/*
* Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2009 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*
* This file was modified by Dolby Laboratories, Inc. The portions of the
* code that are surrounded by "DOLBY..." are copyrighted and
* licensed separately, as follows:
*
* (C) 2015 Dolby Laboratories, Inc.
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#define LOG_TAG "AudioPolicyManagerCustom"
//#define LOG_NDEBUG 0
//#define VERY_VERBOSE_LOGGING
#ifdef VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
#define MIN(a, b) ((a) < (b) ? (a) : (b))
// 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)
// Following delay should be used if the calculated routing delay from all active
// input streams is higher than this value
#define MAX_VOICE_CALL_START_DELAY_MS 100
#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 <policy.h>
#ifdef DOLBY_ENABLE
#include "DolbyAudioPolicy_impl.h"
#endif // DOLBY_END
namespace android {
#ifdef VOICE_CONCURRENCY
audio_output_flags_t AudioPolicyManagerCustom::getFallBackPath()
{
audio_output_flags_t flag = AUDIO_OUTPUT_FLAG_FAST;
char propValue[PROPERTY_VALUE_MAX];
if (property_get("voice.conc.fallbackpath", propValue, NULL)) {
if (!strncmp(propValue, "deep-buffer", 11)) {
flag = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
else if (!strncmp(propValue, "fast", 4)) {
flag = AUDIO_OUTPUT_FLAG_FAST;
}
else {
ALOGD("voice_conc:not a recognised path(%s) in prop voice.conc.fallbackpath",
propValue);
}
}
else {
ALOGD("voice_conc:prop voice.conc.fallbackpath not set");
}
ALOGD("voice_conc:picked up flag(0x%x) from prop voice.conc.fallbackpath",
flag);
return flag;
}
#endif /*VOICE_CONCURRENCY*/
// ----------------------------------------------------------------------------
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
extern "C" AudioPolicyInterface* createAudioPolicyManager(
AudioPolicyClientInterface *clientInterface)
{
return new AudioPolicyManagerCustom(clientInterface);
}
extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
{
delete interface;
}
status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
const char *device_name)
{
ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
device, state, device_address, device_name);
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
sp<DeviceDescriptor> devDesc =
mHwModules.getDeviceDescriptor(device, device_address, device_name);
// handle output devices
if (audio_is_output_device(device)) {
SortedVector <audio_io_handle_t> outputs;
ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
// save a copy of the opened output descriptors before any output is opened or closed
// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
mPreviousOutputs = mOutputs;
switch (state)
{
// handle output device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
if (!strncmp(device_address, "hdmi_spkr", 9)) {
mHdmiAudioDisabled = false;
} else {
mHdmiAudioEvent = true;
}
}
#endif
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);
#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
if (!strncmp(device_address, "hdmi_spkr", 9)) {
mHdmiAudioDisabled = false;
} else {
mHdmiAudioEvent = true;
}
if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
mAvailableOutputDevices.remove(devDesc);
ALOGW("HDMI sink not connected, do not route audio to HDMI out");
return INVALID_OPERATION;
}
}
#endif
if (index >= 0) {
sp<HwModule> module = mHwModules.getModuleForDevice(device);
if (module == 0) {
ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
device);
mAvailableOutputDevices.remove(devDesc);
return INVALID_OPERATION;
}
mAvailableOutputDevices[index]->attach(module);
} else {
return NO_MEMORY;
}
if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
mAvailableOutputDevices.remove(devDesc);
return INVALID_OPERATION;
}
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
// outputs should never be empty here
ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
"checkOutputsForDevice() returned no outputs but status OK");
ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
outputs.size());
// Send connect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
} break;
// handle output device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
if (!strncmp(device_address, "hdmi_spkr", 9)) {
mHdmiAudioDisabled = true;
} else {
mHdmiAudioEvent = false;
}
}
#endif
ALOGW("setDeviceConnectionState() device not connected: %x", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
// Send Disconnect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
// remove device from available output devices
mAvailableOutputDevices.remove(devDesc);
#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
if (!strncmp(device_address, "hdmi_spkr", 9)) {
mHdmiAudioDisabled = true;
} else {
mHdmiAudioEvent = false;
}
}
#endif
checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
} break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
// checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
// output is suspended before any tracks are moved to it
checkA2dpSuspend();
checkOutputForAllStrategies();
// outputs must be closed after checkOutputForAllStrategies() is executed
if (!outputs.isEmpty()) {
for (size_t i = 0; i < outputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
// close unused outputs after device disconnection or direct outputs that have been
// opened by checkOutputsForDevice() to query dynamic parameters
if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
(desc->mDirectOpenCount == 0))) {
closeOutput(outputs[i]);
}
}
// check again after closing A2DP output to reset mA2dpSuspended if needed
checkA2dpSuspend();
}
updateDevicesAndOutputs();
#ifdef DOLBY_ENABLE
// Before closing the opened outputs, update endpoint property with device capabilities
audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AUDIO_STREAM_MUSIC), true);
mDolbyAudioPolicy.setEndpointSystemProperty(audioOutputDevice, mHwModules);
#endif // DOLBY_END
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevice);
}
#ifdef FM_POWER_OPT
// handle FM device connection state to trigger FM AFE loopback
if(device == AUDIO_DEVICE_OUT_FM && hasPrimaryOutput()) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, 1);
newDevice = newDevice | AUDIO_DEVICE_OUT_FM;
} else {
mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, -1);
}
AudioParameter param = AudioParameter();
param.addInt(String8("handle_fm"), (int)newDevice);
mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString());
}
#endif /* FM_POWER_OPT end */
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
// do not force device change on duplicated output because if device is 0, it will
// also force a device 0 for the two outputs it is duplicated to which may override
// a valid device selection on those outputs.
bool force = !desc->isDuplicated()
&& (!device_distinguishes_on_address(device)
// always force when disconnecting (a non-duplicated device)
|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
setOutputDevice(desc, newDevice, force, 0);
}
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is output device
// handle input devices
if (audio_is_input_device(device)) {
SortedVector <audio_io_handle_t> inputs;
ssize_t index = mAvailableInputDevices.indexOf(devDesc);
switch (state)
{
// handle input device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
ALOGW("setDeviceConnectionState() device already connected: %d", device);
return INVALID_OPERATION;
}
sp<HwModule> module = mHwModules.getModuleForDevice(device);
if (module == NULL) {
ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
device);
return INVALID_OPERATION;
}
if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) {
return INVALID_OPERATION;
}
index = mAvailableInputDevices.add(devDesc);
if (index >= 0) {
mAvailableInputDevices[index]->attach(module);
} else {
return NO_MEMORY;
}
// Set connect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
} break;
// handle input device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
ALOGW("setDeviceConnectionState() device not connected: %d", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
// Set Disconnect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress);
mAvailableInputDevices.remove(devDesc);
// Propagate device availability to Engine
mEngine->setDeviceConnectionState(devDesc, state);
} break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
closeAllInputs();
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
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;
}
// 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 AudioPolicyManagerCustom::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);
#ifdef VOICE_CONCURRENCY
char concpropValue[PROPERTY_VALUE_MAX];
if (property_get("voice.playback.conc.disabled", concpropValue, NULL)) {
bool propenabled = atoi(concpropValue) || !strncmp("true", concpropValue, 4);
if (propenabled) {
if (isInCall())
{
ALOGD("\n copl: blocking compress offload on call mode\n");
return false;
}
}
}
#endif
#ifdef RECORD_PLAY_CONCURRENCY
char recConcPropValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;
if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
}
if ((prop_rec_play_enabled) &&
((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCount() > 0))) {
ALOGD("copl: blocking compress offload for record concurrency");
return false;
}
#endif
// 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;
}
//check if it's multi-channel AAC (includes sub formats) and FLAC and VORBIS format
if ((popcount(offloadInfo.channel_mask) > 2) &&
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
return false;
}
#ifdef AUDIO_EXTN_FORMATS_ENABLED
//check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
if ((popcount(offloadInfo.channel_mask) > 2) &&
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && offloadInfo.sample_rate > 48000) ||
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && offloadInfo.sample_rate > 48000) ||
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && offloadInfo.sample_rate > 48000))) {
ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA clips with sample rate > 48kHz");
return false;
}
#endif
//TODO: enable audio offloading with video when ready
const bool allowOffloadWithVideo =
property_get_bool("audio.offload.video", false /* default_value */);
if (offloadInfo.has_video && !allowOffloadWithVideo) {
ALOGV("isOffloadSupported: has_video == true, returning false");
return false;
}
//If duration is less than minimum value defined in property, return false
if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
return false;
}
} else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
//duration checks only valid for MP3/AAC/VORBIS/WMA/ALAC/APE formats,
//do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
if ((offloadInfo.format == AUDIO_FORMAT_MP3) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS)
#ifdef AUDIO_EXTN_FORMATS_ENABLED
|| ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE)
#endif
)
return false;
}
// Do not allow offloading if one non offloadable effect is enabled. This prevents from
// creating an offloaded track and tearing it down immediately after start when audioflinger
// detects there is an active non offloadable effect.
// FIXME: We should check the audio session here but we do not have it in this context.
// This may prevent offloading in rare situations where effects are left active by apps
// in the background.
if (mEffects.isNonOffloadableEffectEnabled()) {
return false;
}
// Check for soundcard status
String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
String8("SND_CARD_STATUS"));
AudioParameter result = AudioParameter(valueStr);
int isonline = 0;
if ((result.getInt(String8("SND_CARD_STATUS"), isonline) == NO_ERROR)
&& !isonline) {
ALOGD("copl: soundcard is offline rejecting offload request");
return false;
}
// See if there is a profile to support this.
// AUDIO_DEVICE_NONE
sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
offloadInfo.sample_rate,
offloadInfo.format,
offloadInfo.channel_mask,
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
return (profile != 0);
}
audio_devices_t AudioPolicyManagerCustom::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
bool fromCache)
{
audio_devices_t device = AUDIO_DEVICE_NONE;
ssize_t index = mAudioPatches.indexOfKey(outputDesc->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 accessibility is active on the output:
// use device for strategy accessibility
// 7: the strategy media is active on the output:
// use device for strategy media
// 8: the strategy DTMF is active on the output:
// use device for strategy DTMF
// 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output:
// use device for strategy t-t-s
if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) &&
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
} else if (isInCall() ||
isStrategyActive(outputDesc, STRATEGY_PHONE)||
isStrategyActive(mPrimaryOutput, STRATEGY_PHONE)) {
device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) {
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)||
(isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION)
&& (!isStrategyActive(mPrimaryOutput,STRATEGY_MEDIA)))) {
device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)||
(isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION_RESPECTFUL)
&& (!isStrategyActive(mPrimaryOutput, STRATEGY_MEDIA)))) {
device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) {
device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) {
device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) {
device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) {
device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
}
ALOGV("getNewOutputDevice() selected device %x", device);
return device;
}
void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)
{
ALOGV("setPhoneState() state %d", state);
// store previous phone state for management of sonification strategy below
audio_devices_t newDevice = AUDIO_DEVICE_NONE;
int oldState = mEngine->getPhoneState();
if (mEngine->setPhoneState(state) != NO_ERROR) {
ALOGW("setPhoneState() invalid or same state %d", state);
return;
}
/// Opens: can these line be executed after the switch of volume curves???
// if leaving call state, handle special case of active streams
// pertaining to sonification strategy see handleIncallSonification()
if (isStateInCall(oldState)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
for (size_t j = 0; j < mOutputs.size(); j++) {
audio_io_handle_t curOutput = mOutputs.keyAt(j);
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
if (stream == AUDIO_STREAM_PATCH) {
continue;
}
handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput);
}
}
// force reevaluating accessibility routing when call starts
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
/**
* Switching to or from incall state or switching between telephony and VoIP lead to force
* routing command.
*/
bool force = ((is_state_in_call(oldState) != is_state_in_call(state))
|| (is_state_in_call(state) && (state != oldState)));
// check for device and output changes triggered by new phone state
checkA2dpSuspend();
checkOutputForAllStrategies();
updateDevicesAndOutputs();
sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
#ifdef VOICE_CONCURRENCY
int voice_call_state = 0;
char propValue[PROPERTY_VALUE_MAX];
bool prop_playback_enabled = false, prop_rec_enabled=false, prop_voip_enabled = false;
if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
prop_playback_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
}
if(property_get("voice.record.conc.disabled", propValue, NULL)) {
prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
}
if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
}
bool mode_in_call = (AUDIO_MODE_IN_CALL != oldState) && (AUDIO_MODE_IN_CALL == state);
//query if it is a actual voice call initiated by telephony
if (mode_in_call) {
String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, String8("in_call"));
AudioParameter result = AudioParameter(valueStr);
if (result.getInt(String8("in_call"), voice_call_state) == NO_ERROR)
ALOGD("voice_conc:SetPhoneState: Voice call state = %d", voice_call_state);
}
if (mode_in_call && voice_call_state && !mvoice_call_state) {
ALOGD("voice_conc:Entering to call mode oldState :: %d state::%d ",
oldState, state);
mvoice_call_state = voice_call_state;
if (prop_rec_enabled) {
//Close all active inputs
audio_io_handle_t activeInput = mInputs.getActiveInput();
if (activeInput != 0) {
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
switch(activeDesc->mInputSource) {
case AUDIO_SOURCE_VOICE_UPLINK:
case AUDIO_SOURCE_VOICE_DOWNLINK:
case AUDIO_SOURCE_VOICE_CALL:
ALOGD("voice_conc:FOUND active input during call active: %d",activeDesc->mInputSource);
break;
case AUDIO_SOURCE_VOICE_COMMUNICATION:
if(prop_voip_enabled) {
ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",activeDesc->mInputSource);
stopInput(activeInput, activeDesc->mSessions.itemAt(0));
releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
}
break;
default:
ALOGD("voice_conc:CLOSING input on call setup for inputSource: %d",activeDesc->mInputSource);
stopInput(activeInput, activeDesc->mSessions.itemAt(0));
releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
break;
}
}
} else if (prop_voip_enabled) {
audio_io_handle_t activeInput = mInputs.getActiveInput();
if (activeInput != 0) {
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeDesc->mInputSource) {
ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeDesc->mInputSource);
stopInput(activeInput, activeDesc->mSessions.itemAt(0));
releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
}
}
}
if (prop_playback_enabled) {
// Move tracks associated to this strategy from previous output to new output
for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
if (i == AUDIO_STREAM_PATCH) {
ALOGV("voice_conc:not calling invalidate for AUDIO_STREAM_PATCH");
continue;
}
if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
if ((AUDIO_STREAM_MUSIC == i) ||
(AUDIO_STREAM_VOICE_CALL == i) ) {
ALOGD("voice_conc:Invalidate stream type %d", i);
mpClientInterface->invalidateStream((audio_stream_type_t)i);
}
} else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
ALOGD("voice_conc:Invalidate stream type %d", i);
mpClientInterface->invalidateStream((audio_stream_type_t)i);
}
}
}
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
ALOGD("voice_conc:ouput desc / profile is NULL");
continue;
}
if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
if (((!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY))
&& prop_playback_enabled) {
ALOGD("voice_conc:calling suspendOutput on call mode for primary output");
mpClientInterface->suspendOutput(mOutputs.keyAt(i));
} //Close compress all sessions
else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
&& prop_playback_enabled) {
ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
closeOutput(mOutputs.keyAt(i));
}
else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX)
&& prop_voip_enabled) {
ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output");
closeOutput(mOutputs.keyAt(i));
}
} else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)
&& prop_playback_enabled) {
ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
closeOutput(mOutputs.keyAt(i));
}
}
}
}
if ((AUDIO_MODE_IN_CALL == oldState || AUDIO_MODE_IN_COMMUNICATION == oldState) &&
(AUDIO_MODE_NORMAL == state) && prop_playback_enabled && mvoice_call_state) {
ALOGD("voice_conc:EXITING from call mode oldState :: %d state::%d \n",oldState, state);
mvoice_call_state = 0;
if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
//restore PCM (deep-buffer) output after call termination
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
ALOGD("voice_conc:ouput desc / profile is NULL");
continue;
}
if (!outputDesc->isDuplicated() && outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
ALOGD("voice_conc:calling restoreOutput after call mode for primary output");
mpClientInterface->restoreOutput(mOutputs.keyAt(i));
}
}
}
//call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
if (i == AUDIO_STREAM_PATCH) {
ALOGV("voice_conc:not calling invalidate for AUDIO_STREAM_PATCH");
continue;
}
if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
if ((AUDIO_STREAM_MUSIC == i) ||
(AUDIO_STREAM_VOICE_CALL == i) ) {
mpClientInterface->invalidateStream((audio_stream_type_t)i);
}
} else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
mpClientInterface->invalidateStream((audio_stream_type_t)i);
}
}
}
#endif
#ifdef RECORD_PLAY_CONCURRENCY
char recConcPropValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;
if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
}
if (prop_rec_play_enabled) {
if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
ALOGD("phone state changed to MODE_IN_COMM invlaidating music and voice streams");
// call invalidate for voice streams, so that it can use deepbuffer with VoIP out device from HAL
mpClientInterface->invalidateStream(AUDIO_STREAM_VOICE_CALL);
// call invalidate for music, so that compress will fallback to deep-buffer with VoIP out device
mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
// close compress output to make sure session will be closed before timeout(60sec)
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
ALOGD("ouput desc / profile is NULL");
continue;
}
if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
ALOGD("calling closeOutput on call mode for COMPRESS output");
closeOutput(mOutputs.keyAt(i));
}
}
} else if ((oldState == AUDIO_MODE_IN_COMMUNICATION) &&
(mEngine->getPhoneState() == AUDIO_MODE_NORMAL)) {
// call invalidate for music so that music can fallback to compress
mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
}
}
#endif
mPrevPhoneState = oldState;
int delayMs = 0;
if (isStateInCall(state)) {
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
// mute media and sonification strategies and delay device switch by the largest
// latency of any output where either strategy is active.
// This avoid sending the ring tone or music tail into the earpiece or headset.
if ((isStrategyActive(desc, STRATEGY_MEDIA,
SONIFICATION_HEADSET_MUSIC_DELAY,
sysTime) ||
isStrategyActive(desc, STRATEGY_SONIFICATION,
SONIFICATION_HEADSET_MUSIC_DELAY,
sysTime)) &&
(delayMs < (int)desc->latency()*2)) {
delayMs = desc->latency()*2;
}
setStrategyMute(STRATEGY_MEDIA, true, desc);
setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS,
getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
setStrategyMute(STRATEGY_SONIFICATION, true, desc);
setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS,
getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
}
ALOGV("Setting the delay from %dms to %dms", delayMs,
MIN(delayMs, MAX_VOICE_CALL_START_DELAY_MS));
delayMs = MIN(delayMs, MAX_VOICE_CALL_START_DELAY_MS);
}
if (hasPrimaryOutput()) {
// Note that despite the fact that getNewOutputDevice() is called on the primary output,
// the device returned is not necessarily reachable via this output
audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
// force routing command to audio hardware when ending call
// even if no device change is needed
if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
rxDevice = mPrimaryOutput->device();
}
if (state == AUDIO_MODE_IN_CALL) {
updateCallRouting(rxDevice, delayMs);
} else if (oldState == AUDIO_MODE_IN_CALL) {
if (mCallRxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
mCallRxPatch.clear();
}
if (mCallTxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
mCallTxPatch.clear();
}
setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
} else {
setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
}
}
//update device for all non-primary outputs
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_io_handle_t output = mOutputs.keyAt(i);
if (output != mPrimaryOutput->mIoHandle) {
newDevice = getNewOutputDevice(mOutputs.valueFor(output), false /*fromCache*/);
setOutputDevice(mOutputs.valueFor(output), newDevice, (newDevice != AUDIO_DEVICE_NONE));
}
}
// if entering in call state, handle special case of active streams
// pertaining to sonification strategy see handleIncallSonification()
if (isStateInCall(state)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
for (size_t j = 0; j < mOutputs.size(); j++) {
audio_io_handle_t curOutput = mOutputs.keyAt(j);
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
if (stream == AUDIO_STREAM_PATCH) {
continue;
}
handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput);
}
}
}
// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
if (state == AUDIO_MODE_RINGTONE &&
isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
mLimitRingtoneVolume = true;
} else {
mLimitRingtoneVolume = false;
}
}
void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config)
{
ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
if (mEngine->setForceUse(usage, config) != NO_ERROR) {
ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
return;
}
bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
(usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
(usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
// check for device and output changes triggered by new force usage
checkA2dpSuspend();
checkOutputForAllStrategies();
updateDevicesAndOutputs();
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
updateCallRouting(newDevice);
}
// Use reverse loop to make sure any low latency usecases (generally tones)
// are not routed before non LL usecases (generally music).
// We can safely assume that LL output would always have lower index,
// and use this work-around to avoid routing of output with music stream
// from the context of short lived LL output.
// Note: in case output's share backend(HAL sharing is implicit) all outputs
// gets routing update while processing first output itself.
for (size_t i = mOutputs.size(); i > 0; i--) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || outputDesc != mPrimaryOutput) {
setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE));
}
if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
applyStreamVolumes(outputDesc, newDevice, 0, true);
}
}
audio_io_handle_t activeInput = mInputs.getActiveInput();
if (activeInput != 0) {
setInputDevice(activeInput, getNewInputDevice(activeInput));
}
}
status_t AudioPolicyManagerCustom::stopSource(sp<AudioOutputDescriptor> outputDesc1,
audio_stream_type_t stream,
bool forceDeviceUpdate)
{
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
ALOGW("stopSource() invalid stream %d", stream);
return INVALID_OPERATION;
}
// always handle stream stop, check which stream type is stopping
#ifdef NON_WEARABLE_TARGET
sp<AudioOutputDescriptor> outputDesc = outputDesc1;
#else
sp<SwAudioOutputDescriptor> outputDesc = (sp<SwAudioOutputDescriptor>) outputDesc1;
#endif
handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
// handle special case for sonification while in call
if (isInCall() && (outputDesc->mRefCount[stream] == 1)) {
if (outputDesc->isDuplicated()) {
#ifdef NON_WEARABLE_TARGET
handleIncallSonification(stream, false, false, outputDesc->subOutput1()->mIoHandle);
handleIncallSonification(stream, false, false, outputDesc->subOutput2()->mIoHandle);
#else
handleIncallSonification(stream, false, false, outputDesc->mOutput1->mIoHandle);
handleIncallSonification(stream, false, false, outputDesc->mOutput2->mIoHandle);
#endif
}
handleIncallSonification(stream, false, false, outputDesc->mIoHandle);
}
if (outputDesc->mRefCount[stream] > 0) {
// decrement usage count of this stream on the output
outputDesc->changeRefCount(stream, -1);
// store time at which the stream was stopped - see isStreamActive()
if (outputDesc->mRefCount[stream] == 0 || forceDeviceUpdate) {
outputDesc->mStopTime[stream] = systemTime();
audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
// delay the device switch by twice the latency because stopOutput() is executed when
// the track stop() command is received and at that time the audio track buffer can
// still contain data that needs to be drained. The latency only covers the audio HAL
// and kernel buffers. Also the latency does not always include additional delay in the
// audio path (audio DSP, CODEC ...)
setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2);
// force restoring the device selection on other active outputs if it differs from the
// one being selected for this output
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_io_handle_t curOutput = mOutputs.keyAt(i);
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc &&
desc->isActive() &&
outputDesc->sharesHwModuleWith(desc) &&
(newDevice != desc->device())) {
audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/);
setOutputDevice(desc,
dev,
true,
outputDesc->latency()*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");
return INVALID_OPERATION;
}
}
status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputDesc1,
audio_stream_type_t stream,
audio_devices_t device,
uint32_t *delayMs)
{
// cannot start playback of STREAM_TTS if any other output is being used
uint32_t beaconMuteLatency = 0;
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
ALOGW("startSource() invalid stream %d", stream);
return INVALID_OPERATION;
}
#ifdef NON_WEARABLE_TARGET
sp<AudioOutputDescriptor> outputDesc = outputDesc1;
#else
sp<SwAudioOutputDescriptor> outputDesc = (sp<SwAudioOutputDescriptor>) outputDesc1;
#endif
*delayMs = 0;
if (stream == AUDIO_STREAM_TTS) {
ALOGV("\t found BEACON stream");
if (mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
return INVALID_OPERATION;
} else {
beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
}
} else {
// some playback other than beacon starts
beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
}
// increment usage count for this stream on the requested output:
// NOTE that the usage count is the same for duplicated output and hardware output which is
// necessary for a correct control of hardware output routing by startOutput() and stopOutput()
outputDesc->changeRefCount(stream, 1);
if (outputDesc->mRefCount[stream] == 1 || device != AUDIO_DEVICE_NONE) {
// starting an output being rerouted?
if (device == AUDIO_DEVICE_NONE) {
device = getNewOutputDevice(outputDesc, false /*fromCache*/);
}
routing_strategy strategy = getStrategy(stream);
bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
(strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
(beaconMuteLatency > 0);
uint32_t waitMs = beaconMuteLatency;
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() != device) {
force = true;
}
// wait for audio on other active outputs to be presented when starting
// a notification so that audio focus effect can propagate, or that a mute/unmute
// event occurred for beacon
uint32_t latency = desc->latency();
if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
waitMs = latency;
}
}
}
uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force);
// handle special case for sonification while in call
if (isInCall()) {
handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
}
// apply volume rules for current stream and device if necessary
checkAndSetVolume(stream,
mStreams.valueFor(stream).getVolumeIndex(device),
outputDesc,
device);
// update the outputs if starting an output with a stream that can affect notification
// routing
handleNotificationRoutingForStream(stream);
// force reevaluating accessibility routing when ringtone or alarm starts
if (strategy == STRATEGY_SONIFICATION) {
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
}
else {
// handle special case for sonification while in call
if (isInCall()) {
handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
}
}
return NO_ERROR;
}
void AudioPolicyManagerCustom::handleIncallSonification(audio_stream_type_t stream,
bool starting, bool stateChange,
audio_io_handle_t output)
{
if(!hasPrimaryOutput()) {
return;
}
// no action needed for AUDIO_STREAM_PATCH stream type, it's for internal flinger tracks
if (stream == AUDIO_STREAM_PATCH) {
return;
}
// if the stream pertains to sonification strategy and we are in call we must
// mute the stream if it is low visibility. If it is high visibility, we must play a tone
// in the device used for phone strategy and play the tone if the selected device does not
// interfere with the device used for phone strategy
// if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
// many times as there are active tracks on the output
const routing_strategy stream_strategy = getStrategy(stream);
if ((stream_strategy == STRATEGY_SONIFICATION) ||
((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
stream, starting, outputDesc->mDevice, stateChange);
if (outputDesc->mRefCount[stream]) {
int muteCount = 1;
if (stateChange) {
muteCount = outputDesc->mRefCount[stream];
}
if (audio_is_low_visibility(stream)) {
ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
for (int i = 0; i < muteCount; i++) {
setStreamMute(stream, starting, outputDesc);
}
} else {
ALOGV("handleIncallSonification() high visibility");
if (outputDesc->device() &
getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
for (int i = 0; i < muteCount; i++) {
setStreamMute(stream, starting, outputDesc);
}
}
if (starting) {
mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
AUDIO_STREAM_VOICE_CALL);
} else {
mpClientInterface->stopTone();
}
}
}
}
}
void AudioPolicyManagerCustom::handleNotificationRoutingForStream(audio_stream_type_t stream) {
switch(stream) {
case AUDIO_STREAM_MUSIC:
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
updateDevicesAndOutputs();
break;
default:
break;
}
}
#ifdef NON_WEARABLE_TARGET
status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream,
int index,
const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device,
int delayMs, bool force)
#else
status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream,
int index,
const sp<SwAudioOutputDescriptor>& outputDesc,
audio_devices_t device,
int delayMs, bool force)
#endif
{
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
ALOGW("checkAndSetVolume() invalid stream %d", stream);
return INVALID_OPERATION;
}
// do not change actual stream volume if the stream is muted
if (outputDesc->mMuteCount[stream] != 0) {
ALOGVV("checkAndSetVolume() stream %d muted count %d",
stream, outputDesc->mMuteCount[stream]);
return NO_ERROR;
}
audio_policy_forced_cfg_t forceUseForComm =
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION);
// do not change in call volume if bluetooth is connected and vice versa
if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
(stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) {
ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
stream, forceUseForComm);
return INVALID_OPERATION;
}
if (device == AUDIO_DEVICE_NONE) {
device = outputDesc->device();
}
float volumeDb = computeVolume(stream, index, device);
if (outputDesc->isFixedVolume(device)) {
volumeDb = 0.0f;
}
outputDesc->setVolume(volumeDb, stream, device, delayMs, force);
if (stream == AUDIO_STREAM_VOICE_CALL ||
stream == AUDIO_STREAM_BLUETOOTH_SCO) {
float voiceVolume;
// Force voice volume to max for bluetooth SCO as volume is managed by the headset
if (stream == AUDIO_STREAM_VOICE_CALL) {
voiceVolume = (float)index/(float)mStreams.valueFor(stream).getVolumeIndexMax();
} else {
voiceVolume = 1.0;
}
if (voiceVolume != mLastVoiceVolume && ((outputDesc == mPrimaryOutput) ||
isDirectOutput(outputDesc->mIoHandle) || device & AUDIO_DEVICE_OUT_ALL_USB)) {
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
mLastVoiceVolume = voiceVolume;
}
#ifdef FM_POWER_OPT
} else if (stream == AUDIO_STREAM_MUSIC && hasPrimaryOutput() &&
outputDesc == mPrimaryOutput) {
AudioParameter param = AudioParameter();
param.addFloat(String8("fm_volume"), Volume::DbToAmpl(volumeDb));
mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString(), delayMs);
#endif /* FM_POWER_OPT end */
}
return NO_ERROR;
}
bool AudioPolicyManagerCustom::isDirectOutput(audio_io_handle_t output) {
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_io_handle_t curOutput = mOutputs.keyAt(i);
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
return true;
}
}
return false;
}
audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice(
audio_devices_t device,
audio_session_t session __unused,
audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
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 SwAudioOutputDescriptor(NULL,
mpClientInterface);
outputDesc->mDevice = mTestDevice;
outputDesc->mLatency = mTestLatencyMs;
outputDesc->mFlags =
(audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
outputDesc->mRefCount[stream] = 0;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = mTestSamplingRate;
config.channel_mask = mTestChannels;
config.format = mTestFormat;
if (offloadInfo != NULL) {
config.offload_info = *offloadInfo;
}
status = mpClientInterface->openOutput(0,
&mTestOutputs[mCurOutput],
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
if (status == NO_ERROR) {
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mFormat = config.format;
outputDesc->mChannelMask = config.channel_mask;
AudioParameter outputCmd = AudioParameter();
outputCmd.addInt(String8("set_id"),mCurOutput);
mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
addOutput(mTestOutputs[mCurOutput], outputDesc);
}
}
return mTestOutputs[mCurOutput];
}
#endif //AUDIO_POLICY_TEST
if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
(stream != AUDIO_STREAM_MUSIC)) {
// compress should not be used for non-music streams
ALOGE("Offloading only allowed with music stream");
return 0;
}
#ifdef COMPRESS_VOIP_ENABLED
if ((stream == AUDIO_STREAM_VOICE_CALL) &&
(channelMask == 1) &&
(samplingRate == 8000 || samplingRate == 16000)) {
// Allow Voip direct output only if:
// audio mode is MODE_IN_COMMUNCATION; AND
// voip output is not opened already; AND
// requested sample rate matches with that of voip input stream (if opened already)
int value = 0;
uint32_t mode = 0, voipOutCount = 1, voipSampleRate = 1;
String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
String8("audio_mode"));
AudioParameter result = AudioParameter(valueStr);
if (result.getInt(String8("audio_mode"), value) == NO_ERROR) {
mode = value;
}
valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
String8("voip_out_stream_count"));
result = AudioParameter(valueStr);
if (result.getInt(String8("voip_out_stream_count"), value) == NO_ERROR) {
voipOutCount = value;
}
valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
String8("voip_sample_rate"));
result = AudioParameter(valueStr);
if (result.getInt(String8("voip_sample_rate"), value) == NO_ERROR) {
voipSampleRate = value;
}
if ((mode == AUDIO_MODE_IN_COMMUNICATION) && (voipOutCount == 0) &&
((voipSampleRate == 0) || (voipSampleRate == samplingRate))) {
if (audio_is_linear_pcm(format)) {
char propValue[PROPERTY_VALUE_MAX] = {0};
property_get("use.voice.path.for.pcm.voip", propValue, "0");
bool voipPcmSysPropEnabled = !strncmp("true", propValue, sizeof("true"));
if (voipPcmSysPropEnabled && (format == AUDIO_FORMAT_PCM_16_BIT)) {
flags = (audio_output_flags_t)((flags &~AUDIO_OUTPUT_FLAG_FAST) |
AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT);
ALOGD("Set VoIP and Direct output flags for PCM format");
}
}
}
}
#endif
#ifdef VOICE_CONCURRENCY
char propValue[PROPERTY_VALUE_MAX];
bool prop_play_enabled=false, prop_voip_enabled = false;
if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
prop_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
}
if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
}
if (prop_play_enabled && mvoice_call_state) {
//check if voice call is active / running in background
if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
((AUDIO_MODE_IN_CALL == mPrevPhoneState)
&& (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
{
if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
if(prop_voip_enabled) {
ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
flags );
return 0;
}
}
else {
if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
ALOGD("voice_conc:IN call mode adding ULL flags .. flags: %x ", flags );
flags = AUDIO_OUTPUT_FLAG_FAST;
} else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
if (AUDIO_STREAM_MUSIC == stream) {
flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
ALOGD("voice_conc:IN call mode adding deep-buffer flags %x ", flags );
}
else {
flags = AUDIO_OUTPUT_FLAG_FAST;
ALOGD("voice_conc:IN call mode adding fast flags %x ", flags );
}
}
}
}
} else if (prop_voip_enabled && mvoice_call_state) {
//check if voice call is active / running in background
//some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
//return only ULL ouput
if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
((AUDIO_MODE_IN_CALL == mPrevPhoneState)
&& (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
{
if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
flags );
return 0;
}
}
}
#endif
#ifdef RECORD_PLAY_CONCURRENCY
char recConcPropValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;
if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
}
if ((prop_rec_play_enabled) &&
((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCount() > 0))) {
if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
if (AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
// allow VoIP using voice path
// Do nothing
} else if((flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
ALOGD("voice_conc:MODE_IN_COMM is setforcing deep buffer output for non ULL... flags: %x", flags);
// use deep buffer path for all non ULL outputs
flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
} else if ((flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
ALOGD("voice_conc:Record mode is on forcing deep buffer output for non ULL... flags: %x ", flags);
// use deep buffer path for all non ULL outputs
flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
}
if (prop_rec_play_enabled &&
(stream == AUDIO_STREAM_ENFORCED_AUDIBLE)) {
ALOGD("Record conc is on forcing ULL output for ENFORCED_AUDIBLE");
flags = AUDIO_OUTPUT_FLAG_FAST;
}
#endif
#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
/*
* WFD audio routes back to target speaker when starting a ringtone playback.
* This is because primary output is reused for ringtone, so output device is
* updated based on SONIFICATION strategy for both ringtone and music playback.
* The same issue is not seen on remoted_submix HAL based WFD audio because
* primary output is not reused and a new output is created for ringtone playback.
* Issue is fixed by updating output flag to AUDIO_OUTPUT_FLAG_FAST when there is
* a non-music stream playback on WFD, so primary output is not reused for ringtone.
*/
audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
if ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY)
&& (stream != AUDIO_STREAM_MUSIC)) {
ALOGD("WFD audio: use OUTPUT_FLAG_FAST for non music stream. flags:%x", flags );
//For voip paths
if(flags & AUDIO_OUTPUT_FLAG_DIRECT)
flags = AUDIO_OUTPUT_FLAG_DIRECT;
else //route every thing else to ULL path
flags = AUDIO_OUTPUT_FLAG_FAST;
}
#endif
// open a direct output if required by specified parameters
//force direct flag if offload flag is set: offloading implies a direct output stream
// and all common behaviors are driven by checking only the direct flag
// this should normally be set appropriately in the policy configuration file
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
// only allow deep buffering for music stream type
if (stream != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
if (stream == AUDIO_STREAM_TTS) {
flags = AUDIO_OUTPUT_FLAG_TTS;
}
// open a direct output if required by specified parameters
//force direct flag if offload flag is set: offloading implies a direct output stream
// and all common behaviors are driven by checking only the direct flag
// this should normally be set appropriately in the policy configuration file
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
// only allow deep buffering for music stream type
if (stream != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
if (stream == AUDIO_STREAM_TTS) {
flags = AUDIO_OUTPUT_FLAG_TTS;
}
sp<IOProfile> profile;
// skip direct output selection if the request can obviously be attached to a mixed output
// and not explicitly requested
if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
audio_is_linear_pcm(format) && samplingRate <= 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) ||
!mEffects.isNonOffloadableEffectEnabled()) {
profile = getProfileForDirectOutput(device,
samplingRate,
format,
channelMask,
(audio_output_flags_t)flags);
}
if (profile != 0) {
sp<SwAudioOutputDescriptor> outputDesc = NULL;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
outputDesc = desc;
// reuse direct output if currently open and configured with same parameters
if ((samplingRate == outputDesc->mSamplingRate) &&
(format == outputDesc->mFormat) &&
(channelMask == outputDesc->mChannelMask)) {
outputDesc->mDirectOpenCount++;
ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
return mOutputs.keyAt(i);
}
}
}
// close direct output if currently open and configured with different parameters
if (outputDesc != NULL) {
closeOutput(outputDesc->mIoHandle);
}
// if the selected profile is offloaded and no offload info was specified,
// create a default one
audio_offload_info_t defaultOffloadInfo = AUDIO_INFO_INITIALIZER;
if ((profile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
defaultOffloadInfo.sample_rate = samplingRate;
defaultOffloadInfo.channel_mask = channelMask;
defaultOffloadInfo.format = format;
defaultOffloadInfo.stream_type = stream;
defaultOffloadInfo.bit_rate = 0;
defaultOffloadInfo.duration_us = -1;
defaultOffloadInfo.has_video = true; // conservative
defaultOffloadInfo.is_streaming = true; // likely
offloadInfo = &defaultOffloadInfo;
}
outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface);
outputDesc->mDevice = device;
outputDesc->mLatency = 0;
outputDesc->mFlags = (audio_output_flags_t)(outputDesc->mFlags | flags);
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = samplingRate;
config.channel_mask = channelMask;
config.format = format;
if (offloadInfo != NULL) {
config.offload_info = *offloadInfo;
}
status = mpClientInterface->openOutput(profile->getModuleHandle(),
&output,
&config,
&outputDesc->mDevice,
String8(""),
&outputDesc->mLatency,
outputDesc->mFlags);
// only accept an output with the requested parameters
if (status != NO_ERROR ||
(samplingRate != 0 && samplingRate != config.sample_rate) ||
(format != AUDIO_FORMAT_DEFAULT && format != config.format) ||
(channelMask != 0 && channelMask != config.channel_mask)) {
ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
"format %d %d, channelMask %04x %04x", output, samplingRate,
outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
outputDesc->mChannelMask);
if (output != AUDIO_IO_HANDLE_NONE) {
mpClientInterface->closeOutput(output);
}
// fall back to mixer output if possible when the direct output could not be open
if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
goto non_direct_output;
}
return AUDIO_IO_HANDLE_NONE;
}
outputDesc->mSamplingRate = config.sample_rate;
outputDesc->mChannelMask = config.channel_mask;
outputDesc->mFormat = config.format;
outputDesc->mRefCount[stream] = 0;
outputDesc->mStopTime[stream] = 0;
outputDesc->mDirectOpenCount = 1;
audio_io_handle_t srcOutput = getOutputForEffect();
addOutput(output, outputDesc);
audio_io_handle_t dstOutput = getOutputForEffect();
if (dstOutput == output) {
#ifdef DOLBY_ENABLE
status_t status = mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
if (status == NO_ERROR) {
for (size_t i = 0; i < mEffects.size(); i++) {
sp<EffectDescriptor> desc = mEffects.valueAt(i);
if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX) {
// update the mIo member of EffectDescriptor for the global effect
ALOGV("%s updating mIo", __FUNCTION__);
desc->mIo = dstOutput;
}
}
} else {
ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__, srcOutput, dstOutput);
}
#else // DOLBY_END
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
#endif // LINE_ADDED_BY_DOLBY
}
mPreviousOutputs = mOutputs;
ALOGV("getOutput() returns new direct output %d", output);
mpClientInterface->onAudioPortListUpdate();
return output;
}
non_direct_output:
// ignoring channel mask due to downmix capability in mixer
// open a non direct output
// for non direct outputs, only PCM is supported
if (audio_is_linear_pcm(format)) {
// get which output is suitable for the specified stream. The actual
// routing change will happen when startOutput() will be called
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
// at this stage we should ignore the DIRECT flag as no direct output could be found earlier
flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
output = selectOutput(outputs, flags, format);
}
ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
"format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
ALOGV(" getOutputForDevice() returns output %d", output);
return output;
}
status_t AudioPolicyManagerCustom::getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
uid_t uid,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_input_flags_t flags,
audio_port_handle_t selectedDeviceId,
input_type_t *inputType)
{
audio_source_t inputSource = attr->source;
#ifdef VOICE_CONCURRENCY
char propValue[PROPERTY_VALUE_MAX];
bool prop_rec_enabled=false, prop_voip_enabled = false;
if(property_get("voice.record.conc.disabled", propValue, NULL)) {
prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
}
if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
}
if (prop_rec_enabled && mvoice_call_state) {
//check if voice call is active / running in background
//some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
//Need to block input request
if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
(AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
{
switch(inputSource) {
case AUDIO_SOURCE_VOICE_UPLINK:
case AUDIO_SOURCE_VOICE_DOWNLINK:
case AUDIO_SOURCE_VOICE_CALL:
ALOGD("voice_conc:Creating input during incall mode for inputSource: %d",
inputSource);
break;
case AUDIO_SOURCE_VOICE_COMMUNICATION:
if(prop_voip_enabled) {
ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
inputSource);
return NO_INIT;
}
break;
default:
ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
inputSource);
return NO_INIT;
}
}
}//check for VoIP flag
else if(prop_voip_enabled && mvoice_call_state) {
//check if voice call is active / running in background
//some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
//Need to block input request
if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
(AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
{
if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {
ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
return NO_INIT;
}
}
}
#endif
return AudioPolicyManager::getInputForAttr(attr,
input,
session,
uid,
samplingRate,
format,
channelMask,
flags,
selectedDeviceId,
inputType);
}
status_t AudioPolicyManagerCustom::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 (!is_virtual_input_device(inputDesc->mDevice)) {
// for a non-virtual input device, check if there is another (non-virtual) active input
audio_io_handle_t activeInput = mInputs.getActiveInput();
if (activeInput != 0 && activeInput != input) {
// If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
// otherwise the active input continues and the new input cannot be started.
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
if (activeDesc->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;
}
}
}
// Routing?
mInputRoutes.incRouteActivity(session);
#ifdef RECORD_PLAY_CONCURRENCY
mIsInputRequestOnProgress = true;
char getPropValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;
if (property_get("rec.playback.conc.disabled", getPropValue, NULL)) {
prop_rec_play_enabled = atoi(getPropValue) || !strncmp("true", getPropValue, 4);
}
if ((prop_rec_play_enabled) &&(mInputs.activeInputsCount() == 0)){
// send update to HAL on record playback concurrency
AudioParameter param = AudioParameter();
param.add(String8("rec_play_conc_on"), String8("true"));
ALOGD("startInput() setParameters rec_play_conc is setting to ON ");
mpClientInterface->setParameters(0, param.toString());
// Call invalidate to reset all opened non ULL audio tracks
// Move tracks associated to this strategy from previous output to new output
for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
// Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder)
if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE) && (i != AUDIO_STREAM_PATCH)) {
ALOGD("Invalidate on releaseInput for stream :: %d ", i);
//FIXME see fixme on name change
mpClientInterface->invalidateStream((audio_stream_type_t)i);
}
}
// close compress tracks
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
ALOGD("ouput desc / profile is NULL");
continue;
}
if (outputDesc->mProfile->mFlags
& AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
// close compress sessions
ALOGD("calling closeOutput on record conc for COMPRESS output");
closeOutput(mOutputs.keyAt(i));
}
}
}
#endif
if (inputDesc->mRefCount == 0 || mInputRoutes.hasRouteChanged(session)) {
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
&& ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId,
MIX_STATE_MIXING);
}
if (mInputs.activeInputsCount() == 0) {
SoundTrigger::setCaptureState(true);
}
setInputDevice(input, getNewInputDevice(input), true /* force */);
// automatically enable the remote submix output when input is started if not
// used by a policy mix of type MIX_TYPE_RECORDERS
// For remote submix (a virtual device), we open only one input per capture request.
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
String8 address = String8("");
if (inputDesc->mPolicyMix == NULL) {
address = String8("0");
} else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
address = inputDesc->mPolicyMix->mRegistrationId;
}
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address, "remote-submix");
}
}
}
ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
inputDesc->mRefCount++;
#ifdef RECORD_PLAY_CONCURRENCY
mIsInputRequestOnProgress = false;
#endif
return NO_ERROR;
}
status_t AudioPolicyManagerCustom::stopInput(audio_io_handle_t input,
audio_session_t session)
{
status_t status;
status = AudioPolicyManager::stopInput(input, session);
#ifdef RECORD_PLAY_CONCURRENCY
char propValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;
if (property_get("rec.playback.conc.disabled", propValue, NULL)) {
prop_rec_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
}
if ((prop_rec_play_enabled) && (mInputs.activeInputsCount() == 0)) {
//send update to HAL on record playback concurrency
AudioParameter param = AudioParameter();
param.add(String8("rec_play_conc_on"), String8("false"));
ALOGD("stopInput() setParameters rec_play_conc is setting to OFF ");
mpClientInterface->setParameters(0, param.toString());
//call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
//Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder stop tone)
if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE) && (i != AUDIO_STREAM_PATCH)) {
ALOGD(" Invalidate on stopInput for stream :: %d ", i);
//FIXME see fixme on name change
mpClientInterface->invalidateStream((audio_stream_type_t)i);
}
}
}
#endif
return status;
}
AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
: AudioPolicyManager(clientInterface)
{
#ifdef RECORD_PLAY_CONCURRENCY
mIsInputRequestOnProgress = false;
#endif
#ifdef VOICE_CONCURRENCY
mFallBackflag = getFallBackPath();
#endif
}
}