Revert "Revert "zenfone6: add back policy hal""
This reverts commit 3c27cbb94316e88860c92e27285cefa13b223d8a.
Change-Id: I7632f910dd50b793de5f8a2f2ff4a660127a11b2
diff --git a/BoardConfig.mk b/BoardConfig.mk
index f6e68d6..7a91514 100755
--- a/BoardConfig.mk
+++ b/BoardConfig.mk
@@ -57,6 +57,7 @@
DEXPREOPT_GENERATE_APEX_IMAGE := true
# Audio
+USE_CUSTOM_AUDIO_POLICY := 1
USE_XML_AUDIO_POLICY_CONF := 1
# Bluetooth
diff --git a/policy_hal/APMConfigHelper.cpp b/policy_hal/APMConfigHelper.cpp
new file mode 100644
index 0000000..1815e77
--- /dev/null
+++ b/policy_hal/APMConfigHelper.cpp
@@ -0,0 +1,320 @@
+/*
+ * Copyright (c) 2019, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "APMConfigHelper"
+
+#include "APMConfigHelper.h"
+#include <cutils/properties.h>
+#include <log/log.h>
+#include <stdlib.h>
+
+namespace android {
+
+void APMConfigHelper::retrieveConfigs()
+{
+#ifdef AHAL_EXT_ENABLED
+ if (isRemote) {
+ ALOGV("Already retrieved configs from remote");
+ return;
+ }
+
+ ALOGV("Try retrieving configs from remote");
+ mConfigs = getApmConfigs< IAudioHalExt,
+ &IAudioHalExt::getApmConfigs>(mConfigs, isRemote);
+#endif /* AHAL_EXT_ENABLED */
+
+ // apmconfig struct dump
+ ALOGV("apmconfigs retrieved from %s", isRemote? "remote" : "local");
+ ALOGV("audio_offload_video: %d", mConfigs.audio_offload_video);
+ ALOGV("audio_offload_disable: %d", mConfigs.audio_offload_disable);
+ ALOGV("audio_deepbuffer_media: %d", mConfigs.audio_deepbuffer_media);
+ ALOGV("audio_av_streaming_offload_enable: %d", mConfigs.audio_av_streaming_offload_enable);
+ ALOGV("audio_offload_track_enable: %d", mConfigs.audio_offload_track_enable);
+ ALOGV("audio_offload_multiple_enabled: %d", mConfigs.audio_offload_multiple_enabled);
+ ALOGV("voice_dsd_playback_conc_disabled: %d", mConfigs.voice_dsd_playback_conc_disabled);
+ ALOGV("audio_sva_conc_enabled: %d", mConfigs.audio_sva_conc_enabled);
+ ALOGV("audio_va_concurrency_enabled: %d", mConfigs.audio_va_concurrency_enabled);
+ ALOGV("audio_rec_playback_conc_disabled: %d", mConfigs.audio_rec_playback_conc_disabled);
+ ALOGV("voice_path_for_pcm_voip: %d", mConfigs.voice_path_for_pcm_voip);
+ ALOGV("voice_playback_conc_disabled: %d", mConfigs.voice_playback_conc_disabled);
+ ALOGV("voice_record_conc_disabled: %d", mConfigs.voice_record_conc_disabled);
+ ALOGV("voice_voip_conc_disabled: %d", mConfigs.voice_voip_conc_disabled);
+ ALOGV("audio_offload_min_duration_secs: %u", mConfigs.audio_offload_min_duration_secs);
+ ALOGV("voice_conc_fallbackpath: %s", mConfigs.voice_conc_fallbackpath.c_str());
+ ALOGV("audio_extn_hdmi_spk_enabled: %d", mConfigs.audio_extn_hdmi_spk_enabled);
+ ALOGV("audio_extn_formats_enabled: %d", mConfigs.audio_extn_formats_enabled);
+ ALOGV("audio_extn_afe_proxy_enabled: %d", mConfigs.audio_extn_afe_proxy_enabled);
+ ALOGV("compress_voip_enabled: %d", mConfigs.compress_voip_enabled);
+ ALOGV("fm_power_opt: %d", mConfigs.fm_power_opt);
+ ALOGV("voice_concurrency: %d", mConfigs.voice_concurrency);
+ ALOGV("record_play_concurrency: %d", mConfigs.record_play_concurrency);
+ ALOGV("use_xml_audio_policy_conf: %d", mConfigs.use_xml_audio_policy_conf);
+ return;
+}
+
+APMConfigHelper::APMConfigHelper()
+ : isRemote(false)
+{
+ uint32_t audio_offload_min_duration_secs = 0;
+ char voice_conc_fallbackpath[PROPERTY_VALUE_MAX] = "\0";
+
+ audio_offload_min_duration_secs =
+ (uint32_t)property_get_int32("audio.offload.min.duration.secs", 0);
+ property_get("vendor.voice.conc.fallbackpath", voice_conc_fallbackpath, NULL);
+
+ mConfigs = {
+ (bool) property_get_bool("audio.offload.video", true),
+ (bool) property_get_bool("audio.offload.disable", false),
+ (bool) property_get_bool("audio.deep_buffer.media", true),
+ (bool) property_get_bool("vendor.audio.av.streaming.offload.enable", false),
+ (bool) property_get_bool("vendor.audio.offload.track.enable", true),
+ (bool) property_get_bool("vendor.audio.offload.multiple.enabled", false),
+ (bool) property_get_bool("vendor.voice.dsd.playback.conc.disabled", true),
+ (bool) property_get_bool("persist.vendor.audio.sva.conc.enabled", false),
+ (bool) property_get_bool("persist.vendor.audio.va_concurrency_enabled", false),
+ (bool) property_get_bool("vendor.audio.rec.playback.conc.disabled", false),
+ (bool) property_get_bool("vendor.voice.path.for.pcm.voip", true),
+ (bool) property_get_bool("vendor.voice.playback.conc.disabled", false),
+ (bool) property_get_bool("vendor.voice.record.conc.disabled", false),
+ (bool) property_get_bool("vendor.voice.voip.conc.disabled", false),
+ audio_offload_min_duration_secs,
+ voice_conc_fallbackpath,
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ true,
+#else
+ false,
+#endif
+#ifdef AUDIO_EXTN_FORMATS_ENABLED
+ true,
+#else
+ false,
+#endif
+#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
+ true,
+#else
+ false,
+#endif
+#ifdef COMPRESS_VOIP_ENABLED
+ true,
+#else
+ false,
+#endif
+#ifdef FM_POWER_OPT
+ true,
+#else
+ false,
+#endif
+#ifdef VOICE_CONCURRENCY
+ true,
+#else
+ false,
+#endif
+#ifdef RECORD_PLAY_CONCURRENCY
+ true,
+#else
+ false,
+#endif
+#ifdef USE_XML_AUDIO_POLICY_CONF
+ true,
+#else
+ false,
+#endif
+ };
+}
+
+bool APMConfigHelper::isAudioOffloadVideoEnabled()
+{
+ ALOGV("isAudioOffloadVideoEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_offload_video;
+}
+
+bool APMConfigHelper::isAudioOffloadDisabled()
+{
+ ALOGV("isAudioOffloadDisabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_offload_disable;
+}
+
+bool APMConfigHelper::isAudioDeepbufferMediaEnabled()
+{
+ ALOGV("isAudioDeepbufferMediaEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_deepbuffer_media;
+}
+
+bool APMConfigHelper::isAVStreamingOffloadEnabled()
+{
+ ALOGV("isAVStreamingOffloadEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_av_streaming_offload_enable;
+}
+
+bool APMConfigHelper::isAudioTrackOffloadEnabled()
+{
+ ALOGV("isAudioTrackOffloadEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_offload_track_enable;
+}
+
+bool APMConfigHelper::isAudioMultipleOffloadEnable()
+{
+ ALOGV("isAudioMultipleOffloadEnable enter");
+ retrieveConfigs();
+ return mConfigs.audio_offload_multiple_enabled;
+}
+
+bool APMConfigHelper::isVoiceDSDConcDisabled()
+{
+ ALOGV("isVoiceDSDConcDisabled enter");
+ retrieveConfigs();
+ return mConfigs.voice_dsd_playback_conc_disabled;
+}
+
+bool APMConfigHelper::isSVAConcEnabled()
+{
+ ALOGV("isSVAConcEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_sva_conc_enabled;
+}
+
+bool APMConfigHelper::isVAConcEnabled()
+{
+ ALOGV("isVAConcEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_va_concurrency_enabled;
+}
+
+bool APMConfigHelper::isRecPlayConcDisabled()
+{
+ ALOGV("isRecPlayConcDisabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_rec_playback_conc_disabled;
+}
+
+bool APMConfigHelper::useVoicePathForPCMVOIP()
+{
+ ALOGV("useVoicePathForPCMVOIP enter");
+ retrieveConfigs();
+ return mConfigs.voice_path_for_pcm_voip;
+}
+
+bool APMConfigHelper::isVoicePlayConcDisabled()
+{
+ ALOGV("isVoicePlayConcDisabled enter");
+ retrieveConfigs();
+ return mConfigs.voice_playback_conc_disabled;
+}
+
+bool APMConfigHelper::isVoiceRecConcDisabled()
+{
+ ALOGV("isVoiceRecConcDisabled enter");
+ retrieveConfigs();
+ return mConfigs.voice_record_conc_disabled;
+}
+
+bool APMConfigHelper::isVoiceVOIPConcDisabled()
+{
+ ALOGV("isVoiceVOIPConcDisabled enter");
+ retrieveConfigs();
+ return mConfigs.voice_voip_conc_disabled;
+}
+
+uint32_t APMConfigHelper::getAudioOffloadMinDuration()
+{
+ ALOGV("getAudioOffloadMinDuration enter");
+ retrieveConfigs();
+ return mConfigs.audio_offload_min_duration_secs;
+}
+
+string APMConfigHelper::getVoiceConcFallbackPath()
+{
+ ALOGV("getVoiceConcFallbackPath enter");
+ retrieveConfigs();
+ return mConfigs.voice_conc_fallbackpath;
+}
+
+bool APMConfigHelper::isHDMISpkEnabled()
+{
+ ALOGV("isHDMISpkEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_extn_hdmi_spk_enabled;
+}
+
+bool APMConfigHelper::isExtnFormatsEnabled()
+{
+ ALOGV("isExtnFormatsEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_extn_formats_enabled;
+}
+
+bool APMConfigHelper::isAFEProxyEnabled()
+{
+ ALOGV("isAFEProxyEnabled enter");
+ retrieveConfigs();
+ return mConfigs.audio_extn_afe_proxy_enabled;
+}
+
+bool APMConfigHelper::isCompressVOIPEnabled()
+{
+ ALOGV("isCompressVOIPEnabled enter");
+ retrieveConfigs();
+ return mConfigs.compress_voip_enabled;
+}
+
+bool APMConfigHelper::isFMPowerOptEnabled()
+{
+ ALOGV("isFMPowerOptEnabled enter");
+ retrieveConfigs();
+ return mConfigs.fm_power_opt;
+}
+
+bool APMConfigHelper::isVoiceConcEnabled()
+{
+ ALOGV("isVoiceConcEnabled enter");
+ retrieveConfigs();
+ return mConfigs.voice_concurrency;
+}
+
+bool APMConfigHelper::isRecPlayConcEnabled()
+{
+ ALOGV("isRecPlayConcEnabled enter");
+ retrieveConfigs();
+ return mConfigs.record_play_concurrency;
+}
+
+bool APMConfigHelper::useXMLAudioPolicyConf()
+{
+ ALOGV("useXMLAudioPolicyConf enter");
+ retrieveConfigs();
+ return mConfigs.use_xml_audio_policy_conf;
+}
+
+}
diff --git a/policy_hal/APMConfigHelper.h b/policy_hal/APMConfigHelper.h
new file mode 100644
index 0000000..137f2d7
--- /dev/null
+++ b/policy_hal/APMConfigHelper.h
@@ -0,0 +1,124 @@
+/*
+ * Copyright (c) 2019, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef _APM_CONFIG_HELPER_H_
+#define _APM_CONFIG_HELPER_H_
+
+#include <string>
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/RefBase.h>
+
+#ifdef AHAL_EXT_ENABLED
+#include <vendor/qti/hardware/audiohalext/1.0/IAudioHalExt.h>
+#include <vendor/qti/hardware/audiohalext/1.0/types.h>
+#include <configstore/Utils.h>
+#endif
+
+namespace android {
+
+using namespace std;
+
+#ifdef AHAL_EXT_ENABLED
+using namespace vendor::qti::hardware::audiohalext::V1_0;
+using namespace vendor::qti::hardware::audiohalext;
+#else
+struct ApmValues {
+ bool audio_offload_video;
+ bool audio_offload_disable;
+ bool audio_deepbuffer_media;
+ bool audio_av_streaming_offload_enable;
+ bool audio_offload_track_enable;
+ bool audio_offload_multiple_enabled;
+ bool voice_dsd_playback_conc_disabled;
+ bool audio_sva_conc_enabled;
+ bool audio_va_concurrency_enabled;
+ bool audio_rec_playback_conc_disabled;
+ bool voice_path_for_pcm_voip;
+ bool voice_playback_conc_disabled;
+ bool voice_record_conc_disabled;
+ bool voice_voip_conc_disabled;
+ uint32_t audio_offload_min_duration_secs;
+ string voice_conc_fallbackpath;
+ bool audio_extn_hdmi_spk_enabled;
+ bool audio_extn_formats_enabled;
+ bool audio_extn_afe_proxy_enabled;
+ bool compress_voip_enabled;
+ bool fm_power_opt;
+ bool voice_concurrency;
+ bool record_play_concurrency;
+ bool use_xml_audio_policy_conf;
+};
+#endif
+
+class APMConfigHelper : public RefBase {
+public:
+ APMConfigHelper();
+
+ virtual ~APMConfigHelper() {};
+
+ /* member functions to query settigns */
+ bool isAudioOffloadVideoEnabled();
+ bool isAudioOffloadDisabled();
+ bool isAudioDeepbufferMediaEnabled();
+ bool isAVStreamingOffloadEnabled();
+ bool isAudioTrackOffloadEnabled();
+ bool isAudioMultipleOffloadEnable();
+ bool isVoiceDSDConcDisabled();
+ bool isSVAConcEnabled();
+ bool isVAConcEnabled();
+ bool isRecPlayConcDisabled();
+ bool useVoicePathForPCMVOIP();
+ bool isVoicePlayConcDisabled();
+ bool isVoiceRecConcDisabled();
+ bool isVoiceVOIPConcDisabled();
+
+ uint32_t getAudioOffloadMinDuration();
+ string getVoiceConcFallbackPath();
+
+ bool isHDMISpkEnabled();
+ bool isExtnFormatsEnabled();
+ bool isAFEProxyEnabled();
+ bool isCompressVOIPEnabled();
+ bool isFMPowerOptEnabled();
+ bool isVoiceConcEnabled();
+ bool isRecPlayConcEnabled();
+ bool useXMLAudioPolicyConf();
+
+private:
+ inline void retrieveConfigs();
+
+ ApmValues mConfigs;
+ bool isRemote; // configs specified from remote
+
+ DISALLOW_EVIL_CONSTRUCTORS(APMConfigHelper);
+};
+
+}
+
+#endif /* _APM_CONFIG_HELPER_H_ */
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
new file mode 100644
index 0000000..8df84b9
--- /dev/null
+++ b/policy_hal/Android.mk
@@ -0,0 +1,83 @@
+ifeq ($(TARGET_DEVICE),$(filter $(TARGET_DEVICE),zenfone6))
+ifneq ($(USE_LEGACY_AUDIO_POLICY), 1)
+ifeq ($(USE_CUSTOM_AUDIO_POLICY), 1)
+LOCAL_PATH := $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := AudioPolicyManager.cpp \
+ APMConfigHelper.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 \
+ $(call include-path-for, avextension) \
+
+LOCAL_HEADER_LIBRARIES := \
+ libbase_headers \
+ libstagefright_foundation_headers
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libutils \
+ liblog \
+ libsoundtrigger \
+ libaudiopolicymanagerdefault
+
+LOCAL_STATIC_LIBRARIES := \
+ libmedia_helper \
+
+LOCAL_CFLAGS += -Wall -Werror
+
+
+ifeq ($(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 $(AUDIO_FEATURE_ENABLED_EXTN_FORMATS)),true)
+ LOCAL_CFLAGS += -DAUDIO_EXTN_FORMATS_ENABLED
+endif
+
+ifeq ($(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 $(AUDIO_FEATURE_ENABLED_FM_POWER_OPT)),true)
+ LOCAL_CFLAGS += -DFM_POWER_OPT
+endif
+
+ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
+ LOCAL_CFLAGS += -DUSE_XML_AUDIO_POLICY_CONF
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_COMPRESS_VOIP)),true)
+ LOCAL_CFLAGS += -DCOMPRESS_VOIP_ENABLED
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_AHAL_EXT)), true)
+ LOCAL_CFLAGS += -DAHAL_EXT_ENABLED
+ LOCAL_SHARED_LIBRARIES += libhidlbase
+ LOCAL_SHARED_LIBRARIES += libhidltransport
+ LOCAL_SHARED_LIBRARIES += vendor.qti.hardware.audiohalext@1.0
+ LOCAL_SHARED_LIBRARIES += vendor.qti.hardware.audiohalext-utils
+endif
+
+LOCAL_MODULE := libaudiopolicymanager
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif
+endif
+endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
new file mode 100755
index 0000000..830ebe1
--- /dev/null
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -0,0 +1,2149 @@
+/*
+ * Copyright (c) 2013-2019 The Linux Foundation. All rights reserved.
+ * Not a contribution.
+ *
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyManagerCustom"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+// A device mask for all audio input and output devices where matching inputs/outputs on device
+// type alone is not enough: the address must match too
+#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
+#define SAMPLE_RATE_8000 8000
+#include <inttypes.h>
+#include <math.h>
+
+#include <cutils/properties.h>
+#include <utils/Log.h>
+#include <hardware/audio.h>
+#include <hardware/audio_effect.h>
+#include <media/AudioParameter.h>
+#include <soundtrigger/SoundTrigger.h>
+#include "AudioPolicyManager.h"
+#include <policy.h>
+
+namespace android {
+/*audio policy: workaround for truncated touch sounds*/
+//FIXME: workaround for truncated touch sounds
+// to be removed when the problem is handled by system UI
+#define TOUCH_SOUND_FIXED_DELAY_MS 100
+
+sp<APMConfigHelper> AudioPolicyManagerCustom::mApmConfigs = new APMConfigHelper();
+
+audio_output_flags_t AudioPolicyManagerCustom::getFallBackPath()
+{
+ audio_output_flags_t flag = AUDIO_OUTPUT_FLAG_FAST;
+ const char *fallback_path = mApmConfigs->getVoiceConcFallbackPath().c_str();
+
+ if (strlen(fallback_path) > 0) {
+ if (!strncmp(fallback_path, "deep-buffer", 11)) {
+ flag = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ }
+ else if (!strncmp(fallback_path, "fast", 4)) {
+ flag = AUDIO_OUTPUT_FLAG_FAST;
+ }
+ else {
+ ALOGD("voice_conc:not a recognised path(%s) in prop vendor.voice.conc.fallbackpath",
+ fallback_path);
+ }
+ }
+ else {
+ ALOGD("voice_conc:prop vendor.voice.conc.fallbackpath not set");
+ }
+
+ ALOGD("voice_conc:picked up flag(0x%x) from prop vendor.voice.conc.fallbackpath",
+ flag);
+
+ return flag;
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+extern "C" AudioPolicyInterface* createAudioPolicyManager(
+ AudioPolicyClientInterface *clientInterface)
+{
+ return new AudioPolicyManagerCustom(clientInterface);
+}
+
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+{
+ delete interface;
+}
+
+status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t deviceType,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name,
+ audio_format_t encodedFormat)
+{
+ ALOGD("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s format 0x%X",
+ deviceType, state, device_address, device_name, encodedFormat);
+
+ // connect/disconnect only 1 device at a time
+ if (!audio_is_output_device(deviceType) && !audio_is_input_device(deviceType)) return BAD_VALUE;
+
+ sp<DeviceDescriptor> device =
+ mHwModules.getDeviceDescriptor(deviceType, device_address, device_name, encodedFormat,
+ state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
+ if (device == 0) {
+ return INVALID_OPERATION;
+ }
+
+ // handle output devices
+ if (audio_is_output_device(deviceType)) {
+ SortedVector <audio_io_handle_t> outputs;
+
+ ssize_t index = mAvailableOutputDevices.indexOf(device);
+
+ // save a copy of the opened output descriptors before any output is opened or closed
+ // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
+ mPreviousOutputs = mOutputs;
+ switch (state)
+ {
+ // handle output device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ if (mApmConfigs->isHDMISpkEnabled() &&
+ (popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, "hdmi_spkr", 9)) {
+ mHdmiAudioDisabled = false;
+ } else {
+ mHdmiAudioEvent = true;
+ }
+ }
+ ALOGW("setDeviceConnectionState() device already connected: %x", deviceType);
+ return INVALID_OPERATION;
+ }
+ ALOGV("%s() connecting device %s format %x",
+ __func__, device->toString().c_str(), encodedFormat);
+
+ // register new device as available
+ index = mAvailableOutputDevices.add(device);
+ if (mApmConfigs->isHDMISpkEnabled() &&
+ (popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, "hdmi_spkr", 9)) {
+ mHdmiAudioDisabled = false;
+ } else {
+ mHdmiAudioEvent = true;
+ }
+ if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
+ mAvailableOutputDevices.remove(device);
+ ALOGW("HDMI sink not connected, do not route audio to HDMI out");
+ return INVALID_OPERATION;
+ }
+ }
+ if (index >= 0) {
+ sp<HwModule> module = mHwModules.getModuleForDevice(device, encodedFormat);
+ if (module == 0) {
+ ALOGD("setDeviceConnectionState() could not find HW module for device %s",
+ device->toString().c_str());
+ mAvailableOutputDevices.remove(device);
+ return INVALID_OPERATION;
+ }
+ ALOGV("setDeviceConnectionState() module name=%s", module->getName());
+ mAvailableOutputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ // Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
+ // parameters on newly connected devices (instead of opening the outputs...)
+ broadcastDeviceConnectionState(device, state);
+
+ if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
+ mAvailableOutputDevices.remove(device);
+
+ mHwModules.cleanUpForDevice(device);
+
+ broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
+ return INVALID_OPERATION;
+ }
+ if (deviceType == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ chkDpConnAndAllowedForVoice();
+ }
+
+ // outputs should never be empty here
+ ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
+ "checkOutputsForDevice() returned no outputs but status OK");
+ ALOGV("%s() checkOutputsForDevice() returned %zu outputs", __func__, outputs.size());
+
+ } break;
+ // handle output device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ if (mApmConfigs->isHDMISpkEnabled() &&
+ (popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, "hdmi_spkr", 9)) {
+ mHdmiAudioDisabled = true;
+ } else {
+ mHdmiAudioEvent = false;
+ }
+ }
+ ALOGW("setDeviceConnectionState() device not connected: %x", deviceType);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("%s() disconnecting output device %s", __func__, device->toString().c_str());
+
+ // Send Disconnect to HALs
+ broadcastDeviceConnectionState(device, state);
+
+ // remove device from available output devices
+ mAvailableOutputDevices.remove(device);
+
+ mOutputs.clearSessionRoutesForDevice(device);
+ if (mApmConfigs->isHDMISpkEnabled() &&
+ (popcount(deviceType) == 1) && (deviceType & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, "hdmi_spkr", 9)) {
+ mHdmiAudioDisabled = true;
+ } else {
+ mHdmiAudioEvent = false;
+ }
+ }
+ checkOutputsForDevice(device, state, outputs);
+
+ // Reset active device codec
+ device->setEncodedFormat(AUDIO_FORMAT_DEFAULT);
+
+ if (deviceType == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ mEngine->setDpConnAndAllowedForVoice(false);
+ }
+ } break;
+
+ default:
+ ALOGE("%s() invalid state: %x", __func__, state);
+ return BAD_VALUE;
+ }
+
+ // Propagate device availability to Engine
+ setEngineDeviceConnectionState(device, state);
+
+ if (!outputs.isEmpty()) {
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+ // close voip output before track invalidation to allow creation of
+ // new voip stream from restoreTrack
+ if ((desc->mFlags == (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_VOIP_RX)) != 0) {
+ closeOutput(outputs[i]);
+ outputs.remove(outputs[i]);
+ }
+ }
+ }
+
+ // No need to evaluate playback routing when connecting a remote submix
+ // output device used by a dynamic policy of type recorder as no
+ // playback use case is affected.
+ bool doCheckForDeviceAndOutputChanges = true;
+ if (device->type() == AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+ && strncmp(device_address, "0", AUDIO_DEVICE_MAX_ADDRESS_LEN) != 0) {
+ for (audio_io_handle_t output : outputs) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
+ sp<AudioPolicyMix> policyMix = desc->mPolicyMix.promote();
+ if (policyMix != nullptr
+ && policyMix->mMixType == MIX_TYPE_RECORDERS
+ && strncmp(device_address,
+ policyMix->mDeviceAddress.string(),
+ AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
+ doCheckForDeviceAndOutputChanges = false;
+ break;
+ }
+ }
+ }
+
+ auto checkCloseOutputs = [&]() {
+ // outputs must be closed after checkOutputForAllStrategies() is executed
+ if (!outputs.isEmpty()) {
+ for (audio_io_handle_t output : outputs) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
+ // close unused outputs after device disconnection or direct outputs that have
+ // been opened by checkOutputsForDevice() to query dynamic parameters
+ if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+ (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+ (desc->mDirectOpenCount == 0))) {
+ closeOutput(output);
+ }
+ }
+ // check A2DP again after closing A2DP output to reset mA2dpSuspended if needed
+ return true;
+ }
+ return false;
+ };
+
+ if (doCheckForDeviceAndOutputChanges) {
+ checkForDeviceAndOutputChanges(checkCloseOutputs);
+ } else {
+ checkCloseOutputs();
+ }
+
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevices);
+ }
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
+ DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
+ // do not force device change on duplicated output because if device is 0, it will
+ // also force a device 0 for the two outputs it is duplicated to which may override
+ // a valid device selection on those outputs.
+ bool force = !desc->isDuplicated()
+ && (!device_distinguishes_on_address(deviceType)
+ // always force when disconnecting (a non-duplicated device)
+ || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
+ setOutputDevices(desc, newDevices, force, 0);
+ }
+ }
+
+ if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
+ cleanUpForDevice(device);
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is output device
+
+ // handle input devices
+ if (audio_is_input_device(deviceType)) {
+ ssize_t index = mAvailableInputDevices.indexOf(device);
+ switch (state)
+ {
+ // handle input device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ ALOGW("setDeviceConnectionState() device already connected: %d", deviceType);
+ return INVALID_OPERATION;
+ }
+ sp<HwModule> module = mHwModules.getModuleForDevice(device, AUDIO_FORMAT_DEFAULT);
+ if (module == NULL) {
+ ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
+ deviceType);
+ return INVALID_OPERATION;
+ }
+
+ // Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
+ // parameters on newly connected devices (instead of opening the inputs...)
+ broadcastDeviceConnectionState(device, state);
+
+ if (checkInputsForDevice(device, state) != NO_ERROR) {
+ broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
+
+ mHwModules.cleanUpForDevice(device);
+
+ return INVALID_OPERATION;
+ }
+
+ index = mAvailableInputDevices.add(device);
+ if (index >= 0) {
+ mAvailableInputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ } break;
+
+ // handle input device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ ALOGW("setDeviceConnectionState() device not connected: %d", deviceType);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting input device %x", deviceType);
+
+ // Set Disconnect to HALs
+ broadcastDeviceConnectionState(device, state);
+
+ checkInputsForDevice(device, state);
+ mAvailableInputDevices.remove(device);
+
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ // Propagate device availability to Engine
+ setEngineDeviceConnectionState(device, state);
+
+ checkCloseInputs();
+ /*audio policy: fix call volume over USB*/
+ // As the input device list can impact the output device selection, update
+ // getDeviceForStrategy() cache
+ updateDevicesAndOutputs();
+
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevices);
+ }
+
+ if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
+ cleanUpForDevice(device);
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is input device
+
+ ALOGW("setDeviceConnectionState() invalid device: %x", deviceType);
+ return BAD_VALUE;
+}
+
+
+
+void AudioPolicyManagerCustom::chkDpConnAndAllowedForVoice()
+{
+ String8 value;
+ bool connAndAllowed = false;
+ String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+ String8("dp_for_voice"));
+
+ AudioParameter result = AudioParameter(valueStr);
+ if (result.get(String8("dp_for_voice"), value) == NO_ERROR) {
+ connAndAllowed = value.contains("true");
+ }
+ mEngine->setDpConnAndAllowedForVoice(connAndAllowed);
+}
+
+bool AudioPolicyManagerCustom::isInvalidationOfMusicStreamNeeded(const audio_attributes_t &attr)
+{
+ if (followsSameRouting(attr, attributes_initializer(AUDIO_USAGE_MEDIA))) {
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> newOutputDesc = mOutputs.valueAt(i);
+ if (newOutputDesc->mFormat == AUDIO_FORMAT_DSD)
+ return false;
+ }
+ }
+ return true;
+}
+
+
+void AudioPolicyManagerCustom::checkOutputForAttributes(const audio_attributes_t &attr)
+{
+
+ DeviceVector oldDevices = mEngine->getOutputDevicesForAttributes(attr, 0, true /*fromCache*/);
+ DeviceVector newDevices = mEngine->getOutputDevicesForAttributes(attr, 0, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevices(oldDevices, mPreviousOutputs);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevices(newDevices, mOutputs);
+
+ // also take into account external policy-related changes: add all outputs which are
+ // associated with policies in the "before" and "after" output vectors
+ ALOGVV("%s(): policy related outputs", __func__);
+ for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
+ const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ srcOutputs.add(desc->mIoHandle);
+ ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
+ }
+ }
+ for (size_t i = 0 ; i < mOutputs.size() ; i++) {
+ const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ dstOutputs.add(desc->mIoHandle);
+ ALOGVV(" new outputs: adding %d", desc->mIoHandle);
+ }
+ }
+
+ if ((srcOutputs != dstOutputs) && isInvalidationOfMusicStreamNeeded(attr)) {
+ AudioPolicyManager::checkOutputForAttributes(attr);
+ }
+}
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManagerCustom::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+ DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+ ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+ " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
+ offloadInfo.sample_rate, offloadInfo.channel_mask,
+ offloadInfo.format,
+ offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+ offloadInfo.has_video);
+
+ if (mMasterMono) {
+ return false; // no offloading if mono is set.
+ }
+
+ if (mApmConfigs->isVoiceConcEnabled()) {
+ if (mApmConfigs->isVoicePlayConcDisabled() && isInCall()) {
+ ALOGD("\n copl: blocking compress offload on call mode\n");
+ return false;
+ }
+ }
+ if (mApmConfigs->isVoiceDSDConcDisabled() &&
+ isInCall() && (offloadInfo.format == AUDIO_FORMAT_DSD)) {
+ ALOGD("blocking DSD compress offload on call mode");
+ return false;
+ }
+ if (mApmConfigs->isRecPlayConcEnabled()) {
+ if (mApmConfigs->isRecPlayConcDisabled() &&
+ ((true == mIsInputRequestOnProgress) ||
+ (mInputs.activeInputsCountOnDevices(primaryInputDevices) > 0))) {
+ ALOGD("copl: blocking compress offload for record concurrency");
+ return false;
+ }
+ }
+ // Check if stream type is music, then only allow offload as of now.
+ if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+ {
+ ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+ return false;
+ }
+
+ // Check if offload has been disabled
+ bool offloadDisabled = mApmConfigs->isAudioOffloadDisabled();
+ if (offloadDisabled) {
+ ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled);
+ return false;
+ }
+
+ //check if it's multi-channel AAC (includes sub formats) and FLAC format
+ if ((popcount(offloadInfo.channel_mask) > 2) &&
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
+ ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
+ return false;
+ }
+
+ if (mApmConfigs->isExtnFormatsEnabled()) {
+ //check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
+ if ((popcount(offloadInfo.channel_mask) > 2) &&
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && (offloadInfo.sample_rate > 48000)) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.sample_rate > 48000)) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.sample_rate > 48000)) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) {
+ ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
+ return false;
+ }
+
+ // check against wma std bit rate restriction
+ if ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) {
+ int32_t sr_id = -1;
+ uint32_t min_bitrate, max_bitrate;
+ for (int i = 0; i < WMA_STD_NUM_FREQ; i++) {
+ if (offloadInfo.sample_rate == wmaStdSampleRateTbl[i]) {
+ sr_id = i;
+ break;
+ }
+ }
+ if ((sr_id < 0) || (popcount(offloadInfo.channel_mask) > 2)
+ || (popcount(offloadInfo.channel_mask) <= 0)) {
+ ALOGE("invalid sample rate or channel count");
+ return false;
+ }
+
+ min_bitrate = wmaStdMinAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1];
+ max_bitrate = wmaStdMaxAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1];
+ if ((offloadInfo.bit_rate > max_bitrate) || (offloadInfo.bit_rate < min_bitrate)) {
+ ALOGD("offload disabled for WMA clips with unsupported bit rate");
+ ALOGD("bit_rate %d, max_bitrate %d, min_bitrate %d", offloadInfo.bit_rate, max_bitrate, min_bitrate);
+ return false;
+ }
+ }
+
+ // Safely choose the min bitrate as threshold and leave the restriction to NT decoder as we can't distinguish wma pro and wma lossless here.
+ if ((((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_PRO)) ||
+ (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_LOSSLESS))) {
+ ALOGD("offload disabled for WMA_PRO/WMA_LOSSLESS clips with bit rate over maximum supported value");
+ return false;
+ }
+ }
+
+ //TODO: enable audio offloading with video when ready
+ if (offloadInfo.has_video && !mApmConfigs->isAudioOffloadVideoEnabled()) {
+ ALOGV("isOffloadSupported: has_video == true, returning false");
+ return false;
+ }
+
+ if (offloadInfo.has_video && offloadInfo.is_streaming &&
+ !mApmConfigs->isAVStreamingOffloadEnabled()) {
+ ALOGW("offload disabled by vendor.audio.av.streaming.offload.enable %d",
+ mApmConfigs->isAVStreamingOffloadEnabled());
+ return false;
+ }
+
+ //If duration is less than minimum value defined in property, return false
+ if (mApmConfigs->getAudioOffloadMinDuration() > 0) {
+ if (offloadInfo.duration_us < (mApmConfigs->getAudioOffloadMinDuration() * 1000000 )) {
+ ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%u)", mApmConfigs->getAudioOffloadMinDuration());
+ return false;
+ }
+ } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+ ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+ //duration checks only valid for MP3/AAC/ formats,
+ //do not check duration for other audio formats, e.g. AAC/AC3 and amrwb+ formats
+ if ((offloadInfo.format == AUDIO_FORMAT_MP3) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))
+ return false;
+
+ if (mApmConfigs->isExtnFormatsEnabled()) {
+ if (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DSD) ||
+ ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))
+ return false;
+ }
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if (mEffects.isNonOffloadableEffectEnabled()) {
+ return false;
+ }
+
+ // See if there is a profile to support this.
+ // AUDIO_DEVICE_NONE
+ sp<IOProfile> profile = getProfileForOutput(DeviceVector() /*ignore device */,
+ offloadInfo.sample_rate,
+ offloadInfo.format,
+ offloadInfo.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,
+ true /*directOnly*/);
+ ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
+ return (profile != 0);
+}
+
+void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)
+{
+ ALOGD("setPhoneState() state %d", state);
+ // store previous phone state for management of sonification strategy below
+ int oldState = mEngine->getPhoneState();
+
+ if (mEngine->setPhoneState(state) != NO_ERROR) {
+ ALOGW("setPhoneState() invalid or same state %d", state);
+ return;
+ }
+ /// Opens: can these line be executed after the switch of volume curves???
+ if (isStateInCall(oldState)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+
+ // force reevaluating accessibility routing when call stops
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ /**
+ * Switching to or from incall state or switching between telephony and VoIP lead to force
+ * routing command.
+ */
+ bool force = ((is_state_in_call(oldState) != is_state_in_call(state))
+ || (is_state_in_call(state) && (state != oldState)));
+
+ // check for device and output changes triggered by new phone state
+ checkForDeviceAndOutputChanges();
+
+ sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
+ if (mApmConfigs->isVoiceConcEnabled()) {
+ bool prop_playback_enabled = mApmConfigs->isVoicePlayConcDisabled();
+ bool prop_rec_enabled = mApmConfigs->isVoiceRecConcDisabled();
+ bool prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled();
+
+ if ((AUDIO_MODE_IN_CALL != oldState) && (AUDIO_MODE_IN_CALL == state)) {
+ ALOGD("voice_conc:Entering to call mode oldState :: %d state::%d ",
+ oldState, state);
+ mvoice_call_state = state;
+ if (prop_rec_enabled) {
+ //Close all active inputs
+ Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
+ if (activeInputs.size() != 0) {
+ for (size_t i = 0; i < activeInputs.size(); i++) {
+ sp<AudioInputDescriptor> activeInput = activeInputs[i];
+ switch(activeInput->source()) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ ALOGD("voice_conc:FOUND active input during call active: %d",
+ activeInput->source());
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if (prop_voip_enabled) {
+ ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",
+ activeInput->source());
+ RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/);
+ for (const auto& activeClient : activeClients) {
+ closeClient(activeClient->portId());
+ }
+ }
+ break;
+
+ default:
+ ALOGD("voice_conc:CLOSING input on call setup for inputSource: %d",
+ activeInput->source());
+ RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/);
+ for (const auto& activeClient : activeClients) {
+ closeClient(activeClient->portId());
+ }
+ break;
+ }
+ }
+ }
+ } else if (prop_voip_enabled) {
+ Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
+ if (activeInputs.size() != 0) {
+ for (size_t i = 0; i < activeInputs.size(); i++) {
+ sp<AudioInputDescriptor> activeInput = activeInputs[i];
+ if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeInput->source()) {
+ ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeInput->source());
+ RecordClientVector activeClients = activeInput->clientsList(true /*activeOnly*/);
+ for (const auto& activeClient : activeClients) {
+ closeClient(activeClient->portId());
+ }
+ }
+ }
+ }
+ }
+ if (prop_playback_enabled) {
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
+ ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
+ if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+ if ((AUDIO_STREAM_MUSIC == i) ||
+ (AUDIO_STREAM_VOICE_CALL == i) ) {
+ ALOGD("voice_conc:Invalidate stream type %d", i);
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ } else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+ ALOGD("voice_conc:Invalidate stream type %d", i);
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("voice_conc:ouput desc / profile is NULL");
+ continue;
+ }
+
+ bool isFastFallBackNeeded =
+ ((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & outputDesc->mProfile->getFlags());
+
+ if ((AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) && isFastFallBackNeeded) {
+ if (((!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY))
+ && prop_playback_enabled) {
+ ALOGD("voice_conc:calling suspendOutput on call mode for primary output");
+ mpClientInterface->suspendOutput(mOutputs.keyAt(i));
+ } //Close compress all sessions
+ else if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
+ && prop_playback_enabled) {
+ ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ else if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX)
+ && prop_voip_enabled) {
+ ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ } else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+ if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX) {
+ if (prop_voip_enabled) {
+ ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ else if (prop_playback_enabled
+ && (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
+ ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ }
+ }
+
+ if ((AUDIO_MODE_IN_CALL == oldState || AUDIO_MODE_IN_COMMUNICATION == oldState) &&
+ (AUDIO_MODE_NORMAL == state) && prop_playback_enabled && mvoice_call_state) {
+ ALOGD("voice_conc:EXITING from call mode oldState :: %d state::%d \n",oldState, state);
+ mvoice_call_state = 0;
+ if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+ //restore PCM (deep-buffer) output after call termination
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("voice_conc:ouput desc / profile is NULL");
+ continue;
+ }
+ if (!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ ALOGD("voice_conc:calling restoreOutput after call mode for primary output");
+ mpClientInterface->restoreOutput(mOutputs.keyAt(i));
+ }
+ }
+ }
+ //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
+ for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
+ ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
+ if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
+ if ((AUDIO_STREAM_MUSIC == i) ||
+ (AUDIO_STREAM_VOICE_CALL == i) ) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ } else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+ }
+
+ sp<SwAudioOutputDescriptor> outputDesc = NULL;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ outputDesc = mOutputs.valueAt(i);
+ if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("voice_conc:ouput desc / profile is NULL");
+ continue;
+ }
+
+ if (mApmConfigs->isVoiceDSDConcDisabled() &&
+ (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+ (outputDesc->mFormat == AUDIO_FORMAT_DSD)) {
+ ALOGD("voice_conc:calling closeOutput on call mode for DSD COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ // call invalidate for music, so that DSD compress will fallback to deep-buffer.
+ mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+ }
+
+ }
+
+ if (mApmConfigs->isRecPlayConcEnabled()) {
+ if (mApmConfigs->isRecPlayConcDisabled()) {
+ if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
+ ALOGD("phone state changed to MODE_IN_COMM invlaidating music and voice streams");
+ // call invalidate for voice streams, so that it can use deepbuffer with VoIP out device from HAL
+ mpClientInterface->invalidateStream(AUDIO_STREAM_VOICE_CALL);
+ // call invalidate for music, so that compress will fallback to deep-buffer with VoIP out device
+ mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+
+ // close compress output to make sure session will be closed before timeout(60sec)
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("ouput desc / profile is NULL");
+ continue;
+ }
+
+ if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ ALOGD("calling closeOutput on call mode for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ } else if ((oldState == AUDIO_MODE_IN_COMMUNICATION) &&
+ (mEngine->getPhoneState() == AUDIO_MODE_NORMAL)) {
+ // call invalidate for music so that music can fallback to compress
+ mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+ }
+ }
+ }
+ mPrevPhoneState = oldState;
+ int delayMs = 0;
+ if (isStateInCall(state)) {
+ nsecs_t sysTime = systemTime();
+ auto musicStrategy = streamToStrategy(AUDIO_STREAM_MUSIC);
+ auto sonificationStrategy = streamToStrategy(AUDIO_STREAM_ALARM);
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ // mute media and sonification strategies and delay device switch by the largest
+ // latency of any output where either strategy is active.
+ // This avoid sending the ring tone or music tail into the earpiece or headset.
+ if ((desc->isStrategyActive(musicStrategy,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime) ||
+ desc->isStrategyActive(sonificationStrategy,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime)) &&
+ (delayMs < (int)desc->latency()*2)) {
+ delayMs = desc->latency()*2;
+ }
+ setStrategyMute(musicStrategy, true, desc);
+ setStrategyMute(musicStrategy, false, desc, MUTE_TIME_MS,
+ mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA),
+ nullptr, true /*fromCache*/).types());
+ setStrategyMute(sonificationStrategy, true, desc);
+ setStrategyMute(sonificationStrategy, false, desc, MUTE_TIME_MS,
+ mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_ALARM),
+ nullptr, true /*fromCache*/).types());
+ }
+ }
+
+ if (hasPrimaryOutput()) {
+ // Note that despite the fact that getNewOutputDevice() is called on the primary output,
+ // the device returned is not necessarily reachable via this output
+ DeviceVector rxDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (isStateInCall(oldState) && rxDevices.isEmpty()) {
+ rxDevices = mPrimaryOutput->devices();
+ }
+
+ if (state == AUDIO_MODE_IN_CALL) {
+ updateCallRouting(rxDevices, delayMs);
+ } else if (oldState == AUDIO_MODE_IN_CALL) {
+ if (mCallRxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ mCallRxPatch.clear();
+ }
+ if (mCallTxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ mCallTxPatch.clear();
+ }
+ setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
+ } else {
+ setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
+ }
+ }
+ //update device for all non-primary outputs
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ if (output != mPrimaryOutput->mIoHandle) {
+ DeviceVector newDevices = getNewOutputDevices(mOutputs.valueFor(output), false /*fromCache*/);
+ setOutputDevices(mOutputs.valueFor(output), newDevices, !newDevices.isEmpty());
+ }
+ }
+ if (isStateInCall(state)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+
+ // force reevaluating accessibility routing when call starts
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+ mLimitRingtoneVolume = (state == AUDIO_MODE_RINGTONE &&
+ isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY));
+}
+
+void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ ALOGD("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
+ if (config == mEngine->getForceUse(usage)) {
+ return;
+ }
+
+ if (mEngine->setForceUse(usage, config) != NO_ERROR) {
+ ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
+ return;
+ }
+ bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
+
+ // check for device and output changes triggered by new force usage
+ checkForDeviceAndOutputChanges();
+
+ /*audio policy: workaround for truncated touch sounds*/
+ //FIXME: workaround for truncated touch sounds
+ // to be removed when the problem is handled by system UI
+ uint32_t delayMs = 0;
+ uint32_t waitMs = 0;
+ if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) {
+ delayMs = TOUCH_SOUND_FIXED_DELAY_MS;
+ }
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
+ if (forceVolumeReeval && !newDevices.isEmpty()) {
+ applyStreamVolumes(mPrimaryOutput, newDevices.types(), delayMs, true);
+ }
+ waitMs = updateCallRouting(newDevices, delayMs);
+ }
+ // Use reverse loop to make sure any low latency usecases (generally tones)
+ // are not routed before non LL usecases (generally music).
+ // We can safely assume that LL output would always have lower index,
+ // and use this work-around to avoid routing of output with music stream
+ // from the context of short lived LL output.
+ // Note: in case output's share backend(HAL sharing is implicit) all outputs
+ // gets routing update while processing first output itself.
+ for (size_t i = mOutputs.size(); i > 0; i--) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
+ DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
+ waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(),
+ delayMs);
+
+ if (forceVolumeReeval && !newDevices.isEmpty()) {
+ applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
+ }
+ }
+ }
+
+ for (const auto& activeDesc : mInputs.getActiveInputs()) {
+ // Skip for hotword recording as the input device switch
+ // is handled within sound trigger HAL
+ if (activeDesc->isSoundTrigger() &&
+ activeDesc->source() == AUDIO_SOURCE_HOTWORD) {
+ continue;
+ }
+ auto newDevice = getNewInputDevice(activeDesc);
+ // Force new input selection if the new device can not be reached via current input
+ if (activeDesc->mProfile->getSupportedDevices().contains(newDevice)) {
+ setInputDevice(activeDesc->mIoHandle, newDevice);
+ } else {
+ closeInput(activeDesc->mIoHandle);
+ }
+ }
+}
+
+status_t AudioPolicyManagerCustom::stopSource(const sp<SwAudioOutputDescriptor>& outputDesc,
+ const sp<TrackClientDescriptor>& client)
+{
+ audio_stream_type_t stream = client->stream();
+ auto clientVolSrc = client->volumeSource();
+
+ if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ ALOGW("stopSource() invalid stream %d", stream);
+ return INVALID_OPERATION;
+ }
+ // always handle stream stop, check which stream type is stopping
+ handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
+
+ if (outputDesc->getActivityCount(clientVolSrc) > 0) {
+ if (outputDesc->getActivityCount(clientVolSrc) == 1) {
+ // Automatically disable the remote submix input when output is stopped on a
+ // re routing mix of type MIX_TYPE_RECORDERS
+ sp<AudioPolicyMix> policyMix = outputDesc->mPolicyMix.promote();
+ if (audio_is_remote_submix_device(outputDesc->devices().types()) &&
+ policyMix != NULL &&
+ policyMix->mMixType == MIX_TYPE_RECORDERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ policyMix->mDeviceAddress,
+ "remote-submix", AUDIO_FORMAT_DEFAULT);
+ }
+ }
+ bool forceDeviceUpdate = false;
+ if (client->hasPreferredDevice(true)) {
+ checkStrategyRoute(client->strategy(), AUDIO_IO_HANDLE_NONE);
+ forceDeviceUpdate = true;
+ }
+
+ // decrement usage count of this stream on the output
+ outputDesc->setClientActive(client, false);
+
+ // store time at which the stream was stopped - see isStreamActive()
+ if (outputDesc->getActivityCount(clientVolSrc) == 0 || forceDeviceUpdate) {
+ outputDesc->setStopTime(client, systemTime());
+ DeviceVector prevDevices = outputDesc->devices();
+ DeviceVector newDevices = getNewOutputDevices(outputDesc, false /*fromCache*/);
+ // delay the device switch by twice the latency because stopOutput() is executed when
+ // the track stop() command is received and at that time the audio track buffer can
+ // still contain data that needs to be drained. The latency only covers the audio HAL
+ // and kernel buffers. Also the latency does not always include additional delay in the
+ // audio path (audio DSP, CODEC ...)
+ setOutputDevices(outputDesc, newDevices, false, outputDesc->latency()*2);
+
+ // force restoring the device selection on other active outputs if it differs from the
+ // one being selected for this output
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != outputDesc &&
+ desc->isActive() &&
+ outputDesc->sharesHwModuleWith(desc) &&
+ (newDevices != desc->devices())) {
+ DeviceVector dev = getNewOutputDevices(mOutputs.valueFor(curOutput), false /*fromCache*/);
+ bool force = prevDevices != dev;
+ uint32_t delayMs;
+ if (dev == prevDevices) {
+ delayMs = 0;
+ } else {
+ delayMs = outputDesc->latency()*2;
+ }
+ setOutputDevices(desc,
+ dev,
+ force,
+ delayMs);
+ /*audio policy: fix media volume after ringtone*/
+ // re-apply device specific volume if not done by setOutputDevice()
+ if (!force) {
+ applyStreamVolumes(desc, dev.types(), delayMs);
+ }
+ }
+ }
+ // update the outputs if stopping one with a stream that can affect notification routing
+ handleNotificationRoutingForStream(stream);
+ }
+
+ if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE &&
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), false, outputDesc);
+ }
+
+ if (followsSameRouting(client->attributes(), attributes_initializer(AUDIO_USAGE_MEDIA))) {
+ selectOutputForMusicEffects();
+ }
+ return NO_ERROR;
+ } else {
+ ALOGW("stopOutput() refcount is already 0");
+ return INVALID_OPERATION;
+ }
+}
+
+status_t AudioPolicyManagerCustom::startSource(const sp<SwAudioOutputDescriptor>& outputDesc,
+ const sp<TrackClientDescriptor>& client,
+ uint32_t *delayMs)
+{
+ // cannot start playback of STREAM_TTS if any other output is being used
+ uint32_t beaconMuteLatency = 0;
+ audio_stream_type_t stream = client->stream();
+ auto clientVolSrc = client->volumeSource();
+ auto clientStrategy = client->strategy();
+ auto clientAttr = client->attributes();
+
+ if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ ALOGW("startSource() invalid stream %d", stream);
+ return INVALID_OPERATION;
+ }
+
+ *delayMs = 0;
+ if (stream == AUDIO_STREAM_TTS) {
+ ALOGV("\t found BEACON stream");
+ if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive(
+ toVolumeSource(AUDIO_STREAM_TTS) /*sourceToIgnore*/)) {
+ return INVALID_OPERATION;
+ } else {
+ beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
+ }
+ } else {
+ // some playback other than beacon starts
+ beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
+ }
+
+ // force device change if the output is inactive and no audio patch is already present.
+ // check active before incrementing usage count
+ bool force = !outputDesc->isActive() &&
+ (outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE);
+
+ DeviceVector devices;
+ sp<AudioPolicyMix> policyMix = outputDesc->mPolicyMix.promote();
+ const char *address = NULL;
+ if (policyMix != NULL) {
+ audio_devices_t newDeviceType;
+ address = policyMix->mDeviceAddress.string();
+ if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
+ newDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ } else {
+ newDeviceType = policyMix->mDeviceType;
+ }
+ sp device = mAvailableOutputDevices.getDevice(newDeviceType, String8(address),
+ AUDIO_FORMAT_DEFAULT);
+ ALOG_ASSERT(device, "%s: no device found t=%u, a=%s", __func__, newDeviceType, address);
+ devices.add(device);
+ }
+
+ // requiresMuteCheck is false when we can bypass mute strategy.
+ // It covers a common case when there is no materially active audio
+ // and muting would result in unnecessary delay and dropped audio.
+ const uint32_t outputLatencyMs = outputDesc->latency();
+ bool requiresMuteCheck = outputDesc->isActive(outputLatencyMs * 2); // account for drain
+
+ // increment usage count for this stream on the requested output:
+ // NOTE that the usage count is the same for duplicated output and hardware output which is
+ // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->setClientActive(client, true);
+
+ if (client->hasPreferredDevice(true)) {
+ devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
+ if (devices != outputDesc->devices()) {
+ checkStrategyRoute(clientStrategy, outputDesc->mIoHandle);
+ }
+ }
+
+ if (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_MEDIA))) {
+ selectOutputForMusicEffects();
+ }
+
+ if (outputDesc->getActivityCount(clientVolSrc) == 1 || !devices.isEmpty()) {
+ // starting an output being rerouted?
+ if (devices.isEmpty()) {
+ devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
+ }
+ bool shouldWait =
+ (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM)) ||
+ followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_NOTIFICATION)) ||
+ (beaconMuteLatency > 0));
+ uint32_t waitMs = beaconMuteLatency;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != outputDesc) {
+ // An output has a shared device if
+ // - managed by the same hw module
+ // - supports the currently selected device
+ const bool sharedDevice = outputDesc->sharesHwModuleWith(desc)
+ && (!desc->filterSupportedDevices(devices).isEmpty());
+
+ // force a device change if any other output is:
+ // - managed by the same hw module
+ // - supports currently selected device
+ // - has a current device selection that differs from selected device.
+ // - has an active audio patch
+ // In this case, the audio HAL must receive the new device selection so that it can
+ // change the device currently selected by the other output.
+ if (sharedDevice &&
+ desc->devices() != devices &&
+ desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) {
+ force = true;
+ }
+ // wait for audio on other active outputs to be presented when starting
+ // a notification so that audio focus effect can propagate, or that a mute/unmute
+ // event occurred for beacon
+ const uint32_t latencyMs = desc->latency();
+ const bool isActive = desc->isActive(latencyMs * 2); // account for drain
+
+ if (shouldWait && isActive && (waitMs < latencyMs)) {
+ waitMs = latencyMs;
+ }
+
+ // Require mute check if another output is on a shared device
+ // and currently active to have proper drain and avoid pops.
+ // Note restoring AudioTracks onto this output needs to invoke
+ // a volume ramp if there is no mute.
+ requiresMuteCheck |= sharedDevice && isActive;
+ }
+ }
+ const uint32_t muteWaitMs =
+ setOutputDevices(outputDesc, devices, force, 0, NULL, requiresMuteCheck);
+
+ // apply volume rules for current stream and device if necessary
+ auto &curves = getVolumeCurves(client->attributes());
+ checkAndSetVolume(curves, client->volumeSource(),
+ getVolumeCurves(stream).getVolumeIndex(outputDesc->devices().types()),
+ outputDesc,
+ outputDesc->devices().types());
+
+ // update the outputs if starting an output with a stream that can affect notification
+ // routing
+ handleNotificationRoutingForStream(stream);
+
+ // force reevaluating accessibility routing when ringtone or alarm starts
+ if (followsSameRouting(clientAttr, attributes_initializer(AUDIO_USAGE_ALARM))) {
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+ if (waitMs > muteWaitMs) {
+ *delayMs = waitMs - muteWaitMs;
+ }
+
+ }
+
+ if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE &&
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), true, outputDesc);
+ }
+
+ // Automatically enable the remote submix input when output is started on a re routing mix
+ // of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(devices.types()) && policyMix != NULL &&
+ policyMix->mMixType == MIX_TYPE_RECORDERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address,
+ "remote-submix",
+ AUDIO_FORMAT_DEFAULT);
+ }
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerCustom::checkAndSetVolume(IVolumeCurves &curves,
+ VolumeSource volumeSource,
+ int index,
+ const sp<AudioOutputDescriptor>& outputDesc,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+ // do not change actual stream volume if the stream is muted
+ if (outputDesc->isMuted(volumeSource)) {
+ ALOGVV("%s: volume source %d muted count %d active=%d", __func__, volumeSource,
+ outputDesc->getMuteCount(volumeSource), outputDesc->isActive(volumeSource));
+ return NO_ERROR;
+ }
+
+ VolumeSource callVolSrc = toVolumeSource(AUDIO_STREAM_VOICE_CALL);
+ VolumeSource btScoVolSrc = toVolumeSource(AUDIO_STREAM_BLUETOOTH_SCO);
+ bool isVoiceVolSrc = callVolSrc == volumeSource;
+ bool isBtScoVolSrc = btScoVolSrc == volumeSource;
+
+ audio_policy_forced_cfg_t forceUseForComm =
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION);
+ // do not change in call volume if bluetooth is connected and vice versa
+ if ((callVolSrc != btScoVolSrc) &&
+ ((isVoiceVolSrc && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (isBtScoVolSrc && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO))) {
+ ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+ volumeSource, forceUseForComm);
+ return INVALID_OPERATION;
+ }
+
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->devices().types();
+ }
+
+ float volumeDb = computeVolume(curves, volumeSource, index, device);
+ if (outputDesc->isFixedVolume(device)) {
+ volumeDb = 0.0f;
+ }
+
+ outputDesc->setVolume(volumeDb, volumeSource, curves.getStreamTypes(), device, delayMs, force);
+
+ if (isVoiceVolSrc || isBtScoVolSrc) {
+ float voiceVolume;
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (isVoiceVolSrc) {
+ voiceVolume = (float)index/(float)curves.getVolumeIndexMax();
+ } else {
+ voiceVolume = 1.0;
+ }
+
+ if (voiceVolume != mLastVoiceVolume) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ mLastVoiceVolume = voiceVolume;
+ }
+ }
+ return NO_ERROR;
+}
+
+bool static tryForDirectPCM(audio_output_flags_t flags)
+{
+ bool trackDirectPCM = false; // Output request for track created by other apps
+
+ if (flags == AUDIO_OUTPUT_FLAG_NONE) {
+ if (AudioPolicyManagerCustom::mApmConfigs != NULL)
+ trackDirectPCM = AudioPolicyManagerCustom::mApmConfigs->isAudioTrackOffloadEnabled();
+ }
+ return trackDirectPCM;
+}
+
+status_t AudioPolicyManagerCustom::getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uid_t uid,
+ const audio_config_t *config,
+ audio_output_flags_t *flags,
+ audio_port_handle_t *selectedDeviceId,
+ audio_port_handle_t *portId,
+ std::vector<audio_io_handle_t> *secondaryOutputs)
+{
+ audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER;
+ audio_config_t tConfig;
+
+ uint32_t bitWidth = (audio_bytes_per_sample(config->format) * 8);
+
+ memcpy(&tConfig, config, sizeof(audio_config_t));
+ if ((*flags == AUDIO_OUTPUT_FLAG_DIRECT || tryForDirectPCM(*flags)) &&
+ (!memcmp(&config->offload_info, &tOffloadInfo, sizeof(audio_offload_info_t)))) {
+ tConfig.offload_info.sample_rate = config->sample_rate;
+ tConfig.offload_info.channel_mask = config->channel_mask;
+ tConfig.offload_info.format = config->format;
+ tConfig.offload_info.stream_type = *stream;
+ tConfig.offload_info.bit_width = bitWidth;
+ if (attr != NULL) {
+ ALOGV("found attribute .. setting usage %d ", attr->usage);
+ tConfig.offload_info.usage = attr->usage;
+ } else {
+ ALOGI("%s:: attribute is NULL .. no usage set", __func__);
+ }
+ }
+
+ return AudioPolicyManager::getOutputForAttr(attr, output, session, stream,
+ (uid_t)uid, &tConfig,
+ flags,
+ (audio_port_handle_t*)selectedDeviceId,
+ portId,
+ secondaryOutputs);
+}
+
+audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevices(
+ const DeviceVector &devices,
+ audio_session_t session,
+ audio_stream_type_t stream,
+ const audio_config_t *config,
+ audio_output_flags_t *flags,
+ bool forceMutingHaptic)
+{
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ status_t status;
+
+ // Discard haptic channel mask when forcing muting haptic channels.
+ audio_channel_mask_t channelMask = forceMutingHaptic
+ ? (config->channel_mask & ~AUDIO_CHANNEL_HAPTIC_ALL) : config->channel_mask;
+
+ if (stream < AUDIO_STREAM_MIN || stream >= AUDIO_STREAM_CNT) {
+ ALOGE("%s: invalid stream %d", __func__, stream);
+ return AUDIO_IO_HANDLE_NONE;
+ }
+
+ if (((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
+ (stream != AUDIO_STREAM_MUSIC)) {
+ // compress should not be used for non-music streams
+ ALOGE("Offloading only allowed with music stream");
+ return 0;
+ }
+
+ if (mApmConfigs->isCompressVOIPEnabled()) {
+ if (stream == AUDIO_STREAM_VOICE_CALL &&
+ audio_is_linear_pcm(config->format)) {
+ // let voice stream to go with primary output by default
+ // in case direct voip is bypassed
+ bool use_primary_out = true;
+
+ if ((channelMask == 1) &&
+ (config->sample_rate == 8000 || config->sample_rate == 16000 ||
+ config->sample_rate == 32000 || config->sample_rate == 48000)) {
+ // Allow Voip direct output only if:
+ // audio mode is MODE_IN_COMMUNCATION; AND
+ // voip output is not opened already; AND
+ // requested sample rate matches with that of voip input stream (if opened already)
+ int value = 0;
+ uint32_t voipOutCount = 1, voipSampleRate = 1;
+
+ String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+ String8("voip_out_stream_count"));
+ AudioParameter result = AudioParameter(valueStr);
+ if (result.getInt(String8("voip_out_stream_count"), value) == NO_ERROR) {
+ voipOutCount = value;
+ }
+
+ valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
+ String8("voip_sample_rate"));
+ result = AudioParameter(valueStr);
+ if (result.getInt(String8("voip_sample_rate"), value) == NO_ERROR) {
+ voipSampleRate = value;
+ }
+
+ if ((voipOutCount == 0) &&
+ ((voipSampleRate == 0) || (voipSampleRate == config->sample_rate))) {
+ if (mApmConfigs->useVoicePathForPCMVOIP()
+ && (config->format == AUDIO_FORMAT_PCM_16_BIT)) {
+ *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
+ AUDIO_OUTPUT_FLAG_DIRECT);
+ ALOGD("Set VoIP and Direct output flags for PCM format");
+ use_primary_out = false;
+ }
+ }
+ }
+
+ if (use_primary_out) {
+ *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_PRIMARY);
+ }
+ }
+ } else {
+ if (stream == AUDIO_STREAM_VOICE_CALL &&
+ audio_is_linear_pcm(config->format) &&
+ (config->channel_mask == 1) &&
+ (config->sample_rate == 8000 || config->sample_rate == 16000 ||
+ config->sample_rate == 32000 || config->sample_rate == 48000)) {
+ //check if VoIP output is not opened already
+ bool voip_pcm_already_in_use = false;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc->mFlags == (AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_DIRECT)) {
+ //close voip output if currently open by the same client with different device
+ if (desc->mDirectClientSession == session &&
+ desc->devices() != devices) {
+ closeOutput(desc->mIoHandle);
+ } else {
+ voip_pcm_already_in_use = true;
+ ALOGD("VoIP PCM already in use");
+ }
+ break;
+ }
+ }
+
+ if (!voip_pcm_already_in_use) {
+ *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
+ AUDIO_OUTPUT_FLAG_DIRECT);
+ ALOGV("Set VoIP and Direct output flags for PCM format");
+ }
+ }
+ } /* compress_voip_enabled */
+
+ //IF VOIP is going to be started at the same time as when
+ //vr is enabled, get VOIP to fallback to low latency
+ String8 vr_value;
+ String8 value_Str;
+ bool is_vr_mode_on = false;
+ AudioParameter ret;
+
+ value_Str = mpClientInterface->getParameters((audio_io_handle_t)0,
+ String8("vr_audio_mode_on"));
+ ret = AudioParameter(value_Str);
+ if (ret.get(String8("vr_audio_mode_on"), vr_value) == NO_ERROR) {
+ is_vr_mode_on = vr_value.contains("true");
+ ALOGI("VR mode is %d, switch to primary output if request is for fast|raw",
+ is_vr_mode_on);
+ }
+
+ if (is_vr_mode_on) {
+ //check the flags being requested for, and clear FAST|RAW
+ *flags = (audio_output_flags_t)(*flags &
+ (~(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW)));
+
+ }
+
+ if (mApmConfigs->isVoiceConcEnabled()) {
+ bool prop_play_enabled = false, prop_voip_enabled = false;
+ prop_play_enabled = mApmConfigs->isVoicePlayConcDisabled();
+ prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled();
+
+ bool isDeepBufferFallBackNeeded =
+ ((AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & *flags);
+ bool isFastFallBackNeeded =
+ ((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & *flags);
+
+ if (prop_play_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ if ((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+ ((AUDIO_MODE_IN_CALL == mPrevPhoneState)
+ && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+ {
+ if (AUDIO_OUTPUT_FLAG_VOIP_RX & *flags) {
+ if (prop_voip_enabled) {
+ ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
+ *flags );
+ return 0;
+ }
+ }
+ else {
+ if (isFastFallBackNeeded &&
+ (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag)) {
+ ALOGD("voice_conc:IN call mode adding ULL flags .. flags: %x ", *flags );
+ *flags = AUDIO_OUTPUT_FLAG_FAST;
+ } else if (isDeepBufferFallBackNeeded &&
+ (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag)) {
+ if (AUDIO_STREAM_MUSIC == stream) {
+ *flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ ALOGD("voice_conc:IN call mode adding deep-buffer flags %x ", *flags );
+ }
+ else {
+ *flags = AUDIO_OUTPUT_FLAG_FAST;
+ ALOGD("voice_conc:IN call mode adding fast flags %x ", *flags );
+ }
+ }
+ }
+ }
+ } else if (prop_voip_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //return only ULL ouput
+ if ((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+ ((AUDIO_MODE_IN_CALL == mPrevPhoneState)
+ && (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+ {
+ if (AUDIO_OUTPUT_FLAG_VOIP_RX & *flags) {
+ ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
+ *flags );
+ return 0;
+ }
+ }
+ }
+ }
+ if (mApmConfigs->isRecPlayConcEnabled()) {
+ bool prop_rec_play_enabled = mApmConfigs->isRecPlayConcDisabled();
+ DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+ if ((prop_rec_play_enabled) &&
+ ((true == mIsInputRequestOnProgress) ||
+ (mInputs.activeInputsCountOnDevices(primaryInputDevices) > 0))) {
+ if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
+ if (AUDIO_OUTPUT_FLAG_VOIP_RX & *flags) {
+ // allow VoIP using voice path
+ // Do nothing
+ } else if ((*flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+ ALOGD("voice_conc:MODE_IN_COMM is setforcing deep buffer output for non ULL... flags: %x", *flags);
+ // use deep buffer path for all non ULL outputs
+ *flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ }
+ } else if ((*flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+ ALOGD("voice_conc:Record mode is on forcing deep buffer output for non ULL... flags: %x ", *flags);
+ // use deep buffer path for all non ULL outputs
+ *flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
+ }
+ }
+ if (prop_rec_play_enabled &&
+ (stream == AUDIO_STREAM_ENFORCED_AUDIBLE)) {
+ ALOGD("Record conc is on forcing ULL output for ENFORCED_AUDIBLE");
+ *flags = AUDIO_OUTPUT_FLAG_FAST;
+ }
+ }
+
+ /*
+ * WFD audio routes back to target speaker when starting a ringtone playback.
+ * This is because primary output is reused for ringtone, so output device is
+ * updated based on SONIFICATION strategy for both ringtone and music playback.
+ * The same issue is not seen on remoted_submix HAL based WFD audio because
+ * primary output is not reused and a new output is created for ringtone playback.
+ * Issue is fixed by updating output flag to AUDIO_OUTPUT_FLAG_FAST when there is
+ * a non-music stream playback on WFD, so primary output is not reused for ringtone.
+ */
+ if (mApmConfigs->isAFEProxyEnabled()) {
+ audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
+ if ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY)
+ && (stream != AUDIO_STREAM_MUSIC)) {
+ ALOGD("WFD audio: use OUTPUT_FLAG_FAST for non music stream. flags:%x", *flags );
+ //For voip paths
+ if (*flags & AUDIO_OUTPUT_FLAG_DIRECT)
+ *flags = AUDIO_OUTPUT_FLAG_DIRECT;
+ else //route every thing else to ULL path
+ *flags = AUDIO_OUTPUT_FLAG_FAST;
+ }
+ }
+
+ // open a direct output if required by specified parameters
+ // force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ if ((*flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+ *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+
+ // Do internal direct magic here
+ bool offload_disabled = mApmConfigs->isAudioOffloadDisabled();
+ if ((*flags == AUDIO_OUTPUT_FLAG_NONE) &&
+ (stream == AUDIO_STREAM_MUSIC) &&
+ ( !offload_disabled) &&
+ ((config->offload_info.usage == AUDIO_USAGE_MEDIA) ||
+ (config->offload_info.usage == AUDIO_USAGE_GAME))) {
+ audio_output_flags_t old_flags = *flags;
+ *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT);
+ ALOGD("Force Direct Flag .. old flags(0x%x)", old_flags);
+ } else if (*flags == AUDIO_OUTPUT_FLAG_DIRECT &&
+ (offload_disabled || stream != AUDIO_STREAM_MUSIC)) {
+ ALOGD("Offloading is disabled or Stream is not music --> Force Remove Direct Flag");
+ *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE);
+ }
+
+ // check if direct output for pcm/track offload or compress offload already exist
+ bool direct_pcm_already_in_use = false;
+ bool compress_offload_already_in_use = false;
+ if (*flags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc->mFlags == AUDIO_OUTPUT_FLAG_DIRECT) {
+ direct_pcm_already_in_use = true;
+ ALOGD("Direct PCM already in use");
+ }
+ if (desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ compress_offload_already_in_use = true;
+ ALOGD("Compress Offload already in use");
+ }
+ }
+ // prevent direct pcm for non-music stream blindly if direct pcm already in use
+ // for other music stream concurrency is handled after checking direct ouput usage
+ // and checking client
+ if (direct_pcm_already_in_use == true && stream != AUDIO_STREAM_MUSIC) {
+ ALOGD("disabling offload for non music stream as direct pcm is already in use");
+ *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE);
+ }
+ }
+
+ bool forced_deep = false;
+ // only allow deep buffering for music stream type
+ if (stream != AUDIO_STREAM_MUSIC) {
+ *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ } else if (/* stream == AUDIO_STREAM_MUSIC && */
+ (*flags == AUDIO_OUTPUT_FLAG_NONE || *flags == AUDIO_OUTPUT_FLAG_DIRECT ||
+ (*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) &&
+ mApmConfigs->isAudioDeepbufferMediaEnabled() && !isInCall()) {
+ forced_deep = true;
+ }
+
+ if (stream == AUDIO_STREAM_TTS) {
+ *flags = AUDIO_OUTPUT_FLAG_TTS;
+ } else if (devices.types() == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
+ stream == AUDIO_STREAM_MUSIC &&
+ audio_is_linear_pcm(config->format) &&
+ isInCall()) {
+ *flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
+ }
+
+ sp<IOProfile> profile;
+
+ // skip direct output selection if the request can obviously be attached to a mixed output
+ // and not explicitly requested
+ if (((*flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
+ audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX &&
+ audio_channel_count_from_out_mask(channelMask) <= 2) {
+ goto non_direct_output;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled.
+ // This prevents creating an offloaded track and tearing it down immediately after start
+ // when audioflinger detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ //
+ // Supplementary annotation:
+ // For sake of track offload introduced, we need a rollback for both compress offload
+ // and track offload use cases.
+ if ((*flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_DIRECT)) &&
+ (mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {
+ ALOGD("non offloadable effect is enabled, try with non direct output");
+ goto non_direct_output;
+ }
+
+ profile = getProfileForOutput(devices,
+ config->sample_rate,
+ config->format,
+ channelMask,
+ (audio_output_flags_t)*flags,
+ true /* directOnly */);
+
+ if (profile != 0) {
+
+ if (!(*flags & AUDIO_OUTPUT_FLAG_DIRECT) &&
+ (profile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
+ ALOGI("got Direct without requesting ... reject ");
+ profile = NULL;
+ goto non_direct_output;
+ }
+
+ sp<SwAudioOutputDescriptor> outputDesc = NULL;
+ // if multiple concurrent offload decode is supported
+ // do no check for reuse and also don't close previous output if its offload
+ // previous output will be closed during track destruction
+ if (!mApmConfigs->isAudioMultipleOffloadEnable() &&
+ ((*flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)) {
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open by the same client
+ // and configured with same parameters
+ if ((config->sample_rate == desc->mSamplingRate) &&
+ audio_formats_match(config->format, desc->mFormat) &&
+ (channelMask == desc->mChannelMask) &&
+ (session == desc->mDirectClientSession)) {
+ desc->mDirectOpenCount++;
+ ALOGV("getOutputForDevice() reusing direct output %d for session %d",
+ mOutputs.keyAt(i), session);
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+ if (outputDesc != NULL) {
+ if ((((*flags == AUDIO_OUTPUT_FLAG_DIRECT) && direct_pcm_already_in_use) ||
+ ((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+ compress_offload_already_in_use)) &&
+ session != outputDesc->mDirectClientSession) {
+ ALOGV("getOutput() do not reuse direct pcm output because current client (%d) "
+ "is not the same as requesting client (%d) for different output conf",
+ outputDesc->mDirectClientSession, session);
+ goto non_direct_output;
+ }
+ closeOutput(outputDesc->mIoHandle);
+ }
+ }
+ if (!profile->canOpenNewIo()) {
+ goto non_direct_output;
+ }
+
+ outputDesc =
+ new SwAudioOutputDescriptor(profile, mpClientInterface);
+ DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(devices.types());
+ String8 address = outputDevices.size() > 0 ? outputDevices.itemAt(0)->address()
+ : String8("");
+ status = outputDesc->open(config, devices, stream, *flags, &output);
+
+ // only accept an output with the requested parameters
+ if (status != NO_ERROR ||
+ (config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
+ (config->format != AUDIO_FORMAT_DEFAULT &&
+ !audio_formats_match(config->format, outputDesc->mFormat)) ||
+ (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+ ALOGV("getOutputForDevice() failed opening direct output: output %d sample rate %d %d,"
+ "format %d %d, channel mask %04x %04x", output, config->sample_rate,
+ outputDesc->mSamplingRate, config->format, outputDesc->mFormat,
+ channelMask, outputDesc->mChannelMask);
+ //Only close o/p descriptor if successfully opened
+ if (status == NO_ERROR) {
+ outputDesc->close();
+ }
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(config->format) && config->sample_rate <= SAMPLE_RATE_HZ_MAX) {
+ goto non_direct_output;
+ }
+ return AUDIO_IO_HANDLE_NONE;
+ }
+ outputDesc->mDirectOpenCount = 1;
+ outputDesc->mDirectClientSession = session;
+
+ addOutput(output, outputDesc);
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutputForDevice() returns new direct output %d", output);
+ mpClientInterface->onAudioPortListUpdate();
+ return output;
+ }
+
+non_direct_output:
+
+ // A request for HW A/V sync cannot fallback to a mixed output because time
+ // stamps are embedded in audio data
+ if ((*flags & (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ)) != 0) {
+ return AUDIO_IO_HANDLE_NONE;
+ }
+
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // for non direct outputs, only PCM is supported
+ if (audio_is_linear_pcm(config->format)) {
+ // get which output is suitable for the specified stream. The actual
+ // routing change will happen when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
+
+ // at this stage we should ignore the DIRECT flag as no direct output could be found earlier
+ *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+
+ if (forced_deep) {
+ *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ ALOGI("setting force DEEP buffer now ");
+ } else if (*flags == AUDIO_OUTPUT_FLAG_NONE) {
+ // no deep buffer playback is requested hence fallback to primary
+ *flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_PRIMARY);
+ ALOGI("FLAG None hence request for a primary output");
+ }
+
+ output = selectOutput(outputs, *flags, config->format);
+ }
+
+ ALOGW_IF((output == 0), "getOutputForDevice() could not find output for stream %d, "
+ "sampling rate %d, format %#x, channels %#x, flags %#x",
+ stream, config->sample_rate, config->format, channelMask, *flags);
+
+ ALOGV("getOutputForDevice() returns output %d", output);
+
+ return output;
+}
+
+status_t AudioPolicyManagerCustom::getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_unique_id_t riid,
+ audio_session_t session,
+ uid_t uid,
+ const audio_config_base_t *config,
+ audio_input_flags_t flags,
+ audio_port_handle_t *selectedDeviceId,
+ input_type_t *inputType,
+ audio_port_handle_t *portId)
+{
+ audio_source_t inputSource;
+ inputSource = attr->source;
+
+ if (mApmConfigs->isVoiceConcEnabled()) {
+ bool prop_rec_enabled = false, prop_voip_enabled = false;
+ prop_rec_enabled = mApmConfigs->isVoiceRecConcDisabled();
+ prop_voip_enabled = mApmConfigs->isVoiceVOIPConcDisabled();
+
+ if (prop_rec_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //Need to block input request
+ if ((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+ ((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
+ (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+ {
+ switch(inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ ALOGD("voice_conc:Creating input during incall mode for inputSource: %d",
+ inputSource);
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if (prop_voip_enabled) {
+ ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
+ inputSource);
+ return NO_INIT;
+ }
+ break;
+ default:
+ ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
+ inputSource);
+ return NO_INIT;
+ }
+ }
+ }//check for VoIP flag
+ else if (prop_voip_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //Need to block input request
+ if ((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
+ ((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
+ (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
+ {
+ if (inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+ ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
+ return NO_INIT;
+ }
+ }
+ }
+ }
+
+
+ return AudioPolicyManager::getInputForAttr(attr,
+ input,
+ riid,
+ session,
+ uid,
+ config,
+ flags,
+ selectedDeviceId,
+ inputType,
+ portId);
+}
+
+uint32_t AudioPolicyManagerCustom::activeNonSoundTriggerInputsCountOnDevices(audio_devices_t devices) const
+{
+ uint32_t count = 0;
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
+ if (inputDescriptor->isActive() && !inputDescriptor->isSoundTrigger() &&
+ ((devices == AUDIO_DEVICE_IN_DEFAULT) ||
+ ((inputDescriptor->getDeviceType() & devices & ~AUDIO_DEVICE_BIT_IN) != 0))) {
+ count++;
+ }
+ }
+ return count;
+}
+
+status_t AudioPolicyManagerCustom::startInput(audio_port_handle_t portId)
+{
+ ALOGV("%s portId %d", __FUNCTION__, portId);
+
+ sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
+ if (inputDesc == 0) {
+ ALOGW("%s no input for client %d", __FUNCTION__, portId);
+ return BAD_VALUE;
+ }
+
+ audio_io_handle_t input = inputDesc->mIoHandle;
+ sp<RecordClientDescriptor> client = inputDesc->getClient(portId);
+ if (client == NULL) {
+ ALOGW("%s invalid client desc for %d", __FUNCTION__, portId);
+ return BAD_VALUE;
+ }
+
+ if (client->active()) {
+ ALOGW("%s input %d client %d already started", __FUNCTION__, input, client->portId());
+ return INVALID_OPERATION;
+ }
+
+ audio_session_t session = client->session();
+
+ ALOGV("%s input:%d, session:%d)", __FUNCTION__, input, session);
+
+
+// FIXME: disable concurrent capture until UI is ready
+#if 0
+ if (!isConcurentCaptureAllowed(inputDesc, audioSession)) {
+ ALOGW("startInput(%d) failed: other input already started", input);
+ return INVALID_OPERATION;
+ }
+
+ if (isInCall()) {
+ *concurrency |= API_INPUT_CONCURRENCY_CALL;
+ }
+
+ if (mInputs.activeInputsCountOnDevices() != 0) {
+ *concurrency |= API_INPUT_CONCURRENCY_CAPTURE;
+ }
+#endif
+
+ if (mApmConfigs->isRecPlayConcEnabled()) {
+ mIsInputRequestOnProgress = true;
+ DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+ if (mApmConfigs->isRecPlayConcDisabled() && (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0)) {
+ // send update to HAL on record playback concurrency
+ AudioParameter param = AudioParameter();
+ param.add(String8("rec_play_conc_on"), String8("true"));
+ ALOGD("startInput() setParameters rec_play_conc is setting to ON ");
+ mpClientInterface->setParameters(0, param.toString());
+
+ // Call invalidate to reset all opened non ULL audio tracks
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
+ // Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder)
+ if (i != AUDIO_STREAM_ENFORCED_AUDIBLE) {
+ ALOGD("Invalidate on releaseInput for stream :: %d ", i);
+ //FIXME see fixme on name change
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ // close compress tracks
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("ouput desc / profile is NULL");
+ continue;
+ }
+ if (outputDesc->mProfile->getFlags()
+ & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ // close compress sessions
+ ALOGD("calling closeOutput on record conc for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ }
+ }
+ status_t status = inputDesc->start();
+ if (status != NO_ERROR) {
+ return status;
+ }
+
+ // increment activity count before calling getNewInputDevice() below as only active sessions
+ // are considered for device selection
+ inputDesc->setClientActive(client, true);
+
+ // indicate active capture to sound trigger service if starting capture from a mic on
+ // primary HW module
+ sp<DeviceDescriptor> device = getNewInputDevice(inputDesc);
+ setInputDevice(input, device, true /* force */);
+
+ if (inputDesc->activeCount() == 1) {
+ sp<AudioPolicyMix> policyMix = inputDesc->mPolicyMix.promote();
+ // if input maps to a dynamic policy with an activity listener, notify of state change
+ if ((policyMix != NULL)
+ && ((policyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
+ mpClientInterface->onDynamicPolicyMixStateUpdate(policyMix->mDeviceAddress,
+ MIX_STATE_MIXING);
+ }
+
+ DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+ if ((primaryInputDevices.contains(device) && (device->type() & ~AUDIO_DEVICE_BIT_IN)) != 0) {
+ if (mApmConfigs->isVAConcEnabled()) {
+ if (activeNonSoundTriggerInputsCountOnDevices(primaryInputDevices.types()) == 1)
+ SoundTrigger::setCaptureState(true);
+ } else if (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1)
+ SoundTrigger::setCaptureState(true);
+ }
+
+ // automatically enable the remote submix output when input is started if not
+ // used by a policy mix of type MIX_TYPE_RECORDERS
+ // For remote submix (a virtual device), we open only one input per capture request.
+ if (audio_is_remote_submix_device(inputDesc->getDeviceType())) {
+ String8 address = String8("");
+ if (policyMix == NULL) {
+ address = String8("0");
+ } else if (policyMix->mMixType == MIX_TYPE_PLAYERS) {
+ address = policyMix->mDeviceAddress;
+ }
+ if (address != "") {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address, "remote-submix", AUDIO_FORMAT_DEFAULT);
+ }
+ }
+ }
+
+ ALOGV("%s input %d source = %d exit", __FUNCTION__, input, client->source());
+
+ if (mApmConfigs->isRecPlayConcEnabled())
+ mIsInputRequestOnProgress = false;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerCustom::stopInput(audio_port_handle_t portId)
+{
+ status_t status;
+ status = AudioPolicyManager::stopInput(portId);
+ sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
+ if (inputDesc == 0) {
+ ALOGW("stopInput() no input for client %d", portId);
+ return BAD_VALUE;
+ }
+ sp<RecordClientDescriptor> client = inputDesc->getClient(portId);
+ audio_io_handle_t input = inputDesc->mIoHandle;
+
+ ALOGV("stopInput() input %d", input);
+ DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+ if (mApmConfigs->isVAConcEnabled()) {
+ sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
+ if ((primaryInputDevices.contains(inputDesc->getDevice()) &&
+ activeNonSoundTriggerInputsCountOnDevices(primaryInputDevices.types())) == 0) {
+ SoundTrigger::setCaptureState(false);
+ }
+ }
+ if (mApmConfigs->isRecPlayConcEnabled()) {
+ if (mApmConfigs->isRecPlayConcDisabled() &&
+ (mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0)) {
+ //send update to HAL on record playback concurrency
+ AudioParameter param = AudioParameter();
+ param.add(String8("rec_play_conc_on"), String8("false"));
+ ALOGD("stopInput() setParameters rec_play_conc is setting to OFF ");
+ mpClientInterface->setParameters(0, param.toString());
+
+ //call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
+ for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
+ //Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder stop tone)
+ if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE) && (i != AUDIO_STREAM_PATCH)) {
+ ALOGD(" Invalidate on stopInput for stream :: %d ", i);
+ //FIXME see fixme on name change
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+ }
+ return status;
+}
+
+AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
+ : AudioPolicyManager(clientInterface),
+ mFallBackflag(AUDIO_OUTPUT_FLAG_NONE),
+ mHdmiAudioDisabled(false),
+ mHdmiAudioEvent(false),
+ mPrevPhoneState(0),
+ mIsInputRequestOnProgress(false)
+{
+ if (mApmConfigs->useXMLAudioPolicyConf())
+ ALOGD("USE_XML_AUDIO_POLICY_CONF is TRUE");
+ else
+ ALOGD("USE_XML_AUDIO_POLICY_CONF is FALSE");
+
+ if (mApmConfigs->isVoiceConcEnabled())
+ mFallBackflag = getFallBackPath();
+}
+
+}
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
new file mode 100755
index 0000000..61f9821
--- /dev/null
+++ b/policy_hal/AudioPolicyManager.h
@@ -0,0 +1,188 @@
+/*
+ * Copyright (c) 2013-2019 The Linux Foundation. All rights reserved.
+ * Not a contribution.
+ *
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#include <audiopolicy/managerdefault/AudioPolicyManager.h>
+#include <audio_policy_conf.h>
+#include <Volume.h>
+#include "APMConfigHelper.h"
+
+
+namespace android {
+#ifndef AUDIO_EXTN_FORMATS_ENABLED
+#define AUDIO_FORMAT_WMA 0x12000000UL
+#define AUDIO_FORMAT_WMA_PRO 0x13000000UL
+#define AUDIO_FORMAT_FLAC 0x1B000000UL
+#define AUDIO_FORMAT_ALAC 0x1C000000UL
+#define AUDIO_FORMAT_APE 0x1D000000UL
+#endif
+
+#define WMA_STD_NUM_FREQ 7
+#define WMA_STD_NUM_CHANNELS 2
+static uint32_t wmaStdSampleRateTbl[WMA_STD_NUM_FREQ] =
+{
+ 8000, 11025, 16000, 22050, 32000, 44100, 48000
+};
+
+static uint32_t wmaStdMinAvgByteRateTbl[WMA_STD_NUM_FREQ][WMA_STD_NUM_CHANNELS] =
+{
+ {128, 12000},
+ {8016, 8016},
+ {10000, 16000},
+ {16016, 20008},
+ {20000, 24000},
+ {20008, 31960},
+ {63000, 63000}
+};
+
+static uint32_t wmaStdMaxAvgByteRateTbl[WMA_STD_NUM_FREQ][WMA_STD_NUM_CHANNELS] =
+{
+ {8000, 12000},
+ {10168, 10168},
+ {16000, 20000},
+ {20008, 32048},
+ {20000, 48000},
+ {48024, 320032},
+ {256008, 256008}
+};
+
+#define MAX_BITRATE_WMA_PRO 1536000
+#define MAX_BITRATE_WMA_LOSSLESS 1152000
+
+#ifndef AAC_ADTS_OFFLOAD_ENABLED
+#define AUDIO_FORMAT_AAC_ADTS 0x1E000000UL
+#endif
+
+#ifndef AUDIO_EXTN_AFE_PROXY_ENABLED
+#define AUDIO_DEVICE_OUT_PROXY 0x1000000
+#endif
+
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManagerCustom: public AudioPolicyManager
+{
+
+public:
+ AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface);
+
+ virtual ~AudioPolicyManagerCustom() {}
+
+ status_t setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name,
+ audio_format_t encodedFormat);
+ 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_unique_id_t riid,
+ audio_session_t session,
+ uid_t uid,
+ const audio_config_base_t *config,
+ audio_input_flags_t flags,
+ audio_port_handle_t *selectedDeviceId,
+ input_type_t *inputType,
+ audio_port_handle_t *portId);
+ /* count active capture sessions (that are not sound trigger) using one of
+ the specified devices. Ignore devices if AUDIO_DEVICE_IN_DEFAULT is passed */
+ uint32_t activeNonSoundTriggerInputsCountOnDevices(
+ audio_devices_t devices = AUDIO_DEVICE_IN_DEFAULT) const;
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_port_handle_t portId);
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_port_handle_t portId);
+
+ static sp<APMConfigHelper> mApmConfigs;
+
+protected:
+ // check that volume change is permitted, compute and send new volume to audio hardware
+ virtual status_t checkAndSetVolume(IVolumeCurves &curves,
+ VolumeSource volumeSource, int index,
+ const sp<AudioOutputDescriptor>& outputDesc,
+ audio_devices_t device,
+ int delayMs = 0, bool force = false);
+
+ // avoid invalidation for active music stream on previous outputs
+ // which is supported on the new device.
+
+ bool isInvalidationOfMusicStreamNeeded(const audio_attributes_t &attr);
+
+ // Must be called before updateDevicesAndOutputs()
+ void checkOutputForAttributes(const audio_attributes_t &attr);
+
+ // if argument "device" is different from AUDIO_DEVICE_NONE, startSource() will force
+ // the re-evaluation of the output device.
+ status_t startSource(const sp<SwAudioOutputDescriptor>& outputDesc,
+ const sp<TrackClientDescriptor>& client,
+ uint32_t *delayMs);
+ status_t stopSource(const sp<SwAudioOutputDescriptor>& outputDesc,
+ const sp<TrackClientDescriptor>& client);
+ // 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;}
+ static audio_output_flags_t getFallBackPath();
+ int mFallBackflag;
+ //parameter indicates of HDMI speakers disabled
+ bool mHdmiAudioDisabled;
+ //parameter indicates if HDMI plug in/out detected
+ bool mHdmiAudioEvent;
+
+private:
+ // internal method to return the output handle for the given device and format
+ audio_io_handle_t getOutputForDevices(
+ const DeviceVector &devices,
+ audio_session_t session,
+ audio_stream_type_t stream,
+ const audio_config_t *config,
+ audio_output_flags_t *flags,
+ bool forceMutingHaptic = false);
+
+ // 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,
+ std::vector<audio_io_handle_t> *secondaryOutputs);
+
+
+
+
+ // internal method to query hal for whether display-port is connected
+ // and can be used for voip/voice call
+ void chkDpConnAndAllowedForVoice();
+ // Used for voip + voice concurrency usecase
+ int mPrevPhoneState;
+ int mvoice_call_state;
+ // Used for record + playback concurrency
+ bool mIsInputRequestOnProgress;
+};
+};