs for matching profiles if needed. Direct inputs are also opened to
// query for dynamic parameters and will be closed later by setDeviceConnectionState()
for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
sp<IOProfile> profile = profiles[profile_index];
// nothing to do if one input is already opened for this profile
size_t input_index;
for (input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
if (desc->mProfile == profile) {
break;
}
}
if (input_index != mInputs.size()) {
continue;
}
ALOGV("opening input for device 0x%X with params %s", device, address.string());
desc = new AudioInputDescriptor(profile);
desc->mDevice = device;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = desc->mSamplingRate;
config.channel_mask = desc->mChannelMask;
config.format = desc->mFormat;
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
&input,
&config,
&desc->mDevice,
address,
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE /*FIXME*/);
if (status == NO_ERROR) {
desc->mSamplingRate = config.sample_rate;
desc->mChannelMask = config.channel_mask;
desc->mFormat = config.format;
if (!address.isEmpty()) {
char *param = audio_device_address_to_parameter(device, address);
mpClientInterface->setParameters(input, String8(param));
free(param);
}
// Here is where we step through and resolve any "dynamic" fields
String8 reply;
char *value;
if (profile->mSamplingRates[0] == 0) {
reply = mpClientInterface->getParameters(input,
String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
ALOGV("checkInputsForDevice() direct input sup sampling rates %s",
reply.string());
value = strpbrk((char *)reply.string(), "=");
if (value != NULL) {
profile->loadSamplingRates(value + 1);
}
}
if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
reply = mpClientInterface->getParameters(input,
String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
value = strpbrk((char *)reply.string(), "=");
if (value != NULL) {
profile->loadFormats(value + 1);
}
}
if (profile->mChannelMasks[0] == 0) {
reply = mpClientInterface->getParameters(input,
String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
ALOGV("checkInputsForDevice() direct input sup channel masks %s",
reply.string());
value = strpbrk((char *)reply.string(), "=");
if (value != NULL) {
profile->loadInChannels(value + 1);
}
}
if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) ||
((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) {
ALOGW("checkInputsForDevice() direct input missing param");
mpClientInterface->closeInput(input);
input = AUDIO_IO_HANDLE_NONE;
}
if (input != 0) {
addInput(input, desc);
}
} // endif input != 0
if (input == AUDIO_IO_HANDLE_NONE) {
ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
profiles.removeAt(profile_index);
profile_index--;
} else {
inputs.add(input);
ALOGV("checkInputsForDevice(): adding input %d", input);
}
} // end scan profiles
if (profiles.isEmpty()) {
ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
return BAD_VALUE;
}
} else {
// Disconnect
// check if one opened input is not needed any more after disconnecting one device
for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types())) {
ALOGV("checkInputsForDevice(): disconnecting adding input %d",
mInputs.keyAt(input_index));
inputs.add(mInputs.keyAt(input_index));
}
}
// Clear any profiles associated with the disconnected device.
for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) {
if (mHwModules[module_index]->mHandle == 0) {
continue;
}
for (size_t profile_index = 0;
profile_index < mHwModules[module_index]->mInputProfiles.size();
profile_index++) {
sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
if (profile->mSupportedDevices.types() & device) {
ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
profile_index, module_index);
if (profile->mSamplingRates[0] == 0) {
profile->mSamplingRates.clear();
profile->mSamplingRates.add(0);
}
if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
profile->mFormats.clear();
profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
}
if (profile->mChannelMasks[0] == 0) {
profile->mChannelMasks.clear();
profile->mChannelMasks.add(0);
}
}
}
}
} // end disconnect
return NO_ERROR;
}
void AudioPolicyManager::closeOutput(audio_io_handle_t output)
{
ALOGV("closeOutput(%d)", output);
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc == NULL) {
ALOGW("closeOutput() unknown output %d", output);
return;
}
// look for duplicated outputs connected to the output being removed.
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
if (dupOutputDesc->isDuplicated() &&
(dupOutputDesc->mOutput1 == outputDesc ||
dupOutputDesc->mOutput2 == outputDesc)) {
sp<AudioOutputDescriptor> outputDesc2;
if (dupOutputDesc->mOutput1 == outputDesc) {
outputDesc2 = dupOutputDesc->mOutput2;
} else {
outputDesc2 = dupOutputDesc->mOutput1;
}
// As all active tracks on duplicated output will be deleted,
// and as they were also referenced on the other output, the reference
// count for their stream type must be adjusted accordingly on
// the other output.
for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
int refCount = dupOutputDesc->mRefCount[j];
outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
}
audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
mpClientInterface->closeOutput(duplicatedOutput);
mOutputs.removeItem(duplicatedOutput);
}
}
nextAudioPortGeneration();
ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
mAudioPatches.removeItemsAt(index);
mpClientInterface->onAudioPatchListUpdate();
}
AudioParameter param;
param.add(String8("closing"), String8("true"));
mpClientInterface->setParameters(output, param.toString());
mpClientInterface->closeOutput(output);
mOutputs.removeItem(output);
mPreviousOutputs = mOutputs;
}
void AudioPolicyManager::closeInput(audio_io_handle_t input)
{
ALOGV("closeInput(%d)", input);
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
if (inputDesc == NULL) {
ALOGW("closeInput() unknown input %d", input);
return;
}
nextAudioPortGeneration();
ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
mAudioPatches.removeItemsAt(index);
mpClientInterface->onAudioPatchListUpdate();
}
mpClientInterface->closeInput(input);
mInputs.removeItem(input);
}
SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs)
{
SortedVector<audio_io_handle_t> outputs;
ALOGVV("getOutputsForDevice() device %04x", device);
for (size_t i = 0; i < openOutputs.size(); i++) {
ALOGVV("output %d isDuplicated=%d device=%04x",
i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
outputs.add(openOutputs.keyAt(i));
}
}
return outputs;
}
bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
SortedVector<audio_io_handle_t>& outputs2)
{
if (outputs1.size() != outputs2.size()) {
return false;
}
for (size_t i = 0; i < outputs1.size(); i++) {
if (outputs1[i] != outputs2[i]) {
return false;
}
}
return true;
}
void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
{
audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
if (!vectorsEqual(srcOutputs,dstOutputs)) {
ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
strategy, srcOutputs[0], dstOutputs[0]);
// mute strategy while moving tracks from one output to another
for (size_t i = 0; i < srcOutputs.size(); i++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
if (desc->isStrategyActive(strategy)) {
setStrategyMute(strategy, true, srcOutputs[i]);
setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
}
}
// Move effects associated to this strategy from previous output to new output
if (strategy == STRATEGY_MEDIA) {
audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
SortedVector<audio_io_handle_t> moved;
for (size_t i = 0; i < mEffects.size(); i++) {
sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
effectDesc->mIo != fxOutput) {
if (moved.indexOf(effectDesc->mIo) < 0) {
ALOGV("checkOutputForStrategy() moving effect %d to output %d",
mEffects.keyAt(i), fxOutput);
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo,
fxOutput);
moved.add(effectDesc->mIo);
}
effectDesc->mIo = fxOutput;
}
}
}
// Move tracks associated to this strategy from previous output to new output
for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
if (getStrategy((audio_stream_type_t)i) == strategy) {
mpClientInterface->invalidateStream((audio_stream_type_t)i);
}
}
}
}
void AudioPolicyManager::checkOutputForAllStrategies()
{
if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
checkOutputForStrategy(STRATEGY_PHONE);
if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
checkOutputForStrategy(STRATEGY_SONIFICATION);
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
checkOutputForStrategy(STRATEGY_MEDIA);
checkOutputForStrategy(STRATEGY_DTMF);
}
audio_io_handle_t AudioPolicyManager::getA2dpOutput()
{
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
return mOutputs.keyAt(i);
}
}
return 0;
}
void AudioPolicyManager::checkA2dpSuspend()
{
audio_io_handle_t a2dpOutput = getA2dpOutput();
if (a2dpOutput == 0) {
mA2dpSuspended = false;
return;
}
bool isScoConnected =
(mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) != 0;
// suspend A2DP output if:
// (NOT already suspended) &&
// ((SCO device is connected &&
// (forced usage for communication || for record is SCO))) ||
// (phone state is ringing || in call)
//
// restore A2DP output if:
// (Already suspended) &&
// ((SCO device is NOT connected ||
// (forced usage NOT for communication && NOT for record is SCO))) &&
// (phone state is NOT ringing && NOT in call)
//
if (mA2dpSuspended) {
if ((!isScoConnected ||
((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) &&
((mPhoneState != AUDIO_MODE_IN_CALL) &&
(mPhoneState != AUDIO_MODE_RINGTONE))) {
mpClientInterface->restoreOutput(a2dpOutput);
mA2dpSuspended = false;
}
} else {
if ((isScoConnected &&
((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
(mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) ||
((mPhoneState == AUDIO_MODE_IN_CALL) ||
(mPhoneState == AUDIO_MODE_RINGTONE))) {
mpClientInterface->suspendOutput(a2dpOutput);
mA2dpSuspended = true;
}
}
}
audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache)
{
audio_devices_t device = AUDIO_DEVICE_NONE;
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
if (patchDesc->mUid != mUidCached) {
ALOGV("getNewOutputDevice() device %08x forced by patch %d",
outputDesc->device(), outputDesc->mPatchHandle);
return outputDesc->device();
}
}
// check the following by order of priority to request a routing change if necessary:
// 1: the strategy enforced audible is active and enforced on the output:
// use device for strategy enforced audible
// 2: we are in call or the strategy phone is active on the output:
// use device for strategy phone
// 3: the strategy for enforced audible is active but not enforced on the output:
// use the device for strategy enforced audible
// 4: the strategy sonification is active on the output:
// use device for strategy sonification
// 5: the strategy "respectful" sonification is active on the output:
// use device for strategy "respectful" sonification
// 6: the strategy media is active on the output:
// use device for strategy media
// 7: the strategy DTMF is active on the output:
// use device for strategy DTMF
if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE) &&
mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
} else if (isInCall() ||
outputDesc->isStrategyActive(STRATEGY_PHONE)) {
device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
} else if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
} else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
} else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
} else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
} else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
}
ALOGV("getNewOutputDevice() selected device %x", device);
return device;
}
audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
{
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
if (patchDesc->mUid != mUidCached) {
ALOGV("getNewInputDevice() device %08x forced by patch %d",
inputDesc->mDevice, inputDesc->mPatchHandle);
return inputDesc->mDevice;
}
}
audio_devices_t device = getDeviceForInputSource(inputDesc->mInputSource);
ALOGV("getNewInputDevice() selected device %x", device);
return device;
}
uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
return (uint32_t)getStrategy(stream);
}
audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
// By checking the range of stream before calling getStrategy, we avoid
// getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
// and then return STRATEGY_MEDIA, but we want to return the empty set.
if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_CNT) {
return AUDIO_DEVICE_NONE;
}
audio_devices_t devices;
AudioPolicyManager::routing_strategy strategy = getStrategy(stream);
devices = getDeviceForStrategy(strategy, true /*fromCache*/);
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
for (size_t i = 0; i < outputs.size(); i++) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
if (outputDesc->isStrategyActive(strategy)) {
devices = outputDesc->device();
break;
}
}
/*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
and doesn't really need to.*/
if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
devices |= AUDIO_DEVICE_OUT_SPEAKER;
devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
}
return devices;
}
AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
audio_stream_type_t stream) {
// stream to strategy mapping
switch (stream) {
case AUDIO_STREAM_VOICE_CALL:
case AUDIO_STREAM_BLUETOOTH_SCO:
return STRATEGY_PHONE;
case AUDIO_STREAM_RING:
case AUDIO_STREAM_ALARM:
return STRATEGY_SONIFICATION;
case AUDIO_STREAM_NOTIFICATION:
return STRATEGY_SONIFICATION_RESPECTFUL;
case AUDIO_STREAM_DTMF:
return STRATEGY_DTMF;
default:
ALOGE("unknown stream type");
case AUDIO_STREAM_SYSTEM:
// NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
// while key clicks are played produces a poor result
case AUDIO_STREAM_TTS:
case AUDIO_STREAM_MUSIC:
return STRATEGY_MEDIA;
case AUDIO_STREAM_ENFORCED_AUDIBLE:
return STRATEGY_ENFORCED_AUDIBLE;
}
}
uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
// flags to strategy mapping
if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
}
// usage to strategy mapping
switch (attr->usage) {
case AUDIO_USAGE_MEDIA:
case AUDIO_USAGE_GAME:
case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
return (uint32_t) STRATEGY_MEDIA;
case AUDIO_USAGE_VOICE_COMMUNICATION:
return (uint32_t) STRATEGY_PHONE;
case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
return (uint32_t) STRATEGY_DTMF;
case AUDIO_USAGE_ALARM:
case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
return (uint32_t) STRATEGY_SONIFICATION;
case AUDIO_USAGE_NOTIFICATION:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
case AUDIO_USAGE_NOTIFICATION_EVENT:
return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL;
case AUDIO_USAGE_UNKNOWN:
default:
return (uint32_t) STRATEGY_MEDIA;
}
}
void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
switch(stream) {
case AUDIO_STREAM_MUSIC:
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
updateDevicesAndOutputs();
break;
default:
break;
}
}
audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
bool fromCache)
{
uint32_t device = AUDIO_DEVICE_NONE;
if (fromCache) {
ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
strategy, mDeviceForStrategy[strategy]);
return mDeviceForStrategy[strategy];
}
audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
switch (strategy) {
case STRATEGY_SONIFICATION_RESPECTFUL:
if (isInCall()) {
device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
} else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC,
SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
// while media is playing on a remote device, use the the sonification behavior.
// Note that we test this usecase before testing if media is playing because
// the isStreamActive() method only informs about the activity of a stream, not
// if it's for local playback. Note also that we use the same delay between both tests
device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
//user "safe" speaker if available instead of normal speaker to avoid triggering
//other acoustic safety mechanisms for notification
if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
} else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
// while media is playing (or has recently played), use the same device
device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
} else {
// when media is not playing anymore, fall back on the sonification behavior
device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
//user "safe" speaker if available instead of normal speaker to avoid triggering
//other acoustic safety mechanisms for notification
if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
}
break;
case STRATEGY_DTMF:
if (!isInCall()) {
// when off call, DTMF strategy follows the same rules as MEDIA strategy
device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
break;
}
// when in call, DTMF and PHONE strategies follow the same rules
// FALL THROUGH
case STRATEGY_PHONE:
// Force use of only devices on primary output if:
// - in call AND
// - cannot route from voice call RX OR
// - audio HAL version is < 3.0 and TX device is on the primary HW module
if (mPhoneState == AUDIO_MODE_IN_CALL) {
audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
if (((mAvailableInputDevices.types() &
AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
(((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) &&
(hwOutputDesc->getAudioPort()->mModule->mHalVersion <
AUDIO_DEVICE_API_VERSION_3_0))) {
availableOutputDeviceTypes = availablePrimaryOutputDevices();
}
}
// for phone strategy, we first consider the forced use and then the available devices by order
// of priority
switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
case AUDIO_POLICY_FORCE_BT_SCO:
if (!isInCall() || strategy != STRATEGY_DTMF) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
if (device) break;
}
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
if (device) break;
// if SCO device is requested but no SCO device is available, fall back to default case
// FALL THROUGH
default: // FORCE_NONE
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
if (!isInCall() &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
(getA2dpOutput() != 0) && !mA2dpSuspended) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
if (device) break;
}
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
if (device) break;
if (mPhoneState != AUDIO_MODE_IN_CALL) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
if (device) break;
}
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE;
if (device) break;
device = mDefaultOutputDevice->mDeviceType;
if (device == AUDIO_DEVICE_NONE) {
ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
}
break;
case AUDIO_POLICY_FORCE_SPEAKER:
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
// A2DP speaker when forcing to speaker output
if (!isInCall() &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
(getA2dpOutput() != 0) && !mA2dpSuspended) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
if (device) break;
}
if (mPhoneState != AUDIO_MODE_IN_CALL) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
if (device) break;
}
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
if (device) break;
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
if (device) break;
device = mDefaultOutputDevice->mDeviceType;
if (device == AUDIO_DEVICE_NONE) {
ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
}
break;
}
break;
case STRATEGY_SONIFICATION:
// If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
// handleIncallSonification().
if (isInCall()) {
device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
break;
}
// FALL THROUGH
case STRATEGY_ENFORCED_AUDIBLE:
// strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
// except:
// - when in call where it doesn't default to STRATEGY_PHONE behavior
// - in countries where not enforced in which case it follows STRATEGY_MEDIA
if ((strategy == STRATEGY_SONIFICATION) ||
(mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
if (device == AUDIO_DEVICE_NONE) {
ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
}
}
// The second device used for sonification is the same as the device used by media strategy
// FALL THROUGH
case STRATEGY_MEDIA: {
uint32_t device2 = AUDIO_DEVICE_NONE;
if (strategy != STRATEGY_SONIFICATION) {
// no sonification on remote submix (e.g. WFD)
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
}
if ((device2 == AUDIO_DEVICE_NONE) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
(getA2dpOutput() != 0) && !mA2dpSuspended) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
}
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
}
if ((device2 == AUDIO_DEVICE_NONE)) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
}
if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
// no sonification on aux digital (e.g. HDMI)
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
}
if ((device2 == AUDIO_DEVICE_NONE) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
}
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
}
int device3 = AUDIO_DEVICE_NONE;
if (strategy == STRATEGY_MEDIA) {
// ARC, SPDIF and AUX_LINE can co-exist with others.
device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC;
device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF);
device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE);
}
device2 |= device3;
// device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
// STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
device |= device2;
// If hdmi system audio mode is on, remove speaker out of output list.
if ((strategy == STRATEGY_MEDIA) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] ==
AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
device &= ~AUDIO_DEVICE_OUT_SPEAKER;
}
if (device) break;
device = mDefaultOutputDevice->mDeviceType;
if (device == AUDIO_DEVICE_NONE) {
ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
}
} break;
default:
ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
break;
}
ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
return device;
}
void AudioPolicyManager::updateDevicesAndOutputs()
{
for (int i = 0; i < NUM_STRATEGIES; i++) {
mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
}
mPreviousOutputs = mOutputs;
}
uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
audio_devices_t prevDevice,
uint32_t delayMs)
{
// mute/unmute strategies using an incompatible device combination
// if muting, wait for the audio in pcm buffer to be drained before proceeding
// if unmuting, unmute only after the specified delay
if (outputDesc->isDuplicated()) {
return 0;
}
uint32_t muteWaitMs = 0;
audio_devices_t device = outputDesc->device();
bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
for (size_t i = 0; i < NUM_STRATEGIES; i++) {
audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
bool mute = shouldMute && (curDevice & device) && (curDevice != device);
bool doMute = false;
if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
doMute = true;
outputDesc->mStrategyMutedByDevice[i] = true;
} else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
doMute = true;
outputDesc->mStrategyMutedByDevice[i] = false;
}
if (doMute) {
for (size_t j = 0; j < mOutputs.size(); j++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
// skip output if it does not share any device with current output
if ((desc->supportedDevices() & outputDesc->supportedDevices())
== AUDIO_DEVICE_NONE) {
continue;
}
audio_io_handle_t curOutput = mOutputs.keyAt(j);
ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
mute ? "muting" : "unmuting", i, curDevice, curOutput);
setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
if (desc->isStrategyActive((routing_strategy)i)) {
if (mute) {
// FIXME: should not need to double latency if volume could be applied
// immediately by the audioflinger mixer. We must account for the delay
// between now and the next time the audioflinger thread for this output
// will process a buffer (which corresponds to one buffer size,
// usually 1/2 or 1/4 of the latency).
if (muteWaitMs < desc->latency() * 2) {
muteWaitMs = desc->latency() * 2;
}
}
}
}
}
}
// temporary mute output if device selection changes to avoid volume bursts due to
// different per device volumes
if (outputDesc->isActive() && (device != prevDevice)) {
if (muteWaitMs < outputDesc->latency() * 2) {
muteWaitMs = outputDesc->latency() * 2;
}
for (size_t i = 0; i < NUM_STRATEGIES; i++) {
if (outputDesc->isStrategyActive((routing_strategy)i)) {
setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle);
// do tempMute unmute after twice the mute wait time
setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle,
muteWaitMs *2, device);
}
}
}
// wait for the PCM output buffers to empty before proceeding with the rest of the command
if (muteWaitMs > delayMs) {
muteWaitMs -= delayMs;
usleep(muteWaitMs * 1000);
return muteWaitMs;
}
return 0;
}
uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
audio_devices_t device,
bool force,
int delayMs,
audio_patch_handle_t *patchHandle,
const char* address)
{
ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
AudioParameter param;
uint32_t muteWaitMs;
if (outputDesc->isDuplicated()) {
muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs);
muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs);
return muteWaitMs;
}
// no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
// output profile
if ((device != AUDIO_DEVICE_NONE) &&
((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) {
return 0;
}
// filter devices according to output selected
device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types());
audio_devices_t prevDevice = outputDesc->mDevice;
ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
if (device != AUDIO_DEVICE_NONE) {
outputDesc->mDevice = device;
}
muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
// Do not change the routing if:
// - the requested device is AUDIO_DEVICE_NONE
// - the requested device is the same as current device and force is not specified.
// Doing this check here allows the caller to call setOutputDevice() without conditions
if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) {
ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
return muteWaitMs;
}
ALOGV("setOutputDevice() changing device");
// do the routing
if (device == AUDIO_DEVICE_NONE) {
resetOutputDevice(output, delayMs, NULL);
} else {
DeviceVector deviceList = (address == NULL) ?
mAvailableOutputDevices.getDevicesFromType(device)
: mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address));
if (!deviceList.isEmpty()) {
struct audio_patch patch;
outputDesc->toAudioPortConfig(&patch.sources[0]);
patch.num_sources = 1;
patch.num_sinks = 0;
for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
patch.num_sinks++;
}
ssize_t index;
if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
}
sp< AudioPatch> patchDesc;
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (index >= 0) {
patchDesc = mAudioPatches.valueAt(index);
afPatchHandle = patchDesc->mAfPatchHandle;
}
status_t status = mpClientInterface->createAudioPatch(&patch,
&afPatchHandle,
delayMs);
ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
"num_sources %d num_sinks %d",
status, afPatchHandle, patch.num_sources, patch.num_sinks);
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
&patch, mUidCached);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = patch;
}
patchDesc->mAfPatchHandle = afPatchHandle;
patchDesc->mUid = mUidCached;
if (patchHandle) {
*patchHandle = patchDesc->mHandle;
}
outputDesc->mPatchHandle = patchDesc->mHandle;
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
}
}
// inform all input as well
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
if (!isVirtualInputDevice(inputDescriptor->mDevice)) {
AudioParameter inputCmd = AudioParameter();
ALOGV("%s: inform input %d of device:%d", __func__,
inputDescriptor->mIoHandle, device);
inputCmd.addInt(String8(AudioParameter::keyRouting),device);
mpClientInterface->setParameters(inputDescriptor->mIoHandle,
inputCmd.toString(),
delayMs);
}
}
}
// update stream volumes according to new device
applyStreamVolumes(output, device, delayMs);
return muteWaitMs;
}
status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output,
int delayMs,
audio_patch_handle_t *patchHandle)
{
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ssize_t index;
if (patchHandle) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
}
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->mPatchHandle = 0;
removeAudioPatch(patchDesc->mHandle);
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
return status;
}
status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
audio_devices_t device,
bool force,
audio_patch_handle_t *patchHandle)
{
status_t status = NO_ERROR;
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
inputDesc->mDevice = device;
DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
if (!deviceList.isEmpty()) {
struct audio_patch patch;
inputDesc->toAudioPortConfig(&patch.sinks[0]);
// AUDIO_SOURCE_HOTWORD is for internal use only:
// handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
!inputDesc->mIsSoundTrigger) {
patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
}
patch.num_sinks = 1;
//only one input device for now
deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
patch.num_sources = 1;
ssize_t index;
if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
}
sp< AudioPatch> patchDesc;
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (index >= 0) {
patchDesc = mAudioPatches.valueAt(index);
afPatchHandle = patchDesc->mAfPatchHandle;
}
status_t status = mpClientInterface->createAudioPatch(&patch,
&afPatchHandle,
0);
ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
status, afPatchHandle);
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
&patch, mUidCached);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = patch;
}
patchDesc->mAfPatchHandle = afPatchHandle;
patchDesc->mUid = mUidCached;
if (patchHandle) {
*patchHandle = patchDesc->mHandle;
}
inputDesc->mPatchHandle = patchDesc->mHandle;
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
}
}
}
return status;
}
status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input,
audio_patch_handle_t *patchHandle)
{
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
ssize_t index;
if (patchHandle) {
index = mAudioPatches.indexOfKey(*patchHandle);
} else {
index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
}
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->mPatchHandle = 0;
removeAudioPatch(patchDesc->mHandle);
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
return status;
}
sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t 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.
for (size_t i = 0; i < mHwModules.size(); i++)
{
if (mHwModules[i]->mHandle == 0) {
continue;
}
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
{
sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
// profile->log();
if (profile->isCompatibleProfile(device, samplingRate,
&samplingRate /*updatedSamplingRate*/,
format, channelMask, (audio_output_flags_t) flags)) {
return profile;
}
}
}
return NULL;
}
audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
{
uint32_t device = AUDIO_DEVICE_NONE;
audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
~AUDIO_DEVICE_BIT_IN;
switch (inputSource) {
case AUDIO_SOURCE_VOICE_UPLINK:
if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
device = AUDIO_DEVICE_IN_VOICE_CALL;
break;
}
break;
case AUDIO_SOURCE_DEFAULT:
case AUDIO_SOURCE_MIC:
if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
device = AUDIO_DEVICE_IN_WIRED_HEADSET;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
device = AUDIO_DEVICE_IN_USB_DEVICE;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
device = AUDIO_DEVICE_IN_BUILTIN_MIC;
}
break;
case AUDIO_SOURCE_VOICE_COMMUNICATION:
// Allow only use of devices on primary input if in call and HAL does not support routing
// to voice call path.
if ((mPhoneState == AUDIO_MODE_IN_CALL) &&
(mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN;
}
switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
case AUDIO_POLICY_FORCE_BT_SCO:
// if SCO device is requested but no SCO device is available, fall back to default case
if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
break;
}
// FALL THROUGH
default: // FORCE_NONE
if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
device = AUDIO_DEVICE_IN_WIRED_HEADSET;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
device = AUDIO_DEVICE_IN_USB_DEVICE;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
device = AUDIO_DEVICE_IN_BUILTIN_MIC;
}
break;
case AUDIO_POLICY_FORCE_SPEAKER:
if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
device = AUDIO_DEVICE_IN_BACK_MIC;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
device = AUDIO_DEVICE_IN_BUILTIN_MIC;
}
break;
}
break;
case AUDIO_SOURCE_VOICE_RECOGNITION:
case AUDIO_SOURCE_HOTWORD:
if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO &&
availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
device = AUDIO_DEVICE_IN_WIRED_HEADSET;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
device = AUDIO_DEVICE_IN_USB_DEVICE;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
device = AUDIO_DEVICE_IN_BUILTIN_MIC;
}
break;
case AUDIO_SOURCE_CAMCORDER:
if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
device = AUDIO_DEVICE_IN_BACK_MIC;
} else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
device = AUDIO_DEVICE_IN_BUILTIN_MIC;
}
break;
case AUDIO_SOURCE_VOICE_DOWNLINK:
case AUDIO_SOURCE_VOICE_CALL:
if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
device = AUDIO_DEVICE_IN_VOICE_CALL;
}
break;
case AUDIO_SOURCE_REMOTE_SUBMIX:
if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
}
break;
default:
ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
break;
}
ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
return device;
}
bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device)
{
if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
device &= ~AUDIO_DEVICE_BIT_IN;
if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
return true;
}
return false;
}
bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) {
return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL) != 0);
}
audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
{
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> input_descriptor = mInputs.valueAt(i);
if ((input_descriptor->mRefCount > 0)
&& (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
return mInputs.keyAt(i);
}
}
return 0;
}
uint32_t AudioPolicyManager::activeInputsCount() const
{
uint32_t count = 0;
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> desc = mInputs.valueAt(i);
if (desc->mRefCount > 0) {
return count++;
}
}
return count;
}
audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
{
if (device == AUDIO_DEVICE_NONE) {
// this happens when forcing a route update and no track is active on an output.
// In this case the returned category is not important.
device = AUDIO_DEVICE_OUT_SPEAKER;
} else if (popcount(device) > 1) {
// Multiple device selection is either:
// - speaker + one other device: give priority to speaker in this case.
// - one A2DP device + another device: happens with duplicated output. In this case
// retain the device on the A2DP output as the other must not correspond to an active
// selection if not the speaker.
// - HDMI-CEC system audio mode only output: give priority to available item in order.
if (device & AUDIO_DEVICE_OUT_SPEAKER) {
device = AUDIO_DEVICE_OUT_SPEAKER;
} else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
device = AUDIO_DEVICE_OUT_HDMI_ARC;
} else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
device = AUDIO_DEVICE_OUT_AUX_LINE;
} else if (device & AUDIO_DEVICE_OUT_SPDIF) {
device = AUDIO_DEVICE_OUT_SPDIF;
} else {
device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
}
}
/*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
device = AUDIO_DEVICE_OUT_SPEAKER;
ALOGW_IF(popcount(device) != 1,
"getDeviceForVolume() invalid device combination: %08x",
device);
return device;
}
AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
{
switch(getDeviceForVolume(device)) {
case AUDIO_DEVICE_OUT_EARPIECE:
return DEVICE_CATEGORY_EARPIECE;
case AUDIO_DEVICE_OUT_WIRED_HEADSET:
case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
return DEVICE_CATEGORY_HEADSET;
case AUDIO_DEVICE_OUT_LINE:
case AUDIO_DEVICE_OUT_AUX_DIGITAL:
/*USB? Remote submix?*/
return DEVICE_CATEGORY_EXT_MEDIA;
case AUDIO_DEVICE_OUT_SPEAKER:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
case AUDIO_DEVICE_OUT_USB_ACCESSORY:
case AUDIO_DEVICE_OUT_USB_DEVICE:
case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
default:
return DEVICE_CATEGORY_SPEAKER;
}
}
float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
int indexInUi)
{
device_category deviceCategory = getDeviceCategory(device);
const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
// the volume index in the UI is relative to the min and max volume indices for this stream type
int nbSteps = 1 + curve[VOLMAX].mIndex -
curve[VOLMIN].mIndex;
int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
(streamDesc.mIndexMax - streamDesc.mIndexMin);
// find what part of the curve this index volume belongs to, or if it's out of bounds
int segment = 0;
if (volIdx < curve[VOLMIN].mIndex) { // out of bounds
return 0.0f;
} else if (volIdx < curve[VOLKNEE1].mIndex) {
segment = 0;
} else if (volIdx < curve[VOLKNEE2].mIndex) {
segment = 1;
} else if (volIdx <= curve[VOLMAX].mIndex) {
segment = 2;
} else { // out of bounds
return 1.0f;
}
// linear interpolation in the attenuation table in dB
float decibels = curve[segment].mDBAttenuation +
((float)(volIdx - curve[segment].mIndex)) *
( (curve[segment+1].mDBAttenuation -
curve[segment].mDBAttenuation) /
((float)(curve[segment+1].mIndex -
curve[segment].mIndex)) );
float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
curve[segment].mIndex, volIdx,
curve[segment+1].mIndex,
curve[segment].mDBAttenuation,
decibels,
curve[segment+1].mDBAttenuation,
amplification);
return amplification;
}
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = {
{1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
{1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
{1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
{1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
{1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = {
{1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
{1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
};
// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
{1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
{1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
{1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
{0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
};
const AudioPolicyManager::VolumeCurvePoint
AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
{0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
};
const AudioPolicyManager::VolumeCurvePoint
*AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT]
[AudioPolicyManager::DEVICE_CATEGORY_CNT] = {
{ // AUDIO_STREAM_VOICE_CALL
sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_SYSTEM
sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_RING
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_MUSIC
sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_ALARM
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_NOTIFICATION
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_BLUETOOTH_SCO
sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_ENFORCED_AUDIBLE
sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_DTMF
sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
{ // AUDIO_STREAM_TTS
sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
},
};
void AudioPolicyManager::initializeVolumeCurves()
{
for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
mStreams[i].mVolumeCurve[j] =
sVolumeProfiles[i][j];
}
}
// Check availability of DRC on speaker path: if available, override some of the speaker curves
if (mSpeakerDrcEnabled) {
mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
sDefaultSystemVolumeCurveDrc;
mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
sSpeakerSonificationVolumeCurveDrc;
mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
sSpeakerSonificationVolumeCurveDrc;
mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
sSpeakerSonificationVolumeCurveDrc;
mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
sSpeakerMediaVolumeCurveDrc;
}
}
float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
int index,
audio_io_handle_t output,
audio_devices_t device)
{
float volume = 1.0;
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
StreamDescriptor &streamDesc = mStreams[stream];
if (device == AUDIO_DEVICE_NONE) {
device = outputDesc->device();
}
volume = volIndexToAmpl(device, streamDesc, index);
// if a headset is connected, apply the following rules to ring tones and notifications
// to avoid sound level bursts in user's ears:
// - always attenuate ring tones and notifications volume by 6dB
// - if music is playing, always limit the volume to current music volume,
// with a minimum threshold at -36dB so that notification is always perceived.
const routing_strategy stream_strategy = getStrategy(stream);
if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
AUDIO_DEVICE_OUT_WIRED_HEADSET |
AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
((stream_strategy == STRATEGY_SONIFICATION)
|| (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
|| (stream == AUDIO_STREAM_SYSTEM)
|| ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) &&
streamDesc.mCanBeMuted) {
volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
// when the phone is ringing we must consider that music could have been paused just before
// by the music application and behave as if music was active if the last music track was
// just stopped
if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
mLimitRingtoneVolume) {
audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
output,
musicDevice);
float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
if (volume > minVol) {
volume = minVol;
ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
}
}
}
return volume;
}
status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
int index,
audio_io_handle_t output,
audio_devices_t device,
int delayMs,
bool force)
{
// do not change actual stream volume if the stream is muted
if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
ALOGVV("checkAndSetVolume() stream %d muted count %d",
stream, mOutputs.valueFor(output)->mMuteCount[stream]);
return NO_ERROR;
}
// do not change in call volume if bluetooth is connected and vice versa
if ((stream == AUDIO_STREAM_VOICE_CALL &&
mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
(stream == AUDIO_STREAM_BLUETOOTH_SCO &&
mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) {
ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
return INVALID_OPERATION;
}
float volume = computeVolume(stream, index, output, device);
// We actually change the volume if:
// - the float value returned by computeVolume() changed
// - the force flag is set
if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
force) {
mOutputs.valueFor(output)->mCurVolume[stream] = volume;
ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
// Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
// enabled
if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
}
mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
}
if (stream == AUDIO_STREAM_VOICE_CALL ||
stream == AUDIO_STREAM_BLUETOOTH_SCO) {
float voiceVolume;
// Force voice volume to max for bluetooth SCO as volume is managed by the headset
if (stream == AUDIO_STREAM_VOICE_CALL) {
voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
} else {
voiceVolume = 1.0;
}
if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
mLastVoiceVolume = voiceVolume;
}
}
return NO_ERROR;
}
void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
audio_devices_t device,
int delayMs,
bool force)
{
ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
checkAndSetVolume((audio_stream_type_t)stream,
mStreams[stream].getVolumeIndex(device),
output,
device,
delayMs,
force);
}
}
void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
bool on,
audio_io_handle_t output,
int delayMs,
audio_devices_t device)
{
ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
if (getStrategy((audio_stream_type_t)stream) == strategy) {
setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
}
}
}
void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
bool on,
audio_io_handle_t output,
int delayMs,
audio_devices_t device)
{
StreamDescriptor &streamDesc = mStreams[stream];
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (device == AUDIO_DEVICE_NONE) {
device = outputDesc->device();
}
ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
stream, on, output, outputDesc->mMuteCount[stream], device);
if (on) {
if (outputDesc->mMuteCount[stream] == 0) {
if (streamDesc.mCanBeMuted &&
((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
(mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) {
checkAndSetVolume(stream, 0, output, device, delayMs);
}
}
// increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
outputDesc->mMuteCount[stream]++;
} else {
if (outputDesc->mMuteCount[stream] == 0) {
ALOGV("setStreamMute() unmuting non muted stream!");
return;
}
if (--outputDesc->mMuteCount[stream] == 0) {
checkAndSetVolume(stream,
streamDesc.getVolumeIndex(device),
output,
device,
delayMs);
}
}
}
void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
bool starting, bool stateChange)
{
// if the stream pertains to sonification strategy and we are in call we must
// mute the stream if it is low visibility. If it is high visibility, we must play a tone
// in the device used for phone strategy and play the tone if the selected device does not
// interfere with the device used for phone strategy
// if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
// many times as there are active tracks on the output
const routing_strategy stream_strategy = getStrategy(stream);
if ((stream_strategy == STRATEGY_SONIFICATION) ||
((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
stream, starting, outputDesc->mDevice, stateChange);
if (outputDesc->mRefCount[stream]) {
int muteCount = 1;
if (stateChange) {
muteCount = outputDesc->mRefCount[stream];
}
if (audio_is_low_visibility(stream)) {
ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
for (int i = 0; i < muteCount; i++) {
setStreamMute(stream, starting, mPrimaryOutput);
}
} else {
ALOGV("handleIncallSonification() high visibility");
if (outputDesc->device() &
getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
for (int i = 0; i < muteCount; i++) {
setStreamMute(stream, starting, mPrimaryOutput);
}
}
if (starting) {
mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
AUDIO_STREAM_VOICE_CALL);
} else {
mpClientInterface->stopTone();
}
}
}
}
}
bool AudioPolicyManager::isInCall()
{
return isStateInCall(mPhoneState);
}
bool AudioPolicyManager::isStateInCall(int state) {
return ((state == AUDIO_MODE_IN_CALL) ||
(state == AUDIO_MODE_IN_COMMUNICATION));
}
uint32_t AudioPolicyManager::getMaxEffectsCpuLoad()
{
return MAX_EFFECTS_CPU_LOAD;
}
uint32_t AudioPolicyManager::getMaxEffectsMemory()
{
return MAX_EFFECTS_MEMORY;
}
// --- AudioOutputDescriptor class implementation
AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
const sp<IOProfile>& profile)
: mId(0), mIoHandle(0), mLatency(0),
mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0),
mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
{
// clear usage count for all stream types
for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
mRefCount[i] = 0;
mCurVolume[i] = -1.0;
mMuteCount[i] = 0;
mStopTime[i] = 0;
}
for (int i = 0; i < NUM_STRATEGIES; i++) {
mStrategyMutedByDevice[i] = false;
}
if (profile != NULL) {
mFlags = (audio_output_flags_t)profile->mFlags;
mSamplingRate = profile->pickSamplingRate();
mFormat = profile->pickFormat();
mChannelMask = profile->pickChannelMask();
if (profile->mGains.size() > 0) {
profile->mGains[0]->getDefaultConfig(&mGain);
}
}
}
audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const
{
if (isDuplicated()) {
return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
} else {
return mDevice;
}
}
uint32_t AudioPolicyManager::AudioOutputDescriptor::latency()
{
if (isDuplicated()) {
return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
} else {
return mLatency;
}
}
bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
const sp<AudioOutputDescriptor> outputDesc)
{
if (isDuplicated()) {
return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
} else if (outputDesc->isDuplicated()){
return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
} else {
return (mProfile->mModule == outputDesc->mProfile->mModule);
}
}
void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
int delta)
{
// forward usage count change to attached outputs
if (isDuplicated()) {
mOutput1->changeRefCount(stream, delta);
mOutput2->changeRefCount(stream, delta);
}
if ((delta + (int)mRefCount[stream]) < 0) {
ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
delta, stream, mRefCount[stream]);
mRefCount[stream] = 0;
return;
}
mRefCount[stream] += delta;
ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
}
audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices()
{
if (isDuplicated()) {
return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
} else {
return mProfile->mSupportedDevices.types() ;
}
}
bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
{
return isStrategyActive(NUM_STRATEGIES, inPastMs);
}
bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
uint32_t inPastMs,
nsecs_t sysTime) const
{
if ((sysTime == 0) && (inPastMs != 0)) {
sysTime = systemTime();
}
for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
if (((getStrategy((audio_stream_type_t)i) == strategy) ||
(NUM_STRATEGIES == strategy)) &&
isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
return true;
}
}
return false;
}
bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
uint32_t inPastMs,
nsecs_t sysTime) const
{
if (mRefCount[stream] != 0) {
return true;
}
if (inPastMs == 0) {
return false;
}
if (sysTime == 0) {
sysTime = systemTime();
}
if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
return true;
}
return false;
}
void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig(
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
if (srcConfig != NULL) {
dstConfig->config_mask |= srcConfig->config_mask;
}
AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
dstConfig->id = mId;
dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
dstConfig->type = AUDIO_PORT_TYPE_MIX;
dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
dstConfig->ext.mix.handle = mIoHandle;
dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
}
void AudioPolicyManager::AudioOutputDescriptor::toAudioPort(
struct audio_port *port) const
{
ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
mProfile->toAudioPort(port);
port->id = mId;
toAudioPortConfig(&port->active_config);
port->ext.mix.hw_module = mProfile->mModule->mHandle;
port->ext.mix.handle = mIoHandle;
port->ext.mix.latency_class =
mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
}
status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, " ID: %d\n", mId);
result.append(buffer);
snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
result.append(buffer);
snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
result.append(buffer);
snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
result.append(buffer);
snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
result.append(buffer);
snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
result.append(buffer);
snprintf(buffer, SIZE, " Devices %08x\n", device());
result.append(buffer);
snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
result.append(buffer);
for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
result.append(buffer);
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
// --- AudioInputDescriptor class implementation
AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
: mId(0), mIoHandle(0),
mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0), mRefCount(0),
mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
{
if (profile != NULL) {
mSamplingRate = profile->pickSamplingRate();
mFormat = profile->pickFormat();
mChannelMask = profile->pickChannelMask();
if (profile->mGains.size() > 0) {
profile->mGains[0]->getDefaultConfig(&mGain);
}
}
}
void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig(
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
ALOG_ASSERT(mProfile != 0,
"toAudioPortConfig() called on input with null profile %d", mIoHandle);
dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
if (srcConfig != NULL) {
dstConfig->config_mask |= srcConfig->config_mask;
}
AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
dstConfig->id = mId;
dstConfig->role = AUDIO_PORT_ROLE_SINK;
dstConfig->type = AUDIO_PORT_TYPE_MIX;
dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
dstConfig->ext.mix.handle = mIoHandle;
dstConfig->ext.mix.usecase.source = mInputSource;
}
void AudioPolicyManager::AudioInputDescriptor::toAudioPort(
struct audio_port *port) const
{
ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
mProfile->toAudioPort(port);
port->id = mId;
toAudioPortConfig(&port->active_config);
port->ext.mix.hw_module = mProfile->mModule->mHandle;
port->ext.mix.handle = mIoHandle;
port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
}
status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, " ID: %d\n", mId);
result.append(buffer);
snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
result.append(buffer);
snprintf(buffer, SIZE, " Format: %d\n", mFormat);
result.append(buffer);
snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
result.append(buffer);
snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
result.append(buffer);
snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
result.append(buffer);
snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
// --- StreamDescriptor class implementation
AudioPolicyManager::StreamDescriptor::StreamDescriptor()
: mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
{
mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
}
int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device)
{
device = AudioPolicyManager::getDeviceForVolume(device);
// there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
if (mIndexCur.indexOfKey(device) < 0) {
device = AUDIO_DEVICE_OUT_DEFAULT;
}
return mIndexCur.valueFor(device);
}
void AudioPolicyManager::StreamDescriptor::dump(int fd)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "%s %02d %02d ",
mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
result.append(buffer);
for (size_t i = 0; i < mIndexCur.size(); i++) {
snprintf(buffer, SIZE, "%04x : %02d, ",
mIndexCur.keyAt(i),
mIndexCur.valueAt(i));
result.append(buffer);
}
result.append("\n");
write(fd, result.string(), result.size());
}
// --- EffectDescriptor class implementation
status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, " I/O: %d\n", mIo);
result.append(buffer);
snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
result.append(buffer);
snprintf(buffer, SIZE, " Session: %d\n", mSession);
result.append(buffer);
snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
result.append(buffer);
snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
// --- HwModule class implementation
AudioPolicyManager::HwModule::HwModule(const char *name)
: mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
{
}
AudioPolicyManager::HwModule::~HwModule()
{
for (size_t i = 0; i < mOutputProfiles.size(); i++) {
mOutputProfiles[i]->mSupportedDevices.clear();
}
for (size_t i = 0; i < mInputProfiles.size(); i++) {
mInputProfiles[i]->mSupportedDevices.clear();
}
free((void *)mName);
}
status_t AudioPolicyManager::HwModule::loadInput(cnode *root)
{
cnode *node = root->first_child;
sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
while (node) {
if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
profile->loadSamplingRates((char *)node->value);
} else if (strcmp(node->name, FORMATS_TAG) == 0) {
profile->loadFormats((char *)node->value);
} else if (strcmp(node->name, CHANNELS_TAG) == 0) {
profile->loadInChannels((char *)node->value);
} else if (strcmp(node->name, DEVICES_TAG) == 0) {
profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
mDeclaredDevices);
} else if (strcmp(node->name, FLAGS_TAG) == 0) {
profile->mFlags = parseInputFlagNames((char *)node->value);
} else if (strcmp(node->name, GAINS_TAG) == 0) {
profile->loadGains(node);
}
node = node->next;
}
ALOGW_IF(profile->mSupportedDevices.isEmpty(),
"loadInput() invalid supported devices");
ALOGW_IF(profile->mChannelMasks.size() == 0,
"loadInput() invalid supported channel masks");
ALOGW_IF(profile->mSamplingRates.size() == 0,
"loadInput() invalid supported sampling rates");
ALOGW_IF(profile->mFormats.size() == 0,
"loadInput() invalid supported formats");
if (!profile->mSupportedDevices.isEmpty() &&
(profile->mChannelMasks.size() != 0) &&
(profile->mSamplingRates.size() != 0) &&
(profile->mFormats.size() != 0)) {
ALOGV("loadInput() adding input Supported Devices %04x",
profile->mSupportedDevices.types());
mInputProfiles.add(profile);
return NO_ERROR;
} else {
return BAD_VALUE;
}
}
status_t AudioPolicyManager::HwModule::loadOutput(cnode *root)
{
cnode *node = root->first_child;
sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
while (node) {
if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
profile->loadSamplingRates((char *)node->value);
} else if (strcmp(node->name, FORMATS_TAG) == 0) {
profile->loadFormats((char *)node->value);
} else if (strcmp(node->name, CHANNELS_TAG) == 0) {
profile->loadOutChannels((char *)node->value);
} else if (strcmp(node->name, DEVICES_TAG) == 0) {
profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
mDeclaredDevices);
} else if (strcmp(node->name, FLAGS_TAG) == 0) {
profile->mFlags = parseOutputFlagNames((char *)node->value);
} else if (strcmp(node->name, GAINS_TAG) == 0) {
profile->loadGains(node);
}
node = node->next;
}
ALOGW_IF(profile->mSupportedDevices.isEmpty(),
"loadOutput() invalid supported devices");
ALOGW_IF(profile->mChannelMasks.size() == 0,
"loadOutput() invalid supported channel masks");
ALOGW_IF(profile->mSamplingRates.size() == 0,
"loadOutput() invalid supported sampling rates");
ALOGW_IF(profile->mFormats.size() == 0,
"loadOutput() invalid supported formats");
if (!profile->mSupportedDevices.isEmpty() &&
(profile->mChannelMasks.size() != 0) &&
(profile->mSamplingRates.size() != 0) &&
(profile->mFormats.size() != 0)) {
ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
profile->mSupportedDevices.types(), profile->mFlags);
mOutputProfiles.add(profile);
return NO_ERROR;
} else {
return BAD_VALUE;
}
}
status_t AudioPolicyManager::HwModule::loadDevice(cnode *root)
{
cnode *node = root->first_child;
audio_devices_t type = AUDIO_DEVICE_NONE;
while (node) {
if (strcmp(node->name, DEVICE_TYPE) == 0) {
type = parseDeviceNames((char *)node->value);
break;
}
node = node->next;
}
if (type == AUDIO_DEVICE_NONE ||
(!audio_is_input_device(type) && !audio_is_output_device(type))) {
ALOGW("loadDevice() bad type %08x", type);
return BAD_VALUE;
}
sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
deviceDesc->mModule = this;
node = root->first_child;
while (node) {
if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
deviceDesc->mAddress = String8((char *)node->value);
} else if (strcmp(node->name, CHANNELS_TAG) == 0) {
if (audio_is_input_device(type)) {
deviceDesc->loadInChannels((char *)node->value);
} else {
deviceDesc->loadOutChannels((char *)node->value);
}
} else if (strcmp(node->name, GAINS_TAG) == 0) {
deviceDesc->loadGains(node);
}
node = node->next;
}
ALOGV("loadDevice() adding device name %s type %08x address %s",
deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
mDeclaredDevices.add(deviceDesc);
return NO_ERROR;
}
void AudioPolicyManager::HwModule::dump(int fd)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, " - name: %s\n", mName);
result.append(buffer);
snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
result.append(buffer);
snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
result.append(buffer);
write(fd, result.string(), result.size());
if (mOutputProfiles.size()) {
write(fd, " - outputs:\n", strlen(" - outputs:\n"));
for (size_t i = 0; i < mOutputProfiles.size(); i++) {
snprintf(buffer, SIZE, " output %zu:\n", i);
write(fd, buffer, strlen(buffer));
mOutputProfiles[i]->dump(fd);
}
}
if (mInputProfiles.size()) {
write(fd, " - inputs:\n", strlen(" - inputs:\n"));
for (size_t i = 0; i < mInputProfiles.size(); i++) {
snprintf(buffer, SIZE, " input %zu:\n", i);
write(fd, buffer, strlen(buffer));
mInputProfiles[i]->dump(fd);
}
}
if (mDeclaredDevices.size()) {
write(fd, " - devices:\n", strlen(" - devices:\n"));
for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
mDeclaredDevices[i]->dump(fd, 4, i);
}
}
}
// --- AudioPort class implementation
AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type,
audio_port_role_t role, const sp<HwModule>& module) :
mName(name), mType(type), mRole(role), mModule(module), mFlags(0)
{
mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
}
void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const
{
port->role = mRole;
port->type = mType;
unsigned int i;
for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
if (mSamplingRates[i] != 0) {
port->sample_rates[i] = mSamplingRates[i];
}
}
port->num_sample_rates = i;
for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
if (mChannelMasks[i] != 0) {
port->channel_masks[i] = mChannelMasks[i];
}
}
port->num_channel_masks = i;
for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
if (mFormats[i] != 0) {
port->formats[i] = mFormats[i];
}
}
port->num_formats = i;
ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
port->gains[i] = mGains[i]->mGain;
}
port->num_gains = i;
}
void AudioPolicyManager::AudioPort::importAudioPort(const sp<AudioPort> port) {
for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
const uint32_t rate = port->mSamplingRates.itemAt(k);
if (rate != 0) { // skip "dynamic" rates
bool hasRate = false;
for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
if (rate == mSamplingRates.itemAt(l)) {
hasRate = true;
break;
}
}
if (!hasRate) { // never import a sampling rate twice
mSamplingRates.add(rate);
}
}
}
for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
if (mask != 0) { // skip "dynamic" masks
bool hasMask = false;
for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
if (mask == mChannelMasks.itemAt(l)) {
hasMask = true;
break;
}
}
if (!hasMask) { // never import a channel mask twice
mChannelMasks.add(mask);
}
}
}
for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
const audio_format_t format = port->mFormats.itemAt(k);
if (format != 0) { // skip "dynamic" formats
bool hasFormat = false;
for (size_t l = 0 ; l < mFormats.size() ; l++) {
if (format == mFormats.itemAt(l)) {
hasFormat = true;
break;
}
}
if (!hasFormat) { // never import a channel mask twice
mFormats.add(format);
}
}
}
}
void AudioPolicyManager::AudioPort::clearCapabilities() {
mChannelMasks.clear();
mFormats.clear();
mSamplingRates.clear();
}
void AudioPolicyManager::AudioPort::loadSamplingRates(char *name)
{
char *str = strtok(name, "|");
// by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
// rates should be read from the output stream after it is opened for the first time
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
mSamplingRates.add(0);
return;
}
while (str != NULL) {
uint32_t rate = atoi(str);
if (rate != 0) {
ALOGV("loadSamplingRates() adding rate %d", rate);
mSamplingRates.add(rate);
}
str = strtok(NULL, "|");
}
}
void AudioPolicyManager::AudioPort::loadFormats(char *name)
{
char *str = strtok(name, "|");
// by convention, "0' in the first entry in mFormats indicates the supported formats
// should be read from the output stream after it is opened for the first time
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
mFormats.add(AUDIO_FORMAT_DEFAULT);
return;
}
while (str != NULL) {
audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
ARRAY_SIZE(sFormatNameToEnumTable),
str);
if (format != AUDIO_FORMAT_DEFAULT) {
mFormats.add(format);
}
str = strtok(NULL, "|");
}
}
void AudioPolicyManager::AudioPort::loadInChannels(char *name)
{
const char *str = strtok(name, "|");
ALOGV("loadInChannels() %s", name);
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
mChannelMasks.add(0);
return;
}
while (str != NULL) {
audio_channel_mask_t channelMask =
(audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
ARRAY_SIZE(sInChannelsNameToEnumTable),
str);
if (channelMask != 0) {
ALOGV("loadInChannels() adding channelMask %04x", channelMask);
mChannelMasks.add(channelMask);
}
str = strtok(NULL, "|");
}
}
void AudioPolicyManager::AudioPort::loadOutChannels(char *name)
{
const char *str = strtok(name, "|");
ALOGV("loadOutChannels() %s", name);
// by convention, "0' in the first entry in mChannelMasks indicates the supported channel
// masks should be read from the output stream after it is opened for the first time
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
mChannelMasks.add(0);
return;
}
while (str != NULL) {
audio_channel_mask_t channelMask =
(audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
ARRAY_SIZE(sOutChannelsNameToEnumTable),
str);
if (channelMask != 0) {
mChannelMasks.add(channelMask);
}
str = strtok(NULL, "|");
}
return;
}
audio_gain_mode_t AudioPolicyManager::AudioPort::loadGainMode(char *name)
{
const char *str = strtok(name, "|");
ALOGV("loadGainMode() %s", name);
audio_gain_mode_t mode = 0;
while (str != NULL) {
mode |= (audio_gain_mode_t)stringToEnum(sGainModeNameToEnumTable,
ARRAY_SIZE(sGainModeNameToEnumTable),
str);
str = strtok(NULL, "|");
}
return mode;
}
void AudioPolicyManager::AudioPort::loadGain(cnode *root, int index)
{
cnode *node = root->first_child;
sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
while (node) {
if (strcmp(node->name, GAIN_MODE) == 0) {
gain->mGain.mode = loadGainMode((char *)node->value);
} else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
if (mUseInChannelMask) {
gain->mGain.channel_mask =
(audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
ARRAY_SIZE(sInChannelsNameToEnumTable),
(char *)node->value);
} else {
gain->mGain.channel_mask =
(audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
ARRAY_SIZE(sOutChannelsNameToEnumTable),
(char *)node->value);
}
} else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
gain->mGain.min_value = atoi((char *)node->value);
} else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
gain->mGain.max_value = atoi((char *)node->value);
} else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
gain->mGain.default_value = atoi((char *)node->value);
} else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
gain->mGain.step_value = atoi((char *)node->value);
} else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
gain->mGain.min_ramp_ms = atoi((char *)node->value);
} else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
gain->mGain.max_ramp_ms = atoi((char *)node->value);
}
node = node->next;
}
ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
if (gain->mGain.mode == 0) {
return;
}
mGains.add(gain);
}
void AudioPolicyManager::AudioPort::loadGains(cnode *root)
{
cnode *node = root->first_child;
int index = 0;
while (node) {
ALOGV("loadGains() loading gain %s", node->name);
loadGain(node, index++);
node = node->next;
}
}
status_t AudioPolicyManager::AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
{
for (size_t i = 0; i < mSamplingRates.size(); i ++) {
if (mSamplingRates[i] == samplingRate) {
return NO_ERROR;
}
}
return BAD_VALUE;
}
status_t AudioPolicyManager::AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
uint32_t *updatedSamplingRate) const
{
// Search for the closest supported sampling rate that is above (preferred)
// or below (acceptable) the desired sampling rate, within a permitted ratio.
// The sampling rates do not need to be sorted in ascending order.
ssize_t maxBelow = -1;
ssize_t minAbove = -1;
uint32_t candidate;
for (size_t i = 0; i < mSamplingRates.size(); i++) {
candidate = mSamplingRates[i];
if (candidate == samplingRate) {
if (updatedSamplingRate != NULL) {
*updatedSamplingRate = candidate;
}
return NO_ERROR;
}
// candidate < desired
if (candidate < samplingRate) {
if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
maxBelow = i;
}
// candidate > desired
} else {
if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
minAbove = i;
}
}
}
// This uses hard-coded knowledge about AudioFlinger resampling ratios.
// TODO Move these assumptions out.
static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
// due to approximation by an int32_t of the
// phase increments
// Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
if (minAbove >= 0) {
candidate = mSamplingRates[minAbove];
if (candidate / kMaxDownSampleRatio <= samplingRate) {
if (updatedSamplingRate != NULL) {
*updatedSamplingRate = candidate;
}
return NO_ERROR;
}
}
// But if we have to up-sample from a lower sampling rate, that's OK.
if (maxBelow >= 0) {
candidate = mSamplingRates[maxBelow];
if (candidate * kMaxUpSampleRatio >= samplingRate) {
if (updatedSamplingRate != NULL) {
*updatedSamplingRate = candidate;
}
return NO_ERROR;
}
}
// leave updatedSamplingRate unmodified
return BAD_VALUE;
}
status_t AudioPolicyManager::AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
{
for (size_t i = 0; i < mChannelMasks.size(); i++) {
if (mChannelMasks[i] == channelMask) {
return NO_ERROR;
}
}
return BAD_VALUE;
}
status_t AudioPolicyManager::AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
const
{
const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
for (size_t i = 0; i < mChannelMasks.size(); i ++) {
// FIXME Does not handle multi-channel automatic conversions yet
audio_channel_mask_t supported = mChannelMasks[i];
if (supported == channelMask) {
return NO_ERROR;
}
if (isRecordThread) {
// This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
// FIXME Abstract this out to a table.
if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
&& channelMask == AUDIO_CHANNEL_IN_MONO) ||
(supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
|| channelMask == AUDIO_CHANNEL_IN_STEREO))) {
return NO_ERROR;
}
}
}
return BAD_VALUE;
}
status_t AudioPolicyManager::AudioPort::checkFormat(audio_format_t format) const
{
for (size_t i = 0; i < mFormats.size(); i ++) {
if (mFormats[i] == format) {
return NO_ERROR;
}
}
return BAD_VALUE;
}
uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const
{
// special case for uninitialized dynamic profile
if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
return 0;
}
// For direct outputs, pick minimum sampling rate: this helps ensuring that the
// channel count / sampling rate combination chosen will be supported by the connected
// sink
if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
(mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
uint32_t samplingRate = UINT_MAX;
for (size_t i = 0; i < mSamplingRates.size(); i ++) {
if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
samplingRate = mSamplingRates[i];
}
}
return (samplingRate == UINT_MAX) ? 0 : samplingRate;
}
uint32_t samplingRate = 0;
uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
// For mixed output and inputs, use max mixer sampling rates. Do not
// limit sampling rate otherwise
if (mType != AUDIO_PORT_TYPE_MIX) {
maxRate = UINT_MAX;
}
for (size_t i = 0; i < mSamplingRates.size(); i ++) {
if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
samplingRate = mSamplingRates[i];
}
}
return samplingRate;
}
audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const
{
// special case for uninitialized dynamic profile
if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
return AUDIO_CHANNEL_NONE;
}
audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
// For direct outputs, pick minimum channel count: this helps ensuring that the
// channel count / sampling rate combination chosen will be supported by the connected
// sink
if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
(mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
uint32_t channelCount = UINT_MAX;
for (size_t i = 0; i < mChannelMasks.size(); i ++) {
uint32_t cnlCount;
if (mUseInChannelMask) {
cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
} else {
cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
}
if ((cnlCount < channelCount) && (cnlCount > 0)) {
channelMask = mChannelMasks[i];
channelCount = cnlCount;
}
}
return channelMask;
}
uint32_t channelCount = 0;
uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
// For mixed output and inputs, use max mixer channel count. Do not
// limit channel count otherwise
if (mType != AUDIO_PORT_TYPE_MIX) {
maxCount = UINT_MAX;
}
for (size_t i = 0; i < mChannelMasks.size(); i ++) {
uint32_t cnlCount;
if (mUseInChannelMask) {
cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
} else {
cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
}
if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
channelMask = mChannelMasks[i];
channelCount = cnlCount;
}
}
return channelMask;
}
/* format in order of increasing preference */
const audio_format_t AudioPolicyManager::AudioPort::sPcmFormatCompareTable[] = {
AUDIO_FORMAT_DEFAULT,
AUDIO_FORMAT_PCM_16_BIT,
AUDIO_FORMAT_PCM_8_24_BIT,
AUDIO_FORMAT_PCM_24_BIT_PACKED,
AUDIO_FORMAT_PCM_32_BIT,
AUDIO_FORMAT_PCM_FLOAT,
};
int AudioPolicyManager::AudioPort::compareFormats(audio_format_t format1,
audio_format_t format2)
{
// NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
// compressed format and better than any PCM format. This is by design of pickFormat()
if (!audio_is_linear_pcm(format1)) {
if (!audio_is_linear_pcm(format2)) {
return 0;
}
return 1;
}
if (!audio_is_linear_pcm(format2)) {
return -1;
}
int index1 = -1, index2 = -1;
for (size_t i = 0;
(i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
i ++) {
if (sPcmFormatCompareTable[i] == format1) {
index1 = i;
}
if (sPcmFormatCompareTable[i] == format2) {
index2 = i;
}
}
// format1 not found => index1 < 0 => format2 > format1
// format2 not found => index2 < 0 => format2 < format1
return index1 - index2;
}
audio_format_t AudioPolicyManager::AudioPort::pickFormat() const
{
// special case for uninitialized dynamic profile
if (mFormats.size() == 1 && mFormats[0] == 0) {
return AUDIO_FORMAT_DEFAULT;
}
audio_format_t format = AUDIO_FORMAT_DEFAULT;
audio_format_t bestFormat =
AudioPolicyManager::AudioPort::sPcmFormatCompareTable[
ARRAY_SIZE(AudioPolicyManager::AudioPort::sPcmFormatCompareTable) - 1];
// For mixed output and inputs, use best mixer output format. Do not
// limit format otherwise
if ((mType != AUDIO_PORT_TYPE_MIX) ||
((mRole == AUDIO_PORT_ROLE_SOURCE) &&
(((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
bestFormat = AUDIO_FORMAT_INVALID;
}
for (size_t i = 0; i < mFormats.size(); i ++) {
if ((compareFormats(mFormats[i], format) > 0) &&
(compareFormats(mFormats[i], bestFormat) <= 0)) {
format = mFormats[i];
}
}
return format;
}
status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig,
int index) const
{
if (index < 0 || (size_t)index >= mGains.size()) {
return BAD_VALUE;
}
return mGains[index]->checkConfig(gainConfig);
}
void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
if (mName.size() != 0) {
snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
result.append(buffer);
}
if (mSamplingRates.size() != 0) {
snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
result.append(buffer);
for (size_t i = 0; i < mSamplingRates.size(); i++) {
if (i == 0 && mSamplingRates[i] == 0) {
snprintf(buffer, SIZE, "Dynamic");
} else {
snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
}
result.append(buffer);
result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
}
result.append("\n");
}
if (mChannelMasks.size() != 0) {
snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
result.append(buffer);
for (size_t i = 0; i < mChannelMasks.size(); i++) {
ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
if (i == 0 && mChannelMasks[i] == 0) {
snprintf(buffer, SIZE, "Dynamic");
} else {
snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
}
result.append(buffer);
result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
}
result.append("\n");
}
if (mFormats.size() != 0) {
snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
result.append(buffer);
for (size_t i = 0; i < mFormats.size(); i++) {
const char *formatStr = enumToString(sFormatNameToEnumTable,
ARRAY_SIZE(sFormatNameToEnumTable),
mFormats[i]);
if (i == 0 && strcmp(formatStr, "") == 0) {
snprintf(buffer, SIZE, "Dynamic");
} else {
snprintf(buffer, SIZE, "%s", formatStr);
}
result.append(buffer);
result.append(i == (mFormats.size() - 1) ? "" : ", ");
}
result.append("\n");
}
write(fd, result.string(), result.size());
if (mGains.size() != 0) {
snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
write(fd, buffer, strlen(buffer) + 1);
result.append(buffer);
for (size_t i = 0; i < mGains.size(); i++) {
mGains[i]->dump(fd, spaces + 2, i);
}
}
}
// --- AudioGain class implementation
AudioPolicyManager::AudioGain::AudioGain(int index, bool useInChannelMask)
{
mIndex = index;
mUseInChannelMask = useInChannelMask;
memset(&mGain, 0, sizeof(struct audio_gain));
}
void AudioPolicyManager::AudioGain::getDefaultConfig(struct audio_gain_config *config)
{
config->index = mIndex;
config->mode = mGain.mode;
config->channel_mask = mGain.channel_mask;
if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
config->values[0] = mGain.default_value;
} else {
uint32_t numValues;
if (mUseInChannelMask) {
numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
} else {
numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
}
for (size_t i = 0; i < numValues; i++) {
config->values[i] = mGain.default_value;
}
}
if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
config->ramp_duration_ms = mGain.min_ramp_ms;
}
}
status_t AudioPolicyManager::AudioGain::checkConfig(const struct audio_gain_config *config)
{
if ((config->mode & ~mGain.mode) != 0) {
return BAD_VALUE;
}
if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
if ((config->values[0] < mGain.min_value) ||
(config->values[0] > mGain.max_value)) {
return BAD_VALUE;
}
} else {
if ((config->channel_mask & ~mGain.channel_mask) != 0) {
return BAD_VALUE;
}
uint32_t numValues;
if (mUseInChannelMask) {
numValues = audio_channel_count_from_in_mask(config->channel_mask);
} else {
numValues = audio_channel_count_from_out_mask(config->channel_mask);
}
for (size_t i = 0; i < numValues; i++) {
if ((config->values[i] < mGain.min_value) ||
(config->values[i] > mGain.max_value)) {
return BAD_VALUE;
}
}
}
if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
(config->ramp_duration_ms > mGain.max_ramp_ms)) {
return BAD_VALUE;
}
}
return NO_ERROR;
}
void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
result.append(buffer);
write(fd, result.string(), result.size());
}
// --- AudioPortConfig class implementation
AudioPolicyManager::AudioPortConfig::AudioPortConfig()
{
mSamplingRate = 0;
mChannelMask = AUDIO_CHANNEL_NONE;
mFormat = AUDIO_FORMAT_INVALID;
mGain.index = -1;
}
status_t AudioPolicyManager::AudioPortConfig::applyAudioPortConfig(
const struct audio_port_config *config,
struct audio_port_config *backupConfig)
{
struct audio_port_config localBackupConfig;
status_t status = NO_ERROR;
localBackupConfig.config_mask = config->config_mask;
toAudioPortConfig(&localBackupConfig);
sp<AudioPort> audioport = getAudioPort();
if (audioport == 0) {
status = NO_INIT;
goto exit;
}
if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
status = audioport->checkExactSamplingRate(config->sample_rate);
if (status != NO_ERROR) {
goto exit;
}
mSamplingRate = config->sample_rate;
}
if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
status = audioport->checkExactChannelMask(config->channel_mask);
if (status != NO_ERROR) {
goto exit;
}
mChannelMask = config->channel_mask;
}
if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
status = audioport->checkFormat(config->format);
if (status != NO_ERROR) {
goto exit;
}
mFormat = config->format;
}
if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
status = audioport->checkGain(&config->gain, config->gain.index);
if (status != NO_ERROR) {
goto exit;
}
mGain = config->gain;
}
exit:
if (status != NO_ERROR) {
applyAudioPortConfig(&localBackupConfig);
}
if (backupConfig != NULL) {
*backupConfig = localBackupConfig;
}
return status;
}
void AudioPolicyManager::AudioPortConfig::toAudioPortConfig(
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
dstConfig->sample_rate = mSamplingRate;
if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
dstConfig->sample_rate = srcConfig->sample_rate;
}
} else {
dstConfig->sample_rate = 0;
}
if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
dstConfig->channel_mask = mChannelMask;
if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
dstConfig->channel_mask = srcConfig->channel_mask;
}
} else {
dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
}
if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
dstConfig->format = mFormat;
if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
dstConfig->format = srcConfig->format;
}
} else {
dstConfig->format = AUDIO_FORMAT_INVALID;
}
if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
dstConfig->gain = mGain;
if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
dstConfig->gain = srcConfig->gain;
}
} else {
dstConfig->gain.index = -1;
}
if (dstConfig->gain.index != -1) {
dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
} else {
dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
}
}
// --- IOProfile class implementation
AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role,
const sp<HwModule>& module)
: AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
{
}
AudioPolicyManager::IOProfile::~IOProfile()
{
}
// checks if the IO profile is compatible with specified parameters.
// Sampling rate, format and channel mask must be specified in order to
// get a valid a match
bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
uint32_t samplingRate,
uint32_t *updatedSamplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
uint32_t flags) const
{
const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
ALOG_ASSERT(isPlaybackThread != isRecordThread);
if ((mSupportedDevices.types() & device) != device) {
return false;
}
if (samplingRate == 0) {
return false;
}
uint32_t myUpdatedSamplingRate = samplingRate;
if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
return false;
}
if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
NO_ERROR) {
return false;
}
if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
return false;
}
if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
checkExactChannelMask(channelMask) != NO_ERROR)) {
return false;
}
if (isRecordThread && (!audio_is_input_channel(channelMask) ||
checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
return false;
}
if (isPlaybackThread && (mFlags & flags) != flags) {
return false;
}
// The only input flag that is allowed to be different is the fast flag.
// An existing fast stream is compatible with a normal track request.
// An existing normal stream is compatible with a fast track request,
// but the fast request will be denied by AudioFlinger and converted to normal track.
if (isRecordThread && ((mFlags ^ flags) &
~AUDIO_INPUT_FLAG_FAST)) {
return false;
}
if (updatedSamplingRate != NULL) {
*updatedSamplingRate = myUpdatedSamplingRate;
}
return true;
}
void AudioPolicyManager::IOProfile::dump(int fd)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
AudioPort::dump(fd, 4);
snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
result.append(buffer);
snprintf(buffer, SIZE, " - devices:\n");
result.append(buffer);
write(fd, result.string(), result.size());
for (size_t i = 0; i < mSupportedDevices.size(); i++) {
mSupportedDevices[i]->dump(fd, 6, i);
}
}
void AudioPolicyManager::IOProfile::log()
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
ALOGV(" - sampling rates: ");
for (size_t i = 0; i < mSamplingRates.size(); i++) {
ALOGV(" %d", mSamplingRates[i]);
}
ALOGV(" - channel masks: ");
for (size_t i = 0; i < mChannelMasks.size(); i++) {
ALOGV(" 0x%04x", mChannelMasks[i]);
}
ALOGV(" - formats: ");
for (size_t i = 0; i < mFormats.size(); i++) {
ALOGV(" 0x%08x", mFormats[i]);
}
ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
ALOGV(" - flags: 0x%04x\n", mFlags);
}
// --- DeviceDescriptor implementation
AudioPolicyManager::DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
AUDIO_PORT_ROLE_SOURCE,
NULL),
mDeviceType(type), mAddress(""), mId(0)
{
if (mGains.size() > 0) {
mGains[0]->getDefaultConfig(&mGain);
}
}
bool AudioPolicyManager::DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
{
// Devices are considered equal if they:
// - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
// - have the same address or one device does not specify the address
// - have the same channel mask or one device does not specify the channel mask
return (mDeviceType == other->mDeviceType) &&
(mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
(mChannelMask == 0 || other->mChannelMask == 0 ||
mChannelMask == other->mChannelMask);
}
void AudioPolicyManager::DeviceVector::refreshTypes()
{
mDeviceTypes = AUDIO_DEVICE_NONE;
for(size_t i = 0; i < size(); i++) {
mDeviceTypes |= itemAt(i)->mDeviceType;
}
ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
}
ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
{
for(size_t i = 0; i < size(); i++) {
if (item->equals(itemAt(i))) {
return i;
}
}
return -1;
}
ssize_t AudioPolicyManager::DeviceVector::add(const sp<DeviceDescriptor>& item)
{
ssize_t ret = indexOf(item);
if (ret < 0) {
ret = SortedVector::add(item);
if (ret >= 0) {
refreshTypes();
}
} else {
ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
ret = -1;
}
return ret;
}
ssize_t AudioPolicyManager::DeviceVector::remove(const sp<DeviceDescriptor>& item)
{
size_t i;
ssize_t ret = indexOf(item);
if (ret < 0) {
ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
} else {
ret = SortedVector::removeAt(ret);
if (ret >= 0) {
refreshTypes();
}
}
return ret;
}
void AudioPolicyManager::DeviceVector::loadDevicesFromType(audio_devices_t types)
{
DeviceVector deviceList;
uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
types &= ~role_bit;
while (types) {
uint32_t i = 31 - __builtin_clz(types);
uint32_t type = 1 << i;
types &= ~type;
add(new DeviceDescriptor(String8(""), type | role_bit));
}
}
void AudioPolicyManager::DeviceVector::loadDevicesFromName(char *name,
const DeviceVector& declaredDevices)
{
char *devName = strtok(name, "|");
while (devName != NULL) {
if (strlen(devName) != 0) {
audio_devices_t type = stringToEnum(sDeviceNameToEnumTable,
ARRAY_SIZE(sDeviceNameToEnumTable),
devName);
if (type != AUDIO_DEVICE_NONE) {
add(new DeviceDescriptor(String8(""), type));
} else {
sp<DeviceDescriptor> deviceDesc =
declaredDevices.getDeviceFromName(String8(devName));
if (deviceDesc != 0) {
add(deviceDesc);
}
}
}
devName = strtok(NULL, "|");
}
}
sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDevice(
audio_devices_t type, String8 address) const
{
sp<DeviceDescriptor> device;
for (size_t i = 0; i < size(); i++) {
if (itemAt(i)->mDeviceType == type) {
device = itemAt(i);
if (itemAt(i)->mAddress = address) {
break;
}
}
}
ALOGV("DeviceVector::getDevice() for type %d address %s found %p",
type, address.string(), device.get());
return device;
}
sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromId(
audio_port_handle_t id) const
{
sp<DeviceDescriptor> device;
for (size_t i = 0; i < size(); i++) {
ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%zu)->mId %d", id, i, itemAt(i)->mId);
if (itemAt(i)->mId == id) {
device = itemAt(i);
break;
}
}
return device;
}
AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType(
audio_devices_t type) const
{
DeviceVector devices;
for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
devices.add(itemAt(i));
type &= ~itemAt(i)->mDeviceType;
ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
itemAt(i)->mDeviceType, itemAt(i).get());
}
}
return devices;
}
AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromTypeAddr(
audio_devices_t type, String8 address) const
{
DeviceVector devices;
//ALOGV(" looking for device=%x, addr=%s", type, address.string());
for (size_t i = 0; i < size(); i++) {
//ALOGV(" at i=%d: device=%x, addr=%s",
// i, itemAt(i)->mDeviceType, itemAt(i)->mAddress.string());
if (itemAt(i)->mDeviceType == type) {
if (itemAt(i)->mAddress == address) {
//ALOGV(" found matching address %s", address.string());
devices.add(itemAt(i));
}
}
}
return devices;
}
sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromName(
const String8& name) const
{
sp<DeviceDescriptor> device;
for (size_t i = 0; i < size(); i++) {
if (itemAt(i)->mName == name) {
device = itemAt(i);
break;
}
}
return device;
}
void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig(
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
if (srcConfig != NULL) {
dstConfig->config_mask |= srcConfig->config_mask;
}
AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
dstConfig->id = mId;
dstConfig->role = audio_is_output_device(mDeviceType) ?
AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
dstConfig->ext.device.type = mDeviceType;
dstConfig->ext.device.hw_module = mModule->mHandle;
strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
}
void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const
{
ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
AudioPort::toAudioPort(port);
port->id = mId;
toAudioPortConfig(&port->active_config);
port->ext.device.type = mDeviceType;
port->ext.device.hw_module = mModule->mHandle;
strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
}
status_t AudioPolicyManager::DeviceDescriptor::dump(int fd, int spaces, int index) const
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
result.append(buffer);
if (mId != 0) {
snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
result.append(buffer);
}
snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
enumToString(sDeviceNameToEnumTable,
ARRAY_SIZE(sDeviceNameToEnumTable),
mDeviceType));
result.append(buffer);
if (mAddress.size() != 0) {
snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
result.append(buffer);
}
write(fd, result.string(), result.size());
AudioPort::dump(fd, spaces);
return NO_ERROR;
}
status_t AudioPolicyManager::AudioPatch::dump(int fd, int spaces, int index) const
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
result.append(buffer);
snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
result.append(buffer);
for (size_t i = 0; i < mPatch.num_sources; i++) {
if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
mPatch.sources[i].id, enumToString(sDeviceNameToEnumTable,
ARRAY_SIZE(sDeviceNameToEnumTable),
mPatch.sources[i].ext.device.type));
} else {
snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
}
result.append(buffer);
}
snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
result.append(buffer);
for (size_t i = 0; i < mPatch.num_sinks; i++) {
if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
mPatch.sinks[i].id, enumToString(sDeviceNameToEnumTable,
ARRAY_SIZE(sDeviceNameToEnumTable),
mPatch.sinks[i].ext.device.type));
} else {
snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
}
result.append(buffer);
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
// --- audio_policy.conf file parsing
uint32_t AudioPolicyManager::parseOutputFlagNames(char *name)
{
uint32_t flag = 0;
// it is OK to cast name to non const here as we are not going to use it after
// strtok() modifies it
char *flagName = strtok(name, "|");
while (flagName != NULL) {
if (strlen(flagName) != 0) {
flag |= stringToEnum(sOutputFlagNameToEnumTable,
ARRAY_SIZE(sOutputFlagNameToEnumTable),
flagName);
}
flagName = strtok(NULL, "|");
}
//force direct flag if offload flag is set: offloading implies a direct output stream
// and all common behaviors are driven by checking only the direct flag
// this should normally be set appropriately in the policy configuration file
if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
flag |= AUDIO_OUTPUT_FLAG_DIRECT;
}
return flag;
}
uint32_t AudioPolicyManager::parseInputFlagNames(char *name)
{
uint32_t flag = 0;
// it is OK to cast name to non const here as we are not going to use it after
// strtok() modifies it
char *flagName = strtok(name, "|");
while (flagName != NULL) {
if (strlen(flagName) != 0) {
flag |= stringToEnum(sInputFlagNameToEnumTable,
ARRAY_SIZE(sInputFlagNameToEnumTable),
flagName);
}
flagName = strtok(NULL, "|");
}
return flag;
}
audio_devices_t AudioPolicyManager::parseDeviceNames(char *name)
{
uint32_t device = 0;
char *devName = strtok(name, "|");
while (devName != NULL) {
if (strlen(devName) != 0) {
device |= stringToEnum(sDeviceNameToEnumTable,
ARRAY_SIZE(sDeviceNameToEnumTable),
devName);
}
devName = strtok(NULL, "|");
}
return device;
}
void AudioPolicyManager::loadHwModule(cnode *root)
{
status_t status = NAME_NOT_FOUND;
cnode *node;
sp<HwModule> module = new HwModule(root->name);
node = config_find(root, DEVICES_TAG);
if (node != NULL) {
node = node->first_child;
while (node) {
ALOGV("loadHwModule() loading device %s", node->name);
status_t tmpStatus = module->loadDevice(node);
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
status = tmpStatus;
}
node = node->next;
}
}
node = config_find(root, OUTPUTS_TAG);
if (node != NULL) {
node = node->first_child;
while (node) {
ALOGV("loadHwModule() loading output %s", node->name);
status_t tmpStatus = module->loadOutput(node);
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
status = tmpStatus;
}
node = node->next;
}
}
node = config_find(root, INPUTS_TAG);
if (node != NULL) {
node = node->first_child;
while (node) {
ALOGV("loadHwModule() loading input %s", node->name);
status_t tmpStatus = module->loadInput(node);
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
status = tmpStatus;
}
node = node->next;
}
}
loadGlobalConfig(root, module);
if (status == NO_ERROR) {
mHwModules.add(module);
}
}
void AudioPolicyManager::loadHwModules(cnode *root)
{
cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
if (node == NULL) {
return;
}
node = node->first_child;
while (node) {
ALOGV("loadHwModules() loading module %s", node->name);
loadHwModule(node);
node = node->next;
}
}
void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module)
{
cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
if (node == NULL) {
return;
}
DeviceVector declaredDevices;
if (module != NULL) {
declaredDevices = module->mDeclaredDevices;
}
node = node->first_child;
while (node) {
if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
mAvailableOutputDevices.loadDevicesFromName((char *)node->value,
declaredDevices);
ALOGV("loadGlobalConfig() Attached Output Devices %08x",
mAvailableOutputDevices.types());
} else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
audio_devices_t device = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
ARRAY_SIZE(sDeviceNameToEnumTable),
(char *)node->value);
if (device != AUDIO_DEVICE_NONE) {
mDefaultOutputDevice = new DeviceDescriptor(String8(""), device);
} else {
ALOGW("loadGlobalConfig() default device not specified");
}
ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType);
} else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
mAvailableInputDevices.loadDevicesFromName((char *)node->value,
declaredDevices);
ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
} else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
mSpeakerDrcEnabled = stringToBool((char *)node->value);
ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
} else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
uint32_t major, minor;
sscanf((char *)node->value, "%u.%u", &major, &minor);
module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor);
ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u",
module->mHalVersion, major, minor);
}
node = node->next;
}
}
status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
{
cnode *root;
char *data;
data = (char *)load_file(path, NULL);
if (data == NULL) {
return -ENODEV;
}
root = config_node("", "");
config_load(root, data);
loadHwModules(root);
// legacy audio_policy.conf files have one global_configuration section
loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY));
config_free(root);
free(root);
free(data);
ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
return NO_ERROR;
}
void AudioPolicyManager::defaultAudioPolicyConfig(void)
{
sp<HwModule> module;
sp<IOProfile> profile;
sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(String8(""),
AUDIO_DEVICE_IN_BUILTIN_MIC);
mAvailableOutputDevices.add(mDefaultOutputDevice);
mAvailableInputDevices.add(defaultInputDevice);
module = new HwModule("primary");
profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module);
profile->mSamplingRates.add(44100);
profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
profile->mSupportedDevices.add(mDefaultOutputDevice);
profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
module->mOutputProfiles.add(profile);
profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module);
profile->mSamplingRates.add(8000);
profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
profile->mSupportedDevices.add(defaultInputDevice);
module->mInputProfiles.add(profile);
mHwModules.add(module);
}
audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
{
// flags to stream type mapping
if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
return AUDIO_STREAM_ENFORCED_AUDIBLE;
}
if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
return AUDIO_STREAM_BLUETOOTH_SCO;
}
// usage to stream type mapping
switch (attr->usage) {
case AUDIO_USAGE_MEDIA:
case AUDIO_USAGE_GAME:
case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
return AUDIO_STREAM_MUSIC;
case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
return AUDIO_STREAM_SYSTEM;
case AUDIO_USAGE_VOICE_COMMUNICATION:
return AUDIO_STREAM_VOICE_CALL;
case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
return AUDIO_STREAM_DTMF;
case AUDIO_USAGE_ALARM:
return AUDIO_STREAM_ALARM;
case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
return AUDIO_STREAM_RING;
case AUDIO_USAGE_NOTIFICATION:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
case AUDIO_USAGE_NOTIFICATION_EVENT:
return AUDIO_STREAM_NOTIFICATION;
case AUDIO_USAGE_UNKNOWN:
default:
return AUDIO_STREAM_MUSIC;
}
}
}; // namespace android