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-rw-r--r--include/media/AudioSystem.h2
-rw-r--r--libs/audioflinger/Android.mk11
-rw-r--r--libs/audioflinger/AudioPolicyManagerBase.cpp1925
-rw-r--r--libs/audioflinger/AudioPolicyManagerGeneric.cpp945
-rw-r--r--libs/audioflinger/AudioPolicyManagerGeneric.h196
-rw-r--r--libs/audioflinger/AudioPolicyService.cpp45
-rw-r--r--libs/audioflinger/AudioPolicyService.h5
7 files changed, 1964 insertions, 1165 deletions
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index c87007c..f935bb9 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -244,6 +244,8 @@ public:
DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_DEFAULT),
+ DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+ DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
// input devices
DEVICE_IN_COMMUNICATION = 0x10000,
diff --git a/libs/audioflinger/Android.mk b/libs/audioflinger/Android.mk
index f5c03bb..b68bfc1 100644
--- a/libs/audioflinger/Android.mk
+++ b/libs/audioflinger/Android.mk
@@ -47,7 +47,7 @@ include $(BUILD_STATIC_LIBRARY)
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyManagerGeneric.cpp
+ AudioPolicyManagerBase.cpp
LOCAL_SHARED_LIBRARIES := \
libcutils \
@@ -60,7 +60,7 @@ else
LOCAL_SHARED_LIBRARIES += libdl
endif
-LOCAL_MODULE:= libaudiopolicygeneric
+LOCAL_MODULE:= libaudiopolicybase
ifeq ($(BOARD_HAVE_BLUETOOTH),true)
LOCAL_CFLAGS += -DWITH_A2DP
@@ -70,7 +70,7 @@ ifeq ($(AUDIO_POLICY_TEST),true)
LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
endif
-include $(BUILD_SHARED_LIBRARY)
+include $(BUILD_STATIC_LIBRARY)
include $(CLEAR_VARS)
@@ -87,11 +87,10 @@ LOCAL_SHARED_LIBRARIES := \
libutils \
libbinder \
libmedia \
- libhardware_legacy \
- libaudiopolicygeneric
+ libhardware_legacy
ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
- LOCAL_STATIC_LIBRARIES += libaudiointerface
+ LOCAL_STATIC_LIBRARIES += libaudiointerface libaudiopolicybase
LOCAL_CFLAGS += -DGENERIC_AUDIO
else
LOCAL_SHARED_LIBRARIES += libaudio libaudiopolicy
diff --git a/libs/audioflinger/AudioPolicyManagerBase.cpp b/libs/audioflinger/AudioPolicyManagerBase.cpp
new file mode 100644
index 0000000..055dbca
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyManagerBase.cpp
@@ -0,0 +1,1925 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyManagerBase"
+//
+#define LOG_NDEBUG 0
+#include <utils/Log.h>
+#include <hardware_legacy/AudioPolicyManagerBase.h>
+#include <media/mediarecorder.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+
+status_t AudioPolicyManagerBase::setDeviceConnectionState(AudioSystem::audio_devices device,
+ AudioSystem::device_connection_state state,
+ const char *device_address)
+{
+
+ LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
+
+ // connect/disconnect only 1 device at a time
+ if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
+
+ if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
+ LOGE("setDeviceConnectionState() invalid address: %s", device_address);
+ return BAD_VALUE;
+ }
+
+ // handle output devices
+ if (AudioSystem::isOutputDevice(device)) {
+
+#ifndef WITH_A2DP
+ if (AudioSystem::isA2dpDevice(device)) {
+ LOGE("setDeviceConnectionState() invalid device: %x", device);
+ return BAD_VALUE;
+ }
+#endif
+
+ switch (state)
+ {
+ // handle output device connection
+ case AudioSystem::DEVICE_STATE_AVAILABLE:
+ if (mAvailableOutputDevices & device) {
+ LOGW("setDeviceConnectionState() device already connected: %x", device);
+ return INVALID_OPERATION;
+ }
+ LOGV("setDeviceConnectionState() connecting device %x", device);
+
+ // register new device as available
+ mAvailableOutputDevices |= device;
+
+#ifdef WITH_A2DP
+ // handle A2DP device connection
+ if (AudioSystem::isA2dpDevice(device)) {
+ status_t status = handleA2dpConnection(device, device_address);
+ if (status != NO_ERROR) {
+ mAvailableOutputDevices &= ~device;
+ return status;
+ }
+ } else
+#endif
+ {
+ if (AudioSystem::isBluetoothScoDevice(device)) {
+ LOGV("setDeviceConnectionState() BT SCO device, address %s", device_address);
+ // keep track of SCO device address
+ mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+#ifdef WITH_A2DP
+ if ((mA2dpDeviceAddress == mScoDeviceAddress) &&
+ (mPhoneState != AudioSystem::MODE_NORMAL)) {
+ mpClientInterface->suspendOutput(mA2dpOutput);
+ }
+#endif
+ }
+ }
+ break;
+ // handle output device disconnection
+ case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+ if (!(mAvailableOutputDevices & device)) {
+ LOGW("setDeviceConnectionState() device not connected: %x", device);
+ return INVALID_OPERATION;
+ }
+
+
+ LOGV("setDeviceConnectionState() disconnecting device %x", device);
+ // remove device from available output devices
+ mAvailableOutputDevices &= ~device;
+
+#ifdef WITH_A2DP
+ // handle A2DP device disconnection
+ if (AudioSystem::isA2dpDevice(device)) {
+ status_t status = handleA2dpDisconnection(device, device_address);
+ if (status != NO_ERROR) {
+ mAvailableOutputDevices |= device;
+ return status;
+ }
+ } else
+#endif
+ {
+ if (AudioSystem::isBluetoothScoDevice(device)) {
+ mScoDeviceAddress = "";
+#ifdef WITH_A2DP
+ if ((mA2dpDeviceAddress == mScoDeviceAddress) &&
+ (mPhoneState != AudioSystem::MODE_NORMAL)) {
+ mpClientInterface->restoreOutput(mA2dpOutput);
+ }
+#endif
+ }
+ }
+ } break;
+
+ default:
+ LOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ // request routing change if necessary
+ uint32_t newDevice = getNewDevice(mHardwareOutput, false);
+#ifdef WITH_A2DP
+ checkOutputForAllStrategies(newDevice);
+ // A2DP outputs must be closed after checkOutputForAllStrategies() is executed
+ if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE && AudioSystem::isA2dpDevice(device)) {
+ closeA2dpOutputs();
+ }
+#endif
+ updateDeviceForStrategy();
+ setOutputDevice(mHardwareOutput, newDevice);
+
+ if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) {
+ device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
+ } else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO ||
+ device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
+ device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
+ device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else {
+ return NO_ERROR;
+ }
+ }
+ // handle input devices
+ if (AudioSystem::isInputDevice(device)) {
+
+ switch (state)
+ {
+ // handle input device connection
+ case AudioSystem::DEVICE_STATE_AVAILABLE: {
+ if (mAvailableInputDevices & device) {
+ LOGW("setDeviceConnectionState() device already connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ mAvailableInputDevices |= device;
+ }
+ break;
+
+ // handle input device disconnection
+ case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+ if (!(mAvailableInputDevices & device)) {
+ LOGW("setDeviceConnectionState() device not connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ mAvailableInputDevices &= ~device;
+ } break;
+
+ default:
+ LOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+ uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+ if (newDevice != inputDesc->mDevice) {
+ LOGV("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());
+ }
+ }
+
+ return NO_ERROR;
+ }
+
+ LOGW("setDeviceConnectionState() invalid device: %x", device);
+ return BAD_VALUE;
+}
+
+AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(AudioSystem::audio_devices device,
+ const char *device_address)
+{
+ AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
+ String8 address = String8(device_address);
+ if (AudioSystem::isOutputDevice(device)) {
+ if (device & mAvailableOutputDevices) {
+#ifdef WITH_A2DP
+ if (AudioSystem::isA2dpDevice(device) &&
+ address != "" && mA2dpDeviceAddress != address) {
+ return state;
+ }
+#endif
+ if (AudioSystem::isBluetoothScoDevice(device) &&
+ address != "" && mScoDeviceAddress != address) {
+ return state;
+ }
+ state = AudioSystem::DEVICE_STATE_AVAILABLE;
+ }
+ } else if (AudioSystem::isInputDevice(device)) {
+ if (device & mAvailableInputDevices) {
+ state = AudioSystem::DEVICE_STATE_AVAILABLE;
+ }
+ }
+
+ return state;
+}
+
+void AudioPolicyManagerBase::setPhoneState(int state)
+{
+ LOGV("setPhoneState() state %d", state);
+ uint32_t newDevice = 0;
+ if (state < 0 || state >= AudioSystem::NUM_MODES) {
+ LOGW("setPhoneState() invalid state %d", state);
+ return;
+ }
+
+ if (state == mPhoneState ) {
+ LOGW("setPhoneState() setting same state %d", state);
+ return;
+ }
+
+ // if leaving call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ LOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ handleIncallSonification(stream, false, true);
+ }
+ }
+
+ // store previous phone state for management of sonification strategy below
+ int oldState = mPhoneState;
+ mPhoneState = state;
+ bool force = false;
+
+ // are we entering or starting a call
+ if ((oldState != AudioSystem::MODE_IN_CALL) && (state == AudioSystem::MODE_IN_CALL)) {
+ LOGV(" Entering call in setPhoneState()");
+ // force routing command to audio hardware when starting a call
+ // even if no device change is needed
+ force = true;
+ } else if ((oldState == AudioSystem::MODE_IN_CALL) && (state != AudioSystem::MODE_IN_CALL)) {
+ LOGV(" Exiting call in setPhoneState()");
+ // force routing command to audio hardware when exiting a call
+ // even if no device change is needed
+ force = true;
+ }
+
+ // check for device and output changes triggered by new phone state
+ newDevice = getNewDevice(mHardwareOutput, false);
+#ifdef WITH_A2DP
+ checkOutputForAllStrategies(newDevice);
+ // suspend A2DP output if SCO device address is the same as A2DP device address.
+ // no need to check that a SCO device is actually connected as mScoDeviceAddress == ""
+ // if none is connected and the test below will fail.
+ if (mA2dpDeviceAddress == mScoDeviceAddress) {
+ if (oldState == AudioSystem::MODE_NORMAL) {
+ mpClientInterface->suspendOutput(mA2dpOutput);
+ } else if (state == AudioSystem::MODE_NORMAL) {
+ mpClientInterface->restoreOutput(mA2dpOutput);
+ }
+ }
+#endif
+ updateDeviceForStrategy();
+
+ AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (oldState == AudioSystem::MODE_IN_CALL && newDevice == 0) {
+ newDevice = hwOutputDesc->device();
+ }
+ // change routing is necessary
+ setOutputDevice(mHardwareOutput, newDevice, force);
+
+ // if entering in call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (state == AudioSystem::MODE_IN_CALL) {
+ LOGV("setPhoneState() in call state management: new state is %d", state);
+ 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 == AudioSystem::MODE_RINGTONE &&
+ (hwOutputDesc->mRefCount[AudioSystem::MUSIC] ||
+ (systemTime() - mMusicStopTime) < seconds(SONIFICATION_HEADSET_MUSIC_DELAY))) {
+ mLimitRingtoneVolume = true;
+ } else {
+ mLimitRingtoneVolume = false;
+ }
+}
+
+void AudioPolicyManagerBase::setRingerMode(uint32_t mode, uint32_t mask)
+{
+ LOGV("setRingerMode() mode %x, mask %x", mode, mask);
+
+ mRingerMode = mode;
+}
+
+void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+{
+ LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+
+ switch(usage) {
+ case AudioSystem::FOR_COMMUNICATION:
+ if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
+ config != AudioSystem::FORCE_NONE) {
+ LOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AudioSystem::FOR_MEDIA:
+ if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
+ config != AudioSystem::FORCE_WIRED_ACCESSORY && config != AudioSystem::FORCE_NONE) {
+ LOGW("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) {
+ LOGW("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) {
+ LOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+ }
+ mForceUse[usage] = config;
+ break;
+ default:
+ LOGW("setForceUse() invalid usage %d", usage);
+ break;
+ }
+
+ // check for device and output changes triggered by new phone state
+ uint32_t newDevice = getNewDevice(mHardwareOutput, false);
+#ifdef WITH_A2DP
+ checkOutputForAllStrategies(newDevice);
+#endif
+ updateDeviceForStrategy();
+ setOutputDevice(mHardwareOutput, newDevice);
+}
+
+AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
+{
+ return mForceUse[usage];
+}
+
+void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
+{
+ LOGV("setSystemProperty() property %s, value %s", property, value);
+ if (strcmp(property, "ro.camera.sound.forced") == 0) {
+ if (atoi(value)) {
+ LOGV("ENFORCED_AUDIBLE cannot be muted");
+ mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
+ } else {
+ LOGV("ENFORCED_AUDIBLE can be muted");
+ mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
+ }
+ }
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::output_flags flags)
+{
+ audio_io_handle_t output = 0;
+ uint32_t latency = 0;
+ routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+ uint32_t device = getDeviceForStrategy(strategy);
+ LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ LOGV("getOutput() opening test output");
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mSamplingRate = mTestSamplingRate;
+ outputDesc->mFormat = mTestFormat;
+ outputDesc->mChannels = mTestChannels;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannels,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ 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:
+ // 1 a direct output is explicitely requested
+ // 2 the audio format is compressed
+ if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+ (format !=0 && !AudioSystem::isLinearPCM(format))) {
+
+ LOGV("getOutput() opening direct output device %x", device);
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+ outputDesc->mDevice = device;
+ outputDesc->mSamplingRate = samplingRate;
+ outputDesc->mFormat = format;
+ outputDesc->mChannels = channels;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT);
+ outputDesc->mRefCount[stream] = 1;
+ output = mpClientInterface->openOutput(&outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannels,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ // only accept an output with the requeted parameters
+ if (output == 0 ||
+ (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+ (format != 0 && format != outputDesc->mFormat) ||
+ (channels != 0 && channels != outputDesc->mChannels)) {
+ LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d",
+ samplingRate, format, channels);
+ if (output != 0) {
+ mpClientInterface->closeOutput(output);
+ }
+ delete outputDesc;
+ return 0;
+ }
+ addOutput(output, outputDesc);
+ return output;
+ }
+
+ if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO &&
+ channels != AudioSystem::CHANNEL_OUT_STEREO) {
+ return 0;
+ }
+ // open a non direct output
+
+ // get which output is suitable for the specified stream. The actual routing change will happen
+ // when startOutput() will be called
+ uint32_t a2dpDevice = device & AudioSystem::DEVICE_OUT_ALL_A2DP;
+ if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) {
+#ifdef WITH_A2DP
+ if (a2dpUsedForSonification() && a2dpDevice != 0) {
+ // if playing on 2 devices among which one is A2DP, use duplicated output
+ LOGV("getOutput() using duplicated output");
+ LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device);
+ output = mDuplicatedOutput;
+ } else
+#endif
+ {
+ // if playing on 2 devices among which none is A2DP, use hardware output
+ output = mHardwareOutput;
+ }
+ LOGV("getOutput() using output %d for 2 devices %x", output, device);
+ } else {
+#ifdef WITH_A2DP
+ if (a2dpDevice != 0) {
+ // if playing on A2DP device, use a2dp output
+ LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device);
+ output = mA2dpOutput;
+ } else
+#endif
+ {
+ // if playing on not A2DP device, use hardware output
+ output = mHardwareOutput;
+ }
+ }
+
+
+ LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
+ stream, samplingRate, format, channels, flags);
+
+ return output;
+}
+
+status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+ LOGV("startOutput() output %d, stream %d", output, stream);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ LOGW("startOutput() unknow output %d", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+
+#ifdef WITH_A2DP
+ if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) {
+ setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
+ }
+#endif
+
+ // incremenent 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
+ // necassary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->changeRefCount(stream, 1);
+
+ setOutputDevice(output, getNewDevice(output));
+
+ // handle special case for sonification while in call
+ if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ handleIncallSonification(stream, true, false);
+ }
+
+ // apply volume rules for current stream and device if necessary
+ checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device());
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+ LOGV("stopOutput() output %d, stream %d", output, stream);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ LOGW("stopOutput() unknow output %d", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+
+ // handle special case for sonification while in call
+ if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ 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 last music track was stopped - see computeVolume()
+ if (stream == AudioSystem::MUSIC) {
+ mMusicStopTime = systemTime();
+ }
+
+ setOutputDevice(output, getNewDevice(output));
+
+#ifdef WITH_A2DP
+ if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) {
+ setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput, mOutputs.valueFor(mHardwareOutput)->mLatency*2);
+ }
+#endif
+ return NO_ERROR;
+ } else {
+ LOGW("stopOutput() refcount is already 0 for output %d", output);
+ return INVALID_OPERATION;
+ }
+}
+
+void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
+{
+ LOGV("releaseOutput() %d", output);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ LOGW("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->refCount() == 0) {
+ mpClientInterface->closeOutput(output);
+ delete mOutputs.valueAt(index);
+ mOutputs.removeItem(output);
+ mTestOutputs[testIndex] = 0;
+ }
+ return;
+ }
+#endif //AUDIO_POLICY_TEST
+
+ if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
+ mpClientInterface->closeOutput(output);
+ delete mOutputs.valueAt(index);
+ mOutputs.removeItem(output);
+ }
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::audio_in_acoustics acoustics)
+{
+ audio_io_handle_t input = 0;
+ uint32_t device = getDeviceForInputSource(inputSource);
+
+ LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
+
+ if (device == 0) {
+ return 0;
+ }
+
+ // adapt channel selection to input source
+ switch(inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_CALL:
+ channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
+ break;
+ default:
+ break;
+ }
+
+ AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
+
+ inputDesc->mInputSource = inputSource;
+ inputDesc->mDevice = device;
+ inputDesc->mSamplingRate = samplingRate;
+ inputDesc->mFormat = format;
+ inputDesc->mChannels = channels;
+ inputDesc->mAcoustics = acoustics;
+ inputDesc->mRefCount = 0;
+ input = mpClientInterface->openInput(&inputDesc->mDevice,
+ &inputDesc->mSamplingRate,
+ &inputDesc->mFormat,
+ &inputDesc->mChannels,
+ inputDesc->mAcoustics);
+
+ // only accept input with the exact requested set of parameters
+ if (input == 0 ||
+ (samplingRate != inputDesc->mSamplingRate) ||
+ (format != inputDesc->mFormat) ||
+ (channels != inputDesc->mChannels)) {
+ LOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d",
+ samplingRate, format, channels);
+ if (input != 0) {
+ mpClientInterface->closeInput(input);
+ }
+ delete inputDesc;
+ return 0;
+ }
+ mInputs.add(input, inputDesc);
+ return input;
+}
+
+status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
+{
+ LOGV("startInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ LOGW("startInput() unknow 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
+ if (getActiveInput() != 0) {
+ LOGW("startInput() input %d failed: other input already started", input);
+ return INVALID_OPERATION;
+ }
+ }
+
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
+
+ // use Voice Recognition mode or not for this input based on input source
+ int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0;
+ param.addInt(String8("vr_mode"), vr_enabled);
+ LOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled);
+
+ mpClientInterface->setParameters(input, param.toString());
+
+ inputDesc->mRefCount = 1;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
+{
+ LOGV("stopInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ LOGW("stopInput() unknow input %d", input);
+ return BAD_VALUE;
+ }
+ AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+ if (inputDesc->mRefCount == 0) {
+ LOGW("stopInput() input %d already stopped", input);
+ return INVALID_OPERATION;
+ } else {
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), 0);
+ mpClientInterface->setParameters(input, param.toString());
+ inputDesc->mRefCount = 0;
+ return NO_ERROR;
+ }
+}
+
+void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input)
+{
+ LOGV("releaseInput() %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ LOGW("releaseInput() releasing unknown input %d", input);
+ return;
+ }
+ mpClientInterface->closeInput(input);
+ delete mInputs.valueAt(index);
+ mInputs.removeItem(input);
+ LOGV("releaseInput() exit");
+}
+
+void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream,
+ int indexMin,
+ int indexMax)
+{
+ LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+ if (indexMin < 0 || indexMin >= indexMax) {
+ LOGW("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 AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
+{
+
+ if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+ return BAD_VALUE;
+ }
+
+ // Force max volume if stream cannot be muted
+ if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
+
+ LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
+ mStreams[stream].mIndexCur = 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++) {
+ status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), mOutputs.valueAt(i)->device());
+ if (volStatus != NO_ERROR) {
+ status = volStatus;
+ }
+ }
+ return status;
+}
+
+status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
+{
+ if (index == 0) {
+ return BAD_VALUE;
+ }
+ LOGV("getStreamVolumeIndex() stream %d", stream);
+ *index = mStreams[stream].mIndexCur;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::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, " Hardware Output: %d\n", mHardwareOutput);
+ result.append(buffer);
+#ifdef WITH_A2DP
+ snprintf(buffer, SIZE, " A2DP Output: %d\n", mA2dpOutput);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Duplicated Output: %d\n", mDuplicatedOutput);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
+ result.append(buffer);
+#endif
+ snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Ringer mode: %d\n", mRingerMode);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ 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 Index Min Index Max Index Cur Can be muted\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+ snprintf(buffer, SIZE, " %02d", i);
+ mStreams[i].dump(buffer + 3, SIZE);
+ write(fd, buffer, strlen(buffer));
+ }
+
+ return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManagerBase
+// ----------------------------------------------------------------------------
+
+AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
+ :
+#ifdef AUDIO_POLICY_TEST
+ Thread(false),
+#endif //AUDIO_POLICY_TEST
+ mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0), mMusicStopTime(0), mLimitRingtoneVolume(false)
+{
+ mpClientInterface = clientInterface;
+
+ for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
+ mForceUse[i] = AudioSystem::FORCE_NONE;
+ }
+
+ // devices available by default are speaker, ear piece and microphone
+ mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE |
+ AudioSystem::DEVICE_OUT_SPEAKER;
+ mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+
+#ifdef WITH_A2DP
+ mA2dpOutput = 0;
+ mDuplicatedOutput = 0;
+ mA2dpDeviceAddress = String8("");
+#endif
+ mScoDeviceAddress = String8("");
+
+ // open hardware output
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+ outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
+ mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannels,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ if (mHardwareOutput == 0) {
+ LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
+ outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
+ } else {
+ addOutput(mHardwareOutput, outputDesc);
+ setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true);
+ }
+
+ updateDeviceForStrategy();
+#ifdef AUDIO_POLICY_TEST
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
+
+ mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
+ mTestSamplingRate = 44100;
+ mTestFormat = AudioSystem::PCM_16_BIT;
+ mTestChannels = AudioSystem::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
+}
+
+AudioPolicyManagerBase::~AudioPolicyManagerBase()
+{
+#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);
+ }
+ mOutputs.clear();
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ mpClientInterface->closeInput(mInputs.keyAt(i));
+ delete mInputs.valueAt(i);
+ }
+ mInputs.clear();
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManagerBase::threadLoop()
+{
+ LOGV("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) {
+ LOGV("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 = AudioSystem::INVALID_FORMAT;
+ if (value == "PCM 16 bits") {
+ format = AudioSystem::PCM_16_BIT;
+ } else if (value == "PCM 8 bits") {
+ format = AudioSystem::PCM_8_BIT;
+ } else if (value == "Compressed MP3") {
+ format = AudioSystem::MP3;
+ }
+ if (format != AudioSystem::INVALID_FORMAT) {
+ 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 = AudioSystem::CHANNEL_OUT_STEREO;
+ } else if (value == "Channels Mono") {
+ channels = AudioSystem::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"));
+
+ mpClientInterface->closeOutput(mHardwareOutput);
+ delete mOutputs.valueFor(mHardwareOutput);
+ mOutputs.removeItem(mHardwareOutput);
+
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+ outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
+ mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannels,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (mHardwareOutput == 0) {
+ LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
+ outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
+ } else {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
+ addOutput(mHardwareOutput, outputDesc);
+ }
+ }
+
+
+ mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+ }
+ }
+ return false;
+}
+
+void AudioPolicyManagerBase::exit()
+{
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+int AudioPolicyManagerBase::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 AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
+{
+ outputDesc->mId = id;
+ mOutputs.add(id, outputDesc);
+}
+
+
+#ifdef WITH_A2DP
+status_t AudioPolicyManagerBase::handleA2dpConnection(AudioSystem::audio_devices device,
+ const char *device_address)
+{
+ // when an A2DP device is connected, open an A2DP and a duplicated output
+ LOGV("opening A2DP output for device %s", device_address);
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+ outputDesc->mDevice = device;
+ mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannels,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (mA2dpOutput) {
+ // add A2DP output descriptor
+ addOutput(mA2dpOutput, outputDesc);
+ // set initial stream volume for A2DP device
+ applyStreamVolumes(mA2dpOutput, device);
+ if (a2dpUsedForSonification()) {
+ mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput);
+ }
+ if (mDuplicatedOutput != 0 ||
+ !a2dpUsedForSonification()) {
+ // If both A2DP and duplicated outputs are open, send device address to A2DP hardware
+ // interface
+ AudioParameter param;
+ param.add(String8("a2dp_sink_address"), String8(device_address));
+ mpClientInterface->setParameters(mA2dpOutput, param.toString());
+ mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+
+ if (a2dpUsedForSonification()) {
+ // add duplicated output descriptor
+ AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor();
+ dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput);
+ dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput);
+ dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate;
+ dupOutputDesc->mFormat = outputDesc->mFormat;
+ dupOutputDesc->mChannels = outputDesc->mChannels;
+ dupOutputDesc->mLatency = outputDesc->mLatency;
+ addOutput(mDuplicatedOutput, dupOutputDesc);
+ applyStreamVolumes(mDuplicatedOutput, device);
+ }
+ } else {
+ LOGW("getOutput() could not open duplicated output for %d and %d",
+ mHardwareOutput, mA2dpOutput);
+ mpClientInterface->closeOutput(mA2dpOutput);
+ mOutputs.removeItem(mA2dpOutput);
+ mA2dpOutput = 0;
+ delete outputDesc;
+ return NO_INIT;
+ }
+ } else {
+ LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device);
+ delete outputDesc;
+ return NO_INIT;
+ }
+ AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+
+ if (mA2dpDeviceAddress == mScoDeviceAddress) {
+ // It is normal to suspend twice if we are both in call,
+ // and have the hardware audio output routed to BT SCO
+ if (mPhoneState != AudioSystem::MODE_NORMAL) {
+ mpClientInterface->suspendOutput(mA2dpOutput);
+ }
+ if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)hwOutputDesc->device())) {
+ mpClientInterface->suspendOutput(mA2dpOutput);
+ }
+ }
+
+ if (!a2dpUsedForSonification()) {
+ // mute music on A2DP output if a notification or ringtone is playing
+ uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION);
+ for (uint32_t i = 0; i < refCount; i++) {
+ setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::handleA2dpDisconnection(AudioSystem::audio_devices device,
+ const char *device_address)
+{
+ if (mA2dpOutput == 0) {
+ LOGW("setDeviceConnectionState() disconnecting A2DP and no A2DP output!");
+ return INVALID_OPERATION;
+ }
+
+ if (mA2dpDeviceAddress != device_address) {
+ LOGW("setDeviceConnectionState() disconnecting unknow A2DP sink address %s", device_address);
+ return INVALID_OPERATION;
+ }
+
+ // mute media during 2 seconds to avoid outputing sound on hardware output while music stream
+ // is switched from A2DP output and before music is paused by music application
+ setStrategyMute(STRATEGY_MEDIA, true, mHardwareOutput);
+ setStrategyMute(STRATEGY_MEDIA, false, mHardwareOutput, 2000);
+
+ if (!a2dpUsedForSonification()) {
+ // unmute music on A2DP output if a notification or ringtone is playing
+ uint32_t refCount = mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_SONIFICATION);
+ for (uint32_t i = 0; i < refCount; i++) {
+ setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput);
+ }
+ }
+ mA2dpDeviceAddress = "";
+ return NO_ERROR;
+}
+
+void AudioPolicyManagerBase::closeA2dpOutputs()
+{
+ LOGV("setDeviceConnectionState() closing A2DP and duplicated output!");
+
+ if (mDuplicatedOutput != 0) {
+ mpClientInterface->closeOutput(mDuplicatedOutput);
+ delete mOutputs.valueFor(mDuplicatedOutput);
+ mOutputs.removeItem(mDuplicatedOutput);
+ mDuplicatedOutput = 0;
+ }
+ if (mA2dpOutput != 0) {
+ AudioParameter param;
+ param.add(String8("closing"), String8("true"));
+ mpClientInterface->setParameters(mA2dpOutput, param.toString());
+ mpClientInterface->closeOutput(mA2dpOutput);
+ delete mOutputs.valueFor(mA2dpOutput);
+ mOutputs.removeItem(mA2dpOutput);
+ mA2dpOutput = 0;
+ }
+}
+
+void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy, uint32_t &newDevice)
+{
+ uint32_t prevDevice = getDeviceForStrategy(strategy);
+ uint32_t curDevice = getDeviceForStrategy(strategy, false);
+ bool a2dpWasUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(prevDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
+ bool a2dpIsUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(curDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
+ AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+ AudioOutputDescriptor *a2dpOutputDesc;
+
+ if (a2dpWasUsed && !a2dpIsUsed) {
+ bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2);
+
+ if (dupUsed) {
+ LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy);
+ a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
+ } else {
+ LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy);
+ a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput);
+ }
+
+ for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ if (getStrategy((AudioSystem::stream_type)i) == strategy) {
+ mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
+ int refCount = a2dpOutputDesc->mRefCount[i];
+ // in the case of duplicated output, the ref count is first incremented
+ // and then decremented on hardware output tus keeping its value
+ hwOutputDesc->changeRefCount((AudioSystem::stream_type)i, refCount);
+ a2dpOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
+ }
+ }
+ // do not change newDevice is it was already set before this call by a previous call to
+ // getNewDevice() or checkOutputForStrategy() for a strategy with higher priority
+ if (newDevice == 0 && hwOutputDesc->isUsedByStrategy(strategy)) {
+ newDevice = getDeviceForStrategy(strategy, false);
+ }
+ }
+ if (a2dpIsUsed && !a2dpWasUsed) {
+ bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2);
+ audio_io_handle_t a2dpOutput;
+
+ if (dupUsed) {
+ LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy);
+ a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
+ a2dpOutput = mDuplicatedOutput;
+ } else {
+ LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy);
+ a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput);
+ a2dpOutput = mA2dpOutput;
+ }
+
+ for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ if (getStrategy((AudioSystem::stream_type)i) == strategy) {
+ mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, a2dpOutput);
+ int refCount = hwOutputDesc->mRefCount[i];
+ // in the case of duplicated output, the ref count is first incremented
+ // and then decremented on hardware output tus keeping its value
+ a2dpOutputDesc->changeRefCount((AudioSystem::stream_type)i, refCount);
+ hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
+ }
+ }
+ }
+}
+
+void AudioPolicyManagerBase::checkOutputForAllStrategies(uint32_t &newDevice)
+{
+ // Check strategies in order of priority so that once newDevice is set
+ // for a given strategy it is not modified by subsequent calls to
+ // checkOutputForStrategy()
+ checkOutputForStrategy(STRATEGY_PHONE, newDevice);
+ checkOutputForStrategy(STRATEGY_SONIFICATION, newDevice);
+ checkOutputForStrategy(STRATEGY_MEDIA, newDevice);
+ checkOutputForStrategy(STRATEGY_DTMF, newDevice);
+}
+
+#endif
+
+uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
+{
+ uint32_t device = 0;
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ // check the following by order of priority to request a routing change if necessary:
+ // 1: we are in call or the strategy phone is active on the hardware output:
+ // use device for strategy phone
+ // 2: the strategy sonification is active on the hardware output:
+ // use device for strategy sonification
+ // 3: the strategy media is active on the hardware output:
+ // use device for strategy media
+ // 4: the strategy DTMF is active on the hardware output:
+ // use device for strategy DTMF
+ if (mPhoneState == AudioSystem::MODE_IN_CALL ||
+ outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ } else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) {
+ device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ } else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) {
+ device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+ }
+
+ LOGV("getNewDevice() selected device %x", device);
+ return device;
+}
+
+AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::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::NOTIFICATION:
+ case AudioSystem::ALARM:
+ case AudioSystem::ENFORCED_AUDIBLE:
+ return STRATEGY_SONIFICATION;
+ case AudioSystem::DTMF:
+ return STRATEGY_DTMF;
+ default:
+ LOGE("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:
+ return STRATEGY_MEDIA;
+ }
+}
+
+uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, bool fromCache)
+{
+ uint32_t device = 0;
+
+ if (fromCache) {
+ LOGV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]);
+ return mDeviceForStrategy[strategy];
+ }
+
+ switch (strategy) {
+ case STRATEGY_DTMF:
+ if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ // when off call, DTMF strategy follows the same rules as MEDIA strategy
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false);
+ 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 (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AudioSystem::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
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
+ if (device) break;
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE;
+ if (device == 0) {
+ LOGE("getDeviceForStrategy() earpiece device not found");
+ }
+ break;
+
+ case AudioSystem::FORCE_SPEAKER:
+ if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+ if (device == 0) {
+ LOGE("getDeviceForStrategy() speaker device not found");
+ }
+ break;
+ }
+ break;
+
+ case STRATEGY_SONIFICATION:
+
+ // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+ // handleIncallSonification().
+ if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, false);
+ break;
+ }
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+ if (device == 0) {
+ LOGE("getDeviceForStrategy() speaker device not found");
+ }
+ // The second device used for sonification is the same as the device used by media strategy
+ // FALL THROUGH
+
+ case STRATEGY_MEDIA: {
+ uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
+#ifdef WITH_A2DP
+ if (mA2dpOutput != 0) {
+ if (strategy == STRATEGY_SONIFICATION && !a2dpUsedForSonification()) {
+ break;
+ }
+ if (device2 == 0) {
+ device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
+ }
+ if (device2 == 0) {
+ device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ }
+ if (device2 == 0) {
+ device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ }
+ }
+#endif
+ if (device2 == 0) {
+ device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
+ }
+ if (device2 == 0) {
+ device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
+ }
+ if (device2 == 0) {
+ device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+ }
+
+ // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION, 0 otherwise
+ device |= device2;
+ if (device == 0) {
+ LOGE("getDeviceForStrategy() speaker device not found");
+ }
+ } break;
+
+ default:
+ LOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+ break;
+ }
+
+ LOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+ return device;
+}
+
+void AudioPolicyManagerBase::updateDeviceForStrategy()
+{
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false);
+ }
+}
+
+void AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs)
+{
+ LOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs);
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+
+
+ if (outputDesc->isDuplicated()) {
+ setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
+ setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
+ return;
+ }
+#ifdef WITH_A2DP
+ // filter devices according to output selected
+ if (output == mHardwareOutput) {
+ device &= ~AudioSystem::DEVICE_OUT_ALL_A2DP;
+ } else {
+ device &= AudioSystem::DEVICE_OUT_ALL_A2DP;
+ }
+#endif
+
+ uint32_t prevDevice = (uint32_t)outputDesc->device();
+ // Do not change the routing if:
+ // - the requestede device is 0
+ // - 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 == 0 ||
+ (device == prevDevice && !force)) {
+ LOGV("setOutputDevice() setting same device %x or null device for output %d", device, output);
+ return;
+ }
+
+ outputDesc->mDevice = device;
+ // mute media streams if both speaker and headset are selected
+ if (output == mHardwareOutput && AudioSystem::popCount(device) == 2) {
+ setStrategyMute(STRATEGY_MEDIA, true, output);
+ // wait for the PCM output buffers to empty before proceeding with the rest of the command
+ usleep(outputDesc->mLatency*2*1000);
+ }
+#ifdef WITH_A2DP
+ // suspend A2D output if SCO device is selected
+ if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)device)) {
+ if (mA2dpOutput && mScoDeviceAddress == mA2dpDeviceAddress) {
+ mpClientInterface->suspendOutput(mA2dpOutput);
+ }
+ }
+#endif
+ // do the routing
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), (int)device);
+ mpClientInterface->setParameters(mHardwareOutput, param.toString(), delayMs);
+ // update stream volumes according to new device
+ applyStreamVolumes(output, device, delayMs);
+
+#ifdef WITH_A2DP
+ // if disconnecting SCO device, restore A2DP output
+ if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)prevDevice)) {
+ if (mA2dpOutput && mScoDeviceAddress == mA2dpDeviceAddress) {
+ LOGV("restore A2DP output");
+ mpClientInterface->restoreOutput(mA2dpOutput);
+ }
+ }
+#endif
+ // if changing from a combined headset + speaker route, unmute media streams
+ if (output == mHardwareOutput && AudioSystem::popCount(prevDevice) == 2) {
+ setStrategyMute(STRATEGY_MEDIA, false, output, delayMs);
+ }
+}
+
+uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
+{
+ uint32_t device;
+
+ switch(inputSource) {
+ case AUDIO_SOURCE_DEFAULT:
+ case AUDIO_SOURCE_MIC:
+ case AUDIO_SOURCE_VOICE_RECOGNITION:
+ if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
+ mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) {
+ device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
+ } else {
+ device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_CAMCORDER:
+ if (hasBackMicrophone()) {
+ device = AudioSystem::DEVICE_IN_BACK_MIC;
+ } else {
+ device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ device = AudioSystem::DEVICE_IN_VOICE_CALL;
+ break;
+ default:
+ LOGW("getInput() invalid input source %d", inputSource);
+ device = 0;
+ break;
+ }
+ LOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+ return device;
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getActiveInput()
+{
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ if (mInputs.valueAt(i)->mRefCount > 0) {
+ return mInputs.keyAt(i);
+ }
+ }
+ return 0;
+}
+
+float AudioPolicyManagerBase::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device)
+{
+ float volume = 1.0;
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ StreamDescriptor &streamDesc = mStreams[stream];
+
+ if (device == 0) {
+ device = outputDesc->device();
+ }
+
+ int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
+ volume = AudioSystem::linearToLog(volInt);
+
+ // if a heaset 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.
+ if ((device &
+ (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
+ AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+ AudioSystem::DEVICE_OUT_WIRED_HEADSET |
+ AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) &&
+ (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) &&
+ 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 (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) {
+ float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device);
+ float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+ if (volume > minVol) {
+ volume = minVol;
+ LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+ }
+ }
+ }
+
+ return volume;
+}
+
+status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force)
+{
+
+ // do not change actual stream volume if the stream is muted
+ if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+ LOGV("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)) {
+ LOGV("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);
+ // do not set volume if the float value did not change
+ if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || force) {
+ mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+ LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+ if (stream == AudioSystem::VOICE_CALL ||
+ stream == AudioSystem::DTMF ||
+ stream == AudioSystem::BLUETOOTH_SCO) {
+ float voiceVolume = -1.0;
+ // offset value to reflect actual hardware volume that never reaches 0
+ // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
+ volume = 0.01 + 0.99 * volume;
+ if (stream == AudioSystem::VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+ } else if (stream == AudioSystem::BLUETOOTH_SCO) {
+ voiceVolume = 1.0;
+ }
+ if (voiceVolume >= 0 && output == mHardwareOutput) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ }
+ }
+ mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
+ }
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs)
+{
+ LOGV("applyStreamVolumes() for output %d and device %x", output, device);
+
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, device, delayMs);
+ }
+}
+
+void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs)
+{
+ LOGV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
+ setStreamMute(stream, on, output, delayMs);
+ }
+ }
+}
+
+void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs)
+{
+ StreamDescriptor &streamDesc = mStreams[stream];
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+
+ LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]);
+
+ if (on) {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ if (streamDesc.mCanBeMuted) {
+ checkAndSetVolume(stream, 0, output, outputDesc->device(), delayMs);
+ }
+ }
+ // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+ outputDesc->mMuteCount[stream]++;
+ } else {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ LOGW("setStreamMute() unmuting non muted stream!");
+ return;
+ }
+ if (--outputDesc->mMuteCount[stream] == 0) {
+ checkAndSetVolume(stream, streamDesc.mIndexCur, output, outputDesc->device(), delayMs);
+ }
+ }
+}
+
+void AudioPolicyManagerBase::handleIncallSonification(int 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
+
+ if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
+ LOGV("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 (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
+ LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mHardwareOutput);
+ }
+ } else {
+ LOGV("handleIncallSonification() high visibility");
+ if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE)) {
+ LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mHardwareOutput);
+ }
+ }
+ if (starting) {
+ mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
+ } else {
+ mpClientInterface->stopTone();
+ }
+ }
+ }
+ }
+}
+
+// --- AudioOutputDescriptor class implementation
+
+AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor()
+ : mId(0), mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
+ mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0)
+{
+ // clear usage count for all stream types
+ for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+ mRefCount[i] = 0;
+ mCurVolume[i] = -1.0;
+ mMuteCount[i] = 0;
+ }
+}
+
+uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::device()
+{
+ uint32_t device = 0;
+ if (isDuplicated()) {
+ device = mOutput1->mDevice | mOutput2->mDevice;
+ } else {
+ device = mDevice;
+ }
+ return device;
+}
+
+void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type 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) {
+ LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
+ mRefCount[stream] = 0;
+ return;
+ }
+ mRefCount[stream] += delta;
+ LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
+{
+ uint32_t refcount = 0;
+ for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ refcount += mRefCount[i];
+ }
+ return refcount;
+}
+
+uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::strategyRefCount(routing_strategy strategy)
+{
+ uint32_t refCount = 0;
+ for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ if (getStrategy((AudioSystem::stream_type)i) == strategy) {
+ refCount += mRefCount[i];
+ }
+ }
+ return refCount;
+}
+
+
+status_t AudioPolicyManagerBase::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: %d\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
+ 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 < AudioSystem::NUM_STREAM_TYPES; 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
+
+AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor()
+ : mSamplingRate(0), mFormat(0), mChannels(0),
+ mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
+{
+}
+
+status_t AudioPolicyManagerBase::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", mChannels);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
+ 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
+
+void AudioPolicyManagerBase::StreamDescriptor::dump(char* buffer, size_t size)
+{
+ snprintf(buffer, size, " %02d %02d %02d %d\n",
+ mIndexMin,
+ mIndexMax,
+ mIndexCur,
+ mCanBeMuted);
+}
+
+
+}; // namespace android
diff --git a/libs/audioflinger/AudioPolicyManagerGeneric.cpp b/libs/audioflinger/AudioPolicyManagerGeneric.cpp
deleted file mode 100644
index 8cfc204..0000000
--- a/libs/audioflinger/AudioPolicyManagerGeneric.cpp
+++ /dev/null
@@ -1,945 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "AudioPolicyManagerGeneric"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-#include "AudioPolicyManagerGeneric.h"
-#include <media/mediarecorder.h>
-
-namespace android {
-
-
-// ----------------------------------------------------------------------------
-// AudioPolicyInterface implementation
-// ----------------------------------------------------------------------------
-
-
-status_t AudioPolicyManagerGeneric::setDeviceConnectionState(AudioSystem::audio_devices device,
- AudioSystem::device_connection_state state,
- const char *device_address)
-{
-
- LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
-
- // connect/disconnect only 1 device at a time
- if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
-
- if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
- LOGE("setDeviceConnectionState() invalid address: %s", device_address);
- return BAD_VALUE;
- }
-
- // handle output devices
- if (AudioSystem::isOutputDevice(device)) {
- switch (state)
- {
- // handle output device connection
- case AudioSystem::DEVICE_STATE_AVAILABLE:
- if (mAvailableOutputDevices & device) {
- LOGW("setDeviceConnectionState() device already connected: %x", device);
- return INVALID_OPERATION;
- }
- LOGV("setDeviceConnectionState() connecting device %x", device);
-
- // register new device as available
- mAvailableOutputDevices |= device;
- break;
- // handle output device disconnection
- case AudioSystem::DEVICE_STATE_UNAVAILABLE:
- if (!(mAvailableOutputDevices & device)) {
- LOGW("setDeviceConnectionState() device not connected: %x", device);
- return INVALID_OPERATION;
- }
- LOGV("setDeviceConnectionState() disconnecting device %x", device);
- // remove device from available output devices
- mAvailableOutputDevices &= ~device;
- break;
-
- default:
- LOGE("setDeviceConnectionState() invalid state: %x", state);
- return BAD_VALUE;
- }
- return NO_ERROR;
- }
- // handle input devices
- if (AudioSystem::isInputDevice(device)) {
- switch (state)
- {
- // handle input device connection
- case AudioSystem::DEVICE_STATE_AVAILABLE:
- if (mAvailableInputDevices & device) {
- LOGW("setDeviceConnectionState() device already connected: %d", device);
- return INVALID_OPERATION;
- }
- mAvailableInputDevices |= device;
- break;
-
- // handle input device disconnection
- case AudioSystem::DEVICE_STATE_UNAVAILABLE:
- if (!(mAvailableInputDevices & device)) {
- LOGW("setDeviceConnectionState() device not connected: %d", device);
- return INVALID_OPERATION;
- }
- mAvailableInputDevices &= ~device;
- break;
-
- default:
- LOGE("setDeviceConnectionState() invalid state: %x", state);
- return BAD_VALUE;
- }
- return NO_ERROR;
- }
-
- LOGW("setDeviceConnectionState() invalid device: %x", device);
- return BAD_VALUE;
-}
-
-AudioSystem::device_connection_state AudioPolicyManagerGeneric::getDeviceConnectionState(AudioSystem::audio_devices device,
- const char *device_address)
-{
- AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
- String8 address = String8(device_address);
- if (AudioSystem::isOutputDevice(device)) {
- if (device & mAvailableOutputDevices) {
- state = AudioSystem::DEVICE_STATE_AVAILABLE;
- }
- } else if (AudioSystem::isInputDevice(device)) {
- if (device & mAvailableInputDevices) {
- state = AudioSystem::DEVICE_STATE_AVAILABLE;
- }
- }
-
- return state;
-}
-
-void AudioPolicyManagerGeneric::setPhoneState(int state)
-{
- LOGV("setPhoneState() state %d", state);
- uint32_t newDevice = 0;
- if (state < 0 || state >= AudioSystem::NUM_MODES) {
- LOGW("setPhoneState() invalid state %d", state);
- return;
- }
-
- if (state == mPhoneState ) {
- LOGW("setPhoneState() setting same state %d", state);
- return;
- }
- // store previous phone state for management of sonification strategy below
- int oldState = mPhoneState;
- mPhoneState = state;
-
- // if leaving or entering in call state, handle special case of active streams
- // pertaining to sonification strategy see handleIncallSonification()
- if (state == AudioSystem::MODE_IN_CALL ||
- oldState == AudioSystem::MODE_IN_CALL) {
- bool starting = (state == AudioSystem::MODE_IN_CALL) ? true : false;
- LOGV("setPhoneState() in call state management: new state is %d", state);
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- handleIncallSonification(stream, starting);
- }
- }
-}
-
-void AudioPolicyManagerGeneric::setRingerMode(uint32_t mode, uint32_t mask)
-{
- LOGV("setRingerMode() mode %x, mask %x", mode, mask);
-
- mRingerMode = mode;
-}
-
-void AudioPolicyManagerGeneric::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
-{
- LOGV("setForceUse) usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
- mForceUse[usage] = config;
-}
-
-AudioSystem::forced_config AudioPolicyManagerGeneric::getForceUse(AudioSystem::force_use usage)
-{
- return mForceUse[usage];
-}
-
-void AudioPolicyManagerGeneric::setSystemProperty(const char* property, const char* value)
-{
- LOGV("setSystemProperty() property %s, value %s", property, value);
- if (strcmp(property, "ro.camera.sound.forced") == 0) {
- if (atoi(value)) {
- LOGV("ENFORCED_AUDIBLE cannot be muted");
- mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
- } else {
- LOGV("ENFORCED_AUDIBLE can be muted");
- mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
- }
- }
-}
-
-audio_io_handle_t AudioPolicyManagerGeneric::getOutput(AudioSystem::stream_type stream,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channels,
- AudioSystem::output_flags flags)
-{
- LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
-
-#ifdef AUDIO_POLICY_TEST
- if (mCurOutput != 0) {
- LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
- mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
-
- if (mTestOutputs[mCurOutput] == 0) {
- LOGV("getOutput() opening test output");
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
- outputDesc->mDevice = mTestDevice;
- outputDesc->mSamplingRate = mTestSamplingRate;
- outputDesc->mFormat = mTestFormat;
- outputDesc->mChannels = mTestChannels;
- outputDesc->mLatency = mTestLatencyMs;
- outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
- outputDesc->mRefCount[stream] = 0;
- mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannels,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (mTestOutputs[mCurOutput]) {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"),mCurOutput);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
- mOutputs.add(mTestOutputs[mCurOutput], outputDesc);
- }
- }
- return mTestOutputs[mCurOutput];
- }
-#endif //AUDIO_POLICY_TEST
- if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
- (format != 0 && !AudioSystem::isLinearPCM(format)) ||
- (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO && channels != AudioSystem::CHANNEL_OUT_STEREO)) {
- return 0;
- }
-
- return mHardwareOutput;
-}
-
-status_t AudioPolicyManagerGeneric::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
-{
- LOGV("startOutput() output %d, stream %d", output, stream);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- LOGW("startOutput() unknow output %d", output);
- return BAD_VALUE;
- }
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
- // handle special case for sonification while in call
- if (mPhoneState == AudioSystem::MODE_IN_CALL) {
- handleIncallSonification(stream, true);
- }
-
- // incremenent usage count for this stream on the requested output:
- outputDesc->changeRefCount(stream, 1);
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerGeneric::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
-{
- LOGV("stopOutput() output %d, stream %d", output, stream);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- LOGW("stopOutput() unknow output %d", output);
- return BAD_VALUE;
- }
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
- // handle special case for sonification while in call
- if (mPhoneState == AudioSystem::MODE_IN_CALL) {
- handleIncallSonification(stream, false);
- }
-
- if (outputDesc->isUsedByStream(stream)) {
- // decrement usage count of this stream on the output
- outputDesc->changeRefCount(stream, -1);
- return NO_ERROR;
- } else {
- LOGW("stopOutput() refcount is already 0 for output %d", output);
- return INVALID_OPERATION;
- }
-}
-
-void AudioPolicyManagerGeneric::releaseOutput(audio_io_handle_t output)
-{
- LOGV("releaseOutput() %d", output);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- LOGW("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->refCount() == 0) {
- mpClientInterface->closeOutput(output);
- delete mOutputs.valueAt(index);
- mOutputs.removeItem(output);
- mTestOutputs[testIndex] = 0;
- }
- }
-#endif //AUDIO_POLICY_TEST
-}
-
-audio_io_handle_t AudioPolicyManagerGeneric::getInput(int inputSource,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channels,
- AudioSystem::audio_in_acoustics acoustics)
-{
- audio_io_handle_t input = 0;
- uint32_t device;
-
- LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
-
- AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
- inputDesc->mDevice = AudioSystem::DEVICE_IN_BUILTIN_MIC;
- inputDesc->mSamplingRate = samplingRate;
- inputDesc->mFormat = format;
- inputDesc->mChannels = channels;
- inputDesc->mAcoustics = acoustics;
- inputDesc->mRefCount = 0;
- input = mpClientInterface->openInput(&inputDesc->mDevice,
- &inputDesc->mSamplingRate,
- &inputDesc->mFormat,
- &inputDesc->mChannels,
- inputDesc->mAcoustics);
-
- // only accept input with the exact requested set of parameters
- if ((samplingRate != inputDesc->mSamplingRate) ||
- (format != inputDesc->mFormat) ||
- (channels != inputDesc->mChannels)) {
- LOGV("getOutput() failed opening input: samplingRate %d, format %d, channels %d",
- samplingRate, format, channels);
- mpClientInterface->closeInput(input);
- delete inputDesc;
- return 0;
- }
- mInputs.add(input, inputDesc);
- return input;
-}
-
-status_t AudioPolicyManagerGeneric::startInput(audio_io_handle_t input)
-{
- LOGV("startInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- LOGW("startInput() unknow 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
- for (size_t i = 0; i < mInputs.size(); i++) {
- if (mInputs.valueAt(i)->mRefCount > 0) {
- LOGW("startInput() input %d, other input %d already started", input, mInputs.keyAt(i));
- return INVALID_OPERATION;
- }
- }
- }
-
- inputDesc->mRefCount = 1;
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerGeneric::stopInput(audio_io_handle_t input)
-{
- LOGV("stopInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- LOGW("stopInput() unknow input %d", input);
- return BAD_VALUE;
- }
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
- if (inputDesc->mRefCount == 0) {
- LOGW("stopInput() input %d already stopped", input);
- return INVALID_OPERATION;
- } else {
- inputDesc->mRefCount = 0;
- return NO_ERROR;
- }
-}
-
-void AudioPolicyManagerGeneric::releaseInput(audio_io_handle_t input)
-{
- LOGV("releaseInput() %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- LOGW("releaseInput() releasing unknown input %d", input);
- return;
- }
- mpClientInterface->closeInput(input);
- delete mInputs.valueAt(index);
- mInputs.removeItem(input);
-}
-
-
-
-void AudioPolicyManagerGeneric::initStreamVolume(AudioSystem::stream_type stream,
- int indexMin,
- int indexMax)
-{
- LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
- mStreams[stream].mIndexMin = indexMin;
- mStreams[stream].mIndexMax = indexMax;
-}
-
-status_t AudioPolicyManagerGeneric::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
-{
-
- if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
- return BAD_VALUE;
- }
-
- LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
- mStreams[stream].mIndexCur = index;
-
- // do not change actual stream volume if the stream is muted
- if (mStreams[stream].mMuteCount != 0) {
- return NO_ERROR;
- }
-
- // Do not changed 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)) {
- LOGV("setStreamVolumeIndex() cannot set stream %d volume with force use = %d for comm",
- stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
- return INVALID_OPERATION;
- }
-
- // compute and apply stream volume on all outputs according to connected device
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
- uint32_t device = outputDesc->device();
-
- float volume = computeVolume((int)stream, index, device);
-
- LOGV("setStreamVolume() for output %d stream %d, volume %f", mOutputs.keyAt(i), stream, volume);
- mpClientInterface->setStreamVolume(stream, volume, mOutputs.keyAt(i));
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerGeneric::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
-{
- if (index == 0) {
- return BAD_VALUE;
- }
- LOGV("getStreamVolumeIndex() stream %d", stream);
- *index = mStreams[stream].mIndexCur;
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerGeneric::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, " Hardware Output: %d\n", mHardwareOutput);
- result.append(buffer);
- snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
- result.append(buffer);
- snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
- result.append(buffer);
- snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ringer mode: %d\n", mRingerMode);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- 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 Index Min Index Max Index Cur Mute Count Can be muted\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- snprintf(buffer, SIZE, " %02d", i);
- mStreams[i].dump(buffer + 3, SIZE);
- write(fd, buffer, strlen(buffer));
- }
-
- return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManagerGeneric
-// ----------------------------------------------------------------------------
-
-// --- class factory
-
-AudioPolicyManagerGeneric::AudioPolicyManagerGeneric(AudioPolicyClientInterface *clientInterface)
- :
-#ifdef AUDIO_POLICY_TEST
- Thread(false),
-#endif //AUDIO_POLICY_TEST
- mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0)
-{
- mpClientInterface = clientInterface;
-
- for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
- mForceUse[i] = AudioSystem::FORCE_NONE;
- }
-
- // devices available by default are speaker, ear piece and microphone
- mAvailableOutputDevices = AudioSystem::DEVICE_OUT_SPEAKER;
- mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
-
- // open hardware output
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
- outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
- mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannels,
- &outputDesc->mLatency,
- outputDesc->mFlags);
-
- if (mHardwareOutput == 0) {
- LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
- outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
- } else {
- mOutputs.add(mHardwareOutput, outputDesc);
- }
-
-#ifdef AUDIO_POLICY_TEST
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
-
- mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
- mTestSamplingRate = 44100;
- mTestFormat = AudioSystem::PCM_16_BIT;
- mTestChannels = AudioSystem::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
-}
-
-AudioPolicyManagerGeneric::~AudioPolicyManagerGeneric()
-{
-#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);
- }
- mOutputs.clear();
- for (size_t i = 0; i < mInputs.size(); i++) {
- mpClientInterface->closeInput(mInputs.keyAt(i));
- delete mInputs.valueAt(i);
- }
- mInputs.clear();
-}
-
-#ifdef AUDIO_POLICY_TEST
-bool AudioPolicyManagerGeneric::threadLoop()
-{
- LOGV("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) {
- LOGV("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 = AudioSystem::INVALID_FORMAT;
- if (value == "PCM 16 bits") {
- format = AudioSystem::PCM_16_BIT;
- } else if (value == "PCM 8 bits") {
- format = AudioSystem::PCM_8_BIT;
- } else if (value == "Compressed MP3") {
- format = AudioSystem::MP3;
- }
- if (format != AudioSystem::INVALID_FORMAT) {
- 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 = AudioSystem::CHANNEL_OUT_STEREO;
- } else if (value == "Channels Mono") {
- channels = AudioSystem::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"));
-
- mpClientInterface->closeOutput(mHardwareOutput);
- delete mOutputs.valueFor(mHardwareOutput);
- mOutputs.removeItem(mHardwareOutput);
-
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
- outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
- mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannels,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (mHardwareOutput == 0) {
- LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
- outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
- } else {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
- mOutputs.add(mHardwareOutput, outputDesc);
- }
- }
-
-
- mpClientInterface->setParameters(0, String8("test_cmd_policy="));
- }
- }
- return false;
-}
-
-void AudioPolicyManagerGeneric::exit()
-{
- {
- AutoMutex _l(mLock);
- requestExit();
- mWaitWorkCV.signal();
- }
- requestExitAndWait();
-}
-
-int AudioPolicyManagerGeneric::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
-
-// ---
-
-AudioPolicyManagerGeneric::routing_strategy AudioPolicyManagerGeneric::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::NOTIFICATION:
- case AudioSystem::ALARM:
- case AudioSystem::ENFORCED_AUDIBLE:
- return STRATEGY_SONIFICATION;
- case AudioSystem::DTMF:
- return STRATEGY_DTMF;
- default:
- LOGE("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:
- return STRATEGY_MEDIA;
- }
-}
-
-
-float AudioPolicyManagerGeneric::computeVolume(int stream, int index, uint32_t device)
-{
- float volume = 1.0;
-
- StreamDescriptor &streamDesc = mStreams[stream];
-
- // Force max volume if stream cannot be muted
- if (!streamDesc.mCanBeMuted) index = streamDesc.mIndexMax;
-
- int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
- volume = AudioSystem::linearToLog(volInt);
-
- return volume;
-}
-
-void AudioPolicyManagerGeneric::setStreamMute(int stream, bool on, audio_io_handle_t output)
-{
- LOGV("setStreamMute() stream %d, mute %d, output %d", stream, on, output);
-
- StreamDescriptor &streamDesc = mStreams[stream];
-
- if (on) {
- if (streamDesc.mMuteCount++ == 0) {
- if (streamDesc.mCanBeMuted) {
- mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, 0, output);
- }
- }
- } else {
- if (streamDesc.mMuteCount == 0) {
- LOGW("setStreamMute() unmuting non muted stream!");
- return;
- }
- if (--streamDesc.mMuteCount == 0) {
- uint32_t device = mOutputs.valueFor(output)->mDevice;
- float volume = computeVolume(stream, streamDesc.mIndexCur, device);
- mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output);
- }
- }
-}
-
-void AudioPolicyManagerGeneric::handleIncallSonification(int stream, bool starting)
-{
- // 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 (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
- LOGV("handleIncallSonification() stream %d starting %d device %x", stream, starting, outputDesc->mDevice);
- if (outputDesc->isUsedByStream((AudioSystem::stream_type)stream)) {
- if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
- LOGV("handleIncallSonification() low visibility");
- setStreamMute(stream, starting, mHardwareOutput);
- } else {
- if (starting) {
- mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
- } else {
- mpClientInterface->stopTone();
- }
- }
- }
- }
-}
-
-
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManagerGeneric::AudioOutputDescriptor::AudioOutputDescriptor()
- : mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
- mFlags((AudioSystem::output_flags)0), mDevice(0)
-{
- // clear usage count for all stream types
- for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- mRefCount[i] = 0;
- }
-}
-
-uint32_t AudioPolicyManagerGeneric::AudioOutputDescriptor::device()
-{
- return mDevice;
-}
-
-void AudioPolicyManagerGeneric::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta)
-{
- if ((delta + (int)mRefCount[stream]) < 0) {
- LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
- mRefCount[stream] = 0;
- return;
- }
- mRefCount[stream] += delta;
- LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-uint32_t AudioPolicyManagerGeneric::AudioOutputDescriptor::refCount()
-{
- uint32_t refcount = 0;
- for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
- refcount += mRefCount[i];
- }
- return refcount;
-}
-
-status_t AudioPolicyManagerGeneric::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: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
- 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", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, " Stream refCount\n");
- result.append(buffer);
- for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- snprintf(buffer, SIZE, " %02d %d\n", i, mRefCount[i]);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManagerGeneric::AudioInputDescriptor::AudioInputDescriptor()
- : mSamplingRate(0), mFormat(0), mChannels(0),
- mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
-{
-}
-
-status_t AudioPolicyManagerGeneric::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", mChannels);
- result.append(buffer);
- snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
- 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
-
-void AudioPolicyManagerGeneric::StreamDescriptor::dump(char* buffer, size_t size)
-{
- snprintf(buffer, size, " %02d %02d %02d %02d %d\n",
- mIndexMin,
- mIndexMax,
- mIndexCur,
- mMuteCount,
- mCanBeMuted);
-}
-
-}; // namespace android
diff --git a/libs/audioflinger/AudioPolicyManagerGeneric.h b/libs/audioflinger/AudioPolicyManagerGeneric.h
deleted file mode 100644
index 4997cdf..0000000
--- a/libs/audioflinger/AudioPolicyManagerGeneric.h
+++ /dev/null
@@ -1,196 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <hardware_legacy/AudioPolicyInterface.h>
-#include <utils/threads.h>
-
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-#define MAX_DEVICE_ADDRESS_LEN 20
-#define NUM_TEST_OUTPUTS 5
-
-class AudioPolicyManagerGeneric: public AudioPolicyInterface
-#ifdef AUDIO_POLICY_TEST
- , public Thread
-#endif //AUDIO_POLICY_TEST
-{
-
-public:
- AudioPolicyManagerGeneric(AudioPolicyClientInterface *clientInterface);
- virtual ~AudioPolicyManagerGeneric();
-
- // AudioPolicyInterface
- virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
- AudioSystem::device_connection_state state,
- const char *device_address);
- virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
- const char *device_address);
- virtual void setPhoneState(int state);
- virtual void setRingerMode(uint32_t mode, uint32_t mask);
- virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
- virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
- virtual void setSystemProperty(const char* property, const char* value);
- virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channels,
- AudioSystem::output_flags flags);
- virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
- virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
- virtual void releaseOutput(audio_io_handle_t output);
- virtual audio_io_handle_t getInput(int inputSource,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channels,
- AudioSystem::audio_in_acoustics 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 initStreamVolume(AudioSystem::stream_type stream,
- int indexMin,
- int indexMax);
- virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index);
- virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index);
-
- virtual status_t dump(int fd);
-
-private:
-
- enum routing_strategy {
- STRATEGY_MEDIA,
- STRATEGY_PHONE,
- STRATEGY_SONIFICATION,
- STRATEGY_DTMF,
- NUM_STRATEGIES
- };
-
- // 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();
-
- status_t dump(int fd);
-
- uint32_t device();
- void changeRefCount(AudioSystem::stream_type, int delta);
- bool isUsedByStream(AudioSystem::stream_type stream) { return mRefCount[stream] > 0 ? true : false; }
- uint32_t refCount();
-
- uint32_t mSamplingRate; //
- uint32_t mFormat; //
- uint32_t mChannels; // output configuration
- uint32_t mLatency; //
- AudioSystem::output_flags mFlags; //
- uint32_t mDevice; // current device this output is routed to
- uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output
- };
-
- // 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();
-
- status_t dump(int fd);
-
- uint32_t mSamplingRate; //
- uint32_t mFormat; // input configuration
- uint32_t mChannels; //
- AudioSystem::audio_in_acoustics mAcoustics; //
- uint32_t mDevice; // current device this input is routed to
- uint32_t mRefCount; // number of AudioRecord clients using this output
- };
-
- // stream descriptor used for volume control
- class StreamDescriptor
- {
- public:
- StreamDescriptor()
- : mIndexMin(0), mIndexMax(1), mIndexCur(1), mMuteCount(0), mCanBeMuted(true) {}
-
- void dump(char* buffer, size_t size);
-
- int mIndexMin; // min volume index
- int mIndexMax; // max volume index
- int mIndexCur; // current volume index
- int mMuteCount; // mute request counter
- bool mCanBeMuted; // true is the stream can be muted
- };
-
- // return the strategy corresponding to a given stream type
- static routing_strategy getStrategy(AudioSystem::stream_type stream);
- // return the output handle of an output routed to the specified device, 0 if no output
- // is routed to the device
- float computeVolume(int stream, int index, uint32_t device);
- // Mute or unmute the stream on the specified output
- void setStreamMute(int stream, bool on, audio_io_handle_t output);
- // 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(int stream, bool starting);
-
-
-#ifdef AUDIO_POLICY_TEST
- virtual bool threadLoop();
- void exit();
- int testOutputIndex(audio_io_handle_t output);
-#endif //AUDIO_POLICY_TEST
-
-
- AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
- audio_io_handle_t mHardwareOutput; // hardware output handler
-
- KeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs; // list ot output descritors
- KeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors
- uint32_t mAvailableOutputDevices; // bit field of all available output devices
- uint32_t mAvailableInputDevices; // bit field of all available input devices
- int mPhoneState; // current phone state
- uint32_t mRingerMode; // current ringer mode
- AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE]; // current forced use configuration
-
- StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES]; // stream descriptors for volume control
-
-#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
-
-};
-
-};
diff --git a/libs/audioflinger/AudioPolicyService.cpp b/libs/audioflinger/AudioPolicyService.cpp
index aa48019..bb3905c 100644
--- a/libs/audioflinger/AudioPolicyService.cpp
+++ b/libs/audioflinger/AudioPolicyService.cpp
@@ -30,9 +30,10 @@
#include <utils/String16.h>
#include <utils/threads.h>
#include "AudioPolicyService.h"
-#include "AudioPolicyManagerGeneric.h"
+#include <hardware_legacy/AudioPolicyManagerBase.h>
#include <cutils/properties.h>
#include <dlfcn.h>
+#include <hardware_legacy/power.h>
// ----------------------------------------------------------------------------
// the sim build doesn't have gettid
@@ -43,8 +44,9 @@
namespace android {
-static const char* kDeadlockedString = "AudioPolicyService may be deadlocked\n";
-static const char* kCmdDeadlockedString = "AudioPolicyService command thread may be deadlocked\n";
+
+static const char *kDeadlockedString = "AudioPolicyService may be deadlocked\n";
+static const char *kCmdDeadlockedString = "AudioPolicyService command thread may be deadlocked\n";
static const int kDumpLockRetries = 50;
static const int kDumpLockSleep = 20000;
@@ -67,18 +69,18 @@ AudioPolicyService::AudioPolicyService()
char value[PROPERTY_VALUE_MAX];
// start tone playback thread
- mTonePlaybackThread = new AudioCommandThread();
+ mTonePlaybackThread = new AudioCommandThread(String8(""));
// start audio commands thread
- mAudioCommandThread = new AudioCommandThread();
+ mAudioCommandThread = new AudioCommandThread(String8("ApmCommandThread"));
#if (defined GENERIC_AUDIO) || (defined AUDIO_POLICY_TEST)
- mpPolicyManager = new AudioPolicyManagerGeneric(this);
+ mpPolicyManager = new AudioPolicyManagerBase(this);
LOGV("build for GENERIC_AUDIO - using generic audio policy");
#else
// if running in emulation - use the emulator driver
if (property_get("ro.kernel.qemu", value, 0)) {
LOGV("Running in emulation - using generic audio policy");
- mpPolicyManager = new AudioPolicyManagerGeneric(this);
+ mpPolicyManager = new AudioPolicyManagerBase(this);
}
else {
LOGV("Using hardware specific audio policy");
@@ -556,8 +558,8 @@ status_t AudioPolicyService::setVoiceVolume(float volume, int delayMs)
// ----------- AudioPolicyService::AudioCommandThread implementation ----------
-AudioPolicyService::AudioCommandThread::AudioCommandThread()
- : Thread(false)
+AudioPolicyService::AudioCommandThread::AudioCommandThread(String8 name)
+ : Thread(false), mName(name)
{
mpToneGenerator = NULL;
}
@@ -565,18 +567,20 @@ AudioPolicyService::AudioCommandThread::AudioCommandThread()
AudioPolicyService::AudioCommandThread::~AudioCommandThread()
{
+ if (mName != "" && !mAudioCommands.isEmpty()) {
+ release_wake_lock(mName.string());
+ }
mAudioCommands.clear();
if (mpToneGenerator != NULL) delete mpToneGenerator;
}
void AudioPolicyService::AudioCommandThread::onFirstRef()
{
- const size_t SIZE = 256;
- char buffer[SIZE];
-
- snprintf(buffer, SIZE, "AudioCommandThread");
-
- run(buffer, ANDROID_PRIORITY_AUDIO);
+ if (mName != "") {
+ run(mName.string(), ANDROID_PRIORITY_AUDIO);
+ } else {
+ run("AudioCommandThread", ANDROID_PRIORITY_AUDIO);
+ }
}
bool AudioPolicyService::AudioCommandThread::threadLoop()
@@ -657,6 +661,10 @@ bool AudioPolicyService::AudioCommandThread::threadLoop()
break;
}
}
+ // release delayed commands wake lock
+ if (mName != "" && mAudioCommands.isEmpty()) {
+ release_wake_lock(mName.string());
+ }
LOGV("AudioCommandThread() going to sleep");
mWaitWorkCV.waitRelative(mLock, waitTime);
LOGV("AudioCommandThread() waking up");
@@ -815,6 +823,11 @@ void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *comma
command->mTime = systemTime() + milliseconds(delayMs);
+ // acquire wake lock to make sure delayed commands are processed
+ if (mName != "" && mAudioCommands.isEmpty()) {
+ acquire_wake_lock(PARTIAL_WAKE_LOCK, mName.string());
+ }
+
// check same pending commands with later time stamps and eliminate them
for (i = mAudioCommands.size()-1; i >= 0; i--) {
AudioCommand *command2 = mAudioCommands[i];
@@ -883,7 +896,7 @@ void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *comma
removedCommands.clear();
// insert command at the right place according to its time stamp
- LOGV("inserting command: %d at index %ld, num commands %d", command->mCommand, i+1, mAudioCommands.size());
+ LOGV("inserting command: %d at index %d, num commands %d", command->mCommand, (int)i+1, mAudioCommands.size());
mAudioCommands.insertAt(command, i + 1);
}
diff --git a/libs/audioflinger/AudioPolicyService.h b/libs/audioflinger/AudioPolicyService.h
index b9234ec..a13d0bd 100644
--- a/libs/audioflinger/AudioPolicyService.h
+++ b/libs/audioflinger/AudioPolicyService.h
@@ -132,7 +132,7 @@ private:
SET_VOICE_VOLUME
};
- AudioCommandThread ();
+ AudioCommandThread (String8 name);
virtual ~AudioCommandThread();
status_t dump(int fd);
@@ -195,7 +195,8 @@ private:
Condition mWaitWorkCV;
Vector <AudioCommand *> mAudioCommands; // list of pending commands
ToneGenerator *mpToneGenerator; // the tone generator
- AudioCommand mLastCommand;
+ AudioCommand mLastCommand; // last processed command (used by dump)
+ String8 mName; // string used by wake lock fo delayed commands
};
// Internal dump utilities.
generated by cgit v1.1 at 2024-09-12 06:38:57 +0000