Revert "policy_hal: Initial change for the new libaudiopolicymanager"
This reverts commit 0670f1668c90399c3e00d6682b56d3e10c2c0074
Change-Id: Ie03ef15f7f6d6e1cf88c4b39b0d9a0362c17a731
diff --git a/policy_hal/Android.mk b/policy_hal/Android.mk
index b6c01a4..7d365d6 100644
--- a/policy_hal/Android.mk
+++ b/policy_hal/Android.mk
@@ -1,26 +1,49 @@
-ifneq ($(USE_LEGACY_AUDIO_POLICY), 1)
-ifeq ($(USE_CUSTOM_AUDIO_POLICY), 1)
+ifeq ($(strip $(BOARD_USES_EXTN_AUDIO_POLICY_MANAGER)),true)
-LOCAL_PATH:= $(call my-dir)
+LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
-LOCAL_SRC_FILES:= \
- AudioPolicyManager.cpp
+LOCAL_SRC_FILES := AudioPolicyManager.cpp
LOCAL_SHARED_LIBRARIES := \
libcutils \
libutils \
liblog
-LOCAL_C_INCLUDES := \
- $(TOPDIR)frameworks/av/services/audiopolicy \
-
LOCAL_STATIC_LIBRARIES := \
libmedia_helper
-LOCAL_MODULE:= libaudiopolicymanager
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+ libaudiopolicy_legacy
+
+LOCAL_MODULE := audio_policy.$(TARGET_BOARD_PLATFORM)
+LOCAL_MODULE_RELATIVE_PATH := hw
+LOCAL_MODULE_TAGS := optional
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FM)),true)
+LOCAL_CFLAGS += -DAUDIO_EXTN_FM_ENABLED
+endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_PROXY_DEVICE)),true)
+LOCAL_CFLAGS += -DAUDIO_EXTN_AFE_PROXY_ENABLED
+endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_INCALL_MUSIC)),true)
+LOCAL_CFLAGS += -DAUDIO_EXTN_INCALL_MUSIC_ENABLED
+endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_SPK)),true)
+LOCAL_CFLAGS += -DAUDIO_EXTN_HDMI_SPK_ENABLED
+endif
+
+
+ifeq ($(strip $(TARGET_BOARD_PLATFORM)),msm8916)
+LOCAL_CFLAGS += -DVOICE_CONCURRENCY
+LOCAL_CFLAGS += -DWFD_CONCURRENCY
+endif
+
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_MULTIPLE_TUNNEL)), true)
+LOCAL_CFLAGS += -DMULTIPLE_OFFLOAD_ENABLED
+endif
+
include $(BUILD_SHARED_LIBRARY)
endif
-endif
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 019161b..2309730 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2009 The Android Open Source Project
@@ -27,180 +27,57 @@
#define ALOGVV(a...) do { } while(0)
#endif
-// A device mask for all audio input devices that are considered "virtual" when evaluating
-// active inputs in getActiveInput()
-#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX
// 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
-#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 "AudioPolicyManager.h"
-#include "audio_policy_conf.h"
+#include <hardware/audio_effect.h>
+#include <hardware/audio.h>
+#include <math.h>
+#include <hardware_legacy/audio_policy_conf.h>
+#include <cutils/properties.h>
-namespace android {
-
-// ----------------------------------------------------------------------------
-// Definitions for audio_policy.conf file parsing
-// ----------------------------------------------------------------------------
-
-struct StringToEnum {
- const char *name;
- uint32_t value;
-};
-
-#define STRING_TO_ENUM(string) { #string, string }
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-const StringToEnum sDeviceNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
-};
-
-const StringToEnum sFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
-};
-
-const StringToEnum sFormatNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
- STRING_TO_ENUM(AUDIO_FORMAT_MP3),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC),
- STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
-};
-
-const StringToEnum sOutChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
-};
-
-const StringToEnum sInChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
-};
-
-const StringToEnum sGainModeNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
-};
-
-
-uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name)
-{
- for (size_t i = 0; i < size; i++) {
- if (strcmp(table[i].name, name) == 0) {
- ALOGV("stringToEnum() found %s", table[i].name);
- return table[i].value;
- }
- }
- return 0;
-}
-
-const char *AudioPolicyManager::enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value)
-{
- for (size_t i = 0; i < size; i++) {
- if (table[i].value == value) {
- return table[i].name;
- }
- }
- return "";
-}
-
-bool AudioPolicyManager::stringToBool(const char *value)
-{
- return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
-}
-
+namespace android_audio_legacy {
// ----------------------------------------------------------------------------
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
-
+const char* AudioPolicyManager::HDMI_SPKR_STR = "hdmi_spkr";
+int AudioPolicyManager::mvoice_call_state = 0;
status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address)
+ AudioSystem::device_connection_state state,
+ const char *device_address)
{
- String8 address = String8(device_address);
+ SortedVector <audio_io_handle_t> outputs;
- ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
+ ALOGD("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+ if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
+ ALOGE("setDeviceConnectionState() invalid address: %s", device_address);
+ return BAD_VALUE;
+ }
+
// handle output devices
if (audio_is_output_device(device)) {
- SortedVector <audio_io_handle_t> outputs;
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
- devDesc->mAddress = address;
- ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
+ if (!mHasA2dp && audio_is_a2dp_device(device)) {
+ ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device);
+ return BAD_VALUE;
+ }
+ if (!mHasUsb && audio_is_usb_device(device)) {
+ ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device);
+ return BAD_VALUE;
+ }
+ if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) {
+ ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device);
+ return BAD_VALUE;
+ }
// save a copy of the opened output descriptors before any output is opened or closed
// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
@@ -208,46 +85,109 @@
switch (state)
{
// handle output device connection
- case AUDIO_POLICY_DEVICE_STATE_AVAILABLE:
- if (index >= 0) {
+ case AudioSystem::DEVICE_STATE_AVAILABLE:
+ if (mAvailableOutputDevices & device) {
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, HDMI_SPKR_STR, MAX_DEVICE_ADDRESS_LEN)) {
+ mHdmiAudioDisabled = false;
+ } else {
+ mHdmiAudioEvent = true;
+ }
+ }
+#endif
ALOGW("setDeviceConnectionState() device already connected: %x", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() connecting device %x", device);
- if (checkOutputsForDevice(device, state, outputs, address) != NO_ERROR) {
+ if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
return INVALID_OPERATION;
}
- // outputs should never be empty here
- ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
- "checkOutputsForDevice() returned no outputs but status OK");
- ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
+ ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs",
outputs.size());
// register new device as available
- index = mAvailableOutputDevices.add(devDesc);
- if (index >= 0) {
- mAvailableOutputDevices[index]->mId = nextUniqueId();
- HwModule *module = getModuleForDevice(device);
- ALOG_ASSERT(module != NULL, "setDeviceConnectionState():"
- "could not find HW module for device %08x", device);
- mAvailableOutputDevices[index]->mModule = module;
- } else {
- return NO_MEMORY;
- }
+ mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device);
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, HDMI_SPKR_STR, MAX_DEVICE_ADDRESS_LEN)) {
+ mHdmiAudioDisabled = false;
+ } else {
+ mHdmiAudioEvent = true;
+ }
+ if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
+ mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
+ }
+ }
+#endif
+ if (!outputs.isEmpty()) {
+ String8 paramStr;
+ if (mHasA2dp && audio_is_a2dp_device(device)) {
+ // handle A2DP device connection
+ AudioParameter param;
+ param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address));
+ paramStr = param.toString();
+ mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+ mA2dpSuspended = false;
+ } else if (audio_is_bluetooth_sco_device(device)) {
+ // handle SCO device connection
+ mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+ } else if (mHasUsb && audio_is_usb_device(device)) {
+ // handle USB device connection
+ mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+ paramStr = mUsbCardAndDevice;
+ }
+ // not currently handling multiple simultaneous submixes: ignoring remote submix
+ // case and address
+ if (!paramStr.isEmpty()) {
+ for (size_t i = 0; i < outputs.size(); i++) {
+ mpClientInterface->setParameters(outputs[i], paramStr);
+ }
+ }
+ }
break;
// handle output device disconnection
- case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
- if (index < 0) {
+ case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+ if (!(mAvailableOutputDevices & device)) {
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, HDMI_SPKR_STR, MAX_DEVICE_ADDRESS_LEN)) {
+ mHdmiAudioDisabled = true;
+ } else {
+ mHdmiAudioEvent = false;
+ }
+ }
+#endif
ALOGW("setDeviceConnectionState() device not connected: %x", device);
return INVALID_OPERATION;
}
ALOGV("setDeviceConnectionState() disconnecting device %x", device);
// remove device from available output devices
- mAvailableOutputDevices.remove(devDesc);
+ mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
- checkOutputsForDevice(device, state, outputs, address);
+#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
+ if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
+ if (!strncmp(device_address, HDMI_SPKR_STR, MAX_DEVICE_ADDRESS_LEN)) {
+ mHdmiAudioDisabled = true;
+ } else {
+ mHdmiAudioEvent = false;
+ }
+ }
+#endif
+ checkOutputsForDevice(device, state, outputs);
+ if (mHasA2dp && audio_is_a2dp_device(device)) {
+ // handle A2DP device disconnection
+ mA2dpDeviceAddress = "";
+ mA2dpSuspended = false;
+ } else if (audio_is_bluetooth_sco_device(device)) {
+ // handle SCO device disconnection
+ mScoDeviceAddress = "";
+ } else if (mHasUsb && audio_is_usb_device(device)) {
+ // handle USB device disconnection
+ mUsbCardAndDevice = "";
+ }
// not currently handling multiple simultaneous submixes: ignoring remote submix
// case and address
} break;
@@ -257,8 +197,6 @@
return BAD_VALUE;
}
- // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
- // output is suspended before any tracks are moved to it
checkA2dpSuspend();
checkOutputForAllStrategies();
// outputs must be closed after checkOutputForAllStrategies() is executed
@@ -267,122 +205,1180 @@
AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
// close unused outputs after device disconnection or direct outputs that have been
// opened by checkOutputsForDevice() to query dynamic parameters
- if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+ if ((state == AudioSystem::DEVICE_STATE_UNAVAILABLE) ||
(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
(desc->mDirectOpenCount == 0))) {
closeOutput(outputs[i]);
}
}
- // check again after closing A2DP output to reset mA2dpSuspended if needed
- checkA2dpSuspend();
}
updateDevicesAndOutputs();
+ audio_devices_t newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+#ifdef AUDIO_EXTN_FM_ENABLED
+ if(device == AUDIO_DEVICE_OUT_FM) {
+ if (state == AudioSystem::DEVICE_STATE_AVAILABLE) {
+ mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AudioSystem::MUSIC, 1);
+ newDevice = (audio_devices_t)(getNewDevice(mPrimaryOutput, false) | AUDIO_DEVICE_OUT_FM);
+ } else {
+ mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AudioSystem::MUSIC, -1);
+ }
+
+ AudioParameter param = AudioParameter();
+ param.addInt(String8("handle_fm"), (int)newDevice);
+ ALOGV("setDeviceConnectionState() setParameters handle_fm");
+ mpClientInterface->setParameters(mPrimaryOutput, param.toString());
+ }
+#endif
for (size_t i = 0; i < mOutputs.size(); i++) {
// 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.
+ audio_devices_t cachedDevice = getNewDevice(mOutputs.keyAt(i), true /*fromCache*/);
+ AudioOutputDescriptor *desc = mOutputs.valueFor(mOutputs.keyAt(i));
+ if (cachedDevice == AUDIO_DEVICE_OUT_SPEAKER &&
+ device == AUDIO_DEVICE_OUT_PROXY &&
+ (desc->mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
+ ALOGI("Avoid routing touch tone to spkr as proxy is being disconnected");
+ break;
+ }
setOutputDevice(mOutputs.keyAt(i),
- getNewOutputDevice(mOutputs.keyAt(i), true /*fromCache*/),
+ cachedDevice,
!mOutputs.valueAt(i)->isDuplicated(),
0);
}
- mpClientInterface->onAudioPortListUpdate();
- return NO_ERROR;
- } // end if is output device
-
+ if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO ||
+ device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
+ device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if(device == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET){
+ device = AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET;
+ } else {
+ return NO_ERROR;
+ }
+ }
// handle input devices
if (audio_is_input_device(device)) {
- SortedVector <audio_io_handle_t> inputs;
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
- devDesc->mAddress = address;
- ssize_t index = mAvailableInputDevices.indexOf(devDesc);
switch (state)
{
// handle input device connection
- case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
- if (index >= 0) {
+ case AudioSystem::DEVICE_STATE_AVAILABLE: {
+ if (mAvailableInputDevices & device) {
ALOGW("setDeviceConnectionState() device already connected: %d", device);
return INVALID_OPERATION;
}
- HwModule *module = getModuleForDevice(device);
- if (module == NULL) {
- ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
- device);
- return INVALID_OPERATION;
+ mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN);
}
- if (checkInputsForDevice(device, state, inputs, address) != NO_ERROR) {
- return INVALID_OPERATION;
- }
-
- index = mAvailableInputDevices.add(devDesc);
- if (index >= 0) {
- mAvailableInputDevices[index]->mId = nextUniqueId();
- mAvailableInputDevices[index]->mModule = module;
- } else {
- return NO_MEMORY;
- }
- } break;
+ break;
// handle input device disconnection
- case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
- if (index < 0) {
+ case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+ if (!(mAvailableInputDevices & device)) {
ALOGW("setDeviceConnectionState() device not connected: %d", device);
return INVALID_OPERATION;
}
- checkInputsForDevice(device, state, inputs, address);
- mAvailableInputDevices.remove(devDesc);
- } break;
+ mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device);
+ } break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
- closeAllInputs();
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+ audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+ if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+ ALOGV("setDeviceConnectionState() changing device from %x to %x for input %d",
+ inputDesc->mDevice, newDevice, activeInput);
+ inputDesc->mDevice = newDevice;
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+ mpClientInterface->setParameters(activeInput, param.toString());
+ }
+ }
- mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
- } // end if is input device
+ }
ALOGW("setDeviceConnectionState() invalid device: %x", device);
return BAD_VALUE;
}
-audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
- const char *device_address)
+void AudioPolicyManager::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
{
- audio_policy_dev_state_t state = AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
- String8 address = String8(device_address);
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
- devDesc->mAddress = String8(device_address);
- ssize_t index;
- DeviceVector *deviceVector;
+ ALOGD("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
- if (audio_is_output_device(device)) {
- deviceVector = &mAvailableOutputDevices;
- } else if (audio_is_input_device(device)) {
- deviceVector = &mAvailableInputDevices;
- } else {
- ALOGW("getDeviceConnectionState() invalid device type %08x", device);
- return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ bool forceVolumeReeval = false;
+ switch(usage) {
+ case AudioSystem::FOR_COMMUNICATION:
+ if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
+ config != AudioSystem::FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+ return;
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AudioSystem::FOR_MEDIA:
+ if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
+#ifdef AUDIO_EXTN_FM_ENABLED
+ config != AudioSystem::FORCE_SPEAKER &&
+#endif
+ config != AudioSystem::FORCE_WIRED_ACCESSORY &&
+ config != AudioSystem::FORCE_ANALOG_DOCK &&
+ config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE &&
+ config != AudioSystem::FORCE_NO_BT_A2DP) {
+ ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AudioSystem::FOR_RECORD:
+ if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY &&
+ config != AudioSystem::FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AudioSystem::FOR_DOCK:
+ if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK &&
+ config != AudioSystem::FORCE_BT_DESK_DOCK &&
+ config != AudioSystem::FORCE_WIRED_ACCESSORY &&
+ config != AudioSystem::FORCE_ANALOG_DOCK &&
+ config != AudioSystem::FORCE_DIGITAL_DOCK) {
+ ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AudioSystem::FOR_SYSTEM:
+ if (config != AudioSystem::FORCE_NONE &&
+ config != AudioSystem::FORCE_SYSTEM_ENFORCED) {
+ ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ default:
+ ALOGW("setForceUse() invalid usage %d", usage);
+ break;
}
- index = deviceVector->indexOf(devDesc);
- if (index >= 0) {
- return AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
- } else {
- return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ // check for device and output changes triggered by new force usage
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ for (int i = mOutputs.size() -1; i >= 0; i--) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
+ setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+ applyStreamVolumes(output, newDevice, 0, true);
+ }
+ }
+
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+ audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+ if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+ ALOGV("setForceUse() changing device from %x to %x for input %d",
+ inputDesc->mDevice, newDevice, activeInput);
+ inputDesc->mDevice = newDevice;
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+ mpClientInterface->setParameters(activeInput, param.toString());
+ }
+ }
+
+}
+
+audio_io_handle_t AudioPolicyManager::getInput(int inputSource,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ AudioSystem::audio_in_acoustics acoustics)
+{
+ audio_io_handle_t input = 0;
+ audio_devices_t device = getDeviceForInputSource(inputSource);
+
+ ALOGD("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
+ inputSource, samplingRate, format, channelMask, acoustics);
+
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGW("getInput() could not find device for inputSource %d", inputSource);
+ return 0;
+ }
+
+#ifdef VOICE_CONCURRENCY
+
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_rec_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.record.conc.disabled", propValue, NULL)) {
+ prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if (prop_rec_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ //some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
+ //Need to block input request
+ if((AudioSystem::MODE_IN_CALL == mPhoneState) ||
+ ((AudioSystem::MODE_IN_CALL == mPrevPhoneState) &&
+ (AudioSystem::MODE_IN_COMMUNICATION == mPhoneState)))
+ {
+ switch(inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ ALOGD("Creating input during incall mode for inputSource: %d ",inputSource);
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if(prop_voip_enabled) {
+ ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
+ return 0;
+ }
+ break;
+ default:
+ ALOGD("BLOCKING input during incall mode for inputSource: %d ",inputSource);
+ return 0;
+ }
+ }
+ }//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((AudioSystem::MODE_IN_CALL == mPhoneState) ||
+ ((AudioSystem::MODE_IN_CALL == mPrevPhoneState) &&
+ (AudioSystem::MODE_IN_COMMUNICATION == mPhoneState)))
+ {
+ if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+ ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
+ return 0;
+ }
+ }
+ }
+
+#endif
+ IOProfile *profile = getInputProfile(device,
+ samplingRate,
+ format,
+ channelMask);
+ if (profile == NULL) {
+ ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d,"
+ "channelMask %04x",
+ device, samplingRate, format, channelMask);
+ return 0;
+ }
+
+ if (profile->mModule->mHandle == 0) {
+ ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
+ return 0;
+ }
+
+ AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
+
+ inputDesc->mInputSource = inputSource;
+ inputDesc->mDevice = device;
+ inputDesc->mSamplingRate = samplingRate;
+ inputDesc->mFormat = (audio_format_t)format;
+ inputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
+ inputDesc->mRefCount = 0;
+ input = mpClientInterface->openInput(profile->mModule->mHandle,
+ &inputDesc->mDevice,
+ &inputDesc->mSamplingRate,
+ &inputDesc->mFormat,
+ &inputDesc->mChannelMask);
+
+ // only accept input with the exact requested set of parameters
+ if (input == 0 ||
+ (samplingRate != inputDesc->mSamplingRate) ||
+ (format != inputDesc->mFormat) ||
+ (channelMask != inputDesc->mChannelMask)) {
+ ALOGV("getInput() failed opening input: samplingRate %d, format %d, channelMask %d",
+ samplingRate, format, channelMask);
+ if (input != 0) {
+ mpClientInterface->closeInput(input);
+ }
+ delete inputDesc;
+ return 0;
+ }
+ mInputs.add(input, inputDesc);
+ ALOGD("getInput() returns input %d", input);
+
+ return input;
+}
+
+AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(AudioSystem::stream_type stream)
+{
+ // stream to strategy mapping
+ switch (stream) {
+ case AudioSystem::VOICE_CALL:
+ case AudioSystem::BLUETOOTH_SCO:
+ return STRATEGY_PHONE;
+ case AudioSystem::RING:
+ case AudioSystem::ALARM:
+ return STRATEGY_SONIFICATION;
+ case AudioSystem::NOTIFICATION:
+ return STRATEGY_SONIFICATION_RESPECTFUL;
+ case AudioSystem::DTMF:
+ return STRATEGY_DTMF;
+ default:
+ ALOGE("unknown stream type");
+ case AudioSystem::SYSTEM:
+ // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+ // while key clicks are played produces a poor result
+ case AudioSystem::TTS:
+ case AudioSystem::MUSIC:
+#ifdef AUDIO_EXTN_INCALL_MUSIC_ENABLED
+ case AudioSystem::INCALL_MUSIC:
+#endif
+#ifdef QCOM_INCALL_MUSIC_ENABLED
+ case AudioSystem::INCALL_MUSIC:
+#endif
+ return STRATEGY_MEDIA;
+ case AudioSystem::ENFORCED_AUDIBLE:
+ return STRATEGY_ENFORCED_AUDIBLE;
+ }
+
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ if (fromCache) {
+ ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
+ strategy, mDeviceForStrategy[strategy]);
+ return mDeviceForStrategy[strategy];
+ }
+
+ switch (strategy) {
+
+ case STRATEGY_SONIFICATION_RESPECTFUL:
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ } else if (isStreamActiveRemotely(AudioSystem::MUSIC,
+ SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing on a remote device, use the the sonification behavior.
+ // Note that we test this usecase before testing if media is playing because
+ // the isStreamActive() method only informs about the activity of a stream, not
+ // if it's for local playback. Note also that we use the same delay between both tests
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ } else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing (or has recently played), use the same device
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ } else {
+ // when media is not playing anymore, fall back on the sonification behavior
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ }
+
+ break;
+
+ case STRATEGY_DTMF:
+ if (!isInCall()) {
+ // when off call, DTMF strategy follows the same rules as MEDIA strategy
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ break;
+ }
+ // when in call, DTMF and PHONE strategies follow the same rules
+ // FALL THROUGH
+
+ case STRATEGY_PHONE:
+ // for phone strategy, we first consider the forced use and then the available devices by order
+ // of priority
+ switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
+ case AudioSystem::FORCE_BT_SCO:
+ if (!isInCall() || strategy != STRATEGY_DTMF) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+ if (device) break;
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ // FALL THROUGH
+
+ default: // FORCE_NONE
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+ if (mHasA2dp && !isInCall() &&
+ (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0) && !mA2dpSuspended) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ if (device) break;
+ if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ }
+
+ // Allow voice call on USB ANLG DOCK headset
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
+ if (device) break;
+ device = mDefaultOutputDevice;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
+ }
+ break;
+
+ case AudioSystem::FORCE_SPEAKER:
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+ // A2DP speaker when forcing to speaker output
+ if (mHasA2dp && !isInCall() &&
+ (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0) && !mA2dpSuspended) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ if (device) break;
+ }
+ if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device) break;
+ device = mDefaultOutputDevice;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
+ }
+ break;
+ }
+ // FIXME: Why do need to replace with speaker? If voice call is active
+ // We should use device from STRATEGY_PHONE
+#ifdef AUDIO_EXTN_FM_ENABLED
+ if (mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM) {
+ if (mForceUse[AudioSystem::FOR_MEDIA] == AudioSystem::FORCE_SPEAKER) {
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ }
+#endif
+ break;
+
+ case STRATEGY_SONIFICATION:
+
+ // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+ // handleIncallSonification().
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+ break;
+ }
+ // FALL THROUGH
+
+ case STRATEGY_ENFORCED_AUDIBLE:
+ // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
+ // except:
+ // - when in call where it doesn't default to STRATEGY_PHONE behavior
+ // - in countries where not enforced in which case it follows STRATEGY_MEDIA
+
+ if ((strategy == STRATEGY_SONIFICATION) ||
+ (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
+ }
+ }
+ // The second device used for sonification is the same as the device used by media strategy
+ // FALL THROUGH
+
+ case STRATEGY_MEDIA: {
+ uint32_t device2 = AUDIO_DEVICE_NONE;
+
+ if (isInCall() && (device == AUDIO_DEVICE_NONE)) {
+ // when in call, get the device for Phone strategy
+ device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+ break;
+ }
+#ifdef AUDIO_EXTN_FM_ENABLED
+ if (mForceUse[AudioSystem::FOR_MEDIA] == AudioSystem::FORCE_SPEAKER) {
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ break;
+ }
+#endif
+
+ if (strategy != STRATEGY_SONIFICATION) {
+ // no sonification on remote submix (e.g. WFD)
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0) && !mA2dpSuspended) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ }
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ }
+ if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+ && (device2 == AUDIO_DEVICE_NONE)) {
+ // no sonification on aux digital (e.g. HDMI)
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AudioSystem::FOR_DOCK] == AudioSystem::FORCE_ANALOG_DOCK)
+ && (strategy != STRATEGY_SONIFICATION)) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ }
+#ifdef AUDIO_EXTN_FM_ENABLED
+ if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+ && (device2 == AUDIO_DEVICE_NONE)) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM_TX;
+ }
+#endif
+#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
+ if ((strategy != STRATEGY_SONIFICATION) && (device == AUDIO_DEVICE_NONE)
+ && (device2 == AUDIO_DEVICE_NONE)) {
+ // no sonification on WFD sink
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY;
+ }
+#endif
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+
+ // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
+ // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
+ device |= device2;
+ if (device) break;
+ device = mDefaultOutputDevice;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
+ }
+ } break;
+
+ default:
+ ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+ break;
+ }
+
+ ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+ return device;
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForInputSource(int inputSource)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ break;
+ }
+ // FALL THROUGH
+
+ case AUDIO_SOURCE_DEFAULT:
+ case AUDIO_SOURCE_MIC:
+ case AUDIO_SOURCE_VOICE_RECOGNITION:
+ case AUDIO_SOURCE_HOTWORD:
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
+ mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET) {
+ device = AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_CAMCORDER:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ }
+ break;
+ case AUDIO_SOURCE_REMOTE_SUBMIX:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+#ifdef AUDIO_EXTN_FM_ENABLED
+ case AUDIO_SOURCE_FM_RX:
+ device = AUDIO_DEVICE_IN_FM_RX;
+ break;
+ case AUDIO_SOURCE_FM_RX_A2DP:
+ device = AUDIO_DEVICE_IN_FM_RX_A2DP;
+ break;
+#endif
+ default:
+ ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
+ break;
+ }
+ ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+ return device;
+}
+
+AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
+{
+ switch(getDeviceForVolume(device)) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ return DEVICE_CATEGORY_EARPIECE;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+#ifdef AUDIO_EXTN_FM_ENABLED
+ case AUDIO_DEVICE_OUT_FM:
+#endif
+ return DEVICE_CATEGORY_HEADSET;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+ case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+ case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+ case AUDIO_DEVICE_OUT_USB_DEVICE:
+ case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
+ case AUDIO_DEVICE_OUT_PROXY:
+#endif
+ default:
+ return DEVICE_CATEGORY_SPEAKER;
}
}
-void AudioPolicyManager::setPhoneState(audio_mode_t state)
+status_t AudioPolicyManager::checkAndSetVolume(int stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
{
- ALOGV("setPhoneState() state %d", state);
+ ALOGV("checkAndSetVolume: index %d output %d device %x", index, output, device);
+ // do not change actual stream volume if the stream is muted
+ if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+ ALOGVV("checkAndSetVolume() stream %d muted count %d",
+ stream, mOutputs.valueFor(output)->mMuteCount[stream]);
+ return NO_ERROR;
+ }
+
+ // do not change in call volume if bluetooth is connected and vice versa
+ if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
+ (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
+ ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+ stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
+ return INVALID_OPERATION;
+ }
+
+ float volume = computeVolume(stream, index, output, device);
+ // We actually change the volume if:
+ // - the float value returned by computeVolume() changed
+ // - the force flag is set
+ if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
+ force) {
+ mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+ ALOGV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+ // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+ // enabled
+ if (stream == AudioSystem::BLUETOOTH_SCO) {
+ mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
+#ifdef AUDIO_EXTN_FM_ENABLED
+ } else if (stream == AudioSystem::MUSIC &&
+ output == mPrimaryOutput) {
+ float fmVolume = -1.0;
+ fmVolume = computeVolume(stream, index, output, device);
+ if (fmVolume >= 0) {
+ AudioParameter param = AudioParameter();
+ param.addFloat(String8("fm_volume"), fmVolume);
+ ALOGV("checkAndSetVolume setParameters fm_volume, volume=:%f delay=:%d",fmVolume,delayMs*2);
+ //Double delayMs to avoid sound burst while device switch.
+ mpClientInterface->setParameters(mPrimaryOutput, param.toString(), delayMs*2);
+ }
+#endif
+ }
+ mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
+ }
+
+ if (stream == AudioSystem::VOICE_CALL ||
+ stream == AudioSystem::BLUETOOTH_SCO) {
+ float voiceVolume;
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (stream == AudioSystem::VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+ } else {
+ voiceVolume = 1.0;
+ }
+
+ if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ mLastVoiceVolume = voiceVolume;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+
+float AudioPolicyManager::computeVolume(int stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device)
+{
+ float volume = 1.0;
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ // if volume is not 0 (not muted), force media volume to max on digital output
+ if (stream == AudioSystem::MUSIC &&
+ index != mStreams[stream].mIndexMin &&
+ (device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
+ device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET ||
+ device == AUDIO_DEVICE_OUT_USB_ACCESSORY ||
+#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
+ device == AUDIO_DEVICE_OUT_PROXY ||
+#endif
+ device == AUDIO_DEVICE_OUT_USB_DEVICE )) {
+ return 1.0;
+ }
+#ifdef AUDIO_EXTN_INCALL_MUSIC_ENABLED
+ if (stream == AudioSystem::INCALL_MUSIC) {
+ return 1.0;
+ }
+#endif
+ return AudioPolicyManagerBase::computeVolume(stream, index, output, device);
+}
+
+
+audio_io_handle_t AudioPolicyManager::getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ AudioSystem::output_flags flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_io_handle_t output = 0;
+ uint32_t latency = 0;
+ routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ IOProfile *profile = NULL;
+
+#ifdef VOICE_CONCURRENCY
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_play_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
+ prop_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if (prop_play_enabled && mvoice_call_state) {
+ //check if voice call is active / running in background
+ if((AudioSystem::MODE_IN_CALL == mPhoneState) ||
+ ((AudioSystem::MODE_IN_CALL == mPrevPhoneState)
+ && (AudioSystem::MODE_IN_COMMUNICATION == mPhoneState)))
+ {
+ if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
+ if(prop_voip_enabled) {
+ ALOGD(" IN call mode returing no output .. for VoIP usecase flags: %x ", flags );
+ // flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ return 0;
+ }
+ }
+ else {
+ ALOGD(" IN call mode adding ULL flags .. flags: %x ", flags );
+ flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ }
+ }
+ } 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((AudioSystem::MODE_IN_CALL == mPhoneState) ||
+ ((AudioSystem::MODE_IN_CALL == mPrevPhoneState)
+ && (AudioSystem::MODE_IN_COMMUNICATION == mPhoneState)))
+ {
+ if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
+ ALOGD(" IN call mode returing no output .. for VoIP usecase flags: %x ", flags );
+ // flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ return 0;
+ }
+ }
+ }
+#endif
+
+#ifdef WFD_CONCURRENCY
+ if ((mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY)
+ && (stream != AudioSystem::MUSIC)) {
+ ALOGV(" WFD mode adding ULL flags for non music stream.. flags: %x ", flags );
+ //For voip paths
+ if(flags & AudioSystem::OUTPUT_FLAG_DIRECT)
+ flags = AudioSystem::OUTPUT_FLAG_DIRECT;
+ else //route every thing else to ULL path
+ flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_FAST;
+ }
+#endif
+
+ ALOGD(" getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x ",
+ device, stream, samplingRate, format, channelMask, flags);
+
+
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ ALOGV("getOutput() opening test output");
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mSamplingRate = mTestSamplingRate;
+ outputDesc->mFormat = mTestFormat;
+ outputDesc->mChannelMask = mTestChannels;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags,
+ offloadInfo);
+ if (mTestOutputs[mCurOutput]) {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"),mCurOutput);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+ addOutput(mTestOutputs[mCurOutput], outputDesc);
+ }
+ }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+
+ // 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 = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+
+ if ((format == AudioSystem::PCM_16_BIT) &&(AudioSystem::popCount(channelMask) > 2)) {
+ ALOGV("owerwrite flag(%x) for PCM16 multi-channel(CM:%x) playback", flags ,channelMask);
+ flags = (AudioSystem::output_flags)AUDIO_OUTPUT_FLAG_DIRECT;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if ((((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+ !isNonOffloadableEffectEnabled()) &&
+ flags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ profile = getProfileForDirectOutput(device,
+ samplingRate,
+ format,
+ channelMask,
+ (audio_output_flags_t)flags);
+ }
+
+ if (profile != NULL) {
+ AudioOutputDescriptor *outputDesc = NULL;
+
+#ifdef MULTIPLE_OFFLOAD_ENABLED
+ bool multiOffloadEnabled = false;
+ char value[PROPERTY_VALUE_MAX] = {0};
+ property_get("audio.offload.multiple.enabled", value, NULL);
+ if (atoi(value) || !strncmp("true", value, 4))
+ multiOffloadEnabled = true;
+ // 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 (multiOffloadEnabled)
+ goto get_output__new_output_desc;
+#endif
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open and configured with same parameters
+ if ((samplingRate == outputDesc->mSamplingRate) &&
+ (format == outputDesc->mFormat) &&
+ (channelMask == outputDesc->mChannelMask)) {
+ outputDesc->mDirectOpenCount++;
+ ALOGD("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mId);
+ }
+get_output__new_output_desc:
+ outputDesc = new AudioOutputDescriptor(profile);
+ outputDesc->mDevice = device;
+ outputDesc->mSamplingRate = samplingRate;
+ outputDesc->mFormat = (audio_format_t)format;
+ outputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+ outputDesc->mRefCount[stream] = 0;
+ outputDesc->mStopTime[stream] = 0;
+ outputDesc->mDirectOpenCount = 1;
+ output = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags,
+ offloadInfo);
+
+ // only accept an output with the requested parameters
+ if (output == 0 ||
+ (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+ (format != 0 && format != outputDesc->mFormat) ||
+ (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+ ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+ "format %d %d, channelMask %04x %04x", output, samplingRate,
+ outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+ outputDesc->mChannelMask);
+ if (output != 0) {
+ mpClientInterface->closeOutput(output);
+ }
+ delete outputDesc;
+ return 0;
+ }
+ audio_io_handle_t srcOutput = getOutputForEffect();
+ addOutput(output, outputDesc);
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput == output) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+ }
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutput() returns new direct output %d", output);
+ return output;
+ }
+
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // for non direct outputs, only PCM is supported
+ if (audio_is_linear_pcm((audio_format_t)format)) {
+ // get which output is suitable for the specified stream. The actual
+ // routing change will happen when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+ output = selectOutput(outputs, flags);
+ }
+ ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+ "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+ ALOGD("getOutput() returns output %d", output);
+
+ return output;
+}
+
+
+// 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 AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+ ALOGD("copl: isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+ " BitRate=%u, duration=%lld us, has_video=%d",
+ offloadInfo.sample_rate, offloadInfo.channel_mask,
+ offloadInfo.format,
+ offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+ offloadInfo.has_video);
+
+#ifdef VOICE_CONCURRENCY
+ char concpropValue[PROPERTY_VALUE_MAX];
+ if(property_get("voice.playback.conc.disabled", concpropValue, NULL)) {
+ bool propenabled = atoi(concpropValue) || !strncmp("true", concpropValue, 4);
+ if (propenabled) {
+ if(isInCall())
+ {
+ ALOGD("\n copl: blocking compress offload on call mode\n");
+ return false;
+ }
+ }
+ }
+
+#endif
+ // Check if stream type is music, then only allow offload as of now.
+ if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+ {
+ ALOGD("copl: isOffloadSupported: stream_type != MUSIC, returning false");
+ return false;
+ }
+
+ char propValue[PROPERTY_VALUE_MAX];
+ bool pcmOffload = false;
+ if (audio_is_offload_pcm(offloadInfo.format)) {
+ if(property_get("audio.offload.pcm.enable", propValue, NULL)) {
+ bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ if (prop_enabled) {
+ ALOGW("PCM offload property is enabled");
+ pcmOffload = true;
+ }
+ }
+ if (!pcmOffload) {
+ ALOGD("copl: PCM offload disabled by property audio.offload.pcm.enable");
+ return false;
+ }
+ }
+
+ if (!pcmOffload) {
+ // Check if offload has been disabled
+ if (property_get("audio.offload.disable", propValue, "0")) {
+ if (atoi(propValue) != 0) {
+ ALOGD("copl: offload disabled by audio.offload.disable=%s", propValue );
+ return false;
+ }
+ }
+
+ //check if it's multi-channel AAC format
+ if (AudioSystem::popCount(offloadInfo.channel_mask) > 2
+ && offloadInfo.format == AUDIO_FORMAT_AAC) {
+ ALOGD("copl: offload disabled for multi-channel AAC format");
+ return false;
+ }
+
+ if (offloadInfo.has_video)
+ {
+ if(property_get("av.offload.enable", propValue, NULL)) {
+ bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ if (!prop_enabled) {
+ ALOGW("offload disabled by av.offload.enable = %s ", propValue );
+ return false;
+ }
+ } else {
+ return false;
+ }
+
+ if(offloadInfo.is_streaming) {
+ if (property_get("av.streaming.offload.enable", propValue, NULL)) {
+ bool prop_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ if (!prop_enabled) {
+ ALOGW("offload disabled by av.streaming.offload.enable = %s ", propValue );
+ return false;
+ }
+ } else {
+ //Do not offload AV streamnig if the property is not defined
+ return false;
+ }
+ }
+ ALOGD("copl: isOffloadSupported: has_video == true, property\
+ set to enable offload");
+ }
+ }
+
+ //If duration is less than minimum value defined in property, return false
+ if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+ if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+ ALOGD("copl: Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+ return false;
+ }
+ } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+ ALOGD("copl: Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+ //duration checks only valid for MP3/AAC formats,
+ //do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
+ if (offloadInfo.format == AUDIO_FORMAT_MP3 || offloadInfo.format == AUDIO_FORMAT_AAC || pcmOffload)
+ 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 (isNonOffloadableEffectEnabled()) {
+ return false;
+ }
+
+ // See if there is a profile to support this.
+ // AUDIO_DEVICE_NONE
+ IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ offloadInfo.sample_rate,
+ offloadInfo.format,
+ offloadInfo.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ ALOGD("copl: isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
+ return (profile != NULL);
+}
+
+void AudioPolicyManager::setPhoneState(int state)
+
+{
+ ALOGD("setPhoneState() state %d", state);
audio_devices_t newDevice = AUDIO_DEVICE_NONE;
- if (state < 0 || state >= AUDIO_MODE_CNT) {
+ if (state < 0 || state >= AudioSystem::NUM_MODES) {
ALOGW("setPhoneState() invalid state %d", state);
return;
}
@@ -396,8 +1392,8 @@
// pertaining to sonification strategy see handleIncallSonification()
if (isInCall()) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- handleIncallSonification((audio_stream_type_t)stream, false, true);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ handleIncallSonification(stream, false, true);
}
}
@@ -405,6 +1401,7 @@
int oldState = mPhoneState;
mPhoneState = state;
bool force = false;
+ int voice_call_state = 0;
// are we entering or starting a call
if (!isStateInCall(oldState) && isStateInCall(state)) {
@@ -433,7 +1430,7 @@
}
// check for device and output changes triggered by new phone state
- newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
checkA2dpSuspend();
checkOutputForAllStrategies();
updateDevicesAndOutputs();
@@ -445,6 +1442,135 @@
if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
newDevice = hwOutputDesc->device();
}
+#ifdef VOICE_CONCURRENCY
+ char propValue[PROPERTY_VALUE_MAX];
+ bool prop_playback_enabled = false, prop_rec_enabled=false, prop_voip_enabled = false;
+
+ if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
+ prop_playback_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.record.conc.disabled", propValue, NULL)) {
+ prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
+ prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
+ }
+
+ bool mode_in_call = (AudioSystem::MODE_IN_CALL != oldState) && (AudioSystem::MODE_IN_CALL == state);
+ //query if it is a actual voice call initiated by telephony
+ if (mode_in_call) {
+ String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0, String8("in_call"));
+ AudioParameter result = AudioParameter(valueStr);
+ if (result.getInt(String8("in_call"), voice_call_state) == NO_ERROR)
+ ALOGD("SetPhoneState: Voice call state = %d", voice_call_state);
+ }
+
+ if (mode_in_call && voice_call_state) {
+ ALOGD("Entering to call mode oldState :: %d state::%d ",oldState, state);
+ mvoice_call_state = voice_call_state;
+ if (prop_playback_enabled) {
+ //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 = AudioSystem::SYSTEM; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ ALOGV(" Invalidate on call mode for stream :: %d ", i);
+ //FIXME see fixme on name change
+ mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
+ 0 /* ignored */);
+ }
+ }
+
+ if (prop_rec_enabled) {
+ //Close all active inputs
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
+ switch(activeDesc->mInputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ ALOGD("FOUND active input during call active: %d",activeDesc->mInputSource);
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if(prop_voip_enabled) {
+ ALOGD("CLOSING VoIP input source on call setup :%d ",activeDesc->mInputSource);
+ stopInput(activeInput);
+ releaseInput(activeInput);
+ }
+ break;
+
+ default:
+ ALOGD("CLOSING input on call setup for inputSource: %d",activeDesc->mInputSource);
+ stopInput(activeInput);
+ releaseInput(activeInput);
+ break;
+ }
+ }
+ } else if (prop_voip_enabled) {
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
+ if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeDesc->mInputSource) {
+ ALOGD("CLOSING VoIP on call setup : %d",activeDesc->mInputSource);
+ stopInput(activeInput);
+ releaseInput(activeInput);
+ }
+ }
+ }
+
+ //suspend PCM (deep-buffer) output & close compress & direct tracks
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ if (!outputDesc || !outputDesc->mProfile) {
+ ALOGD("ouput desc / profile is NULL");
+ continue;
+ }
+ if (((!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY))
+ && prop_playback_enabled) {
+ ALOGD(" calling suspendOutput on call mode for primary output");
+ mpClientInterface->suspendOutput(mOutputs.keyAt(i));
+ } //Close compress all sessions
+ else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
+ && prop_playback_enabled) {
+ ALOGD(" calling closeOutput on call mode for COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ else if ((outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX)
+ && prop_voip_enabled) {
+ ALOGD(" calling closeOutput on call mode for DIRECT output");
+ closeOutput(mOutputs.keyAt(i));
+ }
+ }
+ }
+
+ if ((AudioSystem::MODE_IN_CALL == oldState) && (AudioSystem::MODE_IN_CALL != state)
+ && prop_playback_enabled && mvoice_call_state) {
+ ALOGD("EXITING from call mode oldState :: %d state::%d \n",oldState, state);
+ mvoice_call_state = 0;
+ //restore PCM (deep-buffer) output after call termination
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ if (!outputDesc || !outputDesc->mProfile) {
+ ALOGD("ouput desc / profile is NULL");
+ continue;
+ }
+ if (!outputDesc->isDuplicated() &&outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ ALOGD("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 = AudioSystem::SYSTEM; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ ALOGD("Invalidate on call mode for stream :: %d ", i);
+ //FIXME see fixme on name change
+ mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
+ 0 /* ignored */);
+ }
+ }
+#endif
+ mPrevPhoneState = oldState;
int delayMs = 0;
if (isStateInCall(state)) {
@@ -479,5279 +1605,35 @@
// pertaining to sonification strategy see handleIncallSonification()
if (isStateInCall(state)) {
ALOGV("setPhoneState() in call state management: new state is %d", state);
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- handleIncallSonification((audio_stream_type_t)stream, true, true);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ handleIncallSonification(stream, true, true);
}
}
// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
- if (state == AUDIO_MODE_RINGTONE &&
- isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+ if (state == AudioSystem::MODE_RINGTONE &&
+ isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
mLimitRingtoneVolume = true;
} else {
mLimitRingtoneVolume = false;
}
+ ALOGD(" End of setPhoneState ... mPhoneState: %d ",mPhoneState);
}
-void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config)
+bool AudioPolicyManager::isStateInCall(int state)
{
- ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
-
- bool forceVolumeReeval = false;
- switch(usage) {
- case AUDIO_POLICY_FORCE_FOR_COMMUNICATION:
- if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO &&
- config != AUDIO_POLICY_FORCE_NONE) {
- ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
- return;
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_MEDIA:
- if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP &&
- config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
- config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
- config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
- config != AUDIO_POLICY_FORCE_NO_BT_A2DP) {
- ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
- return;
- }
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_RECORD:
- if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
- config != AUDIO_POLICY_FORCE_NONE) {
- ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
- return;
- }
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_DOCK:
- if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK &&
- config != AUDIO_POLICY_FORCE_BT_DESK_DOCK &&
- config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
- config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
- config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) {
- ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_SYSTEM:
- if (config != AUDIO_POLICY_FORCE_NONE &&
- config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
- ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- default:
- ALOGW("setForceUse() invalid usage %d", usage);
- break;
- }
-
- // check for device and output changes triggered by new force usage
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- updateDevicesAndOutputs();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_io_handle_t output = mOutputs.keyAt(i);
- audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/);
- setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
- if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
- applyStreamVolumes(output, newDevice, 0, true);
- }
- }
-
- audio_io_handle_t activeInput = getActiveInput();
- if (activeInput != 0) {
- setInputDevice(activeInput, getNewInputDevice(activeInput));
- }
-
+ return ((state == AudioSystem::MODE_IN_CALL) || (state == AudioSystem::MODE_IN_COMMUNICATION) ||
+ ((state == AudioSystem::MODE_RINGTONE) && (mPrevPhoneState == AudioSystem::MODE_IN_CALL)));
}
-audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
{
- return mForceUse[usage];
+ return new AudioPolicyManager(clientInterface);
}
-void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
{
- ALOGV("setSystemProperty() property %s, value %s", property, value);
-}
-
-// Find a direct output profile compatible with the parameters passed, even if the input flags do
-// not explicitly request a direct output
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getProfileForDirectOutput(
- audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags)
-{
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
- sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
- bool found = false;
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
- if (profile->isCompatibleProfile(device, samplingRate, format,
- channelMask,
- AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
- found = true;
- }
- } else {
- if (profile->isCompatibleProfile(device, samplingRate, format,
- channelMask,
- AUDIO_OUTPUT_FLAG_DIRECT)) {
- found = true;
- }
- }
- if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) {
- return profile;
- }
- }
- }
- return 0;
-}
-
-audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo)
-{
- audio_io_handle_t output = 0;
- uint32_t latency = 0;
- routing_strategy strategy = getStrategy(stream);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
- ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
- device, stream, samplingRate, format, channelMask, flags);
-
-#ifdef AUDIO_POLICY_TEST
- if (mCurOutput != 0) {
- ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
- mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
-
- if (mTestOutputs[mCurOutput] == 0) {
- ALOGV("getOutput() opening test output");
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
- outputDesc->mDevice = mTestDevice;
- outputDesc->mSamplingRate = mTestSamplingRate;
- outputDesc->mFormat = mTestFormat;
- outputDesc->mChannelMask = mTestChannels;
- outputDesc->mLatency = mTestLatencyMs;
- outputDesc->mFlags =
- (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
- outputDesc->mRefCount[stream] = 0;
- mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannelMask,
- &outputDesc->mLatency,
- outputDesc->mFlags,
- offloadInfo);
- if (mTestOutputs[mCurOutput]) {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"),mCurOutput);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
- addOutput(mTestOutputs[mCurOutput], outputDesc);
- }
- }
- return mTestOutputs[mCurOutput];
- }
-#endif //AUDIO_POLICY_TEST
-
- // 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);
- }
-
- // 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.
- sp<IOProfile> profile;
- if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
- !isNonOffloadableEffectEnabled()) {
- profile = getProfileForDirectOutput(device,
- samplingRate,
- format,
- channelMask,
- (audio_output_flags_t)flags);
- }
-
- if (profile != 0) {
- AudioOutputDescriptor *outputDesc = NULL;
-
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() && (profile == desc->mProfile)) {
- outputDesc = desc;
- // reuse direct output if currently open and configured with same parameters
- if ((samplingRate == outputDesc->mSamplingRate) &&
- (format == outputDesc->mFormat) &&
- (channelMask == outputDesc->mChannelMask)) {
- outputDesc->mDirectOpenCount++;
- ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
- return mOutputs.keyAt(i);
- }
- }
- }
- // close direct output if currently open and configured with different parameters
- if (outputDesc != NULL) {
- closeOutput(outputDesc->mIoHandle);
- }
- outputDesc = new AudioOutputDescriptor(profile);
- outputDesc->mDevice = device;
- outputDesc->mSamplingRate = samplingRate;
- outputDesc->mFormat = format;
- outputDesc->mChannelMask = channelMask;
- outputDesc->mLatency = 0;
- outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
- outputDesc->mRefCount[stream] = 0;
- outputDesc->mStopTime[stream] = 0;
- outputDesc->mDirectOpenCount = 1;
- output = mpClientInterface->openOutput(profile->mModule->mHandle,
- &outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannelMask,
- &outputDesc->mLatency,
- outputDesc->mFlags,
- offloadInfo);
-
- // only accept an output with the requested parameters
- if (output == 0 ||
- (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
- (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) ||
- (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
- ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
- "format %d %d, channelMask %04x %04x", output, samplingRate,
- outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
- outputDesc->mChannelMask);
- if (output != 0) {
- mpClientInterface->closeOutput(output);
- }
- delete outputDesc;
- return 0;
- }
- audio_io_handle_t srcOutput = getOutputForEffect();
- addOutput(output, outputDesc);
- audio_io_handle_t dstOutput = getOutputForEffect();
- if (dstOutput == output) {
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
- }
- mPreviousOutputs = mOutputs;
- ALOGV("getOutput() returns new direct output %d", output);
- mpClientInterface->onAudioPortListUpdate();
- return output;
- }
-
- // ignoring channel mask due to downmix capability in mixer
-
- // open a non direct output
-
- // for non direct outputs, only PCM is supported
- if (audio_is_linear_pcm(format)) {
- // get which output is suitable for the specified stream. The actual
- // routing change will happen when startOutput() will be called
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
-
- output = selectOutput(outputs, flags);
- }
- ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
- "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
-
- ALOGV("getOutput() returns output %d", output);
-
- return output;
-}
-
-audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
- audio_output_flags_t flags)
-{
- // select one output among several that provide a path to a particular device or set of
- // devices (the list was previously build by getOutputsForDevice()).
- // The priority is as follows:
- // 1: the output with the highest number of requested policy flags
- // 2: the primary output
- // 3: the first output in the list
-
- if (outputs.size() == 0) {
- return 0;
- }
- if (outputs.size() == 1) {
- return outputs[0];
- }
-
- int maxCommonFlags = 0;
- audio_io_handle_t outputFlags = 0;
- audio_io_handle_t outputPrimary = 0;
-
- for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
- if (!outputDesc->isDuplicated()) {
- int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
- if (commonFlags > maxCommonFlags) {
- outputFlags = outputs[i];
- maxCommonFlags = commonFlags;
- ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
- }
- if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
- outputPrimary = outputs[i];
- }
- }
- }
-
- if (outputFlags != 0) {
- return outputFlags;
- }
- if (outputPrimary != 0) {
- return outputPrimary;
- }
-
- return outputs[0];
-}
-
-status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- int session)
-{
- ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("startOutput() unknown output %d", output);
- return BAD_VALUE;
- }
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
- // increment usage count for this stream on the requested output:
- // NOTE that the usage count is the same for duplicated output and hardware output which is
- // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
- outputDesc->changeRefCount(stream, 1);
-
- if (outputDesc->mRefCount[stream] == 1) {
- audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
- routing_strategy strategy = getStrategy(stream);
- bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
- (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
- uint32_t waitMs = 0;
- bool force = false;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- if (desc != outputDesc) {
- // force a device change if any other output is managed by the same hw
- // module and has a current device selection that differs from selected device.
- // In this case, the audio HAL must receive the new device selection so that it can
- // change the device currently selected by the other active output.
- if (outputDesc->sharesHwModuleWith(desc) &&
- desc->device() != newDevice) {
- force = true;
- }
- // wait for audio on other active outputs to be presented when starting
- // a notification so that audio focus effect can propagate.
- uint32_t latency = desc->latency();
- if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
- waitMs = latency;
- }
- }
- }
- uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
-
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, true, false);
- }
-
- // apply volume rules for current stream and device if necessary
- checkAndSetVolume(stream,
- mStreams[stream].getVolumeIndex(newDevice),
- output,
- newDevice);
-
- // update the outputs if starting an output with a stream that can affect notification
- // routing
- handleNotificationRoutingForStream(stream);
- if (waitMs > muteWaitMs) {
- usleep((waitMs - muteWaitMs) * 2 * 1000);
- }
- }
- return NO_ERROR;
-}
-
-
-status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- int session)
-{
- ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("stopOutput() unknown output %d", output);
- return BAD_VALUE;
- }
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, false, false);
- }
-
- if (outputDesc->mRefCount[stream] > 0) {
- // decrement usage count of this stream on the output
- outputDesc->changeRefCount(stream, -1);
- // store time at which the stream was stopped - see isStreamActive()
- if (outputDesc->mRefCount[stream] == 0) {
- outputDesc->mStopTime[stream] = systemTime();
- audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
- // delay the device switch by twice the latency because stopOutput() is executed when
- // the track stop() command is received and at that time the audio track buffer can
- // still contain data that needs to be drained. The latency only covers the audio HAL
- // and kernel buffers. Also the latency does not always include additional delay in the
- // audio path (audio DSP, CODEC ...)
- setOutputDevice(output, newDevice, false, outputDesc->mLatency*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);
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- if (curOutput != output &&
- desc->isActive() &&
- outputDesc->sharesHwModuleWith(desc) &&
- (newDevice != desc->device())) {
- setOutputDevice(curOutput,
- getNewOutputDevice(curOutput, false /*fromCache*/),
- true,
- outputDesc->mLatency*2);
- }
- }
- // update the outputs if stopping one with a stream that can affect notification routing
- handleNotificationRoutingForStream(stream);
- }
- return NO_ERROR;
- } else {
- ALOGW("stopOutput() refcount is already 0 for output %d", output);
- return INVALID_OPERATION;
- }
-}
-
-void AudioPolicyManager::releaseOutput(audio_io_handle_t output)
-{
- ALOGV("releaseOutput() %d", output);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("releaseOutput() releasing unknown output %d", output);
- return;
- }
-
-#ifdef AUDIO_POLICY_TEST
- int testIndex = testOutputIndex(output);
- if (testIndex != 0) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
- if (outputDesc->isActive()) {
- mpClientInterface->closeOutput(output);
- delete mOutputs.valueAt(index);
- mOutputs.removeItem(output);
- mTestOutputs[testIndex] = 0;
- }
- return;
- }
-#endif //AUDIO_POLICY_TEST
-
- AudioOutputDescriptor *desc = mOutputs.valueAt(index);
- if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
- if (desc->mDirectOpenCount <= 0) {
- ALOGW("releaseOutput() invalid open count %d for output %d",
- desc->mDirectOpenCount, output);
- return;
- }
- if (--desc->mDirectOpenCount == 0) {
- closeOutput(output);
- // If effects where present on the output, audioflinger moved them to the primary
- // output by default: move them back to the appropriate output.
- audio_io_handle_t dstOutput = getOutputForEffect();
- if (dstOutput != mPrimaryOutput) {
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
- }
- mpClientInterface->onAudioPortListUpdate();
- }
- }
-}
-
-
-audio_io_handle_t AudioPolicyManager::getInput(audio_source_t inputSource,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_in_acoustics_t acoustics)
-{
- audio_io_handle_t input = 0;
- audio_devices_t device = getDeviceForInputSource(inputSource);
-
- ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
- inputSource, samplingRate, format, channelMask, acoustics);
-
- if (device == AUDIO_DEVICE_NONE) {
- ALOGW("getInput() could not find device for inputSource %d", inputSource);
- return 0;
- }
-
- // adapt channel selection to input source
- switch(inputSource) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
- break;
- case AUDIO_SOURCE_VOICE_CALL:
- channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
- break;
- default:
- break;
- }
-
- sp<IOProfile> profile = getInputProfile(device,
- samplingRate,
- format,
- channelMask);
- if (profile == 0) {
- ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d, "
- "channelMask %04x",
- device, samplingRate, format, channelMask);
- return 0;
- }
-
- if (profile->mModule->mHandle == 0) {
- ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
- return 0;
- }
-
- AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
-
- inputDesc->mInputSource = inputSource;
- inputDesc->mDevice = device;
- inputDesc->mSamplingRate = samplingRate;
- inputDesc->mFormat = format;
- inputDesc->mChannelMask = channelMask;
- inputDesc->mRefCount = 0;
- input = mpClientInterface->openInput(profile->mModule->mHandle,
- &inputDesc->mDevice,
- &inputDesc->mSamplingRate,
- &inputDesc->mFormat,
- &inputDesc->mChannelMask);
-
- // only accept input with the exact requested set of parameters
- if (input == 0 ||
- (samplingRate != inputDesc->mSamplingRate) ||
- (format != inputDesc->mFormat) ||
- (channelMask != inputDesc->mChannelMask)) {
- ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask %x",
- samplingRate, format, channelMask);
- if (input != 0) {
- mpClientInterface->closeInput(input);
- }
- delete inputDesc;
- return 0;
- }
- addInput(input, inputDesc);
- mpClientInterface->onAudioPortListUpdate();
- return input;
-}
-
-status_t AudioPolicyManager::startInput(audio_io_handle_t input)
-{
- ALOGV("startInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("startInput() unknown input %d", input);
- return BAD_VALUE;
- }
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-#ifdef AUDIO_POLICY_TEST
- if (mTestInput == 0)
-#endif //AUDIO_POLICY_TEST
- {
- // refuse 2 active AudioRecord clients at the same time except if the active input
- // uses AUDIO_SOURCE_HOTWORD in which case it is closed.
- audio_io_handle_t activeInput = getActiveInput();
- if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) {
- AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
- if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
- ALOGW("startInput() preempting already started low-priority input %d", activeInput);
- stopInput(activeInput);
- releaseInput(activeInput);
- } else {
- ALOGW("startInput() input %d failed: other input already started", input);
- return INVALID_OPERATION;
- }
- }
- }
-
- setInputDevice(input, getNewInputDevice(input), true /* force */);
-
- // automatically enable the remote submix output when input is started
- if (audio_is_remote_submix_device(inputDesc->mDevice)) {
- setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
- }
-
- ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
-
- inputDesc->mRefCount = 1;
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::stopInput(audio_io_handle_t input)
-{
- ALOGV("stopInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("stopInput() unknown input %d", input);
- return BAD_VALUE;
- }
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
- if (inputDesc->mRefCount == 0) {
- ALOGW("stopInput() input %d already stopped", input);
- return INVALID_OPERATION;
- } else {
- // automatically disable the remote submix output when input is stopped
- if (audio_is_remote_submix_device(inputDesc->mDevice)) {
- setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
- }
-
- resetInputDevice(input);
- inputDesc->mRefCount = 0;
- return NO_ERROR;
- }
-}
-
-void AudioPolicyManager::releaseInput(audio_io_handle_t input)
-{
- ALOGV("releaseInput() %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("releaseInput() releasing unknown input %d", input);
- return;
- }
- mpClientInterface->closeInput(input);
- delete mInputs.valueAt(index);
- mInputs.removeItem(input);
- nextAudioPortGeneration();
- mpClientInterface->onAudioPortListUpdate();
- ALOGV("releaseInput() exit");
-}
-
-void AudioPolicyManager::closeAllInputs() {
- for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- mpClientInterface->closeInput(mInputs.keyAt(input_index));
- }
- mInputs.clear();
- nextAudioPortGeneration();
-}
-
-void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
- int indexMin,
- int indexMax)
-{
- ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
- if (indexMin < 0 || indexMin >= indexMax) {
- ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
- return;
- }
- mStreams[stream].mIndexMin = indexMin;
- mStreams[stream].mIndexMax = indexMax;
-}
-
-status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
- int index,
- audio_devices_t device)
-{
-
- if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
- return BAD_VALUE;
- }
- if (!audio_is_output_device(device)) {
- return BAD_VALUE;
- }
-
- // Force max volume if stream cannot be muted
- if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
-
- ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
- stream, device, index);
-
- // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
- // clear all device specific values
- if (device == AUDIO_DEVICE_OUT_DEFAULT) {
- mStreams[stream].mIndexCur.clear();
- }
- mStreams[stream].mIndexCur.add(device, index);
-
- // compute and apply stream volume on all outputs according to connected device
- status_t status = NO_ERROR;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_devices_t curDevice =
- getDeviceForVolume(mOutputs.valueAt(i)->device());
- if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) {
- status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
- if (volStatus != NO_ERROR) {
- status = volStatus;
- }
- }
- }
- return status;
-}
-
-status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
- int *index,
- audio_devices_t device)
-{
- if (index == NULL) {
- return BAD_VALUE;
- }
- if (!audio_is_output_device(device)) {
- return BAD_VALUE;
- }
- // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
- // the strategy the stream belongs to.
- if (device == AUDIO_DEVICE_OUT_DEFAULT) {
- device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
- }
- device = getDeviceForVolume(device);
-
- *index = mStreams[stream].getVolumeIndex(device);
- ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
- return NO_ERROR;
-}
-
-audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
- const SortedVector<audio_io_handle_t>& outputs)
-{
- // select one output among several suitable for global effects.
- // The priority is as follows:
- // 1: An offloaded output. If the effect ends up not being offloadable,
- // AudioFlinger will invalidate the track and the offloaded output
- // will be closed causing the effect to be moved to a PCM output.
- // 2: A deep buffer output
- // 3: the first output in the list
-
- if (outputs.size() == 0) {
- return 0;
- }
-
- audio_io_handle_t outputOffloaded = 0;
- audio_io_handle_t outputDeepBuffer = 0;
-
- for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
- ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
- if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- outputOffloaded = outputs[i];
- }
- if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
- outputDeepBuffer = outputs[i];
- }
- }
-
- ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
- outputOffloaded, outputDeepBuffer);
- if (outputOffloaded != 0) {
- return outputOffloaded;
- }
- if (outputDeepBuffer != 0) {
- return outputDeepBuffer;
- }
-
- return outputs[0];
-}
-
-audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
-{
- // apply simple rule where global effects are attached to the same output as MUSIC streams
-
- routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
-
- audio_io_handle_t output = selectOutputForEffects(dstOutputs);
- ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
- output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
-
- return output;
-}
-
-status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
- audio_io_handle_t io,
- uint32_t strategy,
- int session,
- int id)
-{
- ssize_t index = mOutputs.indexOfKey(io);
- if (index < 0) {
- index = mInputs.indexOfKey(io);
- if (index < 0) {
- ALOGW("registerEffect() unknown io %d", io);
- return INVALID_OPERATION;
- }
- }
-
- if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
- ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
- desc->name, desc->memoryUsage);
- return INVALID_OPERATION;
- }
- mTotalEffectsMemory += desc->memoryUsage;
- ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
- desc->name, io, strategy, session, id);
- ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
-
- EffectDescriptor *pDesc = new EffectDescriptor();
- memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
- pDesc->mIo = io;
- pDesc->mStrategy = (routing_strategy)strategy;
- pDesc->mSession = session;
- pDesc->mEnabled = false;
-
- mEffects.add(id, pDesc);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::unregisterEffect(int id)
-{
- ssize_t index = mEffects.indexOfKey(id);
- if (index < 0) {
- ALOGW("unregisterEffect() unknown effect ID %d", id);
- return INVALID_OPERATION;
- }
-
- EffectDescriptor *pDesc = mEffects.valueAt(index);
-
- setEffectEnabled(pDesc, false);
-
- if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
- ALOGW("unregisterEffect() memory %d too big for total %d",
- pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
- pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
- }
- mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
- ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
- pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
-
- mEffects.removeItem(id);
- delete pDesc;
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled)
-{
- ssize_t index = mEffects.indexOfKey(id);
- if (index < 0) {
- ALOGW("unregisterEffect() unknown effect ID %d", id);
- return INVALID_OPERATION;
- }
-
- return setEffectEnabled(mEffects.valueAt(index), enabled);
-}
-
-status_t AudioPolicyManager::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
-{
- if (enabled == pDesc->mEnabled) {
- ALOGV("setEffectEnabled(%s) effect already %s",
- enabled?"true":"false", enabled?"enabled":"disabled");
- return INVALID_OPERATION;
- }
-
- if (enabled) {
- if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
- ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
- pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
- return INVALID_OPERATION;
- }
- mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
- ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
- } else {
- if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
- ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
- pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
- pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
- }
- mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
- ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
- }
- pDesc->mEnabled = enabled;
- return NO_ERROR;
-}
-
-bool AudioPolicyManager::isNonOffloadableEffectEnabled()
-{
- for (size_t i = 0; i < mEffects.size(); i++) {
- const EffectDescriptor * const pDesc = mEffects.valueAt(i);
- if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) &&
- ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
- ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
- pDesc->mDesc.name, pDesc->mSession);
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
-{
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
- if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream,
- uint32_t inPastMs) const
-{
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
- if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
- outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::isSourceActive(audio_source_t source) const
-{
- for (size_t i = 0; i < mInputs.size(); i++) {
- const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i);
- if ((inputDescriptor->mInputSource == (int)source ||
- (source == AUDIO_SOURCE_VOICE_RECOGNITION &&
- inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD))
- && (inputDescriptor->mRefCount > 0)) {
- return true;
- }
- }
- return false;
-}
-
-
-status_t AudioPolicyManager::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
- result.append(buffer);
-
- snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
- result.append(buffer);
- snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for communications %d\n",
- mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]);
- result.append(buffer);
-
- snprintf(buffer, SIZE, " Available output devices:\n");
- result.append(buffer);
- write(fd, result.string(), result.size());
- for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
- mAvailableOutputDevices[i]->dump(fd, 2, i);
- }
- snprintf(buffer, SIZE, "\n Available input devices:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
- mAvailableInputDevices[i]->dump(fd, 2, i);
- }
-
- snprintf(buffer, SIZE, "\nHW Modules dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mHwModules.size(); i++) {
- snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1);
- write(fd, buffer, strlen(buffer));
- mHwModules[i]->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nOutputs dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mOutputs.size(); i++) {
- snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mOutputs.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nInputs dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mInputs.size(); i++) {
- snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mInputs.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nStreams dump:\n");
- write(fd, buffer, strlen(buffer));
- snprintf(buffer, SIZE,
- " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
- write(fd, buffer, strlen(buffer));
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- snprintf(buffer, SIZE, " %02zu ", i);
- write(fd, buffer, strlen(buffer));
- mStreams[i].dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
- (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
- write(fd, buffer, strlen(buffer));
-
- snprintf(buffer, SIZE, "Registered effects:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mEffects.size(); i++) {
- snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mEffects.valueAt(i)->dump(fd);
- }
-
-
- return NO_ERROR;
-}
-
-// 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 AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
-{
- ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
- " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
- offloadInfo.sample_rate, offloadInfo.channel_mask,
- offloadInfo.format,
- offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
- offloadInfo.has_video);
-
- // Check if offload has been disabled
- char propValue[PROPERTY_VALUE_MAX];
- if (property_get("audio.offload.disable", propValue, "0")) {
- if (atoi(propValue) != 0) {
- ALOGV("offload disabled by audio.offload.disable=%s", propValue );
- return false;
- }
- }
-
- // Check if stream type is music, then only allow offload as of now.
- if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
- {
- ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
- return false;
- }
-
- //TODO: enable audio offloading with video when ready
- if (offloadInfo.has_video)
- {
- ALOGV("isOffloadSupported: has_video == true, returning false");
- return false;
- }
-
- //If duration is less than minimum value defined in property, return false
- if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
- if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
- ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
- return false;
- }
- } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
- ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
- 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 (isNonOffloadableEffectEnabled()) {
- return false;
- }
-
- // See if there is a profile to support this.
- // AUDIO_DEVICE_NONE
- sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
- offloadInfo.sample_rate,
- offloadInfo.format,
- offloadInfo.channel_mask,
- AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
- ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
- return (profile != 0);
-}
-
-status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role,
- audio_port_type_t type,
- unsigned int *num_ports,
- struct audio_port *ports,
- unsigned int *generation)
-{
- if (num_ports == NULL || (*num_ports != 0 && ports == NULL) ||
- generation == NULL) {
- return BAD_VALUE;
- }
- ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports);
- if (ports == NULL) {
- *num_ports = 0;
- }
-
- size_t portsWritten = 0;
- size_t portsMax = *num_ports;
- *num_ports = 0;
- if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) {
- if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
- for (size_t i = 0;
- i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) {
- mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]);
- }
- *num_ports += mAvailableOutputDevices.size();
- }
- if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
- for (size_t i = 0;
- i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) {
- mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]);
- }
- *num_ports += mAvailableInputDevices.size();
- }
- }
- if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) {
- if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
- for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) {
- mInputs[i]->toAudioPort(&ports[portsWritten++]);
- }
- *num_ports += mInputs.size();
- }
- if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
- for (size_t i = 0; i < mOutputs.size() && portsWritten < portsMax; i++) {
- mOutputs[i]->toAudioPort(&ports[portsWritten++]);
- }
- *num_ports += mOutputs.size();
- }
- }
- *generation = curAudioPortGeneration();
- ALOGV("listAudioPorts() got %d ports needed %d", portsWritten, *num_ports);
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused)
-{
- return NO_ERROR;
-}
-
-AudioPolicyManager::AudioOutputDescriptor *AudioPolicyManager::getOutputFromId(
- audio_port_handle_t id) const
-{
- AudioOutputDescriptor *outputDesc = NULL;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- outputDesc = mOutputs.valueAt(i);
- if (outputDesc->mId == id) {
- break;
- }
- }
- return outputDesc;
-}
-
-AudioPolicyManager::AudioInputDescriptor *AudioPolicyManager::getInputFromId(
- audio_port_handle_t id) const
-{
- AudioInputDescriptor *inputDesc = NULL;
- for (size_t i = 0; i < mInputs.size(); i++) {
- inputDesc = mInputs.valueAt(i);
- if (inputDesc->mId == id) {
- break;
- }
- }
- return inputDesc;
-}
-
-AudioPolicyManager::HwModule *AudioPolicyManager::getModuleForDevice(audio_devices_t device) const
-{
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- if (audio_is_output_device(device)) {
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) {
- return mHwModules[i];
- }
- }
- } else {
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
- if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() &
- device & ~AUDIO_DEVICE_BIT_IN) {
- return mHwModules[i];
- }
- }
- }
- }
- return NULL;
-}
-
-AudioPolicyManager::HwModule *AudioPolicyManager::getModuleFromName(const char *name) const
-{
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (strcmp(mHwModules[i]->mName, name) == 0) {
- return mHwModules[i];
- }
- }
- return NULL;
-}
-
-
-status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
- audio_patch_handle_t *handle,
- uid_t uid)
-{
- ALOGV("createAudioPatch()");
-
- if (handle == NULL || patch == NULL) {
- return BAD_VALUE;
- }
- ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
-
- if (patch->num_sources > 1 || patch->num_sinks > 1) {
- return INVALID_OPERATION;
- }
- if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE ||
- patch->sinks[0].role != AUDIO_PORT_ROLE_SINK) {
- return INVALID_OPERATION;
- }
-
- sp<AudioPatch> patchDesc;
- ssize_t index = mAudioPatches.indexOfKey(*handle);
-
- ALOGV("createAudioPatch sink id %d role %d type %d", patch->sinks[0].id, patch->sinks[0].role,
- patch->sinks[0].type);
- ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
- patch->sources[0].role,
- patch->sources[0].type);
-
- if (index >= 0) {
- patchDesc = mAudioPatches.valueAt(index);
- ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
- mUidCached, patchDesc->mUid, uid);
- if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
- return INVALID_OPERATION;
- }
- } else {
- *handle = 0;
- }
-
- if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
- // TODO add support for mix to mix connection
- if (patch->sinks[0].type != AUDIO_PORT_TYPE_DEVICE) {
- ALOGV("createAudioPatch() source mix sink not device");
- return BAD_VALUE;
- }
- // output mix to output device connection
- AudioOutputDescriptor *outputDesc = getOutputFromId(patch->sources[0].id);
- if (outputDesc == NULL) {
- ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
- return BAD_VALUE;
- }
- if (patchDesc != 0) {
- if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
- ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
- patchDesc->mPatch.sources[0].id, patch->sources[0].id);
- return BAD_VALUE;
- }
- }
- sp<DeviceDescriptor> devDesc =
- mAvailableOutputDevices.getDeviceFromId(patch->sinks[0].id);
- if (devDesc == 0) {
- ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[0].id);
- return BAD_VALUE;
- }
-
- if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mType,
- patch->sources[0].sample_rate,
- patch->sources[0].format,
- patch->sources[0].channel_mask,
- AUDIO_OUTPUT_FLAG_NONE)) {
- return INVALID_OPERATION;
- }
- // TODO: reconfigure output format and channels here
- ALOGV("createAudioPatch() setting device %08x on output %d",
- devDesc->mType, outputDesc->mIoHandle);
- setOutputDevice(outputDesc->mIoHandle,
- devDesc->mType,
- true,
- 0,
- handle);
- index = mAudioPatches.indexOfKey(*handle);
- if (index >= 0) {
- if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
- ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
- }
- patchDesc = mAudioPatches.valueAt(index);
- patchDesc->mUid = uid;
- ALOGV("createAudioPatch() success");
- } else {
- ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
- return INVALID_OPERATION;
- }
- } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
- if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
- // input device to input mix connection
- AudioInputDescriptor *inputDesc = getInputFromId(patch->sinks[0].id);
- if (inputDesc == NULL) {
- return BAD_VALUE;
- }
- if (patchDesc != 0) {
- if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) {
- return BAD_VALUE;
- }
- }
- sp<DeviceDescriptor> devDesc =
- mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
- if (devDesc == 0) {
- return BAD_VALUE;
- }
-
- if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mType,
- patch->sinks[0].sample_rate,
- patch->sinks[0].format,
- patch->sinks[0].channel_mask,
- AUDIO_OUTPUT_FLAG_NONE)) {
- return INVALID_OPERATION;
- }
- // TODO: reconfigure output format and channels here
- ALOGV("createAudioPatch() setting device %08x on output %d",
- devDesc->mType, inputDesc->mIoHandle);
- setInputDevice(inputDesc->mIoHandle,
- devDesc->mType,
- true,
- handle);
- index = mAudioPatches.indexOfKey(*handle);
- if (index >= 0) {
- if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
- ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
- }
- patchDesc = mAudioPatches.valueAt(index);
- patchDesc->mUid = uid;
- ALOGV("createAudioPatch() success");
- } else {
- ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
- return INVALID_OPERATION;
- }
- } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
- // device to device connection
- if (patchDesc != 0) {
- if (patchDesc->mPatch.sources[0].id != patch->sources[0].id &&
- patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) {
- return BAD_VALUE;
- }
- }
-
- sp<DeviceDescriptor> srcDeviceDesc =
- mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
- sp<DeviceDescriptor> sinkDeviceDesc =
- mAvailableOutputDevices.getDeviceFromId(patch->sinks[0].id);
- if (srcDeviceDesc == 0 || sinkDeviceDesc == 0) {
- return BAD_VALUE;
- }
- //update source and sink with our own data as the data passed in the patch may
- // be incomplete.
- struct audio_patch newPatch = *patch;
- srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
- sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[0], &patch->sinks[0]);
-
- // TODO: add support for devices on different HW modules
- if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) {
- return INVALID_OPERATION;
- }
- // TODO: check from routing capabilities in config file and other conflicting patches
-
- audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- if (index >= 0) {
- afPatchHandle = patchDesc->mAfPatchHandle;
- }
-
- status_t status = mpClientInterface->createAudioPatch(&newPatch,
- &afPatchHandle,
- 0);
- ALOGV("createAudioPatch() patch panel returned %d patchHandle %d",
- status, afPatchHandle);
- if (status == NO_ERROR) {
- if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &newPatch, uid);
- addAudioPatch(patchDesc->mHandle, patchDesc);
- } else {
- patchDesc->mPatch = newPatch;
- }
- patchDesc->mAfPatchHandle = afPatchHandle;
- *handle = patchDesc->mHandle;
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- } else {
- ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
- status);
- return INVALID_OPERATION;
- }
- } else {
- return BAD_VALUE;
- }
- } else {
- return BAD_VALUE;
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle,
- uid_t uid)
-{
- ALOGV("releaseAudioPatch() patch %d", handle);
-
- ssize_t index = mAudioPatches.indexOfKey(handle);
-
- if (index < 0) {
- return BAD_VALUE;
- }
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
- mUidCached, patchDesc->mUid, uid);
- if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
- return INVALID_OPERATION;
- }
-
- struct audio_patch *patch = &patchDesc->mPatch;
- patchDesc->mUid = mUidCached;
- if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
- AudioOutputDescriptor *outputDesc = getOutputFromId(patch->sources[0].id);
- if (outputDesc == NULL) {
- ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
- return BAD_VALUE;
- }
-
- setOutputDevice(outputDesc->mIoHandle,
- getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/),
- true,
- 0,
- NULL);
- } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
- if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
- AudioInputDescriptor *inputDesc = getInputFromId(patch->sinks[0].id);
- if (inputDesc == NULL) {
- ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
- return BAD_VALUE;
- }
- setInputDevice(inputDesc->mIoHandle,
- getNewInputDevice(inputDesc->mIoHandle),
- true,
- NULL);
- } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
- audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle;
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d",
- status, patchDesc->mAfPatchHandle);
- removeAudioPatch(patchDesc->mHandle);
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- } else {
- return BAD_VALUE;
- }
- } else {
- return BAD_VALUE;
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches,
- struct audio_patch *patches,
- unsigned int *generation)
-{
- if (num_patches == NULL || (*num_patches != 0 && patches == NULL) ||
- generation == NULL) {
- return BAD_VALUE;
- }
- ALOGV("listAudioPatches() num_patches %d patches %p available patches %d",
- *num_patches, patches, mAudioPatches.size());
- if (patches == NULL) {
- *num_patches = 0;
- }
-
- size_t patchesWritten = 0;
- size_t patchesMax = *num_patches;
- for (size_t i = 0;
- i < mAudioPatches.size() && patchesWritten < patchesMax; i++) {
- patches[patchesWritten] = mAudioPatches[i]->mPatch;
- patches[patchesWritten++].id = mAudioPatches[i]->mHandle;
- ALOGV("listAudioPatches() patch %d num_sources %d num_sinks %d",
- i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks);
- }
- *num_patches = mAudioPatches.size();
-
- *generation = curAudioPortGeneration();
- ALOGV("listAudioPatches() got %d patches needed %d", patchesWritten, *num_patches);
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config)
-{
- ALOGV("setAudioPortConfig()");
-
- if (config == NULL) {
- return BAD_VALUE;
- }
- ALOGV("setAudioPortConfig() on port handle %d", config->id);
- // Only support gain configuration for now
- if (config->config_mask != AUDIO_PORT_CONFIG_GAIN || config->gain.index < 0) {
- return BAD_VALUE;
- }
-
- sp<AudioPort> portDesc;
- struct audio_port_config portConfig;
- if (config->type == AUDIO_PORT_TYPE_MIX) {
- if (config->role == AUDIO_PORT_ROLE_SOURCE) {
- AudioOutputDescriptor *outputDesc = getOutputFromId(config->id);
- if (outputDesc == NULL) {
- return BAD_VALUE;
- }
- portDesc = outputDesc->mProfile;
- outputDesc->toAudioPortConfig(&portConfig);
- } else if (config->role == AUDIO_PORT_ROLE_SINK) {
- AudioInputDescriptor *inputDesc = getInputFromId(config->id);
- if (inputDesc == NULL) {
- return BAD_VALUE;
- }
- portDesc = inputDesc->mProfile;
- inputDesc->toAudioPortConfig(&portConfig);
- } else {
- return BAD_VALUE;
- }
- } else if (config->type == AUDIO_PORT_TYPE_DEVICE) {
- sp<DeviceDescriptor> deviceDesc;
- if (config->role == AUDIO_PORT_ROLE_SOURCE) {
- deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id);
- } else if (config->role == AUDIO_PORT_ROLE_SINK) {
- deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id);
- } else {
- return BAD_VALUE;
- }
- if (deviceDesc == NULL) {
- return BAD_VALUE;
- }
- portDesc = deviceDesc;
- deviceDesc->toAudioPortConfig(&portConfig);
- } else {
- return BAD_VALUE;
- }
-
- if ((size_t)config->gain.index >= portDesc->mGains.size()) {
- return INVALID_OPERATION;
- }
- const struct audio_gain *gain = &portDesc->mGains[config->gain.index]->mGain;
- if ((config->gain.mode & ~gain->mode) != 0) {
- return BAD_VALUE;
- }
- if ((config->gain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- if ((config->gain.values[0] < gain->min_value) ||
- (config->gain.values[0] > gain->max_value)) {
- return BAD_VALUE;
- }
- } else {
- if ((config->gain.channel_mask & ~gain->channel_mask) != 0) {
- return BAD_VALUE;
- }
- size_t numValues = popcount(config->gain.channel_mask);
- for (size_t i = 0; i < numValues; i++) {
- if ((config->gain.values[i] < gain->min_value) ||
- (config->gain.values[i] > gain->max_value)) {
- return BAD_VALUE;
- }
- }
- }
- if ((config->gain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- if ((config->gain.ramp_duration_ms < gain->min_ramp_ms) ||
- (config->gain.ramp_duration_ms > gain->max_ramp_ms)) {
- return BAD_VALUE;
- }
- }
-
- portConfig.gain = config->gain;
-
- status_t status = mpClientInterface->setAudioPortConfig(&portConfig, 0);
-
- return status;
-}
-
-void AudioPolicyManager::clearAudioPatches(uid_t uid)
-{
- for (ssize_t i = 0; i < (ssize_t)mAudioPatches.size(); i++) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
- if (patchDesc->mUid == uid) {
- // releaseAudioPatch() removes the patch from mAudioPatches
- if (releaseAudioPatch(mAudioPatches.keyAt(i), uid) == NO_ERROR) {
- i--;
- }
- }
- }
-}
-
-status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle,
- const sp<AudioPatch>& patch)
-{
- ssize_t index = mAudioPatches.indexOfKey(handle);
-
- if (index >= 0) {
- ALOGW("addAudioPatch() patch %d already in", handle);
- return ALREADY_EXISTS;
- }
- mAudioPatches.add(handle, patch);
- ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d"
- "sink handle %d",
- handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks,
- patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id);
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle)
-{
- ssize_t index = mAudioPatches.indexOfKey(handle);
-
- if (index < 0) {
- ALOGW("removeAudioPatch() patch %d not in", handle);
- return ALREADY_EXISTS;
- }
- ALOGV("removeAudioPatch() handle %d af handle %d", handle,
- mAudioPatches.valueAt(index)->mAfPatchHandle);
- mAudioPatches.removeItemsAt(index);
- return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManager
-// ----------------------------------------------------------------------------
-
-uint32_t AudioPolicyManager::nextUniqueId()
-{
- return android_atomic_inc(&mNextUniqueId);
-}
-
-uint32_t AudioPolicyManager::nextAudioPortGeneration()
-{
- return android_atomic_inc(&mAudioPortGeneration);
-}
-
-AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
- :
-#ifdef AUDIO_POLICY_TEST
- Thread(false),
-#endif //AUDIO_POLICY_TEST
- mPrimaryOutput((audio_io_handle_t)0),
- mPhoneState(AUDIO_MODE_NORMAL),
- mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
- mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
- mA2dpSuspended(false),
- mSpeakerDrcEnabled(false), mNextUniqueId(1),
- mAudioPortGeneration(1)
-{
- mUidCached = getuid();
- mpClientInterface = clientInterface;
-
- for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
- mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
- }
-
- mDefaultOutputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_OUT_SPEAKER);
- if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
- if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
- ALOGE("could not load audio policy configuration file, setting defaults");
- defaultAudioPolicyConfig();
- }
- }
- // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
-
- // must be done after reading the policy
- initializeVolumeCurves();
-
- // open all output streams needed to access attached devices
- audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
- audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
- for (size_t i = 0; i < mHwModules.size(); i++) {
- mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
- if (mHwModules[i]->mHandle == 0) {
- ALOGW("could not open HW module %s", mHwModules[i]->mName);
- continue;
- }
- // open all output streams needed to access attached devices
- // except for direct output streams that are only opened when they are actually
- // required by an app.
- // This also validates mAvailableOutputDevices list
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
-
- if (outProfile->mSupportedDevices.isEmpty()) {
- ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName);
- continue;
- }
-
- audio_devices_t profileTypes = outProfile->mSupportedDevices.types();
- if ((profileTypes & outputDeviceTypes) &&
- ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
-
- outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice->mDeviceType & profileTypes);
- audio_io_handle_t output = mpClientInterface->openOutput(
- outProfile->mModule->mHandle,
- &outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannelMask,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (output == 0) {
- ALOGW("Cannot open output stream for device %08x on hw module %s",
- outputDesc->mDevice,
- mHwModules[i]->mName);
- delete outputDesc;
- } else {
- for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
- audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType;
- ssize_t index =
- mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
- // give a valid ID to an attached device once confirmed it is reachable
- if ((index >= 0) && (mAvailableOutputDevices[index]->mId == 0)) {
- mAvailableOutputDevices[index]->mId = nextUniqueId();
- mAvailableOutputDevices[index]->mModule = mHwModules[i];
- }
- }
- if (mPrimaryOutput == 0 &&
- outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
- mPrimaryOutput = output;
- }
- addOutput(output, outputDesc);
- ALOGI("CSTOR setOutputDevice %08x", outputDesc->mDevice);
- setOutputDevice(output,
- outputDesc->mDevice,
- true);
- }
- }
- }
- // open input streams needed to access attached devices to validate
- // mAvailableInputDevices list
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
- {
- const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
-
- if (inProfile->mSupportedDevices.isEmpty()) {
- ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName);
- continue;
- }
-
- audio_devices_t profileTypes = inProfile->mSupportedDevices.types();
- if (profileTypes & inputDeviceTypes) {
- AudioInputDescriptor *inputDesc = new AudioInputDescriptor(inProfile);
-
- inputDesc->mInputSource = AUDIO_SOURCE_MIC;
- inputDesc->mDevice = inProfile->mSupportedDevices[0]->mDeviceType;
- audio_io_handle_t input = mpClientInterface->openInput(
- inProfile->mModule->mHandle,
- &inputDesc->mDevice,
- &inputDesc->mSamplingRate,
- &inputDesc->mFormat,
- &inputDesc->mChannelMask);
-
- if (input != 0) {
- for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
- audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType;
- ssize_t index =
- mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
- // give a valid ID to an attached device once confirmed it is reachable
- if ((index >= 0) && (mAvailableInputDevices[index]->mId == 0)) {
- mAvailableInputDevices[index]->mId = nextUniqueId();
- mAvailableInputDevices[index]->mModule = mHwModules[i];
- }
- }
- mpClientInterface->closeInput(input);
- } else {
- ALOGW("Cannot open input stream for device %08x on hw module %s",
- inputDesc->mDevice,
- mHwModules[i]->mName);
- }
- delete inputDesc;
- }
- }
- }
- // make sure all attached devices have been allocated a unique ID
- for (size_t i = 0; i < mAvailableOutputDevices.size();) {
- if (mAvailableOutputDevices[i]->mId == 0) {
- ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
- mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
- continue;
- }
- i++;
- }
- for (size_t i = 0; i < mAvailableInputDevices.size();) {
- if (mAvailableInputDevices[i]->mId == 0) {
- ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
- mAvailableInputDevices.remove(mAvailableInputDevices[i]);
- continue;
- }
- i++;
- }
- // make sure default device is reachable
- if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
- ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType);
- }
-
- ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
-
- updateDevicesAndOutputs();
-
-#ifdef AUDIO_POLICY_TEST
- if (mPrimaryOutput != 0) {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
-
- mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
- mTestSamplingRate = 44100;
- mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
- mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
- mTestLatencyMs = 0;
- mCurOutput = 0;
- mDirectOutput = false;
- for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
- mTestOutputs[i] = 0;
- }
-
- const size_t SIZE = 256;
- char buffer[SIZE];
- snprintf(buffer, SIZE, "AudioPolicyManagerTest");
- run(buffer, ANDROID_PRIORITY_AUDIO);
- }
-#endif //AUDIO_POLICY_TEST
-}
-
-AudioPolicyManager::~AudioPolicyManager()
-{
-#ifdef AUDIO_POLICY_TEST
- exit();
-#endif //AUDIO_POLICY_TEST
- for (size_t i = 0; i < mOutputs.size(); i++) {
- mpClientInterface->closeOutput(mOutputs.keyAt(i));
- delete mOutputs.valueAt(i);
- }
- for (size_t i = 0; i < mInputs.size(); i++) {
- mpClientInterface->closeInput(mInputs.keyAt(i));
- delete mInputs.valueAt(i);
- }
- for (size_t i = 0; i < mHwModules.size(); i++) {
- delete mHwModules[i];
- }
- mAvailableOutputDevices.clear();
- mAvailableInputDevices.clear();
-}
-
-status_t AudioPolicyManager::initCheck()
-{
- return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
-}
-
-#ifdef AUDIO_POLICY_TEST
-bool AudioPolicyManager::threadLoop()
-{
- ALOGV("entering threadLoop()");
- while (!exitPending())
- {
- String8 command;
- int valueInt;
- String8 value;
-
- Mutex::Autolock _l(mLock);
- mWaitWorkCV.waitRelative(mLock, milliseconds(50));
-
- command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
- AudioParameter param = AudioParameter(command);
-
- if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
- valueInt != 0) {
- ALOGV("Test command %s received", command.string());
- String8 target;
- if (param.get(String8("target"), target) != NO_ERROR) {
- target = "Manager";
- }
- if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_output"));
- mCurOutput = valueInt;
- }
- if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_direct"));
- if (value == "false") {
- mDirectOutput = false;
- } else if (value == "true") {
- mDirectOutput = true;
- }
- }
- if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_input"));
- mTestInput = valueInt;
- }
-
- if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_format"));
- int format = AUDIO_FORMAT_INVALID;
- if (value == "PCM 16 bits") {
- format = AUDIO_FORMAT_PCM_16_BIT;
- } else if (value == "PCM 8 bits") {
- format = AUDIO_FORMAT_PCM_8_BIT;
- } else if (value == "Compressed MP3") {
- format = AUDIO_FORMAT_MP3;
- }
- if (format != AUDIO_FORMAT_INVALID) {
- if (target == "Manager") {
- mTestFormat = format;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("format"), format);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
- if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_channels"));
- int channels = 0;
-
- if (value == "Channels Stereo") {
- channels = AUDIO_CHANNEL_OUT_STEREO;
- } else if (value == "Channels Mono") {
- channels = AUDIO_CHANNEL_OUT_MONO;
- }
- if (channels != 0) {
- if (target == "Manager") {
- mTestChannels = channels;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("channels"), channels);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
- if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_sampleRate"));
- if (valueInt >= 0 && valueInt <= 96000) {
- int samplingRate = valueInt;
- if (target == "Manager") {
- mTestSamplingRate = samplingRate;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("sampling_rate"), samplingRate);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
-
- if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_reopen"));
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
- mpClientInterface->closeOutput(mPrimaryOutput);
-
- audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
-
- delete mOutputs.valueFor(mPrimaryOutput);
- mOutputs.removeItem(mPrimaryOutput);
-
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
- outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
- mPrimaryOutput = mpClientInterface->openOutput(moduleHandle,
- &outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannelMask,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (mPrimaryOutput == 0) {
- ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
- outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
- } else {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
- addOutput(mPrimaryOutput, outputDesc);
- }
- }
-
-
- mpClientInterface->setParameters(0, String8("test_cmd_policy="));
- }
- }
- return false;
-}
-
-void AudioPolicyManager::exit()
-{
- {
- AutoMutex _l(mLock);
- requestExit();
- mWaitWorkCV.signal();
- }
- requestExitAndWait();
-}
-
-int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
-{
- for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
- if (output == mTestOutputs[i]) return i;
- }
- return 0;
-}
-#endif //AUDIO_POLICY_TEST
-
-// ---
-
-void AudioPolicyManager::addOutput(audio_io_handle_t output, AudioOutputDescriptor *outputDesc)
-{
- outputDesc->mIoHandle = output;
- outputDesc->mId = nextUniqueId();
- mOutputs.add(output, outputDesc);
- nextAudioPortGeneration();
-}
-
-void AudioPolicyManager::addInput(audio_io_handle_t input, AudioInputDescriptor *inputDesc)
-{
- inputDesc->mIoHandle = input;
- inputDesc->mId = nextUniqueId();
- mInputs.add(input, inputDesc);
- nextAudioPortGeneration();
-}
-
-String8 AudioPolicyManager::addressToParameter(audio_devices_t device, const String8 address)
-{
- if (device & AUDIO_DEVICE_OUT_ALL_A2DP) {
- return String8("a2dp_sink_address=")+address;
- }
- return address;
-}
-
-status_t AudioPolicyManager::checkOutputsForDevice(audio_devices_t device,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& outputs,
- const String8 address)
-{
- AudioOutputDescriptor *desc;
-
- if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
- // first list already open outputs that can be routed to this device
- for (size_t i = 0; i < mOutputs.size(); i++) {
- desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) {
- ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
- outputs.add(mOutputs.keyAt(i));
- }
- }
- // then look for output profiles that can be routed to this device
- SortedVector< sp<IOProfile> > profiles;
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) {
- ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
- profiles.add(mHwModules[i]->mOutputProfiles[j]);
- }
- }
- }
-
- if (profiles.isEmpty() && outputs.isEmpty()) {
- ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
- return BAD_VALUE;
- }
-
- // open outputs for matching profiles if needed. Direct outputs are also opened to
- // query for dynamic parameters and will be closed later by setDeviceConnectionState()
- for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
- sp<IOProfile> profile = profiles[profile_index];
-
- // nothing to do if one output is already opened for this profile
- size_t j;
- for (j = 0; j < mOutputs.size(); j++) {
- desc = mOutputs.valueAt(j);
- if (!desc->isDuplicated() && desc->mProfile == profile) {
- break;
- }
- }
- if (j != mOutputs.size()) {
- continue;
- }
-
- ALOGV("opening output for device %08x with params %s", device, address.string());
- desc = new AudioOutputDescriptor(profile);
- desc->mDevice = device;
- audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
- offloadInfo.sample_rate = desc->mSamplingRate;
- offloadInfo.format = desc->mFormat;
- offloadInfo.channel_mask = desc->mChannelMask;
-
- audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle,
- &desc->mDevice,
- &desc->mSamplingRate,
- &desc->mFormat,
- &desc->mChannelMask,
- &desc->mLatency,
- desc->mFlags,
- &offloadInfo);
- if (output != 0) {
- // Here is where the out_set_parameters() for card & device gets called
- if (!address.isEmpty()) {
- mpClientInterface->setParameters(output, addressToParameter(device, address));
- }
-
- // Here is where we step through and resolve any "dynamic" fields
- String8 reply;
- char *value;
- if (profile->mSamplingRates[0] == 0) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
- ALOGV("checkOutputsForDevice() direct output sup sampling rates %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadSamplingRates(value + 1);
- }
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
- ALOGV("checkOutputsForDevice() direct output sup formats %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadFormats(value + 1);
- }
- }
- if (profile->mChannelMasks[0] == 0) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
- ALOGV("checkOutputsForDevice() direct output sup channel masks %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadOutChannels(value + 1);
- }
- }
- if (((profile->mSamplingRates[0] == 0) &&
- (profile->mSamplingRates.size() < 2)) ||
- ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
- (profile->mFormats.size() < 2)) ||
- ((profile->mChannelMasks[0] == 0) &&
- (profile->mChannelMasks.size() < 2))) {
- ALOGW("checkOutputsForDevice() direct output missing param");
- mpClientInterface->closeOutput(output);
- output = 0;
- } else if (profile->mSamplingRates[0] == 0) {
- mpClientInterface->closeOutput(output);
- desc->mSamplingRate = profile->mSamplingRates[1];
- offloadInfo.sample_rate = desc->mSamplingRate;
- output = mpClientInterface->openOutput(
- profile->mModule->mHandle,
- &desc->mDevice,
- &desc->mSamplingRate,
- &desc->mFormat,
- &desc->mChannelMask,
- &desc->mLatency,
- desc->mFlags,
- &offloadInfo);
- }
-
- if (output != 0) {
- addOutput(output, desc);
- if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
- audio_io_handle_t duplicatedOutput = 0;
-
- // set initial stream volume for device
- applyStreamVolumes(output, device, 0, true);
-
- //TODO: configure audio effect output stage here
-
- // open a duplicating output thread for the new output and the primary output
- duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
- mPrimaryOutput);
- if (duplicatedOutput != 0) {
- // add duplicated output descriptor
- AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL);
- dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
- dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
- dupOutputDesc->mSamplingRate = desc->mSamplingRate;
- dupOutputDesc->mFormat = desc->mFormat;
- dupOutputDesc->mChannelMask = desc->mChannelMask;
- dupOutputDesc->mLatency = desc->mLatency;
- addOutput(duplicatedOutput, dupOutputDesc);
- applyStreamVolumes(duplicatedOutput, device, 0, true);
- } else {
- ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
- mPrimaryOutput, output);
- mpClientInterface->closeOutput(output);
- mOutputs.removeItem(output);
- nextAudioPortGeneration();
- output = 0;
- }
- }
- }
- }
- if (output == 0) {
- ALOGW("checkOutputsForDevice() could not open output for device %x", device);
- delete desc;
- profiles.removeAt(profile_index);
- profile_index--;
- } else {
- outputs.add(output);
- ALOGV("checkOutputsForDevice(): adding output %d", output);
- }
- }
-
- if (profiles.isEmpty()) {
- ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
- return BAD_VALUE;
- }
- } else { // Disconnect
- // check if one opened output is not needed any more after disconnecting one device
- for (size_t i = 0; i < mOutputs.size(); i++) {
- desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() &&
- !(desc->mProfile->mSupportedDevices.types() &
- mAvailableOutputDevices.types())) {
- ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i));
- outputs.add(mOutputs.keyAt(i));
- }
- }
- // Clear any profiles associated with the disconnected device.
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
- if (profile->mSupportedDevices.types() & device) {
- ALOGV("checkOutputsForDevice(): "
- "clearing direct output profile %zu on module %zu", j, i);
- if (profile->mSamplingRates[0] == 0) {
- profile->mSamplingRates.clear();
- profile->mSamplingRates.add(0);
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.clear();
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- }
- if (profile->mChannelMasks[0] == 0) {
- profile->mChannelMasks.clear();
- profile->mChannelMasks.add(0);
- }
- }
- }
- }
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& inputs,
- const String8 address)
-{
- AudioInputDescriptor *desc;
- if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
- // first list already open inputs that can be routed to this device
- for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
- ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
- inputs.add(mInputs.keyAt(input_index));
- }
- }
-
- // then look for input profiles that can be routed to this device
- SortedVector< sp<IOProfile> > profiles;
- for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
- {
- if (mHwModules[module_idx]->mHandle == 0) {
- continue;
- }
- for (size_t profile_index = 0;
- profile_index < mHwModules[module_idx]->mInputProfiles.size();
- profile_index++)
- {
- if (mHwModules[module_idx]->mInputProfiles[profile_index]->mSupportedDevices.types()
- & (device & ~AUDIO_DEVICE_BIT_IN)) {
- ALOGV("checkInputsForDevice(): adding profile %d from module %d",
- profile_index, module_idx);
- profiles.add(mHwModules[module_idx]->mInputProfiles[profile_index]);
- }
- }
- }
-
- if (profiles.isEmpty() && inputs.isEmpty()) {
- ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
- return BAD_VALUE;
- }
-
- // open inputs for matching profiles if needed. Direct inputs are also opened to
- // query for dynamic parameters and will be closed later by setDeviceConnectionState()
- for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
-
- sp<IOProfile> profile = profiles[profile_index];
- // nothing to do if one input is already opened for this profile
- size_t input_index;
- for (input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (desc->mProfile == profile) {
- break;
- }
- }
- if (input_index != mInputs.size()) {
- continue;
- }
-
- ALOGV("opening input for device 0x%X with params %s", device, address.string());
- desc = new AudioInputDescriptor(profile);
- desc->mDevice = device;
-
- audio_io_handle_t input = mpClientInterface->openInput(profile->mModule->mHandle,
- &desc->mDevice,
- &desc->mSamplingRate,
- &desc->mFormat,
- &desc->mChannelMask);
-
- if (input != 0) {
- if (!address.isEmpty()) {
- mpClientInterface->setParameters(input, addressToParameter(device, address));
- }
-
- // Here is where we step through and resolve any "dynamic" fields
- String8 reply;
- char *value;
- if (profile->mSamplingRates[0] == 0) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
- ALOGV("checkInputsForDevice() direct input sup sampling rates %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadSamplingRates(value + 1);
- }
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
- ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadFormats(value + 1);
- }
- }
- if (profile->mChannelMasks[0] == 0) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
- ALOGV("checkInputsForDevice() direct input sup channel masks %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadInChannels(value + 1);
- }
- }
- if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
- ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) ||
- ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) {
- ALOGW("checkInputsForDevice() direct input missing param");
- mpClientInterface->closeInput(input);
- input = 0;
- }
-
- if (input != 0) {
- addInput(input, desc);
- }
- } // endif input != 0
-
- if (input == 0) {
- ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
- delete desc;
- profiles.removeAt(profile_index);
- profile_index--;
- } else {
- inputs.add(input);
- ALOGV("checkInputsForDevice(): adding input %d", input);
- }
- } // end scan profiles
-
- if (profiles.isEmpty()) {
- ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
- return BAD_VALUE;
- }
- } else {
- // Disconnect
- // check if one opened input is not needed any more after disconnecting one device
- for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types())) {
- ALOGV("checkInputsForDevice(): disconnecting adding input %d",
- mInputs.keyAt(input_index));
- inputs.add(mInputs.keyAt(input_index));
- }
- }
- // Clear any profiles associated with the disconnected device.
- for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) {
- if (mHwModules[module_index]->mHandle == 0) {
- continue;
- }
- for (size_t profile_index = 0;
- profile_index < mHwModules[module_index]->mInputProfiles.size();
- profile_index++) {
- sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
- if (profile->mSupportedDevices.types() & device) {
- ALOGV("checkInputsForDevice(): clearing direct input profile %d on module %d",
- profile_index, module_index);
- if (profile->mSamplingRates[0] == 0) {
- profile->mSamplingRates.clear();
- profile->mSamplingRates.add(0);
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.clear();
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- }
- if (profile->mChannelMasks[0] == 0) {
- profile->mChannelMasks.clear();
- profile->mChannelMasks.add(0);
- }
- }
- }
- }
- } // end disconnect
-
- return NO_ERROR;
-}
-
-
-void AudioPolicyManager::closeOutput(audio_io_handle_t output)
-{
- ALOGV("closeOutput(%d)", output);
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- if (outputDesc == NULL) {
- ALOGW("closeOutput() unknown output %d", output);
- return;
- }
-
- // look for duplicated outputs connected to the output being removed.
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i);
- if (dupOutputDesc->isDuplicated() &&
- (dupOutputDesc->mOutput1 == outputDesc ||
- dupOutputDesc->mOutput2 == outputDesc)) {
- AudioOutputDescriptor *outputDesc2;
- if (dupOutputDesc->mOutput1 == outputDesc) {
- outputDesc2 = dupOutputDesc->mOutput2;
- } else {
- outputDesc2 = dupOutputDesc->mOutput1;
- }
- // As all active tracks on duplicated output will be deleted,
- // and as they were also referenced on the other output, the reference
- // count for their stream type must be adjusted accordingly on
- // the other output.
- for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
- int refCount = dupOutputDesc->mRefCount[j];
- outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
- }
- audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
- ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
-
- mpClientInterface->closeOutput(duplicatedOutput);
- delete mOutputs.valueFor(duplicatedOutput);
- mOutputs.removeItem(duplicatedOutput);
- }
- }
-
- AudioParameter param;
- param.add(String8("closing"), String8("true"));
- mpClientInterface->setParameters(output, param.toString());
-
- mpClientInterface->closeOutput(output);
- delete outputDesc;
- mOutputs.removeItem(output);
- mPreviousOutputs = mOutputs;
- nextAudioPortGeneration();
-}
-
-SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs)
-{
- SortedVector<audio_io_handle_t> outputs;
-
- ALOGVV("getOutputsForDevice() device %04x", device);
- for (size_t i = 0; i < openOutputs.size(); i++) {
- ALOGVV("output %d isDuplicated=%d device=%04x",
- i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
- if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
- ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
- outputs.add(openOutputs.keyAt(i));
- }
- }
- return outputs;
-}
-
-bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
- SortedVector<audio_io_handle_t>& outputs2)
-{
- if (outputs1.size() != outputs2.size()) {
- return false;
- }
- for (size_t i = 0; i < outputs1.size(); i++) {
- if (outputs1[i] != outputs2[i]) {
- return false;
- }
- }
- return true;
-}
-
-void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
-{
- audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
- audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
- SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
-
- if (!vectorsEqual(srcOutputs,dstOutputs)) {
- ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
- strategy, srcOutputs[0], dstOutputs[0]);
- // mute strategy while moving tracks from one output to another
- for (size_t i = 0; i < srcOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]);
- if (desc->isStrategyActive(strategy)) {
- setStrategyMute(strategy, true, srcOutputs[i]);
- setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
- }
- }
-
- // Move effects associated to this strategy from previous output to new output
- if (strategy == STRATEGY_MEDIA) {
- audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
- SortedVector<audio_io_handle_t> moved;
- for (size_t i = 0; i < mEffects.size(); i++) {
- EffectDescriptor *desc = mEffects.valueAt(i);
- if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
- desc->mIo != fxOutput) {
- if (moved.indexOf(desc->mIo) < 0) {
- ALOGV("checkOutputForStrategy() moving effect %d to output %d",
- mEffects.keyAt(i), fxOutput);
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo,
- fxOutput);
- moved.add(desc->mIo);
- }
- desc->mIo = fxOutput;
- }
- }
- }
- // Move tracks associated to this strategy from previous output to new output
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- if (getStrategy((audio_stream_type_t)i) == strategy) {
- mpClientInterface->invalidateStream((audio_stream_type_t)i);
- }
- }
- }
-}
-
-void AudioPolicyManager::checkOutputForAllStrategies()
-{
- checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
- checkOutputForStrategy(STRATEGY_PHONE);
- checkOutputForStrategy(STRATEGY_SONIFICATION);
- checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
- checkOutputForStrategy(STRATEGY_MEDIA);
- checkOutputForStrategy(STRATEGY_DTMF);
-}
-
-audio_io_handle_t AudioPolicyManager::getA2dpOutput()
-{
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
- if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
- return mOutputs.keyAt(i);
- }
- }
-
- return 0;
-}
-
-void AudioPolicyManager::checkA2dpSuspend()
-{
- audio_io_handle_t a2dpOutput = getA2dpOutput();
- if (a2dpOutput == 0) {
- mA2dpSuspended = false;
- return;
- }
-
- bool isScoConnected =
- (mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) != 0;
- // suspend A2DP output if:
- // (NOT already suspended) &&
- // ((SCO device is connected &&
- // (forced usage for communication || for record is SCO))) ||
- // (phone state is ringing || in call)
- //
- // restore A2DP output if:
- // (Already suspended) &&
- // ((SCO device is NOT connected ||
- // (forced usage NOT for communication && NOT for record is SCO))) &&
- // (phone state is NOT ringing && NOT in call)
- //
- if (mA2dpSuspended) {
- if ((!isScoConnected ||
- ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) &&
- ((mPhoneState != AUDIO_MODE_IN_CALL) &&
- (mPhoneState != AUDIO_MODE_RINGTONE))) {
-
- mpClientInterface->restoreOutput(a2dpOutput);
- mA2dpSuspended = false;
- }
- } else {
- if ((isScoConnected &&
- ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
- (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) ||
- ((mPhoneState == AUDIO_MODE_IN_CALL) ||
- (mPhoneState == AUDIO_MODE_RINGTONE))) {
-
- mpClientInterface->suspendOutput(a2dpOutput);
- mA2dpSuspended = true;
- }
- }
-}
-
-audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache)
-{
- audio_devices_t device = AUDIO_DEVICE_NONE;
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-
- ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
- if (index >= 0) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- if (patchDesc->mUid != mUidCached) {
- ALOGV("getNewOutputDevice() device %08x forced by patch %d",
- outputDesc->device(), outputDesc->mPatchHandle);
- return outputDesc->device();
- }
- }
-
- // check the following by order of priority to request a routing change if necessary:
- // 1: the strategy enforced audible is active on the output:
- // use device for strategy enforced audible
- // 2: we are in call or the strategy phone is active on the output:
- // use device for strategy phone
- // 3: the strategy sonification is active on the output:
- // use device for strategy sonification
- // 4: the strategy "respectful" sonification is active on the output:
- // use device for strategy "respectful" sonification
- // 5: the strategy media is active on the output:
- // use device for strategy media
- // 6: the strategy DTMF is active on the output:
- // use device for strategy DTMF
- if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
- device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
- } else if (isInCall() ||
- outputDesc->isStrategyActive(STRATEGY_PHONE)) {
- device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
- device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
- device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
- }
-
- ALOGV("getNewOutputDevice() selected device %x", device);
- return device;
-}
-
-audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
-{
- AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
-
- ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
- if (index >= 0) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- if (patchDesc->mUid != mUidCached) {
- ALOGV("getNewInputDevice() device %08x forced by patch %d",
- inputDesc->mDevice, inputDesc->mPatchHandle);
- return inputDesc->mDevice;
- }
- }
-
- audio_devices_t device = getDeviceForInputSource(inputDesc->mInputSource);
-
- ALOGV("getNewInputDevice() selected device %x", device);
- return device;
-}
-
-uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
- return (uint32_t)getStrategy(stream);
-}
-
-audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
- // By checking the range of stream before calling getStrategy, we avoid
- // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
- // and then return STRATEGY_MEDIA, but we want to return the empty set.
- if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_CNT) {
- return AUDIO_DEVICE_NONE;
- }
- audio_devices_t devices;
- AudioPolicyManager::routing_strategy strategy = getStrategy(stream);
- devices = getDeviceForStrategy(strategy, true /*fromCache*/);
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
- for (size_t i = 0; i < outputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
- if (outputDesc->isStrategyActive(strategy)) {
- devices = outputDesc->device();
- break;
- }
- }
- return devices;
-}
-
-AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
- audio_stream_type_t stream) {
- // stream to strategy mapping
- switch (stream) {
- case AUDIO_STREAM_VOICE_CALL:
- case AUDIO_STREAM_BLUETOOTH_SCO:
- return STRATEGY_PHONE;
- case AUDIO_STREAM_RING:
- case AUDIO_STREAM_ALARM:
- return STRATEGY_SONIFICATION;
- case AUDIO_STREAM_NOTIFICATION:
- return STRATEGY_SONIFICATION_RESPECTFUL;
- case AUDIO_STREAM_DTMF:
- return STRATEGY_DTMF;
- default:
- ALOGE("unknown stream type");
- case AUDIO_STREAM_SYSTEM:
- // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
- // while key clicks are played produces a poor result
- case AUDIO_STREAM_TTS:
- case AUDIO_STREAM_MUSIC:
- return STRATEGY_MEDIA;
- case AUDIO_STREAM_ENFORCED_AUDIBLE:
- return STRATEGY_ENFORCED_AUDIBLE;
- }
-}
-
-void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
- switch(stream) {
- case AUDIO_STREAM_MUSIC:
- checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
- updateDevicesAndOutputs();
- break;
- default:
- break;
- }
-}
-
-audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
- bool fromCache)
-{
- uint32_t device = AUDIO_DEVICE_NONE;
-
- if (fromCache) {
- ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
- strategy, mDeviceForStrategy[strategy]);
- return mDeviceForStrategy[strategy];
- }
- audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
- switch (strategy) {
-
- case STRATEGY_SONIFICATION_RESPECTFUL:
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC,
- SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
- // while media is playing on a remote device, use the the sonification behavior.
- // Note that we test this usecase before testing if media is playing because
- // the isStreamActive() method only informs about the activity of a stream, not
- // if it's for local playback. Note also that we use the same delay between both tests
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
- // while media is playing (or has recently played), use the same device
- device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
- } else {
- // when media is not playing anymore, fall back on the sonification behavior
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- }
-
- break;
-
- case STRATEGY_DTMF:
- if (!isInCall()) {
- // when off call, DTMF strategy follows the same rules as MEDIA strategy
- device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
- break;
- }
- // when in call, DTMF and PHONE strategies follow the same rules
- // FALL THROUGH
-
- case STRATEGY_PHONE:
- // for phone strategy, we first consider the forced use and then the available devices by order
- // of priority
- switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
- case AUDIO_POLICY_FORCE_BT_SCO:
- if (!isInCall() || strategy != STRATEGY_DTMF) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
- if (device) break;
- }
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
- if (device) break;
- // if SCO device is requested but no SCO device is available, fall back to default case
- // FALL THROUGH
-
- default: // FORCE_NONE
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
- if (!isInCall() &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- if (device) break;
- }
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- if (device) break;
- if (mPhoneState != AUDIO_MODE_IN_CALL) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- }
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE;
- if (device) break;
- device = mDefaultOutputDevice->mDeviceType;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
- }
- break;
-
- case AUDIO_POLICY_FORCE_SPEAKER:
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
- // A2DP speaker when forcing to speaker output
- if (!isInCall() &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- if (device) break;
- }
- if (mPhoneState != AUDIO_MODE_IN_CALL) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- }
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
- if (device) break;
- device = mDefaultOutputDevice->mDeviceType;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
- }
- break;
- }
- break;
-
- case STRATEGY_SONIFICATION:
-
- // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
- // handleIncallSonification().
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
- break;
- }
- // FALL THROUGH
-
- case STRATEGY_ENFORCED_AUDIBLE:
- // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
- // except:
- // - when in call where it doesn't default to STRATEGY_PHONE behavior
- // - in countries where not enforced in which case it follows STRATEGY_MEDIA
-
- if ((strategy == STRATEGY_SONIFICATION) ||
- (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
- }
- }
- // The second device used for sonification is the same as the device used by media strategy
- // FALL THROUGH
-
- case STRATEGY_MEDIA: {
- uint32_t device2 = AUDIO_DEVICE_NONE;
- if (strategy != STRATEGY_SONIFICATION) {
- // no sonification on remote submix (e.g. WFD)
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
- }
- if ((device2 == AUDIO_DEVICE_NONE) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- }
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- }
- if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
- // no sonification on aux digital (e.g. HDMI)
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- }
- if ((device2 == AUDIO_DEVICE_NONE) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
- }
-
- // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
- // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
- device |= device2;
- if (device) break;
- device = mDefaultOutputDevice->mDeviceType;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
- }
- } break;
-
- default:
- ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
- break;
- }
-
- ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
- return device;
-}
-
-void AudioPolicyManager::updateDevicesAndOutputs()
-{
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
- }
- mPreviousOutputs = mOutputs;
-}
-
-uint32_t AudioPolicyManager::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
- audio_devices_t prevDevice,
- uint32_t delayMs)
-{
- // mute/unmute strategies using an incompatible device combination
- // if muting, wait for the audio in pcm buffer to be drained before proceeding
- // if unmuting, unmute only after the specified delay
- if (outputDesc->isDuplicated()) {
- return 0;
- }
-
- uint32_t muteWaitMs = 0;
- audio_devices_t device = outputDesc->device();
- bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
-
- for (size_t i = 0; i < NUM_STRATEGIES; i++) {
- audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
- bool mute = shouldMute && (curDevice & device) && (curDevice != device);
- bool doMute = false;
-
- if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
- doMute = true;
- outputDesc->mStrategyMutedByDevice[i] = true;
- } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
- doMute = true;
- outputDesc->mStrategyMutedByDevice[i] = false;
- }
- if (doMute) {
- for (size_t j = 0; j < mOutputs.size(); j++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(j);
- // skip output if it does not share any device with current output
- if ((desc->supportedDevices() & outputDesc->supportedDevices())
- == AUDIO_DEVICE_NONE) {
- continue;
- }
- audio_io_handle_t curOutput = mOutputs.keyAt(j);
- ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
- mute ? "muting" : "unmuting", i, curDevice, curOutput);
- setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
- if (desc->isStrategyActive((routing_strategy)i)) {
- if (mute) {
- // FIXME: should not need to double latency if volume could be applied
- // immediately by the audioflinger mixer. We must account for the delay
- // between now and the next time the audioflinger thread for this output
- // will process a buffer (which corresponds to one buffer size,
- // usually 1/2 or 1/4 of the latency).
- if (muteWaitMs < desc->latency() * 2) {
- muteWaitMs = desc->latency() * 2;
- }
- }
- }
- }
- }
- }
-
- // temporary mute output if device selection changes to avoid volume bursts due to
- // different per device volumes
- if (outputDesc->isActive() && (device != prevDevice)) {
- if (muteWaitMs < outputDesc->latency() * 2) {
- muteWaitMs = outputDesc->latency() * 2;
- }
- for (size_t i = 0; i < NUM_STRATEGIES; i++) {
- if (outputDesc->isStrategyActive((routing_strategy)i)) {
- setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle);
- // do tempMute unmute after twice the mute wait time
- setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle,
- muteWaitMs *2, device);
- }
- }
- }
-
- // wait for the PCM output buffers to empty before proceeding with the rest of the command
- if (muteWaitMs > delayMs) {
- muteWaitMs -= delayMs;
- usleep(muteWaitMs * 1000);
- return muteWaitMs;
- }
- return 0;
-}
-
-uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
- audio_devices_t device,
- bool force,
- int delayMs,
- audio_patch_handle_t *patchHandle)
-{
- ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- AudioParameter param;
- uint32_t muteWaitMs;
-
- if (outputDesc->isDuplicated()) {
- muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs);
- muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs);
- return muteWaitMs;
- }
- // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
- // output profile
- if ((device != AUDIO_DEVICE_NONE) &&
- ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) {
- return 0;
- }
-
- // filter devices according to output selected
- device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types());
-
- audio_devices_t prevDevice = outputDesc->mDevice;
-
- ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
-
- if (device != AUDIO_DEVICE_NONE) {
- outputDesc->mDevice = device;
- }
- muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
-
- // Do not change the routing if:
- // - the requested device is AUDIO_DEVICE_NONE
- // - the requested device is the same as current device and force is not specified.
- // Doing this check here allows the caller to call setOutputDevice() without conditions
- if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) {
- ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
- return muteWaitMs;
- }
-
- ALOGV("setOutputDevice() changing device");
-
- // do the routing
- if (device == AUDIO_DEVICE_NONE) {
- resetOutputDevice(output, delayMs, NULL);
- } else {
- DeviceVector deviceList = mAvailableOutputDevices.getDevicesFromType(device);
- if (!deviceList.isEmpty()) {
- struct audio_patch patch;
- outputDesc->toAudioPortConfig(&patch.sources[0]);
- patch.num_sources = 1;
- patch.num_sinks = 0;
- for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
- deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
- patch.num_sinks++;
- }
- ssize_t index;
- if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
- index = mAudioPatches.indexOfKey(*patchHandle);
- } else {
- index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
- }
- sp< AudioPatch> patchDesc;
- audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- if (index >= 0) {
- patchDesc = mAudioPatches.valueAt(index);
- afPatchHandle = patchDesc->mAfPatchHandle;
- }
-
- status_t status = mpClientInterface->createAudioPatch(&patch,
- &afPatchHandle,
- delayMs);
- ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
- "num_sources %d num_sinks %d",
- status, afPatchHandle, patch.num_sources, patch.num_sinks);
- if (status == NO_ERROR) {
- if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
- addAudioPatch(patchDesc->mHandle, patchDesc);
- } else {
- patchDesc->mPatch = patch;
- }
- patchDesc->mAfPatchHandle = afPatchHandle;
- patchDesc->mUid = mUidCached;
- if (patchHandle) {
- *patchHandle = patchDesc->mHandle;
- }
- outputDesc->mPatchHandle = patchDesc->mHandle;
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- }
- }
- }
-
- // update stream volumes according to new device
- applyStreamVolumes(output, device, delayMs);
-
- return muteWaitMs;
-}
-
-status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output,
- int delayMs,
- audio_patch_handle_t *patchHandle)
-{
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- ssize_t index;
- if (patchHandle) {
- index = mAudioPatches.indexOfKey(*patchHandle);
- } else {
- index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
- }
- if (index < 0) {
- return INVALID_OPERATION;
- }
- sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs);
- ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
- outputDesc->mPatchHandle = 0;
- removeAudioPatch(patchDesc->mHandle);
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- return status;
-}
-
-status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
- audio_devices_t device,
- bool force,
- audio_patch_handle_t *patchHandle)
-{
- status_t status = NO_ERROR;
-
- AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
- if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
- inputDesc->mDevice = device;
-
- DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
- if (!deviceList.isEmpty()) {
- struct audio_patch patch;
- inputDesc->toAudioPortConfig(&patch.sinks[0]);
- patch.num_sinks = 1;
- //only one input device for now
- deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
- patch.num_sources = 1;
- ssize_t index;
- if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
- index = mAudioPatches.indexOfKey(*patchHandle);
- } else {
- index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
- }
- sp< AudioPatch> patchDesc;
- audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- if (index >= 0) {
- patchDesc = mAudioPatches.valueAt(index);
- afPatchHandle = patchDesc->mAfPatchHandle;
- }
-
- status_t status = mpClientInterface->createAudioPatch(&patch,
- &afPatchHandle,
- 0);
- ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
- status, afPatchHandle);
- if (status == NO_ERROR) {
- if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
- addAudioPatch(patchDesc->mHandle, patchDesc);
- } else {
- patchDesc->mPatch = patch;
- }
- patchDesc->mAfPatchHandle = afPatchHandle;
- patchDesc->mUid = mUidCached;
- if (patchHandle) {
- *patchHandle = patchDesc->mHandle;
- }
- inputDesc->mPatchHandle = patchDesc->mHandle;
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- }
- }
- }
- return status;
-}
-
-status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input,
- audio_patch_handle_t *patchHandle)
-{
- AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
- ssize_t index;
- if (patchHandle) {
- index = mAudioPatches.indexOfKey(*patchHandle);
- } else {
- index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
- }
- if (index < 0) {
- return INVALID_OPERATION;
- }
- sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
- inputDesc->mPatchHandle = 0;
- removeAudioPatch(patchDesc->mHandle);
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- return status;
-}
-
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask)
-{
- // Choose an input profile based on the requested capture parameters: select the first available
- // profile supporting all requested parameters.
-
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
- {
- sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
- // profile->log();
- if (profile->isCompatibleProfile(device, samplingRate, format,
- channelMask, AUDIO_OUTPUT_FLAG_NONE)) {
- return profile;
- }
- }
- }
- return NULL;
-}
-
-audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
-{
- uint32_t device = AUDIO_DEVICE_NONE;
- audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
- ~AUDIO_DEVICE_BIT_IN;
- switch (inputSource) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
- device = AUDIO_DEVICE_IN_VOICE_CALL;
- break;
- }
- // FALL THROUGH
-
- case AUDIO_SOURCE_DEFAULT:
- case AUDIO_SOURCE_MIC:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
- break;
- }
- // FALL THROUGH
-
- case AUDIO_SOURCE_VOICE_RECOGNITION:
- case AUDIO_SOURCE_HOTWORD:
- case AUDIO_SOURCE_VOICE_COMMUNICATION:
- if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO &&
- availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_CAMCORDER:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
- device = AUDIO_DEVICE_IN_BACK_MIC;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- case AUDIO_SOURCE_VOICE_CALL:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
- device = AUDIO_DEVICE_IN_VOICE_CALL;
- }
- break;
- case AUDIO_SOURCE_REMOTE_SUBMIX:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
- device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
- }
- break;
- default:
- ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
- break;
- }
- ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
- return device;
-}
-
-bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device)
-{
- if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
- device &= ~AUDIO_DEVICE_BIT_IN;
- if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
- return true;
- }
- return false;
-}
-
-audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
-{
- for (size_t i = 0; i < mInputs.size(); i++) {
- const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i);
- if ((input_descriptor->mRefCount > 0)
- && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
- return mInputs.keyAt(i);
- }
- }
- return 0;
-}
-
-
-audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
-{
- if (device == AUDIO_DEVICE_NONE) {
- // this happens when forcing a route update and no track is active on an output.
- // In this case the returned category is not important.
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (popcount(device) > 1) {
- // Multiple device selection is either:
- // - speaker + one other device: give priority to speaker in this case.
- // - one A2DP device + another device: happens with duplicated output. In this case
- // retain the device on the A2DP output as the other must not correspond to an active
- // selection if not the speaker.
- if (device & AUDIO_DEVICE_OUT_SPEAKER) {
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else {
- device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
- }
- }
-
- ALOGW_IF(popcount(device) != 1,
- "getDeviceForVolume() invalid device combination: %08x",
- device);
-
- return device;
-}
-
-AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
-{
- switch(getDeviceForVolume(device)) {
- case AUDIO_DEVICE_OUT_EARPIECE:
- return DEVICE_CATEGORY_EARPIECE;
- case AUDIO_DEVICE_OUT_WIRED_HEADSET:
- case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
- return DEVICE_CATEGORY_HEADSET;
- case AUDIO_DEVICE_OUT_SPEAKER:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
- case AUDIO_DEVICE_OUT_AUX_DIGITAL:
- case AUDIO_DEVICE_OUT_USB_ACCESSORY:
- case AUDIO_DEVICE_OUT_USB_DEVICE:
- case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
- default:
- return DEVICE_CATEGORY_SPEAKER;
- }
-}
-
-float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi)
-{
- device_category deviceCategory = getDeviceCategory(device);
- const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
-
- // the volume index in the UI is relative to the min and max volume indices for this stream type
- int nbSteps = 1 + curve[VOLMAX].mIndex -
- curve[VOLMIN].mIndex;
- int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
- (streamDesc.mIndexMax - streamDesc.mIndexMin);
-
- // find what part of the curve this index volume belongs to, or if it's out of bounds
- int segment = 0;
- if (volIdx < curve[VOLMIN].mIndex) { // out of bounds
- return 0.0f;
- } else if (volIdx < curve[VOLKNEE1].mIndex) {
- segment = 0;
- } else if (volIdx < curve[VOLKNEE2].mIndex) {
- segment = 1;
- } else if (volIdx <= curve[VOLMAX].mIndex) {
- segment = 2;
- } else { // out of bounds
- return 1.0f;
- }
-
- // linear interpolation in the attenuation table in dB
- float decibels = curve[segment].mDBAttenuation +
- ((float)(volIdx - curve[segment].mIndex)) *
- ( (curve[segment+1].mDBAttenuation -
- curve[segment].mDBAttenuation) /
- ((float)(curve[segment+1].mIndex -
- curve[segment].mIndex)) );
-
- float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
- ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
- curve[segment].mIndex, volIdx,
- curve[segment+1].mIndex,
- curve[segment].mDBAttenuation,
- decibels,
- curve[segment+1].mDBAttenuation,
- amplification);
-
- return amplification;
-}
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
-};
-
-// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
-// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
-// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
-// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT]
- [AudioPolicyManager::DEVICE_CATEGORY_CNT] = {
- { // AUDIO_STREAM_VOICE_CALL
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_SYSTEM
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_RING
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_MUSIC
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_ALARM
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_NOTIFICATION
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_BLUETOOTH_SCO
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_ENFORCED_AUDIBLE
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_DTMF
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
- { // AUDIO_STREAM_TTS
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
- },
-};
-
-void AudioPolicyManager::initializeVolumeCurves()
-{
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
- mStreams[i].mVolumeCurve[j] =
- sVolumeProfiles[i][j];
- }
- }
-
- // Check availability of DRC on speaker path: if available, override some of the speaker curves
- if (mSpeakerDrcEnabled) {
- mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sDefaultSystemVolumeCurveDrc;
- mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- }
-}
-
-float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
- int index,
- audio_io_handle_t output,
- audio_devices_t device)
-{
- float volume = 1.0;
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- StreamDescriptor &streamDesc = mStreams[stream];
-
- if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->device();
- }
-
- // if volume is not 0 (not muted), force media volume to max on digital output
- if (stream == AUDIO_STREAM_MUSIC &&
- index != mStreams[stream].mIndexMin &&
- (device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
- device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET ||
- device == AUDIO_DEVICE_OUT_USB_ACCESSORY ||
- device == AUDIO_DEVICE_OUT_USB_DEVICE)) {
- return 1.0;
- }
-
- volume = volIndexToAmpl(device, streamDesc, index);
-
- // if a headset is connected, apply the following rules to ring tones and notifications
- // to avoid sound level bursts in user's ears:
- // - always attenuate ring tones and notifications volume by 6dB
- // - if music is playing, always limit the volume to current music volume,
- // with a minimum threshold at -36dB so that notification is always perceived.
- const routing_strategy stream_strategy = getStrategy(stream);
- if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
- AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
- AUDIO_DEVICE_OUT_WIRED_HEADSET |
- AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
- ((stream_strategy == STRATEGY_SONIFICATION)
- || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
- || (stream == AUDIO_STREAM_SYSTEM)
- || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) &&
- streamDesc.mCanBeMuted) {
- volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
- // when the phone is ringing we must consider that music could have been paused just before
- // by the music application and behave as if music was active if the last music track was
- // just stopped
- if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
- mLimitRingtoneVolume) {
- audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
- float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
- mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
- output,
- musicDevice);
- float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
- musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
- if (volume > minVol) {
- volume = minVol;
- ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
- }
- }
- }
-
- return volume;
-}
-
-status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
- int index,
- audio_io_handle_t output,
- audio_devices_t device,
- int delayMs,
- bool force)
-{
-
- // do not change actual stream volume if the stream is muted
- if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
- ALOGVV("checkAndSetVolume() stream %d muted count %d",
- stream, mOutputs.valueFor(output)->mMuteCount[stream]);
- return NO_ERROR;
- }
-
- // do not change in call volume if bluetooth is connected and vice versa
- if ((stream == AUDIO_STREAM_VOICE_CALL &&
- mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
- (stream == AUDIO_STREAM_BLUETOOTH_SCO &&
- mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) {
- ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
- stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
- return INVALID_OPERATION;
- }
-
- float volume = computeVolume(stream, index, output, device);
- // We actually change the volume if:
- // - the float value returned by computeVolume() changed
- // - the force flag is set
- if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
- force) {
- mOutputs.valueFor(output)->mCurVolume[stream] = volume;
- ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
- // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
- // enabled
- if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
- mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
- }
- mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
- }
-
- if (stream == AUDIO_STREAM_VOICE_CALL ||
- stream == AUDIO_STREAM_BLUETOOTH_SCO) {
- float voiceVolume;
- // Force voice volume to max for bluetooth SCO as volume is managed by the headset
- if (stream == AUDIO_STREAM_VOICE_CALL) {
- voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
- } else {
- voiceVolume = 1.0;
- }
-
- if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
- mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
- mLastVoiceVolume = voiceVolume;
- }
- }
-
- return NO_ERROR;
-}
-
-void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
- audio_devices_t device,
- int delayMs,
- bool force)
-{
- ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
-
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- checkAndSetVolume((audio_stream_type_t)stream,
- mStreams[stream].getVolumeIndex(device),
- output,
- device,
- delayMs,
- force);
- }
-}
-
-void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
- bool on,
- audio_io_handle_t output,
- int delayMs,
- audio_devices_t device)
-{
- ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- if (getStrategy((audio_stream_type_t)stream) == strategy) {
- setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
- }
- }
-}
-
-void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
- bool on,
- audio_io_handle_t output,
- int delayMs,
- audio_devices_t device)
-{
- StreamDescriptor &streamDesc = mStreams[stream];
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->device();
- }
-
- ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
- stream, on, output, outputDesc->mMuteCount[stream], device);
-
- if (on) {
- if (outputDesc->mMuteCount[stream] == 0) {
- if (streamDesc.mCanBeMuted &&
- ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
- (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) {
- checkAndSetVolume(stream, 0, output, device, delayMs);
- }
- }
- // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
- outputDesc->mMuteCount[stream]++;
- } else {
- if (outputDesc->mMuteCount[stream] == 0) {
- ALOGV("setStreamMute() unmuting non muted stream!");
- return;
- }
- if (--outputDesc->mMuteCount[stream] == 0) {
- checkAndSetVolume(stream,
- streamDesc.getVolumeIndex(device),
- output,
- device,
- delayMs);
- }
- }
-}
-
-void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
- bool starting, bool stateChange)
-{
- // if the stream pertains to sonification strategy and we are in call we must
- // mute the stream if it is low visibility. If it is high visibility, we must play a tone
- // in the device used for phone strategy and play the tone if the selected device does not
- // interfere with the device used for phone strategy
- // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
- // many times as there are active tracks on the output
- const routing_strategy stream_strategy = getStrategy(stream);
- if ((stream_strategy == STRATEGY_SONIFICATION) ||
- ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
- ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
- stream, starting, outputDesc->mDevice, stateChange);
- if (outputDesc->mRefCount[stream]) {
- int muteCount = 1;
- if (stateChange) {
- muteCount = outputDesc->mRefCount[stream];
- }
- if (audio_is_low_visibility(stream)) {
- ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
- for (int i = 0; i < muteCount; i++) {
- setStreamMute(stream, starting, mPrimaryOutput);
- }
- } else {
- ALOGV("handleIncallSonification() high visibility");
- if (outputDesc->device() &
- getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
- ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
- for (int i = 0; i < muteCount; i++) {
- setStreamMute(stream, starting, mPrimaryOutput);
- }
- }
- if (starting) {
- mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
- AUDIO_STREAM_VOICE_CALL);
- } else {
- mpClientInterface->stopTone();
- }
- }
- }
- }
-}
-
-bool AudioPolicyManager::isInCall()
-{
- return isStateInCall(mPhoneState);
-}
-
-bool AudioPolicyManager::isStateInCall(int state) {
- return ((state == AUDIO_MODE_IN_CALL) ||
- (state == AUDIO_MODE_IN_COMMUNICATION));
-}
-
-uint32_t AudioPolicyManager::getMaxEffectsCpuLoad()
-{
- return MAX_EFFECTS_CPU_LOAD;
-}
-
-uint32_t AudioPolicyManager::getMaxEffectsMemory()
-{
- return MAX_EFFECTS_MEMORY;
-}
-
-
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
- const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
- mChannelMask(0), mLatency(0),
- mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0),
- mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
-{
- // clear usage count for all stream types
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- mRefCount[i] = 0;
- mCurVolume[i] = -1.0;
- mMuteCount[i] = 0;
- mStopTime[i] = 0;
- }
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mStrategyMutedByDevice[i] = false;
- }
- if (profile != NULL) {
- mSamplingRate = profile->mSamplingRates[0];
- mFormat = profile->mFormats[0];
- mChannelMask = profile->mChannelMasks[0];
- mFlags = profile->mFlags;
- }
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
- } else {
- return mDevice;
- }
-}
-
-uint32_t AudioPolicyManager::AudioOutputDescriptor::latency()
-{
- if (isDuplicated()) {
- return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
- } else {
- return mLatency;
- }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
- const AudioOutputDescriptor *outputDesc)
-{
- if (isDuplicated()) {
- return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
- } else if (outputDesc->isDuplicated()){
- return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
- } else {
- return (mProfile->mModule == outputDesc->mProfile->mModule);
- }
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
- int delta)
-{
- // forward usage count change to attached outputs
- if (isDuplicated()) {
- mOutput1->changeRefCount(stream, delta);
- mOutput2->changeRefCount(stream, delta);
- }
- if ((delta + (int)mRefCount[stream]) < 0) {
- ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
- delta, stream, mRefCount[stream]);
- mRefCount[stream] = 0;
- return;
- }
- mRefCount[stream] += delta;
- ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices()
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
- } else {
- return mProfile->mSupportedDevices.types() ;
- }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
-{
- return isStrategyActive(NUM_STRATEGIES, inPastMs);
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if ((sysTime == 0) && (inPastMs != 0)) {
- sysTime = systemTime();
- }
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- if (((getStrategy((audio_stream_type_t)i) == strategy) ||
- (NUM_STRATEGIES == strategy)) &&
- isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if (mRefCount[stream] != 0) {
- return true;
- }
- if (inPastMs == 0) {
- return false;
- }
- if (sysTime == 0) {
- sysTime = systemTime();
- }
- if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
- return true;
- }
- return false;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->sample_rate = mSamplingRate;
- dstConfig->channel_mask = mChannelMask;
- dstConfig->format = mFormat;
- dstConfig->gain.index = -1;
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT;
- // use supplied variable configuration parameters if any
- if (srcConfig != NULL) {
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- dstConfig->sample_rate = srcConfig->sample_rate;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- dstConfig->format = srcConfig->format;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = srcConfig->gain;
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- }
- }
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class =
- mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
- result.append(buffer);
- snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", device());
- result.append(buffer);
- snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
- result.append(buffer);
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
- i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0), mSamplingRate(0),
- mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
- mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0), mRefCount(0),
- mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile)
-{
- if (profile != NULL) {
- mSamplingRate = profile->mSamplingRates[0];
- mFormat = profile->mFormats[0];
- mChannelMask = profile->mChannelMasks[0];
- }
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SINK;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->sample_rate = mSamplingRate;
- dstConfig->channel_mask = mChannelMask;
- dstConfig->format = mFormat;
- dstConfig->gain.index = -1;
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT;
- // use supplied variable configuration parameters if any
- if (srcConfig != NULL) {
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- dstConfig->sample_rate = srcConfig->sample_rate;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- dstConfig->format = srcConfig->format;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = srcConfig->gain;
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- }
- }
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.source = mInputSource;
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-AudioPolicyManager::StreamDescriptor::StreamDescriptor()
- : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
-{
- mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
-}
-
-int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device)
-{
- device = AudioPolicyManager::getDeviceForVolume(device);
- // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
- if (mIndexCur.indexOfKey(device) < 0) {
- device = AUDIO_DEVICE_OUT_DEFAULT;
- }
- return mIndexCur.valueFor(device);
-}
-
-void AudioPolicyManager::StreamDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%s %02d %02d ",
- mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
- result.append(buffer);
- for (size_t i = 0; i < mIndexCur.size(); i++) {
- snprintf(buffer, SIZE, "%04x : %02d, ",
- mIndexCur.keyAt(i),
- mIndexCur.valueAt(i));
- result.append(buffer);
- }
- result.append("\n");
-
- write(fd, result.string(), result.size());
-}
-
-// --- EffectDescriptor class implementation
-
-status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " I/O: %d\n", mIo);
- result.append(buffer);
- snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
- result.append(buffer);
- snprintf(buffer, SIZE, " Session: %d\n", mSession);
- result.append(buffer);
- snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
- result.append(buffer);
- snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- HwModule class implementation
-
-AudioPolicyManager::HwModule::HwModule(const char *name)
- : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0)
-{
-}
-
-AudioPolicyManager::HwModule::~HwModule()
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- mOutputProfiles[i]->mSupportedDevices.clear();
- }
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- mInputProfiles[i]->mSupportedDevices.clear();
- }
- free((void *)mName);
-}
-
-status_t AudioPolicyManager::HwModule::loadInput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadInChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadInput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadInput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadInput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadInput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadInput() adding input Supported Devices %04x",
- profile->mSupportedDevices.types());
-
- mInputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManager::HwModule::loadOutput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadOutChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = parseFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadOutput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadOutput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadOutput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadOutput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
- profile->mSupportedDevices.types(), profile->mFlags);
-
- mOutputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManager::HwModule::loadDevice(cnode *root)
-{
- cnode *node = root->first_child;
-
- audio_devices_t type = AUDIO_DEVICE_NONE;
- while (node) {
- if (strcmp(node->name, DEVICE_TYPE) == 0) {
- type = parseDeviceNames((char *)node->value);
- break;
- }
- node = node->next;
- }
- if (type == AUDIO_DEVICE_NONE ||
- (!audio_is_input_device(type) && !audio_is_output_device(type))) {
- ALOGW("loadDevice() bad type %08x", type);
- return BAD_VALUE;
- }
- sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
- deviceDesc->mModule = this;
-
- node = root->first_child;
- while (node) {
- if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
- deviceDesc->mAddress = String8((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- if (audio_is_input_device(type)) {
- deviceDesc->loadInChannels((char *)node->value);
- } else {
- deviceDesc->loadOutChannels((char *)node->value);
- }
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- deviceDesc->loadGains(node);
- }
- node = node->next;
- }
-
- ALOGV("loadDevice() adding device name %s type %08x address %s",
- deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
-
- mDeclaredDevices.add(deviceDesc);
-
- return NO_ERROR;
-}
-
-void AudioPolicyManager::HwModule::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " - name: %s\n", mName);
- result.append(buffer);
- snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
- result.append(buffer);
- write(fd, result.string(), result.size());
- if (mOutputProfiles.size()) {
- write(fd, " - outputs:\n", strlen(" - outputs:\n"));
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " output %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mOutputProfiles[i]->dump(fd);
- }
- }
- if (mInputProfiles.size()) {
- write(fd, " - inputs:\n", strlen(" - inputs:\n"));
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " input %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mInputProfiles[i]->dump(fd);
- }
- }
- if (mDeclaredDevices.size()) {
- write(fd, " - devices:\n", strlen(" - devices:\n"));
- for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
- mDeclaredDevices[i]->dump(fd, 4, i);
- }
- }
-}
-
-// --- AudioPort class implementation
-
-void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const
-{
- port->role = mRole;
- port->type = mType;
- unsigned int i;
- for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
- port->sample_rates[i] = mSamplingRates[i];
- }
- port->num_sample_rates = i;
- for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
- port->channel_masks[i] = mChannelMasks[i];
- }
- port->num_channel_masks = i;
- for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
- port->formats[i] = mFormats[i];
- }
- port->num_formats = i;
-
- ALOGV("AudioPort::toAudioPort() num gains %d", mGains.size());
-
- for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
- port->gains[i] = mGains[i]->mGain;
- }
- port->num_gains = i;
-}
-
-
-void AudioPolicyManager::AudioPort::loadSamplingRates(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
- // rates should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mSamplingRates.add(0);
- return;
- }
-
- while (str != NULL) {
- uint32_t rate = atoi(str);
- if (rate != 0) {
- ALOGV("loadSamplingRates() adding rate %d", rate);
- mSamplingRates.add(rate);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadFormats(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mFormats indicates the supported formats
- // should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mFormats.add(AUDIO_FORMAT_DEFAULT);
- return;
- }
-
- while (str != NULL) {
- audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- str);
- if (format != AUDIO_FORMAT_DEFAULT) {
- mFormats.add(format);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadInChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadInChannels() %s", name);
-
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- ALOGV("loadInChannels() adding channelMask %04x", channelMask);
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadOutChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadOutChannels() %s", name);
-
- // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
- // masks should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-audio_gain_mode_t AudioPolicyManager::AudioPort::loadGainMode(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadGainMode() %s", name);
- audio_gain_mode_t mode = 0;
- while (str != NULL) {
- mode |= (audio_gain_mode_t)stringToEnum(sGainModeNameToEnumTable,
- ARRAY_SIZE(sGainModeNameToEnumTable),
- str);
- str = strtok(NULL, "|");
- }
- return mode;
-}
-
-void AudioPolicyManager::AudioPort::loadGain(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<AudioGain> gain = new AudioGain();
-
- while (node) {
- if (strcmp(node->name, GAIN_MODE) == 0) {
- gain->mGain.mode = loadGainMode((char *)node->value);
- } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
- if ((mType == AUDIO_PORT_TYPE_DEVICE && mRole == AUDIO_PORT_ROLE_SOURCE) ||
- (mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK)) {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- (char *)node->value);
- } else {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- (char *)node->value);
- }
- } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
- gain->mGain.min_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
- gain->mGain.max_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
- gain->mGain.default_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
- gain->mGain.step_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
- gain->mGain.min_ramp_ms = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
- gain->mGain.max_ramp_ms = atoi((char *)node->value);
- }
- node = node->next;
- }
-
- ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
- gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
-
- if (gain->mGain.mode == 0) {
- return;
- }
- mGains.add(gain);
-}
-
-void AudioPolicyManager::AudioPort::loadGains(cnode *root)
-{
- cnode *node = root->first_child;
- while (node) {
- ALOGV("loadGains() loading gain %s", node->name);
- loadGain(node);
- node = node->next;
- }
-}
-
-void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- if (mName.size() != 0) {
- snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
- result.append(buffer);
- }
-
- if (mSamplingRates.size() != 0) {
- snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
- result.append(buffer);
- result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mChannelMasks.size() != 0) {
- snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
- result.append(buffer);
- result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mFormats.size() != 0) {
- snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mFormats.size(); i++) {
- snprintf(buffer, SIZE, "%-48s", enumToString(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- mFormats[i]));
- result.append(buffer);
- result.append(i == (mFormats.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
- write(fd, result.string(), result.size());
- if (mGains.size() != 0) {
- snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
- write(fd, buffer, strlen(buffer) + 1);
- result.append(buffer);
- for (size_t i = 0; i < mGains.size(); i++) {
- mGains[i]->dump(fd, spaces + 2, i);
- }
- }
-}
-
-// --- AudioGain class implementation
-
-AudioPolicyManager::AudioGain::AudioGain()
-{
- memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-}
-
-// --- IOProfile class implementation
-
-AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role,
- HwModule *module)
- : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module), mFlags((audio_output_flags_t)0)
-{
-}
-
-AudioPolicyManager::IOProfile::~IOProfile()
-{
-}
-
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags) const
-{
- if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) {
- return false;
- }
-
- if ((mSupportedDevices.types() & device) != device) {
- return false;
- }
- if ((mFlags & flags) != flags) {
- return false;
- }
- size_t i;
- for (i = 0; i < mSamplingRates.size(); i++)
- {
- if (mSamplingRates[i] == samplingRate) {
- break;
- }
- }
- if (i == mSamplingRates.size()) {
- return false;
- }
- for (i = 0; i < mFormats.size(); i++)
- {
- if (mFormats[i] == format) {
- break;
- }
- }
- if (i == mFormats.size()) {
- return false;
- }
- for (i = 0; i < mChannelMasks.size(); i++)
- {
- if (mChannelMasks[i] == channelMask) {
- break;
- }
- }
- if (i == mChannelMasks.size()) {
- return false;
- }
- return true;
-}
-
-void AudioPolicyManager::IOProfile::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- AudioPort::dump(fd, 4);
-
- snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " - devices:\n");
- result.append(buffer);
- write(fd, result.string(), result.size());
- for (size_t i = 0; i < mSupportedDevices.size(); i++) {
- mSupportedDevices[i]->dump(fd, 6, i);
- }
-}
-
-void AudioPolicyManager::IOProfile::log()
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- ALOGV(" - sampling rates: ");
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- ALOGV(" %d", mSamplingRates[i]);
- }
-
- ALOGV(" - channel masks: ");
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV(" 0x%04x", mChannelMasks[i]);
- }
-
- ALOGV(" - formats: ");
- for (size_t i = 0; i < mFormats.size(); i++) {
- ALOGV(" 0x%08x", mFormats[i]);
- }
-
- ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
- ALOGV(" - flags: 0x%04x\n", mFlags);
-}
-
-
-// --- DeviceDescriptor implementation
-
-bool AudioPolicyManager::DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
-{
- // Devices are considered equal if they:
- // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
- // - have the same address or one device does not specify the address
- // - have the same channel mask or one device does not specify the channel mask
- return (mDeviceType == other->mDeviceType) &&
- (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
- (mChannelMask == 0 || other->mChannelMask == 0 ||
- mChannelMask == other->mChannelMask);
-}
-
-void AudioPolicyManager::DeviceVector::refreshTypes()
-{
- mDeviceTypes = AUDIO_DEVICE_NONE;
- for(size_t i = 0; i < size(); i++) {
- mDeviceTypes |= itemAt(i)->mDeviceType;
- }
- ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
-}
-
-ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
-{
- for(size_t i = 0; i < size(); i++) {
- if (item->equals(itemAt(i))) {
- return i;
- }
- }
- return -1;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::add(const sp<DeviceDescriptor>& item)
-{
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ret = SortedVector::add(item);
- if (ret >= 0) {
- refreshTypes();
- }
- } else {
- ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
- ret = -1;
- }
- return ret;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::remove(const sp<DeviceDescriptor>& item)
-{
- size_t i;
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
- } else {
- ret = SortedVector::removeAt(ret);
- if (ret >= 0) {
- refreshTypes();
- }
- }
- return ret;
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromType(audio_devices_t types)
-{
- DeviceVector deviceList;
-
- uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
- types &= ~role_bit;
-
- while (types) {
- uint32_t i = 31 - __builtin_clz(types);
- uint32_t type = 1 << i;
- types &= ~type;
- add(new DeviceDescriptor(String8(""), type | role_bit));
- }
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromName(char *name,
- const DeviceVector& declaredDevices)
-{
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- audio_devices_t type = stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- if (type != AUDIO_DEVICE_NONE) {
- add(new DeviceDescriptor(String8(""), type));
- } else {
- sp<DeviceDescriptor> deviceDesc =
- declaredDevices.getDeviceFromName(String8(devName));
- if (deviceDesc != 0) {
- add(deviceDesc);
- }
- }
- }
- devName = strtok(NULL, "|");
- }
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDevice(
- audio_devices_t type, String8 address) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- device = itemAt(i);
- if (itemAt(i)->mAddress = address) {
- break;
- }
- }
- }
- ALOGV("DeviceVector::getDevice() for type %d address %s found %p",
- type, address.string(), device.get());
- return device;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromId(
- audio_port_handle_t id) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%d)->mId %d", id, i, itemAt(i)->mId);
- if (itemAt(i)->mId == id) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType(
- audio_devices_t type) const
-{
- DeviceVector devices;
- for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
- if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
- devices.add(itemAt(i));
- type &= ~itemAt(i)->mDeviceType;
- ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
- itemAt(i)->mDeviceType, itemAt(i).get());
- }
- }
- return devices;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromName(
- const String8& name) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mName == name) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- dstConfig->id = mId;
- dstConfig->role = audio_is_output_device(mDeviceType) ?
- AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
- dstConfig->channel_mask = mChannelMask;
- dstConfig->gain.index = -1;
- dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK;
- // use supplied variable configuration parameters if any
- if (srcConfig != NULL) {
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- if (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = srcConfig->gain;
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- }
- }
- dstConfig->ext.device.type = mDeviceType;
- dstConfig->ext.device.hw_module = mModule->mHandle;
- strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const
-{
- ALOGV("DeviceVector::toAudioPort() handle %d type %x", mId, mDeviceType);
- AudioPort::toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.device.type = mDeviceType;
- port->ext.device.hw_module = mModule->mHandle;
- strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-status_t AudioPolicyManager::DeviceDescriptor::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
- result.append(buffer);
- if (mId != 0) {
- snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
- enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mDeviceType));
- result.append(buffer);
- if (mAddress.size() != 0) {
- snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
- result.append(buffer);
- }
- if (mChannelMask != AUDIO_CHANNEL_NONE) {
- snprintf(buffer, SIZE, "%*s- channel mask: %08x\n", spaces, "", mChannelMask);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
- AudioPort::dump(fd, spaces);
-
- return NO_ERROR;
-}
-
-
-// --- audio_policy.conf file parsing
-
-audio_output_flags_t AudioPolicyManager::parseFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sFlagNameToEnumTable,
- ARRAY_SIZE(sFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- //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 ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flag |= AUDIO_OUTPUT_FLAG_DIRECT;
- }
-
- return (audio_output_flags_t)flag;
-}
-
-audio_devices_t AudioPolicyManager::parseDeviceNames(char *name)
-{
- uint32_t device = 0;
-
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- device |= stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- }
- devName = strtok(NULL, "|");
- }
- return device;
-}
-
-void AudioPolicyManager::loadHwModule(cnode *root)
-{
- status_t status = NAME_NOT_FOUND;
- cnode *node;
- HwModule *module = new HwModule(root->name);
-
- node = config_find(root, DEVICES_TAG);
- if (node != NULL) {
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModule() loading device %s", node->name);
- status_t tmpStatus = module->loadDevice(node);
- if (status == NAME_NOT_FOUND || status == NO_ERROR) {
- status = tmpStatus;
- }
- node = node->next;
- }
- }
- node = config_find(root, OUTPUTS_TAG);
- if (node != NULL) {
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModule() loading output %s", node->name);
- status_t tmpStatus = module->loadOutput(node);
- if (status == NAME_NOT_FOUND || status == NO_ERROR) {
- status = tmpStatus;
- }
- node = node->next;
- }
- }
- node = config_find(root, INPUTS_TAG);
- if (node != NULL) {
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModule() loading input %s", node->name);
- status_t tmpStatus = module->loadInput(node);
- if (status == NAME_NOT_FOUND || status == NO_ERROR) {
- status = tmpStatus;
- }
- node = node->next;
- }
- }
- loadGlobalConfig(root, module);
-
- if (status == NO_ERROR) {
- mHwModules.add(module);
- } else {
- delete module;
- }
-}
-
-void AudioPolicyManager::loadHwModules(cnode *root)
-{
- cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
- if (node == NULL) {
- return;
- }
-
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModules() loading module %s", node->name);
- loadHwModule(node);
- node = node->next;
- }
-}
-
-void AudioPolicyManager::loadGlobalConfig(cnode *root, HwModule *module)
-{
- cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
- if (node == NULL) {
- return;
- }
- DeviceVector declaredDevices;
- if (module != NULL) {
- declaredDevices = module->mDeclaredDevices;
- }
-
- node = node->first_child;
- while (node) {
- if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
- mAvailableOutputDevices.loadDevicesFromName((char *)node->value,
- declaredDevices);
- ALOGV("loadGlobalConfig() Attached Output Devices %08x",
- mAvailableOutputDevices.types());
- } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
- audio_devices_t device = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- (char *)node->value);
- if (device != AUDIO_DEVICE_NONE) {
- mDefaultOutputDevice = new DeviceDescriptor(String8(""), device);
- } else {
- ALOGW("loadGlobalConfig() default device not specified");
- }
- ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType);
- } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
- mAvailableInputDevices.loadDevicesFromName((char *)node->value,
- declaredDevices);
- ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
- } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
- mSpeakerDrcEnabled = stringToBool((char *)node->value);
- ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
- }
- node = node->next;
- }
-}
-
-status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
-{
- cnode *root;
- char *data;
-
- data = (char *)load_file(path, NULL);
- if (data == NULL) {
- return -ENODEV;
- }
- root = config_node("", "");
- config_load(root, data);
-
- loadHwModules(root);
- // legacy audio_policy.conf files have one global_configuration section
- loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY));
- config_free(root);
- free(root);
- free(data);
-
- ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
-
- return NO_ERROR;
-}
-
-void AudioPolicyManager::defaultAudioPolicyConfig(void)
-{
- HwModule *module;
- sp<IOProfile> profile;
- sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_IN_BUILTIN_MIC);
- mAvailableOutputDevices.add(mDefaultOutputDevice);
- mAvailableInputDevices.add(defaultInputDevice);
-
- module = new HwModule("primary");
-
- profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module);
- profile->mSamplingRates.add(44100);
- profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
- profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
- profile->mSupportedDevices.add(mDefaultOutputDevice);
- profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
- module->mOutputProfiles.add(profile);
-
- profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module);
- profile->mSamplingRates.add(8000);
- profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
- profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
- profile->mSupportedDevices.add(defaultInputDevice);
- module->mInputProfiles.add(profile);
-
- mHwModules.add(module);
+ delete interface;
}
}; // namespace android
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index 0dd015a..9f2b473 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2009 The Android Open Source Project
@@ -20,470 +20,53 @@
#include <stdint.h>
#include <sys/types.h>
-#include <cutils/config_utils.h>
-#include <cutils/misc.h>
#include <utils/Timers.h>
#include <utils/Errors.h>
#include <utils/KeyedVector.h>
-#include <utils/SortedVector.h>
-#include "AudioPolicyInterface.h"
+#include <hardware_legacy/AudioPolicyManagerBase.h>
-namespace android {
+namespace android_audio_legacy {
// ----------------------------------------------------------------------------
-// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
-#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
-// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
-#define SONIFICATION_HEADSET_VOLUME_MIN 0.016
-// Time in milliseconds during which we consider that music is still active after a music
-// track was stopped - see computeVolume()
-#define SONIFICATION_HEADSET_MUSIC_DELAY 5000
-// Time in milliseconds after media stopped playing during which we consider that the
-// sonification should be as unobtrusive as during the time media was playing.
-#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
-// Time in milliseconds during witch some streams are muted while the audio path
-// is switched
-#define MUTE_TIME_MS 2000
-
-#define NUM_TEST_OUTPUTS 5
-
-#define NUM_VOL_CURVE_KNEES 2
-
-// Default minimum length allowed for offloading a compressed track
-// Can be overridden by the audio.offload.min.duration.secs property
-#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManager implements audio policy manager behavior common to all platforms.
-// ----------------------------------------------------------------------------
-
-class AudioPolicyManager: public AudioPolicyInterface
-#ifdef AUDIO_POLICY_TEST
- , public Thread
-#endif //AUDIO_POLICY_TEST
+class AudioPolicyManager: public AudioPolicyManagerBase
{
public:
- AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
- virtual ~AudioPolicyManager();
+ AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+ : AudioPolicyManagerBase(clientInterface) {
+ mHdmiAudioDisabled = false;
+ mHdmiAudioEvent = false; }
- // AudioPolicyInterface
+ virtual ~AudioPolicyManager() {}
+
virtual status_t setDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
+ AudioSystem::device_connection_state state,
const char *device_address);
- virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
- const char *device_address);
- virtual void setPhoneState(audio_mode_t state);
- virtual void setForceUse(audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config);
- virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
- virtual void setSystemProperty(const char* property, const char* value);
- virtual status_t initCheck();
- virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+ virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+ virtual audio_io_handle_t getInput(int inputSource,
uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo);
- virtual status_t startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- int session = 0);
- virtual status_t stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- int session = 0);
- virtual void releaseOutput(audio_io_handle_t output);
- virtual audio_io_handle_t getInput(audio_source_t inputSource,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_in_acoustics_t acoustics);
-
- // indicates to the audio policy manager that the input starts being used.
- virtual status_t startInput(audio_io_handle_t input);
-
- // indicates to the audio policy manager that the input stops being used.
- virtual status_t stopInput(audio_io_handle_t input);
- virtual void releaseInput(audio_io_handle_t input);
- virtual void closeAllInputs();
- virtual void initStreamVolume(audio_stream_type_t stream,
- int indexMin,
- int indexMax);
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
- int index,
- audio_devices_t device);
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
- int *index,
- audio_devices_t device);
-
- // return the strategy corresponding to a given stream type
- virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
-
- // return the enabled output devices for the given stream type
- virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
-
- virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
- virtual status_t registerEffect(const effect_descriptor_t *desc,
- audio_io_handle_t io,
- uint32_t strategy,
- int session,
- int id);
- virtual status_t unregisterEffect(int id);
- virtual status_t setEffectEnabled(int id, bool enabled);
-
- virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
- // return whether a stream is playing remotely, override to change the definition of
- // local/remote playback, used for instance by notification manager to not make
- // media players lose audio focus when not playing locally
- virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
- virtual bool isSourceActive(audio_source_t source) const;
-
- virtual status_t dump(int fd);
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::audio_in_acoustics acoustics);
+ virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate = 0,
+ uint32_t format = AudioSystem::FORMAT_DEFAULT,
+ uint32_t channels = 0,
+ AudioSystem::output_flags flags =
+ AudioSystem::OUTPUT_FLAG_INDIRECT,
+ const audio_offload_info_t *offloadInfo = NULL);
virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
- virtual status_t listAudioPorts(audio_port_role_t role,
- audio_port_type_t type,
- unsigned int *num_ports,
- struct audio_port *ports,
- unsigned int *generation);
- virtual status_t getAudioPort(struct audio_port *port);
- virtual status_t createAudioPatch(const struct audio_patch *patch,
- audio_patch_handle_t *handle,
- uid_t uid);
- virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
- uid_t uid);
- virtual status_t listAudioPatches(unsigned int *num_patches,
- struct audio_patch *patches,
- unsigned int *generation);
- virtual status_t setAudioPortConfig(const struct audio_port_config *config);
- virtual void clearAudioPatches(uid_t uid);
+ virtual void setPhoneState(int state);
+ // true if given state represents a device in a telephony or VoIP call
+ virtual bool isStateInCall(int state);
protected:
-
- enum routing_strategy {
- STRATEGY_MEDIA,
- STRATEGY_PHONE,
- STRATEGY_SONIFICATION,
- STRATEGY_SONIFICATION_RESPECTFUL,
- STRATEGY_DTMF,
- STRATEGY_ENFORCED_AUDIBLE,
- NUM_STRATEGIES
- };
-
- // 4 points to define the volume attenuation curve, each characterized by the volume
- // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
- // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
-
- enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
-
- class VolumeCurvePoint
- {
- public:
- int mIndex;
- float mDBAttenuation;
- };
-
- // device categories used for volume curve management.
- enum device_category {
- DEVICE_CATEGORY_HEADSET,
- DEVICE_CATEGORY_SPEAKER,
- DEVICE_CATEGORY_EARPIECE,
- DEVICE_CATEGORY_CNT
- };
-
- class HwModule;
-
- class AudioGain: public RefBase
- {
- public:
- AudioGain();
- virtual ~AudioGain() {}
-
- void dump(int fd, int spaces, int index) const;
-
- struct audio_gain mGain;
- };
-
- class AudioPort: public RefBase
- {
- public:
- AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, HwModule *module) :
- mName(name), mType(type), mRole(role), mModule(module) {}
- virtual ~AudioPort() {}
-
- virtual void toAudioPort(struct audio_port *port) const;
-
- void loadSamplingRates(char *name);
- void loadFormats(char *name);
- void loadOutChannels(char *name);
- void loadInChannels(char *name);
-
- audio_gain_mode_t loadGainMode(char *name);
- void loadGain(cnode *root);
- void loadGains(cnode *root);
-
- void dump(int fd, int spaces) const;
-
- String8 mName;
- audio_port_type_t mType;
- audio_port_role_t mRole;
- // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
- // indicates the supported parameters should be read from the output stream
- // after it is opened for the first time
- Vector <uint32_t> mSamplingRates; // supported sampling rates
- Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
- Vector <audio_format_t> mFormats; // supported audio formats
- Vector < sp<AudioGain> > mGains; // gain controllers
- HwModule *mModule; // audio HW module exposing this I/O stream
- };
-
- class AudioPatch: public RefBase
- {
- public:
- AudioPatch(audio_patch_handle_t handle,
- const struct audio_patch *patch, uid_t uid) :
- mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) {}
-
- audio_patch_handle_t mHandle;
- struct audio_patch mPatch;
- uid_t mUid;
- audio_patch_handle_t mAfPatchHandle;
- };
-
- class DeviceDescriptor: public AudioPort
- {
- public:
- DeviceDescriptor(const String8& name, audio_devices_t type, String8 address,
- audio_channel_mask_t channelMask) :
- AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
- audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
- AUDIO_PORT_ROLE_SOURCE,
- NULL),
- mDeviceType(type), mAddress(address),
- mChannelMask(channelMask), mId(0) {}
-
- DeviceDescriptor(String8 name, audio_devices_t type) :
- AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
- audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
- AUDIO_PORT_ROLE_SOURCE,
- NULL),
- mDeviceType(type), mAddress(""),
- mChannelMask(0), mId(0) {}
- virtual ~DeviceDescriptor() {}
-
- bool equals(const sp<DeviceDescriptor>& other) const;
- void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
-
- virtual void toAudioPort(struct audio_port *port) const;
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_devices_t mDeviceType;
- String8 mAddress;
- audio_channel_mask_t mChannelMask;
- audio_port_handle_t mId;
- };
-
- class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
- {
- public:
- DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
-
- ssize_t add(const sp<DeviceDescriptor>& item);
- ssize_t remove(const sp<DeviceDescriptor>& item);
- ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
-
- audio_devices_t types() const { return mDeviceTypes; }
-
- void loadDevicesFromType(audio_devices_t types);
- void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
-
- sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
- DeviceVector getDevicesFromType(audio_devices_t types) const;
- sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
- sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
-
- private:
- void refreshTypes();
- audio_devices_t mDeviceTypes;
- };
-
- // the IOProfile class describes the capabilities of an output or input stream.
- // It is currently assumed that all combination of listed parameters are supported.
- // It is used by the policy manager to determine if an output or input is suitable for
- // a given use case, open/close it accordingly and connect/disconnect audio tracks
- // to/from it.
- class IOProfile : public AudioPort
- {
- public:
- IOProfile(const String8& name, audio_port_role_t role, HwModule *module);
- virtual ~IOProfile();
-
- bool isCompatibleProfile(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags) const;
-
- void dump(int fd);
- void log();
-
- DeviceVector mSupportedDevices; // supported devices
- // (devices this output can be routed to)
- audio_output_flags_t mFlags; // attribute flags (e.g primary output,
- // direct output...). For outputs only.
- };
-
- class HwModule {
- public:
- HwModule(const char *name);
- ~HwModule();
-
- status_t loadOutput(cnode *root);
- status_t loadInput(cnode *root);
- status_t loadDevice(cnode *root);
-
- void dump(int fd);
-
- const char *const mName; // base name of the audio HW module (primary, a2dp ...)
- audio_module_handle_t mHandle;
- Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
- Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module
- DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf
-
- };
-
- // default volume curve
- static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT];
- // default volume curve for media strategy
- static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT];
- // volume curve for media strategy on speakers
- static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT];
- // volume curve for sonification strategy on speakers
- static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
- // default volume curves per stream and device category. See initializeVolumeCurves()
- static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT];
-
- // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
- // and keep track of the usage of this output by each audio stream type.
- class AudioOutputDescriptor
- {
- public:
- AudioOutputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_devices_t device() const;
- void changeRefCount(audio_stream_type_t stream, int delta);
-
- bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
- audio_devices_t supportedDevices();
- uint32_t latency();
- bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc);
- bool isActive(uint32_t inPastMs = 0) const;
- bool isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs = 0,
- nsecs_t sysTime = 0) const;
- bool isStrategyActive(routing_strategy strategy,
- uint32_t inPastMs = 0,
- nsecs_t sysTime = 0) const;
-
- void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- void toAudioPort(struct audio_port *port) const;
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // output handle
- uint32_t mSamplingRate; //
- audio_format_t mFormat; //
- audio_channel_mask_t mChannelMask; // output configuration
- uint32_t mLatency; //
- audio_output_flags_t mFlags; //
- audio_devices_t mDevice; // current device this output is routed to
- audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
- nsecs_t mStopTime[AUDIO_STREAM_CNT];
- AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output
- AudioOutputDescriptor *mOutput2; // used by duplicated outputs: second output
- float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
- int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
- const sp<IOProfile> mProfile; // I/O profile this output derives from
- bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
- // device selection. See checkDeviceMuteStrategies()
- uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
- };
-
- // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
- // and keep track of the usage of this input.
- class AudioInputDescriptor
- {
- public:
- AudioInputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // input handle
- uint32_t mSamplingRate; //
- audio_format_t mFormat; // input configuration
- audio_channel_mask_t mChannelMask; //
- audio_devices_t mDevice; // current device this input is routed to
- audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount; // number of AudioRecord clients using this output
- audio_source_t mInputSource; // input source selected by application (mediarecorder.h)
- const sp<IOProfile> mProfile; // I/O profile this output derives from
-
- void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- void toAudioPort(struct audio_port *port) const;
- };
-
- // stream descriptor used for volume control
- class StreamDescriptor
- {
- public:
- StreamDescriptor();
-
- int getVolumeIndex(audio_devices_t device);
- void dump(int fd);
-
- int mIndexMin; // min volume index
- int mIndexMax; // max volume index
- KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device
- bool mCanBeMuted; // true is the stream can be muted
-
- const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
- };
-
- // stream descriptor used for volume control
- class EffectDescriptor
- {
- public:
-
- status_t dump(int fd);
-
- int mIo; // io the effect is attached to
- routing_strategy mStrategy; // routing strategy the effect is associated to
- int mSession; // audio session the effect is on
- effect_descriptor_t mDesc; // effect descriptor
- bool mEnabled; // enabled state: CPU load being used or not
- };
-
- void addOutput(audio_io_handle_t output, AudioOutputDescriptor *outputDesc);
- void addInput(audio_io_handle_t input, AudioInputDescriptor *inputDesc);
-
// return the strategy corresponding to a given stream type
- static routing_strategy getStrategy(audio_stream_type_t stream);
+ static routing_strategy getStrategy(AudioSystem::stream_type stream);
// return appropriate device for streams handled by the specified strategy according to current
// phone state, connected devices...
@@ -497,258 +80,32 @@
// where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
// before updateDevicesAndOutputs() is called.
virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
- bool fromCache);
-
- // change the route of the specified output. Returns the number of ms we have slept to
- // allow new routing to take effect in certain cases.
- uint32_t setOutputDevice(audio_io_handle_t output,
- audio_devices_t device,
- bool force = false,
- int delayMs = 0,
- audio_patch_handle_t *patchHandle = NULL);
- status_t resetOutputDevice(audio_io_handle_t output,
- int delayMs = 0,
- audio_patch_handle_t *patchHandle = NULL);
- status_t setInputDevice(audio_io_handle_t input,
- audio_devices_t device,
- bool force = false,
- audio_patch_handle_t *patchHandle = NULL);
- status_t resetInputDevice(audio_io_handle_t input,
- audio_patch_handle_t *patchHandle = NULL);
-
+ bool fromCache = true);
// select input device corresponding to requested audio source
- virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
-
- // return io handle of active input or 0 if no input is active
- // Only considers inputs from physical devices (e.g. main mic, headset mic) when
- // ignoreVirtualInputs is true.
- audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
-
- // initialize volume curves for each strategy and device category
- void initializeVolumeCurves();
+ virtual audio_devices_t getDeviceForInputSource(int inputSource);
// compute the actual volume for a given stream according to the requested index and a particular
// device
- virtual float computeVolume(audio_stream_type_t stream, int index,
- audio_io_handle_t output, audio_devices_t device);
+ virtual float computeVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device);
// check that volume change is permitted, compute and send new volume to audio hardware
- status_t checkAndSetVolume(audio_stream_type_t stream, int index, audio_io_handle_t output,
- audio_devices_t device, int delayMs = 0, bool force = false);
-
- // apply all stream volumes to the specified output and device
- void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
-
- // Mute or unmute all streams handled by the specified strategy on the specified output
- void setStrategyMute(routing_strategy strategy,
- bool on,
- audio_io_handle_t output,
- int delayMs = 0,
- audio_devices_t device = (audio_devices_t)0);
-
- // Mute or unmute the stream on the specified output
- void setStreamMute(audio_stream_type_t stream,
- bool on,
- audio_io_handle_t output,
- int delayMs = 0,
- audio_devices_t device = (audio_devices_t)0);
-
- // handle special cases for sonification strategy while in call: mute streams or replace by
- // a special tone in the device used for communication
- void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
-
- // true if device is in a telephony or VoIP call
- virtual bool isInCall();
-
- // true if given state represents a device in a telephony or VoIP call
- virtual bool isStateInCall(int state);
-
- // when a device is connected, checks if an open output can be routed
- // to this device. If none is open, tries to open one of the available outputs.
- // Returns an output suitable to this device or 0.
- // when a device is disconnected, checks if an output is not used any more and
- // returns its handle if any.
- // transfers the audio tracks and effects from one output thread to another accordingly.
- status_t checkOutputsForDevice(audio_devices_t device,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& outputs,
- const String8 address);
-
- status_t checkInputsForDevice(audio_devices_t device,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& inputs,
- const String8 address);
-
- // close an output and its companion duplicating output.
- void closeOutput(audio_io_handle_t output);
-
- // checks and if necessary changes outputs used for all strategies.
- // must be called every time a condition that affects the output choice for a given strategy
- // changes: connected device, phone state, force use...
- // Must be called before updateDevicesAndOutputs()
- void checkOutputForStrategy(routing_strategy strategy);
-
- // Same as checkOutputForStrategy() but for a all strategies in order of priority
- void checkOutputForAllStrategies();
-
- // manages A2DP output suspend/restore according to phone state and BT SCO usage
- void checkA2dpSuspend();
-
- // returns the A2DP output handle if it is open or 0 otherwise
- audio_io_handle_t getA2dpOutput();
-
- // selects the most appropriate device on output for current state
- // must be called every time a condition that affects the device choice for a given output is
- // changed: connected device, phone state, force use, output start, output stop..
- // see getDeviceForStrategy() for the use of fromCache parameter
- audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
-
- // updates cache of device used by all strategies (mDeviceForStrategy[])
- // must be called every time a condition that affects the device choice for a given strategy is
- // changed: connected device, phone state, force use...
- // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
- // Must be called after checkOutputForAllStrategies()
- void updateDevicesAndOutputs();
-
- // selects the most appropriate device on input for current state
- audio_devices_t getNewInputDevice(audio_io_handle_t input);
-
- virtual uint32_t getMaxEffectsCpuLoad();
- virtual uint32_t getMaxEffectsMemory();
-#ifdef AUDIO_POLICY_TEST
- virtual bool threadLoop();
- void exit();
- int testOutputIndex(audio_io_handle_t output);
-#endif //AUDIO_POLICY_TEST
-
- status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled);
+ status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
// returns the category the device belongs to with regard to volume curve management
static device_category getDeviceCategory(audio_devices_t device);
- // extract one device relevant for volume control from multiple device selection
- static audio_devices_t getDeviceForVolume(audio_devices_t device);
+ static const char* HDMI_SPKR_STR;
- SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs);
- bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
- SortedVector<audio_io_handle_t>& outputs2);
+ //parameter indicates of HDMI speakers disabled from the Qualcomm settings
+ bool mHdmiAudioDisabled;
- // mute/unmute strategies using an incompatible device combination
- // if muting, wait for the audio in pcm buffer to be drained before proceeding
- // if unmuting, unmute only after the specified delay
- // Returns the number of ms waited
- uint32_t checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
- audio_devices_t prevDevice,
- uint32_t delayMs);
-
- audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
- audio_output_flags_t flags);
- sp<IOProfile> getInputProfile(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask);
- sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags);
-
- audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
-
- bool isNonOffloadableEffectEnabled();
-
- status_t addAudioPatch(audio_patch_handle_t handle,
- const sp<AudioPatch>& patch);
- status_t removeAudioPatch(audio_patch_handle_t handle);
-
- AudioOutputDescriptor *getOutputFromId(audio_port_handle_t id) const;
- AudioInputDescriptor *getInputFromId(audio_port_handle_t id) const;
- HwModule *getModuleForDevice(audio_devices_t device) const;
- HwModule *getModuleFromName(const char *name) const;
- //
- // Audio policy configuration file parsing (audio_policy.conf)
- //
- static uint32_t stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name);
- static const char *enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value);
- static bool stringToBool(const char *value);
- static audio_output_flags_t parseFlagNames(char *name);
- static audio_devices_t parseDeviceNames(char *name);
- void loadHwModule(cnode *root);
- void loadHwModules(cnode *root);
- void loadGlobalConfig(cnode *root, HwModule *module);
- status_t loadAudioPolicyConfig(const char *path);
- void defaultAudioPolicyConfig(void);
-
-
- uid_t mUidCached;
- AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
- audio_io_handle_t mPrimaryOutput; // primary output handle
- // list of descriptors for outputs currently opened
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;
- // copy of mOutputs before setDeviceConnectionState() opens new outputs
- // reset to mOutputs when updateDevicesAndOutputs() is called.
- DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs;
- DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors
- DeviceVector mAvailableOutputDevices; // all available output devices
- DeviceVector mAvailableInputDevices; // all available input devices
- int mPhoneState; // current phone state
- audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration
-
- StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control
- bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
- audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
- float mLastVoiceVolume; // last voice volume value sent to audio HAL
-
- // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
- static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
- // Maximum memory allocated to audio effects in KB
- static const uint32_t MAX_EFFECTS_MEMORY = 512;
- uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
- uint32_t mTotalEffectsMemory; // current memory used by effects
- KeyedVector<int, EffectDescriptor *> mEffects; // list of registered audio effects
- bool mA2dpSuspended; // true if A2DP output is suspended
- sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
- bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
- // to boost soft sounds, used to adjust volume curves accordingly
-
- Vector <HwModule *> mHwModules;
- volatile int32_t mNextUniqueId;
- volatile int32_t mAudioPortGeneration;
-
- DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
-
-#ifdef AUDIO_POLICY_TEST
- Mutex mLock;
- Condition mWaitWorkCV;
-
- int mCurOutput;
- bool mDirectOutput;
- audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
- int mTestInput;
- uint32_t mTestDevice;
- uint32_t mTestSamplingRate;
- uint32_t mTestFormat;
- uint32_t mTestChannels;
- uint32_t mTestLatencyMs;
-#endif //AUDIO_POLICY_TEST
+ //parameter indicates if HDMI plug in/out detected
+ bool mHdmiAudioEvent;
private:
- static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi);
- // updates device caching and output for streams that can influence the
- // routing of notifications
- void handleNotificationRoutingForStream(audio_stream_type_t stream);
- static bool isVirtualInputDevice(audio_devices_t device);
- uint32_t nextUniqueId();
- uint32_t nextAudioPortGeneration();
- uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
- // converts device address to string sent to audio HAL via setParameters
- static String8 addressToParameter(audio_devices_t device, const String8 address);
-};
+ // Used for voip + voice concurrency usecase
+ int mPrevPhoneState;
+ static int mvoice_call_state;
};
+};