audio: msm8909w caf release LW.BR.1.0-00410-8x09w.0
MSM8909w Audio HAL code copied from CAF release
LW.BR.1.0-00410-8x09w.0
dbcce50 hal: Port wcd9326 changes to 8909
410c530 hal: update error handling for pcm_prepare failures
ff79309 hal: fix compilation issues with audio FM extention
762d7eb policy_hal: add support for fm device loopback
7c418f9 audio_policy: modify few methods to appropriately override base
8b12163 audio: Add support to enable split A2DP
a0559fa Revert "Revert "policy_hal: Function prototype correction for custom policy"."
Fixed makefiles to be compatible with PDK without kernel source
Change-Id: I9c6f2139adee62426b877516deeb41d4ed8052b2
diff --git a/msm8909/policy_hal/Android.mk b/msm8909/policy_hal/Android.mk
new file mode 100644
index 0000000..29bb01a
--- /dev/null
+++ b/msm8909/policy_hal/Android.mk
@@ -0,0 +1,69 @@
+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 \
+ $(call include-path-for, avextension)
+
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libutils \
+ liblog \
+ libsoundtrigger \
+ libaudiopolicymanagerdefault \
+ libserviceutility
+
+LOCAL_STATIC_LIBRARIES := \
+ libmedia_helper \
+
+ifeq ($(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 $(AUDIO_FEATURE_ENABLED_EXTN_FORMATS)),true)
+LOCAL_CFLAGS += -DAUDIO_EXTN_FORMATS_ENABLED
+endif
+
+ifeq ($(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 $(AUDIO_FEATURE_ENABLED_COMPRESS_VOIP)),true)
+ LOCAL_CFLAGS += -DCOMPRESS_VOIP_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_NON_WEARABLE_TARGET)),true)
+ LOCAL_CFLAGS += -DNON_WEARABLE_TARGET
+else
+ LOCAL_CFLAGS += -Wno-error -fpermissive
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FM_POWER_OPT)),true)
+LOCAL_CFLAGS += -DFM_POWER_OPT
+endif
+
+LOCAL_MODULE := libaudiopolicymanager
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif
+endif
diff --git a/msm8909/policy_hal/AudioPolicyManager.cpp b/msm8909/policy_hal/AudioPolicyManager.cpp
new file mode 100644
index 0000000..a279c1e
--- /dev/null
+++ b/msm8909/policy_hal/AudioPolicyManager.cpp
@@ -0,0 +1,1951 @@
+/*
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ * Not a contribution.
+ *
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ * This file was modified by Dolby Laboratories, Inc. The portions of the
+ * code that are surrounded by "DOLBY..." are copyrighted and
+ * licensed separately, as follows:
+ *
+ * (C) 2015 Dolby Laboratories, Inc.
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyManagerCustom"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#define MIN(a, b) ((a) < (b) ? (a) : (b))
+
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+// A device mask for all audio input and output devices where matching inputs/outputs on device
+// type alone is not enough: the address must match too
+#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
+// Following delay should be used if the calculated routing delay from all active
+// input streams is higher than this value
+#define MAX_VOICE_CALL_START_DELAY_MS 100
+
+#include <inttypes.h>
+#include <math.h>
+
+#include <cutils/properties.h>
+#include <utils/Log.h>
+#include <hardware/audio.h>
+#include <hardware/audio_effect.h>
+#include <media/AudioParameter.h>
+#include <soundtrigger/SoundTrigger.h>
+#include "AudioPolicyManager.h"
+#include <policy.h>
+#ifdef DOLBY_ENABLE
+#include "DolbyAudioPolicy_impl.h"
+#endif // DOLBY_END
+
+namespace android {
+#ifdef VOICE_CONCURRENCY
+audio_output_flags_t AudioPolicyManagerCustom::getFallBackPath()
+{
+ audio_output_flags_t flag = AUDIO_OUTPUT_FLAG_FAST;
+ char propValue[PROPERTY_VALUE_MAX];
+
+ if (property_get("voice.conc.fallbackpath", propValue, NULL)) {
+ if (!strncmp(propValue, "deep-buffer", 11)) {
+ flag = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ }
+ else if (!strncmp(propValue, "fast", 4)) {
+ flag = AUDIO_OUTPUT_FLAG_FAST;
+ }
+ else {
+ ALOGD("voice_conc:not a recognised path(%s) in prop voice.conc.fallbackpath",
+ propValue);
+ }
+ }
+ else {
+ ALOGD("voice_conc:prop voice.conc.fallbackpath not set");
+ }
+
+ ALOGD("voice_conc:picked up flag(0x%x) from prop voice.conc.fallbackpath",
+ flag);
+
+ return flag;
+}
+#endif /*VOICE_CONCURRENCY*/
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+extern "C" AudioPolicyInterface* createAudioPolicyManager(
+ AudioPolicyClientInterface *clientInterface)
+{
+ return new AudioPolicyManagerCustom(clientInterface);
+}
+
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+{
+ delete interface;
+}
+
+status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name)
+{
+ ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
+ device, state, device_address, device_name);
+
+ // connect/disconnect only 1 device at a time
+ if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+
+ sp<DeviceDescriptor> devDesc =
+ mHwModules.getDeviceDescriptor(device, device_address, device_name);
+
+ // handle output devices
+ if (audio_is_output_device(device)) {
+ SortedVector <audio_io_handle_t> outputs;
+
+ ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
+
+ // save a copy of the opened output descriptors before any output is opened or closed
+ // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
+ mPreviousOutputs = mOutputs;
+ switch (state)
+ {
+ // handle output device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, "hdmi_spkr", 9)) {
+ mHdmiAudioDisabled = false;
+ } else {
+ mHdmiAudioEvent = true;
+ }
+ }
+#endif
+ ALOGW("setDeviceConnectionState() device already connected: %x", device);
+ return INVALID_OPERATION;
+ }
+ ALOGV("setDeviceConnectionState() connecting device %x", device);
+
+ // register new device as available
+ index = mAvailableOutputDevices.add(devDesc);
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, "hdmi_spkr", 9)) {
+ mHdmiAudioDisabled = false;
+ } else {
+ mHdmiAudioEvent = true;
+ }
+ if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
+ mAvailableOutputDevices.remove(devDesc);
+ ALOGW("HDMI sink not connected, do not route audio to HDMI out");
+ return INVALID_OPERATION;
+ }
+ }
+#endif
+ if (index >= 0) {
+ sp<HwModule> module = mHwModules.getModuleForDevice(device);
+ if (module == 0) {
+ ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
+ device);
+ mAvailableOutputDevices.remove(devDesc);
+ return INVALID_OPERATION;
+ }
+ mAvailableOutputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
+ mAvailableOutputDevices.remove(devDesc);
+ return INVALID_OPERATION;
+ }
+ // Propagate device availability to Engine
+ mEngine->setDeviceConnectionState(devDesc, state);
+
+ // outputs should never be empty here
+ ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
+ "checkOutputsForDevice() returned no outputs but status OK");
+ ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
+ outputs.size());
+
+ // Send connect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ } break;
+ // handle output device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, "hdmi_spkr", 9)) {
+ mHdmiAudioDisabled = true;
+ } else {
+ mHdmiAudioEvent = false;
+ }
+ }
+#endif
+ ALOGW("setDeviceConnectionState() device not connected: %x", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
+
+ // Send Disconnect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ // remove device from available output devices
+ mAvailableOutputDevices.remove(devDesc);
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, "hdmi_spkr", 9)) {
+ mHdmiAudioDisabled = true;
+ } else {
+ mHdmiAudioEvent = false;
+ }
+ }
+#endif
+
+ checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
+
+ // Propagate device availability to Engine
+ mEngine->setDeviceConnectionState(devDesc, state);
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
+ // output is suspended before any tracks are moved to it
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ // outputs must be closed after checkOutputForAllStrategies() is executed
+ if (!outputs.isEmpty()) {
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+ // close unused outputs after device disconnection or direct outputs that have been
+ // opened by checkOutputsForDevice() to query dynamic parameters
+ if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+ (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+ (desc->mDirectOpenCount == 0))) {
+ closeOutput(outputs[i]);
+ }
+ }
+ // check again after closing A2DP output to reset mA2dpSuspended if needed
+ checkA2dpSuspend();
+ }
+
+ updateDevicesAndOutputs();
+#ifdef DOLBY_ENABLE
+ // Before closing the opened outputs, update endpoint property with device capabilities
+ audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AUDIO_STREAM_MUSIC), true);
+ mDolbyAudioPolicy.setEndpointSystemProperty(audioOutputDevice, mHwModules);
+#endif // DOLBY_END
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+
+#ifdef FM_POWER_OPT
+ // handle FM device connection state to trigger FM AFE loopback
+ if(device == AUDIO_DEVICE_OUT_FM && hasPrimaryOutput()) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, 1);
+ newDevice = newDevice | AUDIO_DEVICE_OUT_FM;
+ } else {
+ mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, -1);
+ }
+ AudioParameter param = AudioParameter();
+ param.addInt(String8("handle_fm"), (int)newDevice);
+ mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString());
+ }
+#endif /* FM_POWER_OPT end */
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
+ audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
+ // do not force device change on duplicated output because if device is 0, it will
+ // also force a device 0 for the two outputs it is duplicated to which may override
+ // a valid device selection on those outputs.
+ bool force = !desc->isDuplicated()
+ && (!device_distinguishes_on_address(device)
+ // always force when disconnecting (a non-duplicated device)
+ || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
+ setOutputDevice(desc, newDevice, force, 0);
+ }
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is output device
+
+ // handle input devices
+ if (audio_is_input_device(device)) {
+ SortedVector <audio_io_handle_t> inputs;
+
+ ssize_t index = mAvailableInputDevices.indexOf(devDesc);
+ switch (state)
+ {
+ // handle input device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ ALOGW("setDeviceConnectionState() device already connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ sp<HwModule> module = mHwModules.getModuleForDevice(device);
+ if (module == NULL) {
+ ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
+ device);
+ return INVALID_OPERATION;
+ }
+ if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) {
+ return INVALID_OPERATION;
+ }
+
+ index = mAvailableInputDevices.add(devDesc);
+ if (index >= 0) {
+ mAvailableInputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ // Set connect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ // Propagate device availability to Engine
+ mEngine->setDeviceConnectionState(devDesc, state);
+ } break;
+
+ // handle input device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ ALOGW("setDeviceConnectionState() device not connected: %d", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
+
+ // Set Disconnect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress);
+ mAvailableInputDevices.remove(devDesc);
+
+ // Propagate device availability to Engine
+ mEngine->setDeviceConnectionState(devDesc, state);
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ closeAllInputs();
+
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is input device
+
+ ALOGW("setDeviceConnectionState() invalid device: %x", device);
+ return BAD_VALUE;
+}
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManagerCustom::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+ ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+ " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
+ offloadInfo.sample_rate, offloadInfo.channel_mask,
+ offloadInfo.format,
+ offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+ offloadInfo.has_video);
+#ifdef VOICE_CONCURRENCY
+ char concpropValue[PROPERTY_VALUE_MAX];
+ if (property_get("voice.playback.conc.disabled", concpropValue, NULL)) {
+ bool propenabled = atoi(concpropValue) || !strncmp("true", concpropValue, 4);
+ if (propenabled) {
+ if (isInCall())
+ {
+ ALOGD("\n copl: blocking compress offload on call mode\n");
+ return false;
+ }
+ }
+ }
+#endif
+#ifdef RECORD_PLAY_CONCURRENCY
+ char recConcPropValue[PROPERTY_VALUE_MAX];
+ bool prop_rec_play_enabled = false;
+
+ if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
+ prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
+ }
+
+ if ((prop_rec_play_enabled) &&
+ ((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCount() > 0))) {
+ ALOGD("copl: blocking compress offload for record concurrency");
+ return false;
+ }
+#endif
+ // Check if offload has been disabled
+ char propValue[PROPERTY_VALUE_MAX];
+ if (property_get("audio.offload.disable", propValue, "0")) {
+ if (atoi(propValue) != 0) {
+ ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+ return false;
+ }
+ }
+
+ // Check if stream type is music, then only allow offload as of now.
+ if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+ {
+ ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+ return false;
+ }
+ //check if it's multi-channel AAC (includes sub formats) and FLAC and VORBIS format
+ if ((popcount(offloadInfo.channel_mask) > 2) &&
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
+ ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
+ return false;
+ }
+#ifdef AUDIO_EXTN_FORMATS_ENABLED
+ //check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
+ if ((popcount(offloadInfo.channel_mask) > 2) &&
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && offloadInfo.sample_rate > 48000) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && offloadInfo.sample_rate > 48000) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && offloadInfo.sample_rate > 48000))) {
+ ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA clips with sample rate > 48kHz");
+ return false;
+ }
+#endif
+ //TODO: enable audio offloading with video when ready
+ const bool allowOffloadWithVideo =
+ property_get_bool("audio.offload.video", false /* default_value */);
+ if (offloadInfo.has_video && !allowOffloadWithVideo) {
+ ALOGV("isOffloadSupported: has_video == true, returning false");
+ return false;
+ }
+
+ //If duration is less than minimum value defined in property, return false
+ if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+ if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+ ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+ return false;
+ }
+ } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+ ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+ //duration checks only valid for MP3/AAC/VORBIS/WMA/ALAC/APE formats,
+ //do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
+ if ((offloadInfo.format == AUDIO_FORMAT_MP3) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS)
+#ifdef AUDIO_EXTN_FORMATS_ENABLED
+ || ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE)
+#endif
+ )
+ return false;
+
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if (mEffects.isNonOffloadableEffectEnabled()) {
+ return false;
+ }
+ // Check for soundcard status
+ String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+ String8("SND_CARD_STATUS"));
+ AudioParameter result = AudioParameter(valueStr);
+ int isonline = 0;
+ if ((result.getInt(String8("SND_CARD_STATUS"), isonline) == NO_ERROR)
+ && !isonline) {
+ ALOGD("copl: soundcard is offline rejecting offload request");
+ return false;
+ }
+ // See if there is a profile to support this.
+ // AUDIO_DEVICE_NONE
+ sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ offloadInfo.sample_rate,
+ offloadInfo.format,
+ offloadInfo.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
+ return (profile != 0);
+}
+audio_devices_t AudioPolicyManagerCustom::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
+ bool fromCache)
+{
+ audio_devices_t device = AUDIO_DEVICE_NONE;
+
+ ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ if (patchDesc->mUid != mUidCached) {
+ ALOGV("getNewOutputDevice() device %08x forced by patch %d",
+ outputDesc->device(), outputDesc->mPatchHandle);
+ return outputDesc->device();
+ }
+ }
+
+ // check the following by order of priority to request a routing change if necessary:
+ // 1: the strategy enforced audible is active and enforced on the output:
+ // use device for strategy enforced audible
+ // 2: we are in call or the strategy phone is active on the output:
+ // use device for strategy phone
+ // 3: the strategy for enforced audible is active but not enforced on the output:
+ // use the device for strategy enforced audible
+ // 4: the strategy sonification is active on the output:
+ // use device for strategy sonification
+ // 5: the strategy "respectful" sonification is active on the output:
+ // use device for strategy "respectful" sonification
+ // 6: the strategy accessibility is active on the output:
+ // use device for strategy accessibility
+ // 7: the strategy media is active on the output:
+ // use device for strategy media
+ // 8: the strategy DTMF is active on the output:
+ // use device for strategy DTMF
+ // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output:
+ // use device for strategy t-t-s
+ if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) &&
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isInCall() ||
+ isStrategyActive(outputDesc, STRATEGY_PHONE)||
+ isStrategyActive(mPrimaryOutput, STRATEGY_PHONE)) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)||
+ (isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION)
+ && (!isStrategyActive(mPrimaryOutput,STRATEGY_MEDIA)))) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)||
+ (isStrategyActive(mPrimaryOutput,STRATEGY_SONIFICATION_RESPECTFUL)
+ && (!isStrategyActive(mPrimaryOutput, STRATEGY_MEDIA)))) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) {
+ device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) {
+ device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) {
+ device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
+ device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) {
+ device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
+ }
+
+ ALOGV("getNewOutputDevice() selected device %x", device);
+ return device;
+}
+void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)
+{
+ ALOGV("setPhoneState() state %d", state);
+ // store previous phone state for management of sonification strategy below
+ audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+ int oldState = mEngine->getPhoneState();
+
+ if (mEngine->setPhoneState(state) != NO_ERROR) {
+ ALOGW("setPhoneState() invalid or same state %d", state);
+ return;
+ }
+ /// Opens: can these line be executed after the switch of volume curves???
+ // if leaving call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isStateInCall(oldState)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (size_t j = 0; j < mOutputs.size(); j++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(j);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+
+ handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput);
+ }
+ }
+
+ // force reevaluating accessibility routing when call starts
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ /**
+ * Switching to or from incall state or switching between telephony and VoIP lead to force
+ * routing command.
+ */
+ bool force = ((is_state_in_call(oldState) != is_state_in_call(state))
+ || (is_state_in_call(state) && (state != oldState)));
+
+ // check for device and output changes triggered by new phone state
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+
+ sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
+#ifdef VOICE_CONCURRENCY
+ int voice_call_state = 0;
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_playback_enabled = false, prop_rec_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
+ prop_playback_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.record.conc.disabled", propValue, NULL)) {
+ prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ bool mode_in_call = (AUDIO_MODE_IN_CALL != oldState) && (AUDIO_MODE_IN_CALL == state);
+ //query if it is a actual voice call initiated by telephony
+ if (mode_in_call) {
+ String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, String8("in_call"));
+ AudioParameter result = AudioParameter(valueStr);
+ if (result.getInt(String8("in_call"), voice_call_state) == NO_ERROR)
+ ALOGD("voice_conc:SetPhoneState: Voice call state = %d", voice_call_state);
+ }
+
+ if (mode_in_call && voice_call_state && !mvoice_call_state) {
+ ALOGD("voice_conc:Entering to call mode oldState :: %d state::%d ",
+ oldState, state);
+ mvoice_call_state = voice_call_state;
+ if (prop_rec_enabled) {
+ //Close all active inputs
+ audio_io_handle_t activeInput = mInputs.getActiveInput();
+ if (activeInput != 0) {
+ sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
+ switch(activeDesc->mInputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ ALOGD("voice_conc:FOUND active input during call active: %d",activeDesc->mInputSource);
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if(prop_voip_enabled) {
+ ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",activeDesc->mInputSource);
+ stopInput(activeInput, activeDesc->mSessions.itemAt(0));
+ releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
+ }
+ break;
+
+ default:
+ ALOGD("voice_conc:CLOSING input on call setup for inputSource: %d",activeDesc->mInputSource);
+ stopInput(activeInput, activeDesc->mSessions.itemAt(0));
+ releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
+ break;
+ }
+ }
+ } else if (prop_voip_enabled) {
+ audio_io_handle_t activeInput = mInputs.getActiveInput();
+ if (activeInput != 0) {
+ sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
+ if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeDesc->mInputSource) {
+ ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeDesc->mInputSource);
+ stopInput(activeInput, activeDesc->mSessions.itemAt(0));
+ releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
+ }
+ }
+ }
+ if (prop_playback_enabled) {
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
+ ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
+ if (i == AUDIO_STREAM_PATCH) {
+ ALOGV("voice_conc:not calling invalidate for AUDIO_STREAM_PATCH");
+ continue;
+ }
+ if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+ if ((AUDIO_STREAM_MUSIC == i) ||
+ (AUDIO_STREAM_VOICE_CALL == i) ) {
+ ALOGD("voice_conc:Invalidate stream type %d", i);
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ } else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+ ALOGD("voice_conc:Invalidate stream type %d", i);
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("voice_conc:ouput desc / profile is NULL");
+ continue;
+ }
+
+ if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+ if (((!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY))
+ && prop_playback_enabled) {
+ ALOGD("voice_conc:calling suspendOutput on call mode for primary output");
+ mpClientInterface->suspendOutput(mOutputs.keyAt(i));
+ } //Close compress all sessions
+ else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
+ && prop_playback_enabled) {
+ ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX)
+ && prop_voip_enabled) {
+ ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ } else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+ if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)
+ && prop_playback_enabled) {
+ ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ }
+ }
+
+ if ((AUDIO_MODE_IN_CALL == oldState || AUDIO_MODE_IN_COMMUNICATION == oldState) &&
+ (AUDIO_MODE_NORMAL == state) && prop_playback_enabled && mvoice_call_state) {
+ ALOGD("voice_conc:EXITING from call mode oldState :: %d state::%d \n",oldState, state);
+ mvoice_call_state = 0;
+ if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+ //restore PCM (deep-buffer) output after call termination
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("voice_conc:ouput desc / profile is NULL");
+ continue;
+ }
+ if (!outputDesc->isDuplicated() && outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ ALOGD("voice_conc:calling restoreOutput after call mode for primary output");
+ mpClientInterface->restoreOutput(mOutputs.keyAt(i));
+ }
+ }
+ }
+ //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
+ for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
+ ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
+ if (i == AUDIO_STREAM_PATCH) {
+ ALOGV("voice_conc:not calling invalidate for AUDIO_STREAM_PATCH");
+ continue;
+ }
+ if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+ if ((AUDIO_STREAM_MUSIC == i) ||
+ (AUDIO_STREAM_VOICE_CALL == i) ) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ } else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+
+#endif
+#ifdef RECORD_PLAY_CONCURRENCY
+ char recConcPropValue[PROPERTY_VALUE_MAX];
+ bool prop_rec_play_enabled = false;
+
+ if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
+ prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
+ }
+ if (prop_rec_play_enabled) {
+ if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
+ ALOGD("phone state changed to MODE_IN_COMM invlaidating music and voice streams");
+ // call invalidate for voice streams, so that it can use deepbuffer with VoIP out device from HAL
+ mpClientInterface->invalidateStream(AUDIO_STREAM_VOICE_CALL);
+ // call invalidate for music, so that compress will fallback to deep-buffer with VoIP out device
+ mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+
+ // close compress output to make sure session will be closed before timeout(60sec)
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("ouput desc / profile is NULL");
+ continue;
+ }
+
+ if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ ALOGD("calling closeOutput on call mode for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ } else if ((oldState == AUDIO_MODE_IN_COMMUNICATION) &&
+ (mEngine->getPhoneState() == AUDIO_MODE_NORMAL)) {
+ // call invalidate for music so that music can fallback to compress
+ mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+ }
+ }
+#endif
+ mPrevPhoneState = oldState;
+ int delayMs = 0;
+ if (isStateInCall(state)) {
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ // mute media and sonification strategies and delay device switch by the largest
+ // latency of any output where either strategy is active.
+ // This avoid sending the ring tone or music tail into the earpiece or headset.
+ if ((isStrategyActive(desc, STRATEGY_MEDIA,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime) ||
+ isStrategyActive(desc, STRATEGY_SONIFICATION,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime)) &&
+ (delayMs < (int)desc->latency()*2)) {
+ delayMs = desc->latency()*2;
+ }
+ setStrategyMute(STRATEGY_MEDIA, true, desc);
+ setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+ setStrategyMute(STRATEGY_SONIFICATION, true, desc);
+ setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+ }
+ ALOGV("Setting the delay from %dms to %dms", delayMs,
+ MIN(delayMs, MAX_VOICE_CALL_START_DELAY_MS));
+ delayMs = MIN(delayMs, MAX_VOICE_CALL_START_DELAY_MS);
+ }
+
+ if (hasPrimaryOutput()) {
+ // Note that despite the fact that getNewOutputDevice() is called on the primary output,
+ // the device returned is not necessarily reachable via this output
+ audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
+ rxDevice = mPrimaryOutput->device();
+ }
+
+ if (state == AUDIO_MODE_IN_CALL) {
+ updateCallRouting(rxDevice, delayMs);
+ } else if (oldState == AUDIO_MODE_IN_CALL) {
+ if (mCallRxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ mCallRxPatch.clear();
+ }
+ if (mCallTxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ mCallTxPatch.clear();
+ }
+ setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ } else {
+ setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ }
+ }
+ //update device for all non-primary outputs
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ if (output != mPrimaryOutput->mIoHandle) {
+ newDevice = getNewOutputDevice(mOutputs.valueFor(output), false /*fromCache*/);
+ setOutputDevice(mOutputs.valueFor(output), newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ }
+ }
+ // if entering in call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isStateInCall(state)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (size_t j = 0; j < mOutputs.size(); j++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(j);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput);
+ }
+ }
+ }
+
+ // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+ if (state == AUDIO_MODE_RINGTONE &&
+ isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+ mLimitRingtoneVolume = true;
+ } else {
+ mLimitRingtoneVolume = false;
+ }
+}
+
+void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
+
+ if (mEngine->setForceUse(usage, config) != NO_ERROR) {
+ ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
+ return;
+ }
+ bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
+
+ // check for device and output changes triggered by new force usage
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ // Use reverse loop to make sure any low latency usecases (generally tones)
+ // are not routed before non LL usecases (generally music).
+ // We can safely assume that LL output would always have lower index,
+ // and use this work-around to avoid routing of output with music stream
+ // from the context of short lived LL output.
+ // Note: in case output's share backend(HAL sharing is implicit) all outputs
+ // gets routing update while processing first output itself.
+ for (size_t i = mOutputs.size(); i > 0; i--) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
+ audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || outputDesc != mPrimaryOutput) {
+ setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ }
+ if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+ applyStreamVolumes(outputDesc, newDevice, 0, true);
+ }
+ }
+
+ audio_io_handle_t activeInput = mInputs.getActiveInput();
+ if (activeInput != 0) {
+ setInputDevice(activeInput, getNewInputDevice(activeInput));
+ }
+
+}
+
+status_t AudioPolicyManagerCustom::stopSource(sp<AudioOutputDescriptor> outputDesc1,
+ audio_stream_type_t stream,
+ bool forceDeviceUpdate)
+{
+ if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ ALOGW("stopSource() invalid stream %d", stream);
+ return INVALID_OPERATION;
+ }
+
+ // always handle stream stop, check which stream type is stopping
+#ifdef NON_WEARABLE_TARGET
+ sp<AudioOutputDescriptor> outputDesc = outputDesc1;
+#else
+ sp<SwAudioOutputDescriptor> outputDesc = (sp<SwAudioOutputDescriptor>) outputDesc1;
+#endif
+ handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
+
+ // handle special case for sonification while in call
+ if (isInCall() && (outputDesc->mRefCount[stream] == 1)) {
+ if (outputDesc->isDuplicated()) {
+#ifdef NON_WEARABLE_TARGET
+ handleIncallSonification(stream, false, false, outputDesc->subOutput1()->mIoHandle);
+ handleIncallSonification(stream, false, false, outputDesc->subOutput2()->mIoHandle);
+#else
+ handleIncallSonification(stream, false, false, outputDesc->mOutput1->mIoHandle);
+ handleIncallSonification(stream, false, false, outputDesc->mOutput2->mIoHandle);
+#endif
+ }
+ handleIncallSonification(stream, false, false, outputDesc->mIoHandle);
+ }
+
+ if (outputDesc->mRefCount[stream] > 0) {
+ // decrement usage count of this stream on the output
+ outputDesc->changeRefCount(stream, -1);
+
+ // store time at which the stream was stopped - see isStreamActive()
+ if (outputDesc->mRefCount[stream] == 0 || forceDeviceUpdate) {
+ outputDesc->mStopTime[stream] = systemTime();
+ audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
+ // delay the device switch by twice the latency because stopOutput() is executed when
+ // the track stop() command is received and at that time the audio track buffer can
+ // still contain data that needs to be drained. The latency only covers the audio HAL
+ // and kernel buffers. Also the latency does not always include additional delay in the
+ // audio path (audio DSP, CODEC ...)
+ setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2);
+
+ // force restoring the device selection on other active outputs if it differs from the
+ // one being selected for this output
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != outputDesc &&
+ desc->isActive() &&
+ outputDesc->sharesHwModuleWith(desc) &&
+ (newDevice != desc->device())) {
+ audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/);
+ setOutputDevice(desc,
+ dev,
+ true,
+ outputDesc->latency()*2);
+ }
+ }
+ // update the outputs if stopping one with a stream that can affect notification routing
+ handleNotificationRoutingForStream(stream);
+ }
+ return NO_ERROR;
+ } else {
+ ALOGW("stopOutput() refcount is already 0");
+ return INVALID_OPERATION;
+ }
+}
+status_t AudioPolicyManagerCustom::startSource(sp<AudioOutputDescriptor> outputDesc1,
+ audio_stream_type_t stream,
+ audio_devices_t device,
+ uint32_t *delayMs)
+{
+ // cannot start playback of STREAM_TTS if any other output is being used
+ uint32_t beaconMuteLatency = 0;
+ if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ ALOGW("startSource() invalid stream %d", stream);
+ return INVALID_OPERATION;
+ }
+
+#ifdef NON_WEARABLE_TARGET
+ sp<AudioOutputDescriptor> outputDesc = outputDesc1;
+#else
+ sp<SwAudioOutputDescriptor> outputDesc = (sp<SwAudioOutputDescriptor>) outputDesc1;
+#endif
+
+ *delayMs = 0;
+ if (stream == AUDIO_STREAM_TTS) {
+ ALOGV("\t found BEACON stream");
+ if (mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
+ return INVALID_OPERATION;
+ } else {
+ beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
+ }
+ } else {
+ // some playback other than beacon starts
+ beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
+ }
+
+ // increment usage count for this stream on the requested output:
+ // NOTE that the usage count is the same for duplicated output and hardware output which is
+ // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->changeRefCount(stream, 1);
+
+ if (outputDesc->mRefCount[stream] == 1 || device != AUDIO_DEVICE_NONE) {
+ // starting an output being rerouted?
+ if (device == AUDIO_DEVICE_NONE) {
+ device = getNewOutputDevice(outputDesc, false /*fromCache*/);
+ }
+ routing_strategy strategy = getStrategy(stream);
+ bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+ (strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
+ (beaconMuteLatency > 0);
+ uint32_t waitMs = beaconMuteLatency;
+ bool force = false;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != outputDesc) {
+ // force a device change if any other output is managed by the same hw
+ // module and has a current device selection that differs from selected device.
+ // In this case, the audio HAL must receive the new device selection so that it can
+ // change the device currently selected by the other active output.
+ if (outputDesc->sharesHwModuleWith(desc) &&
+ desc->device() != device) {
+ force = true;
+ }
+ // wait for audio on other active outputs to be presented when starting
+ // a notification so that audio focus effect can propagate, or that a mute/unmute
+ // event occurred for beacon
+ uint32_t latency = desc->latency();
+ if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+ waitMs = latency;
+ }
+ }
+ }
+ uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
+ }
+
+ // apply volume rules for current stream and device if necessary
+ checkAndSetVolume(stream,
+ mStreams.valueFor(stream).getVolumeIndex(device),
+ outputDesc,
+ device);
+
+ // update the outputs if starting an output with a stream that can affect notification
+ // routing
+ handleNotificationRoutingForStream(stream);
+
+ // force reevaluating accessibility routing when ringtone or alarm starts
+ if (strategy == STRATEGY_SONIFICATION) {
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+ }
+ else {
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
+ }
+ }
+ return NO_ERROR;
+}
+void AudioPolicyManagerCustom::handleIncallSonification(audio_stream_type_t stream,
+ bool starting, bool stateChange,
+ audio_io_handle_t output)
+{
+ if(!hasPrimaryOutput()) {
+ return;
+ }
+ // no action needed for AUDIO_STREAM_PATCH stream type, it's for internal flinger tracks
+ if (stream == AUDIO_STREAM_PATCH) {
+ return;
+ }
+ // if the stream pertains to sonification strategy and we are in call we must
+ // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+ // in the device used for phone strategy and play the tone if the selected device does not
+ // interfere with the device used for phone strategy
+ // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
+ // many times as there are active tracks on the output
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((stream_strategy == STRATEGY_SONIFICATION) ||
+ ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
+ stream, starting, outputDesc->mDevice, stateChange);
+ if (outputDesc->mRefCount[stream]) {
+ int muteCount = 1;
+ if (stateChange) {
+ muteCount = outputDesc->mRefCount[stream];
+ }
+ if (audio_is_low_visibility(stream)) {
+ ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, outputDesc);
+ }
+ } else {
+ ALOGV("handleIncallSonification() high visibility");
+ if (outputDesc->device() &
+ getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
+ ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, outputDesc);
+ }
+ }
+ if (starting) {
+ mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
+ AUDIO_STREAM_VOICE_CALL);
+ } else {
+ mpClientInterface->stopTone();
+ }
+ }
+ }
+ }
+}
+void AudioPolicyManagerCustom::handleNotificationRoutingForStream(audio_stream_type_t stream) {
+ switch(stream) {
+ case AUDIO_STREAM_MUSIC:
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ updateDevicesAndOutputs();
+ break;
+ default:
+ break;
+ }
+}
+#ifdef NON_WEARABLE_TARGET
+status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream,
+ int index,
+ const sp<AudioOutputDescriptor>& outputDesc,
+ audio_devices_t device,
+ int delayMs, bool force)
+#else
+status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream,
+ int index,
+ const sp<SwAudioOutputDescriptor>& outputDesc,
+ audio_devices_t device,
+ int delayMs, bool force)
+
+#endif
+{
+ if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ ALOGW("checkAndSetVolume() invalid stream %d", stream);
+ return INVALID_OPERATION;
+ }
+
+ // do not change actual stream volume if the stream is muted
+ if (outputDesc->mMuteCount[stream] != 0) {
+ ALOGVV("checkAndSetVolume() stream %d muted count %d",
+ stream, outputDesc->mMuteCount[stream]);
+ return NO_ERROR;
+ }
+ audio_policy_forced_cfg_t forceUseForComm =
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION);
+ // do not change in call volume if bluetooth is connected and vice versa
+ if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) {
+ ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+ stream, forceUseForComm);
+ return INVALID_OPERATION;
+ }
+
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ float volumeDb = computeVolume(stream, index, device);
+ if (outputDesc->isFixedVolume(device)) {
+ volumeDb = 0.0f;
+ }
+
+ outputDesc->setVolume(volumeDb, stream, device, delayMs, force);
+
+ if (stream == AUDIO_STREAM_VOICE_CALL ||
+ stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ float voiceVolume;
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (stream == AUDIO_STREAM_VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams.valueFor(stream).getVolumeIndexMax();
+ } else {
+ voiceVolume = 1.0;
+ }
+
+ if (voiceVolume != mLastVoiceVolume && ((outputDesc == mPrimaryOutput) ||
+ isDirectOutput(outputDesc->mIoHandle) || device & AUDIO_DEVICE_OUT_ALL_USB)) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ mLastVoiceVolume = voiceVolume;
+ }
+#ifdef FM_POWER_OPT
+ } else if (stream == AUDIO_STREAM_MUSIC && hasPrimaryOutput() &&
+ outputDesc == mPrimaryOutput) {
+ AudioParameter param = AudioParameter();
+ param.addFloat(String8("fm_volume"), Volume::DbToAmpl(volumeDb));
+ mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString(), delayMs);
+#endif /* FM_POWER_OPT end */
+ }
+
+ return NO_ERROR;
+}
+bool AudioPolicyManagerCustom::isDirectOutput(audio_io_handle_t output) {
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
+ return true;
+ }
+ }
+ return false;
+}
+audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice(
+ audio_devices_t device,
+ audio_session_t session __unused,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ uint32_t latency = 0;
+ status_t status;
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ ALOGV("getOutput() opening test output");
+ sp<AudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(NULL,
+ mpClientInterface);
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags =
+ (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = mTestSamplingRate;
+ config.channel_mask = mTestChannels;
+ config.format = mTestFormat;
+ if (offloadInfo != NULL) {
+ config.offload_info = *offloadInfo;
+ }
+ status = mpClientInterface->openOutput(0,
+ &mTestOutputs[mCurOutput],
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (status == NO_ERROR) {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mFormat = config.format;
+ outputDesc->mChannelMask = config.channel_mask;
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"),mCurOutput);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+ addOutput(mTestOutputs[mCurOutput], outputDesc);
+ }
+ }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+ if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
+ (stream != AUDIO_STREAM_MUSIC)) {
+ // compress should not be used for non-music streams
+ ALOGE("Offloading only allowed with music stream");
+ return 0;
+ }
+
+#ifdef COMPRESS_VOIP_ENABLED
+ if ((stream == AUDIO_STREAM_VOICE_CALL) &&
+ (channelMask == 1) &&
+ (samplingRate == 8000 || samplingRate == 16000)) {
+ // Allow Voip direct output only if:
+ // audio mode is MODE_IN_COMMUNCATION; AND
+ // voip output is not opened already; AND
+ // requested sample rate matches with that of voip input stream (if opened already)
+ int value = 0;
+ uint32_t mode = 0, voipOutCount = 1, voipSampleRate = 1;
+ String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+ String8("audio_mode"));
+ AudioParameter result = AudioParameter(valueStr);
+ if (result.getInt(String8("audio_mode"), value) == NO_ERROR) {
+ mode = value;
+ }
+
+ valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+ String8("voip_out_stream_count"));
+ result = AudioParameter(valueStr);
+ if (result.getInt(String8("voip_out_stream_count"), value) == NO_ERROR) {
+ voipOutCount = value;
+ }
+
+ valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+ String8("voip_sample_rate"));
+ result = AudioParameter(valueStr);
+ if (result.getInt(String8("voip_sample_rate"), value) == NO_ERROR) {
+ voipSampleRate = value;
+ }
+
+ if ((mode == AUDIO_MODE_IN_COMMUNICATION) && (voipOutCount == 0) &&
+ ((voipSampleRate == 0) || (voipSampleRate == samplingRate))) {
+ if (audio_is_linear_pcm(format)) {
+ char propValue[PROPERTY_VALUE_MAX] = {0};
+ property_get("use.voice.path.for.pcm.voip", propValue, "0");
+ bool voipPcmSysPropEnabled = !strncmp("true", propValue, sizeof("true"));
+ if (voipPcmSysPropEnabled && (format == AUDIO_FORMAT_PCM_16_BIT)) {
+ flags = (audio_output_flags_t)((flags &~AUDIO_OUTPUT_FLAG_FAST) |
+ AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT);
+ ALOGD("Set VoIP and Direct output flags for PCM format");
+ }
+ }
+ }
+ }
+#endif
+
+#ifdef VOICE_CONCURRENCY
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_play_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
+ prop_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if (prop_play_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+ ((AUDIO_MODE_IN_CALL == mPrevPhoneState)
+ && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+ {
+ if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
+ if(prop_voip_enabled) {
+ ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
+ flags );
+ return 0;
+ }
+ }
+ else {
+ if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+ ALOGD("voice_conc:IN call mode adding ULL flags .. flags: %x ", flags );
+ flags = AUDIO_OUTPUT_FLAG_FAST;
+ } else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+ if (AUDIO_STREAM_MUSIC == stream) {
+ flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ ALOGD("voice_conc:IN call mode adding deep-buffer flags %x ", flags );
+ }
+ else {
+ flags = AUDIO_OUTPUT_FLAG_FAST;
+ ALOGD("voice_conc:IN call mode adding fast flags %x ", flags );
+ }
+ }
+ }
+ }
+ } else if (prop_voip_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //return only ULL ouput
+ if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+ ((AUDIO_MODE_IN_CALL == mPrevPhoneState)
+ && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+ {
+ if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
+ ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
+ flags );
+ return 0;
+ }
+ }
+ }
+#endif
+#ifdef RECORD_PLAY_CONCURRENCY
+ char recConcPropValue[PROPERTY_VALUE_MAX];
+ bool prop_rec_play_enabled = false;
+
+ if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
+ prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
+ }
+ if ((prop_rec_play_enabled) &&
+ ((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCount() > 0))) {
+ if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
+ if (AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
+ // allow VoIP using voice path
+ // Do nothing
+ } else if((flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+ ALOGD("voice_conc:MODE_IN_COMM is setforcing deep buffer output for non ULL... flags: %x", flags);
+ // use deep buffer path for all non ULL outputs
+ flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ }
+ } else if ((flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+ ALOGD("voice_conc:Record mode is on forcing deep buffer output for non ULL... flags: %x ", flags);
+ // use deep buffer path for all non ULL outputs
+ flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ }
+ }
+ if (prop_rec_play_enabled &&
+ (stream == AUDIO_STREAM_ENFORCED_AUDIBLE)) {
+ ALOGD("Record conc is on forcing ULL output for ENFORCED_AUDIBLE");
+ flags = AUDIO_OUTPUT_FLAG_FAST;
+ }
+#endif
+#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
+ /*
+ * WFD audio routes back to target speaker when starting a ringtone playback.
+ * This is because primary output is reused for ringtone, so output device is
+ * updated based on SONIFICATION strategy for both ringtone and music playback.
+ * The same issue is not seen on remoted_submix HAL based WFD audio because
+ * primary output is not reused and a new output is created for ringtone playback.
+ * Issue is fixed by updating output flag to AUDIO_OUTPUT_FLAG_FAST when there is
+ * a non-music stream playback on WFD, so primary output is not reused for ringtone.
+ */
+ audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
+ if ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY)
+ && (stream != AUDIO_STREAM_MUSIC)) {
+ ALOGD("WFD audio: use OUTPUT_FLAG_FAST for non music stream. flags:%x", flags );
+ //For voip paths
+ if(flags & AUDIO_OUTPUT_FLAG_DIRECT)
+ flags = AUDIO_OUTPUT_FLAG_DIRECT;
+ else //route every thing else to ULL path
+ flags = AUDIO_OUTPUT_FLAG_FAST;
+ }
+#endif
+ // open a direct output if required by specified parameters
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ // only allow deep buffering for music stream type
+ if (stream != AUDIO_STREAM_MUSIC) {
+ flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ }
+ if (stream == AUDIO_STREAM_TTS) {
+ flags = AUDIO_OUTPUT_FLAG_TTS;
+ }
+
+ // open a direct output if required by specified parameters
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ // only allow deep buffering for music stream type
+ if (stream != AUDIO_STREAM_MUSIC) {
+ flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ }
+ if (stream == AUDIO_STREAM_TTS) {
+ flags = AUDIO_OUTPUT_FLAG_TTS;
+ }
+
+ sp<IOProfile> profile;
+
+ // skip direct output selection if the request can obviously be attached to a mixed output
+ // and not explicitly requested
+ if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
+ audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE &&
+ audio_channel_count_from_out_mask(channelMask) <= 2) {
+ goto non_direct_output;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+
+ if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+ !mEffects.isNonOffloadableEffectEnabled()) {
+ profile = getProfileForDirectOutput(device,
+ samplingRate,
+ format,
+ channelMask,
+ (audio_output_flags_t)flags);
+ }
+
+ if (profile != 0) {
+ sp<SwAudioOutputDescriptor> outputDesc = NULL;
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open and configured with same parameters
+ if ((samplingRate == outputDesc->mSamplingRate) &&
+ (format == outputDesc->mFormat) &&
+ (channelMask == outputDesc->mChannelMask)) {
+ outputDesc->mDirectOpenCount++;
+ ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mIoHandle);
+ }
+
+ // if the selected profile is offloaded and no offload info was specified,
+ // create a default one
+ audio_offload_info_t defaultOffloadInfo = AUDIO_INFO_INITIALIZER;
+ if ((profile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ defaultOffloadInfo.sample_rate = samplingRate;
+ defaultOffloadInfo.channel_mask = channelMask;
+ defaultOffloadInfo.format = format;
+ defaultOffloadInfo.stream_type = stream;
+ defaultOffloadInfo.bit_rate = 0;
+ defaultOffloadInfo.duration_us = -1;
+ defaultOffloadInfo.has_video = true; // conservative
+ defaultOffloadInfo.is_streaming = true; // likely
+ offloadInfo = &defaultOffloadInfo;
+ }
+
+ outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface);
+ outputDesc->mDevice = device;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags = (audio_output_flags_t)(outputDesc->mFlags | flags);
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = samplingRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+ if (offloadInfo != NULL) {
+ config.offload_info = *offloadInfo;
+ }
+ status = mpClientInterface->openOutput(profile->getModuleHandle(),
+ &output,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ // only accept an output with the requested parameters
+ if (status != NO_ERROR ||
+ (samplingRate != 0 && samplingRate != config.sample_rate) ||
+ (format != AUDIO_FORMAT_DEFAULT && format != config.format) ||
+ (channelMask != 0 && channelMask != config.channel_mask)) {
+ ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+ "format %d %d, channelMask %04x %04x", output, samplingRate,
+ outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+ outputDesc->mChannelMask);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ mpClientInterface->closeOutput(output);
+ }
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
+ return AUDIO_IO_HANDLE_NONE;
+ }
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+ outputDesc->mRefCount[stream] = 0;
+ outputDesc->mStopTime[stream] = 0;
+ outputDesc->mDirectOpenCount = 1;
+
+ audio_io_handle_t srcOutput = getOutputForEffect();
+ addOutput(output, outputDesc);
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput == output) {
+#ifdef DOLBY_ENABLE
+ status_t status = mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+ if (status == NO_ERROR) {
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ sp<EffectDescriptor> desc = mEffects.valueAt(i);
+ if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX) {
+ // update the mIo member of EffectDescriptor for the global effect
+ ALOGV("%s updating mIo", __FUNCTION__);
+ desc->mIo = dstOutput;
+ }
+ }
+ } else {
+ ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__, srcOutput, dstOutput);
+ }
+#else // DOLBY_END
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+#endif // LINE_ADDED_BY_DOLBY
+ }
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutput() returns new direct output %d", output);
+ mpClientInterface->onAudioPortListUpdate();
+ return output;
+ }
+
+non_direct_output:
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // for non direct outputs, only PCM is supported
+ if (audio_is_linear_pcm(format)) {
+ // get which output is suitable for the specified stream. The actual
+ // routing change will happen when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+ // at this stage we should ignore the DIRECT flag as no direct output could be found earlier
+ flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+ output = selectOutput(outputs, flags, format);
+ }
+ ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+ "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+ ALOGV(" getOutputForDevice() returns output %d", output);
+
+ return output;
+}
+
+status_t AudioPolicyManagerCustom::getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uid_t uid,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ audio_port_handle_t selectedDeviceId,
+ input_type_t *inputType)
+{
+ audio_source_t inputSource = attr->source;
+#ifdef VOICE_CONCURRENCY
+
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_rec_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.record.conc.disabled", propValue, NULL)) {
+ prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if (prop_rec_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //Need to block input request
+ if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+ ((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
+ (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+ {
+ switch(inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ ALOGD("voice_conc:Creating input during incall mode for inputSource: %d",
+ inputSource);
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if(prop_voip_enabled) {
+ ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
+ inputSource);
+ return NO_INIT;
+ }
+ break;
+ default:
+ ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
+ inputSource);
+ return NO_INIT;
+ }
+ }
+ }//check for VoIP flag
+ else if(prop_voip_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //Need to block input request
+ if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+ ((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
+ (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+ {
+ if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+ ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
+ return NO_INIT;
+ }
+ }
+ }
+
+#endif
+
+ return AudioPolicyManager::getInputForAttr(attr,
+ input,
+ session,
+ uid,
+ samplingRate,
+ format,
+ channelMask,
+ flags,
+ selectedDeviceId,
+ inputType);
+}
+status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("startInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("startInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("startInput() unknown session %d on input %d", session, input);
+ return BAD_VALUE;
+ }
+
+ // virtual input devices are compatible with other input devices
+ if (!is_virtual_input_device(inputDesc->mDevice)) {
+
+ // for a non-virtual input device, check if there is another (non-virtual) active input
+ audio_io_handle_t activeInput = mInputs.getActiveInput();
+ if (activeInput != 0 && activeInput != input) {
+
+ // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
+ // otherwise the active input continues and the new input cannot be started.
+ sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
+ if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
+ ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
+ stopInput(activeInput, activeDesc->mSessions.itemAt(0));
+ releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
+ } else {
+ ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
+ return INVALID_OPERATION;
+ }
+ }
+ }
+
+ // Routing?
+ mInputRoutes.incRouteActivity(session);
+#ifdef RECORD_PLAY_CONCURRENCY
+ mIsInputRequestOnProgress = true;
+
+ char getPropValue[PROPERTY_VALUE_MAX];
+ bool prop_rec_play_enabled = false;
+
+ if (property_get("rec.playback.conc.disabled", getPropValue, NULL)) {
+ prop_rec_play_enabled = atoi(getPropValue) || !strncmp("true", getPropValue, 4);
+ }
+
+ if ((prop_rec_play_enabled) &&(mInputs.activeInputsCount() == 0)){
+ // send update to HAL on record playback concurrency
+ AudioParameter param = AudioParameter();
+ param.add(String8("rec_play_conc_on"), String8("true"));
+ ALOGD("startInput() setParameters rec_play_conc is setting to ON ");
+ mpClientInterface->setParameters(0, param.toString());
+
+ // Call invalidate to reset all opened non ULL audio tracks
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
+ // Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder)
+ if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE && (i != AUDIO_STREAM_PATCH)) {
+ ALOGD("Invalidate on releaseInput for stream :: %d ", i);
+ //FIXME see fixme on name change
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ // close compress tracks
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("ouput desc / profile is NULL");
+ continue;
+ }
+ if (outputDesc->mProfile->mFlags
+ & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ // close compress sessions
+ ALOGD("calling closeOutput on record conc for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ }
+#endif
+
+ if (inputDesc->mRefCount == 0 || mInputRoutes.hasRouteChanged(session)) {
+ // if input maps to a dynamic policy with an activity listener, notify of state change
+ if ((inputDesc->mPolicyMix != NULL)
+ && ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
+ mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mRegistrationId,
+ MIX_STATE_MIXING);
+ }
+
+ if (mInputs.activeInputsCount() == 0) {
+ SoundTrigger::setCaptureState(true);
+ }
+ setInputDevice(input, getNewInputDevice(input), true /* force */);
+
+ // automatically enable the remote submix output when input is started if not
+ // used by a policy mix of type MIX_TYPE_RECORDERS
+ // For remote submix (a virtual device), we open only one input per capture request.
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ String8 address = String8("");
+ if (inputDesc->mPolicyMix == NULL) {
+ address = String8("0");
+ } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
+ address = inputDesc->mPolicyMix->mRegistrationId;
+ }
+ if (address != "") {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address, "remote-submix");
+ }
+ }
+ }
+
+ ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
+
+ inputDesc->mRefCount++;
+#ifdef RECORD_PLAY_CONCURRENCY
+ mIsInputRequestOnProgress = false;
+#endif
+ return NO_ERROR;
+}
+status_t AudioPolicyManagerCustom::stopInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ status_t status;
+ status = AudioPolicyManager::stopInput(input, session);
+#ifdef RECORD_PLAY_CONCURRENCY
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_rec_play_enabled = false;
+
+ if (property_get("rec.playback.conc.disabled", propValue, NULL)) {
+ prop_rec_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if ((prop_rec_play_enabled) && (mInputs.activeInputsCount() == 0)) {
+
+ //send update to HAL on record playback concurrency
+ AudioParameter param = AudioParameter();
+ param.add(String8("rec_play_conc_on"), String8("false"));
+ ALOGD("stopInput() setParameters rec_play_conc is setting to OFF ");
+ mpClientInterface->setParameters(0, param.toString());
+
+ //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
+ for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
+ //Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder stop tone)
+ if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE) && (i != AUDIO_STREAM_PATCH)) {
+ ALOGD(" Invalidate on stopInput for stream :: %d ", i);
+ //FIXME see fixme on name change
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+#endif
+ return status;
+}
+
+AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
+ : AudioPolicyManager(clientInterface)
+{
+#ifdef RECORD_PLAY_CONCURRENCY
+ mIsInputRequestOnProgress = false;
+#endif
+
+
+#ifdef VOICE_CONCURRENCY
+ mFallBackflag = getFallBackPath();
+#endif
+}
+}
diff --git a/msm8909/policy_hal/AudioPolicyManager.h b/msm8909/policy_hal/AudioPolicyManager.h
new file mode 100644
index 0000000..64a9a01
--- /dev/null
+++ b/msm8909/policy_hal/AudioPolicyManager.h
@@ -0,0 +1,164 @@
+/*
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ * Not a contribution.
+ *
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#include <audiopolicy/managerdefault/AudioPolicyManager.h>
+#include <audio_policy_conf.h>
+#include <Volume.h>
+
+
+namespace android {
+#ifndef FLAC_OFFLOAD_ENABLED
+#define AUDIO_FORMAT_FLAC 0x1D000000UL
+#endif
+
+#ifndef WMA_OFFLOAD_ENABLED
+#define AUDIO_FORMAT_WMA 0x13000000UL
+#define AUDIO_FORMAT_WMA_PRO 0x14000000UL
+#endif
+
+#ifndef ALAC_OFFLOAD_ENABLED
+#define AUDIO_FORMAT_ALAC 0x1F000000UL
+#endif
+
+#ifndef APE_OFFLOAD_ENABLED
+#define AUDIO_FORMAT_APE 0x20000000UL
+#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,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ audio_port_handle_t selectedDeviceId,
+ input_type_t *inputType);
+ // 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);
+ // 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);
+
+protected:
+
+#ifdef NON_WEARABLE_TARGET
+ status_t checkAndSetVolume(audio_stream_type_t stream,
+ int index,
+ const sp<AudioOutputDescriptor>& outputDesc,
+ audio_devices_t device,
+ int delayMs = 0, bool force = false);
+#else
+ status_t checkAndSetVolume(audio_stream_type_t stream,
+ int index,
+ const sp<SwAudioOutputDescriptor>& outputDesc,
+ audio_devices_t device,
+ int delayMs = 0, bool force = false);
+#endif
+
+ // selects the most appropriate device on output for current state
+ // must be called every time a condition that affects the device choice for a given output is
+ // changed: connected device, phone state, force use, output start, output stop..
+ // see getDeviceForStrategy() for the use of fromCache parameter
+ audio_devices_t getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
+ bool fromCache);
+ // 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(sp<AudioOutputDescriptor> outputDesc,
+ audio_stream_type_t stream,
+ audio_devices_t device,
+ uint32_t *delayMs);
+ status_t stopSource(sp<AudioOutputDescriptor> outputDesc,
+ audio_stream_type_t stream,
+ bool forceDeviceUpdate);
+ // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON 313
+ // returns 0 if no mute/unmute event happened, the largest latency of the device where 314
+ // the mute/unmute happened 315
+ 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*/
+
+ // 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:
+ static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi);
+ // updates device caching and output for streams that can influence the
+ // routing of notifications
+ void handleNotificationRoutingForStream(audio_stream_type_t stream);
+ static bool isVirtualInputDevice(audio_devices_t device);
+ static bool deviceDistinguishesOnAddress(audio_devices_t device);
+ uint32_t nextUniqueId();
+ // 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);
+ // Used for voip + voice concurrency usecase
+ int mPrevPhoneState;
+ int mvoice_call_state;
+#ifdef RECORD_PLAY_CONCURRENCY
+ // Used for record + playback concurrency
+ bool mIsInputRequestOnProgress;
+#endif
+
+
+};
+
+};