Refactor AudioPolicyManager
AudioPolicyManager implementation is now split into the
following files:
files managerdefault/Gains.*
class AudioGain
class VolumeCurvePoint
class StreamDescriptor
files managerdefault/Devices.*
class DeviceDescriptor
class DeviceVector
files managerdefault/Ports.*
class AudioPort
class AudioPortConfig
class AudioPatch
files managerdefault/IOProfile.*
class IOProfile
files managerdefault/HwModule.*
class HwModule
files managerdefault/AudioInputDescriptor.*
class AudioInputDescriptor
files managerdefault/AudioOutputDescriptor.*
class AudioOutputDescriptor
All files for libaudiopolicyservice are moved under service/
All files for libaudiopolicymanager are moved under manager/
Change-Id: I43758be1894e37d34db194b51a19ae24461e066e
diff --git a/services/audiopolicy/managerdefault/ApmImplDefinitions.h b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
new file mode 100644
index 0000000..620979b
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+enum routing_strategy {
+ STRATEGY_MEDIA,
+ STRATEGY_PHONE,
+ STRATEGY_SONIFICATION,
+ STRATEGY_SONIFICATION_RESPECTFUL,
+ STRATEGY_DTMF,
+ STRATEGY_ENFORCED_AUDIBLE,
+ STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
+ STRATEGY_ACCESSIBILITY,
+ STRATEGY_REROUTING,
+ NUM_STRATEGIES
+};
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
new file mode 100644
index 0000000..f4054c8
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
@@ -0,0 +1,100 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioInputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
+ : mId(0), mIoHandle(0),
+ mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
+ mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
+{
+ if (profile != NULL) {
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ }
+}
+
+void AudioInputDescriptor::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ ALOG_ASSERT(mProfile != 0,
+ "toAudioPortConfig() called on input with null profile %d", mIoHandle);
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = AUDIO_PORT_ROLE_SINK;
+ dstConfig->type = AUDIO_PORT_TYPE_MIX;
+ dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+ dstConfig->ext.mix.handle = mIoHandle;
+ dstConfig->ext.mix.usecase.source = mInputSource;
+}
+
+void AudioInputDescriptor::toAudioPort(
+ struct audio_port *port) const
+{
+ ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
+
+ mProfile->toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.mix.hw_module = mProfile->mModule->mHandle;
+ port->ext.mix.handle = mIoHandle;
+ port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioInputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " ID: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.h b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
new file mode 100644
index 0000000..02579e6
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+// 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 AudioPortConfig
+{
+public:
+ AudioInputDescriptor(const sp<IOProfile>& profile);
+
+ status_t dump(int fd);
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // input handle
+ audio_devices_t mDevice; // current device this input is routed to
+ AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
+ audio_patch_handle_t mPatchHandle;
+ uint32_t mRefCount; // number of AudioRecord clients using
+ // this input
+ uint32_t mOpenRefCount;
+ audio_source_t mInputSource; // input source selected by application
+ //(mediarecorder.h)
+ const sp<IOProfile> mProfile; // I/O profile this output derives from
+ SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
+ bool mIsSoundTrigger; // used by a soundtrigger capture
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+ void toAudioPort(struct audio_port *port) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
new file mode 100644
index 0000000..4b85972
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
@@ -0,0 +1,221 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioOutputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+AudioOutputDescriptor::AudioOutputDescriptor(
+ const sp<IOProfile>& profile)
+ : mId(0), mIoHandle(0), mLatency(0),
+ mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
+ mPatchHandle(0),
+ mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+ // clear usage count for all stream types
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ mRefCount[i] = 0;
+ mCurVolume[i] = -1.0;
+ mMuteCount[i] = 0;
+ mStopTime[i] = 0;
+ }
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mStrategyMutedByDevice[i] = false;
+ }
+ if (profile != NULL) {
+ mFlags = (audio_output_flags_t)profile->mFlags;
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ }
+}
+
+audio_devices_t AudioOutputDescriptor::device() const
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+ } else {
+ return mDevice;
+ }
+}
+
+uint32_t AudioOutputDescriptor::latency()
+{
+ if (isDuplicated()) {
+ return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+ } else {
+ return mLatency;
+ }
+}
+
+bool AudioOutputDescriptor::sharesHwModuleWith(
+ const sp<AudioOutputDescriptor> outputDesc)
+{
+ if (isDuplicated()) {
+ return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+ } else if (outputDesc->isDuplicated()){
+ return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+ } else {
+ return (mProfile->mModule == outputDesc->mProfile->mModule);
+ }
+}
+
+void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+ int delta)
+{
+ // forward usage count change to attached outputs
+ if (isDuplicated()) {
+ mOutput1->changeRefCount(stream, delta);
+ mOutput2->changeRefCount(stream, delta);
+ }
+ if ((delta + (int)mRefCount[stream]) < 0) {
+ ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
+ delta, stream, mRefCount[stream]);
+ mRefCount[stream] = 0;
+ return;
+ }
+ mRefCount[stream] += delta;
+ ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioOutputDescriptor::supportedDevices()
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+ } else {
+ return mProfile->mSupportedDevices.types() ;
+ }
+}
+
+bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+ return isStrategyActive(NUM_STRATEGIES, inPastMs);
+}
+
+bool AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if ((sysTime == 0) && (inPastMs != 0)) {
+ sysTime = systemTime();
+ }
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ if (i == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (((AudioPolicyManager::getStrategy((audio_stream_type_t)i) == strategy) ||
+ (NUM_STRATEGIES == strategy)) &&
+ isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if (mRefCount[stream] != 0) {
+ return true;
+ }
+ if (inPastMs == 0) {
+ return false;
+ }
+ if (sysTime == 0) {
+ sysTime = systemTime();
+ }
+ if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+ return true;
+ }
+ return false;
+}
+
+void AudioOutputDescriptor::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
+
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_MIX;
+ dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+ dstConfig->ext.mix.handle = mIoHandle;
+ dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
+}
+
+void AudioOutputDescriptor::toAudioPort(
+ struct audio_port *port) const
+{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
+ mProfile->toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.mix.hw_module = mProfile->mModule->mHandle;
+ port->ext.mix.handle = mIoHandle;
+ port->ext.mix.latency_class =
+ mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioOutputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " ID: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", device());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+ result.append(buffer);
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
+ i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.h b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
new file mode 100644
index 0000000..32f46e4
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2015 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 "ApmImplDefinitions.h"
+
+namespace android {
+
+// 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 AudioPortConfig
+{
+public:
+ AudioOutputDescriptor(const sp<IOProfile>& profile);
+
+ status_t dump(int fd);
+
+ audio_devices_t device() const;
+ void changeRefCount(audio_stream_type_t stream, int delta);
+
+ bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+ audio_devices_t supportedDevices();
+ uint32_t latency();
+ bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
+ bool isActive(uint32_t inPastMs = 0) const;
+ bool isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs = 0,
+ nsecs_t sysTime = 0) const;
+ bool isStrategyActive(routing_strategy strategy,
+ uint32_t inPastMs = 0,
+ nsecs_t sysTime = 0) const;
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+ void toAudioPort(struct audio_port *port) const;
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // output handle
+ uint32_t mLatency; //
+ audio_output_flags_t mFlags; //
+ audio_devices_t mDevice; // current device this output is routed to
+ AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
+ audio_patch_handle_t mPatchHandle;
+ uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
+ nsecs_t mStopTime[AUDIO_STREAM_CNT];
+ sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
+ sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output
+ float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
+ int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
+ const sp<IOProfile> mProfile; // I/O profile this output derives from
+ bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
+ // device selection. See checkDeviceMuteStrategies()
+ uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
new file mode 100644
index 0000000..b48dc80
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -0,0 +1,5766 @@
+/*
+ * 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 "APM::AudioPolicyManager"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+// A device mask for all audio input devices that are considered "virtual" when evaluating
+// active inputs in getActiveInput()
+#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER)
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+// A device mask for all audio input and output devices where matching inputs/outputs on device
+// type alone is not enough: the address must match too
+#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
+
+#include <inttypes.h>
+#include <math.h>
+
+#include <cutils/properties.h>
+#include <utils/Log.h>
+#include <hardware/audio.h>
+#include <hardware/audio_effect.h>
+#include <media/AudioParameter.h>
+#include <media/AudioPolicyHelper.h>
+#include <soundtrigger/SoundTrigger.h>
+#include "AudioPolicyManager.h"
+#include "audio_policy_conf.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name)
+{
+ return setDeviceConnectionStateInt(device, state, device_address, device_name);
+}
+
+status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name)
+{
+ ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
+- device, state, device_address, device_name);
+
+ // connect/disconnect only 1 device at a time
+ if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+
+ sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address, device_name);
+
+ // handle output devices
+ if (audio_is_output_device(device)) {
+ SortedVector <audio_io_handle_t> outputs;
+
+ ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
+
+ // save a copy of the opened output descriptors before any output is opened or closed
+ // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
+ mPreviousOutputs = mOutputs;
+ switch (state)
+ {
+ // handle output device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ ALOGW("setDeviceConnectionState() device already connected: %x", device);
+ return INVALID_OPERATION;
+ }
+ ALOGV("setDeviceConnectionState() connecting device %x", device);
+
+ // register new device as available
+ index = mAvailableOutputDevices.add(devDesc);
+ if (index >= 0) {
+ sp<HwModule> module = getModuleForDevice(device);
+ if (module == 0) {
+ ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
+ device);
+ mAvailableOutputDevices.remove(devDesc);
+ return INVALID_OPERATION;
+ }
+ mAvailableOutputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
+ mAvailableOutputDevices.remove(devDesc);
+ return INVALID_OPERATION;
+ }
+ // outputs should never be empty here
+ ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
+ "checkOutputsForDevice() returned no outputs but status OK");
+ ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
+ outputs.size());
+
+ // Send connect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ } break;
+ // handle output device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ ALOGW("setDeviceConnectionState() device not connected: %x", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
+
+ // Send Disconnect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ // remove device from available output devices
+ mAvailableOutputDevices.remove(devDesc);
+
+ checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
+ // output is suspended before any tracks are moved to it
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ // outputs must be closed after checkOutputForAllStrategies() is executed
+ if (!outputs.isEmpty()) {
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+ // close unused outputs after device disconnection or direct outputs that have been
+ // opened by checkOutputsForDevice() to query dynamic parameters
+ if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+ (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+ (desc->mDirectOpenCount == 0))) {
+ closeOutput(outputs[i]);
+ }
+ }
+ // check again after closing A2DP output to reset mA2dpSuspended if needed
+ checkA2dpSuspend();
+ }
+
+ updateDevicesAndOutputs();
+ if (mPhoneState == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
+ audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i),
+ true /*fromCache*/);
+ // do not force device change on duplicated output because if device is 0, it will
+ // also force a device 0 for the two outputs it is duplicated to which may override
+ // a valid device selection on those outputs.
+ bool force = !mOutputs.valueAt(i)->isDuplicated()
+ && (!deviceDistinguishesOnAddress(device)
+ // always force when disconnecting (a non-duplicated device)
+ || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
+ setOutputDevice(output, newDevice, force, 0);
+ }
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is output device
+
+ // handle input devices
+ if (audio_is_input_device(device)) {
+ SortedVector <audio_io_handle_t> inputs;
+
+ ssize_t index = mAvailableInputDevices.indexOf(devDesc);
+ switch (state)
+ {
+ // handle input device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ ALOGW("setDeviceConnectionState() device already connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ sp<HwModule> module = getModuleForDevice(device);
+ if (module == NULL) {
+ ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
+ device);
+ return INVALID_OPERATION;
+ }
+ if (checkInputsForDevice(device, state, inputs, devDesc->mAddress) != NO_ERROR) {
+ return INVALID_OPERATION;
+ }
+
+ index = mAvailableInputDevices.add(devDesc);
+ if (index >= 0) {
+ mAvailableInputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ // Set connect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ } break;
+
+ // handle input device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ ALOGW("setDeviceConnectionState() device not connected: %d", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
+
+ // Set Disconnect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ checkInputsForDevice(device, state, inputs, devDesc->mAddress);
+ mAvailableInputDevices.remove(devDesc);
+
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ closeAllInputs();
+
+ if (mPhoneState == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is input device
+
+ ALOGW("setDeviceConnectionState() invalid device: %x", device);
+ return BAD_VALUE;
+}
+
+audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
+ const char *device_address)
+{
+ sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address, "");
+ DeviceVector *deviceVector;
+
+ if (audio_is_output_device(device)) {
+ deviceVector = &mAvailableOutputDevices;
+ } else if (audio_is_input_device(device)) {
+ deviceVector = &mAvailableInputDevices;
+ } else {
+ ALOGW("getDeviceConnectionState() invalid device type %08x", device);
+ return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ }
+
+ ssize_t index = deviceVector->indexOf(devDesc);
+ if (index >= 0) {
+ return AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
+ } else {
+ return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ }
+}
+
+sp<DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name)
+{
+ String8 address = (device_address == NULL) ? String8("") : String8(device_address);
+ // handle legacy remote submix case where the address was not always specified
+ if (deviceDistinguishesOnAddress(device) && (address.length() == 0)) {
+ address = String8("0");
+ }
+
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ DeviceVector deviceList =
+ mHwModules[i]->mDeclaredDevices.getDevicesFromTypeAddr(device, address);
+ if (!deviceList.isEmpty()) {
+ return deviceList.itemAt(0);
+ }
+ deviceList = mHwModules[i]->mDeclaredDevices.getDevicesFromType(device);
+ if (!deviceList.isEmpty()) {
+ return deviceList.itemAt(0);
+ }
+ }
+
+ sp<DeviceDescriptor> devDesc =
+ new DeviceDescriptor(String8(device_name != NULL ? device_name : ""), device);
+ devDesc->mAddress = address;
+ return devDesc;
+}
+
+void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs)
+{
+ bool createTxPatch = false;
+ struct audio_patch patch;
+ patch.num_sources = 1;
+ patch.num_sinks = 1;
+ status_t status;
+ audio_patch_handle_t afPatchHandle;
+ DeviceVector deviceList;
+
+ audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+ ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice);
+
+ // release existing RX patch if any
+ if (mCallRxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ mCallRxPatch.clear();
+ }
+ // release TX patch if any
+ if (mCallTxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ mCallTxPatch.clear();
+ }
+
+ // If the RX device is on the primary HW module, then use legacy routing method for voice calls
+ // via setOutputDevice() on primary output.
+ // Otherwise, create two audio patches for TX and RX path.
+ if (availablePrimaryOutputDevices() & rxDevice) {
+ setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs);
+ // If the TX device is also on the primary HW module, setOutputDevice() will take care
+ // of it due to legacy implementation. If not, create a patch.
+ if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN)
+ == AUDIO_DEVICE_NONE) {
+ createTxPatch = true;
+ }
+ } else {
+ // create RX path audio patch
+ deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() selected device not in output device list");
+ sp<DeviceDescriptor> rxSinkDeviceDesc = deviceList.itemAt(0);
+ deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() no telephony RX device");
+ sp<DeviceDescriptor> rxSourceDeviceDesc = deviceList.itemAt(0);
+
+ rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
+ rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
+
+ // request to reuse existing output stream if one is already opened to reach the RX device
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(rxDevice, mOutputs);
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "updateCallRouting() RX device output is duplicated");
+ outputDesc->toAudioPortConfig(&patch.sources[1]);
+ patch.num_sources = 2;
+ }
+
+ afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
+ ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch",
+ status);
+ if (status == NO_ERROR) {
+ mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &patch, mUidCached);
+ mCallRxPatch->mAfPatchHandle = afPatchHandle;
+ mCallRxPatch->mUid = mUidCached;
+ }
+ createTxPatch = true;
+ }
+ if (createTxPatch) {
+
+ struct audio_patch patch;
+ patch.num_sources = 1;
+ patch.num_sinks = 1;
+ deviceList = mAvailableInputDevices.getDevicesFromType(txDevice);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() selected device not in input device list");
+ sp<DeviceDescriptor> txSourceDeviceDesc = deviceList.itemAt(0);
+ txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
+ deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() no telephony TX device");
+ sp<DeviceDescriptor> txSinkDeviceDesc = deviceList.itemAt(0);
+ txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
+
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs);
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ // request to reuse existing output stream if one is already opened to reach the TX
+ // path output device
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "updateCallRouting() RX device output is duplicated");
+ outputDesc->toAudioPortConfig(&patch.sources[1]);
+ patch.num_sources = 2;
+ }
+
+ afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
+ ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch",
+ status);
+ if (status == NO_ERROR) {
+ mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &patch, mUidCached);
+ mCallTxPatch->mAfPatchHandle = afPatchHandle;
+ mCallTxPatch->mUid = mUidCached;
+ }
+ }
+}
+
+void AudioPolicyManager::setPhoneState(audio_mode_t state)
+{
+ ALOGV("setPhoneState() state %d", state);
+ if (state < 0 || state >= AUDIO_MODE_CNT) {
+ ALOGW("setPhoneState() invalid state %d", state);
+ return;
+ }
+
+ if (state == mPhoneState ) {
+ ALOGW("setPhoneState() setting same state %d", state);
+ return;
+ }
+
+ // if leaving call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isInCall()) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ handleIncallSonification((audio_stream_type_t)stream, false, true);
+ }
+
+ // force reevaluating accessibility routing when call starts
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ // 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 (!isStateInCall(oldState) && isStateInCall(state)) {
+ ALOGV(" Entering call in setPhoneState()");
+ // force routing command to audio hardware when starting a call
+ // even if no device change is needed
+ force = true;
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
+ mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+ ApmGains::sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+ }
+ } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+ ALOGV(" Exiting call in setPhoneState()");
+ // force routing command to audio hardware when exiting a call
+ // even if no device change is needed
+ force = true;
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
+ mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+ ApmGains::sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+ }
+ } else if (isStateInCall(state) && (state != oldState)) {
+ ALOGV(" Switching between telephony and VoIP in setPhoneState()");
+ // force routing command to audio hardware when switching between telephony and VoIP
+ // even if no device change is needed
+ force = true;
+ }
+
+ // check for device and output changes triggered by new phone state
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+
+ sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+ int delayMs = 0;
+ if (isStateInCall(state)) {
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ // mute media and sonification strategies and delay device switch by the largest
+ // latency of any output where either strategy is active.
+ // This avoid sending the ring tone or music tail into the earpiece or headset.
+ if ((desc->isStrategyActive(STRATEGY_MEDIA,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime) ||
+ desc->isStrategyActive(STRATEGY_SONIFICATION,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime)) &&
+ (delayMs < (int)desc->mLatency*2)) {
+ delayMs = desc->mLatency*2;
+ }
+ setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+ setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+ }
+ }
+
+ // Note that despite the fact that getNewOutputDevice() is called on the primary output,
+ // the device returned is not necessarily reachable via this output
+ audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
+ rxDevice = hwOutputDesc->device();
+ }
+
+ if (state == AUDIO_MODE_IN_CALL) {
+ updateCallRouting(rxDevice, delayMs);
+ } else if (oldState == AUDIO_MODE_IN_CALL) {
+ if (mCallRxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ mCallRxPatch.clear();
+ }
+ if (mCallTxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ mCallTxPatch.clear();
+ }
+ setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ } else {
+ setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ }
+ // if entering in call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isStateInCall(state)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ handleIncallSonification((audio_stream_type_t)stream, true, true);
+ }
+ }
+
+ // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+ if (state == AUDIO_MODE_RINGTONE &&
+ isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+ mLimitRingtoneVolume = true;
+ } else {
+ mLimitRingtoneVolume = false;
+ }
+}
+
+void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+
+ bool forceVolumeReeval = false;
+ switch(usage) {
+ case AUDIO_POLICY_FORCE_FOR_COMMUNICATION:
+ if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO &&
+ config != AUDIO_POLICY_FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+ return;
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_MEDIA:
+ if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP &&
+ config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+ config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) {
+ ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_RECORD:
+ if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_DOCK:
+ if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK &&
+ config != AUDIO_POLICY_FORCE_BT_DESK_DOCK &&
+ config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+ config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) {
+ ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_SYSTEM:
+ if (config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO:
+ if (config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) {
+ ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config);
+ }
+ mForceUse[usage] = config;
+ break;
+ default:
+ ALOGW("setForceUse() invalid usage %d", usage);
+ break;
+ }
+
+ // check for device and output changes triggered by new force usage
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ if (mPhoneState == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/);
+ if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
+ setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ }
+ if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+ applyStreamVolumes(output, newDevice, 0, true);
+ }
+ }
+
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ setInputDevice(activeInput, getNewInputDevice(activeInput));
+ }
+
+}
+
+audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
+{
+ return mForceUse[usage];
+}
+
+void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
+{
+ ALOGV("setSystemProperty() property %s, value %s", property, value);
+}
+
+// Find a direct output profile compatible with the parameters passed, even if the input flags do
+// not explicitly request a direct output
+sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput(
+ audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags)
+{
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ bool found = profile->isCompatibleProfile(device, String8(""), samplingRate,
+ NULL /*updatedSamplingRate*/, format, channelMask,
+ flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ?
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT);
+ if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) {
+ return profile;
+ }
+ }
+ }
+ return 0;
+}
+
+audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ routing_strategy strategy = getStrategy(stream);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
+ device, stream, samplingRate, format, channelMask, flags);
+
+ return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE,
+ stream, samplingRate,format, channelMask,
+ flags, offloadInfo);
+}
+
+status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_attributes_t attributes;
+ if (attr != NULL) {
+ if (!isValidAttributes(attr)) {
+ ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+ attr->usage, attr->content_type, attr->flags,
+ attr->tags);
+ return BAD_VALUE;
+ }
+ attributes = *attr;
+ } else {
+ if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) {
+ ALOGE("getOutputForAttr(): invalid stream type");
+ return BAD_VALUE;
+ }
+ stream_type_to_audio_attributes(*stream, &attributes);
+ }
+
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ sp<AudioOutputDescriptor> desc;
+ if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) {
+ for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
+ if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) ||
+ (RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) {
+ desc = mPolicyMixes[i]->mOutput;
+ break;
+ }
+ if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attributes.tags + strlen("addr="),
+ mPolicyMixes[i]->mMix.mRegistrationId.string(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
+ desc = mPolicyMixes[i]->mOutput;
+ break;
+ }
+ }
+ } else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) {
+ if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE &&
+ strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attributes.tags + strlen("addr="),
+ mPolicyMixes[i]->mMix.mRegistrationId.string(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
+ desc = mPolicyMixes[i]->mOutput;
+ }
+ }
+ if (desc != 0) {
+ if (!audio_is_linear_pcm(format)) {
+ return BAD_VALUE;
+ }
+ desc->mPolicyMix = &mPolicyMixes[i]->mMix;
+ *stream = streamTypefromAttributesInt(&attributes);
+ *output = desc->mIoHandle;
+ ALOGV("getOutputForAttr() returns output %d", *output);
+ return NO_ERROR;
+ }
+ }
+ if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+ ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
+ return BAD_VALUE;
+ }
+
+ ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x",
+ attributes.usage, attributes.content_type, attributes.tags, attributes.flags);
+
+ routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+
+ if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
+ }
+
+ ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x",
+ device, samplingRate, format, channelMask, flags);
+
+ *stream = streamTypefromAttributesInt(&attributes);
+ *output = getOutputForDevice(device, session, *stream,
+ samplingRate, format, channelMask,
+ flags, offloadInfo);
+ if (*output == AUDIO_IO_HANDLE_NONE) {
+ return INVALID_OPERATION;
+ }
+ return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForDevice(
+ audio_devices_t device,
+ audio_session_t session __unused,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ uint32_t latency = 0;
+ status_t status;
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ ALOGV("getOutput() opening test output");
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags =
+ (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = mTestSamplingRate;
+ config.channel_mask = mTestChannels;
+ config.format = mTestFormat;
+ if (offloadInfo != NULL) {
+ config.offload_info = *offloadInfo;
+ }
+ status = mpClientInterface->openOutput(0,
+ &mTestOutputs[mCurOutput],
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (status == NO_ERROR) {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mFormat = config.format;
+ outputDesc->mChannelMask = config.channel_mask;
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"),mCurOutput);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+ addOutput(mTestOutputs[mCurOutput], outputDesc);
+ }
+ }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+
+ // open a direct output if required by specified parameters
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ // only allow deep buffering for music stream type
+ if (stream != AUDIO_STREAM_MUSIC) {
+ flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ }
+
+ sp<IOProfile> profile;
+
+ // skip direct output selection if the request can obviously be attached to a mixed output
+ // and not explicitly requested
+ if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
+ audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE &&
+ audio_channel_count_from_out_mask(channelMask) <= 2) {
+ goto non_direct_output;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+
+ if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+ !isNonOffloadableEffectEnabled()) {
+ profile = getProfileForDirectOutput(device,
+ samplingRate,
+ format,
+ channelMask,
+ (audio_output_flags_t)flags);
+ }
+
+ if (profile != 0) {
+ sp<AudioOutputDescriptor> outputDesc = NULL;
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open and configured with same parameters
+ if ((samplingRate == outputDesc->mSamplingRate) &&
+ (format == outputDesc->mFormat) &&
+ (channelMask == outputDesc->mChannelMask)) {
+ outputDesc->mDirectOpenCount++;
+ ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mIoHandle);
+ }
+ outputDesc = new AudioOutputDescriptor(profile);
+ outputDesc->mDevice = device;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = samplingRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+ if (offloadInfo != NULL) {
+ config.offload_info = *offloadInfo;
+ }
+ status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ // only accept an output with the requested parameters
+ if (status != NO_ERROR ||
+ (samplingRate != 0 && samplingRate != config.sample_rate) ||
+ (format != AUDIO_FORMAT_DEFAULT && format != config.format) ||
+ (channelMask != 0 && channelMask != config.channel_mask)) {
+ ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+ "format %d %d, channelMask %04x %04x", output, samplingRate,
+ outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+ outputDesc->mChannelMask);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ mpClientInterface->closeOutput(output);
+ }
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
+ return AUDIO_IO_HANDLE_NONE;
+ }
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+ outputDesc->mRefCount[stream] = 0;
+ outputDesc->mStopTime[stream] = 0;
+ outputDesc->mDirectOpenCount = 1;
+
+ audio_io_handle_t srcOutput = getOutputForEffect();
+ addOutput(output, outputDesc);
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput == output) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+ }
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutput() returns new direct output %d", output);
+ mpClientInterface->onAudioPortListUpdate();
+ return output;
+ }
+
+non_direct_output:
+
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // for non direct outputs, only PCM is supported
+ if (audio_is_linear_pcm(format)) {
+ // get which output is suitable for the specified stream. The actual
+ // routing change will happen when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+ // at this stage we should ignore the DIRECT flag as no direct output could be found earlier
+ flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+ output = selectOutput(outputs, flags, format);
+ }
+ ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+ "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+ ALOGV("getOutput() returns output %d", output);
+
+ return output;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ audio_output_flags_t flags,
+ audio_format_t format)
+{
+ // select one output among several that provide a path to a particular device or set of
+ // devices (the list was previously build by getOutputsForDevice()).
+ // The priority is as follows:
+ // 1: the output with the highest number of requested policy flags
+ // 2: the primary output
+ // 3: the first output in the list
+
+ if (outputs.size() == 0) {
+ return 0;
+ }
+ if (outputs.size() == 1) {
+ return outputs[0];
+ }
+
+ int maxCommonFlags = 0;
+ audio_io_handle_t outputFlags = 0;
+ audio_io_handle_t outputPrimary = 0;
+
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
+ if (!outputDesc->isDuplicated()) {
+ // if a valid format is specified, skip output if not compatible
+ if (format != AUDIO_FORMAT_INVALID) {
+ if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (format != outputDesc->mFormat) {
+ continue;
+ }
+ } else if (!audio_is_linear_pcm(format)) {
+ continue;
+ }
+ }
+
+ int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
+ if (commonFlags > maxCommonFlags) {
+ outputFlags = outputs[i];
+ maxCommonFlags = commonFlags;
+ ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
+ }
+ if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ outputPrimary = outputs[i];
+ }
+ }
+ }
+
+ if (outputFlags != 0) {
+ return outputFlags;
+ }
+ if (outputPrimary != 0) {
+ return outputPrimary;
+ }
+
+ return outputs[0];
+}
+
+status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session)
+{
+ ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("startOutput() unknown output %d", output);
+ return BAD_VALUE;
+ }
+
+ // cannot start playback of STREAM_TTS if any other output is being used
+ uint32_t beaconMuteLatency = 0;
+ if (stream == AUDIO_STREAM_TTS) {
+ ALOGV("\t found BEACON stream");
+ if (isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
+ return INVALID_OPERATION;
+ } else {
+ beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
+ }
+ } else {
+ // some playback other than beacon starts
+ beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
+ }
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+
+ // increment usage count for this stream on the requested output:
+ // NOTE that the usage count is the same for duplicated output and hardware output which is
+ // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->changeRefCount(stream, 1);
+
+ if (outputDesc->mRefCount[stream] == 1) {
+ // starting an output being rerouted?
+ audio_devices_t newDevice;
+ if (outputDesc->mPolicyMix != NULL) {
+ newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ } else {
+ newDevice = getNewOutputDevice(output, false /*fromCache*/);
+ }
+ routing_strategy strategy = getStrategy(stream);
+ bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+ (strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
+ (beaconMuteLatency > 0);
+ uint32_t waitMs = beaconMuteLatency;
+ bool force = false;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != outputDesc) {
+ // force a device change if any other output is managed by the same hw
+ // module and has a current device selection that differs from selected device.
+ // In this case, the audio HAL must receive the new device selection so that it can
+ // change the device currently selected by the other active output.
+ if (outputDesc->sharesHwModuleWith(desc) &&
+ desc->device() != newDevice) {
+ force = true;
+ }
+ // wait for audio on other active outputs to be presented when starting
+ // a notification so that audio focus effect can propagate, or that a mute/unmute
+ // event occurred for beacon
+ uint32_t latency = desc->latency();
+ if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+ waitMs = latency;
+ }
+ }
+ }
+ uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false);
+ }
+
+ // apply volume rules for current stream and device if necessary
+ checkAndSetVolume(stream,
+ mStreams[stream].getVolumeIndex(newDevice),
+ output,
+ newDevice);
+
+ // update the outputs if starting an output with a stream that can affect notification
+ // routing
+ handleNotificationRoutingForStream(stream);
+
+ // Automatically enable the remote submix input when output is started on a re routing mix
+ // of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL &&
+ outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ outputDesc->mPolicyMix->mRegistrationId,
+ "remote-submix");
+ }
+
+ // force reevaluating accessibility routing when ringtone or alarm starts
+ if (strategy == STRATEGY_SONIFICATION) {
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ if (waitMs > muteWaitMs) {
+ usleep((waitMs - muteWaitMs) * 2 * 1000);
+ }
+ }
+ return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session)
+{
+ ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("stopOutput() unknown output %d", output);
+ return BAD_VALUE;
+ }
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+
+ // always handle stream stop, check which stream type is stopping
+ handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, false, false);
+ }
+
+ if (outputDesc->mRefCount[stream] > 0) {
+ // decrement usage count of this stream on the output
+ outputDesc->changeRefCount(stream, -1);
+ // store time at which the stream was stopped - see isStreamActive()
+ if (outputDesc->mRefCount[stream] == 0) {
+ // Automatically disable the remote submix input when output is stopped on a
+ // re routing mix of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(outputDesc->mDevice) &&
+ outputDesc->mPolicyMix != NULL &&
+ outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ outputDesc->mPolicyMix->mRegistrationId,
+ "remote-submix");
+ }
+
+ outputDesc->mStopTime[stream] = systemTime();
+ audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
+ // delay the device switch by twice the latency because stopOutput() is executed when
+ // the track stop() command is received and at that time the audio track buffer can
+ // still contain data that needs to be drained. The latency only covers the audio HAL
+ // and kernel buffers. Also the latency does not always include additional delay in the
+ // audio path (audio DSP, CODEC ...)
+ setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+
+ // force restoring the device selection on other active outputs if it differs from the
+ // one being selected for this output
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (curOutput != output &&
+ desc->isActive() &&
+ outputDesc->sharesHwModuleWith(desc) &&
+ (newDevice != desc->device())) {
+ setOutputDevice(curOutput,
+ getNewOutputDevice(curOutput, false /*fromCache*/),
+ true,
+ outputDesc->mLatency*2);
+ }
+ }
+ // update the outputs if stopping one with a stream that can affect notification routing
+ handleNotificationRoutingForStream(stream);
+ }
+ return NO_ERROR;
+ } else {
+ ALOGW("stopOutput() refcount is already 0 for output %d", output);
+ return INVALID_OPERATION;
+ }
+}
+
+void AudioPolicyManager::releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream __unused,
+ audio_session_t session __unused)
+{
+ ALOGV("releaseOutput() %d", output);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("releaseOutput() releasing unknown output %d", output);
+ return;
+ }
+
+#ifdef AUDIO_POLICY_TEST
+ int testIndex = testOutputIndex(output);
+ if (testIndex != 0) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->isActive()) {
+ mpClientInterface->closeOutput(output);
+ mOutputs.removeItem(output);
+ mTestOutputs[testIndex] = 0;
+ }
+ return;
+ }
+#endif //AUDIO_POLICY_TEST
+
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index);
+ if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (desc->mDirectOpenCount <= 0) {
+ ALOGW("releaseOutput() invalid open count %d for output %d",
+ desc->mDirectOpenCount, output);
+ return;
+ }
+ if (--desc->mDirectOpenCount == 0) {
+ closeOutput(output);
+ // If effects where present on the output, audioflinger moved them to the primary
+ // output by default: move them back to the appropriate output.
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput != mPrimaryOutput) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
+ }
+ mpClientInterface->onAudioPortListUpdate();
+ }
+ }
+}
+
+
+status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ input_type_t *inputType)
+{
+ ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x,"
+ "session %d, flags %#x",
+ attr->source, samplingRate, format, channelMask, session, flags);
+
+ *input = AUDIO_IO_HANDLE_NONE;
+ *inputType = API_INPUT_INVALID;
+ audio_devices_t device;
+ // handle legacy remote submix case where the address was not always specified
+ String8 address = String8("");
+ bool isSoundTrigger = false;
+ audio_source_t inputSource = attr->source;
+ audio_source_t halInputSource;
+ AudioMix *policyMix = NULL;
+
+ if (inputSource == AUDIO_SOURCE_DEFAULT) {
+ inputSource = AUDIO_SOURCE_MIC;
+ }
+ halInputSource = inputSource;
+
+ if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
+ strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ address = String8(attr->tags + strlen("addr="));
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index < 0) {
+ ALOGW("getInputForAttr() no policy for address %s", address.string());
+ return BAD_VALUE;
+ }
+ if (mPolicyMixes[index]->mMix.mMixType != MIX_TYPE_PLAYERS) {
+ ALOGW("getInputForAttr() bad policy mix type for address %s", address.string());
+ return BAD_VALUE;
+ }
+ policyMix = &mPolicyMixes[index]->mMix;
+ *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
+ } else {
+ device = getDeviceAndMixForInputSource(inputSource, &policyMix);
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGW("getInputForAttr() could not find device for source %d", inputSource);
+ return BAD_VALUE;
+ }
+ if (policyMix != NULL) {
+ address = policyMix->mRegistrationId;
+ if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
+ // there is an external policy, but this input is attached to a mix of recorders,
+ // meaning it receives audio injected into the framework, so the recorder doesn't
+ // know about it and is therefore considered "legacy"
+ *inputType = API_INPUT_LEGACY;
+ } else {
+ // recording a mix of players defined by an external policy, we're rerouting for
+ // an external policy
+ *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
+ }
+ } else if (audio_is_remote_submix_device(device)) {
+ address = String8("0");
+ *inputType = API_INPUT_MIX_CAPTURE;
+ } else {
+ *inputType = API_INPUT_LEGACY;
+ }
+ // adapt channel selection to input source
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_CALL:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ default:
+ break;
+ }
+ if (inputSource == AUDIO_SOURCE_HOTWORD) {
+ ssize_t index = mSoundTriggerSessions.indexOfKey(session);
+ if (index >= 0) {
+ *input = mSoundTriggerSessions.valueFor(session);
+ isSoundTrigger = true;
+ flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
+ ALOGV("SoundTrigger capture on session %d input %d", session, *input);
+ } else {
+ halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
+ }
+ }
+ }
+
+ sp<IOProfile> profile = getInputProfile(device, address,
+ samplingRate, format, channelMask,
+ flags);
+ if (profile == 0) {
+ //retry without flags
+ audio_input_flags_t log_flags = flags;
+ flags = AUDIO_INPUT_FLAG_NONE;
+ profile = getInputProfile(device, address,
+ samplingRate, format, channelMask,
+ flags);
+ if (profile == 0) {
+ ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u,"
+ "format %#x, channelMask 0x%X, flags %#x",
+ device, samplingRate, format, channelMask, log_flags);
+ return BAD_VALUE;
+ }
+ }
+
+ if (profile->mModule->mHandle == 0) {
+ ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName);
+ return NO_INIT;
+ }
+
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = samplingRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+
+ status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
+ input,
+ &config,
+ &device,
+ address,
+ halInputSource,
+ flags);
+
+ // only accept input with the exact requested set of parameters
+ if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE ||
+ (samplingRate != config.sample_rate) ||
+ (format != config.format) ||
+ (channelMask != config.channel_mask)) {
+ ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x",
+ samplingRate, format, channelMask);
+ if (*input != AUDIO_IO_HANDLE_NONE) {
+ mpClientInterface->closeInput(*input);
+ }
+ return BAD_VALUE;
+ }
+
+ sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
+ inputDesc->mInputSource = inputSource;
+ inputDesc->mRefCount = 0;
+ inputDesc->mOpenRefCount = 1;
+ inputDesc->mSamplingRate = samplingRate;
+ inputDesc->mFormat = format;
+ inputDesc->mChannelMask = channelMask;
+ inputDesc->mDevice = device;
+ inputDesc->mSessions.add(session);
+ inputDesc->mIsSoundTrigger = isSoundTrigger;
+ inputDesc->mPolicyMix = policyMix;
+
+ ALOGV("getInputForAttr() returns input type = %d", inputType);
+
+ addInput(*input, inputDesc);
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::startInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("startInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("startInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("startInput() unknown session %d on input %d", session, input);
+ return BAD_VALUE;
+ }
+
+ // virtual input devices are compatible with other input devices
+ if (!isVirtualInputDevice(inputDesc->mDevice)) {
+
+ // for a non-virtual input device, check if there is another (non-virtual) active input
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0 && activeInput != input) {
+
+ // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
+ // otherwise the active input continues and the new input cannot be started.
+ sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
+ if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
+ ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
+ stopInput(activeInput, activeDesc->mSessions.itemAt(0));
+ releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
+ } else {
+ ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
+ return INVALID_OPERATION;
+ }
+ }
+ }
+
+ if (inputDesc->mRefCount == 0) {
+ if (activeInputsCount() == 0) {
+ SoundTrigger::setCaptureState(true);
+ }
+ setInputDevice(input, getNewInputDevice(input), true /* force */);
+
+ // automatically enable the remote submix output when input is started if not
+ // used by a policy mix of type MIX_TYPE_RECORDERS
+ // For remote submix (a virtual device), we open only one input per capture request.
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ String8 address = String8("");
+ if (inputDesc->mPolicyMix == NULL) {
+ address = String8("0");
+ } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
+ address = inputDesc->mPolicyMix->mRegistrationId;
+ }
+ if (address != "") {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address, "remote-submix");
+ }
+ }
+ }
+
+ ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
+
+ inputDesc->mRefCount++;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::stopInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("stopInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("stopInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("stopInput() unknown session %d on input %d", session, input);
+ return BAD_VALUE;
+ }
+
+ if (inputDesc->mRefCount == 0) {
+ ALOGW("stopInput() input %d already stopped", input);
+ return INVALID_OPERATION;
+ }
+
+ inputDesc->mRefCount--;
+ if (inputDesc->mRefCount == 0) {
+
+ // automatically disable the remote submix output when input is stopped if not
+ // used by a policy mix of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ String8 address = String8("");
+ if (inputDesc->mPolicyMix == NULL) {
+ address = String8("0");
+ } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
+ address = inputDesc->mPolicyMix->mRegistrationId;
+ }
+ if (address != "") {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address, "remote-submix");
+ }
+ }
+
+ resetInputDevice(input);
+
+ if (activeInputsCount() == 0) {
+ SoundTrigger::setCaptureState(false);
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::releaseInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("releaseInput() %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("releaseInput() releasing unknown input %d", input);
+ return;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+ ALOG_ASSERT(inputDesc != 0);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("releaseInput() unknown session %d on input %d", session, input);
+ return;
+ }
+ inputDesc->mSessions.remove(session);
+ if (inputDesc->mOpenRefCount == 0) {
+ ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount);
+ return;
+ }
+ inputDesc->mOpenRefCount--;
+ if (inputDesc->mOpenRefCount > 0) {
+ ALOGV("releaseInput() exit > 0");
+ return;
+ }
+
+ closeInput(input);
+ mpClientInterface->onAudioPortListUpdate();
+ ALOGV("releaseInput() exit");
+}
+
+void AudioPolicyManager::closeAllInputs() {
+ bool patchRemoved = false;
+
+ for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index);
+ ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (patch_index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(patch_index);
+ patchRemoved = true;
+ }
+ mpClientInterface->closeInput(mInputs.keyAt(input_index));
+ }
+ mInputs.clear();
+ nextAudioPortGeneration();
+
+ if (patchRemoved) {
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+}
+
+void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax)
+{
+ ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+ if (indexMin < 0 || indexMin >= indexMax) {
+ ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
+ return;
+ }
+ mStreams[stream].mIndexMin = indexMin;
+ mStreams[stream].mIndexMax = indexMax;
+ //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
+ if (stream == AUDIO_STREAM_MUSIC) {
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMin = indexMin;
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMax = indexMax;
+ }
+}
+
+status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
+{
+
+ if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+
+ // Force max volume if stream cannot be muted
+ if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
+
+ ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
+ stream, device, index);
+
+ // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
+ // clear all device specific values
+ if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+ mStreams[stream].mIndexCur.clear();
+ }
+ mStreams[stream].mIndexCur.add(device, index);
+
+ // update volume on all outputs whose current device is also selected by the same
+ // strategy as the device specified by the caller
+ audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+
+
+ //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
+ audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE;
+ if (stream == AUDIO_STREAM_MUSIC) {
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexCur.add(device, index);
+ accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/);
+ }
+ if ((device != AUDIO_DEVICE_OUT_DEFAULT) &&
+ (device & (strategyDevice | accessibilityDevice)) == 0) {
+ return NO_ERROR;
+ }
+ status_t status = NO_ERROR;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_devices_t curDevice =
+ ApmGains::getDeviceForVolume(mOutputs.valueAt(i)->device());
+ if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
+ status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
+ if (volStatus != NO_ERROR) {
+ status = volStatus;
+ }
+ }
+ if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) {
+ status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY,
+ index, mOutputs.keyAt(i), curDevice);
+ }
+ }
+ return status;
+}
+
+status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
+{
+ if (index == NULL) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+ // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
+ // the strategy the stream belongs to.
+ if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+ device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+ }
+ device = ApmGains::getDeviceForVolume(device);
+
+ *index = mStreams[stream].getVolumeIndex(device);
+ ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
+ return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
+ const SortedVector<audio_io_handle_t>& outputs)
+{
+ // select one output among several suitable for global effects.
+ // The priority is as follows:
+ // 1: An offloaded output. If the effect ends up not being offloadable,
+ // AudioFlinger will invalidate the track and the offloaded output
+ // will be closed causing the effect to be moved to a PCM output.
+ // 2: A deep buffer output
+ // 3: the first output in the list
+
+ if (outputs.size() == 0) {
+ return 0;
+ }
+
+ audio_io_handle_t outputOffloaded = 0;
+ audio_io_handle_t outputDeepBuffer = 0;
+
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+ ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
+ if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ outputOffloaded = outputs[i];
+ }
+ if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
+ outputDeepBuffer = outputs[i];
+ }
+ }
+
+ ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
+ outputOffloaded, outputDeepBuffer);
+ if (outputOffloaded != 0) {
+ return outputOffloaded;
+ }
+ if (outputDeepBuffer != 0) {
+ return outputDeepBuffer;
+ }
+
+ return outputs[0];
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
+{
+ // apply simple rule where global effects are attached to the same output as MUSIC streams
+
+ routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
+
+ audio_io_handle_t output = selectOutputForEffects(dstOutputs);
+ ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
+ output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
+
+ return output;
+}
+
+status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id)
+{
+ ssize_t index = mOutputs.indexOfKey(io);
+ if (index < 0) {
+ index = mInputs.indexOfKey(io);
+ if (index < 0) {
+ ALOGW("registerEffect() unknown io %d", io);
+ return INVALID_OPERATION;
+ }
+ }
+
+ if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
+ ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
+ desc->name, desc->memoryUsage);
+ return INVALID_OPERATION;
+ }
+ mTotalEffectsMemory += desc->memoryUsage;
+ ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
+ desc->name, io, strategy, session, id);
+ ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
+
+ sp<EffectDescriptor> effectDesc = new EffectDescriptor();
+ memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t));
+ effectDesc->mIo = io;
+ effectDesc->mStrategy = (routing_strategy)strategy;
+ effectDesc->mSession = session;
+ effectDesc->mEnabled = false;
+
+ mEffects.add(id, effectDesc);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::unregisterEffect(int id)
+{
+ ssize_t index = mEffects.indexOfKey(id);
+ if (index < 0) {
+ ALOGW("unregisterEffect() unknown effect ID %d", id);
+ return INVALID_OPERATION;
+ }
+
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(index);
+
+ setEffectEnabled(effectDesc, false);
+
+ if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) {
+ ALOGW("unregisterEffect() memory %d too big for total %d",
+ effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+ effectDesc->mDesc.memoryUsage = mTotalEffectsMemory;
+ }
+ mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage;
+ ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
+ effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+
+ mEffects.removeItem(id);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled)
+{
+ ssize_t index = mEffects.indexOfKey(id);
+ if (index < 0) {
+ ALOGW("unregisterEffect() unknown effect ID %d", id);
+ return INVALID_OPERATION;
+ }
+
+ return setEffectEnabled(mEffects.valueAt(index), enabled);
+}
+
+status_t AudioPolicyManager::setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled)
+{
+ if (enabled == effectDesc->mEnabled) {
+ ALOGV("setEffectEnabled(%s) effect already %s",
+ enabled?"true":"false", enabled?"enabled":"disabled");
+ return INVALID_OPERATION;
+ }
+
+ if (enabled) {
+ if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
+ ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
+ effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10);
+ return INVALID_OPERATION;
+ }
+ mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad;
+ ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
+ } else {
+ if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) {
+ ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
+ effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
+ effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
+ }
+ mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad;
+ ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
+ }
+ effectDesc->mEnabled = enabled;
+ return NO_ERROR;
+}
+
+bool AudioPolicyManager::isNonOffloadableEffectEnabled()
+{
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
+ if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) &&
+ ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
+ ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
+ effectDesc->mDesc.name, effectDesc->mSession);
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream,
+ uint32_t inPastMs) const
+{
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+ outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+ // do not consider re routing (when the output is going to a dynamic policy)
+ // as "remote playback"
+ if (outputDesc->mPolicyMix == NULL) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isSourceActive(audio_source_t source) const
+{
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
+ if (inputDescriptor->mRefCount == 0) {
+ continue;
+ }
+ if (inputDescriptor->mInputSource == (int)source) {
+ return true;
+ }
+ // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it
+ // corresponds to an active capture triggered by a hardware hotword recognition
+ if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) &&
+ (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) {
+ // FIXME: we should not assume that the first session is the active one and keep
+ // activity count per session. Same in startInput().
+ ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0));
+ if (index >= 0) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+// Register a list of custom mixes with their attributes and format.
+// When a mix is registered, corresponding input and output profiles are
+// added to the remote submix hw module. The profile contains only the
+// parameters (sampling rate, format...) specified by the mix.
+// The corresponding input remote submix device is also connected.
+//
+// When a remote submix device is connected, the address is checked to select the
+// appropriate profile and the corresponding input or output stream is opened.
+//
+// When capture starts, getInputForAttr() will:
+// - 1 look for a mix matching the address passed in attribtutes tags if any
+// - 2 if none found, getDeviceForInputSource() will:
+// - 2.1 look for a mix matching the attributes source
+// - 2.2 if none found, default to device selection by policy rules
+// At this time, the corresponding output remote submix device is also connected
+// and active playback use cases can be transferred to this mix if needed when reconnecting
+// after AudioTracks are invalidated
+//
+// When playback starts, getOutputForAttr() will:
+// - 1 look for a mix matching the address passed in attribtutes tags if any
+// - 2 if none found, look for a mix matching the attributes usage
+// - 3 if none found, default to device and output selection by policy rules.
+
+status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes)
+{
+ sp<HwModule> module;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
+ mHwModules[i]->mHandle != 0) {
+ module = mHwModules[i];
+ break;
+ }
+ }
+
+ if (module == 0) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("registerPolicyMixes() num mixes %d", mixes.size());
+
+ for (size_t i = 0; i < mixes.size(); i++) {
+ String8 address = mixes[i].mRegistrationId;
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index >= 0) {
+ ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string());
+ continue;
+ }
+ audio_config_t outputConfig = mixes[i].mFormat;
+ audio_config_t inputConfig = mixes[i].mFormat;
+ // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
+ // stereo and let audio flinger do the channel conversion if needed.
+ outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ module->addOutputProfile(address, &outputConfig,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
+ module->addInputProfile(address, &inputConfig,
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
+ sp<AudioPolicyMix> policyMix = new AudioPolicyMix();
+ policyMix->mMix = mixes[i];
+ mPolicyMixes.add(address, policyMix);
+ if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.string(), "remote-submix");
+ } else {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.string(), "remote-submix");
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes)
+{
+ sp<HwModule> module;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
+ mHwModules[i]->mHandle != 0) {
+ module = mHwModules[i];
+ break;
+ }
+ }
+
+ if (module == 0) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size());
+
+ for (size_t i = 0; i < mixes.size(); i++) {
+ String8 address = mixes[i].mRegistrationId;
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index < 0) {
+ ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string());
+ continue;
+ }
+
+ mPolicyMixes.removeItemsAt(index);
+
+ if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) ==
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
+ {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.string(), "remote-submix");
+ }
+
+ if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
+ {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.string(), "remote-submix");
+ }
+ module->removeOutputProfile(address);
+ module->removeInputProfile(address);
+ }
+ return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
+ result.append(buffer);
+
+ snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for communications %d\n",
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n",
+ mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO]);
+ result.append(buffer);
+
+ snprintf(buffer, SIZE, " Available output devices:\n");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
+ mAvailableOutputDevices[i]->dump(fd, 2, i);
+ }
+ snprintf(buffer, SIZE, "\n Available input devices:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
+ mAvailableInputDevices[i]->dump(fd, 2, i);
+ }
+
+ snprintf(buffer, SIZE, "\nHW Modules dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1);
+ write(fd, buffer, strlen(buffer));
+ mHwModules[i]->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nOutputs dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mOutputs.valueAt(i)->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nInputs dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mInputs.valueAt(i)->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nStreams dump:\n");
+ write(fd, buffer, strlen(buffer));
+ snprintf(buffer, SIZE,
+ " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) {
+ snprintf(buffer, SIZE, " %02zu ", i);
+ write(fd, buffer, strlen(buffer));
+ mStreams[i].dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
+ (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
+ write(fd, buffer, strlen(buffer));
+
+ snprintf(buffer, SIZE, "Registered effects:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mEffects.valueAt(i)->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nAudio Patches:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mAudioPatches.size(); i++) {
+ mAudioPatches[i]->dump(fd, 2, i);
+ }
+
+ return NO_ERROR;
+}
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+ ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+ " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
+ offloadInfo.sample_rate, offloadInfo.channel_mask,
+ offloadInfo.format,
+ offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+ offloadInfo.has_video);
+
+ // Check if offload has been disabled
+ char propValue[PROPERTY_VALUE_MAX];
+ if (property_get("audio.offload.disable", propValue, "0")) {
+ if (atoi(propValue) != 0) {
+ ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+ return false;
+ }
+ }
+
+ // Check if stream type is music, then only allow offload as of now.
+ if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+ {
+ ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+ return false;
+ }
+
+ //TODO: enable audio offloading with video when ready
+ if (offloadInfo.has_video)
+ {
+ ALOGV("isOffloadSupported: has_video == true, returning false");
+ return false;
+ }
+
+ //If duration is less than minimum value defined in property, return false
+ if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+ if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+ ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+ return false;
+ }
+ } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+ ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+ return false;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if (isNonOffloadableEffectEnabled()) {
+ return false;
+ }
+
+ // See if there is a profile to support this.
+ // AUDIO_DEVICE_NONE
+ sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ offloadInfo.sample_rate,
+ offloadInfo.format,
+ offloadInfo.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
+ return (profile != 0);
+}
+
+status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role,
+ audio_port_type_t type,
+ unsigned int *num_ports,
+ struct audio_port *ports,
+ unsigned int *generation)
+{
+ if (num_ports == NULL || (*num_ports != 0 && ports == NULL) ||
+ generation == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports);
+ if (ports == NULL) {
+ *num_ports = 0;
+ }
+
+ size_t portsWritten = 0;
+ size_t portsMax = *num_ports;
+ *num_ports = 0;
+ if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) {
+ if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0;
+ i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) {
+ mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mAvailableOutputDevices.size();
+ }
+ if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0;
+ i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) {
+ mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mAvailableInputDevices.size();
+ }
+ }
+ if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) {
+ if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) {
+ mInputs[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mInputs.size();
+ }
+ if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
+ size_t numOutputs = 0;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ if (!mOutputs[i]->isDuplicated()) {
+ numOutputs++;
+ if (portsWritten < portsMax) {
+ mOutputs[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ }
+ }
+ *num_ports += numOutputs;
+ }
+ }
+ *generation = curAudioPortGeneration();
+ ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused)
+{
+ return NO_ERROR;
+}
+
+sp<AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
+ audio_port_handle_t id) const
+{
+ sp<AudioOutputDescriptor> outputDesc = NULL;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ outputDesc = mOutputs.valueAt(i);
+ if (outputDesc->mId == id) {
+ break;
+ }
+ }
+ return outputDesc;
+}
+
+sp<AudioInputDescriptor> AudioPolicyManager::getInputFromId(
+ audio_port_handle_t id) const
+{
+ sp<AudioInputDescriptor> inputDesc = NULL;
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ inputDesc = mInputs.valueAt(i);
+ if (inputDesc->mId == id) {
+ break;
+ }
+ }
+ return inputDesc;
+}
+
+sp <HwModule> AudioPolicyManager::getModuleForDevice(
+ audio_devices_t device) const
+{
+ sp <HwModule> module;
+
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ if (audio_is_output_device(device)) {
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) {
+ return mHwModules[i];
+ }
+ }
+ } else {
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
+ if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() &
+ device & ~AUDIO_DEVICE_BIT_IN) {
+ return mHwModules[i];
+ }
+ }
+ }
+ }
+ return module;
+}
+
+sp <HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
+{
+ sp <HwModule> module;
+
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (strcmp(mHwModules[i]->mName, name) == 0) {
+ return mHwModules[i];
+ }
+ }
+ return module;
+}
+
+audio_devices_t AudioPolicyManager::availablePrimaryOutputDevices()
+{
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types();
+ return devices & mAvailableOutputDevices.types();
+}
+
+audio_devices_t AudioPolicyManager::availablePrimaryInputDevices()
+{
+ audio_module_handle_t primaryHandle =
+ mOutputs.valueFor(mPrimaryOutput)->mProfile->mModule->mHandle;
+ audio_devices_t devices = AUDIO_DEVICE_NONE;
+ for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
+ if (mAvailableInputDevices[i]->mModule->mHandle == primaryHandle) {
+ devices |= mAvailableInputDevices[i]->mDeviceType;
+ }
+ }
+ return devices;
+}
+
+status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ uid_t uid)
+{
+ ALOGV("createAudioPatch()");
+
+ if (handle == NULL || patch == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
+
+ if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
+ patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
+ return BAD_VALUE;
+ }
+ // only one source per audio patch supported for now
+ if (patch->num_sources > 1) {
+ return INVALID_OPERATION;
+ }
+
+ if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) {
+ return INVALID_OPERATION;
+ }
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) {
+ return INVALID_OPERATION;
+ }
+ }
+
+ sp<AudioPatch> patchDesc;
+ ssize_t index = mAudioPatches.indexOfKey(*handle);
+
+ ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
+ patch->sources[0].role,
+ patch->sources[0].type);
+#if LOG_NDEBUG == 0
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ ALOGV("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id,
+ patch->sinks[i].role,
+ patch->sinks[i].type);
+ }
+#endif
+
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
+ mUidCached, patchDesc->mUid, uid);
+ if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+ return INVALID_OPERATION;
+ }
+ } else {
+ *handle = 0;
+ }
+
+ if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
+ if (outputDesc == NULL) {
+ ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
+ return BAD_VALUE;
+ }
+ ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
+ outputDesc->mIoHandle);
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
+ ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
+ patchDesc->mPatch.sources[0].id, patch->sources[0].id);
+ return BAD_VALUE;
+ }
+ }
+ DeviceVector devices;
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ // Only support mix to devices connection
+ // TODO add support for mix to mix connection
+ if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
+ ALOGV("createAudioPatch() source mix but sink is not a device");
+ return INVALID_OPERATION;
+ }
+ sp<DeviceDescriptor> devDesc =
+ mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
+ if (devDesc == 0) {
+ ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id);
+ return BAD_VALUE;
+ }
+
+ if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
+ devDesc->mAddress,
+ patch->sources[0].sample_rate,
+ NULL, // updatedSamplingRate
+ patch->sources[0].format,
+ patch->sources[0].channel_mask,
+ AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
+ ALOGV("createAudioPatch() profile not supported for device %08x",
+ devDesc->mDeviceType);
+ return INVALID_OPERATION;
+ }
+ devices.add(devDesc);
+ }
+ if (devices.size() == 0) {
+ return INVALID_OPERATION;
+ }
+
+ // TODO: reconfigure output format and channels here
+ ALOGV("createAudioPatch() setting device %08x on output %d",
+ devices.types(), outputDesc->mIoHandle);
+ setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle);
+ index = mAudioPatches.indexOfKey(*handle);
+ if (index >= 0) {
+ if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
+ ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
+ }
+ patchDesc = mAudioPatches.valueAt(index);
+ patchDesc->mUid = uid;
+ ALOGV("createAudioPatch() success");
+ } else {
+ ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
+ return INVALID_OPERATION;
+ }
+ } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+ // input device to input mix connection
+ // only one sink supported when connecting an input device to a mix
+ if (patch->num_sinks > 1) {
+ return INVALID_OPERATION;
+ }
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
+ if (inputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) {
+ return BAD_VALUE;
+ }
+ }
+ sp<DeviceDescriptor> devDesc =
+ mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
+ if (devDesc == 0) {
+ return BAD_VALUE;
+ }
+
+ if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
+ devDesc->mAddress,
+ patch->sinks[0].sample_rate,
+ NULL, /*updatedSampleRate*/
+ patch->sinks[0].format,
+ patch->sinks[0].channel_mask,
+ // FIXME for the parameter type,
+ // and the NONE
+ (audio_output_flags_t)
+ AUDIO_INPUT_FLAG_NONE)) {
+ return INVALID_OPERATION;
+ }
+ // TODO: reconfigure output format and channels here
+ ALOGV("createAudioPatch() setting device %08x on output %d",
+ devDesc->mDeviceType, inputDesc->mIoHandle);
+ setInputDevice(inputDesc->mIoHandle, devDesc->mDeviceType, true, handle);
+ index = mAudioPatches.indexOfKey(*handle);
+ if (index >= 0) {
+ if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
+ ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
+ }
+ patchDesc = mAudioPatches.valueAt(index);
+ patchDesc->mUid = uid;
+ ALOGV("createAudioPatch() success");
+ } else {
+ ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
+ return INVALID_OPERATION;
+ }
+ } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ // device to device connection
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
+ return BAD_VALUE;
+ }
+ }
+ sp<DeviceDescriptor> srcDeviceDesc =
+ mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
+ if (srcDeviceDesc == 0) {
+ return BAD_VALUE;
+ }
+
+ //update source and sink with our own data as the data passed in the patch may
+ // be incomplete.
+ struct audio_patch newPatch = *patch;
+ srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
+
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
+ ALOGV("createAudioPatch() source device but one sink is not a device");
+ return INVALID_OPERATION;
+ }
+
+ sp<DeviceDescriptor> sinkDeviceDesc =
+ mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
+ if (sinkDeviceDesc == 0) {
+ return BAD_VALUE;
+ }
+ sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
+
+ if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) {
+ // only one sink supported when connected devices across HW modules
+ if (patch->num_sinks > 1) {
+ return INVALID_OPERATION;
+ }
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(sinkDeviceDesc->mDeviceType,
+ mOutputs);
+ // if the sink device is reachable via an opened output stream, request to go via
+ // this output stream by adding a second source to the patch description
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (outputDesc->isDuplicated()) {
+ return INVALID_OPERATION;
+ }
+ outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]);
+ newPatch.num_sources = 2;
+ }
+ }
+ }
+ // TODO: check from routing capabilities in config file and other conflicting patches
+
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&newPatch,
+ &afPatchHandle,
+ 0);
+ ALOGV("createAudioPatch() patch panel returned %d patchHandle %d",
+ status, afPatchHandle);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &newPatch, uid);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = newPatch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ *handle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ } else {
+ ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
+ status);
+ return INVALID_OPERATION;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle,
+ uid_t uid)
+{
+ ALOGV("releaseAudioPatch() patch %d", handle);
+
+ ssize_t index = mAudioPatches.indexOfKey(handle);
+
+ if (index < 0) {
+ return BAD_VALUE;
+ }
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
+ mUidCached, patchDesc->mUid, uid);
+ if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+ return INVALID_OPERATION;
+ }
+
+ struct audio_patch *patch = &patchDesc->mPatch;
+ patchDesc->mUid = mUidCached;
+ if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
+ if (outputDesc == NULL) {
+ ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
+ return BAD_VALUE;
+ }
+
+ setOutputDevice(outputDesc->mIoHandle,
+ getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/),
+ true,
+ 0,
+ NULL);
+ } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
+ if (inputDesc == NULL) {
+ ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
+ return BAD_VALUE;
+ }
+ setInputDevice(inputDesc->mIoHandle,
+ getNewInputDevice(inputDesc->mIoHandle),
+ true,
+ NULL);
+ } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle;
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d",
+ status, patchDesc->mAfPatchHandle);
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ } else {
+ return BAD_VALUE;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches,
+ unsigned int *generation)
+{
+ if (num_patches == NULL || (*num_patches != 0 && patches == NULL) ||
+ generation == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu",
+ *num_patches, patches, mAudioPatches.size());
+ if (patches == NULL) {
+ *num_patches = 0;
+ }
+
+ size_t patchesWritten = 0;
+ size_t patchesMax = *num_patches;
+ for (size_t i = 0;
+ i < mAudioPatches.size() && patchesWritten < patchesMax; i++) {
+ patches[patchesWritten] = mAudioPatches[i]->mPatch;
+ patches[patchesWritten++].id = mAudioPatches[i]->mHandle;
+ ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
+ i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks);
+ }
+ *num_patches = mAudioPatches.size();
+
+ *generation = curAudioPortGeneration();
+ ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config)
+{
+ ALOGV("setAudioPortConfig()");
+
+ if (config == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("setAudioPortConfig() on port handle %d", config->id);
+ // Only support gain configuration for now
+ if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) {
+ return INVALID_OPERATION;
+ }
+
+ sp<AudioPortConfig> audioPortConfig;
+ if (config->type == AUDIO_PORT_TYPE_MIX) {
+ if (config->role == AUDIO_PORT_ROLE_SOURCE) {
+ sp<AudioOutputDescriptor> outputDesc = getOutputFromId(config->id);
+ if (outputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "setAudioPortConfig() called on duplicated output %d",
+ outputDesc->mIoHandle);
+ audioPortConfig = outputDesc;
+ } else if (config->role == AUDIO_PORT_ROLE_SINK) {
+ sp<AudioInputDescriptor> inputDesc = getInputFromId(config->id);
+ if (inputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ audioPortConfig = inputDesc;
+ } else {
+ return BAD_VALUE;
+ }
+ } else if (config->type == AUDIO_PORT_TYPE_DEVICE) {
+ sp<DeviceDescriptor> deviceDesc;
+ if (config->role == AUDIO_PORT_ROLE_SOURCE) {
+ deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id);
+ } else if (config->role == AUDIO_PORT_ROLE_SINK) {
+ deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id);
+ } else {
+ return BAD_VALUE;
+ }
+ if (deviceDesc == NULL) {
+ return BAD_VALUE;
+ }
+ audioPortConfig = deviceDesc;
+ } else {
+ return BAD_VALUE;
+ }
+
+ struct audio_port_config backupConfig;
+ status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig);
+ if (status == NO_ERROR) {
+ struct audio_port_config newConfig;
+ audioPortConfig->toAudioPortConfig(&newConfig, config);
+ status = mpClientInterface->setAudioPortConfig(&newConfig, 0);
+ }
+ if (status != NO_ERROR) {
+ audioPortConfig->applyAudioPortConfig(&backupConfig);
+ }
+
+ return status;
+}
+
+void AudioPolicyManager::clearAudioPatches(uid_t uid)
+{
+ for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
+ if (patchDesc->mUid == uid) {
+ releaseAudioPatch(mAudioPatches.keyAt(i), uid);
+ }
+ }
+}
+
+status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device)
+{
+ *session = (audio_session_t)mpClientInterface->newAudioUniqueId();
+ *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId();
+ *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD);
+
+ mSoundTriggerSessions.add(*session, *ioHandle);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::releaseSoundTriggerSession(audio_session_t session)
+{
+ ssize_t index = mSoundTriggerSessions.indexOfKey(session);
+ if (index < 0) {
+ ALOGW("acquireSoundTriggerSession() session %d not registered", session);
+ return BAD_VALUE;
+ }
+
+ mSoundTriggerSessions.removeItem(session);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle,
+ const sp<AudioPatch>& patch)
+{
+ ssize_t index = mAudioPatches.indexOfKey(handle);
+
+ if (index >= 0) {
+ ALOGW("addAudioPatch() patch %d already in", handle);
+ return ALREADY_EXISTS;
+ }
+ mAudioPatches.add(handle, patch);
+ ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d"
+ "sink handle %d",
+ handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks,
+ patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle)
+{
+ ssize_t index = mAudioPatches.indexOfKey(handle);
+
+ if (index < 0) {
+ ALOGW("removeAudioPatch() patch %d not in", handle);
+ return ALREADY_EXISTS;
+ }
+ ALOGV("removeAudioPatch() handle %d af handle %d", handle,
+ mAudioPatches.valueAt(index)->mAfPatchHandle);
+ mAudioPatches.removeItemsAt(index);
+ return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager
+// ----------------------------------------------------------------------------
+
+uint32_t AudioPolicyManager::nextUniqueId()
+{
+ return android_atomic_inc(&mNextUniqueId);
+}
+
+uint32_t AudioPolicyManager::nextAudioPortGeneration()
+{
+ return android_atomic_inc(&mAudioPortGeneration);
+}
+
+int32_t volatile AudioPolicyManager::mNextUniqueId = 1;
+
+AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+ :
+#ifdef AUDIO_POLICY_TEST
+ Thread(false),
+#endif //AUDIO_POLICY_TEST
+ mPrimaryOutput((audio_io_handle_t)0),
+ mPhoneState(AUDIO_MODE_NORMAL),
+ mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
+ mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
+ mA2dpSuspended(false),
+ mSpeakerDrcEnabled(false),
+ mAudioPortGeneration(1),
+ mBeaconMuteRefCount(0),
+ mBeaconPlayingRefCount(0),
+ mBeaconMuted(false)
+{
+ mUidCached = getuid();
+ mpClientInterface = clientInterface;
+
+ for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
+ mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
+ }
+
+ mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER);
+ if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
+ if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
+ ALOGE("could not load audio policy configuration file, setting defaults");
+ defaultAudioPolicyConfig();
+ }
+ }
+ // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
+
+ // must be done after reading the policy
+ initializeVolumeCurves();
+
+ // open all output streams needed to access attached devices
+ audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
+ audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
+ if (mHwModules[i]->mHandle == 0) {
+ ALOGW("could not open HW module %s", mHwModules[i]->mName);
+ continue;
+ }
+ // open all output streams needed to access attached devices
+ // except for direct output streams that are only opened when they are actually
+ // required by an app.
+ // This also validates mAvailableOutputDevices list
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
+
+ if (outProfile->mSupportedDevices.isEmpty()) {
+ ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName);
+ continue;
+ }
+
+ if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+ continue;
+ }
+ audio_devices_t profileType = outProfile->mSupportedDevices.types();
+ if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) {
+ profileType = mDefaultOutputDevice->mDeviceType;
+ } else {
+ // chose first device present in mSupportedDevices also part of
+ // outputDeviceTypes
+ for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
+ profileType = outProfile->mSupportedDevices[k]->mDeviceType;
+ if ((profileType & outputDeviceTypes) != 0) {
+ break;
+ }
+ }
+ }
+ if ((profileType & outputDeviceTypes) == 0) {
+ continue;
+ }
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile);
+
+ outputDesc->mDevice = profileType;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = outputDesc->mSamplingRate;
+ config.channel_mask = outputDesc->mChannelMask;
+ config.format = outputDesc->mFormat;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle,
+ &output,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ if (status != NO_ERROR) {
+ ALOGW("Cannot open output stream for device %08x on hw module %s",
+ outputDesc->mDevice,
+ mHwModules[i]->mName);
+ } else {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+
+ for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
+ audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType;
+ ssize_t index =
+ mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
+ // give a valid ID to an attached device once confirmed it is reachable
+ if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
+ mAvailableOutputDevices[index]->attach(mHwModules[i]);
+ }
+ }
+ if (mPrimaryOutput == 0 &&
+ outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ mPrimaryOutput = output;
+ }
+ addOutput(output, outputDesc);
+ setOutputDevice(output,
+ outputDesc->mDevice,
+ true);
+ }
+ }
+ // open input streams needed to access attached devices to validate
+ // mAvailableInputDevices list
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ {
+ const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
+
+ if (inProfile->mSupportedDevices.isEmpty()) {
+ ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName);
+ continue;
+ }
+ // chose first device present in mSupportedDevices also part of
+ // inputDeviceTypes
+ audio_devices_t profileType = AUDIO_DEVICE_NONE;
+ for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
+ profileType = inProfile->mSupportedDevices[k]->mDeviceType;
+ if (profileType & inputDeviceTypes) {
+ break;
+ }
+ }
+ if ((profileType & inputDeviceTypes) == 0) {
+ continue;
+ }
+ sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(inProfile);
+
+ inputDesc->mInputSource = AUDIO_SOURCE_MIC;
+ inputDesc->mDevice = profileType;
+
+ // find the address
+ DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType);
+ // the inputs vector must be of size 1, but we don't want to crash here
+ String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress
+ : String8("");
+ ALOGV(" for input device 0x%x using address %s", profileType, address.string());
+ ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
+
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = inputDesc->mSamplingRate;
+ config.channel_mask = inputDesc->mChannelMask;
+ config.format = inputDesc->mFormat;
+ audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle,
+ &input,
+ &config,
+ &inputDesc->mDevice,
+ address,
+ AUDIO_SOURCE_MIC,
+ AUDIO_INPUT_FLAG_NONE);
+
+ if (status == NO_ERROR) {
+ for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
+ audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType;
+ ssize_t index =
+ mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
+ // give a valid ID to an attached device once confirmed it is reachable
+ if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) {
+ mAvailableInputDevices[index]->attach(mHwModules[i]);
+ }
+ }
+ mpClientInterface->closeInput(input);
+ } else {
+ ALOGW("Cannot open input stream for device %08x on hw module %s",
+ inputDesc->mDevice,
+ mHwModules[i]->mName);
+ }
+ }
+ }
+ // make sure all attached devices have been allocated a unique ID
+ for (size_t i = 0; i < mAvailableOutputDevices.size();) {
+ if (!mAvailableOutputDevices[i]->isAttached()) {
+ ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
+ mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
+ continue;
+ }
+ i++;
+ }
+ for (size_t i = 0; i < mAvailableInputDevices.size();) {
+ if (!mAvailableInputDevices[i]->isAttached()) {
+ ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
+ mAvailableInputDevices.remove(mAvailableInputDevices[i]);
+ continue;
+ }
+ i++;
+ }
+ // make sure default device is reachable
+ if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
+ ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType);
+ }
+
+ ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
+
+ updateDevicesAndOutputs();
+
+#ifdef AUDIO_POLICY_TEST
+ if (mPrimaryOutput != 0) {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+
+ mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ mTestSamplingRate = 44100;
+ mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
+ mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
+ mTestLatencyMs = 0;
+ mCurOutput = 0;
+ mDirectOutput = false;
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ mTestOutputs[i] = 0;
+ }
+
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ snprintf(buffer, SIZE, "AudioPolicyManagerTest");
+ run(buffer, ANDROID_PRIORITY_AUDIO);
+ }
+#endif //AUDIO_POLICY_TEST
+}
+
+AudioPolicyManager::~AudioPolicyManager()
+{
+#ifdef AUDIO_POLICY_TEST
+ exit();
+#endif //AUDIO_POLICY_TEST
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ mpClientInterface->closeOutput(mOutputs.keyAt(i));
+ }
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ mpClientInterface->closeInput(mInputs.keyAt(i));
+ }
+ mAvailableOutputDevices.clear();
+ mAvailableInputDevices.clear();
+ mOutputs.clear();
+ mInputs.clear();
+ mHwModules.clear();
+}
+
+status_t AudioPolicyManager::initCheck()
+{
+ return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManager::threadLoop()
+{
+ ALOGV("entering threadLoop()");
+ while (!exitPending())
+ {
+ String8 command;
+ int valueInt;
+ String8 value;
+
+ Mutex::Autolock _l(mLock);
+ mWaitWorkCV.waitRelative(mLock, milliseconds(50));
+
+ command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
+ AudioParameter param = AudioParameter(command);
+
+ if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
+ valueInt != 0) {
+ ALOGV("Test command %s received", command.string());
+ String8 target;
+ if (param.get(String8("target"), target) != NO_ERROR) {
+ target = "Manager";
+ }
+ if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_output"));
+ mCurOutput = valueInt;
+ }
+ if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_direct"));
+ if (value == "false") {
+ mDirectOutput = false;
+ } else if (value == "true") {
+ mDirectOutput = true;
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_input"));
+ mTestInput = valueInt;
+ }
+
+ if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_format"));
+ int format = AUDIO_FORMAT_INVALID;
+ if (value == "PCM 16 bits") {
+ format = AUDIO_FORMAT_PCM_16_BIT;
+ } else if (value == "PCM 8 bits") {
+ format = AUDIO_FORMAT_PCM_8_BIT;
+ } else if (value == "Compressed MP3") {
+ format = AUDIO_FORMAT_MP3;
+ }
+ if (format != AUDIO_FORMAT_INVALID) {
+ if (target == "Manager") {
+ mTestFormat = format;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("format"), format);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+ if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_channels"));
+ int channels = 0;
+
+ if (value == "Channels Stereo") {
+ channels = AUDIO_CHANNEL_OUT_STEREO;
+ } else if (value == "Channels Mono") {
+ channels = AUDIO_CHANNEL_OUT_MONO;
+ }
+ if (channels != 0) {
+ if (target == "Manager") {
+ mTestChannels = channels;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("channels"), channels);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_sampleRate"));
+ if (valueInt >= 0 && valueInt <= 96000) {
+ int samplingRate = valueInt;
+ if (target == "Manager") {
+ mTestSamplingRate = samplingRate;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("sampling_rate"), samplingRate);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+
+ if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_reopen"));
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ mpClientInterface->closeOutput(mPrimaryOutput);
+
+ audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
+
+ mOutputs.removeItem(mPrimaryOutput);
+
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = outputDesc->mSamplingRate;
+ config.channel_mask = outputDesc->mChannelMask;
+ config.format = outputDesc->mFormat;
+ status_t status = mpClientInterface->openOutput(moduleHandle,
+ &mPrimaryOutput,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (status != NO_ERROR) {
+ ALOGE("Failed to reopen hardware output stream, "
+ "samplingRate: %d, format %d, channels %d",
+ outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
+ } else {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+ addOutput(mPrimaryOutput, outputDesc);
+ }
+ }
+
+
+ mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+ }
+ }
+ return false;
+}
+
+void AudioPolicyManager::exit()
+{
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
+{
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ if (output == mTestOutputs[i]) return i;
+ }
+ return 0;
+}
+#endif //AUDIO_POLICY_TEST
+
+// ---
+
+void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc)
+{
+ outputDesc->mIoHandle = output;
+ outputDesc->mId = nextUniqueId();
+ mOutputs.add(output, outputDesc);
+ nextAudioPortGeneration();
+}
+
+void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc)
+{
+ inputDesc->mIoHandle = input;
+ inputDesc->mId = nextUniqueId();
+ mInputs.add(input, inputDesc);
+ nextAudioPortGeneration();
+}
+
+void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+ const audio_devices_t device /*in*/,
+ const String8 address /*in*/,
+ SortedVector<audio_io_handle_t>& outputs /*out*/) {
+ sp<DeviceDescriptor> devDesc =
+ desc->mProfile->mSupportedDevices.getDevice(device, address);
+ if (devDesc != 0) {
+ ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s",
+ desc->mIoHandle, address.string());
+ outputs.add(desc->mIoHandle);
+ }
+}
+
+status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& outputs,
+ const String8 address)
+{
+ audio_devices_t device = devDesc->mDeviceType;
+ sp<AudioOutputDescriptor> desc;
+ // erase all current sample rates, formats and channel masks
+ devDesc->clearCapabilities();
+
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ // first list already open outputs that can be routed to this device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) {
+ if (!deviceDistinguishesOnAddress(device)) {
+ ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
+ outputs.add(mOutputs.keyAt(i));
+ } else {
+ ALOGV(" checking address match due to device 0x%x", device);
+ findIoHandlesByAddress(desc, device, address, outputs);
+ }
+ }
+ }
+ // then look for output profiles that can be routed to this device
+ SortedVector< sp<IOProfile> > profiles;
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ if (profile->mSupportedDevices.types() & device) {
+ if (!deviceDistinguishesOnAddress(device) ||
+ address == profile->mSupportedDevices[0]->mAddress) {
+ profiles.add(profile);
+ ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
+ }
+ }
+ }
+ }
+
+ ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size());
+
+ if (profiles.isEmpty() && outputs.isEmpty()) {
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ return BAD_VALUE;
+ }
+
+ // open outputs for matching profiles if needed. Direct outputs are also opened to
+ // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+ for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+ sp<IOProfile> profile = profiles[profile_index];
+
+ // nothing to do if one output is already opened for this profile
+ size_t j;
+ for (j = 0; j < outputs.size(); j++) {
+ desc = mOutputs.valueFor(outputs.itemAt(j));
+ if (!desc->isDuplicated() && desc->mProfile == profile) {
+ // matching profile: save the sample rates, format and channel masks supported
+ // by the profile in our device descriptor
+ devDesc->importAudioPort(profile);
+ break;
+ }
+ }
+ if (j != outputs.size()) {
+ continue;
+ }
+
+ ALOGV("opening output for device %08x with params %s profile %p",
+ device, address.string(), profile.get());
+ desc = new AudioOutputDescriptor(profile);
+ desc->mDevice = device;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = desc->mSamplingRate;
+ config.channel_mask = desc->mChannelMask;
+ config.format = desc->mFormat;
+ config.offload_info.sample_rate = desc->mSamplingRate;
+ config.offload_info.channel_mask = desc->mChannelMask;
+ config.offload_info.format = desc->mFormat;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &desc->mDevice,
+ address,
+ &desc->mLatency,
+ desc->mFlags);
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+
+ // Here is where the out_set_parameters() for card & device gets called
+ if (!address.isEmpty()) {
+ char *param = audio_device_address_to_parameter(device, address);
+ mpClientInterface->setParameters(output, String8(param));
+ free(param);
+ }
+
+ // Here is where we step through and resolve any "dynamic" fields
+ String8 reply;
+ char *value;
+ if (profile->mSamplingRates[0] == 0) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+ ALOGV("checkOutputsForDevice() supported sampling rates %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadSamplingRates(value + 1);
+ }
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+ ALOGV("checkOutputsForDevice() supported formats %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadFormats(value + 1);
+ }
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+ ALOGV("checkOutputsForDevice() supported channel masks %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadOutChannels(value + 1);
+ }
+ }
+ if (((profile->mSamplingRates[0] == 0) &&
+ (profile->mSamplingRates.size() < 2)) ||
+ ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+ (profile->mFormats.size() < 2)) ||
+ ((profile->mChannelMasks[0] == 0) &&
+ (profile->mChannelMasks.size() < 2))) {
+ ALOGW("checkOutputsForDevice() missing param");
+ mpClientInterface->closeOutput(output);
+ output = AUDIO_IO_HANDLE_NONE;
+ } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 ||
+ profile->mChannelMasks[0] == 0) {
+ mpClientInterface->closeOutput(output);
+ config.sample_rate = profile->pickSamplingRate();
+ config.channel_mask = profile->pickChannelMask();
+ config.format = profile->pickFormat();
+ config.offload_info.sample_rate = config.sample_rate;
+ config.offload_info.channel_mask = config.channel_mask;
+ config.offload_info.format = config.format;
+ status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &desc->mDevice,
+ address,
+ &desc->mLatency,
+ desc->mFlags);
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+ } else {
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ }
+
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ addOutput(output, desc);
+ if (deviceDistinguishesOnAddress(device) && address != "0") {
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index >= 0) {
+ mPolicyMixes[index]->mOutput = desc;
+ desc->mPolicyMix = &mPolicyMixes[index]->mMix;
+ } else {
+ ALOGE("checkOutputsForDevice() cannot find policy for address %s",
+ address.string());
+ }
+ } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
+ // no duplicated output for direct outputs and
+ // outputs used by dynamic policy mixes
+ audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
+
+ // set initial stream volume for device
+ applyStreamVolumes(output, device, 0, true);
+
+ //TODO: configure audio effect output stage here
+
+ // open a duplicating output thread for the new output and the primary output
+ duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
+ mPrimaryOutput);
+ if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) {
+ // add duplicated output descriptor
+ sp<AudioOutputDescriptor> dupOutputDesc =
+ new AudioOutputDescriptor(NULL);
+ dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
+ dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
+ dupOutputDesc->mSamplingRate = desc->mSamplingRate;
+ dupOutputDesc->mFormat = desc->mFormat;
+ dupOutputDesc->mChannelMask = desc->mChannelMask;
+ dupOutputDesc->mLatency = desc->mLatency;
+ addOutput(duplicatedOutput, dupOutputDesc);
+ applyStreamVolumes(duplicatedOutput, device, 0, true);
+ } else {
+ ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
+ mPrimaryOutput, output);
+ mpClientInterface->closeOutput(output);
+ mOutputs.removeItem(output);
+ nextAudioPortGeneration();
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ }
+ }
+ } else {
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ ALOGW("checkOutputsForDevice() could not open output for device %x", device);
+ profiles.removeAt(profile_index);
+ profile_index--;
+ } else {
+ outputs.add(output);
+ devDesc->importAudioPort(profile);
+
+ if (deviceDistinguishesOnAddress(device)) {
+ ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
+ device, address.string());
+ setOutputDevice(output, device, true/*force*/, 0/*delay*/,
+ NULL/*patch handle*/, address.string());
+ }
+ ALOGV("checkOutputsForDevice(): adding output %d", output);
+ }
+ }
+
+ if (profiles.isEmpty()) {
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ return BAD_VALUE;
+ }
+ } else { // Disconnect
+ // check if one opened output is not needed any more after disconnecting one device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated()) {
+ // exact match on device
+ if (deviceDistinguishesOnAddress(device) &&
+ (desc->mProfile->mSupportedDevices.types() == device)) {
+ findIoHandlesByAddress(desc, device, address, outputs);
+ } else if (!(desc->mProfile->mSupportedDevices.types()
+ & mAvailableOutputDevices.types())) {
+ ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
+ mOutputs.keyAt(i));
+ outputs.add(mOutputs.keyAt(i));
+ }
+ }
+ }
+ // Clear any profiles associated with the disconnected device.
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ if (profile->mSupportedDevices.types() & device) {
+ ALOGV("checkOutputsForDevice(): "
+ "clearing direct output profile %zu on module %zu", j, i);
+ if (profile->mSamplingRates[0] == 0) {
+ profile->mSamplingRates.clear();
+ profile->mSamplingRates.add(0);
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ profile->mFormats.clear();
+ profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ profile->mChannelMasks.clear();
+ profile->mChannelMasks.add(0);
+ }
+ }
+ }
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& inputs,
+ const String8 address)
+{
+ sp<AudioInputDescriptor> desc;
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ // first list already open inputs that can be routed to this device
+ for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
+ ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
+ inputs.add(mInputs.keyAt(input_index));
+ }
+ }
+
+ // then look for input profiles that can be routed to this device
+ SortedVector< sp<IOProfile> > profiles;
+ for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
+ {
+ if (mHwModules[module_idx]->mHandle == 0) {
+ continue;
+ }
+ for (size_t profile_index = 0;
+ profile_index < mHwModules[module_idx]->mInputProfiles.size();
+ profile_index++)
+ {
+ sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index];
+
+ if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
+ if (!deviceDistinguishesOnAddress(device) ||
+ address == profile->mSupportedDevices[0]->mAddress) {
+ profiles.add(profile);
+ ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
+ profile_index, module_idx);
+ }
+ }
+ }
+ }
+
+ if (profiles.isEmpty() && inputs.isEmpty()) {
+ ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+ return BAD_VALUE;
+ }
+
+ // open inputs for matching profiles if needed. Direct inputs are also opened to
+ // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+ for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+
+ sp<IOProfile> profile = profiles[profile_index];
+ // nothing to do if one input is already opened for this profile
+ size_t input_index;
+ for (input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (desc->mProfile == profile) {
+ break;
+ }
+ }
+ if (input_index != mInputs.size()) {
+ continue;
+ }
+
+ ALOGV("opening input for device 0x%X with params %s", device, address.string());
+ desc = new AudioInputDescriptor(profile);
+ desc->mDevice = device;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = desc->mSamplingRate;
+ config.channel_mask = desc->mChannelMask;
+ config.format = desc->mFormat;
+ audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
+ &input,
+ &config,
+ &desc->mDevice,
+ address,
+ AUDIO_SOURCE_MIC,
+ AUDIO_INPUT_FLAG_NONE /*FIXME*/);
+
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+
+ if (!address.isEmpty()) {
+ char *param = audio_device_address_to_parameter(device, address);
+ mpClientInterface->setParameters(input, String8(param));
+ free(param);
+ }
+
+ // Here is where we step through and resolve any "dynamic" fields
+ String8 reply;
+ char *value;
+ if (profile->mSamplingRates[0] == 0) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+ ALOGV("checkInputsForDevice() direct input sup sampling rates %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadSamplingRates(value + 1);
+ }
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+ ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadFormats(value + 1);
+ }
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+ ALOGV("checkInputsForDevice() direct input sup channel masks %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadInChannels(value + 1);
+ }
+ }
+ if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
+ ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) ||
+ ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) {
+ ALOGW("checkInputsForDevice() direct input missing param");
+ mpClientInterface->closeInput(input);
+ input = AUDIO_IO_HANDLE_NONE;
+ }
+
+ if (input != 0) {
+ addInput(input, desc);
+ }
+ } // endif input != 0
+
+ if (input == AUDIO_IO_HANDLE_NONE) {
+ ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
+ profiles.removeAt(profile_index);
+ profile_index--;
+ } else {
+ inputs.add(input);
+ ALOGV("checkInputsForDevice(): adding input %d", input);
+ }
+ } // end scan profiles
+
+ if (profiles.isEmpty()) {
+ ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+ return BAD_VALUE;
+ }
+ } else {
+ // Disconnect
+ // check if one opened input is not needed any more after disconnecting one device
+ for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types() &
+ ~AUDIO_DEVICE_BIT_IN)) {
+ ALOGV("checkInputsForDevice(): disconnecting adding input %d",
+ mInputs.keyAt(input_index));
+ inputs.add(mInputs.keyAt(input_index));
+ }
+ }
+ // Clear any profiles associated with the disconnected device.
+ for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) {
+ if (mHwModules[module_index]->mHandle == 0) {
+ continue;
+ }
+ for (size_t profile_index = 0;
+ profile_index < mHwModules[module_index]->mInputProfiles.size();
+ profile_index++) {
+ sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
+ if (profile->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) {
+ ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
+ profile_index, module_index);
+ if (profile->mSamplingRates[0] == 0) {
+ profile->mSamplingRates.clear();
+ profile->mSamplingRates.add(0);
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ profile->mFormats.clear();
+ profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ profile->mChannelMasks.clear();
+ profile->mChannelMasks.add(0);
+ }
+ }
+ }
+ }
+ } // end disconnect
+
+ return NO_ERROR;
+}
+
+
+void AudioPolicyManager::closeOutput(audio_io_handle_t output)
+{
+ ALOGV("closeOutput(%d)", output);
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (outputDesc == NULL) {
+ ALOGW("closeOutput() unknown output %d", output);
+ return;
+ }
+
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ if (mPolicyMixes[i]->mOutput == outputDesc) {
+ mPolicyMixes[i]->mOutput.clear();
+ }
+ }
+
+ // look for duplicated outputs connected to the output being removed.
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
+ if (dupOutputDesc->isDuplicated() &&
+ (dupOutputDesc->mOutput1 == outputDesc ||
+ dupOutputDesc->mOutput2 == outputDesc)) {
+ sp<AudioOutputDescriptor> outputDesc2;
+ if (dupOutputDesc->mOutput1 == outputDesc) {
+ outputDesc2 = dupOutputDesc->mOutput2;
+ } else {
+ outputDesc2 = dupOutputDesc->mOutput1;
+ }
+ // As all active tracks on duplicated output will be deleted,
+ // and as they were also referenced on the other output, the reference
+ // count for their stream type must be adjusted accordingly on
+ // the other output.
+ for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
+ int refCount = dupOutputDesc->mRefCount[j];
+ outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
+ }
+ audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
+ ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
+
+ mpClientInterface->closeOutput(duplicatedOutput);
+ mOutputs.removeItem(duplicatedOutput);
+ }
+ }
+
+ nextAudioPortGeneration();
+
+ ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(index);
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+
+ AudioParameter param;
+ param.add(String8("closing"), String8("true"));
+ mpClientInterface->setParameters(output, param.toString());
+
+ mpClientInterface->closeOutput(output);
+ mOutputs.removeItem(output);
+ mPreviousOutputs = mOutputs;
+}
+
+void AudioPolicyManager::closeInput(audio_io_handle_t input)
+{
+ ALOGV("closeInput(%d)", input);
+
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ if (inputDesc == NULL) {
+ ALOGW("closeInput() unknown input %d", input);
+ return;
+ }
+
+ nextAudioPortGeneration();
+
+ ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(index);
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+
+ mpClientInterface->closeInput(input);
+ mInputs.removeItem(input);
+}
+
+SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs)
+{
+ SortedVector<audio_io_handle_t> outputs;
+
+ ALOGVV("getOutputsForDevice() device %04x", device);
+ for (size_t i = 0; i < openOutputs.size(); i++) {
+ ALOGVV("output %d isDuplicated=%d device=%04x",
+ i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
+ if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
+ ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
+ outputs.add(openOutputs.keyAt(i));
+ }
+ }
+ return outputs;
+}
+
+bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2)
+{
+ if (outputs1.size() != outputs2.size()) {
+ return false;
+ }
+ for (size_t i = 0; i < outputs1.size(); i++) {
+ if (outputs1[i] != outputs2[i]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
+{
+ audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
+ audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+ // also take into account external policy-related changes: add all outputs which are
+ // associated with policies in the "before" and "after" output vectors
+ ALOGVV("checkOutputForStrategy(): policy related outputs");
+ for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
+ const sp<AudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ srcOutputs.add(desc->mIoHandle);
+ ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
+ }
+ }
+ for (size_t i = 0 ; i < mOutputs.size() ; i++) {
+ const sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ dstOutputs.add(desc->mIoHandle);
+ ALOGVV(" new outputs: adding %d", desc->mIoHandle);
+ }
+ }
+
+ if (!vectorsEqual(srcOutputs,dstOutputs)) {
+ ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
+ strategy, srcOutputs[0], dstOutputs[0]);
+ // mute strategy while moving tracks from one output to another
+ for (size_t i = 0; i < srcOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
+ if (desc->isStrategyActive(strategy)) {
+ setStrategyMute(strategy, true, srcOutputs[i]);
+ setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
+ }
+ }
+
+ // Move effects associated to this strategy from previous output to new output
+ if (strategy == STRATEGY_MEDIA) {
+ audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
+ SortedVector<audio_io_handle_t> moved;
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
+ if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
+ effectDesc->mIo != fxOutput) {
+ if (moved.indexOf(effectDesc->mIo) < 0) {
+ ALOGV("checkOutputForStrategy() moving effect %d to output %d",
+ mEffects.keyAt(i), fxOutput);
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo,
+ fxOutput);
+ moved.add(effectDesc->mIo);
+ }
+ effectDesc->mIo = fxOutput;
+ }
+ }
+ }
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ if (i == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (getStrategy((audio_stream_type_t)i) == strategy) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+}
+
+void AudioPolicyManager::checkOutputForAllStrategies()
+{
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
+ checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+ checkOutputForStrategy(STRATEGY_PHONE);
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
+ checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+ checkOutputForStrategy(STRATEGY_SONIFICATION);
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ checkOutputForStrategy(STRATEGY_ACCESSIBILITY);
+ checkOutputForStrategy(STRATEGY_MEDIA);
+ checkOutputForStrategy(STRATEGY_DTMF);
+ checkOutputForStrategy(STRATEGY_REROUTING);
+}
+
+audio_io_handle_t AudioPolicyManager::getA2dpOutput()
+{
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ return mOutputs.keyAt(i);
+ }
+ }
+
+ return 0;
+}
+
+void AudioPolicyManager::checkA2dpSuspend()
+{
+ audio_io_handle_t a2dpOutput = getA2dpOutput();
+ if (a2dpOutput == 0) {
+ mA2dpSuspended = false;
+ return;
+ }
+
+ bool isScoConnected =
+ ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET &
+ ~AUDIO_DEVICE_BIT_IN) != 0) ||
+ ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0);
+ // suspend A2DP output if:
+ // (NOT already suspended) &&
+ // ((SCO device is connected &&
+ // (forced usage for communication || for record is SCO))) ||
+ // (phone state is ringing || in call)
+ //
+ // restore A2DP output if:
+ // (Already suspended) &&
+ // ((SCO device is NOT connected ||
+ // (forced usage NOT for communication && NOT for record is SCO))) &&
+ // (phone state is NOT ringing && NOT in call)
+ //
+ if (mA2dpSuspended) {
+ if ((!isScoConnected ||
+ ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) &&
+ ((mPhoneState != AUDIO_MODE_IN_CALL) &&
+ (mPhoneState != AUDIO_MODE_RINGTONE))) {
+
+ mpClientInterface->restoreOutput(a2dpOutput);
+ mA2dpSuspended = false;
+ }
+ } else {
+ if ((isScoConnected &&
+ ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) ||
+ ((mPhoneState == AUDIO_MODE_IN_CALL) ||
+ (mPhoneState == AUDIO_MODE_RINGTONE))) {
+
+ mpClientInterface->suspendOutput(a2dpOutput);
+ mA2dpSuspended = true;
+ }
+ }
+}
+
+audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache)
+{
+ audio_devices_t device = AUDIO_DEVICE_NONE;
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+
+ ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ if (patchDesc->mUid != mUidCached) {
+ ALOGV("getNewOutputDevice() device %08x forced by patch %d",
+ outputDesc->device(), outputDesc->mPatchHandle);
+ return outputDesc->device();
+ }
+ }
+
+ // check the following by order of priority to request a routing change if necessary:
+ // 1: the strategy enforced audible is active and enforced on the output:
+ // use device for strategy enforced audible
+ // 2: we are in call or the strategy phone is active on the output:
+ // use device for strategy phone
+ // 3: the strategy for enforced audible is active but not enforced on the output:
+ // use the device for strategy enforced audible
+ // 4: the strategy sonification is active on the output:
+ // use device for strategy sonification
+ // 5: the strategy "respectful" sonification is active on the output:
+ // use device for strategy "respectful" sonification
+ // 6: the strategy accessibility is active on the output:
+ // use device for strategy accessibility
+ // 7: the strategy media is active on the output:
+ // use device for strategy media
+ // 8: the strategy DTMF is active on the output:
+ // use device for strategy DTMF
+ // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output:
+ // use device for strategy t-t-s
+ if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE) &&
+ mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isInCall() ||
+ outputDesc->isStrategyActive(STRATEGY_PHONE)) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_ACCESSIBILITY)) {
+ device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
+ device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
+ device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
+ device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_REROUTING)) {
+ device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
+ }
+
+ ALOGV("getNewOutputDevice() selected device %x", device);
+ return device;
+}
+
+audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
+{
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+
+ ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ if (patchDesc->mUid != mUidCached) {
+ ALOGV("getNewInputDevice() device %08x forced by patch %d",
+ inputDesc->mDevice, inputDesc->mPatchHandle);
+ return inputDesc->mDevice;
+ }
+ }
+
+ audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->mInputSource);
+
+ ALOGV("getNewInputDevice() selected device %x", device);
+ return device;
+}
+
+uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
+ return (uint32_t)getStrategy(stream);
+}
+
+audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
+ // By checking the range of stream before calling getStrategy, we avoid
+ // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
+ // and then return STRATEGY_MEDIA, but we want to return the empty set.
+ if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) {
+ return AUDIO_DEVICE_NONE;
+ }
+ audio_devices_t devices;
+ routing_strategy strategy = getStrategy(stream);
+ devices = getDeviceForStrategy(strategy, true /*fromCache*/);
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
+ if (outputDesc->isStrategyActive(strategy)) {
+ devices = outputDesc->device();
+ break;
+ }
+ }
+
+ /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
+ and doesn't really need to.*/
+ if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
+ devices |= AUDIO_DEVICE_OUT_SPEAKER;
+ devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ }
+
+ return devices;
+}
+
+routing_strategy AudioPolicyManager::getStrategy(
+ audio_stream_type_t stream) {
+
+ ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH");
+
+ // stream to strategy mapping
+ switch (stream) {
+ case AUDIO_STREAM_VOICE_CALL:
+ case AUDIO_STREAM_BLUETOOTH_SCO:
+ return STRATEGY_PHONE;
+ case AUDIO_STREAM_RING:
+ case AUDIO_STREAM_ALARM:
+ return STRATEGY_SONIFICATION;
+ case AUDIO_STREAM_NOTIFICATION:
+ return STRATEGY_SONIFICATION_RESPECTFUL;
+ case AUDIO_STREAM_DTMF:
+ return STRATEGY_DTMF;
+ default:
+ ALOGE("unknown stream type %d", stream);
+ case AUDIO_STREAM_SYSTEM:
+ // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+ // while key clicks are played produces a poor result
+ case AUDIO_STREAM_MUSIC:
+ return STRATEGY_MEDIA;
+ case AUDIO_STREAM_ENFORCED_AUDIBLE:
+ return STRATEGY_ENFORCED_AUDIBLE;
+ case AUDIO_STREAM_TTS:
+ return STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
+ case AUDIO_STREAM_ACCESSIBILITY:
+ return STRATEGY_ACCESSIBILITY;
+ case AUDIO_STREAM_REROUTING:
+ return STRATEGY_REROUTING;
+ }
+}
+
+uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
+ // flags to strategy mapping
+ if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
+ return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
+ }
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
+ }
+
+ // usage to strategy mapping
+ switch (attr->usage) {
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ if (isStreamActive(AUDIO_STREAM_RING) || isStreamActive(AUDIO_STREAM_ALARM)) {
+ return (uint32_t) STRATEGY_SONIFICATION;
+ }
+ if (isInCall()) {
+ return (uint32_t) STRATEGY_PHONE;
+ }
+ return (uint32_t) STRATEGY_ACCESSIBILITY;
+
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return (uint32_t) STRATEGY_MEDIA;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return (uint32_t) STRATEGY_PHONE;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return (uint32_t) STRATEGY_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return (uint32_t) STRATEGY_SONIFICATION;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return (uint32_t) STRATEGY_MEDIA;
+ }
+}
+
+void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
+ switch(stream) {
+ case AUDIO_STREAM_MUSIC:
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ updateDevicesAndOutputs();
+ break;
+ default:
+ break;
+ }
+}
+
+bool AudioPolicyManager::isAnyOutputActive(audio_stream_type_t streamToIgnore) {
+ for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) {
+ if (s == (size_t) streamToIgnore) {
+ continue;
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ if (outputDesc->mRefCount[s] != 0) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+uint32_t AudioPolicyManager::handleEventForBeacon(int event) {
+ switch(event) {
+ case STARTING_OUTPUT:
+ mBeaconMuteRefCount++;
+ break;
+ case STOPPING_OUTPUT:
+ if (mBeaconMuteRefCount > 0) {
+ mBeaconMuteRefCount--;
+ }
+ break;
+ case STARTING_BEACON:
+ mBeaconPlayingRefCount++;
+ break;
+ case STOPPING_BEACON:
+ if (mBeaconPlayingRefCount > 0) {
+ mBeaconPlayingRefCount--;
+ }
+ break;
+ }
+
+ if (mBeaconMuteRefCount > 0) {
+ // any playback causes beacon to be muted
+ return setBeaconMute(true);
+ } else {
+ // no other playback: unmute when beacon starts playing, mute when it stops
+ return setBeaconMute(mBeaconPlayingRefCount == 0);
+ }
+}
+
+uint32_t AudioPolicyManager::setBeaconMute(bool mute) {
+ ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d",
+ mute, mBeaconMuteRefCount, mBeaconPlayingRefCount);
+ // keep track of muted state to avoid repeating mute/unmute operations
+ if (mBeaconMuted != mute) {
+ // mute/unmute AUDIO_STREAM_TTS on all outputs
+ ALOGV("\t muting %d", mute);
+ uint32_t maxLatency = 0;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ setStreamMute(AUDIO_STREAM_TTS, mute/*on*/,
+ desc->mIoHandle,
+ 0 /*delay*/, AUDIO_DEVICE_NONE);
+ const uint32_t latency = desc->latency() * 2;
+ if (latency > maxLatency) {
+ maxLatency = latency;
+ }
+ }
+ mBeaconMuted = mute;
+ return maxLatency;
+ }
+ return 0;
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ if (fromCache) {
+ ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
+ strategy, mDeviceForStrategy[strategy]);
+ return mDeviceForStrategy[strategy];
+ }
+ audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
+ switch (strategy) {
+
+ case STRATEGY_TRANSMITTED_THROUGH_SPEAKER:
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ if (!device) {
+ ALOGE("getDeviceForStrategy() no device found for "\
+ "STRATEGY_TRANSMITTED_THROUGH_SPEAKER");
+ }
+ break;
+
+ case STRATEGY_SONIFICATION_RESPECTFUL:
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC,
+ SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing on a remote device, use the the sonification behavior.
+ // Note that we test this usecase before testing if media is playing because
+ // the isStreamActive() method only informs about the activity of a stream, not
+ // if it's for local playback. Note also that we use the same delay between both tests
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ //user "safe" speaker if available instead of normal speaker to avoid triggering
+ //other acoustic safety mechanisms for notification
+ if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
+ device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing (or has recently played), use the same device
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ } else {
+ // when media is not playing anymore, fall back on the sonification behavior
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ //user "safe" speaker if available instead of normal speaker to avoid triggering
+ //other acoustic safety mechanisms for notification
+ if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
+ device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ }
+
+ break;
+
+ case STRATEGY_DTMF:
+ if (!isInCall()) {
+ // when off call, DTMF strategy follows the same rules as MEDIA strategy
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ break;
+ }
+ // when in call, DTMF and PHONE strategies follow the same rules
+ // FALL THROUGH
+
+ case STRATEGY_PHONE:
+ // Force use of only devices on primary output if:
+ // - in call AND
+ // - cannot route from voice call RX OR
+ // - audio HAL version is < 3.0 and TX device is on the primary HW module
+ if (mPhoneState == AUDIO_MODE_IN_CALL) {
+ audio_devices_t txDevice =
+ getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+ sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+ if (((mAvailableInputDevices.types() &
+ AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
+ (((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+ (hwOutputDesc->getAudioPort()->mModule->mHalVersion <
+ AUDIO_DEVICE_API_VERSION_3_0))) {
+ availableOutputDeviceTypes = availablePrimaryOutputDevices();
+ }
+ }
+ // for phone strategy, we first consider the forced use and then the available devices by order
+ // of priority
+ switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
+ case AUDIO_POLICY_FORCE_BT_SCO:
+ if (!isInCall() || strategy != STRATEGY_DTMF) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ if (device) break;
+ }
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+ if (device) break;
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ // FALL THROUGH
+
+ default: // FORCE_NONE
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+ if (!isInCall() &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0)) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ if (device) break;
+ }
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ if (mPhoneState != AUDIO_MODE_IN_CALL) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE;
+ if (device) break;
+ device = mDefaultOutputDevice->mDeviceType;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
+ }
+ break;
+
+ case AUDIO_POLICY_FORCE_SPEAKER:
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+ // A2DP speaker when forcing to speaker output
+ if (!isInCall() &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0)) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ if (device) break;
+ }
+ if (!isInCall()) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
+ if (device) break;
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device) break;
+ device = mDefaultOutputDevice->mDeviceType;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
+ }
+ break;
+ }
+ break;
+
+ case STRATEGY_SONIFICATION:
+
+ // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+ // handleIncallSonification().
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+ break;
+ }
+ // FALL THROUGH
+
+ case STRATEGY_ENFORCED_AUDIBLE:
+ // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
+ // except:
+ // - when in call where it doesn't default to STRATEGY_PHONE behavior
+ // - in countries where not enforced in which case it follows STRATEGY_MEDIA
+
+ if ((strategy == STRATEGY_SONIFICATION) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
+ device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
+ }
+ }
+ // The second device used for sonification is the same as the device used by media strategy
+ // FALL THROUGH
+
+ // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now
+ case STRATEGY_ACCESSIBILITY:
+ if (strategy == STRATEGY_ACCESSIBILITY) {
+ // do not route accessibility prompts to a digital output currently configured with a
+ // compressed format as they would likely not be mixed and dropped.
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ audio_devices_t devices = desc->device() &
+ (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
+ if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
+ devices != AUDIO_DEVICE_NONE) {
+ availableOutputDeviceTypes = availableOutputDeviceTypes & ~devices;
+ }
+ }
+ }
+ // FALL THROUGH
+
+ case STRATEGY_REROUTING:
+ case STRATEGY_MEDIA: {
+ uint32_t device2 = AUDIO_DEVICE_NONE;
+ if (strategy != STRATEGY_SONIFICATION) {
+ // no sonification on remote submix (e.g. WFD)
+ if (mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ }
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0)) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ }
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE)) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+ // no sonification on aux digital (e.g. HDMI)
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ int device3 = AUDIO_DEVICE_NONE;
+ if (strategy == STRATEGY_MEDIA) {
+ // ARC, SPDIF and AUX_LINE can co-exist with others.
+ device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC;
+ device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF);
+ device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE);
+ }
+
+ device2 |= device3;
+ // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
+ // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
+ device |= device2;
+
+ // If hdmi system audio mode is on, remove speaker out of output list.
+ if ((strategy == STRATEGY_MEDIA) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] ==
+ AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
+ device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+ }
+
+ if (device) break;
+ device = mDefaultOutputDevice->mDeviceType;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
+ }
+ } break;
+
+ default:
+ ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+ break;
+ }
+
+ ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+ return device;
+}
+
+void AudioPolicyManager::updateDevicesAndOutputs()
+{
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ }
+ mPreviousOutputs = mOutputs;
+}
+
+uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs)
+{
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ if (outputDesc->isDuplicated()) {
+ return 0;
+ }
+
+ uint32_t muteWaitMs = 0;
+ audio_devices_t device = outputDesc->device();
+ bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
+
+ for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+ audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types();
+ bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+ bool doMute = false;
+
+ if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = true;
+ } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = false;
+ }
+ if (doMute) {
+ for (size_t j = 0; j < mOutputs.size(); j++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
+ // skip output if it does not share any device with current output
+ if ((desc->supportedDevices() & outputDesc->supportedDevices())
+ == AUDIO_DEVICE_NONE) {
+ continue;
+ }
+ audio_io_handle_t curOutput = mOutputs.keyAt(j);
+ ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
+ mute ? "muting" : "unmuting", i, curDevice, curOutput);
+ setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
+ if (desc->isStrategyActive((routing_strategy)i)) {
+ if (mute) {
+ // FIXME: should not need to double latency if volume could be applied
+ // immediately by the audioflinger mixer. We must account for the delay
+ // between now and the next time the audioflinger thread for this output
+ // will process a buffer (which corresponds to one buffer size,
+ // usually 1/2 or 1/4 of the latency).
+ if (muteWaitMs < desc->latency() * 2) {
+ muteWaitMs = desc->latency() * 2;
+ }
+ }
+ }
+ }
+ }
+ }
+
+ // temporary mute output if device selection changes to avoid volume bursts due to
+ // different per device volumes
+ if (outputDesc->isActive() && (device != prevDevice)) {
+ if (muteWaitMs < outputDesc->latency() * 2) {
+ muteWaitMs = outputDesc->latency() * 2;
+ }
+ for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+ if (outputDesc->isStrategyActive((routing_strategy)i)) {
+ setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle);
+ // do tempMute unmute after twice the mute wait time
+ setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle,
+ muteWaitMs *2, device);
+ }
+ }
+ }
+
+ // wait for the PCM output buffers to empty before proceeding with the rest of the command
+ if (muteWaitMs > delayMs) {
+ muteWaitMs -= delayMs;
+ usleep(muteWaitMs * 1000);
+ return muteWaitMs;
+ }
+ return 0;
+}
+
+uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force,
+ int delayMs,
+ audio_patch_handle_t *patchHandle,
+ const char* address)
+{
+ ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ AudioParameter param;
+ uint32_t muteWaitMs;
+
+ if (outputDesc->isDuplicated()) {
+ muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs);
+ muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs);
+ return muteWaitMs;
+ }
+ // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+ // output profile
+ if ((device != AUDIO_DEVICE_NONE) &&
+ ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) {
+ return 0;
+ }
+
+ // filter devices according to output selected
+ device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types());
+
+ audio_devices_t prevDevice = outputDesc->mDevice;
+
+ ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
+
+ if (device != AUDIO_DEVICE_NONE) {
+ outputDesc->mDevice = device;
+ }
+ muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
+
+ // Do not change the routing if:
+ // the requested device is AUDIO_DEVICE_NONE
+ // OR the requested device is the same as current device
+ // AND force is not specified
+ // AND the output is connected by a valid audio patch.
+ // Doing this check here allows the caller to call setOutputDevice() without conditions
+ if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force &&
+ outputDesc->mPatchHandle != 0) {
+ ALOGV("setOutputDevice() setting same device %04x or null device for output %d",
+ device, output);
+ return muteWaitMs;
+ }
+
+ ALOGV("setOutputDevice() changing device");
+
+ // do the routing
+ if (device == AUDIO_DEVICE_NONE) {
+ resetOutputDevice(output, delayMs, NULL);
+ } else {
+ DeviceVector deviceList = (address == NULL) ?
+ mAvailableOutputDevices.getDevicesFromType(device)
+ : mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address));
+ if (!deviceList.isEmpty()) {
+ struct audio_patch patch;
+ outputDesc->toAudioPortConfig(&patch.sources[0]);
+ patch.num_sources = 1;
+ patch.num_sinks = 0;
+ for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
+ deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
+ patch.num_sinks++;
+ }
+ ssize_t index;
+ if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ }
+ sp< AudioPatch> patchDesc;
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&patch,
+ &afPatchHandle,
+ delayMs);
+ ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
+ "num_sources %d num_sinks %d",
+ status, afPatchHandle, patch.num_sources, patch.num_sinks);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &patch, mUidCached);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = patch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ patchDesc->mUid = mUidCached;
+ if (patchHandle) {
+ *patchHandle = patchDesc->mHandle;
+ }
+ outputDesc->mPatchHandle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+ }
+
+ // inform all input as well
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
+ if (!isVirtualInputDevice(inputDescriptor->mDevice)) {
+ AudioParameter inputCmd = AudioParameter();
+ ALOGV("%s: inform input %d of device:%d", __func__,
+ inputDescriptor->mIoHandle, device);
+ inputCmd.addInt(String8(AudioParameter::keyRouting),device);
+ mpClientInterface->setParameters(inputDescriptor->mIoHandle,
+ inputCmd.toString(),
+ delayMs);
+ }
+ }
+ }
+
+ // update stream volumes according to new device
+ applyStreamVolumes(output, device, delayMs);
+
+ return muteWaitMs;
+}
+
+status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output,
+ int delayMs,
+ audio_patch_handle_t *patchHandle)
+{
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ssize_t index;
+ if (patchHandle) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ }
+ if (index < 0) {
+ return INVALID_OPERATION;
+ }
+ sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs);
+ ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
+ outputDesc->mPatchHandle = 0;
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ return status;
+}
+
+status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
+ audio_devices_t device,
+ bool force,
+ audio_patch_handle_t *patchHandle)
+{
+ status_t status = NO_ERROR;
+
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
+ inputDesc->mDevice = device;
+
+ DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
+ if (!deviceList.isEmpty()) {
+ struct audio_patch patch;
+ inputDesc->toAudioPortConfig(&patch.sinks[0]);
+ // AUDIO_SOURCE_HOTWORD is for internal use only:
+ // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
+ if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
+ !inputDesc->mIsSoundTrigger) {
+ patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
+ }
+ patch.num_sinks = 1;
+ //only one input device for now
+ deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
+ patch.num_sources = 1;
+ ssize_t index;
+ if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ }
+ sp< AudioPatch> patchDesc;
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&patch,
+ &afPatchHandle,
+ 0);
+ ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
+ status, afPatchHandle);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
+ &patch, mUidCached);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = patch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ patchDesc->mUid = mUidCached;
+ if (patchHandle) {
+ *patchHandle = patchDesc->mHandle;
+ }
+ inputDesc->mPatchHandle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+ }
+ }
+ return status;
+}
+
+status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input,
+ audio_patch_handle_t *patchHandle)
+{
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ ssize_t index;
+ if (patchHandle) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ }
+ if (index < 0) {
+ return INVALID_OPERATION;
+ }
+ sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
+ inputDesc->mPatchHandle = 0;
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ return status;
+}
+
+sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
+ String8 address,
+ uint32_t& samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags)
+{
+ // Choose an input profile based on the requested capture parameters: select the first available
+ // profile supporting all requested parameters.
+
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
+ // profile->log();
+ if (profile->isCompatibleProfile(device, address, samplingRate,
+ &samplingRate /*updatedSamplingRate*/,
+ format, channelMask, (audio_output_flags_t) flags)) {
+
+ return profile;
+ }
+ }
+ }
+ return NULL;
+}
+
+
+audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource,
+ AudioMix **policyMix)
+{
+ audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
+ ~AUDIO_DEVICE_BIT_IN;
+
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ if (mPolicyMixes[i]->mMix.mMixType != MIX_TYPE_RECORDERS) {
+ continue;
+ }
+ for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
+ if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource == inputSource) ||
+ (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource != inputSource)) {
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ if (policyMix != NULL) {
+ *policyMix = &mPolicyMixes[i]->mMix;
+ }
+ return AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+ }
+ }
+ }
+
+ return getDeviceForInputSource(inputSource);
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+ audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
+ ~AUDIO_DEVICE_BIT_IN;
+
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ break;
+ }
+ break;
+
+ case AUDIO_SOURCE_DEFAULT:
+ case AUDIO_SOURCE_MIC:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
+ } else if ((mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO) &&
+ (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ // Allow only use of devices on primary input if in call and HAL does not support routing
+ // to voice call path.
+ if ((mPhoneState == AUDIO_MODE_IN_CALL) &&
+ (mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
+ availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN;
+ }
+
+ switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
+ case AUDIO_POLICY_FORCE_BT_SCO:
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ break;
+ }
+ // FALL THROUGH
+
+ default: // FORCE_NONE
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+
+ case AUDIO_POLICY_FORCE_SPEAKER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ }
+ break;
+
+ case AUDIO_SOURCE_VOICE_RECOGNITION:
+ case AUDIO_SOURCE_HOTWORD:
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO &&
+ availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_CAMCORDER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ }
+ break;
+ case AUDIO_SOURCE_REMOTE_SUBMIX:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+ case AUDIO_SOURCE_FM_TUNER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) {
+ device = AUDIO_DEVICE_IN_FM_TUNER;
+ }
+ break;
+ default:
+ ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
+ break;
+ }
+ ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+ return device;
+}
+
+bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device)
+{
+ if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+ device &= ~AUDIO_DEVICE_BIT_IN;
+ if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
+ return true;
+ }
+ return false;
+}
+
+bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) {
+ return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0);
+}
+
+audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
+{
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> input_descriptor = mInputs.valueAt(i);
+ if ((input_descriptor->mRefCount > 0)
+ && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
+ return mInputs.keyAt(i);
+ }
+ }
+ return 0;
+}
+
+uint32_t AudioPolicyManager::activeInputsCount() const
+{
+ uint32_t count = 0;
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> desc = mInputs.valueAt(i);
+ if (desc->mRefCount > 0) {
+ count++;
+ }
+ }
+ return count;
+}
+
+
+void AudioPolicyManager::initializeVolumeCurves()
+{
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
+ mStreams[i].mVolumeCurve[j] =
+ ApmGains::sVolumeProfiles[i][j];
+ }
+ }
+
+ // Check availability of DRC on speaker path: if available, override some of the speaker curves
+ if (mSpeakerDrcEnabled) {
+ mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sDefaultSystemVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_RING].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerMediaVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerMediaVolumeCurveDrc;
+ }
+}
+
+float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device)
+{
+ float volume = 1.0;
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ StreamDescriptor &streamDesc = mStreams[stream];
+
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ volume = ApmGains::volIndexToAmpl(device, streamDesc, index);
+
+ // if a headset is connected, apply the following rules to ring tones and notifications
+ // to avoid sound level bursts in user's ears:
+ // - always attenuate ring tones and notifications volume by 6dB
+ // - if music is playing, always limit the volume to current music volume,
+ // with a minimum threshold at -36dB so that notification is always perceived.
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+ AUDIO_DEVICE_OUT_WIRED_HEADSET |
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
+ ((stream_strategy == STRATEGY_SONIFICATION)
+ || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
+ || (stream == AUDIO_STREAM_SYSTEM)
+ || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) &&
+ streamDesc.mCanBeMuted) {
+ volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
+ // when the phone is ringing we must consider that music could have been paused just before
+ // by the music application and behave as if music was active if the last music track was
+ // just stopped
+ if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
+ mLimitRingtoneVolume) {
+ audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
+ float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
+ mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
+ output,
+ musicDevice);
+ float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
+ musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+ if (volume > minVol) {
+ volume = minVol;
+ ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+ }
+ }
+ }
+
+ return volume;
+}
+
+status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+
+ // do not change actual stream volume if the stream is muted
+ if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+ ALOGVV("checkAndSetVolume() stream %d muted count %d",
+ stream, mOutputs.valueFor(output)->mMuteCount[stream]);
+ return NO_ERROR;
+ }
+
+ // do not change in call volume if bluetooth is connected and vice versa
+ if ((stream == AUDIO_STREAM_VOICE_CALL &&
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (stream == AUDIO_STREAM_BLUETOOTH_SCO &&
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) {
+ ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+ stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
+ return INVALID_OPERATION;
+ }
+
+ float volume = computeVolume(stream, index, output, device);
+ // unit gain if rerouting to external policy
+ if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index >= 0) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->mPolicyMix != NULL) {
+ ALOGV("max gain when rerouting for output=%d", output);
+ volume = 1.0f;
+ }
+ }
+
+ }
+ // We actually change the volume if:
+ // - the float value returned by computeVolume() changed
+ // - the force flag is set
+ if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
+ force) {
+ mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+ ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+ // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+ // enabled
+ if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
+ }
+ mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
+ }
+
+ if (stream == AUDIO_STREAM_VOICE_CALL ||
+ stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ float voiceVolume;
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (stream == AUDIO_STREAM_VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+ } else {
+ voiceVolume = 1.0;
+ }
+
+ if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ mLastVoiceVolume = voiceVolume;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+ ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
+
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ checkAndSetVolume((audio_stream_type_t)stream,
+ mStreams[stream].getVolumeIndex(device),
+ output,
+ device,
+ delayMs,
+ force);
+ }
+}
+
+void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (getStrategy((audio_stream_type_t)stream) == strategy) {
+ setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
+ }
+ }
+}
+
+void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ StreamDescriptor &streamDesc = mStreams[stream];
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
+ stream, on, output, outputDesc->mMuteCount[stream], device);
+
+ if (on) {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ if (streamDesc.mCanBeMuted &&
+ ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) {
+ checkAndSetVolume(stream, 0, output, device, delayMs);
+ }
+ }
+ // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+ outputDesc->mMuteCount[stream]++;
+ } else {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ ALOGV("setStreamMute() unmuting non muted stream!");
+ return;
+ }
+ if (--outputDesc->mMuteCount[stream] == 0) {
+ checkAndSetVolume(stream,
+ streamDesc.getVolumeIndex(device),
+ output,
+ device,
+ delayMs);
+ }
+ }
+}
+
+void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
+ bool starting, bool stateChange)
+{
+ // if the stream pertains to sonification strategy and we are in call we must
+ // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+ // in the device used for phone strategy and play the tone if the selected device does not
+ // interfere with the device used for phone strategy
+ // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
+ // many times as there are active tracks on the output
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((stream_strategy == STRATEGY_SONIFICATION) ||
+ ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
+ stream, starting, outputDesc->mDevice, stateChange);
+ if (outputDesc->mRefCount[stream]) {
+ int muteCount = 1;
+ if (stateChange) {
+ muteCount = outputDesc->mRefCount[stream];
+ }
+ if (audio_is_low_visibility(stream)) {
+ ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mPrimaryOutput);
+ }
+ } else {
+ ALOGV("handleIncallSonification() high visibility");
+ if (outputDesc->device() &
+ getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
+ ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mPrimaryOutput);
+ }
+ }
+ if (starting) {
+ mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
+ AUDIO_STREAM_VOICE_CALL);
+ } else {
+ mpClientInterface->stopTone();
+ }
+ }
+ }
+ }
+}
+
+bool AudioPolicyManager::isInCall()
+{
+ return isStateInCall(mPhoneState);
+}
+
+bool AudioPolicyManager::isStateInCall(int state) {
+ return ((state == AUDIO_MODE_IN_CALL) ||
+ (state == AUDIO_MODE_IN_COMMUNICATION));
+}
+
+uint32_t AudioPolicyManager::getMaxEffectsCpuLoad()
+{
+ return MAX_EFFECTS_CPU_LOAD;
+}
+
+uint32_t AudioPolicyManager::getMaxEffectsMemory()
+{
+ return MAX_EFFECTS_MEMORY;
+}
+
+
+// --- EffectDescriptor class implementation
+
+status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " I/O: %d\n", mIo);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Session: %d\n", mSession);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+
+// --- audio_policy.conf file parsing
+// TODO candidate to be moved to ConfigParsingUtils
+void AudioPolicyManager::loadHwModule(cnode *root)
+{
+ status_t status = NAME_NOT_FOUND;
+ cnode *node;
+ sp<HwModule> module = new HwModule(root->name);
+
+ node = config_find(root, DEVICES_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading device %s", node->name);
+ status_t tmpStatus = module->loadDevice(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ node = config_find(root, OUTPUTS_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading output %s", node->name);
+ status_t tmpStatus = module->loadOutput(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ node = config_find(root, INPUTS_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading input %s", node->name);
+ status_t tmpStatus = module->loadInput(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ loadGlobalConfig(root, module);
+
+ if (status == NO_ERROR) {
+ mHwModules.add(module);
+ }
+}
+
+// TODO candidate to be moved to ConfigParsingUtils
+void AudioPolicyManager::loadHwModules(cnode *root)
+{
+ cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
+ if (node == NULL) {
+ return;
+ }
+
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModules() loading module %s", node->name);
+ loadHwModule(node);
+ node = node->next;
+ }
+}
+
+// TODO candidate to be moved to ConfigParsingUtils
+void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module)
+{
+ cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
+
+ if (node == NULL) {
+ return;
+ }
+ DeviceVector declaredDevices;
+ if (module != NULL) {
+ declaredDevices = module->mDeclaredDevices;
+ }
+
+ node = node->first_child;
+ while (node) {
+ if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
+ mAvailableOutputDevices.loadDevicesFromName((char *)node->value,
+ declaredDevices);
+ ALOGV("loadGlobalConfig() Attached Output Devices %08x",
+ mAvailableOutputDevices.types());
+ } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
+ audio_devices_t device = (audio_devices_t)ConfigParsingUtils::stringToEnum(
+ sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ (char *)node->value);
+ if (device != AUDIO_DEVICE_NONE) {
+ mDefaultOutputDevice = new DeviceDescriptor(String8("default-output"), device);
+ } else {
+ ALOGW("loadGlobalConfig() default device not specified");
+ }
+ ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType);
+ } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
+ mAvailableInputDevices.loadDevicesFromName((char *)node->value,
+ declaredDevices);
+ ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
+ } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
+ mSpeakerDrcEnabled = ConfigParsingUtils::stringToBool((char *)node->value);
+ ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
+ } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
+ uint32_t major, minor;
+ sscanf((char *)node->value, "%u.%u", &major, &minor);
+ module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor);
+ ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u",
+ module->mHalVersion, major, minor);
+ }
+ node = node->next;
+ }
+}
+
+// TODO candidate to be moved to ConfigParsingUtils
+status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
+{
+ cnode *root;
+ char *data;
+
+ data = (char *)load_file(path, NULL);
+ if (data == NULL) {
+ return -ENODEV;
+ }
+ root = config_node("", "");
+ config_load(root, data);
+
+ loadHwModules(root);
+ // legacy audio_policy.conf files have one global_configuration section
+ loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY));
+ config_free(root);
+ free(root);
+ free(data);
+
+ ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::defaultAudioPolicyConfig(void)
+{
+ sp<HwModule> module;
+ sp<IOProfile> profile;
+ sp<DeviceDescriptor> defaultInputDevice =
+ new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC);
+ mAvailableOutputDevices.add(mDefaultOutputDevice);
+ mAvailableInputDevices.add(defaultInputDevice);
+
+ module = new HwModule("primary");
+
+ profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module);
+ profile->mSamplingRates.add(44100);
+ profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+ profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
+ profile->mSupportedDevices.add(mDefaultOutputDevice);
+ profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
+ module->mOutputProfiles.add(profile);
+
+ profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module);
+ profile->mSamplingRates.add(8000);
+ profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+ profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
+ profile->mSupportedDevices.add(defaultInputDevice);
+ module->mInputProfiles.add(profile);
+
+ mHwModules.add(module);
+}
+
+audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
+{
+ // flags to stream type mapping
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return AUDIO_STREAM_ENFORCED_AUDIBLE;
+ }
+ if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+ return AUDIO_STREAM_BLUETOOTH_SCO;
+ }
+ if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
+ return AUDIO_STREAM_TTS;
+ }
+
+ // usage to stream type mapping
+ switch (attr->usage) {
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ return AUDIO_STREAM_MUSIC;
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ if (isStreamActive(AUDIO_STREAM_ALARM)) {
+ return AUDIO_STREAM_ALARM;
+ }
+ if (isStreamActive(AUDIO_STREAM_RING)) {
+ return AUDIO_STREAM_RING;
+ }
+ if (isInCall()) {
+ return AUDIO_STREAM_VOICE_CALL;
+ }
+ return AUDIO_STREAM_ACCESSIBILITY;
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return AUDIO_STREAM_SYSTEM;
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return AUDIO_STREAM_VOICE_CALL;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return AUDIO_STREAM_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ return AUDIO_STREAM_ALARM;
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return AUDIO_STREAM_RING;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return AUDIO_STREAM_NOTIFICATION;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return AUDIO_STREAM_MUSIC;
+ }
+}
+
+bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) {
+ // has flags that map to a strategy?
+ if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) {
+ return true;
+ }
+
+ // has known usage?
+ switch (paa->usage) {
+ case AUDIO_USAGE_UNKNOWN:
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_VIRTUAL_SOURCE:
+ break;
+ default:
+ return false;
+ }
+ return true;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
new file mode 100644
index 0000000..61ea6f2
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -0,0 +1,560 @@
+/*
+ * 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 <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include <media/AudioPolicy.h>
+#include "AudioPolicyInterface.h"
+
+#include "Gains.h"
+#include "Ports.h"
+#include "ConfigParsingUtils.h"
+#include "Devices.h"
+#include "IOProfile.h"
+#include "HwModule.h"
+#include "AudioInputDescriptor.h"
+#include "AudioOutputDescriptor.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN 0.016
+// Time in milliseconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY 5000
+// Time in milliseconds after media stopped playing during which we consider that the
+// sonification should be as unobtrusive as during the time media was playing.
+#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
+// Time in milliseconds during witch some streams are muted while the audio path
+// is switched
+#define MUTE_TIME_MS 2000
+
+#define NUM_TEST_OUTPUTS 5
+
+#define NUM_VOL_CURVE_KNEES 2
+
+// Default minimum length allowed for offloading a compressed track
+// Can be overridden by the audio.offload.min.duration.secs property
+#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+
+#define MAX_MIXER_SAMPLING_RATE 48000
+#define MAX_MIXER_CHANNEL_COUNT 8
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager implements audio policy manager behavior common to all platforms.
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManager: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+ , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+ AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+ virtual ~AudioPolicyManager();
+
+ // AudioPolicyInterface
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name);
+ virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+ const char *device_address);
+ virtual void setPhoneState(audio_mode_t state);
+ virtual void setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config);
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+ virtual void setSystemProperty(const char* property, const char* value);
+ virtual status_t initCheck();
+ virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ virtual status_t getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ virtual status_t startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual void releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual status_t getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ input_type_t *inputType);
+
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_io_handle_t input,
+ audio_session_t session);
+
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input,
+ audio_session_t session);
+ virtual void releaseInput(audio_io_handle_t input,
+ audio_session_t session);
+ virtual void closeAllInputs();
+ virtual void initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax);
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
+
+ // return the strategy corresponding to a given stream type
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+ // return the strategy corresponding to the given audio attributes
+ virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
+
+ // return the enabled output devices for the given stream type
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
+ virtual status_t registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id);
+ virtual status_t unregisterEffect(int id);
+ virtual status_t setEffectEnabled(int id, bool enabled);
+
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+ // return whether a stream is playing remotely, override to change the definition of
+ // local/remote playback, used for instance by notification manager to not make
+ // media players lose audio focus when not playing locally
+ // For the base implementation, "remotely" means playing during screen mirroring which
+ // uses an output for playback with a non-empty, non "0" address.
+ virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+ virtual bool isSourceActive(audio_source_t source) const;
+
+ virtual status_t dump(int fd);
+
+ virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+ virtual status_t listAudioPorts(audio_port_role_t role,
+ audio_port_type_t type,
+ unsigned int *num_ports,
+ struct audio_port *ports,
+ unsigned int *generation);
+ virtual status_t getAudioPort(struct audio_port *port);
+ virtual status_t createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ uid_t uid);
+ virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
+ uid_t uid);
+ virtual status_t listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches,
+ unsigned int *generation);
+ virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+ virtual void clearAudioPatches(uid_t uid);
+
+ virtual status_t acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device);
+
+ virtual status_t releaseSoundTriggerSession(audio_session_t session);
+
+ virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
+ virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
+
+ // Audio policy configuration file parsing (audio_policy.conf)
+ // TODO candidates to be moved to ConfigParsingUtils
+ void loadHwModule(cnode *root);
+ void loadHwModules(cnode *root);
+ void loadGlobalConfig(cnode *root, const sp<HwModule>& module);
+ status_t loadAudioPolicyConfig(const char *path);
+ void defaultAudioPolicyConfig(void);
+
+ // return the strategy corresponding to a given stream type
+ static routing_strategy getStrategy(audio_stream_type_t stream);
+
+ static uint32_t nextUniqueId();
+protected:
+
+ class EffectDescriptor : public RefBase
+ {
+ public:
+
+ status_t dump(int fd);
+
+ int mIo; // io the effect is attached to
+ routing_strategy mStrategy; // routing strategy the effect is associated to
+ int mSession; // audio session the effect is on
+ effect_descriptor_t mDesc; // effect descriptor
+ bool mEnabled; // enabled state: CPU load being used or not
+ };
+
+ void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
+ void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
+
+ // return appropriate device for streams handled by the specified strategy according to current
+ // phone state, connected devices...
+ // if fromCache is true, the device is returned from mDeviceForStrategy[],
+ // otherwise it is determine by current state
+ // (device connected,phone state, force use, a2dp output...)
+ // This allows to:
+ // 1 speed up process when the state is stable (when starting or stopping an output)
+ // 2 access to either current device selection (fromCache == true) or
+ // "future" device selection (fromCache == false) when called from a context
+ // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+ // before updateDevicesAndOutputs() is called.
+ virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache);
+
+ // change the route of the specified output. Returns the number of ms we have slept to
+ // allow new routing to take effect in certain cases.
+ virtual uint32_t setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force = false,
+ int delayMs = 0,
+ audio_patch_handle_t *patchHandle = NULL,
+ const char* address = NULL);
+ status_t resetOutputDevice(audio_io_handle_t output,
+ int delayMs = 0,
+ audio_patch_handle_t *patchHandle = NULL);
+ status_t setInputDevice(audio_io_handle_t input,
+ audio_devices_t device,
+ bool force = false,
+ audio_patch_handle_t *patchHandle = NULL);
+ status_t resetInputDevice(audio_io_handle_t input,
+ audio_patch_handle_t *patchHandle = NULL);
+
+ // select input device corresponding to requested audio source
+ virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+
+ // return io handle of active input or 0 if no input is active
+ // Only considers inputs from physical devices (e.g. main mic, headset mic) when
+ // ignoreVirtualInputs is true.
+ audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+ uint32_t activeInputsCount() const;
+
+ // initialize volume curves for each strategy and device category
+ void initializeVolumeCurves();
+
+ // compute the actual volume for a given stream according to the requested index and a particular
+ // device
+ virtual float computeVolume(audio_stream_type_t stream, int index,
+ audio_io_handle_t output, audio_devices_t device);
+
+ // check that volume change is permitted, compute and send new volume to audio hardware
+ virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs = 0, bool force = false);
+
+ // apply all stream volumes to the specified output and device
+ void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+ // Mute or unmute all streams handled by the specified strategy on the specified output
+ void setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
+ // Mute or unmute the stream on the specified output
+ void setStreamMute(audio_stream_type_t stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
+ // handle special cases for sonification strategy while in call: mute streams or replace by
+ // a special tone in the device used for communication
+ void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
+
+ // true if device is in a telephony or VoIP call
+ virtual bool isInCall();
+
+ // true if given state represents a device in a telephony or VoIP call
+ virtual bool isStateInCall(int state);
+
+ // when a device is connected, checks if an open output can be routed
+ // to this device. If none is open, tries to open one of the available outputs.
+ // Returns an output suitable to this device or 0.
+ // when a device is disconnected, checks if an output is not used any more and
+ // returns its handle if any.
+ // transfers the audio tracks and effects from one output thread to another accordingly.
+ status_t checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& outputs,
+ const String8 address);
+
+ status_t checkInputsForDevice(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& inputs,
+ const String8 address);
+
+ // close an output and its companion duplicating output.
+ void closeOutput(audio_io_handle_t output);
+
+ // close an input.
+ void closeInput(audio_io_handle_t input);
+
+ // checks and if necessary changes outputs used for all strategies.
+ // must be called every time a condition that affects the output choice for a given strategy
+ // changes: connected device, phone state, force use...
+ // Must be called before updateDevicesAndOutputs()
+ void checkOutputForStrategy(routing_strategy strategy);
+
+ // Same as checkOutputForStrategy() but for a all strategies in order of priority
+ void checkOutputForAllStrategies();
+
+ // manages A2DP output suspend/restore according to phone state and BT SCO usage
+ void checkA2dpSuspend();
+
+ // returns the A2DP output handle if it is open or 0 otherwise
+ audio_io_handle_t getA2dpOutput();
+
+ // selects the most appropriate device on output for current state
+ // must be called every time a condition that affects the device choice for a given output is
+ // changed: connected device, phone state, force use, output start, output stop..
+ // see getDeviceForStrategy() for the use of fromCache parameter
+ audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
+
+ // updates cache of device used by all strategies (mDeviceForStrategy[])
+ // must be called every time a condition that affects the device choice for a given strategy is
+ // changed: connected device, phone state, force use...
+ // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
+ // Must be called after checkOutputForAllStrategies()
+ void updateDevicesAndOutputs();
+
+ // selects the most appropriate device on input for current state
+ audio_devices_t getNewInputDevice(audio_io_handle_t input);
+
+ virtual uint32_t getMaxEffectsCpuLoad();
+ virtual uint32_t getMaxEffectsMemory();
+#ifdef AUDIO_POLICY_TEST
+ virtual bool threadLoop();
+ void exit();
+ int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+ status_t setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled);
+
+ SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
+ bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2);
+
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ // Returns the number of ms waited
+ virtual uint32_t checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs);
+
+ audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ audio_output_flags_t flags,
+ audio_format_t format);
+ // samplingRate parameter is an in/out and so may be modified
+ sp<IOProfile> getInputProfile(audio_devices_t device,
+ String8 address,
+ uint32_t& samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags);
+ sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags);
+
+ audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
+
+ bool isNonOffloadableEffectEnabled();
+
+ virtual status_t addAudioPatch(audio_patch_handle_t handle,
+ const sp<AudioPatch>& patch);
+ virtual status_t removeAudioPatch(audio_patch_handle_t handle);
+
+ sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
+ sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
+ sp<HwModule> getModuleForDevice(audio_devices_t device) const;
+ sp<HwModule> getModuleFromName(const char *name) const;
+ audio_devices_t availablePrimaryOutputDevices();
+ audio_devices_t availablePrimaryInputDevices();
+
+ void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
+
+
+ uid_t mUidCached;
+ AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
+ audio_io_handle_t mPrimaryOutput; // primary output handle
+ // list of descriptors for outputs currently opened
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs;
+ // copy of mOutputs before setDeviceConnectionState() opens new outputs
+ // reset to mOutputs when updateDevicesAndOutputs() is called.
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > mInputs; // list of input descriptors
+ DeviceVector mAvailableOutputDevices; // all available output devices
+ DeviceVector mAvailableInputDevices; // all available input devices
+ int mPhoneState; // current phone state
+ audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration
+
+ StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control
+ bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
+ audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+ float mLastVoiceVolume; // last voice volume value sent to audio HAL
+
+ // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
+ static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
+ // Maximum memory allocated to audio effects in KB
+ static const uint32_t MAX_EFFECTS_MEMORY = 512;
+ uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
+ uint32_t mTotalEffectsMemory; // current memory used by effects
+ KeyedVector<int, sp<EffectDescriptor> > mEffects; // list of registered audio effects
+ bool mA2dpSuspended; // true if A2DP output is suspended
+ sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
+ bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
+ // to boost soft sounds, used to adjust volume curves accordingly
+
+ Vector < sp<HwModule> > mHwModules;
+ static volatile int32_t mNextUniqueId;
+ volatile int32_t mAudioPortGeneration;
+
+ DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
+
+ DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions;
+
+ sp<AudioPatch> mCallTxPatch;
+ sp<AudioPatch> mCallRxPatch;
+
+ // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
+ // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
+ enum {
+ STARTING_OUTPUT,
+ STARTING_BEACON,
+ STOPPING_OUTPUT,
+ STOPPING_BEACON
+ };
+ uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon
+ uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
+ bool mBeaconMuted; // has STREAM_TTS been muted
+
+ // custom mix entry in mPolicyMixes
+ class AudioPolicyMix : public RefBase {
+ public:
+ AudioPolicyMix() {}
+
+ AudioMix mMix; // Audio policy mix descriptor
+ sp<AudioOutputDescriptor> mOutput; // Corresponding output stream
+ };
+ DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
+
+
+#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
+
+ static bool isVirtualInputDevice(audio_devices_t device);
+
+ uint32_t nextAudioPortGeneration();
+private:
+ // updates device caching and output for streams that can influence the
+ // routing of notifications
+ void handleNotificationRoutingForStream(audio_stream_type_t stream);
+ static bool deviceDistinguishesOnAddress(audio_devices_t device);
+ // find the outputs on a given output descriptor that have the given address.
+ // to be called on an AudioOutputDescriptor whose supported devices (as defined
+ // in mProfile->mSupportedDevices) matches the device whose address is to be matched.
+ // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
+ // where addresses are used to distinguish between one connected device and another.
+ void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+ const audio_devices_t device /*in*/,
+ const String8 address /*in*/,
+ SortedVector<audio_io_handle_t>& outputs /*out*/);
+ uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
+ // internal method to return the output handle for the given device and format
+ audio_io_handle_t getOutputForDevice(
+ audio_devices_t device,
+ audio_session_t session,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ // internal function to derive a stream type value from audio attributes
+ audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
+ // return true if any output is playing anything besides the stream to ignore
+ bool isAnyOutputActive(audio_stream_type_t streamToIgnore);
+ // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
+ // returns 0 if no mute/unmute event happened, the largest latency of the device where
+ // the mute/unmute happened
+ uint32_t handleEventForBeacon(int event);
+ uint32_t setBeaconMute(bool mute);
+ bool isValidAttributes(const audio_attributes_t *paa);
+
+ // select input device corresponding to requested audio source and return associated policy
+ // mix if any. Calls getDeviceForInputSource().
+ audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
+ AudioMix **policyMix = NULL);
+
+ // Called by setDeviceConnectionState().
+ status_t setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name);
+ sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name);
+};
+
+};
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
new file mode 100644
index 0000000..1afd487
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
@@ -0,0 +1,121 @@
+/*
+ * Copyright (C) 2015 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 "APM::ConfigParsingUtils"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+//static
+uint32_t ConfigParsingUtils::stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (strcmp(table[i].name, name) == 0) {
+ ALOGV("stringToEnum() found %s", table[i].name);
+ return table[i].value;
+ }
+ }
+ return 0;
+}
+
+//static
+const char *ConfigParsingUtils::enumToString(const struct StringToEnum *table,
+ size_t size,
+ uint32_t value)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (table[i].value == value) {
+ return table[i].name;
+ }
+ }
+ return "";
+}
+
+//static
+bool ConfigParsingUtils::stringToBool(const char *value)
+{
+ return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+
+// --- audio_policy.conf file parsing
+//static
+uint32_t ConfigParsingUtils::parseOutputFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= ConfigParsingUtils::stringToEnum(sOutputFlagNameToEnumTable,
+ ARRAY_SIZE(sOutputFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+ }
+
+ return flag;
+}
+
+//static
+uint32_t ConfigParsingUtils::parseInputFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= stringToEnum(sInputFlagNameToEnumTable,
+ ARRAY_SIZE(sInputFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ return flag;
+}
+
+//static
+audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name)
+{
+ uint32_t device = 0;
+
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ device |= stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ }
+ devName = strtok(NULL, "|");
+ }
+ return device;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.h b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
new file mode 100644
index 0000000..7969661
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
@@ -0,0 +1,159 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// Definitions for audio_policy.conf file parsing
+// ----------------------------------------------------------------------------
+
+struct StringToEnum {
+ const char *name;
+ uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+const StringToEnum sDeviceNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
+};
+
+const StringToEnum sOutputFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
+};
+
+const StringToEnum sInputFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
+};
+
+const StringToEnum sFormatNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+ STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
+ STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
+ STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+ STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
+};
+
+const StringToEnum sOutChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const StringToEnum sInChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+const StringToEnum sGainModeNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
+};
+
+class ConfigParsingUtils
+{
+public:
+ static uint32_t stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name);
+ static const char *enumToString(const struct StringToEnum *table,
+ size_t size,
+ uint32_t value);
+ static bool stringToBool(const char *value);
+ static uint32_t parseOutputFlagNames(char *name);
+ static uint32_t parseInputFlagNames(char *name);
+ static audio_devices_t parseDeviceNames(char *name);
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.cpp b/services/audiopolicy/managerdefault/Devices.cpp
new file mode 100644
index 0000000..13c8bbc
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Devices.cpp
@@ -0,0 +1,282 @@
+/*
+ * Copyright (C) 2015 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 "APM::Devices"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+String8 DeviceDescriptor::emptyNameStr = String8("");
+
+DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
+ AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
+ audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+ AUDIO_PORT_ROLE_SOURCE,
+ NULL),
+ mDeviceType(type), mAddress("")
+{
+
+}
+
+bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
+{
+ // Devices are considered equal if they:
+ // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
+ // - have the same address or one device does not specify the address
+ // - have the same channel mask or one device does not specify the channel mask
+ return (mDeviceType == other->mDeviceType) &&
+ (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
+ (mChannelMask == 0 || other->mChannelMask == 0 ||
+ mChannelMask == other->mChannelMask);
+}
+
+void DeviceDescriptor::loadGains(cnode *root)
+{
+ AudioPort::loadGains(root);
+ if (mGains.size() > 0) {
+ mGains[0]->getDefaultConfig(&mGain);
+ }
+}
+
+void DeviceVector::refreshTypes()
+{
+ mDeviceTypes = AUDIO_DEVICE_NONE;
+ for(size_t i = 0; i < size(); i++) {
+ mDeviceTypes |= itemAt(i)->mDeviceType;
+ }
+ ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
+}
+
+ssize_t DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
+{
+ for(size_t i = 0; i < size(); i++) {
+ if (item->equals(itemAt(i))) {
+ return i;
+ }
+ }
+ return -1;
+}
+
+ssize_t DeviceVector::add(const sp<DeviceDescriptor>& item)
+{
+ ssize_t ret = indexOf(item);
+
+ if (ret < 0) {
+ ret = SortedVector::add(item);
+ if (ret >= 0) {
+ refreshTypes();
+ }
+ } else {
+ ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
+ ret = -1;
+ }
+ return ret;
+}
+
+ssize_t DeviceVector::remove(const sp<DeviceDescriptor>& item)
+{
+ size_t i;
+ ssize_t ret = indexOf(item);
+
+ if (ret < 0) {
+ ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
+ } else {
+ ret = SortedVector::removeAt(ret);
+ if (ret >= 0) {
+ refreshTypes();
+ }
+ }
+ return ret;
+}
+
+void DeviceVector::loadDevicesFromType(audio_devices_t types)
+{
+ DeviceVector deviceList;
+
+ uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
+ types &= ~role_bit;
+
+ while (types) {
+ uint32_t i = 31 - __builtin_clz(types);
+ uint32_t type = 1 << i;
+ types &= ~type;
+ add(new DeviceDescriptor(String8("device_type"), type | role_bit));
+ }
+}
+
+void DeviceVector::loadDevicesFromName(char *name,
+ const DeviceVector& declaredDevices)
+{
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ if (type != AUDIO_DEVICE_NONE) {
+ sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type);
+ if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
+ type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
+ dev->mAddress = String8("0");
+ }
+ add(dev);
+ } else {
+ sp<DeviceDescriptor> deviceDesc =
+ declaredDevices.getDeviceFromName(String8(devName));
+ if (deviceDesc != 0) {
+ add(deviceDesc);
+ }
+ }
+ }
+ devName = strtok(NULL, "|");
+ }
+}
+
+sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, String8 address) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mDeviceType == type) {
+ if (address == "" || itemAt(i)->mAddress == address) {
+ device = itemAt(i);
+ if (itemAt(i)->mAddress == address) {
+ break;
+ }
+ }
+ }
+ }
+ ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
+ type, address.string(), device.get());
+ return device;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->getHandle() == id) {
+ device = itemAt(i);
+ break;
+ }
+ }
+ return device;
+}
+
+DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const
+{
+ DeviceVector devices;
+ for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
+ if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
+ devices.add(itemAt(i));
+ type &= ~itemAt(i)->mDeviceType;
+ ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
+ itemAt(i)->mDeviceType, itemAt(i).get());
+ }
+ }
+ return devices;
+}
+
+DeviceVector DeviceVector::getDevicesFromTypeAddr(
+ audio_devices_t type, String8 address) const
+{
+ DeviceVector devices;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mDeviceType == type) {
+ if (itemAt(i)->mAddress == address) {
+ devices.add(itemAt(i));
+ }
+ }
+ }
+ return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromName(const String8& name) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mName == name) {
+ device = itemAt(i);
+ break;
+ }
+ }
+ return device;
+}
+
+void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = audio_is_output_device(mDeviceType) ?
+ AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
+ dstConfig->ext.device.type = mDeviceType;
+
+ //TODO Understand why this test is necessary. i.e. why at boot time does it crash
+ // without the test?
+ // This has been demonstrated to NOT be true (at start up)
+ // ALOG_ASSERT(mModule != NULL);
+ dstConfig->ext.device.hw_module = mModule != NULL ? mModule->mHandle : NULL;
+ strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+void DeviceDescriptor::toAudioPort(struct audio_port *port) const
+{
+ ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
+ AudioPort::toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.device.type = mDeviceType;
+ port->ext.device.hw_module = mModule->mHandle;
+ strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+status_t DeviceDescriptor::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ if (mId != 0) {
+ snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
+ result.append(buffer);
+ }
+ snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
+ ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mDeviceType));
+ result.append(buffer);
+ if (mAddress.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+ AudioPort::dump(fd, spaces);
+
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.h b/services/audiopolicy/managerdefault/Devices.h
new file mode 100644
index 0000000..65e1416
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Devices.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class AudioPort;
+class AudioPortConfig;
+
+class DeviceDescriptor: public AudioPort, public AudioPortConfig
+{
+public:
+ DeviceDescriptor(const String8& name, audio_devices_t type);
+
+ virtual ~DeviceDescriptor() {}
+
+ bool equals(const sp<DeviceDescriptor>& other) const;
+
+ // AudioPortConfig
+ virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+
+ // AudioPort
+ virtual void loadGains(cnode *root);
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ status_t dump(int fd, int spaces, int index) const;
+
+ audio_devices_t mDeviceType;
+ String8 mAddress;
+ audio_port_handle_t mId;
+
+ static String8 emptyNameStr;
+};
+
+class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
+{
+public:
+ DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
+
+ ssize_t add(const sp<DeviceDescriptor>& item);
+ ssize_t remove(const sp<DeviceDescriptor>& item);
+ ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
+
+ audio_devices_t types() const { return mDeviceTypes; }
+
+ void loadDevicesFromType(audio_devices_t types);
+ void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
+
+ sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
+ DeviceVector getDevicesFromType(audio_devices_t types) const;
+ sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
+ sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
+ DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
+ const;
+
+private:
+ void refreshTypes();
+ audio_devices_t mDeviceTypes;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.cpp b/services/audiopolicy/managerdefault/Gains.cpp
new file mode 100644
index 0000000..4aca26d
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Gains.cpp
@@ -0,0 +1,446 @@
+/*
+ * Copyright (C) 2015 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 "APM::Gains"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "AudioPolicyManager.h"
+
+#include <math.h>
+
+namespace android {
+
+const VolumeCurvePoint
+ApmGains::sDefaultVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+
+const VolumeCurvePoint
+ApmGains::sDefaultMediaVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sExtMediaSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sHeadsetSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultVoiceVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sLinearVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSilentVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sFullScaleVolumeCurve[ApmGains::VOLCNT] = {
+ {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint *ApmGains::sVolumeProfiles[AUDIO_STREAM_CNT]
+ [ApmGains::DEVICE_CATEGORY_CNT] = {
+ { // AUDIO_STREAM_VOICE_CALL
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_SYSTEM
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_RING
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_MUSIC
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ALARM
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_NOTIFICATION
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_BLUETOOTH_SCO
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ENFORCED_AUDIBLE
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_DTMF
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_TTS
+ // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ACCESSIBILITY
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_REROUTING
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_PATCH
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+};
+
+//static
+audio_devices_t ApmGains::getDeviceForVolume(audio_devices_t device)
+{
+ if (device == AUDIO_DEVICE_NONE) {
+ // this happens when forcing a route update and no track is active on an output.
+ // In this case the returned category is not important.
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (popcount(device) > 1) {
+ // Multiple device selection is either:
+ // - speaker + one other device: give priority to speaker in this case.
+ // - one A2DP device + another device: happens with duplicated output. In this case
+ // retain the device on the A2DP output as the other must not correspond to an active
+ // selection if not the speaker.
+ // - HDMI-CEC system audio mode only output: give priority to available item in order.
+ if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
+ device = AUDIO_DEVICE_OUT_HDMI_ARC;
+ } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
+ device = AUDIO_DEVICE_OUT_AUX_LINE;
+ } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
+ device = AUDIO_DEVICE_OUT_SPDIF;
+ } else {
+ device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+ }
+ }
+
+ /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
+ if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+
+ ALOGW_IF(popcount(device) != 1,
+ "getDeviceForVolume() invalid device combination: %08x",
+ device);
+
+ return device;
+}
+
+//static
+ApmGains::device_category ApmGains::getDeviceCategory(audio_devices_t device)
+{
+ switch(getDeviceForVolume(device)) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ return ApmGains::DEVICE_CATEGORY_EARPIECE;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+ return ApmGains::DEVICE_CATEGORY_HEADSET;
+ case AUDIO_DEVICE_OUT_LINE:
+ case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+ /*USB? Remote submix?*/
+ return ApmGains::DEVICE_CATEGORY_EXT_MEDIA;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+ case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+ case AUDIO_DEVICE_OUT_USB_DEVICE:
+ case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+ default:
+ return ApmGains::DEVICE_CATEGORY_SPEAKER;
+ }
+}
+
+//static
+float ApmGains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi)
+{
+ ApmGains::device_category deviceCategory = ApmGains::getDeviceCategory(device);
+ const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
+
+ // the volume index in the UI is relative to the min and max volume indices for this stream type
+ int nbSteps = 1 + curve[ApmGains::VOLMAX].mIndex -
+ curve[ApmGains::VOLMIN].mIndex;
+ int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
+ (streamDesc.mIndexMax - streamDesc.mIndexMin);
+
+ // find what part of the curve this index volume belongs to, or if it's out of bounds
+ int segment = 0;
+ if (volIdx < curve[ApmGains::VOLMIN].mIndex) { // out of bounds
+ return 0.0f;
+ } else if (volIdx < curve[ApmGains::VOLKNEE1].mIndex) {
+ segment = 0;
+ } else if (volIdx < curve[ApmGains::VOLKNEE2].mIndex) {
+ segment = 1;
+ } else if (volIdx <= curve[ApmGains::VOLMAX].mIndex) {
+ segment = 2;
+ } else { // out of bounds
+ return 1.0f;
+ }
+
+ // linear interpolation in the attenuation table in dB
+ float decibels = curve[segment].mDBAttenuation +
+ ((float)(volIdx - curve[segment].mIndex)) *
+ ( (curve[segment+1].mDBAttenuation -
+ curve[segment].mDBAttenuation) /
+ ((float)(curve[segment+1].mIndex -
+ curve[segment].mIndex)) );
+
+ float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+ ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+ curve[segment].mIndex, volIdx,
+ curve[segment+1].mIndex,
+ curve[segment].mDBAttenuation,
+ decibels,
+ curve[segment+1].mDBAttenuation,
+ amplification);
+
+ return amplification;
+}
+
+
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+ mIndex = index;
+ mUseInChannelMask = useInChannelMask;
+ memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+ config->index = mIndex;
+ config->mode = mGain.mode;
+ config->channel_mask = mGain.channel_mask;
+ if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ config->values[0] = mGain.default_value;
+ } else {
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ config->values[i] = mGain.default_value;
+ }
+ }
+ if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ config->ramp_duration_ms = mGain.min_ramp_ms;
+ }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+ if ((config->mode & ~mGain.mode) != 0) {
+ return BAD_VALUE;
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ if ((config->values[0] < mGain.min_value) ||
+ (config->values[0] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ } else {
+ if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+ return BAD_VALUE;
+ }
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(config->channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(config->channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ if ((config->values[i] < mGain.min_value) ||
+ (config->values[i] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ }
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+ (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+ return BAD_VALUE;
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioGain::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+}
+
+
+// --- StreamDescriptor class implementation
+
+StreamDescriptor::StreamDescriptor()
+ : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+ mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int StreamDescriptor::getVolumeIndex(audio_devices_t device)
+{
+ device = ApmGains::getDeviceForVolume(device);
+ // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+ if (mIndexCur.indexOfKey(device) < 0) {
+ device = AUDIO_DEVICE_OUT_DEFAULT;
+ }
+ return mIndexCur.valueFor(device);
+}
+
+void StreamDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%s %02d %02d ",
+ mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+ result.append(buffer);
+ for (size_t i = 0; i < mIndexCur.size(); i++) {
+ snprintf(buffer, SIZE, "%04x : %02d, ",
+ mIndexCur.keyAt(i),
+ mIndexCur.valueAt(i));
+ result.append(buffer);
+ }
+ result.append("\n");
+
+ write(fd, result.string(), result.size());
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.h b/services/audiopolicy/managerdefault/Gains.h
new file mode 100644
index 0000000..b4ab129
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Gains.h
@@ -0,0 +1,112 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class VolumeCurvePoint
+{
+public:
+ int mIndex;
+ float mDBAttenuation;
+};
+
+class StreamDescriptor;
+
+class ApmGains
+{
+public :
+ // 4 points to define the volume attenuation curve, each characterized by the volume
+ // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
+ // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+ enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
+
+ // device categories used for volume curve management.
+ enum device_category {
+ DEVICE_CATEGORY_HEADSET,
+ DEVICE_CATEGORY_SPEAKER,
+ DEVICE_CATEGORY_EARPIECE,
+ DEVICE_CATEGORY_EXT_MEDIA,
+ DEVICE_CATEGORY_CNT
+ };
+
+ // returns the category the device belongs to with regard to volume curve management
+ static ApmGains::device_category getDeviceCategory(audio_devices_t device);
+
+ // extract one device relevant for volume control from multiple device selection
+ static audio_devices_t getDeviceForVolume(audio_devices_t device);
+
+ static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi);
+
+ // default volume curve
+ static const VolumeCurvePoint sDefaultVolumeCurve[ApmGains::VOLCNT];
+ // default volume curve for media strategy
+ static const VolumeCurvePoint sDefaultMediaVolumeCurve[ApmGains::VOLCNT];
+ // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
+ static const VolumeCurvePoint sExtMediaSystemVolumeCurve[ApmGains::VOLCNT];
+ // volume curve for media strategy on speakers
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT];
+ // volume curve for sonification strategy on speakers
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sHeadsetSystemVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultVoiceVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sLinearVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSilentVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sFullScaleVolumeCurve[ApmGains::VOLCNT];
+ // default volume curves per stream and device category. See initializeVolumeCurves()
+ static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+
+class AudioGain: public RefBase
+{
+public:
+ AudioGain(int index, bool useInChannelMask);
+ virtual ~AudioGain() {}
+
+ void dump(int fd, int spaces, int index) const;
+
+ void getDefaultConfig(struct audio_gain_config *config);
+ status_t checkConfig(const struct audio_gain_config *config);
+ int mIndex;
+ struct audio_gain mGain;
+ bool mUseInChannelMask;
+};
+
+
+// stream descriptor used for volume control
+class StreamDescriptor
+{
+public:
+ StreamDescriptor();
+
+ int getVolumeIndex(audio_devices_t device);
+ void dump(int fd);
+
+ int mIndexMin; // min volume index
+ int mIndexMax; // max volume index
+ KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device
+ bool mCanBeMuted; // true is the stream can be muted
+
+ const VolumeCurvePoint *mVolumeCurve[ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.cpp b/services/audiopolicy/managerdefault/HwModule.cpp
new file mode 100644
index 0000000..a04bdc8
--- /dev/null
+++ b/services/audiopolicy/managerdefault/HwModule.cpp
@@ -0,0 +1,279 @@
+/*
+ * Copyright (C) 2015 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 "APM::HwModule"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+#include "audio_policy_conf.h"
+#include <hardware/audio.h>
+
+namespace android {
+
+HwModule::HwModule(const char *name)
+ : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
+ mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
+{
+}
+
+HwModule::~HwModule()
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ mOutputProfiles[i]->mSupportedDevices.clear();
+ }
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ mInputProfiles[i]->mSupportedDevices.clear();
+ }
+ free((void *)mName);
+}
+
+status_t HwModule::loadInput(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ profile->loadSamplingRates((char *)node->value);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ profile->loadFormats((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ profile->loadInChannels((char *)node->value);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+ mDeclaredDevices);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = ConfigParsingUtils::parseInputFlagNames((char *)node->value);
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ profile->loadGains(node);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+ "loadInput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadInput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadInput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadInput() invalid supported formats");
+ if (!profile->mSupportedDevices.isEmpty() &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadInput() adding input Supported Devices %04x",
+ profile->mSupportedDevices.types());
+
+ mInputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t HwModule::loadOutput(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ profile->loadSamplingRates((char *)node->value);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ profile->loadFormats((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ profile->loadOutChannels((char *)node->value);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+ mDeclaredDevices);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = ConfigParsingUtils::parseOutputFlagNames((char *)node->value);
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ profile->loadGains(node);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+ "loadOutput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadOutput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadOutput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadOutput() invalid supported formats");
+ if (!profile->mSupportedDevices.isEmpty() &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
+ profile->mSupportedDevices.types(), profile->mFlags);
+
+ mOutputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t HwModule::loadDevice(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ audio_devices_t type = AUDIO_DEVICE_NONE;
+ while (node) {
+ if (strcmp(node->name, DEVICE_TYPE) == 0) {
+ type = ConfigParsingUtils::parseDeviceNames((char *)node->value);
+ break;
+ }
+ node = node->next;
+ }
+ if (type == AUDIO_DEVICE_NONE ||
+ (!audio_is_input_device(type) && !audio_is_output_device(type))) {
+ ALOGW("loadDevice() bad type %08x", type);
+ return BAD_VALUE;
+ }
+ sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
+ deviceDesc->mModule = this;
+
+ node = root->first_child;
+ while (node) {
+ if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
+ deviceDesc->mAddress = String8((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ if (audio_is_input_device(type)) {
+ deviceDesc->loadInChannels((char *)node->value);
+ } else {
+ deviceDesc->loadOutChannels((char *)node->value);
+ }
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ deviceDesc->loadGains(node);
+ }
+ node = node->next;
+ }
+
+ ALOGV("loadDevice() adding device name %s type %08x address %s",
+ deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
+
+ mDeclaredDevices.add(deviceDesc);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ mOutputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::removeOutputProfile(String8 name)
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ if (mOutputProfiles[i]->mName == name) {
+ mOutputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+status_t HwModule::addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
+
+ mInputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::removeInputProfile(String8 name)
+{
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ if (mInputProfiles[i]->mName == name) {
+ mInputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+
+void HwModule::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " - name: %s\n", mName);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ if (mOutputProfiles.size()) {
+ write(fd, " - outputs:\n", strlen(" - outputs:\n"));
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " output %zu:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mOutputProfiles[i]->dump(fd);
+ }
+ }
+ if (mInputProfiles.size()) {
+ write(fd, " - inputs:\n", strlen(" - inputs:\n"));
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " input %zu:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mInputProfiles[i]->dump(fd);
+ }
+ }
+ if (mDeclaredDevices.size()) {
+ write(fd, " - devices:\n", strlen(" - devices:\n"));
+ for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
+ mDeclaredDevices[i]->dump(fd, 4, i);
+ }
+ }
+}
+
+} //namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.h b/services/audiopolicy/managerdefault/HwModule.h
new file mode 100644
index 0000000..f814dd9
--- /dev/null
+++ b/services/audiopolicy/managerdefault/HwModule.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class HwModule : public RefBase
+{
+public:
+ HwModule(const char *name);
+ ~HwModule();
+
+ status_t loadOutput(cnode *root);
+ status_t loadInput(cnode *root);
+ status_t loadDevice(cnode *root);
+
+ status_t addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeOutputProfile(String8 name);
+ status_t addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeInputProfile(String8 name);
+
+ void dump(int fd);
+
+ const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+ uint32_t mHalVersion; // audio HAL API version
+ audio_module_handle_t mHandle;
+ Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
+ Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module
+ DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.cpp b/services/audiopolicy/managerdefault/IOProfile.cpp
new file mode 100644
index 0000000..538ac1a
--- /dev/null
+++ b/services/audiopolicy/managerdefault/IOProfile.cpp
@@ -0,0 +1,139 @@
+/*
+ * Copyright (C) 2015 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 "APM::IOProfile"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+IOProfile::IOProfile(const String8& name, audio_port_role_t role,
+ const sp<HwModule>& module)
+ : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
+{
+}
+
+IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool IOProfile::isCompatibleProfile(audio_devices_t device,
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const
+{
+ const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
+ const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+ ALOG_ASSERT(isPlaybackThread != isRecordThread);
+
+ if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
+ return false;
+ }
+
+ if (samplingRate == 0) {
+ return false;
+ }
+ uint32_t myUpdatedSamplingRate = samplingRate;
+ if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
+ return false;
+ }
+ if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
+ NO_ERROR) {
+ return false;
+ }
+
+ if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
+ return false;
+ }
+
+ if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
+ checkExactChannelMask(channelMask) != NO_ERROR)) {
+ return false;
+ }
+ if (isRecordThread && (!audio_is_input_channel(channelMask) ||
+ checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
+ return false;
+ }
+
+ if (isPlaybackThread && (mFlags & flags) != flags) {
+ return false;
+ }
+ // The only input flag that is allowed to be different is the fast flag.
+ // An existing fast stream is compatible with a normal track request.
+ // An existing normal stream is compatible with a fast track request,
+ // but the fast request will be denied by AudioFlinger and converted to normal track.
+ if (isRecordThread && ((mFlags ^ flags) &
+ ~AUDIO_INPUT_FLAG_FAST)) {
+ return false;
+ }
+
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = myUpdatedSamplingRate;
+ }
+ return true;
+}
+
+void IOProfile::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ AudioPort::dump(fd, 4);
+
+ snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - devices:\n");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ for (size_t i = 0; i < mSupportedDevices.size(); i++) {
+ mSupportedDevices[i]->dump(fd, 6, i);
+ }
+}
+
+void IOProfile::log()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ ALOGV(" - sampling rates: ");
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ ALOGV(" %d", mSamplingRates[i]);
+ }
+
+ ALOGV(" - channel masks: ");
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ ALOGV(" 0x%04x", mChannelMasks[i]);
+ }
+
+ ALOGV(" - formats: ");
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ ALOGV(" 0x%08x", mFormats[i]);
+ }
+
+ ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
+ ALOGV(" - flags: 0x%04x\n", mFlags);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.h b/services/audiopolicy/managerdefault/IOProfile.h
new file mode 100644
index 0000000..3317969
--- /dev/null
+++ b/services/audiopolicy/managerdefault/IOProfile.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class HwModule;
+
+// the IOProfile class describes the capabilities of an output or input stream.
+// It is currently assumed that all combination of listed parameters are supported.
+// It is used by the policy manager to determine if an output or input is suitable for
+// a given use case, open/close it accordingly and connect/disconnect audio tracks
+// to/from it.
+class IOProfile : public AudioPort
+{
+public:
+ IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
+ virtual ~IOProfile();
+
+ // This method is used for both output and input.
+ // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
+ // For input, flags is interpreted as audio_input_flags_t.
+ // TODO: merge audio_output_flags_t and audio_input_flags_t.
+ bool isCompatibleProfile(audio_devices_t device,
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const;
+
+ void dump(int fd);
+ void log();
+
+ DeviceVector mSupportedDevices; // supported devices
+ // (devices this output can be routed to)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.cpp b/services/audiopolicy/managerdefault/Ports.cpp
new file mode 100644
index 0000000..3e55cee
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Ports.cpp
@@ -0,0 +1,844 @@
+/*
+ * Copyright (C) 2015 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 "APM::Ports"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+#include "audio_policy_conf.h"
+
+namespace android {
+
+// --- AudioPort class implementation
+
+AudioPort::AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module) :
+ mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0)
+{
+ mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
+ ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
+}
+
+void AudioPort::attach(const sp<HwModule>& module) {
+ mId = AudioPolicyManager::nextUniqueId();
+ mModule = module;
+}
+
+void AudioPort::toAudioPort(struct audio_port *port) const
+{
+ port->role = mRole;
+ port->type = mType;
+ strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
+ unsigned int i;
+ for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
+ if (mSamplingRates[i] != 0) {
+ port->sample_rates[i] = mSamplingRates[i];
+ }
+ }
+ port->num_sample_rates = i;
+ for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
+ if (mChannelMasks[i] != 0) {
+ port->channel_masks[i] = mChannelMasks[i];
+ }
+ }
+ port->num_channel_masks = i;
+ for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
+ if (mFormats[i] != 0) {
+ port->formats[i] = mFormats[i];
+ }
+ }
+ port->num_formats = i;
+
+ ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+
+ for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
+ port->gains[i] = mGains[i]->mGain;
+ }
+ port->num_gains = i;
+}
+
+void AudioPort::importAudioPort(const sp<AudioPort> port) {
+ for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
+ const uint32_t rate = port->mSamplingRates.itemAt(k);
+ if (rate != 0) { // skip "dynamic" rates
+ bool hasRate = false;
+ for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
+ if (rate == mSamplingRates.itemAt(l)) {
+ hasRate = true;
+ break;
+ }
+ }
+ if (!hasRate) { // never import a sampling rate twice
+ mSamplingRates.add(rate);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
+ const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
+ if (mask != 0) { // skip "dynamic" masks
+ bool hasMask = false;
+ for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
+ if (mask == mChannelMasks.itemAt(l)) {
+ hasMask = true;
+ break;
+ }
+ }
+ if (!hasMask) { // never import a channel mask twice
+ mChannelMasks.add(mask);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
+ const audio_format_t format = port->mFormats.itemAt(k);
+ if (format != 0) { // skip "dynamic" formats
+ bool hasFormat = false;
+ for (size_t l = 0 ; l < mFormats.size() ; l++) {
+ if (format == mFormats.itemAt(l)) {
+ hasFormat = true;
+ break;
+ }
+ }
+ if (!hasFormat) { // never import a channel mask twice
+ mFormats.add(format);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mGains.size() ; k++) {
+ sp<AudioGain> gain = port->mGains.itemAt(k);
+ if (gain != 0) {
+ bool hasGain = false;
+ for (size_t l = 0 ; l < mGains.size() ; l++) {
+ if (gain == mGains.itemAt(l)) {
+ hasGain = true;
+ break;
+ }
+ }
+ if (!hasGain) { // never import a gain twice
+ mGains.add(gain);
+ }
+ }
+ }
+}
+
+void AudioPort::clearCapabilities() {
+ mChannelMasks.clear();
+ mFormats.clear();
+ mSamplingRates.clear();
+ mGains.clear();
+}
+
+void AudioPort::loadSamplingRates(char *name)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+ // rates should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mSamplingRates.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ uint32_t rate = atoi(str);
+ if (rate != 0) {
+ ALOGV("loadSamplingRates() adding rate %d", rate);
+ mSamplingRates.add(rate);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadFormats(char *name)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mFormats indicates the supported formats
+ // should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mFormats.add(AUDIO_FORMAT_DEFAULT);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_format_t format = (audio_format_t)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ str);
+ if (format != AUDIO_FORMAT_DEFAULT) {
+ mFormats.add(format);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadInChannels(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadInChannels() %s", name);
+
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+ mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadOutChannels(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadOutChannels() %s", name);
+
+ // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+ // masks should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+ return;
+}
+
+audio_gain_mode_t AudioPort::loadGainMode(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadGainMode() %s", name);
+ audio_gain_mode_t mode = 0;
+ while (str != NULL) {
+ mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable,
+ ARRAY_SIZE(sGainModeNameToEnumTable),
+ str);
+ str = strtok(NULL, "|");
+ }
+ return mode;
+}
+
+void AudioPort::loadGain(cnode *root, int index)
+{
+ cnode *node = root->first_child;
+
+ sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
+
+ while (node) {
+ if (strcmp(node->name, GAIN_MODE) == 0) {
+ gain->mGain.mode = loadGainMode((char *)node->value);
+ } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
+ if (mUseInChannelMask) {
+ gain->mGain.channel_mask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ (char *)node->value);
+ } else {
+ gain->mGain.channel_mask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ (char *)node->value);
+ }
+ } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
+ gain->mGain.min_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
+ gain->mGain.max_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
+ gain->mGain.default_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
+ gain->mGain.step_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
+ gain->mGain.min_ramp_ms = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
+ gain->mGain.max_ramp_ms = atoi((char *)node->value);
+ }
+ node = node->next;
+ }
+
+ ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
+ gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
+
+ if (gain->mGain.mode == 0) {
+ return;
+ }
+ mGains.add(gain);
+}
+
+void AudioPort::loadGains(cnode *root)
+{
+ cnode *node = root->first_child;
+ int index = 0;
+ while (node) {
+ ALOGV("loadGains() loading gain %s", node->name);
+ loadGain(node, index++);
+ node = node->next;
+ }
+}
+
+status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if (mSamplingRates[i] == samplingRate) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t *updatedSamplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ // Search for the closest supported sampling rate that is above (preferred)
+ // or below (acceptable) the desired sampling rate, within a permitted ratio.
+ // The sampling rates do not need to be sorted in ascending order.
+ ssize_t maxBelow = -1;
+ ssize_t minAbove = -1;
+ uint32_t candidate;
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ candidate = mSamplingRates[i];
+ if (candidate == samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ // candidate < desired
+ if (candidate < samplingRate) {
+ if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
+ maxBelow = i;
+ }
+ // candidate > desired
+ } else {
+ if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
+ minAbove = i;
+ }
+ }
+ }
+ // This uses hard-coded knowledge about AudioFlinger resampling ratios.
+ // TODO Move these assumptions out.
+ static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
+ static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
+ // due to approximation by an int32_t of the
+ // phase increments
+ // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
+ if (minAbove >= 0) {
+ candidate = mSamplingRates[minAbove];
+ if (candidate / kMaxDownSampleRatio <= samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ }
+ // But if we have to up-sample from a lower sampling rate, that's OK.
+ if (maxBelow >= 0) {
+ candidate = mSamplingRates[maxBelow];
+ if (candidate * kMaxUpSampleRatio >= samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ }
+ // leave updatedSamplingRate unmodified
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
+{
+ if (mChannelMasks.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ if (mChannelMasks[i] == channelMask) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
+ const
+{
+ if (mChannelMasks.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ // FIXME Does not handle multi-channel automatic conversions yet
+ audio_channel_mask_t supported = mChannelMasks[i];
+ if (supported == channelMask) {
+ return NO_ERROR;
+ }
+ if (isRecordThread) {
+ // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
+ // FIXME Abstract this out to a table.
+ if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
+ && channelMask == AUDIO_CHANNEL_IN_MONO) ||
+ (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
+ || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
+ return NO_ERROR;
+ }
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkFormat(audio_format_t format) const
+{
+ if (mFormats.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if (mFormats[i] == format) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+
+uint32_t AudioPort::pickSamplingRate() const
+{
+ // special case for uninitialized dynamic profile
+ if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
+ return 0;
+ }
+
+ // For direct outputs, pick minimum sampling rate: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t samplingRate = UINT_MAX;
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return (samplingRate == UINT_MAX) ? 0 : samplingRate;
+ }
+
+ uint32_t samplingRate = 0;
+ uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
+
+ // For mixed output and inputs, use max mixer sampling rates. Do not
+ // limit sampling rate otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxRate = UINT_MAX;
+ }
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return samplingRate;
+}
+
+audio_channel_mask_t AudioPort::pickChannelMask() const
+{
+ // special case for uninitialized dynamic profile
+ if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
+ return AUDIO_CHANNEL_NONE;
+ }
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
+
+ // For direct outputs, pick minimum channel count: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t channelCount = UINT_MAX;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount < channelCount) && (cnlCount > 0)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+ }
+
+ uint32_t channelCount = 0;
+ uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
+
+ // For mixed output and inputs, use max mixer channel count. Do not
+ // limit channel count otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxCount = UINT_MAX;
+ }
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+}
+
+/* format in order of increasing preference */
+const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
+ AUDIO_FORMAT_DEFAULT,
+ AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_FORMAT_PCM_8_24_BIT,
+ AUDIO_FORMAT_PCM_24_BIT_PACKED,
+ AUDIO_FORMAT_PCM_32_BIT,
+ AUDIO_FORMAT_PCM_FLOAT,
+};
+
+int AudioPort::compareFormats(audio_format_t format1,
+ audio_format_t format2)
+{
+ // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
+ // compressed format and better than any PCM format. This is by design of pickFormat()
+ if (!audio_is_linear_pcm(format1)) {
+ if (!audio_is_linear_pcm(format2)) {
+ return 0;
+ }
+ return 1;
+ }
+ if (!audio_is_linear_pcm(format2)) {
+ return -1;
+ }
+
+ int index1 = -1, index2 = -1;
+ for (size_t i = 0;
+ (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
+ i ++) {
+ if (sPcmFormatCompareTable[i] == format1) {
+ index1 = i;
+ }
+ if (sPcmFormatCompareTable[i] == format2) {
+ index2 = i;
+ }
+ }
+ // format1 not found => index1 < 0 => format2 > format1
+ // format2 not found => index2 < 0 => format2 < format1
+ return index1 - index2;
+}
+
+audio_format_t AudioPort::pickFormat() const
+{
+ // special case for uninitialized dynamic profile
+ if (mFormats.size() == 1 && mFormats[0] == 0) {
+ return AUDIO_FORMAT_DEFAULT;
+ }
+
+ audio_format_t format = AUDIO_FORMAT_DEFAULT;
+ audio_format_t bestFormat =
+ AudioPort::sPcmFormatCompareTable[
+ ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1];
+ // For mixed output and inputs, use best mixer output format. Do not
+ // limit format otherwise
+ if ((mType != AUDIO_PORT_TYPE_MIX) ||
+ ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
+ bestFormat = AUDIO_FORMAT_INVALID;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if ((compareFormats(mFormats[i], format) > 0) &&
+ (compareFormats(mFormats[i], bestFormat) <= 0)) {
+ format = mFormats[i];
+ }
+ }
+ return format;
+}
+
+status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig,
+ int index) const
+{
+ if (index < 0 || (size_t)index >= mGains.size()) {
+ return BAD_VALUE;
+ }
+ return mGains[index]->checkConfig(gainConfig);
+}
+
+void AudioPort::dump(int fd, int spaces) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ if (mName.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
+ result.append(buffer);
+ }
+
+ if (mSamplingRates.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ if (i == 0 && mSamplingRates[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ }
+ result.append(buffer);
+ result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (mChannelMasks.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
+
+ if (i == 0 && mChannelMasks[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+ }
+ result.append(buffer);
+ result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (mFormats.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ mFormats[i]);
+ if (i == 0 && strcmp(formatStr, "") == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%s", formatStr);
+ }
+ result.append(buffer);
+ result.append(i == (mFormats.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+ write(fd, result.string(), result.size());
+ if (mGains.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
+ write(fd, buffer, strlen(buffer) + 1);
+ result.append(buffer);
+ for (size_t i = 0; i < mGains.size(); i++) {
+ mGains[i]->dump(fd, spaces + 2, i);
+ }
+ }
+}
+
+
+// --- AudioPortConfig class implementation
+
+AudioPortConfig::AudioPortConfig()
+{
+ mSamplingRate = 0;
+ mChannelMask = AUDIO_CHANNEL_NONE;
+ mFormat = AUDIO_FORMAT_INVALID;
+ mGain.index = -1;
+}
+
+status_t AudioPortConfig::applyAudioPortConfig(
+ const struct audio_port_config *config,
+ struct audio_port_config *backupConfig)
+{
+ struct audio_port_config localBackupConfig;
+ status_t status = NO_ERROR;
+
+ localBackupConfig.config_mask = config->config_mask;
+ toAudioPortConfig(&localBackupConfig);
+
+ sp<AudioPort> audioport = getAudioPort();
+ if (audioport == 0) {
+ status = NO_INIT;
+ goto exit;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ status = audioport->checkExactSamplingRate(config->sample_rate);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mSamplingRate = config->sample_rate;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ status = audioport->checkExactChannelMask(config->channel_mask);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mChannelMask = config->channel_mask;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ status = audioport->checkFormat(config->format);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mFormat = config->format;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ status = audioport->checkGain(&config->gain, config->gain.index);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mGain = config->gain;
+ }
+
+exit:
+ if (status != NO_ERROR) {
+ applyAudioPortConfig(&localBackupConfig);
+ }
+ if (backupConfig != NULL) {
+ *backupConfig = localBackupConfig;
+ }
+ return status;
+}
+
+void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ dstConfig->sample_rate = mSamplingRate;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
+ dstConfig->sample_rate = srcConfig->sample_rate;
+ }
+ } else {
+ dstConfig->sample_rate = 0;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ dstConfig->channel_mask = mChannelMask;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
+ dstConfig->channel_mask = srcConfig->channel_mask;
+ }
+ } else {
+ dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ dstConfig->format = mFormat;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
+ dstConfig->format = srcConfig->format;
+ }
+ } else {
+ dstConfig->format = AUDIO_FORMAT_INVALID;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ dstConfig->gain = mGain;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
+ dstConfig->gain = srcConfig->gain;
+ }
+ } else {
+ dstConfig->gain.index = -1;
+ }
+ if (dstConfig->gain.index != -1) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
+ } else {
+ dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+ }
+}
+
+
+// --- AudioPatch class implementation
+
+AudioPatch::AudioPatch(audio_patch_handle_t handle,
+ const struct audio_patch *patch, uid_t uid) :
+ mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0)
+{}
+
+status_t AudioPatch::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
+ result.append(buffer);
+ for (size_t i = 0; i < mPatch.num_sources; i++) {
+ if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
+ snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+ mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mPatch.sources[i].ext.device.type));
+ } else {
+ snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+ mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
+ }
+ result.append(buffer);
+ }
+ snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
+ result.append(buffer);
+ for (size_t i = 0; i < mPatch.num_sinks; i++) {
+ if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
+ snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+ mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mPatch.sinks[i].ext.device.type));
+ } else {
+ snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+ mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
+ }
+ result.append(buffer);
+ }
+
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.h b/services/audiopolicy/managerdefault/Ports.h
new file mode 100644
index 0000000..f6e0e93
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Ports.h
@@ -0,0 +1,122 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+namespace android {
+
+class HwModule;
+
+class AudioPort: public virtual RefBase
+{
+public:
+ AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module);
+ virtual ~AudioPort() {}
+
+ audio_port_handle_t getHandle() { return mId; }
+
+ void attach(const sp<HwModule>& module);
+ bool isAttached() { return mId != 0; }
+
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ void importAudioPort(const sp<AudioPort> port);
+ void clearCapabilities();
+
+ void loadSamplingRates(char *name);
+ void loadFormats(char *name);
+ void loadOutChannels(char *name);
+ void loadInChannels(char *name);
+
+ audio_gain_mode_t loadGainMode(char *name);
+ void loadGain(cnode *root, int index);
+ virtual void loadGains(cnode *root);
+
+ // searches for an exact match
+ status_t checkExactSamplingRate(uint32_t samplingRate) const;
+ // searches for a compatible match, and returns the best match via updatedSamplingRate
+ status_t checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t *updatedSamplingRate) const;
+ // searches for an exact match
+ status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
+ // searches for a compatible match, currently implemented for input channel masks only
+ status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
+ status_t checkFormat(audio_format_t format) const;
+ status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
+
+ uint32_t pickSamplingRate() const;
+ audio_channel_mask_t pickChannelMask() const;
+ audio_format_t pickFormat() const;
+
+ static const audio_format_t sPcmFormatCompareTable[];
+ static int compareFormats(audio_format_t format1, audio_format_t format2);
+
+ void dump(int fd, int spaces) const;
+
+ String8 mName;
+ audio_port_type_t mType;
+ audio_port_role_t mRole;
+ bool mUseInChannelMask;
+ // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
+ // indicates the supported parameters should be read from the output stream
+ // after it is opened for the first time
+ Vector <uint32_t> mSamplingRates; // supported sampling rates
+ Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
+ Vector <audio_format_t> mFormats; // supported audio formats
+ Vector < sp<AudioGain> > mGains; // gain controllers
+ sp<HwModule> mModule; // audio HW module exposing this I/O stream
+ uint32_t mFlags; // attribute flags (e.g primary output,
+ // direct output...).
+
+
+protected:
+ //TODO - clarify the role of mId in this case, both an "attached" indicator
+ // and a unique ID for identifying a port to the (upcoming) selection API,
+ // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor.
+ audio_port_handle_t mId;
+};
+
+class AudioPortConfig: public virtual RefBase
+{
+public:
+ AudioPortConfig();
+ virtual ~AudioPortConfig() {}
+
+ status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const = 0;
+ virtual sp<AudioPort> getAudioPort() const = 0;
+ uint32_t mSamplingRate;
+ audio_format_t mFormat;
+ audio_channel_mask_t mChannelMask;
+ struct audio_gain_config mGain;
+};
+
+
+class AudioPatch: public RefBase
+{
+public:
+ AudioPatch(audio_patch_handle_t handle, const struct audio_patch *patch, uid_t uid);
+
+ status_t dump(int fd, int spaces, int index) const;
+
+ audio_patch_handle_t mHandle;
+ struct audio_patch mPatch;
+ uid_t mUid;
+ audio_patch_handle_t mAfPatchHandle;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/audio_policy_conf.h b/services/audiopolicy/managerdefault/audio_policy_conf.h
new file mode 100644
index 0000000..2535a67
--- /dev/null
+++ b/services/audiopolicy/managerdefault/audio_policy_conf.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+
+#ifndef ANDROID_AUDIO_POLICY_CONF_H
+#define ANDROID_AUDIO_POLICY_CONF_H
+
+
+/////////////////////////////////////////////////
+// Definitions for audio policy configuration file (audio_policy.conf)
+/////////////////////////////////////////////////
+
+#define AUDIO_HARDWARE_MODULE_ID_MAX_LEN 32
+
+#define AUDIO_POLICY_CONFIG_FILE "/system/etc/audio_policy.conf"
+#define AUDIO_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_policy.conf"
+
+// global configuration
+#define GLOBAL_CONFIG_TAG "global_configuration"
+
+#define ATTACHED_OUTPUT_DEVICES_TAG "attached_output_devices"
+#define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device"
+#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices"
+#define SPEAKER_DRC_ENABLED_TAG "speaker_drc_enabled"
+#define AUDIO_HAL_VERSION_TAG "audio_hal_version"
+
+// hw modules descriptions
+#define AUDIO_HW_MODULE_TAG "audio_hw_modules"
+
+#define OUTPUTS_TAG "outputs"
+#define INPUTS_TAG "inputs"
+
+#define SAMPLING_RATES_TAG "sampling_rates"
+#define FORMATS_TAG "formats"
+#define CHANNELS_TAG "channel_masks"
+#define DEVICES_TAG "devices"
+#define FLAGS_TAG "flags"
+
+#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and
+ // "formats" in outputs descriptors indicating that supported
+ // values should be queried after opening the output.
+
+#define DEVICES_TAG "devices"
+#define DEVICE_TYPE "type"
+#define DEVICE_ADDRESS "address"
+
+#define MIXERS_TAG "mixers"
+#define MIXER_TYPE "type"
+#define MIXER_TYPE_MUX "mux"
+#define MIXER_TYPE_MIX "mix"
+
+#define GAINS_TAG "gains"
+#define GAIN_MODE "mode"
+#define GAIN_CHANNELS "channel_mask"
+#define GAIN_MIN_VALUE "min_value_mB"
+#define GAIN_MAX_VALUE "max_value_mB"
+#define GAIN_DEFAULT_VALUE "default_value_mB"
+#define GAIN_STEP_VALUE "step_value_mB"
+#define GAIN_MIN_RAMP_MS "min_ramp_ms"
+#define GAIN_MAX_RAMP_MS "max_ramp_ms"
+
+
+
+#endif // ANDROID_AUDIO_POLICY_CONF_H