diff options
Diffstat (limited to 'services/audiopolicy/managerdefault')
20 files changed, 9562 insertions, 0 deletions
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..53ec0f6 --- /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 (!isInCall()) { + 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..b2d9763 --- /dev/null +++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.h @@ -0,0 +1,161 @@ +/* + * 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 } +#ifndef ARRAY_SIZE +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) +#endif + +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..5b1401e --- /dev/null +++ b/services/audiopolicy/managerdefault/Devices.cpp @@ -0,0 +1,286 @@ +/* + * 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; + bool isOutput = audio_is_output_devices(type); + type &= ~AUDIO_DEVICE_BIT_IN; + for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) { + bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType); + audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN; + if ((isOutput == curIsOutput) && ((type & curType) != 0)) { + devices.add(itemAt(i)); + type &= ~curType; + 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 != 0 ? mModule->mHandle : AUDIO_IO_HANDLE_NONE; + 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..af2fbda --- /dev/null +++ b/services/audiopolicy/managerdefault/Devices.h @@ -0,0 +1,74 @@ +/* + * 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; + + 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..8000914 --- /dev/null +++ b/services/audiopolicy/managerdefault/IOProfile.cpp @@ -0,0 +1,147 @@ +/* + * 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) { + // just check types if multiple devices are selected + if (popcount(device & ~AUDIO_DEVICE_BIT_IN) > 1) { + if ((mSupportedDevices.types() & device) != device) { + return false; + } + } else if (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 |