audio: Remove policy hal directory
Remove policy hal directory and move it to the new
qcom-opensource audio project.
Change-Id: I0c1d1845b90e4938194868f5ab317694d5722f0b
diff --git a/Android.mk b/Android.mk
index 2f4054f..429c4cb 100644
--- a/Android.mk
+++ b/Android.mk
@@ -12,7 +12,6 @@
endif
include $(MY_LOCAL_PATH)/voice_processing/Android.mk
include $(MY_LOCAL_PATH)/mm-audio/Android.mk
-include $(MY_LOCAL_PATH)/policy_hal/Android.mk
include $(MY_LOCAL_PATH)/visualizer/Android.mk
include $(MY_LOCAL_PATH)/post_proc/Android.mk
include $(MY_LOCAL_PATH)/qahw_api/Android.mk
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
deleted file mode 100644
index 5b7f6f1..0000000
--- a/policy_hal/Android.mk
+++ /dev/null
@@ -1,96 +0,0 @@
-# This file was modified by Dolby Laboratories, Inc. The portions of the
-# code that are surrounded by "DOLBY..." are copyrighted and
-# licensed separately, as follows:
-#
-# (C) 2016 Dolby Laboratories, Inc.
-# All rights reserved.
-#
-# This program is protected under international and U.S. Copyright laws as
-# an unpublished work. This program is confidential and proprietary to the
-# copyright owners. Reproduction or disclosure, in whole or in part, or the
-# production of derivative works therefrom without the express permission of
-# the copyright owners is prohibited.
-#
-ifneq ($(USE_LEGACY_AUDIO_POLICY), 1)
-ifeq ($(USE_CUSTOM_AUDIO_POLICY), 1)
-LOCAL_PATH := $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := AudioPolicyManager.cpp
-
-LOCAL_C_INCLUDES := $(TOPDIR)frameworks/av/services \
- $(TOPDIR)frameworks/av/services/audioflinger \
- $(call include-path-for, audio-effects) \
- $(call include-path-for, audio-utils) \
- $(TOPDIR)frameworks/av/services/audiopolicy/common/include \
- $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface \
- $(TOPDIR)frameworks/av/services/audiopolicy \
- $(TOPDIR)frameworks/av/services/audiopolicy/common/managerdefinitions/include \
- $(call include-path-for, avextension) \
- $(TOPDIR)system/core/base/include
-
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libutils \
- liblog \
- libsoundtrigger \
- libaudiopolicymanagerdefault \
- libserviceutility
-
-LOCAL_STATIC_LIBRARIES := \
- libmedia_helper \
-
-LOCAL_CFLAGS += -Wall -Werror
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_VOICE_CONCURRENCY)),true)
-LOCAL_CFLAGS += -DVOICE_CONCURRENCY
-endif
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_RECORD_PLAY_CONCURRENCY)),true)
-LOCAL_CFLAGS += -DRECORD_PLAY_CONCURRENCY
-endif
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_PCM_OFFLOAD)),true)
- LOCAL_CFLAGS += -DPCM_OFFLOAD_ENABLED
-endif
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_PCM_OFFLOAD_24)),true)
- LOCAL_CFLAGS += -DPCM_OFFLOAD_ENABLED_24
-endif
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_EXTN_FORMATS)),true)
- LOCAL_CFLAGS += -DAUDIO_EXTN_FORMATS_ENABLED
-endif
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AAC_ADTS_OFFLOAD)),true)
- LOCAL_CFLAGS += -DAAC_ADTS_OFFLOAD_ENABLED
-endif
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_SPK)),true)
- LOCAL_CFLAGS += -DAUDIO_EXTN_HDMI_SPK_ENABLED
-endif
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_PROXY_DEVICE)),true)
- LOCAL_CFLAGS += -DAUDIO_EXTN_AFE_PROXY_ENABLED
-endif
-
-ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FM_POWER_OPT)),true)
-LOCAL_CFLAGS += -DFM_POWER_OPT
-endif
-# DOLBY_START
-ifeq ($(strip $(DOLBY_ENABLE)),true)
-LOCAL_CFLAGS += $(dolby_cflags)
-endif
-# DOLBY_END
-
-ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-LOCAL_CFLAGS += -DUSE_XML_AUDIO_POLICY_CONF
-endif
-
-LOCAL_MODULE := libaudiopolicymanager
-
-include $(BUILD_SHARED_LIBRARY)
-
-endif
-endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
deleted file mode 100644
index 72bd644..0000000
--- a/policy_hal/AudioPolicyManager.cpp
+++ /dev/null
@@ -1,2246 +0,0 @@
-/*
- * Copyright (c) 2013-2017 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
-
-// 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>
-#ifdef DOLBY_ENABLE
-#include "DolbyAudioPolicy_impl.h"
-#endif // DOLBY_END
-
-#ifndef AUDIO_OUTPUT_FLAG_VOIP_RX
-#define AUDIO_OUTPUT_FLAG_VOIP_RX 0x800
-#endif
-
-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
-#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*/
-
-void AudioPolicyManagerCustom::moveGlobalEffect()
-{
- audio_io_handle_t dstOutput = getOutputForEffect();
- if (hasPrimaryOutput() && dstOutput != mPrimaryOutput->mIoHandle) {
-#ifdef DOLBY_ENABLE
- status_t status =
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
- mPrimaryOutput->mIoHandle,
- 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__,
- mPrimaryOutput->mIoHandle, dstOutput);
- }
-#else // DOLBY_END
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
- mPrimaryOutput->mIoHandle, dstOutput);
-#endif
- }
-}
-
-// ----------------------------------------------------------------------------
-// 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)
-{
- ALOGD("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;
- }
-
- // 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, devDesc->mAddress);
-
- if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
- mAvailableOutputDevices.remove(devDesc);
-
- broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- devDesc->mAddress);
- 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());
-
- } 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
- broadcastDeviceConnectionState(device, state, devDesc->mAddress);
-
- // 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();
- }
-
-#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 = AUDIO_DEVICE_NONE;
- if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
- mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, 1);
- newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false)|AUDIO_DEVICE_OUT_FM);
- mFMIsActive = true;
- mPrimaryOutput->mDevice = newDevice & ~AUDIO_DEVICE_OUT_FM;
- } else {
- newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false));
- mFMIsActive = false;
- 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 */
-
- 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);
- }
-
- 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);
- }
- }
-
- if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
- cleanUpForDevice(devDesc);
- }
-
- 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;
- }
-
- // 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, devDesc->mAddress);
-
- if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) {
- broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- devDesc->mAddress);
- return INVALID_OPERATION;
- }
-
- index = mAvailableInputDevices.add(devDesc);
- if (index >= 0) {
- mAvailableInputDevices[index]->attach(module);
- } else {
- return NO_MEMORY;
- }
-
- // 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
- broadcastDeviceConnectionState(device, state, devDesc->mAddress);
-
- 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();
- /*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()) {
- audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
- updateCallRouting(newDevice);
- }
-
- if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
- cleanUpForDevice(devDesc);
- }
-
- mpClientInterface->onAudioPortListUpdate();
- return NO_ERROR;
- } // end if is input device
-
- ALOGW("setDeviceConnectionState() invalid device: %x", device);
- return BAD_VALUE;
-}
-
-bool AudioPolicyManagerCustom::isInvalidationOfMusicStreamNeeded(routing_strategy strategy)
-{
- if (strategy == STRATEGY_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::checkOutputForStrategy(routing_strategy strategy)
-{
- audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
- audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mOutputs);
- SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
-
- // also take into account external policy-related changes: add all outputs which are
- // associated with policies in the "before" and "after" output vectors
- ALOGV("checkOutputForStrategy(): policy related outputs");
- 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);
- ALOGV(" 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);
- ALOGV(" new outputs: adding %d", desc->mIoHandle);
- }
- }
-
- if (!vectorsEqual(srcOutputs,dstOutputs) && isInvalidationOfMusicStreamNeeded(strategy)) {
- AudioPolicyManager::checkOutputForStrategy(strategy);
- }
-}
-
-// 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);
-
- if (mMasterMono) {
- return false; // no offloading if mono is set.
- }
-
-#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
- if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
- isInCall() && (offloadInfo.format == AUDIO_FORMAT_DSD)) {
- ALOGD("blocking DSD compress offload on call mode");
- return false;
- }
-#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.activeInputsCountOnDevices() > 0))) {
- ALOGD("copl: blocking compress offload for record concurrency");
- return false;
- }
-#endif
- // 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 = property_get_bool("audio.offload.disable", false);
- 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;
- }
-
-#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)) ||
- ((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;
- }
-#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;
- }
-
- const bool allowOffloadStreamingWithVideo = property_get_bool("av.streaming.offload.enable",
- false /*default value*/);
- if (offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) {
- ALOGW("offload disabled by av.streaming.offload.enable %d",allowOffloadStreamingWithVideo);
- return false;
- }
-
- //If duration is less than minimum value defined in property, return false
- char propValue[PROPERTY_VALUE_MAX];
- 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/ 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_FLAC) ||
- ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))
- return false;
-
-#ifdef AUDIO_EXTN_FORMATS_ENABLED
- 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;
-#endif
- }
-
- // 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 = 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);
-}
-
-void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)
-{
- ALOGD("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_FOR_POLICY_CNT; stream++) {
- handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput);
- }
- }
-
- // 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
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- updateDevicesAndOutputs();
-
- sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
-#ifdef VOICE_CONCURRENCY
- 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);
- }
-
- 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->inputSource()) {
- 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->inputSource());
- break;
-
- case AUDIO_SOURCE_VOICE_COMMUNICATION:
- if(prop_voip_enabled) {
- ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",activeInput->inputSource());
- AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
- audio_session_t activeSession = activeSessions.keyAt(0);
- stopInput(activeInput->mIoHandle, activeSession);
- releaseInput(activeInput->mIoHandle, activeSession);
- }
- break;
-
- default:
- ALOGD("voice_conc:CLOSING input on call setup for inputSource: %d",activeInput->inputSource());
- AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
- audio_session_t activeSession = activeSessions.keyAt(0);
- stopInput(activeInput->mIoHandle, activeSession);
- releaseInput(activeInput->mIoHandle, activeSession);
- 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->inputSource()) {
- ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeInput->inputSource());
- AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
- audio_session_t activeSession = activeSessions.keyAt(0);
- stopInput(activeInput->mIoHandle, activeSession);
- releaseInput(activeInput->mIoHandle, activeSession);
- }
- }
- }
- }
- 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 effects where present on any of the above closed outputs,
- // audioflinger moved them to the primary output by default
- // move them back to the appropriate output.
- moveGlobalEffect();
- }
-
- 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);
- }
- }
- }
-
-#endif
-
- 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 (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
- (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);
- }
-
- }
-
-#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->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
- ALOGD("calling closeOutput on call mode for COMPRESS output");
- closeOutput(mOutputs.keyAt(i));
- }
- }
- // If effects where present on any of the above closed outputs,
- // audioflinger moved them to the primary output by default
- // move them back to the appropriate output.
- moveGlobalEffect();
- } 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*/));
- }
- }
-
- 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_FOR_POLICY_CNT; stream++) {
- handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput);
- }
- }
-
- // 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
- 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)
-{
- ALOGD("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();
-
- /*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()) {
- audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
- waitMs = updateCallRouting(newDevice, 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);
- audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
- if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
- waitMs = setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE),
- delayMs);
- }
- if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
- applyStreamVolumes(outputDesc, newDevice, waitMs, true);
- }
- }
-
- Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
- for (size_t i = 0; i < activeInputs.size(); i++) {
- sp<AudioInputDescriptor> activeDesc = activeInputs[i];
- audio_devices_t newDevice = getNewInputDevice(activeDesc);
- // Force new input selection if the new device can not be reached via current input
- if (activeDesc->mProfile->getSupportedDevices().types() &
- (newDevice & ~AUDIO_DEVICE_BIT_IN)) {
- setInputDevice(activeDesc->mIoHandle, newDevice);
- } else {
- closeInput(activeDesc->mIoHandle);
- }
- }
-}
-
-status_t AudioPolicyManagerCustom::stopSource(const sp<AudioOutputDescriptor>& outputDesc,
- 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
- handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
-
- // handle special case for sonification while in call
- if (isInCall()) {
- if (outputDesc->isDuplicated()) {
- handleIncallSonification(stream, false, false, outputDesc->subOutput1()->mIoHandle);
- handleIncallSonification(stream, false, false, outputDesc->subOutput2()->mIoHandle);
- }
- 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 prevDevice = outputDesc->device();
- 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*/);
- bool force = desc->device() != dev;
- uint32_t delayMs;
- if (dev == prevDevice) {
- delayMs = 0;
- } else {
- delayMs = outputDesc->latency()*2;
- }
- setOutputDevice(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, delayMs);
- }
- }
- }
- // 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(const sp<AudioOutputDescriptor>& outputDesc,
- audio_stream_type_t stream,
- audio_devices_t device,
- const char *address,
- 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;
- }
-
- *delayMs = 0;
- if (stream == AUDIO_STREAM_TTS) {
- ALOGV("\t found BEACON stream");
- if (!mTtsOutputAvailable && 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);
- }
-
- // 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);
-
- // 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;
- 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
- // - has a current device selection that differs from selected device.
- // - supports currently 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 active output.
- if (outputDesc->sharesHwModuleWith(desc) &&
- desc->device() != device &&
- desc->supportedDevices() & device &&
- 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
- uint32_t latency = desc->latency();
- if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
- waitMs = latency;
- }
- }
- }
- uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force, 0, NULL, address);
-
- // 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,
- mVolumeCurves->getVolumeIndex(stream, 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);
- }
- if (waitMs > muteWaitMs) {
- *delayMs = waitMs - muteWaitMs;
- }
-
- } 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;
- }
-}
-
-status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream,
- int index,
- const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t device,
- int delayMs,
- bool force)
-{
- 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)mVolumeCurves->getVolumeIndexMax(stream);
- } else {
- voiceVolume = 1.0;
- }
-
- if (voiceVolume != mLastVoiceVolume) {
- mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
- mLastVoiceVolume = voiceVolume;
- }
-#ifdef FM_POWER_OPT
- } else if (stream == AUDIO_STREAM_MUSIC && hasPrimaryOutput() &&
- outputDesc == mPrimaryOutput && mFMIsActive) {
- /* Avoid unnecessary set_parameter calls as it puts the primary
- outputs FastMixer in HOT_IDLE leading to breaks in audio */
- if (volumeDb != mPrevFMVolumeDb) {
- mPrevFMVolumeDb = volumeDb;
- AudioParameter param = AudioParameter();
- param.addFloat(String8("fm_volume"), Volume::DbToAmpl(volumeDb));
- //Double delayMs to avoid sound burst while device switch.
- mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString(), delayMs*2);
- }
-#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;
-}
-
-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) {
- trackDirectPCM = property_get_bool("audio.offload.track.enable", true);
- }
- 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)
-{
- 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);
-}
-
-audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice(
- audio_devices_t device,
- audio_session_t session,
- audio_stream_type_t stream,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo)
-{
- audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- 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;
- }
-
- if ((stream == AUDIO_STREAM_VOICE_CALL) &&
- (channelMask == 1) &&
- (samplingRate == 8000 || samplingRate == 16000 ||
- samplingRate == 32000 || samplingRate == 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 mode = 0, voipOutCount = 1, voipSampleRate = 1;
- bool is_vr_mode_on = false;
- 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)(AUDIO_OUTPUT_FLAG_VOIP_RX |
- AUDIO_OUTPUT_FLAG_DIRECT);
- ALOGD("Set VoIP and Direct output flags for PCM format");
- }
- }
- }
- //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;
- valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
- String8("vr_audio_mode_on"));
- result = AudioParameter(valueStr);
- if (result.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)));
-
- }
-
- }
-
-#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);
- }
-
- 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;
- }
- }
- }
-#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.activeInputsCountOnDevices() > 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);
- }
-
- // Do internal direct magic here
- bool offload_disabled = property_get_bool("audio.offload.disable", false);
- if ((flags == AUDIO_OUTPUT_FLAG_NONE) &&
- (stream == AUDIO_STREAM_MUSIC) &&
- (offloadInfo != NULL) && !offload_disabled &&
- ((offloadInfo->usage == AUDIO_USAGE_MEDIA) || (offloadInfo->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);
- }
-
- 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 &&
- property_get_bool("audio.deep_buffer.media", false /* default_value */)) {
- forced_deep = true;
- }
-
- 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 <= 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 = getProfileForDirectOutput(device,
- samplingRate,
- format,
- channelMask,
- (audio_output_flags_t)flags);
-
- 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 (!(property_get_bool("audio.offload.multiple.enabled", false) &&
- ((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 ((samplingRate == outputDesc->mSamplingRate) &&
- audio_formats_match(format, outputDesc->mFormat) &&
- (channelMask == outputDesc->mChannelMask)) {
- if (session == outputDesc->mDirectClientSession) {
- outputDesc->mDirectOpenCount++;
- ALOGV("getOutput() reusing direct output %d for session %d",
- mOutputs.keyAt(i), session);
- return mOutputs.keyAt(i);
- } else {
- ALOGV("getOutput() do not reuse direct output because current client (%d) "
- "is not the same as requesting client (%d)",
- outputDesc->mDirectClientSession, session);
- goto non_direct_output;
- }
- }
- }
- }
- // 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->getFlags() & 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 && !audio_formats_match(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 <= SAMPLE_RATE_HZ_MAX) {
- 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;
- outputDesc->mDirectClientSession = session;
-
- 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:
-
- // 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) != 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(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);
-
- 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, 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,
- 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;
-#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,
- config,
- flags,
- selectedDeviceId,
- inputType,
- portId);
-}
-
-
-status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input,
- audio_session_t session,
- concurrency_type__mask_t *concurrency)
-{
- ALOGV("startInput() input %d", input);
- *concurrency = API_INPUT_CONCURRENCY_NONE;
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("startInput() unknown input %d", input);
- return BAD_VALUE;
- }
- sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
-
- sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
- if (audioSession == 0) {
- ALOGW("startInput() unknown session %d on input %d", session, input);
- return BAD_VALUE;
- }
-
- 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;
- }
-#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.activeInputsCountOnDevices() == 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));
- }
- }
- // If effects where present on any of the above closed outputs,
- // audioflinger moved them to the primary output by default
- // move them back to the appropriate output.
- moveGlobalEffect();
- }
-#endif
-
- // increment activity count before calling getNewInputDevice() below as only active sessions
- // are considered for device selection
- audioSession->changeActiveCount(1);
-
- // Routing?
- mInputRoutes.incRouteActivity(session);
-
- if (audioSession->activeCount() == 1 || mInputRoutes.hasRouteChanged(session)) {
- // indicate active capture to sound trigger service if starting capture from a mic on
- // primary HW module
- audio_devices_t device = getNewInputDevice(inputDesc);
- setInputDevice(input, device, true /* force */);
-
- if (inputDesc->getAudioSessionCount(true/*activeOnly*/) == 1) {
- // 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->mDeviceAddress,
- MIX_STATE_MIXING);
- }
-
- audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
- if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
- 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->mDevice)) {
- String8 address = String8("");
- if (inputDesc->mPolicyMix == NULL) {
- address = String8("0");
- } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
- address = inputDesc->mPolicyMix->mDeviceAddress;
- }
- if (address != "") {
- setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address, "remote-submix");
- }
- }
- }
- }
-
- ALOGV("AudioPolicyManager::startInput() input source = %d", audioSession->inputSource());
-#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.activeInputsCountOnDevices() == 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;
-}
-
-void AudioPolicyManagerCustom::closeAllInputs() {
- bool patchRemoved = false;
-
- for(size_t input_index = mInputs.size(); input_index > 0; input_index--) {
- sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index-1);
- ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
- if (patch_index >= 0) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
- (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- mAudioPatches.removeItemsAt(patch_index);
- patchRemoved = true;
- }
- mpClientInterface->closeInput(mInputs.keyAt(input_index-1));
- }
- mInputs.clear();
- SoundTrigger::setCaptureState(false);
- nextAudioPortGeneration();
-
- if (patchRemoved) {
- mpClientInterface->onAudioPatchListUpdate();
- }
-}
-
-AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
- : AudioPolicyManager(clientInterface),
- mHdmiAudioDisabled(false),
- mHdmiAudioEvent(false),
-#ifndef FM_POWER_OPT
- mPrevPhoneState(0)
-#else
- mPrevPhoneState(0),
- mPrevFMVolumeDb(0.0f),
- mFMIsActive(false)
-#endif
-{
-
-#ifdef USE_XML_AUDIO_POLICY_CONF
- ALOGD("USE_XML_AUDIO_POLICY_CONF is TRUE");
-#else
- ALOGD("USE_XML_AUDIO_POLICY_CONF is FALSE");
-#endif
-
-#ifdef RECORD_PLAY_CONCURRENCY
- mIsInputRequestOnProgress = false;
-#endif
-
-
-#ifdef VOICE_CONCURRENCY
- mFallBackflag = getFallBackPath();
-#endif
-}
-}
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
deleted file mode 100644
index 433380b..0000000
--- a/policy_hal/AudioPolicyManager.h
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- * Copyright (c) 2013-2017, 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.
- */
-
-
-#include <audiopolicy/managerdefault/AudioPolicyManager.h>
-#include <audio_policy_conf.h>
-#include <Volume.h>
-
-
-namespace android {
-#ifndef AUDIO_EXTN_FORMATS_ENABLED
-#define AUDIO_FORMAT_WMA 0x12000000UL
-#define AUDIO_FORMAT_WMA_PRO 0x13000000UL
-#define AUDIO_FORMAT_FLAC 0x1B000000UL
-#define AUDIO_FORMAT_ALAC 0x1C000000UL
-#define AUDIO_FORMAT_APE 0x1D000000UL
-#endif
-
-#define WMA_STD_NUM_FREQ 7
-#define WMA_STD_NUM_CHANNELS 2
-static uint32_t wmaStdSampleRateTbl[WMA_STD_NUM_FREQ] =
-{
- 8000, 11025, 16000, 22050, 32000, 44100, 48000
-};
-
-static uint32_t wmaStdMinAvgByteRateTbl[WMA_STD_NUM_FREQ][WMA_STD_NUM_CHANNELS] =
-{
- {128, 12000},
- {8016, 8016},
- {10000, 16000},
- {16016, 20008},
- {20000, 24000},
- {20008, 31960},
- {63000, 63000}
-};
-
-static uint32_t wmaStdMaxAvgByteRateTbl[WMA_STD_NUM_FREQ][WMA_STD_NUM_CHANNELS] =
-{
- {8000, 12000},
- {10168, 10168},
- {16000, 20000},
- {20008, 32048},
- {20000, 48000},
- {48024, 320032},
- {256008, 256008}
-};
-
-#define MAX_BITRATE_WMA_PRO 1536000
-#define MAX_BITRATE_WMA_LOSSLESS 1152000
-
-#ifndef AAC_ADTS_OFFLOAD_ENABLED
-#define AUDIO_FORMAT_AAC_ADTS 0x1E000000UL
-#endif
-
-#ifndef AUDIO_EXTN_AFE_PROXY_ENABLED
-#define AUDIO_DEVICE_OUT_PROXY 0x1000000
-#endif
-
-// ----------------------------------------------------------------------------
-
-class AudioPolicyManagerCustom: public AudioPolicyManager
-{
-
-public:
- AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface);
-
- virtual ~AudioPolicyManagerCustom() {}
-
- status_t setDeviceConnectionStateInt(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address,
- const char *device_name);
- virtual void setPhoneState(audio_mode_t state);
- virtual void setForceUse(audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config);
-
- virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
-
- virtual status_t getInputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *input,
- 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);
- // indicates to the audio policy manager that the input starts being used.
- virtual status_t startInput(audio_io_handle_t input,
- audio_session_t session,
- concurrency_type__mask_t *concurrency);
- // indicates to the audio policy manager that the input stops being used.
- virtual status_t stopInput(audio_io_handle_t input,
- audio_session_t session);
-
- virtual void closeAllInputs();
-
-protected:
-
- status_t checkAndSetVolume(audio_stream_type_t stream,
- int index,
- const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t device,
- int delayMs = 0, bool force = false);
-
- // avoid invalidation for active music stream on previous outputs
- // which is supported on the new device.
- bool isInvalidationOfMusicStreamNeeded(routing_strategy strategy);
-
- // Must be called before updateDevicesAndOutputs()
- void checkOutputForStrategy(routing_strategy strategy);
-
- // returns true if given output is direct output
- bool isDirectOutput(audio_io_handle_t output);
-
- // if argument "device" is different from AUDIO_DEVICE_NONE, startSource() will force
- // the re-evaluation of the output device.
- status_t startSource(const sp<AudioOutputDescriptor>& outputDesc,
- audio_stream_type_t stream,
- audio_devices_t device,
- const char *address,
- uint32_t *delayMs);
- status_t stopSource(const sp<AudioOutputDescriptor>& outputDesc,
- audio_stream_type_t stream,
- bool forceDeviceUpdate);
- // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
- // returns 0 if no mute/unmute event happened, the largest latency of the device where
- // the mute/unmute happened
- uint32_t handleEventForBeacon(int){return 0;}
- uint32_t setBeaconMute(bool){return 0;}
-#ifdef VOICE_CONCURRENCY
- static audio_output_flags_t getFallBackPath();
- int mFallBackflag;
-#endif /*VOICE_CONCURRENCY*/
- void moveGlobalEffect();
-
- // handle special cases for sonification strategy while in call: mute streams or replace by
- // a special tone in the device used for communication
- void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange, audio_io_handle_t output);
- //parameter indicates of HDMI speakers disabled
- bool mHdmiAudioDisabled;
- //parameter indicates if HDMI plug in/out detected
- bool mHdmiAudioEvent;
-private:
- // updates device caching and output for streams that can influence the
- // routing of notifications
- void handleNotificationRoutingForStream(audio_stream_type_t stream);
- // internal method to return the output handle for the given device and format
- audio_io_handle_t getOutputForDevice(
- audio_devices_t device,
- audio_session_t session,
- audio_stream_type_t stream,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo);
- // internal method to fill offload info in case of Direct PCM
- status_t 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);
- // Used for voip + voice concurrency usecase
- int mPrevPhoneState;
-#ifdef VOICE_CONCURRENCY
- int mvoice_call_state;
-#endif
-#ifdef RECORD_PLAY_CONCURRENCY
- // Used for record + playback concurrency
- bool mIsInputRequestOnProgress;
-#endif
-
-#ifdef FM_POWER_OPT
- float mPrevFMVolumeDb;
- bool mFMIsActive;
-#endif
-};
-};