move native services under services/
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Icd7336f7289db35df9c8c1857a5122bb8a6f1c86
diff --git a/services/audioflinger/AudioPolicyManagerBase.cpp b/services/audioflinger/AudioPolicyManagerBase.cpp
new file mode 100644
index 0000000..c8b3f48
--- /dev/null
+++ b/services/audioflinger/AudioPolicyManagerBase.cpp
@@ -0,0 +1,1972 @@
+/*
+ * 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 (mA2dpOutput != 0 &&
+ 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 (mA2dpOutput != 0 &&
+ 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 a SCO device is present.
+ if (mA2dpOutput != 0 && 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();
+ }
+
+ // when changing from ring tone to in call mode, mute the ringing tone
+ // immediately and delay the route change to avoid sending the ring tone
+ // tail into the earpiece or headset.
+ int delayMs = 0;
+ if (state == AudioSystem::MODE_IN_CALL && oldState == AudioSystem::MODE_RINGTONE) {
+ // delay the device change command by twice the output latency to have some margin
+ // and be sure that audio buffers not yet affected by the mute are out when
+ // we actually apply the route change
+ delayMs = hwOutputDesc->mLatency*2;
+ setStreamMute(AudioSystem::RING, true, mHardwareOutput);
+ }
+
+ // change routing is necessary
+ setOutputDevice(mHardwareOutput, newDevice, force, delayMs);
+
+ // 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);
+ // unmute the ringing tone after a sufficient delay if it was muted before
+ // setting output device above
+ if (oldState == AudioSystem::MODE_RINGTONE) {
+ setStreamMute(AudioSystem::RING, false, mHardwareOutput, MUTE_TIME_MS);
+ }
+ 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);
+
+ bool forceVolumeReeval = false;
+ 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);
+ }
+ forceVolumeReeval = true;
+ 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);
+ if (forceVolumeReeval) {
+ applyStreamVolumes(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 required by specified parameters
+ if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) {
+
+ 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] = 0;
+ 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
+ if (output != mHardwareOutput) {
+ setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true);
+ }
+ 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 (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 strategy to avoid outputting 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, MUTE_TIME_MS);
+
+ 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 if 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;
+#ifdef WITH_A2DP
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+ if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device) break;
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ if (device) break;
+ }
+#endif
+ 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;
+ }
+#ifdef WITH_A2DP
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+ // A2DP speaker when forcing to speaker output
+ if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ if (device) break;
+ }
+#endif
+ 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;
+ if (device2 == 0) {
+ device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
+ }
+ if (device2 == 0) {
+ device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
+ }
+#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_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 == mA2dpOutput) {
+ 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 A2DP output if SCO device is selected
+ if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)device)) {
+ if (mA2dpOutput != 0) {
+ 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 != 0) {
+ 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 headset is connected, apply the following rules to ring tones and notifications
+ // to avoid sound level bursts in user's ears:
+ // - always attenuate ring tones and notifications volume by 6dB
+ // - if music is playing, always limit the volume to current music volume,
+ // with a minimum threshold at -36dB so that notification is always perceived.
+ 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();
+ }
+ }
+ }
+ }
+}
+
+bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channels,
+ AudioSystem::output_flags flags,
+ uint32_t device)
+{
+ return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+ (format !=0 && !AudioSystem::isLinearPCM(format)));
+}
+
+// --- 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