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LeafOS / LeafOS-Project / android_vendor_qcom_opensource_audio / 2c9add7250b8fa0590f6106fc837510549f76367 / . / policy_hal / AudioPolicyManager.cpp
blob: 830ebe1ca2f2020091719a48c6f784e55e156a6e [file] [log] [blame]
/*
* Copyright (c) 2013-2019 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.
*/
#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
// 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)
#define SAMPLE_RATE_8000 8000
#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>
namespace android {
/*audio policy: workaround for truncated touch sounds*/
//FIXME: workaround for truncated touch sounds
// to be removed when the problem is handled by system UI
#define TOUCH_SOUND_FIXED_DELAY_MS 100
sp<APMConfigHelper> AudioPolicyManagerCustom::mApmConfigs = new APMConfigHelper();
audio_output_flags_t AudioPolicyManagerCustom::getFallBackPath()
{
audio_output_flags_t flag = AUDIO_OUTPUT_FLAG_FAST;
const char *fallback_path = mApmConfigs->getVoiceConcFallbackPath().c_str();
if (strlen(fallback_path) > 0) {
if (!strncmp(fallback_path, "deep-buffer", 11)) {
flag = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
else if (!strncmp(fallback_path, "fast", 4)) {
flag = AUDIO_OUTPUT_FLAG_FAST;
}
else {
ALOGD("voice_conc:not a recognised path(%s) in prop vendor.voice.conc.fallbackpath",
fallback_path);
}
}
else {
ALOGD("voice_conc:prop vendor.voice.conc.fallbackpath not set");
}
ALOGD("voice_conc:picked up flag(0x%x) from prop vendor.voice.conc.fallbackpath",
flag);
return flag;
}
// ----------------------------------------------------------------------------
// 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 deviceType,
audio_policy_dev_state_t state,
const char *device_address,
const char *device_name,
audio_format_t encodedFormat)
{
ALOGD("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s format 0x%X",
deviceType, state, device_address, device_name, encodedFormat);
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(deviceType) && !audio_is_input_device(deviceType)) return BAD_VALUE;
sp<DeviceDescriptor> device =
mHwModules.getDeviceDescriptor(deviceType, device_address, device_name, encodedFormat,
state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
if (device == 0) {
return INVALID_OPERATION;
}
// handle output devices
if (audio_is_output_device(deviceType)) {
SortedVector <audio_io_handle_t> outputs;
ssize_t index = mAvailableOutputDevices.indexOf(device);
// save a copy of the opened output descriptors before any output is opened or closed
// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
mPreviousOutputs = mOutputs;
switch (state)
{
// handle output device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
if (mApmConfigs->isHDMISpkEnabled() &&
(popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
if (!strncmp(device_address, "hdmi_spkr", 9)) {
mHdmiAudioDisabled = false;
} else {
mHdmiAudioEvent = true;
}
}
ALOGW("setDeviceConnectionState() device already connected: %x", deviceType);
return INVALID_OPERATION;
}
ALOGV("%s() connecting device %s format %x",
__func__, device->toString().c_str(), encodedFormat);
// register new device as available
index = mAvailableOutputDevices.add(device);
if (mApmConfigs->isHDMISpkEnabled() &&
(popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
if (!strncmp(device_address, "hdmi_spkr", 9)) {
mHdmiAudioDisabled = false;
} else {
mHdmiAudioEvent = true;
}
if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
mAvailableOutputDevices.remove(device);
ALOGW("HDMI sink not connected, do not route audio to HDMI out");
return INVALID_OPERATION;
}
}
if (index >= 0) {
sp<HwModule> module = mHwModules.getModuleForDevice(device, encodedFormat);
if (module == 0) {
ALOGD("setDeviceConnectionState() could not find HW module for device %s",
device->toString().c_str());
mAvailableOutputDevices.remove(device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() module name=%s", module->getName());
mAvailableOutputDevices[index]->attach(module);
} else {
return NO_MEMORY;
}
// Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
// parameters on newly connected devices (instead of opening the outputs...)
broadcastDeviceConnectionState(device, state);
if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
mAvailableOutputDevices.remove(device);
mHwModules.cleanUpForDevice(device);
broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
return INVALID_OPERATION;
}
if (deviceType == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
chkDpConnAndAllowedForVoice();
}
// outputs should never be empty here
ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
"checkOutputsForDevice() returned no outputs but status OK");
ALOGV("%s() checkOutputsForDevice() returned %zu outputs", __func__, outputs.size());
} break;
// handle output device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
if (mApmConfigs->isHDMISpkEnabled() &&
(popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
if (!strncmp(device_address, "hdmi_spkr", 9)) {
mHdmiAudioDisabled = true;
} else {
mHdmiAudioEvent = false;
}
}
ALOGW("setDeviceConnectionState() device not connected: %x", deviceType);
return INVALID_OPERATION;
}
ALOGV("%s() disconnecting output device %s", __func__, device->toString().c_str());
// Send Disconnect to HALs
broadcastDeviceConnectionState(device, state);
// remove device from available output devices
mAvailableOutputDevices.remove(device);
mOutputs.clearSessionRoutesForDevice(device);
if (mApmConfigs->isHDMISpkEnabled() &&
(popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
if (!strncmp(device_address, "hdmi_spkr", 9)) {
mHdmiAudioDisabled = true;
} else {
mHdmiAudioEvent = false;
}
}
checkOutputsForDevice(device, state, outputs);
// Reset active device codec
device->setEncodedFormat(AUDIO_FORMAT_DEFAULT);
if (deviceType == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
mEngine->setDpConnAndAllowedForVoice(false);
}
} break;
default:
ALOGE("%s() invalid state: %x", __func__, state);
return BAD_VALUE;
}
// Propagate device availability to Engine
setEngineDeviceConnectionState(device, state);
if (!outputs.isEmpty()) {
for (size_t i = 0; i < outputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
// close voip output before track invalidation to allow creation of
// new voip stream from restoreTrack
if ((desc->mFlags == (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_VOIP_RX)) != 0) {
closeOutput(outputs[i]);
outputs.remove(outputs[i]);
}
}
}
// No need to evaluate playback routing when connecting a remote submix
// output device used by a dynamic policy of type recorder as no
// playback use case is affected.
bool doCheckForDeviceAndOutputChanges = true;
if (device->type() == AUDIO_DEVICE_OUT_REMOTE_SUBMIX
&& strncmp(device_address, "0", AUDIO_DEVICE_MAX_ADDRESS_LEN) != 0) {
for (audio_io_handle_t output : outputs) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
sp<AudioPolicyMix> policyMix = desc->mPolicyMix.promote();
if (policyMix != nullptr
&& policyMix->mMixType == MIX_TYPE_RECORDERS
&& strncmp(device_address,
policyMix->mDeviceAddress.string(),
AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
doCheckForDeviceAndOutputChanges = false;
break;
}
}
}
auto checkCloseOutputs = [&]() {
// outputs must be closed after checkOutputForAllStrategies() is executed
if (!outputs.isEmpty()) {
for (audio_io_handle_t output : outputs) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
// close unused outputs after device disconnection or direct outputs that have
// been opened by checkOutputsForDevice() to query dynamic parameters
if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
(desc->mDirectOpenCount == 0))) {
closeOutput(output);
}
}
// check A2DP again after closing A2DP output to reset mA2dpSuspended if needed
return true;
}
return false;
};
if (doCheckForDeviceAndOutputChanges) {
checkForDeviceAndOutputChanges(checkCloseOutputs);
} else {
checkCloseOutputs();
}
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevices);
}
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
// do not force device change on duplicated output because if device is 0, it will
// also force a device 0 for the two outputs it is duplicated to which may override
// a valid device selection on those outputs.
bool force = !desc->isDuplicated()
&& (!device_distinguishes_on_address(deviceType)
// always force when disconnecting (a non-duplicated device)
|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
setOutputDevices(desc, newDevices, force, 0);
}
}
if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
cleanUpForDevice(device);
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is output device
// handle input devices
if (audio_is_input_device(deviceType)) {
ssize_t index = mAvailableInputDevices.indexOf(device);
switch (state)
{
// handle input device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
ALOGW("setDeviceConnectionState() device already connected: %d", deviceType);
return INVALID_OPERATION;
}
sp<HwModule> module = mHwModules.getModuleForDevice(device, AUDIO_FORMAT_DEFAULT);
if (module == NULL) {
ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
deviceType);
return INVALID_OPERATION;
}
// Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
// parameters on newly connected devices (instead of opening the inputs...)
broadcastDeviceConnectionState(device, state);
if (checkInputsForDevice(device, state) != NO_ERROR) {
broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
mHwModules.cleanUpForDevice(device);
return INVALID_OPERATION;
}
index = mAvailableInputDevices.add(device);
if (index >= 0) {
mAvailableInputDevices[index]->attach(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", deviceType);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() disconnecting input device %x", deviceType);
// Set Disconnect to HALs
broadcastDeviceConnectionState(device, state);
checkInputsForDevice(device, state);
mAvailableInputDevices.remove(device);
} break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
// Propagate device availability to Engine
setEngineDeviceConnectionState(device, state);
checkCloseInputs();
/*audio policy: fix call volume over USB*/
// As the input device list can impact the output device selection, update
// getDeviceForStrategy() cache
updateDevicesAndOutputs();
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
updateCallRouting(newDevices);
}
if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
cleanUpForDevice(device);
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is input device
ALOGW("setDeviceConnectionState() invalid device: %x", deviceType);
return BAD_VALUE;
}
void AudioPolicyManagerCustom::chkDpConnAndAllowedForVoice()
{
String8 value;
bool connAndAllowed = false;
String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
String8("dp_for_voice"));
AudioParameter result = AudioParameter(valueStr);
if (result.get(String8("dp_for_voice"), value) == NO_ERROR) {
connAndAllowed = value.contains("true");
}
mEngine->setDpConnAndAllowedForVoice(connAndAllowed);
}
bool AudioPolicyManagerCustom::isInvalidationOfMusicStreamNeeded(const audio_attributes_t &attr)
{
if (followsSameRouting(attr, attributes_initializer(AUDIO_USAGE_MEDIA))) {
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> newOutputDesc = mOutputs.valueAt(i);
if (newOutputDesc->mFormat == AUDIO_FORMAT_DSD)
return false;
}
}
return true;
}
void AudioPolicyManagerCustom::checkOutputForAttributes(const audio_attributes_t &attr)
{
DeviceVector oldDevices = mEngine->getOutputDevicesForAttributes(attr, 0, true /*fromCache*/);
DeviceVector newDevices = mEngine->getOutputDevicesForAttributes(attr, 0, false /*fromCache*/);
SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevices(oldDevices, mPreviousOutputs);
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevices(newDevices, mOutputs);
// also take into account external policy-related changes: add all outputs which are
// associated with policies in the "before" and "after" output vectors
ALOGVV("%s(): policy related outputs", __func__);
for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
if (desc != 0 && desc->mPolicyMix != NULL) {
srcOutputs.add(desc->mIoHandle);
ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
}
}
for (size_t i = 0 ; i < mOutputs.size() ; i++) {
const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != 0 && desc->mPolicyMix != NULL) {
dstOutputs.add(desc->mIoHandle);
ALOGVV(" new outputs: adding %d", desc->mIoHandle);
}
}
if ((srcOutputs != dstOutputs) && isInvalidationOfMusicStreamNeeded(attr)) {
AudioPolicyManager::checkOutputForAttributes(attr);
}
}
// 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)
{
DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
" BitRate=%u, duration=%" PRId64 " us, has_video=%d",
offloadInfo.sample_rate, offloadInfo.channel_mask,
offloadInfo.format,
offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
offloadInfo.has_video);
if (mMasterMono) {
return false; // no offloading if mono is set.
}
if (mApmConfigs->isVoiceConcEnabled()) {
if (mApmConfigs->isVoicePlayConcDisabled() && isInCall()) {
ALOGD("\n copl: blocking compress offload on call mode\n");
return false;
}
}
if (mApmConfigs->isVoiceDSDConcDisabled() &&
isInCall() && (offloadInfo.format == AUDIO_FORMAT_DSD)) {
ALOGD("blocking DSD compress offload on call mode");
return false;
}
if (mApmConfigs->isRecPlayConcEnabled()) {
if (mApmConfigs->isRecPlayConcDisabled() &&
((true == mIsInputRequestOnProgress) ||
(mInputs.activeInputsCountOnDevices(primaryInputDevices) > 0))) {
ALOGD("copl: blocking compress offload for record concurrency");
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 offload has been disabled
bool offloadDisabled = mApmConfigs->isAudioOffloadDisabled();
if (offloadDisabled) {
ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled);
return false;
}
//check if it's multi-channel AAC (includes sub formats) and FLAC format
if ((popcount(offloadInfo.channel_mask) > 2) &&
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
return false;
}
if (mApmConfigs->isExtnFormatsEnabled()) {
//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)) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) {
ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
return false;
}
// check against wma std bit rate restriction
if ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) {
int32_t sr_id = -1;
uint32_t min_bitrate, max_bitrate;
for (int i = 0; i < WMA_STD_NUM_FREQ; i++) {
if (offloadInfo.sample_rate == wmaStdSampleRateTbl[i]) {
sr_id = i;
break;
}
}
if ((sr_id < 0) || (popcount(offloadInfo.channel_mask) > 2)
|| (popcount(offloadInfo.channel_mask) <= 0)) {
ALOGE("invalid sample rate or channel count");
return false;
}
min_bitrate = wmaStdMinAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1];
max_bitrate = wmaStdMaxAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1];
if ((offloadInfo.bit_rate > max_bitrate) || (offloadInfo.bit_rate < min_bitrate)) {
ALOGD("offload disabled for WMA clips with unsupported bit rate");
ALOGD("bit_rate %d, max_bitrate %d, min_bitrate %d", offloadInfo.bit_rate, max_bitrate, min_bitrate);
return false;
}
}
// Safely choose the min bitrate as threshold and leave the restriction to NT decoder as we can't distinguish wma pro and wma lossless here.
if ((((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_PRO)) ||
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_LOSSLESS))) {
ALOGD("offload disabled for WMA_PRO/WMA_LOSSLESS clips with bit rate over maximum supported value");
return false;
}
}
//TODO: enable audio offloading with video when ready
if (offloadInfo.has_video && !mApmConfigs->isAudioOffloadVideoEnabled()) {
ALOGV("isOffloadSupported: has_video == true, returning false");
return false;
}
if (offloadInfo.has_video && offloadInfo.is_streaming &&
!mApmConfigs->isAVStreamingOffloadEnabled()) {
ALOGW("offload disabled by vendor.audio.av.streaming.offload.enable %d",
mApmConfigs->isAVStreamingOffloadEnabled());
return false;
}
//If duration is less than minimum value defined in property, return false
if (mApmConfigs->getAudioOffloadMinDuration() > 0) {
if (offloadInfo.duration_us < (mApmConfigs->getAudioOffloadMinDuration() * 1000000 )) {
ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%u)", mApmConfigs->getAudioOffloadMinDuration());
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/ formats,
//do not check duration for other audio formats, e.g. 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_FLAC) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))
return false;
if (mApmConfigs->isExtnFormatsEnabled()) {
if (((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) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DSD) ||
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))
return false;
}
}
// Do not allow offloading if one non offloadable effect is enabled. This prevents from
// creating an offloaded track and tearing it down immediately after start when audioflinger
// detects there is an active non offloadable effect.
// FIXME: We should check the audio session here but we do not have it in this context.
// This may prevent offloading in rare situations where effects are left active by apps
// in the background.
if (mEffects.isNonOffloadableEffectEnabled()) {
return false;
}
// See if there is a profile to support this.
// AUDIO_DEVICE_NONE
sp<IOProfile> profile = getProfileForOutput(DeviceVector() /*ignore device */,
offloadInfo.sample_rate,
offloadInfo.format,
offloadInfo.channel_mask,
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,
true /*directOnly*/);
ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
return (profile != 0);
}
void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)
{
ALOGD("setPhoneState() state %d", state);
// store previous phone state for management of sonification strategy below
int oldState = mEngine->getPhoneState();
if (mEngine->setPhoneState(state) != NO_ERROR) {
ALOGW("setPhoneState() invalid or same state %d", state);
return;
}
/// Opens: can these line be executed after the switch of volume curves???
if (isStateInCall(oldState)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
// force reevaluating accessibility routing when call stops
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
/**
* Switching to or from incall state or switching between telephony and VoIP lead to force
* routing command.
*/
bool force = ((is_state_in_call(oldState) != is_state_in_call(state))
|| (is_state_in_call(state) && (state != oldState)));
// check for device and output changes triggered by new phone state
checkForDeviceAndOutputChanges();
sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
if (mApmConfigs->isVoiceConcEnabled()) {
bool prop_playback_enabled = mApmConfigs->isVoicePlayConcDisabled();
bool prop_rec_enabled = mApmConfigs->isVoiceRecConcDisabled();
bool prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled();
if ((AUDIO_MODE_IN_CALL != oldState) && (AUDIO_MODE_IN_CALL == state)) {
ALOGD("voice_conc:Entering to call mode oldState :: %d state::%d ",
oldState, state);
mvoice_call_state = state;
if (prop_rec_enabled) {
//Close all active inputs
Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
if (activeInputs.size() != 0) {
for (size_t i = 0; i < activeInputs.size(); i++) {
sp<AudioInputDescriptor> activeInput = activeInputs[i];
switch(activeInput->source()) {
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",
activeInput->source());
break;
case AUDIO_SOURCE_VOICE_COMMUNICATION:
if (prop_voip_enabled) {
ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",
activeInput->source());
RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/);
for (const auto& activeClient : activeClients) {
closeClient(activeClient->portId());
}
}
break;
default:
ALOGD("voice_conc:CLOSING input on call setup for inputSource: %d",
activeInput->source());
RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/);
for (const auto& activeClient : activeClients) {
closeClient(activeClient->portId());
}
break;
}
}
}
} else if (prop_voip_enabled) {
Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
if (activeInputs.size() != 0) {
for (size_t i = 0; i < activeInputs.size(); i++) {
sp<AudioInputDescriptor> activeInput = activeInputs[i];
if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeInput->source()) {
ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeInput->source());
RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/);
for (const auto& activeClient : activeClients) {
closeClient(activeClient->portId());
}
}
}
}
}
if (prop_playback_enabled) {
// Move tracks associated to this strategy from previous output to new output
for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
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;
}
bool isFastFallBackNeeded =
((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & outputDesc->mProfile->getFlags());
if ((AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) && isFastFallBackNeeded) {
if (((!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & 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->getFlags() & 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->getFlags() & 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->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX) {
if (prop_voip_enabled) {
ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output");
closeOutput(mOutputs.keyAt(i));
}
}
else if (prop_playback_enabled
&& (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
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->getFlags() & 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 < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
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);
}
}
}
}
sp<SwAudioOutputDescriptor> outputDesc = NULL;
for (size_t i = 0; i < mOutputs.size(); i++) {
outputDesc = mOutputs.valueAt(i);
if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
ALOGD("voice_conc:ouput desc / profile is NULL");
continue;
}
if (mApmConfigs->isVoiceDSDConcDisabled() &&
(outputDesc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
(outputDesc->mFormat == AUDIO_FORMAT_DSD)) {
ALOGD("voice_conc:calling closeOutput on call mode for DSD COMPRESS output");
closeOutput(mOutputs.keyAt(i));
// call invalidate for music, so that DSD compress will fallback to deep-buffer.
mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
}
}
if (mApmConfigs->isRecPlayConcEnabled()) {
if (mApmConfigs->isRecPlayConcDisabled()) {
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->getFlags() & 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);
}
}
}
mPrevPhoneState = oldState;
int delayMs = 0;
if (isStateInCall(state)) {
nsecs_t sysTime = systemTime();
auto musicStrategy = streamToStrategy(AUDIO_STREAM_MUSIC);
auto sonificationStrategy = streamToStrategy(AUDIO_STREAM_ALARM);
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
// mute media and sonification strategies and delay device switch by the largest
// latency of any output where either strategy is active.
// This avoid sending the ring tone or music tail into the earpiece or headset.
if ((desc->isStrategyActive(musicStrategy,
SONIFICATION_HEADSET_MUSIC_DELAY,
sysTime) ||
desc->isStrategyActive(sonificationStrategy,
SONIFICATION_HEADSET_MUSIC_DELAY,
sysTime)) &&
(delayMs < (int)desc->latency()*2)) {
delayMs = desc->latency()*2;
}
setStrategyMute(musicStrategy, true, desc);
setStrategyMute(musicStrategy, false, desc, MUTE_TIME_MS,
mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA),
nullptr, true /*fromCache*/).types());
setStrategyMute(sonificationStrategy, true, desc);
setStrategyMute(sonificationStrategy, false, desc, MUTE_TIME_MS,
mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_ALARM),
nullptr, true /*fromCache*/).types());
}
}
if (hasPrimaryOutput()) {
// Note that despite the fact that getNewOutputDevice() is called on the primary output,
// the device returned is not necessarily reachable via this output
DeviceVector rxDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
// force routing command to audio hardware when ending call
// even if no device change is needed
if (isStateInCall(oldState) && rxDevices.isEmpty()) {
rxDevices = mPrimaryOutput->devices();
}
if (state == AUDIO_MODE_IN_CALL) {
updateCallRouting(rxDevices, delayMs);
} else if (oldState == AUDIO_MODE_IN_CALL) {
if (mCallRxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
mCallRxPatch.clear();
}
if (mCallTxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
mCallTxPatch.clear();
}
setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
} else {
setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
}
}
//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) {
DeviceVector newDevices = getNewOutputDevices(mOutputs.valueFor(output), false /*fromCache*/);
setOutputDevices(mOutputs.valueFor(output), newDevices, !newDevices.isEmpty());
}
}
if (isStateInCall(state)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
// force reevaluating accessibility routing when call starts
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
mLimitRingtoneVolume = (state == AUDIO_MODE_RINGTONE &&
isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY));
}
void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config)
{
ALOGD("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
if (config == mEngine->getForceUse(usage)) {
return;
}
if (mEngine->setForceUse(usage, config) != NO_ERROR) {
ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
return;
}
bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
(usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
(usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
// check for device and output changes triggered by new force usage
checkForDeviceAndOutputChanges();
/*audio policy: workaround for truncated touch sounds*/
//FIXME: workaround for truncated touch sounds
// to be removed when the problem is handled by system UI
uint32_t delayMs = 0;
uint32_t waitMs = 0;
if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) {
delayMs = TOUCH_SOUND_FIXED_DELAY_MS;
}
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
if (forceVolumeReeval && !newDevices.isEmpty()) {
applyStreamVolumes(mPrimaryOutput, newDevices.types(), delayMs, true);
}
waitMs = updateCallRouting(newDevices, delayMs);
}
// 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);
DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(),
delayMs);
if (forceVolumeReeval && !newDevices.isEmpty()) {
applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
}
}
}
for (const auto& activeDesc : mInputs.getActiveInputs()) {
// Skip for hotword recording as the input device switch
// is handled within sound trigger HAL
if (activeDesc->isSoundTrigger() &&
activeDesc->source() == AUDIO_SOURCE_HOTWORD) {
continue;
}
auto newDevice = getNewInputDevice(activeDesc);
// Force new input selection if the new device can not be reached via current input
if (activeDesc->mProfile->getSupportedDevices().contains(newDevice)) {
setInputDevice(activeDesc->mIoHandle, newDevice);
} else {
closeInput(activeDesc->mIoHandle);
}
}
}
status_t AudioPolicyManagerCustom::stopSource(const sp<SwAudioOutputDescriptor>& outputDesc,
const sp<TrackClientDescriptor>& client)
{
audio_stream_type_t stream = client->stream();
auto clientVolSrc = client->volumeSource();
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
handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
if (outputDesc->getActivityCount(clientVolSrc) > 0) {
if (outputDesc->getActivityCount(clientVolSrc) == 1) {
// Automatically disable the remote submix input when output is stopped on a
// re routing mix of type MIX_TYPE_RECORDERS
sp<AudioPolicyMix> policyMix = outputDesc->mPolicyMix.promote();
if (audio_is_remote_submix_device(outputDesc->devices().types()) &&
policyMix != NULL &&
policyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
policyMix->mDeviceAddress,
"remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
bool forceDeviceUpdate = false;
if (client->hasPreferredDevice(true)) {
checkStrategyRoute(client->strategy(), AUDIO_IO_HANDLE_NONE);
forceDeviceUpdate = true;
}
// decrement usage count of this stream on the output
outputDesc->setClientActive(client, false);
// store time at which the stream was stopped - see isStreamActive()
if (outputDesc->getActivityCount(clientVolSrc) == 0 || forceDeviceUpdate) {
outputDesc->setStopTime(client, systemTime());
DeviceVector prevDevices = outputDesc->devices();
DeviceVector newDevices = getNewOutputDevices(outputDesc, false /*fromCache*/);
// delay the device switch by twice the latency because stopOutput() is executed when
// the track stop() command is received and at that time the audio track buffer can
// still contain data that needs to be drained. The latency only covers the audio HAL
// and kernel buffers. Also the latency does not always include additional delay in the
// audio path (audio DSP, CODEC ...)
setOutputDevices(outputDesc, newDevices, false, outputDesc->latency()*2);
// force restoring the device selection on other active outputs if it differs from the
// one being selected for this output
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_io_handle_t curOutput = mOutputs.keyAt(i);
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc &&
desc->isActive() &&
outputDesc->sharesHwModuleWith(desc) &&
(newDevices != desc->devices())) {
DeviceVector dev = getNewOutputDevices(mOutputs.valueFor(curOutput), false /*fromCache*/);
bool force = prevDevices != dev;
uint32_t delayMs;
if (dev == prevDevices) {
delayMs = 0;
} else {
delayMs = outputDesc->latency()*2;
}
setOutputDevices(desc,
dev,
force,
delayMs);
/*audio policy: fix media volume after ringtone*/
// re-apply device specific volume if not done by setOutputDevice()
if (!force) {
applyStreamVolumes(desc, dev.types(), delayMs);
}
}
}
// update the outputs if stopping one with a stream that can affect notification routing
handleNotificationRoutingForStream(stream);
}
if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE &&
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), false, outputDesc);
}
if (followsSameRouting(client->attributes(), attributes_initializer(AUDIO_USAGE_MEDIA))) {
selectOutputForMusicEffects();
}
return NO_ERROR;
} else {
ALOGW("stopOutput() refcount is already 0");
return INVALID_OPERATION;
}
}
status_t AudioPolicyManagerCustom::startSource(const sp<SwAudioOutputDescriptor>& outputDesc,
const sp<TrackClientDescriptor>& client,
uint32_t *delayMs)
{
// cannot start playback of STREAM_TTS if any other output is being used
uint32_t beaconMuteLatency = 0;
audio_stream_type_t stream = client->stream();
auto clientVolSrc = client->volumeSource();
auto clientStrategy = client->strategy();
auto clientAttr = client->attributes();
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
ALOGW("startSource() invalid stream %d", stream);
return INVALID_OPERATION;
}
*delayMs = 0;
if (stream == AUDIO_STREAM_TTS) {
ALOGV("\t found BEACON stream");
if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive(
toVolumeSource(AUDIO_STREAM_TTS) /*sourceToIgnore*/)) {
return INVALID_OPERATION;
} else {
beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
}
} else {
// some playback other than beacon starts
beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
}
// force device change if the output is inactive and no audio patch is already present.
// check active before incrementing usage count
bool force = !outputDesc->isActive() &&
(outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE);
DeviceVector devices;
sp<AudioPolicyMix> policyMix = outputDesc->mPolicyMix.promote();
const char *address = NULL;
if (policyMix != NULL) {
audio_devices_t newDeviceType;
address = policyMix->mDeviceAddress.string();
if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
newDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
} else {
newDeviceType = policyMix->mDeviceType;
}
sp device = mAvailableOutputDevices.getDevice(newDeviceType, String8(address),
AUDIO_FORMAT_DEFAULT);
ALOG_ASSERT(device, "%s: no device found t=%u, a=%s", __func__, newDeviceType, address);
devices.add(device);
}
// requiresMuteCheck is false when we can bypass mute strategy.
// It covers a common case when there is no materially active audio
// and muting would result in unnecessary delay and dropped audio.
const uint32_t outputLatencyMs = outputDesc->latency();
bool requiresMuteCheck = outputDesc->isActive(outputLatencyMs * 2); // account for drain
// increment usage count for this stream on the requested output:
// NOTE that the usage count is the same for duplicated output and hardware output which is
// necessary for a correct control of hardware output routing by startOutput() and stopOutput()
outputDesc->setClientActive(client, true);
if (client->hasPreferredDevice(true)) {
devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
if (devices != outputDesc->devices()) {
checkStrategyRoute(clientStrategy, outputDesc->mIoHandle);
}
}
if (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_MEDIA))) {
selectOutputForMusicEffects();
}
if (outputDesc->getActivityCount(clientVolSrc) == 1 || !devices.isEmpty()) {
// starting an output being rerouted?
if (devices.isEmpty()) {
devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
}
bool shouldWait =
(followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM)) ||
followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_NOTIFICATION)) ||
(beaconMuteLatency > 0));
uint32_t waitMs = beaconMuteLatency;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc) {
// An output has a shared device if
// - managed by the same hw module
// - supports the currently selected device
const bool sharedDevice = outputDesc->sharesHwModuleWith(desc)
&& (!desc->filterSupportedDevices(devices).isEmpty());
// force a device change if any other output is:
// - managed by the same hw module
// - supports currently selected device
// - has a current device selection that differs from selected device.
// - has an active audio patch
// In this case, the audio HAL must receive the new device selection so that it can
// change the device currently selected by the other output.
if (sharedDevice &&
desc->devices() != devices &&
desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) {
force = true;
}
// wait for audio on other active outputs to be presented when starting
// a notification so that audio focus effect can propagate, or that a mute/unmute
// event occurred for beacon
const uint32_t latencyMs = desc->latency();
const bool isActive = desc->isActive(latencyMs * 2); // account for drain
if (shouldWait && isActive && (waitMs < latencyMs)) {
waitMs = latencyMs;
}
// Require mute check if another output is on a shared device
// and currently active to have proper drain and avoid pops.
// Note restoring AudioTracks onto this output needs to invoke
// a volume ramp if there is no mute.
requiresMuteCheck |= sharedDevice && isActive;
}
}
const uint32_t muteWaitMs =
setOutputDevices(outputDesc, devices, force, 0, NULL, requiresMuteCheck);
// apply volume rules for current stream and device if necessary
auto &curves = getVolumeCurves(client->attributes());
checkAndSetVolume(curves, client->volumeSource(),
getVolumeCurves(stream).getVolumeIndex(outputDesc->devices().types()),
outputDesc,
outputDesc->devices().types());
// update the outputs if starting an output with a stream that can affect notification
// routing
handleNotificationRoutingForStream(stream);
// force reevaluating accessibility routing when ringtone or alarm starts
if (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM))) {
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
}
if (waitMs > muteWaitMs) {
*delayMs = waitMs - muteWaitMs;
}
}
if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE &&
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), true, outputDesc);
}
// Automatically enable the remote submix input when output is started on a re routing mix
// of type MIX_TYPE_RECORDERS
if (audio_is_remote_submix_device(devices.types()) && policyMix != NULL &&
policyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address,
"remote-submix",
AUDIO_FORMAT_DEFAULT);
}
return NO_ERROR;
}
status_t AudioPolicyManagerCustom::checkAndSetVolume(IVolumeCurves &curves,
VolumeSource volumeSource,
int index,
const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device,
int delayMs,
bool force)
{
// do not change actual stream volume if the stream is muted
if (outputDesc->isMuted(volumeSource)) {
ALOGVV("%s: volume source %d muted count %d active=%d", __func__, volumeSource,
outputDesc->getMuteCount(volumeSource), outputDesc->isActive(volumeSource));
return NO_ERROR;
}
VolumeSource callVolSrc = toVolumeSource(AUDIO_STREAM_VOICE_CALL);
VolumeSource btScoVolSrc = toVolumeSource(AUDIO_STREAM_BLUETOOTH_SCO);
bool isVoiceVolSrc = callVolSrc == volumeSource;
bool isBtScoVolSrc = btScoVolSrc == volumeSource;
audio_policy_forced_cfg_t forceUseForComm =
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION);
// do not change in call volume if bluetooth is connected and vice versa
if ((callVolSrc != btScoVolSrc) &&
((isVoiceVolSrc && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
(isBtScoVolSrc && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO))) {
ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
volumeSource, forceUseForComm);
return INVALID_OPERATION;
}
if (device == AUDIO_DEVICE_NONE) {
device = outputDesc->devices().types();
}
float volumeDb = computeVolume(curves, volumeSource, index, device);
if (outputDesc->isFixedVolume(device)) {
volumeDb = 0.0f;
}
outputDesc->setVolume(volumeDb, volumeSource, curves.getStreamTypes(), device, delayMs, force);
if (isVoiceVolSrc || isBtScoVolSrc) {
float voiceVolume;
// Force voice volume to max for bluetooth SCO as volume is managed by the headset
if (isVoiceVolSrc) {
voiceVolume = (float)index/(float)curves.getVolumeIndexMax();
} else {
voiceVolume = 1.0;
}
if (voiceVolume != mLastVoiceVolume) {
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
mLastVoiceVolume = voiceVolume;
}
}
return NO_ERROR;
}
bool static tryForDirectPCM(audio_output_flags_t flags)
{
bool trackDirectPCM = false; // Output request for track created by other apps
if (flags == AUDIO_OUTPUT_FLAG_NONE) {
if (AudioPolicyManagerCustom::mApmConfigs != NULL)
trackDirectPCM = AudioPolicyManagerCustom::mApmConfigs->isAudioTrackOffloadEnabled();
}
return trackDirectPCM;
}
status_t AudioPolicyManagerCustom::getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
uid_t uid,
const audio_config_t *config,
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs)
{
audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER;
audio_config_t tConfig;
uint32_t bitWidth = (audio_bytes_per_sample(config->format) * 8);
memcpy(&tConfig, config, sizeof(audio_config_t));
if ((*flags == AUDIO_OUTPUT_FLAG_DIRECT || tryForDirectPCM(*flags)) &&
(!memcmp(&config->offload_info, &tOffloadInfo, sizeof(audio_offload_info_t)))) {
tConfig.offload_info.sample_rate = config->sample_rate;
tConfig.offload_info.channel_mask = config->channel_mask;
tConfig.offload_info.format = config->format;
tConfig.offload_info.stream_type = *stream;
tConfig.offload_info.bit_width = bitWidth;
if (attr != NULL) {
ALOGV("found attribute .. setting usage %d ", attr->usage);
tConfig.offload_info.usage = attr->usage;
} else {
ALOGI("%s:: attribute is NULL .. no usage set", __func__);
}
}
return AudioPolicyManager::getOutputForAttr(attr, output, session, stream,
(uid_t)uid, &tConfig,
flags,
(audio_port_handle_t*)selectedDeviceId,
portId,
secondaryOutputs);
}
audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevices(
const DeviceVector &devices,
audio_session_t session,
audio_stream_type_t stream,
const audio_config_t *config,
audio_output_flags_t *flags,
bool forceMutingHaptic)
{
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status;
// Discard haptic channel mask when forcing muting haptic channels.
audio_channel_mask_t channelMask = forceMutingHaptic
? (config->channel_mask & ~AUDIO_CHANNEL_HAPTIC_ALL) : config->channel_mask;
if (stream < AUDIO_STREAM_MIN || stream >= AUDIO_STREAM_CNT) {
ALOGE("%s: invalid stream %d", __func__, stream);
return AUDIO_IO_HANDLE_NONE;
}
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;
}
if (mApmConfigs->isCompressVOIPEnabled()) {
if (stream == AUDIO_STREAM_VOICE_CALL &&
audio_is_linear_pcm(config->format)) {
// let voice stream to go with primary output by default
// in case direct voip is bypassed
bool use_primary_out = true;
if ((channelMask == 1) &&
(config->sample_rate == 8000 || config->sample_rate == 16000 ||
config->sample_rate == 32000 || config->sample_rate == 48000)) {
// 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 voipOutCount = 1, voipSampleRate = 1;
String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
String8("voip_out_stream_count"));
AudioParameter 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 ((voipOutCount == 0) &&
((voipSampleRate == 0) || (voipSampleRate == config->sample_rate))) {
if (mApmConfigs->useVoicePathForPCMVOIP()
&& (config->format == AUDIO_FORMAT_PCM_16_BIT)) {
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
AUDIO_OUTPUT_FLAG_DIRECT);
ALOGD("Set VoIP and Direct output flags for PCM format");
use_primary_out = false;
}
}
}
if (use_primary_out) {
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY);
}
}
} else {
if (stream == AUDIO_STREAM_VOICE_CALL &&
audio_is_linear_pcm(config->format) &&
(config->channel_mask == 1) &&
(config->sample_rate == 8000 || config->sample_rate == 16000 ||
config->sample_rate == 32000 || config->sample_rate == 48000)) {
//check if VoIP output is not opened already
bool voip_pcm_already_in_use = false;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc->mFlags == (AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT)) {
//close voip output if currently open by the same client with different device
if (desc->mDirectClientSession == session &&
desc->devices() != devices) {
closeOutput(desc->mIoHandle);
} else {
voip_pcm_already_in_use = true;
ALOGD("VoIP PCM already in use");
}
break;
}
}
if (!voip_pcm_already_in_use) {
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
AUDIO_OUTPUT_FLAG_DIRECT);
ALOGV("Set VoIP and Direct output flags for PCM format");
}
}
} /* compress_voip_enabled */
//IF VOIP is going to be started at the same time as when
//vr is enabled, get VOIP to fallback to low latency
String8 vr_value;
String8 value_Str;
bool is_vr_mode_on = false;
AudioParameter ret;
value_Str = mpClientInterface->getParameters((audio_io_handle_t)0,
String8("vr_audio_mode_on"));
ret = AudioParameter(value_Str);
if (ret.get(String8("vr_audio_mode_on"), vr_value) == NO_ERROR) {
is_vr_mode_on = vr_value.contains("true");
ALOGI("VR mode is %d, switch to primary output if request is for fast|raw",
is_vr_mode_on);
}
if (is_vr_mode_on) {
//check the flags being requested for, and clear FAST|RAW
*flags = (audio_output_flags_t)(*flags &
(~(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW)));
}
if (mApmConfigs->isVoiceConcEnabled()) {
bool prop_play_enabled = false, prop_voip_enabled = false;
prop_play_enabled = mApmConfigs->isVoicePlayConcDisabled();
prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled();
bool isDeepBufferFallBackNeeded =
((AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & *flags);
bool isFastFallBackNeeded =
((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & *flags);
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 (isFastFallBackNeeded &&
(AUDIO_OUTPUT_FLAG_FAST == mFallBackflag)) {
ALOGD("voice_conc:IN call mode adding ULL flags .. flags: %x ", *flags );
*flags = AUDIO_OUTPUT_FLAG_FAST;
} else if (isDeepBufferFallBackNeeded &&
(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;
}
}
}
}
if (mApmConfigs->isRecPlayConcEnabled()) {
bool prop_rec_play_enabled = mApmConfigs->isRecPlayConcDisabled();
DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
if ((prop_rec_play_enabled) &&
((true == mIsInputRequestOnProgress) ||
(mInputs.activeInputsCountOnDevices(primaryInputDevices) > 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;
}
}
/*
* 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.
*/
if (mApmConfigs->isAFEProxyEnabled()) {
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;
}
}
// 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);
}
// Do internal direct magic here
bool offload_disabled = mApmConfigs->isAudioOffloadDisabled();
if ((*flags == AUDIO_OUTPUT_FLAG_NONE) &&
(stream == AUDIO_STREAM_MUSIC) &&
( !offload_disabled) &&
((config->offload_info.usage == AUDIO_USAGE_MEDIA) ||
(config->offload_info.usage == AUDIO_USAGE_GAME))) {
audio_output_flags_t old_flags = *flags;
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT);
ALOGD("Force Direct Flag .. old flags(0x%x)", old_flags);
} else if (*flags == AUDIO_OUTPUT_FLAG_DIRECT &&
(offload_disabled || stream != AUDIO_STREAM_MUSIC)) {
ALOGD("Offloading is disabled or Stream is not music --> Force Remove Direct Flag");
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE);
}
// check if direct output for pcm/track offload or compress offload already exist
bool direct_pcm_already_in_use = false;
bool compress_offload_already_in_use = false;
if (*flags & AUDIO_OUTPUT_FLAG_DIRECT) {
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc->mFlags == AUDIO_OUTPUT_FLAG_DIRECT) {
direct_pcm_already_in_use = true;
ALOGD("Direct PCM already in use");
}
if (desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
compress_offload_already_in_use = true;
ALOGD("Compress Offload already in use");
}
}
// prevent direct pcm for non-music stream blindly if direct pcm already in use
// for other music stream concurrency is handled after checking direct ouput usage
// and checking client
if (direct_pcm_already_in_use == true && stream != AUDIO_STREAM_MUSIC) {
ALOGD("disabling offload for non music stream as direct pcm is already in use");
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE);
}
}
bool forced_deep = false;
// only allow deep buffering for music stream type
if (stream != AUDIO_STREAM_MUSIC) {
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
} else if (/* stream == AUDIO_STREAM_MUSIC && */
(*flags == AUDIO_OUTPUT_FLAG_NONE || *flags == AUDIO_OUTPUT_FLAG_DIRECT ||
(*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) &&
mApmConfigs->isAudioDeepbufferMediaEnabled() && !isInCall()) {
forced_deep = true;
}
if (stream == AUDIO_STREAM_TTS) {
*flags = AUDIO_OUTPUT_FLAG_TTS;
} else if (devices.types() == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
stream == AUDIO_STREAM_MUSIC &&
audio_is_linear_pcm(config->format) &&
isInCall()) {
*flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
}
sp<IOProfile> profile;
// skip direct output selection if the request can obviously be attached to a mixed output
// and not explicitly requested
if (((*flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX &&
audio_channel_count_from_out_mask(channelMask) <= 2) {
goto non_direct_output;
}
// Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled.
// This prevents creating an offloaded track and tearing it down immediately after start
// when audioflinger detects there is an active non offloadable effect.
// FIXME: We should check the audio session here but we do not have it in this context.
// This may prevent offloading in rare situations where effects are left active by apps
// in the background.
//
// Supplementary annotation:
// For sake of track offload introduced, we need a rollback for both compress offload
// and track offload use cases.
if ((*flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_DIRECT)) &&
(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {
ALOGD("non offloadable effect is enabled, try with non direct output");
goto non_direct_output;
}
profile = getProfileForOutput(devices,
config->sample_rate,
config->format,
channelMask,
(audio_output_flags_t)*flags,
true /* directOnly */);
if (profile != 0) {
if (!(*flags & AUDIO_OUTPUT_FLAG_DIRECT) &&
(profile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
ALOGI("got Direct without requesting ... reject ");
profile = NULL;
goto non_direct_output;
}
sp<SwAudioOutputDescriptor> outputDesc = NULL;
// if multiple concurrent offload decode is supported
// do no check for reuse and also don't close previous output if its offload
// previous output will be closed during track destruction
if (!mApmConfigs->isAudioMultipleOffloadEnable() &&
((*flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)) {
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 by the same client
// and configured with same parameters
if ((config->sample_rate == desc->mSamplingRate) &&
audio_formats_match(config->format, desc->mFormat) &&
(channelMask == desc->mChannelMask) &&
(session == desc->mDirectClientSession)) {
desc->mDirectOpenCount++;
ALOGV("getOutputForDevice() reusing direct output %d for session %d",
mOutputs.keyAt(i), session);
return mOutputs.keyAt(i);
}
}
}
if (outputDesc != NULL) {
if ((((*flags == AUDIO_OUTPUT_FLAG_DIRECT) && direct_pcm_already_in_use) ||
((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
compress_offload_already_in_use)) &&
session != outputDesc->mDirectClientSession) {
ALOGV("getOutput() do not reuse direct pcm output because current client (%d) "
"is not the same as requesting client (%d) for different output conf",
outputDesc->mDirectClientSession, session);
goto non_direct_output;
}
closeOutput(outputDesc->mIoHandle);
}
}
if (!profile->canOpenNewIo()) {
goto non_direct_output;
}
outputDesc =
new SwAudioOutputDescriptor(profile, mpClientInterface);
DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(devices.types());
String8 address = outputDevices.size() > 0 ? outputDevices.itemAt(0)->address()
: String8("");
status = outputDesc->open(config, devices, stream, *flags, &output);
// only accept an output with the requested parameters
if (status != NO_ERROR ||
(config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
(config->format != AUDIO_FORMAT_DEFAULT &&
!audio_formats_match(config->format, outputDesc->mFormat)) ||
(channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
ALOGV("getOutputForDevice() failed opening direct output: output %d sample rate %d %d,"
"format %d %d, channel mask %04x %04x", output, config->sample_rate,
outputDesc->mSamplingRate, config->format, outputDesc->mFormat,
channelMask, outputDesc->mChannelMask);
//Only close o/p descriptor if successfully opened
if (status == NO_ERROR) {
outputDesc->close();
}
// fall back to mixer output if possible when the direct output could not be open
if (audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX) {
goto non_direct_output;
}
return AUDIO_IO_HANDLE_NONE;
}
outputDesc->mDirectOpenCount = 1;
outputDesc->mDirectClientSession = session;
addOutput(output, outputDesc);
mPreviousOutputs = mOutputs;
ALOGV("getOutputForDevice() returns new direct output %d", output);
mpClientInterface->onAudioPortListUpdate();
return output;
}
non_direct_output:
// A request for HW A/V sync cannot fallback to a mixed output because time
// stamps are embedded in audio data
if ((*flags & (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)) != 0) {
return AUDIO_IO_HANDLE_NONE;
}
// ignoring channel mask due to downmix capability in mixer
// open a non direct output
// for non direct outputs, only PCM is supported
if (audio_is_linear_pcm(config->format)) {
// get which output is suitable for the specified stream. The actual
// routing change will happen when startOutput() will be called
SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
// at this stage we should ignore the DIRECT flag as no direct output could be found earlier
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
if (forced_deep) {
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
ALOGI("setting force DEEP buffer now ");
} else if (*flags == AUDIO_OUTPUT_FLAG_NONE) {
// no deep buffer playback is requested hence fallback to primary
*flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_PRIMARY);
ALOGI("FLAG None hence request for a primary output");
}
output = selectOutput(outputs, *flags, config->format);
}
ALOGW_IF((output == 0), "getOutputForDevice() could not find output for stream %d, "
"sampling rate %d, format %#x, channels %#x, flags %#x",
stream, config->sample_rate, config->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_unique_id_t riid,
audio_session_t session,
uid_t uid,
const audio_config_base_t *config,
audio_input_flags_t flags,
audio_port_handle_t *selectedDeviceId,
input_type_t *inputType,
audio_port_handle_t *portId)
{
audio_source_t inputSource;
inputSource = attr->source;
if (mApmConfigs->isVoiceConcEnabled()) {
bool prop_rec_enabled = false, prop_voip_enabled = false;
prop_rec_enabled = mApmConfigs->isVoiceRecConcDisabled();
prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled();
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;
}
}
}
}
return AudioPolicyManager::getInputForAttr(attr,
input,
riid,
session,
uid,
config,
flags,
selectedDeviceId,
inputType,
portId);
}
uint32_t AudioPolicyManagerCustom::activeNonSoundTriggerInputsCountOnDevices(audio_devices_t devices) const
{
uint32_t count = 0;
for (size_t i = 0; i < mInputs.size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
if (inputDescriptor->isActive() && !inputDescriptor->isSoundTrigger() &&
((devices == AUDIO_DEVICE_IN_DEFAULT) ||
((inputDescriptor->getDeviceType() & devices & ~AUDIO_DEVICE_BIT_IN) != 0))) {
count++;
}
}
return count;
}
status_t AudioPolicyManagerCustom::startInput(audio_port_handle_t portId)
{
ALOGV("%s portId %d", __FUNCTION__, portId);
sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
if (inputDesc == 0) {
ALOGW("%s no input for client %d", __FUNCTION__, portId);
return BAD_VALUE;
}
audio_io_handle_t input = inputDesc->mIoHandle;
sp<RecordClientDescriptor> client = inputDesc->getClient(portId);
if (client == NULL) {
ALOGW("%s invalid client desc for %d", __FUNCTION__, portId);
return BAD_VALUE;
}
if (client->active()) {
ALOGW("%s input %d client %d already started", __FUNCTION__, input, client->portId());
return INVALID_OPERATION;
}
audio_session_t session = client->session();
ALOGV("%s input:%d, session:%d)", __FUNCTION__, input, session);
// FIXME: disable concurrent capture until UI is ready
#if 0
if (!isConcurentCaptureAllowed(inputDesc, audioSession)) {
ALOGW("startInput(%d) failed: other input already started", input);
return INVALID_OPERATION;
}
if (isInCall()) {
*concurrency |= API_INPUT_CONCURRENCY_CALL;
}
if (mInputs.activeInputsCountOnDevices() != 0) {
*concurrency |= API_INPUT_CONCURRENCY_CAPTURE;
}
#endif
if (mApmConfigs->isRecPlayConcEnabled()) {
mIsInputRequestOnProgress = true;
DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
if (mApmConfigs->isRecPlayConcDisabled() && (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 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 < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
// Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder)
if (i != AUDIO_STREAM_ENFORCED_AUDIBLE) {
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->getFlags()
& AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
// close compress sessions
ALOGD("calling closeOutput on record conc for COMPRESS output");
closeOutput(mOutputs.keyAt(i));
}
}
}
}
status_t status = inputDesc->start();
if (status != NO_ERROR) {
return status;
}
// increment activity count before calling getNewInputDevice() below as only active sessions
// are considered for device selection
inputDesc->setClientActive(client, true);
// indicate active capture to sound trigger service if starting capture from a mic on
// primary HW module
sp<DeviceDescriptor> device = getNewInputDevice(inputDesc);
setInputDevice(input, device, true /* force */);
if (inputDesc->activeCount() == 1) {
sp<AudioPolicyMix> policyMix = inputDesc->mPolicyMix.promote();
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((policyMix != NULL)
&& ((policyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
mpClientInterface->onDynamicPolicyMixStateUpdate(policyMix->mDeviceAddress,
MIX_STATE_MIXING);
}
DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
if ((primaryInputDevices.contains(device) && (device->type() & ~AUDIO_DEVICE_BIT_IN)) != 0) {
if (mApmConfigs->isVAConcEnabled()) {
if (activeNonSoundTriggerInputsCountOnDevices(primaryInputDevices.types()) == 1)
SoundTrigger::setCaptureState(true);
} else if (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1)
SoundTrigger::setCaptureState(true);
}
// automatically enable the remote submix output when input is started if not
// used by a policy mix of type MIX_TYPE_RECORDERS
// For remote submix (a virtual device), we open only one input per capture request.
if (audio_is_remote_submix_device(inputDesc->getDeviceType())) {
String8 address = String8("");
if (policyMix == NULL) {
address = String8("0");
} else if (policyMix->mMixType == MIX_TYPE_PLAYERS) {
address = policyMix->mDeviceAddress;
}
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address, "remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
}
ALOGV("%s input %d source = %d exit", __FUNCTION__, input, client->source());
if (mApmConfigs->isRecPlayConcEnabled())
mIsInputRequestOnProgress = false;
return NO_ERROR;
}
status_t AudioPolicyManagerCustom::stopInput(audio_port_handle_t portId)
{
status_t status;
status = AudioPolicyManager::stopInput(portId);
sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
if (inputDesc == 0) {
ALOGW("stopInput() no input for client %d", portId);
return BAD_VALUE;
}
sp<RecordClientDescriptor> client = inputDesc->getClient(portId);
audio_io_handle_t input = inputDesc->mIoHandle;
ALOGV("stopInput() input %d", input);
DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
if (mApmConfigs->isVAConcEnabled()) {
sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
if ((primaryInputDevices.contains(inputDesc->getDevice()) &&
activeNonSoundTriggerInputsCountOnDevices(primaryInputDevices.types())) == 0) {
SoundTrigger::setCaptureState(false);
}
}
if (mApmConfigs->isRecPlayConcEnabled()) {
if (mApmConfigs->isRecPlayConcDisabled() &&
(mInputs.activeInputsCountOnDevices(primaryInputDevices) == 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);
}
}
}
}
return status;
}
AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
: AudioPolicyManager(clientInterface),
mFallBackflag(AUDIO_OUTPUT_FLAG_NONE),
mHdmiAudioDisabled(false),
mHdmiAudioEvent(false),
mPrevPhoneState(0),
mIsInputRequestOnProgress(false)
{
if (mApmConfigs->useXMLAudioPolicyConf())
ALOGD("USE_XML_AUDIO_POLICY_CONF is TRUE");
else
ALOGD("USE_XML_AUDIO_POLICY_CONF is FALSE");
if (mApmConfigs->isVoiceConcEnabled())
mFallBackflag = getFallBackPath();
}
}