From 98cc191247388132b6fd8a4ecd07abd6e4c5a0ed Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Fran=C3=A7ois=20Gaffie?= Date: Wed, 18 Mar 2015 17:52:40 +0100 Subject: Split ManagerDefault into Manager and ManagerDefinitions MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch splits the managerdefault into a manager and a manager defintion library that contains all pillar elements of a policy manager. It renames the file with the name of the main class they contains. It splits the AudioPort into AudioPort and AudioPatch. Change-Id: I992cf0b8aed895805cc003ba0980d2c9e92c985b Signed-off-by: François Gaffie --- media/mediaserver/Android.mk | 3 +- services/audiopolicy/Android.mk | 23 +- services/audiopolicy/common/Android.mk | 9 + .../common/managerdefinitions/Android.mk | 29 + .../include/ApmImplDefinitions.h | 34 + .../common/managerdefinitions/include/AudioGain.h | 119 +++ .../include/AudioInputDescriptor.h | 59 ++ .../include/AudioOutputDescriptor.h | 76 ++ .../common/managerdefinitions/include/AudioPatch.h | 41 + .../common/managerdefinitions/include/AudioPort.h | 126 +++ .../include/ConfigParsingUtils.h | 166 ++++ .../managerdefinitions/include/DeviceDescriptor.h | 80 ++ .../common/managerdefinitions/include/HwModule.h | 58 ++ .../common/managerdefinitions/include/IOProfile.h | 58 ++ .../managerdefinitions/include/audio_policy_conf.h | 70 ++ .../common/managerdefinitions/src/AudioGain.cpp | 448 +++++++++++ .../src/AudioInputDescriptor.cpp | 110 +++ .../src/AudioOutputDescriptor.cpp | 223 ++++++ .../common/managerdefinitions/src/AudioPatch.cpp | 84 ++ .../common/managerdefinitions/src/AudioPort.cpp | 798 +++++++++++++++++++ .../managerdefinitions/src/ConfigParsingUtils.cpp | 122 +++ .../managerdefinitions/src/DeviceDescriptor.cpp | 289 +++++++ .../common/managerdefinitions/src/HwModule.cpp | 282 +++++++ .../common/managerdefinitions/src/IOProfile.cpp | 149 ++++ .../managerdefault/ApmImplDefinitions.h | 34 - .../managerdefault/AudioInputDescriptor.cpp | 104 --- .../managerdefault/AudioInputDescriptor.h | 58 -- .../managerdefault/AudioOutputDescriptor.cpp | 217 ------ .../managerdefault/AudioOutputDescriptor.h | 75 -- .../managerdefault/AudioPolicyManager.cpp | 29 +- .../managerdefault/AudioPolicyManager.h | 20 +- .../managerdefault/ConfigParsingUtils.cpp | 122 --- .../managerdefault/ConfigParsingUtils.h | 166 ---- services/audiopolicy/managerdefault/Devices.cpp | 289 ------- services/audiopolicy/managerdefault/Devices.h | 80 -- services/audiopolicy/managerdefault/Gains.cpp | 448 ----------- services/audiopolicy/managerdefault/Gains.h | 119 --- services/audiopolicy/managerdefault/HwModule.cpp | 282 ------- services/audiopolicy/managerdefault/HwModule.h | 58 -- services/audiopolicy/managerdefault/IOProfile.cpp | 149 ---- services/audiopolicy/managerdefault/IOProfile.h | 58 -- services/audiopolicy/managerdefault/Ports.cpp | 848 --------------------- services/audiopolicy/managerdefault/Ports.h | 138 ---- .../audiopolicy/managerdefault/audio_policy_conf.h | 70 -- 44 files changed, 3461 insertions(+), 3359 deletions(-) create mode 100644 services/audiopolicy/common/Android.mk create mode 100644 services/audiopolicy/common/managerdefinitions/Android.mk create mode 100644 services/audiopolicy/common/managerdefinitions/include/ApmImplDefinitions.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/AudioGain.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/AudioPatch.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/AudioPort.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/HwModule.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/IOProfile.h create mode 100644 services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h create mode 100644 services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp create mode 100644 services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp create mode 100644 services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp create mode 100644 services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp create mode 100644 services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp create mode 100644 services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp create mode 100644 services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp create mode 100644 services/audiopolicy/common/managerdefinitions/src/HwModule.cpp create mode 100644 services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp delete mode 100644 services/audiopolicy/managerdefault/ApmImplDefinitions.h delete mode 100644 services/audiopolicy/managerdefault/AudioInputDescriptor.cpp delete mode 100644 services/audiopolicy/managerdefault/AudioInputDescriptor.h delete mode 100644 services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp delete mode 100644 services/audiopolicy/managerdefault/AudioOutputDescriptor.h delete mode 100644 services/audiopolicy/managerdefault/ConfigParsingUtils.cpp delete mode 100644 services/audiopolicy/managerdefault/ConfigParsingUtils.h delete mode 100644 services/audiopolicy/managerdefault/Devices.cpp delete mode 100644 services/audiopolicy/managerdefault/Devices.h delete mode 100644 services/audiopolicy/managerdefault/Gains.cpp delete mode 100644 services/audiopolicy/managerdefault/Gains.h delete mode 100644 services/audiopolicy/managerdefault/HwModule.cpp delete mode 100644 services/audiopolicy/managerdefault/HwModule.h delete mode 100644 services/audiopolicy/managerdefault/IOProfile.cpp delete mode 100644 services/audiopolicy/managerdefault/IOProfile.h delete mode 100644 services/audiopolicy/managerdefault/Ports.cpp delete mode 100644 services/audiopolicy/managerdefault/Ports.h delete mode 100644 services/audiopolicy/managerdefault/audio_policy_conf.h diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk index baf65f6..0f36a90 100644 --- a/media/mediaserver/Android.mk +++ b/media/mediaserver/Android.mk @@ -31,13 +31,14 @@ LOCAL_SHARED_LIBRARIES := \ libradioservice LOCAL_STATIC_LIBRARIES := \ - libregistermsext + libregistermsext LOCAL_C_INCLUDES := \ frameworks/av/media/libmediaplayerservice \ frameworks/av/services/medialog \ frameworks/av/services/audioflinger \ frameworks/av/services/audiopolicy \ + frameworks/av/services/audiopolicy/common/managerdefinitions/include \ frameworks/av/services/camera/libcameraservice \ frameworks/av/services/mediaresourcemanager \ $(call include-path-for, audio-utils) \ diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk index 351ed79..fb56fd8 100644 --- a/services/audiopolicy/Android.mk +++ b/services/audiopolicy/Android.mk @@ -16,6 +16,7 @@ else LOCAL_SRC_FILES += \ service/AudioPolicyInterfaceImpl.cpp \ service/AudioPolicyClientImpl.cpp + endif LOCAL_C_INCLUDES := \ @@ -39,7 +40,8 @@ LOCAL_SHARED_LIBRARIES += \ endif LOCAL_STATIC_LIBRARIES := \ - libmedia_helper + libmedia_helper \ + libaudiopolicycomponents LOCAL_MODULE:= libaudiopolicyservice @@ -54,14 +56,6 @@ include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ managerdefault/AudioPolicyManager.cpp \ - managerdefault/ConfigParsingUtils.cpp \ - managerdefault/Devices.cpp \ - managerdefault/Gains.cpp \ - managerdefault/HwModule.cpp \ - managerdefault/IOProfile.cpp \ - managerdefault/Ports.cpp \ - managerdefault/AudioInputDescriptor.cpp \ - managerdefault/AudioOutputDescriptor.cpp LOCAL_SHARED_LIBRARIES := \ libcutils \ @@ -70,7 +64,8 @@ LOCAL_SHARED_LIBRARIES := \ libsoundtrigger LOCAL_STATIC_LIBRARIES := \ - libmedia_helper + libmedia_helper \ + libaudiopolicycomponents LOCAL_MODULE:= libaudiopolicymanagerdefault @@ -86,9 +81,17 @@ LOCAL_SRC_FILES:= \ LOCAL_SHARED_LIBRARIES := \ libaudiopolicymanagerdefault +LOCAL_STATIC_LIBRARIES := \ + libaudiopolicycomponents + LOCAL_MODULE:= libaudiopolicymanager include $(BUILD_SHARED_LIBRARY) endif endif + +####################################################################### +# Recursive call sub-folder Android.mk +# +include $(call all-makefiles-under,$(LOCAL_PATH)) diff --git a/services/audiopolicy/common/Android.mk b/services/audiopolicy/common/Android.mk new file mode 100644 index 0000000..dcce8e3 --- /dev/null +++ b/services/audiopolicy/common/Android.mk @@ -0,0 +1,9 @@ + +LOCAL_PATH := $(call my-dir) +include $(CLEAR_VARS) + +####################################################################### +# Recursive call sub-folder Android.mk +# +include $(call all-makefiles-under,$(LOCAL_PATH)) + diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk new file mode 100644 index 0000000..d9d1477 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/Android.mk @@ -0,0 +1,29 @@ +LOCAL_PATH:= $(call my-dir) + +include $(CLEAR_VARS) + +LOCAL_SRC_FILES:= \ + src/DeviceDescriptor.cpp \ + src/AudioGain.cpp \ + src/HwModule.cpp \ + src/IOProfile.cpp \ + src/AudioPort.cpp \ + src/AudioPatch.cpp \ + src/AudioInputDescriptor.cpp \ + src/AudioOutputDescriptor.cpp \ + src/ConfigParsingUtils.cpp \ + +LOCAL_SHARED_LIBRARIES := \ + libcutils \ + libutils \ + liblog \ + +LOCAL_C_INCLUDES += \ + $(LOCAL_PATH)/include \ + +LOCAL_EXPORT_C_INCLUDE_DIRS := \ + $(LOCAL_PATH)/include + +LOCAL_MODULE := libaudiopolicycomponents + +include $(BUILD_STATIC_LIBRARY) diff --git a/services/audiopolicy/common/managerdefinitions/include/ApmImplDefinitions.h b/services/audiopolicy/common/managerdefinitions/include/ApmImplDefinitions.h new file mode 100644 index 0000000..62927da --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/ApmImplDefinitions.h @@ -0,0 +1,34 @@ +/* + * 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. + */ + +#pragma once + +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/common/managerdefinitions/include/AudioGain.h b/services/audiopolicy/common/managerdefinitions/include/AudioGain.h new file mode 100644 index 0000000..f638c8e --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioGain.h @@ -0,0 +1,119 @@ +/* + * 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. + */ + +#pragma once + +#include +#include +#include +#include + +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 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/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h new file mode 100644 index 0000000..599c295 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h @@ -0,0 +1,59 @@ +/* + * 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. + */ + +#pragma once + +#include "AudioPort.h" +#include +#include +#include + +namespace android { + +class IOProfile; +class AudioMix; + +// 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& profile); + void setIoHandle(audio_io_handle_t ioHandle); + + 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 mProfile; // I/O profile this output derives from + SortedVector 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 getAudioPort() const { return mProfile; } + void toAudioPort(struct audio_port *port) const; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h new file mode 100644 index 0000000..8469b81 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h @@ -0,0 +1,76 @@ +/* + * 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. + */ + +#pragma once + +#include "AudioPort.h" +#include "ApmImplDefinitions.h" +#include +#include +#include + +namespace android { + +class IOProfile; +class AudioMix; + +// 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& profile); + + status_t dump(int fd); + + audio_devices_t device() const; + void changeRefCount(audio_stream_type_t stream, int delta); + + void setIoHandle(audio_io_handle_t ioHandle); + bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } + audio_devices_t supportedDevices(); + uint32_t latency(); + bool sharesHwModuleWith(const sp outputDesc); + bool isActive(uint32_t inPastMs = 0) const; + bool isStreamActive(audio_stream_type_t stream, + 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 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 mOutput1; // used by duplicated outputs: first output + sp 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 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/common/managerdefinitions/include/AudioPatch.h b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h new file mode 100644 index 0000000..6e8fb32 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h @@ -0,0 +1,41 @@ +/* + * 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. + */ + +#pragma once + +#include +#include +#include + +namespace android { + +class AudioPatch : public RefBase +{ +public: + AudioPatch(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; + +private: + static volatile int32_t mNextUniqueId; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h new file mode 100644 index 0000000..52e845c --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h @@ -0,0 +1,126 @@ +/* + * 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. + */ + +#pragma once + +#include +#include +#include +#include +#include +#include + +#define MAX_MIXER_SAMPLING_RATE 48000 +#define MAX_MIXER_CHANNEL_COUNT 8 + +namespace android { + +class HwModule; +class AudioGain; + +class AudioPort: public virtual RefBase +{ +public: + AudioPort(const String8& name, audio_port_type_t type, + audio_port_role_t role, const sp& module); + virtual ~AudioPort() {} + + audio_port_handle_t getHandle() { return mId; } + + void attach(const sp& module); + bool isAttached() { return mId != 0; } + + static audio_port_handle_t getNextUniqueId(); + + virtual void toAudioPort(struct audio_port *port) const; + + void importAudioPort(const sp 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 mSamplingRates; // supported sampling rates + Vector mChannelMasks; // supported channel masks + Vector mFormats; // supported audio formats + Vector < sp > mGains; // gain controllers + sp 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; + +private: + static volatile int32_t mNextUniqueId; +}; + +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 getAudioPort() const = 0; + uint32_t mSamplingRate; + audio_format_t mFormat; + audio_channel_mask_t mChannelMask; + struct audio_gain_config mGain; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h new file mode 100644 index 0000000..45e96d9 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h @@ -0,0 +1,166 @@ +/* + * 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. + */ + +#pragma once + +#include +#include + +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/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h new file mode 100644 index 0000000..e63ea2b --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h @@ -0,0 +1,80 @@ +/* + * 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. + */ + +#pragma once + +#include "AudioPort.h" +#include +#include +#include +#include +#include + +namespace android { + +class DeviceDescriptor: public AudioPort, public AudioPortConfig +{ +public: + DeviceDescriptor(const String8& name, audio_devices_t type); + + virtual ~DeviceDescriptor() {} + + bool equals(const sp& other) const; + + // AudioPortConfig + virtual sp 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 > +{ +public: + DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} + + ssize_t add(const sp& item); + ssize_t remove(const sp& item); + ssize_t indexOf(const sp& item) const; + + audio_devices_t types() const { return mDeviceTypes; } + + void loadDevicesFromType(audio_devices_t types); + void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); + + sp getDevice(audio_devices_t type, String8 address) const; + DeviceVector getDevicesFromType(audio_devices_t types) const; + sp getDeviceFromId(audio_port_handle_t id) const; + sp 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/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h new file mode 100644 index 0000000..3d4e2ad --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h @@ -0,0 +1,58 @@ +/* + * 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. + */ + +#pragma once + +#include "DeviceDescriptor.h" +#include +#include +#include +#include +#include +#include + +namespace android { + +class IOProfile; + +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 > mOutputProfiles; // output profiles exposed by this module + Vector < sp > mInputProfiles; // input profiles exposed by this module + DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h new file mode 100644 index 0000000..095e759 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h @@ -0,0 +1,58 @@ +/* + * 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. + */ + +#pragma once + +#include "AudioPort.h" +#include "DeviceDescriptor.h" +#include +#include + +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& 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/common/managerdefinitions/include/audio_policy_conf.h b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h new file mode 100644 index 0000000..441bf7b --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h @@ -0,0 +1,70 @@ +/* + * 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. + */ + +#pragma once + +///////////////////////////////////////////////// +// 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" diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp new file mode 100644 index 0000000..ebd3548 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp @@ -0,0 +1,448 @@ +/* + * 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::AudioGain" +//#define LOG_NDEBUG 0 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +#include "AudioGain.h" +#include +#include + +#include + +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/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp new file mode 100644 index 0000000..fa7761a --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp @@ -0,0 +1,110 @@ +/* + * 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 "AudioInputDescriptor.h" +#include "IOProfile.h" +#include "AudioGain.h" +#include "HwModule.h" +#include + +namespace android { + +AudioInputDescriptor::AudioInputDescriptor(const sp& 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::setIoHandle(audio_io_handle_t ioHandle) +{ + mId = AudioPort::getNextUniqueId(); + mIoHandle = ioHandle; +} + +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/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp new file mode 100644 index 0000000..4c5509d --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp @@ -0,0 +1,223 @@ +/* + * 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 "AudioOutputDescriptor.h" +#include "IOProfile.h" +#include "AudioGain.h" +#include "HwModule.h" +#include + +namespace android { + +AudioOutputDescriptor::AudioOutputDescriptor( + const sp& 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; + } +} + +void AudioOutputDescriptor::setIoHandle(audio_io_handle_t ioHandle) +{ + mId = AudioPort::getNextUniqueId(); + mIoHandle = ioHandle; +} + +uint32_t AudioOutputDescriptor::latency() +{ + if (isDuplicated()) { + return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; + } else { + return mLatency; + } +} + +bool AudioOutputDescriptor::sharesHwModuleWith( + const sp 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 +{ + nsecs_t sysTime = 0; + if (inPastMs != 0) { + sysTime = systemTime(); + } + for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { + if (i == AUDIO_STREAM_PATCH) { + continue; + } + if (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/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp new file mode 100644 index 0000000..89e7f81 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp @@ -0,0 +1,84 @@ +/* + * 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::AudioPatch" +//#define LOG_NDEBUG 0 + +#include "AudioPatch.h" +#include "ConfigParsingUtils.h" +#include +#include + +namespace android { + +int32_t volatile AudioPatch::mNextUniqueId = 1; + +AudioPatch::AudioPatch(const struct audio_patch *patch, uid_t uid) : + mHandle(static_cast(android_atomic_inc(&mNextUniqueId))), + 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/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp new file mode 100644 index 0000000..d34aac1 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp @@ -0,0 +1,798 @@ +/* + * 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::AudioPort" +//#define LOG_NDEBUG 0 + +#include "AudioPort.h" +#include "HwModule.h" +#include "AudioGain.h" +#include "ConfigParsingUtils.h" +#include "audio_policy_conf.h" + +namespace android { + +int32_t volatile AudioPort::mNextUniqueId = 1; + +// --- AudioPort class implementation + +AudioPort::AudioPort(const String8& name, audio_port_type_t type, + audio_port_role_t role, const sp& 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& module) +{ + mId = getNextUniqueId(); + mModule = module; +} + +audio_port_handle_t AudioPort::getNextUniqueId() +{ + return static_cast(android_atomic_inc(&mNextUniqueId)); +} + +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 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 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 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 = 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; + } +} + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp new file mode 100644 index 0000000..300f35a --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp @@ -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. + */ + +#define LOG_TAG "APM::ConfigParsingUtils" +//#define LOG_NDEBUG 0 + +#include "ConfigParsingUtils.h" +#include + +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/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp new file mode 100644 index 0000000..7f098ab --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp @@ -0,0 +1,289 @@ +/* + * 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 "DeviceDescriptor.h" +#include "AudioGain.h" +#include "HwModule.h" +#include "ConfigParsingUtils.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& 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& 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& 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& 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 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 deviceDesc = + declaredDevices.getDeviceFromName(String8(devName)); + if (deviceDesc != 0) { + add(deviceDesc); + } + } + } + devName = strtok(NULL, "|"); + } +} + +sp DeviceVector::getDevice(audio_devices_t type, String8 address) const +{ + sp 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 DeviceVector::getDeviceFromId(audio_port_handle_t id) const +{ + sp 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 DeviceVector::getDeviceFromName(const String8& name) const +{ + sp 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/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp new file mode 100644 index 0000000..a0ea185 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp @@ -0,0 +1,282 @@ +/* + * Copyright (C) 2015 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#define LOG_TAG "APM::HwModule" +//#define LOG_NDEBUG 0 + +#include "HwModule.h" +#include "IOProfile.h" +#include "AudioGain.h" +#include "ConfigParsingUtils.h" +#include "audio_policy_conf.h" +#include + +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 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 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 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 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 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 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 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/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp new file mode 100644 index 0000000..0214a2b --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp @@ -0,0 +1,149 @@ +/* + * 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 "IOProfile.h" +#include "HwModule.h" +#include "AudioGain.h" + +namespace android { + +IOProfile::IOProfile(const String8& name, audio_port_role_t role, + const sp& 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/ApmImplDefinitions.h b/services/audiopolicy/managerdefault/ApmImplDefinitions.h deleted file mode 100644 index 62927da..0000000 --- a/services/audiopolicy/managerdefault/ApmImplDefinitions.h +++ /dev/null @@ -1,34 +0,0 @@ -/* - * 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. - */ - -#pragma once - -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 deleted file mode 100644 index ce6b1e7..0000000 --- a/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp +++ /dev/null @@ -1,104 +0,0 @@ -/* - * 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 "AudioInputDescriptor.h" -#include "IOProfile.h" -#include "Gains.h" -#include "HwModule.h" -#include - -namespace android { - -AudioInputDescriptor::AudioInputDescriptor(const sp& 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 deleted file mode 100644 index ce96228..0000000 --- a/services/audiopolicy/managerdefault/AudioInputDescriptor.h +++ /dev/null @@ -1,58 +0,0 @@ -/* - * 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. - */ - -#pragma once - -#include "Ports.h" -#include -#include -#include - -namespace android { - -class IOProfile; -class AudioMix; - -// 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& 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 mProfile; // I/O profile this output derives from - SortedVector 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 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 deleted file mode 100644 index 4dd9316..0000000 --- a/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp +++ /dev/null @@ -1,217 +0,0 @@ -/* - * 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 "AudioOutputDescriptor.h" -#include "IOProfile.h" -#include "Gains.h" -#include "HwModule.h" -#include - -namespace android { - -AudioOutputDescriptor::AudioOutputDescriptor( - const sp& 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 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 -{ - nsecs_t sysTime = 0; - if (inPastMs != 0) { - sysTime = systemTime(); - } - for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { - if (i == AUDIO_STREAM_PATCH) { - continue; - } - if (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 deleted file mode 100644 index f7a06ee..0000000 --- a/services/audiopolicy/managerdefault/AudioOutputDescriptor.h +++ /dev/null @@ -1,75 +0,0 @@ -/* - * 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. - */ - -#pragma once - -#include "Ports.h" -#include "ApmImplDefinitions.h" -#include -#include -#include - -namespace android { - -class IOProfile; -class AudioMix; - -// 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& 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 outputDesc); - bool isActive(uint32_t inPastMs = 0) const; - bool isStreamActive(audio_stream_type_t stream, - 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 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 mOutput1; // used by duplicated outputs: first output - sp 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 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 index 730d32f..a110ada 100644 --- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp +++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp @@ -393,8 +393,7 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs 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 = new AudioPatch(&patch, mUidCached); mCallRxPatch->mAfPatchHandle = afPatchHandle; mCallRxPatch->mUid = mUidCached; } @@ -436,8 +435,7 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs 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 = new AudioPatch(&patch, mUidCached); mCallTxPatch->mAfPatchHandle = afPatchHandle; mCallTxPatch->mUid = mUidCached; } @@ -2634,8 +2632,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, status, afPatchHandle); if (status == NO_ERROR) { if (index < 0) { - patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &newPatch, uid); + patchDesc = new AudioPatch(&newPatch, uid); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = newPatch; @@ -2880,19 +2877,11 @@ status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle) // ---------------------------------------------------------------------------- // 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 @@ -3314,16 +3303,14 @@ int AudioPolicyManager::testOutputIndex(audio_io_handle_t output) void AudioPolicyManager::addOutput(audio_io_handle_t output, sp outputDesc) { - outputDesc->mIoHandle = output; - outputDesc->mId = nextUniqueId(); + outputDesc->setIoHandle(output); mOutputs.add(output, outputDesc); nextAudioPortGeneration(); } void AudioPolicyManager::addInput(audio_io_handle_t input, sp inputDesc) { - inputDesc->mIoHandle = input; - inputDesc->mId = nextUniqueId(); + inputDesc->setIoHandle(input); mInputs.add(input, inputDesc); nextAudioPortGeneration(); } @@ -4827,8 +4814,7 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, 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); + patchDesc = new AudioPatch(&patch, mUidCached); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = patch; @@ -4934,8 +4920,7 @@ status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input, status, afPatchHandle); if (status == NO_ERROR) { if (index < 0) { - patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); + patchDesc = new AudioPatch(&patch, mUidCached); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = patch; diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h index 8308d54..279dc93 100644 --- a/services/audiopolicy/managerdefault/AudioPolicyManager.h +++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h @@ -27,14 +27,15 @@ #include #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" +#include +#include +#include +#include +#include +#include +#include +#include +#include namespace android { @@ -208,8 +209,6 @@ public: // 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 @@ -453,7 +452,6 @@ protected: // to boost soft sounds, used to adjust volume curves accordingly Vector < sp > mHwModules; - static volatile int32_t mNextUniqueId; volatile int32_t mAudioPortGeneration; DefaultKeyedVector > mAudioPatches; diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp deleted file mode 100644 index 300f35a..0000000 --- a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp +++ /dev/null @@ -1,122 +0,0 @@ -/* - * 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 "ConfigParsingUtils.h" -#include - -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 deleted file mode 100644 index 45e96d9..0000000 --- a/services/audiopolicy/managerdefault/ConfigParsingUtils.h +++ /dev/null @@ -1,166 +0,0 @@ -/* - * 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. - */ - -#pragma once - -#include -#include - -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 deleted file mode 100644 index 574cff5..0000000 --- a/services/audiopolicy/managerdefault/Devices.cpp +++ /dev/null @@ -1,289 +0,0 @@ -/* - * 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 "Devices.h" -#include "Gains.h" -#include "HwModule.h" -#include "ConfigParsingUtils.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& 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& 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& 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& 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 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 deviceDesc = - declaredDevices.getDeviceFromName(String8(devName)); - if (deviceDesc != 0) { - add(deviceDesc); - } - } - } - devName = strtok(NULL, "|"); - } -} - -sp DeviceVector::getDevice(audio_devices_t type, String8 address) const -{ - sp 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 DeviceVector::getDeviceFromId(audio_port_handle_t id) const -{ - sp 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 DeviceVector::getDeviceFromName(const String8& name) const -{ - sp 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 deleted file mode 100644 index 7ccf559..0000000 --- a/services/audiopolicy/managerdefault/Devices.h +++ /dev/null @@ -1,80 +0,0 @@ -/* - * 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. - */ - -#pragma once - -#include "Ports.h" -#include -#include -#include -#include -#include - -namespace android { - -class DeviceDescriptor: public AudioPort, public AudioPortConfig -{ -public: - DeviceDescriptor(const String8& name, audio_devices_t type); - - virtual ~DeviceDescriptor() {} - - bool equals(const sp& other) const; - - // AudioPortConfig - virtual sp 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 > -{ -public: - DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} - - ssize_t add(const sp& item); - ssize_t remove(const sp& item); - ssize_t indexOf(const sp& item) const; - - audio_devices_t types() const { return mDeviceTypes; } - - void loadDevicesFromType(audio_devices_t types); - void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); - - sp getDevice(audio_devices_t type, String8 address) const; - DeviceVector getDevicesFromType(audio_devices_t types) const; - sp getDeviceFromId(audio_port_handle_t id) const; - sp 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 deleted file mode 100644 index 98a8d1c..0000000 --- a/services/audiopolicy/managerdefault/Gains.cpp +++ /dev/null @@ -1,448 +0,0 @@ -/* - * 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 "Gains.h" -#include -#include - -#include - -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 deleted file mode 100644 index f638c8e..0000000 --- a/services/audiopolicy/managerdefault/Gains.h +++ /dev/null @@ -1,119 +0,0 @@ -/* - * 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. - */ - -#pragma once - -#include -#include -#include -#include - -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 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 deleted file mode 100644 index 39dc889..0000000 --- a/services/audiopolicy/managerdefault/HwModule.cpp +++ /dev/null @@ -1,282 +0,0 @@ -/* - * 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 "HwModule.h" -#include "IOProfile.h" -#include "Gains.h" -#include "ConfigParsingUtils.h" -#include "audio_policy_conf.h" -#include - -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 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 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 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 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 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 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 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 deleted file mode 100644 index d9e6cdf..0000000 --- a/services/audiopolicy/managerdefault/HwModule.h +++ /dev/null @@ -1,58 +0,0 @@ -/* - * 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. - */ - -#pragma once - -#include "Devices.h" -#include -#include -#include -#include -#include -#include - -namespace android { - -class IOProfile; - -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 > mOutputProfiles; // output profiles exposed by this module - Vector < sp > 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 deleted file mode 100644 index f82ff92..0000000 --- a/services/audiopolicy/managerdefault/IOProfile.cpp +++ /dev/null @@ -1,149 +0,0 @@ -/* - * 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 "IOProfile.h" -#include "HwModule.h" -#include "Gains.h" - -namespace android { - -IOProfile::IOProfile(const String8& name, audio_port_role_t role, - const sp& 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 deleted file mode 100644 index a9d93d1..0000000 --- a/services/audiopolicy/managerdefault/IOProfile.h +++ /dev/null @@ -1,58 +0,0 @@ -/* - * 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. - */ - -#pragma once - -#include "Ports.h" -#include "Devices.h" -#include -#include - -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& 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 deleted file mode 100644 index 326f9c5..0000000 --- a/services/audiopolicy/managerdefault/Ports.cpp +++ /dev/null @@ -1,848 +0,0 @@ -/* - * 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 "Ports.h" -#include "HwModule.h" -#include "Gains.h" -#include "ConfigParsingUtils.h" -#include "audio_policy_conf.h" - -namespace android { - -int32_t volatile AudioPort::mNextUniqueId = 1; - -// --- AudioPort class implementation - -AudioPort::AudioPort(const String8& name, audio_port_type_t type, - audio_port_role_t role, const sp& 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& module) { - mId = android_atomic_inc(&mNextUniqueId); - 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 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 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 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 = 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 deleted file mode 100644 index 6e0e2fe..0000000 --- a/services/audiopolicy/managerdefault/Ports.h +++ /dev/null @@ -1,138 +0,0 @@ -/* - * 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. - */ - -#pragma once - -#include -#include -#include -#include -#include -#include - -#define MAX_MIXER_SAMPLING_RATE 48000 -#define MAX_MIXER_CHANNEL_COUNT 8 - -namespace android { - -class HwModule; -class AudioGain; - -class AudioPort: public virtual RefBase -{ -public: - AudioPort(const String8& name, audio_port_type_t type, - audio_port_role_t role, const sp& module); - virtual ~AudioPort() {} - - audio_port_handle_t getHandle() { return mId; } - - void attach(const sp& module); - bool isAttached() { return mId != 0; } - - virtual void toAudioPort(struct audio_port *port) const; - - void importAudioPort(const sp 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 mSamplingRates; // supported sampling rates - Vector mChannelMasks; // supported channel masks - Vector mFormats; // supported audio formats - Vector < sp > mGains; // gain controllers - sp 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; - -private: - static volatile int32_t mNextUniqueId; -}; - -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 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 deleted file mode 100644 index 441bf7b..0000000 --- a/services/audiopolicy/managerdefault/audio_policy_conf.h +++ /dev/null @@ -1,70 +0,0 @@ -/* - * 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. - */ - -#pragma once - -///////////////////////////////////////////////// -// 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" -- cgit v1.1