diff --git a/Android.mk b/Android.mk index 2f4054f7..429c4cb2 100644 --- a/Android.mk +++ b/Android.mk @@ -12,7 +12,6 @@ include $(MY_LOCAL_PATH)/hal/Android.mk 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 $(MY_LOCAL_PATH)/qahw_api/Android.mk diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk deleted file mode 100644 index 5b7f6f1b..00000000 --- a/policy_hal/Android.mk +++ /dev/null @@ -1,96 +0,0 @@ -# This file was modified by Dolby Laboratories, Inc. The portions of the -# code that are surrounded by "DOLBY..." are copyrighted and -# licensed separately, as follows: -# -# (C) 2016 Dolby Laboratories, Inc. -# All rights reserved. -# -# This program is protected under international and U.S. Copyright laws as -# an unpublished work. This program is confidential and proprietary to the -# copyright owners. Reproduction or disclosure, in whole or in part, or the -# production of derivative works therefrom without the express permission of -# the copyright owners is prohibited. -# -ifneq ($(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 72bd6447..00000000 --- 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 -#include - -#include -#include -#include -#include -#include -#include -#include "AudioPolicyManager.h" -#include -#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 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 devDesc = - mHwModules.getDeviceDescriptor(device, device_address, device_name); - - // handle output devices - if (audio_is_output_device(device)) { - SortedVector 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 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 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 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 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 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 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 srcOutputs = getOutputsForDevice(oldDevice, mOutputs); - SortedVector 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 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 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 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 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 > activeInputs = mInputs.getActiveInputs(); - if (activeInputs.size() != 0) { - for (size_t i = 0; i < activeInputs.size(); i++) { - sp 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 > activeInputs = mInputs.getActiveInputs(); - if (activeInputs.size() != 0) { - for (size_t i = 0; i < activeInputs.size(); i++) { - sp 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 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 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 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 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 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 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 > activeInputs = mInputs.getActiveInputs(); - for (size_t i = 0; i < activeInputs.size(); i++) { - sp 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& 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 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& 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 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 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& 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 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 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 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 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 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 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 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 inputDesc = mInputs.valueAt(index); - - sp 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 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 inputDesc = mInputs.valueAt(input_index-1); - ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle()); - if (patch_index >= 0) { - sp 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 433380bb..00000000 --- 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 -#include -#include - - -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& 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& outputDesc, - audio_stream_type_t stream, - audio_devices_t device, - const char *address, - uint32_t *delayMs); - status_t stopSource(const sp& 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 -}; -};