AudioPolicyService是策略的制定者,比如什么时候打开音频接口设备、某种Stream类型的音频对应什么设备等等。而AudioFlinger则是策略的执行者,例如具体如何与音频设备通信,如何维护现有系统中的音频设备,以及多个音频流的混音如何处理等等都得由它来完成。AudioPolicyService根据用户配置来指导AudioFlinger加载设备接口,起到路由功能。
AudioPolicyService启动过程
AudioPolicyService服务运行在mediaserver进程中,随着mediaserver进程启动而启动。
frameworks\av\media\mediaserver\ Main_mediaserver.cpp
1 2 3 4 5 6 7 8 9 10 11 12 13 | int main( int argc, char ** argv) { sp proc(ProcessState::self()); sp sm = defaultServiceManager(); ALOGI( "ServiceManager: %p" , sm.get()); VolumeManager::instantiate(); // volumemanager have to be started before audioflinger AudioFlinger::instantiate(); MediaPlayerService::instantiate(); CameraService::instantiate(); AudioPolicyService::instantiate(); ProcessState::self()->startThreadPool(); IPCThreadState::self()->joinThreadPool(); } |
AudioPolicyService继承了模板类BinderService,该类用于注册native service。
frameworks\native\include\binder\ BinderService.h
1 2 3 4 5 6 7 8 9 10 | template class BinderService { public : static status_t publish(bool allowIsolated = false ) { sp sm(defaultServiceManager()); return sm->addService(String16(SERVICE::getServiceName()), new SERVICE(), allowIsolated); } static void instantiate() { publish(); } }; |
BinderService是一个模板类,该类的publish函数就是完成向ServiceManager注册服务。
1 | static const char *getServiceName() { return "media.audio_policy" ; } |
AudioPolicyService注册名为media.audio_policy的服务。
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 | AudioPolicyService::AudioPolicyService() : BnAudioPolicyService() , mpAudioPolicyDev(NULL) , mpAudioPolicy(NULL) { char value[PROPERTY_VALUE_MAX]; const struct hw_module_t *module; int forced_val; int rc; Mutex::Autolock _l(mLock); // start tone playback thread mTonePlaybackThread = new AudioCommandThread(String8( "ApmTone" ), this ); // start audio commands thread mAudioCommandThread = new AudioCommandThread(String8( "ApmAudio" ), this ); // start output activity command thread mOutputCommandThread = new AudioCommandThread(String8( "ApmOutput" ), this ); /* instantiate the audio policy manager */ /* 加载audio_policy.default.so库得到audio_policy_module模块 */ rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module); if (rc) return ; /* 通过audio_policy_module模块打开audio_policy_device设备 */ rc = audio_policy_dev_open(module, &mpAudioPolicyDev); ALOGE_IF(rc, "couldn't open audio policy device (%s)" , strerror(-rc)); if (rc) return ; //通过audio_policy_device设备创建audio_policy rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this , &mpAudioPolicy); ALOGE_IF(rc, "couldn't create audio policy (%s)" , strerror(-rc)); if (rc) return ; rc = mpAudioPolicy->init_check(mpAudioPolicy); ALOGE_IF(rc, "couldn't init_check the audio policy (%s)" , strerror(-rc)); if (rc) return ; /* SPRD: maybe set this property better, but here just change the default value @{ */ property_get( "ro.camera.sound.forced" , value, "1" ); forced_val = strtol(value, NULL, 0 ); ALOGV( "setForceUse() !forced_val=%d " ,!forced_val); mpAudioPolicy->set_can_mute_enforced_audible(mpAudioPolicy, !forced_val); ALOGI( "Loaded audio policy from %s (%s)" , module->name, module->id); // 读取audio_effects.conf文件 if (access(AUDIO_EFFECT_VENDOR_CONFIG_FILE, R_OK) == 0 ) { loadPreProcessorConfig(AUDIO_EFFECT_VENDOR_CONFIG_FILE); } else if (access(AUDIO_EFFECT_DEFAULT_CONFIG_FILE, R_OK) == 0 ) { loadPreProcessorConfig(AUDIO_EFFECT_DEFAULT_CONFIG_FILE); } } |
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- 创建AudioCommandThread (ApmTone、ApmAudio、ApmOutput)加载legacy_ap_module打开legacy_ap_device创建legacy_audio_policy读取audio_effects.conf
创建AudioCommandThread线程
在AudioPolicyService对象构造过程中,分别创建了ApmTone、ApmAudio、ApmOutput三个AudioCommandThread线程:
1、 ApmTone用于播放tone音;
2、 ApmAudio用于执行audio命令;
3、ApmOutput用于执行输出命令;
在第一次强引用AudioCommandThread线程对象时,AudioCommandThread的onFirstRef函数被回调,在此启动线程
1 2 3 4 | void AudioPolicyService::AudioCommandThread::onFirstRef() { run(mName.string(), ANDROID_PRIORITY_AUDIO); } |
这里采用异步方式来执行audio command,当需要执行上表中的命令时,首先将命令投递到AudioCommandThread的mAudioCommands命令向量表中,然后通过mWaitWorkCV.signal()唤醒AudioCommandThread线程,被唤醒的AudioCommandThread线程执行完command后,又通过mWaitWorkCV.waitRelative(mLock, waitTime)睡眠等待命令到来。
加载audio_policy_module模块
audio_policy硬件抽象层动态库位于/system/lib/hw/目录下,命名为:audio_policy.$(TARGET_BOARD_PLATFORM).so。audiopolicy的硬件抽象层定义在hardware\libhardware_legacy\audio\audio_policy_hal.cpp中,AUDIO_POLICY_HARDWARE_MODULE_ID硬件抽象模块定义如下:
hardware\libhardware_legacy\audio\ audio_policy_hal.cpp【audio_policy.scx15.so】
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 | struct legacy_ap_module HAL_MODULE_INFO_SYM = { module: { common: { tag: HARDWARE_MODULE_TAG, version_major: 1 , version_minor: 0 , id: AUDIO_POLICY_HARDWARE_MODULE_ID, name: "LEGACY Audio Policy HAL" , author: "The Android Open Source Project" , methods: &legacy_ap_module_methods, dso : NULL, reserved : { 0 }, }, }, }; |
legacy_ap_module继承于audio_policy_module。
关于hw_get_module函数加载硬件抽象层模块的过程请参考Android硬件抽象Hardware库加载过程源码分析。
打开audio_policy_device设备
hardware\libhardware\include\hardware\ audio_policy.h
1 2 3 4 5 6 | static inline int audio_policy_dev_open( const hw_module_t* module, struct audio_policy_device** device) { return module->methods->open(module, AUDIO_POLICY_INTERFACE, (hw_device_t**)device); } |
通过legacy_ap_module模块的open方法来打开一个legacy_ap_device设备。
hardware\libhardware_legacy\audio\ audio_policy_hal.cpp
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 | static int legacy_ap_dev_open( const hw_module_t* module, const char * name, hw_device_t** device) { struct legacy_ap_device *dev; if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0 ) return -EINVAL; dev = (struct legacy_ap_device *)calloc( 1 , sizeof(*dev)); if (!dev) return -ENOMEM; dev->device.common.tag = HARDWARE_DEVICE_TAG; dev->device.common.version = 0 ; dev->device.common.module = const_cast<hw_module_t*>(module); dev->device.common.close = legacy_ap_dev_close; dev->device.create_audio_policy = create_legacy_ap; dev->device.destroy_audio_policy = destroy_legacy_ap; *device = &dev->device.common; return 0 ; } |
打开得到一个legacy_ap_device设备,通过该抽象设备可以创建一个audio_policy对象。
创建audio_policy对象
在打开legacy_ap_device设备时,该设备的create_audio_policy成员初始化为create_legacy_ap函数指针,我们通过legacy_ap_device设备可以创建一个legacy_audio_policy对象。
1 2 | rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this , &mpAudioPolicy); |
这里通过audio_policy_device设备创建audio策略对象
hardware\libhardware_legacy\audio\ audio_policy_hal.cpp
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 | static int create_legacy_ap( const struct audio_policy_device *device, struct audio_policy_service_ops *aps_ops, void *service, struct audio_policy **ap) { struct legacy_audio_policy *lap; int ret; if (!service || !aps_ops) return -EINVAL; lap = (struct legacy_audio_policy *)calloc( 1 , sizeof(*lap)); if (!lap) return -ENOMEM; lap->policy.set_device_connection_state = ap_set_device_connection_state; … lap->policy.dump = ap_dump; lap->policy.is_offload_supported = ap_is_offload_supported; lap->service = service; lap->aps_ops = aps_ops; lap->service_client = new AudioPolicyCompatClient(aps_ops, service); if (!lap->service_client) { ret = -ENOMEM; goto err_new_compat_client; } lap->apm = createAudioPolicyManager(lap->service_client); if (!lap->apm) { ret = -ENOMEM; goto err_create_apm; } *ap = &lap->policy; return 0 ; err_create_apm: delete lap->service_client; err_new_compat_client: free(lap); *ap = NULL; return ret; } |
audio_policy实现在audio_policy_hal.cpp中,audio_policy_service_ops实现在AudioPolicyService.cpp中。create_audio_policy()函数就是创建并初始化一个legacy_audio_policy对象。
audio_policy与AudioPolicyService、AudioPolicyCompatClient之间的关系如下:
AudioPolicyClient创建
hardware\libhardware_legacy\audio\ AudioPolicyCompatClient.h
1 2 | AudioPolicyCompatClient(struct audio_policy_service_ops *serviceOps, void *service) : mServiceOps(serviceOps) , mService(service) {} |
AudioPolicyCompatClient是对audio_policy_service_ops的封装类,对外提供audio_policy_service_ops数据结构中定义的接口。
AudioPolicyManager创建
1 2 3 4 5 | extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface) { ALOGI( "SPRD policy manager created." ); return new AudioPolicyManagerSPRD(clientInterface); } |
使用AudioPolicyClientInterface对象来构造AudioPolicyManagerSPRD对象,AudioPolicyManagerSPRD继承于AudioPolicyManagerBase,而AudioPolicyManagerBase又继承于AudioPolicyInterface。
hardware\libhardware_legacy\audio\ AudioPolicyManagerBase.cpp
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 | AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface) : #ifdef AUDIO_POLICY_TEST Thread( false ), #endif //AUDIO_POLICY_TEST //变量初始化 mPrimaryOutput((audio_io_handle_t) 0 ), mAvailableOutputDevices(AUDIO_DEVICE_NONE), mPhoneState(AudioSystem::MODE_NORMAL), mLimitRingtoneVolume( false ), mLastVoiceVolume(- 1 .0f), mTotalEffectsCpuLoad( 0 ), mTotalEffectsMemory( 0 ), mA2dpSuspended( false ), mHasA2dp( false ), mHasUsb( false ), mHasRemoteSubmix( false ), mSpeakerDrcEnabled( false ), mFmOffGoing( false ) { //引用AudioPolicyCompatClient对象,这样音频管理器AudioPolicyManager就可以使用audio_policy_service_ops中的接口 mpClientInterface = clientInterface; for ( int i = 0 ; i < AudioSystem::NUM_FORCE_USE; i++) { mForceUse[i] = AudioSystem::FORCE_NONE; } mA2dpDeviceAddress = String8( "" ); mScoDeviceAddress = String8( "" ); mUsbCardAndDevice = String8( "" ); /** * 优先加载/vendor/etc/audio_policy.conf配置文件,如果该配置文件不存在,则 * 加载/system/etc/audio_policy.conf配置文件,如果该文件还是不存在,则通过 * 函数defaultAudioPolicyConfig()来设置默认音频接口 */ if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) { if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) { ALOGE( "could not load audio policy configuration file, setting defaults" ); defaultAudioPolicyConfig(); } } //设置各种音频流对应的音量调节点,must be done after reading the policy initializeVolumeCurves(); // open all output streams needed to access attached devices for (size_t i = 0 ; i < mHwModules.size(); i++) { //通过名称打开对应的音频接口硬件抽象库 mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName); if (mHwModules[i]->mHandle == 0 ) { ALOGW( "could not open HW module %s" , mHwModules[i]->mName); continue ; } // open all output streams needed to access attached devices // except for direct output streams that are only opened when they are actually // required by an app. for (size_t j = 0 ; j < mHwModules[i]->mOutputProfiles.size(); j++) { const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j]; //打开mAttachedOutputDevices对应的输出 if ((outProfile->mSupportedDevices & mAttachedOutputDevices) && ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0 )) { //将输出IOProfile封装为AudioOutputDescriptor对象 AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile); //设置当前音频接口的默认输出设备 outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice & outProfile->mSupportedDevices); //打开输出,在AudioFlinger中创建PlaybackThread线程,并返回该线程的id audio_io_handle_t output = mpClientInterface->openOutput( outProfile->mModule->mHandle, &outputDesc->mDevice, &outputDesc->mSamplingRate, &outputDesc->mFormat, &outputDesc->mChannelMask, &outputDesc->mLatency, outputDesc->mFlags); if (output == 0 ) { delete outputDesc; } else { //设置可以使用的输出设备为mAttachedOutputDevices mAvailableOutputDevices =(audio_devices_t)(mAvailableOutputDevices | (outProfile->mSupportedDevices & mAttachedOutputDevices)); if (mPrimaryOutput == 0 && outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { mPrimaryOutput = output; } //将输出描述符对象AudioOutputDescriptor及创建的PlaybackThread线程id以键值对形式保存 addOutput(output, outputDesc); //设置默认输出设备 setOutputDevice(output,(audio_devices_t)(mDefaultOutputDevice & outProfile->mSupportedDevices), true ); } } } } ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices), "Not output found for attached devices %08x" , (mAttachedOutputDevices & ~mAvailableOutputDevices)); ALOGE_IF((mPrimaryOutput == 0 ), "Failed to open primary output" ); updateDevicesAndOutputs(); // add for bug158794 start char bootvalue[PROPERTY_VALUE_MAX]; // prop sys.boot_completed will set 1 when system ready (ActivityManagerService.java)... property_get( "sys.boot_completed" , bootvalue, "" ); if (strncmp( "1" , bootvalue, 1 ) != 0 ) { startReadingThread(); } // add for bug158794 end #ifdef AUDIO_POLICY_TEST ... #endif //AUDIO_POLICY_TEST } |
AudioPolicyManagerBase对象构造过程中主要完成以下几个步骤:
1、 loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE)加载audio_policy.conf配置文件;
2、 initializeVolumeCurves()初始化各种音频流对应的音量调节点;
3、 加载audio policy硬件抽象库:mpClientInterface->loadHwModule(mHwModules[i]->mName)
4、 打开attached_output_devices输出:
mpClientInterface->openOutput();
5、 保存输出设备描述符对象:addOutput(output, outputDesc);
读取audio_policy.conf文件
Android为每种音频接口定义了对应的硬件抽象层,且编译为单独的so库。
每种音频接口定义了不同的输入输出,一个接口可以具有多个输入或者输出,每个输入输出有可以支持不同的音频设备。通过读取audio_policy.conf文件可以获取系统支持的音频接口参数。
audio_policy.conf文件定义了两种音频配置信息:
1、 当前系统支持的音频输入输出设备及默认输入输出设备;
这些信息时通过global_configuration配置项来设置,在global_configuration中定义了三种音频设备信息:
attached_output_devices:已连接的输出设备;
default_output_device:默认输出设备;
attached_input_devices:已连接的输入设备;
1、 系统支持的音频接口信息;
audio_policy.conf定义了系统支持的所有音频接口参数信息,比如primary、a2dp、usb等,对于primary定义如下:
a2dp定义:
usb定义:
每种音频接口包含输入输出,每种输入输出又包含多种输入输出配置,每种输入输出配置又支持多种音频设备。AudioPolicyManagerBase首先加载/vendor/etc/audio_policy.conf,如果该文件不存在,则加/system/etc/audio_policy.conf。
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 | status_t AudioPolicyManagerBase::loadAudioPolicyConfig( const char *path) { cnode *root; char *data; data = ( char *)load_file(path, NULL); if (data == NULL) { return -ENODEV; } root = config_node( "" , "" ); //读取配置文件 config_load(root, data); //解析global_configuration loadGlobalConfig(root); //解析audio_hw_modules loadHwModules(root); config_free(root); free(root); free(data); ALOGI( "loadAudioPolicyConfig() loaded %s\n" , path); return NO_ERROR; } |
通过loadGlobalConfig(root)函数来读取这些全局配置信息。
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 | void AudioPolicyManagerBase::loadGlobalConfig(cnode *root) { cnode *node = config_find(root, GLOBAL_CONFIG_TAG); if (node == NULL) { return ; } node = node->first_child; while (node) { //attached_output_devices AUDIO_DEVICE_OUT_EARPIECE if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0 ) { mAttachedOutputDevices = parseDeviceNames(( char *)node->value); ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE, "loadGlobalConfig() no attached output devices" ); ALOGV( "loadGlobalConfig()mAttachedOutputDevices%04x" , mAttachedOutputDevices); //default_output_device AUDIO_DEVICE_OUT_SPEAKER } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0 ) { mDefaultOutputDevice= (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,ARRAY_SIZE(sDeviceNameToEnumTable),( char *)node->value); ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE, "loadGlobalConfig() default device not specified" ); ALOGV( "loadGlobalConfig() mDefaultOutputDevice %04x" , mDefaultOutputDevice); //attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0 ) { mAvailableInputDevices = parseDeviceNames(( char *)node->value) & ~AUDIO_DEVICE_BIT_IN; ALOGV( "loadGlobalConfig() mAvailableInputDevices %04x" , mAvailableInputDevices); //speaker_drc_enabled } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0 ) { mSpeakerDrcEnabled = stringToBool(( char *)node->value); ALOGV( "loadGlobalConfig() mSpeakerDrcEnabled = %d" , mSpeakerDrcEnabled); } node = node->next; } } |
audio_policy.conf同时定义了多个audio 接口,每一个audio 接口包含若干output和input,而每个output和input又同时支持多种输入输出模式,每种输入输出模式又支持若干种设备。
通过loadHwModules ()函数来加载系统配置的所有audio 接口:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 | void AudioPolicyManagerBase::loadHwModules(cnode *root) { //audio_hw_modules cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); if (node == NULL) { return ; } node = node->first_child; while (node) { ALOGV( "loadHwModules() loading module %s" , node->name); //加载音频接口 loadHwModule(node); node = node->next; } } |
由于audio_policy.conf可以定义多个音频接口,因此该函数循环调用loadHwModule()来解析每个音频接口参数信息。Android定义HwModule类来描述每一个audio 接口参数,定义IOProfile类来描述输入输出模式配置。
到此就将audio_policy.conf文件中音频接口配置信息解析到了AudioPolicyManagerBase的成员变量mHwModules、mAttachedOutputDevices、mDefaultOutputDevice、mAvailableInputDevices中。
初始化音量调节点
音量调节点设置在Android4.1与Android4.4中的实现完全不同,在Android4.1中是通过VolumeManager服务来管理,通过devicevolume.xml文件来配置,但Android4.4取消了VolumeManager服务,将音量控制放到AudioPolicyManagerBase中。在AudioPolicyManagerBase中定义了音量调节对应的音频流描述符数组:
1 | StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES]; |
initializeVolumeCurves()函数就是初始化该数组元素:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 | void AudioPolicyManagerBase::initializeVolumeCurves() { for ( int i = 0 ; i < AUDIO_STREAM_CNT; i++) { for ( int j = 0 ; j < DEVICE_CATEGORY_CNT; j++) { mStreams[i].mVolumeCurve[j] = sVolumeProfiles[i][j]; } } // Check availability of DRC on speaker path: if available, override some of the speaker curves if (mSpeakerDrcEnabled) { mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sDefaultSystemVolumeCurveDrc; mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sSpeakerSonificationVolumeCurveDrc; mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] = sSpeakerSonificationVolumeCurveDrc; mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =sSpeakerSonificationVolumeCurveDrc; } } |
sVolumeProfiles数组定义了不同音频设备下不同音频流对应的音量调节档位,定义如下:
数组元素为音量调节档位,每种模式下的音量调节都包含4个档位,定义如下:
加载audio_module模块
AudioPolicyManager通过读取audio_policy.conf配置文件,可以知道系统当前支持那些音频接口以及attached的输入输出设备、默认输出设备。接下来就需要加载这些音频接口的硬件抽象库。
这三中音频接口硬件抽象定义如下:
/vendor/sprd/open-source/libs/audio/audio_hw.c 【audio.primary.scx15.so】
1 2 3 4 5 6 7 8 9 10 11 | struct audio_module HAL_MODULE_INFO_SYM = { .common = { .tag = HARDWARE_MODULE_TAG, .module_api_version = AUDIO_MODULE_API_VERSION_0_1, .hal_api_version = HARDWARE_HAL_API_VERSION, .id = AUDIO_HARDWARE_MODULE_ID, .name = "Spreadtrum Audio HW HAL" , .author = "The Android Open Source Project" , .methods = &hal_module_methods, }, }; |
external/bluetooth/bluedroid/audio_a2dp_hw/audio_a2dp_hw.c【audio.a2dp.default.so】
1 2 3 4 5 6 7 8 9 10 11 | struct audio_module HAL_MODULE_INFO_SYM = { .common = { .tag = HARDWARE_MODULE_TAG, .version_major = 1 , .version_minor = 0 , .id = AUDIO_HARDWARE_MODULE_ID, .name = "A2DP Audio HW HAL" , .author = "The Android Open Source Project" , .methods = &hal_module_methods, }, }; |
hardware/libhardware/modules/usbaudio/audio_hw.c【audio. usb.default.so】
1 2 3 4 5 6 7 8 9 10 11 | struct audio_module HAL_MODULE_INFO_SYM = { .common = { .tag = HARDWARE_MODULE_TAG, .module_api_version = AUDIO_MODULE_API_VERSION_0_1, .hal_api_version = HARDWARE_HAL_API_VERSION, .id = AUDIO_HARDWARE_MODULE_ID, .name = "USB audio HW HAL" , .author = "The Android Open Source Project" , .methods = &hal_module_methods, }, }; |
AudioPolicyClientInterface提供了加载音频接口硬件抽象库的接口函数,通过前面的介绍,我们知道,AudioPolicyCompatClient通过代理audio_policy_service_ops实现AudioPolicyClientInterface接口。
hardware\libhardware_legacy\audio\ AudioPolicyCompatClient.cpp
1 2 3 4 | audio_module_handle_t AudioPolicyCompatClient::loadHwModule( const char *moduleName) { return mServiceOps->load_hw_module(mService, moduleName); } |
AudioPolicyCompatClient将音频模块加载工作交给audio_policy_service_ops
frameworks\av\services\audioflinger\ AudioPolicyService.cpp
1 2 3 4 5 6 7 8 9 | static audio_module_handle_t aps_load_hw_module( void *service, const char *name) { sp af = AudioSystem::get_audio_flinger(); if (af == 0 ) { ALOGW( "%s: could not get AudioFlinger" , __func__); return 0 ; } return af->loadHwModule(name); } |
AudioPolicyService又将其转交给AudioFlinger
frameworks\av\services\audioflinger\ AudioFlinger.cpp
1 2 3 4 5 6 7 8 | audio_module_handle_t AudioFlinger::loadHwModule( const char *name) { if (!settingsAllowed()) { return 0 ; } Mutex::Autolock _l(mLock); return loadHwModule_l(name); } |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 | audio_module_handle_t AudioFlinger::loadHwModule_l( const char *name) { for (size_t i = 0 ; i < mAudioHwDevs.size(); i++) { if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0 ) { ALOGW( "loadHwModule() module %s already loaded" , name); return mAudioHwDevs.keyAt(i); } } audio_hw_device_t *dev; //加载音频接口对应的so库,得到对应的音频接口设备audio_hw_device_t int rc = load_audio_interface(name, &dev); if (rc) { ALOGI( "loadHwModule() error %d loading module %s " , rc, name); return 0 ; } mHardwareStatus = AUDIO_HW_INIT; rc = dev->init_check(dev); mHardwareStatus = AUDIO_HW_IDLE; if (rc) { ALOGI( "loadHwModule() init check error %d for module %s " , rc, name); return 0 ; } if ((mMasterVolumeSupportLvl != MVS_NONE) && (NULL != dev->set_master_volume)) { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; dev->set_master_volume(dev, mMasterVolume); mHardwareStatus = AUDIO_HW_IDLE; } audio_module_handle_t handle = nextUniqueId(); mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); ALOGI( "loadHwModule() Loaded %s audio interface from %s (%s) handle %d" , name, dev->common.module->name, dev->common.module->id, handle); return handle; } |
函数首先加载系统定义的音频接口对应的so库,并打开该音频接口的抽象硬件设备audio_hw_device_t,为每个音频接口设备生成独一无二的ID号,同时将打开的音频接口设备封装为AudioHwDevice对象,将系统中所有的音频接口设备保存到AudioFlinger的成员变量mAudioHwDevs中。
函数load_audio_interface根据音频接口名称来打开抽象的音频接口设备audio_hw_device_t。
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 | static int load_audio_interface( const char *if_name, audio_hw_device_t **dev) { const hw_module_t *mod; int rc; //根据名字加载audio_module模块 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)" , __func__, AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); if (rc) { goto out; } //打开audio_device设备 rc = audio_hw_device_open(mod, dev); ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)" , __func__, AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); if (rc) { goto out; } if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { ALOGE( "%s wrong audio hw device version %04x" , __func__, (*dev)->common.version); rc = BAD_VALUE; goto out; } return 0 ; out: *dev = NULL; return rc; } |
hardware\libhardware\include\hardware\ Audio.h
1 2 3 4 5 6 | static inline int audio_hw_device_open( const struct hw_module_t* module, struct audio_hw_device** device) { return module->methods->open(module, AUDIO_HARDWARE_INTERFACE, (struct hw_device_t**)device); } |
hardware\libhardware_legacy\audio\ audio_hw_hal.cpp
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 | static int legacy_adev_open( const hw_module_t* module, const char * name, hw_device_t** device) { struct legacy_audio_device *ladev; int ret; if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0 ) return -EINVAL; ladev = (struct legacy_audio_device *)calloc( 1 , sizeof(*ladev)); if (!ladev) return -ENOMEM; ladev->device.common.tag = HARDWARE_DEVICE_TAG; ladev->device.common.version = AUDIO_DEVICE_API_VERSION_1_0; ladev->device.common.module = const_cast<hw_module_t*>(module); ladev->device.common.close = legacy_adev_close; ladev->device.get_supported_devices = adev_get_supported_devices; … ladev->device.dump = adev_dump; ladev->hwif = createAudioHardware(); if (!ladev->hwif) { ret = -EIO; goto err_create_audio_hw; } *device = &ladev->device.common; return 0 ; err_create_audio_hw: free(ladev); return ret; } |
打开音频接口设备过程其实就是构造并初始化legacy_audio_device对象过程,legacy_audio_device数据结构关系如下:
legacy_adev_open函数就是创建并初始化一个legacy_audio_device对象:
到此就加载完系统定义的所有音频接口,并生成相应的数据对象,如下图所示:
打开音频输出
AudioPolicyService加载完所有音频接口后,就知道了系统支持的所有音频接口参数,可以为音频输出提供决策。
为了能正常播放音频数据,需要创建抽象的音频输出接口对象,打开音频输出过程如下:
1 2 3 4 5 6 7 8 9 10 11 12 13 | audio_io_handle_t AudioPolicyCompatClient::openOutput(audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask, uint32_t *pLatencyMs, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { return mServiceOps->open_output_on_module(mService,module, pDevices, pSamplingRate, pFormat, pChannelMask, pLatencyMs, flags, offloadInfo); } |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 | static audio_io_handle_t aps_open_output_on_module( void *service, audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask, uint32_t *pLatencyMs, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { sp af = AudioSystem::get_audio_flinger(); if (af == 0 ) { ALOGW( "%s: could not get AudioFlinger" , __func__); return 0 ; } return af->openOutput(module, pDevices, pSamplingRate, pFormat, pChannelMask, pLatencyMs, flags, offloadInfo); } |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 | audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask, uint32_t *pLatencyMs, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { PlaybackThread *thread = NULL; struct audio_config config; config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0 ; config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0 ; config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; if (offloadInfo) { config.offload_info = *offloadInfo; } //创建一个音频输出流对象audio_stream_out_t audio_stream_out_t *outStream = NULL; AudioHwDevice *outHwDev; ALOGV( "openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x" , module, (pDevices != NULL) ? *pDevices : 0 , config.sample_rate, config.format, config.channel_mask, flags); ALOGV( "openOutput(), offloadInfo %p version 0x%04x" , offloadInfo, offloadInfo == NULL ? - 1 : offloadInfo->version ); if (pDevices == NULL || *pDevices == 0 ) { return 0 ; } Mutex::Autolock _l(mLock); //从音频接口列表mAudioHwDevs中查找出对应的音频接口,如果找不到,则重新加载音频接口动态库 outHwDev = findSuitableHwDev_l(module, *pDevices); if (outHwDev == NULL) return 0 ; //取出module对应的audio_hw_device_t设备 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); //为音频输出流生成一个独一无二的id号 audio_io_handle_t id = nextUniqueId(); mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; //打开音频输出流 status_t status = hwDevHal->open_output_stream(hwDevHal, id, *pDevices, (audio_output_flags_t)flags, &config, &outStream); mHardwareStatus = AUDIO_HW_IDLE; ALOGV( "openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " "Channels %x, status %d" , outStream, config.sample_rate, config.format, config.channel_mask, status); if (status == NO_ERROR && outStream != NULL) { //使用AudioStreamOut来封装音频输出流audio_stream_out_t AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); //根据flag标志位,创建不同类型的线程 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { thread = new OffloadThread( this , output, id, *pDevices); ALOGV( "openOutput() created offload output: ID %d thread %p" , id, thread); } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || (config.format != AUDIO_FORMAT_PCM_16_BIT) || (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { thread = new DirectOutputThread( this , output, id, *pDevices); ALOGV( "openOutput() created direct output: ID %d thread %p" , id, thread); } else { thread = new MixerThread( this , output, id, *pDevices); ALOGV( "openOutput() created mixer output: ID %d thread %p" , id, thread); } //将创建的线程及id以键值对的形式保存在mPlaybackThreads中 mPlaybackThreads.add(id, thread); if (pSamplingRate != NULL) { *pSamplingRate = config.sample_rate; } if (pFormat != NULL) { *pFormat = config.format; } if (pChannelMask != NULL) { *pChannelMask = config.channel_mask; } if (pLatencyMs != NULL) { *pLatencyMs = thread->latency(); } // notify client processes of the new output creation thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); // the first primary output opened designates the primary hw device if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { ALOGI( "Using module %d has the primary audio interface" , module); mPrimaryHardwareDev = outHwDev; AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MODE; hwDevHal->set_mode(hwDevHal, mMode); mHardwareStatus = AUDIO_HW_IDLE; } return id; } return 0 ; } |
打开音频输出流过程其实就是创建AudioStreamOut对象及PlaybackThread线程过程。首先通过抽象的音频接口设备audio_hw_device_t来创建输出流对象legacy_stream_out。
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 | static int adev_open_output_stream(struct audio_hw_device *dev, audio_io_handle_t handle, audio_devices_t devices, audio_output_flags_t flags, struct audio_config *config, struct audio_stream_out **stream_out) { struct legacy_audio_device *ladev = to_ladev(dev); status_t status; struct legacy_stream_out *out; int ret; //分配一个legacy_stream_out对象 out = (struct legacy_stream_out *)calloc( 1 , sizeof(*out)); if (!out) return -ENOMEM; devices = convert_audio_device(devices, HAL_API_REV_2_0, HAL_API_REV_1_0); //创建AudioStreamOut对象 out->legacy_out = ladev->hwif->openOutputStream(devices, ( int *) &config->format, &config->channel_mask, &config->sample_rate, &status); if (!out->legacy_out) { ret = status; goto err_open; } //初始化成员变量audio_stream out->stream.common.get_sample_rate = out_get_sample_rate; … *stream_out = &out->stream; return 0 ; err_open: free(out); *stream_out = NULL; return ret; } |
由于legacy_audio_device的成员变量hwif的类型为AudioHardwareInterface,因此通过调用AudioHardwareInterface的接口openOutputStream()来创建AudioStreamOut对象。
1 2 3 4 5 6 7 8 9 10 11 12 13 | AudioStreamOut* AudioHardwareStub::openOutputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) { AudioStreamOutStub* out = new AudioStreamOutStub(); status_t lStatus = out->set(format, channels, sampleRate); if (status) { *status = lStatus; } if (lStatus == NO_ERROR) return out; delete out; return 0 ; } |
打开音频输出后,在AudioFlinger与AudioPolicyService中的表现形式如下:
打开音频输入
1 2 3 4 5 6 7 8 | audio_io_handle_t AudioPolicyCompatClient::openInput(audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask) { return mServiceOps->open_input_on_module(mService, module, pDevices,pSamplingRate, pFormat, pChannelMask); } |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 | static audio_io_handle_t aps_open_input_on_module( void *service, audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask) { sp af = AudioSystem::get_audio_flinger(); if (af == 0 ) { ALOGW( "%s: could not get AudioFlinger" , __func__); return 0 ; } return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask); } |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 | audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask) { status_t status; RecordThread *thread = NULL; struct audio_config config; config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0 ; config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0 ; config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; uint32_t reqSamplingRate = config.sample_rate; audio_format_t reqFormat = config.format; audio_channel_mask_t reqChannels = config.channel_mask; audio_stream_in_t *inStream = NULL; AudioHwDevice *inHwDev; if (pDevices == NULL || *pDevices == 0 ) { return 0 ; } Mutex::Autolock _l(mLock); inHwDev = findSuitableHwDev_l(module, *pDevices); if (inHwDev == NULL) return 0 ; audio_hw_device_t *inHwHal = inHwDev->hwDevice(); audio_io_handle_t id = nextUniqueId(); status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,&inStream); ALOGV( "openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " "status %d" , inStream, config.sample_rate, config.format, config.channel_mask, status); // If the input could not be opened with the requested parameters and we can handle the // conversion internally, try to open again with the proposed parameters. The AudioFlinger can // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. if (status == BAD_VALUE &&reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && (config.sample_rate <= 2 * reqSamplingRate) && (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { ALOGV( "openInput() reopening with proposed sampling rate and channel mask" ); inStream = NULL; status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); } if (status == NO_ERROR && inStream != NULL) { #ifdef TEE_SINK // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, // or (re-)create if current Pipe is idle and does not match the new format ... #endif AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); // Start record thread // RecordThread requires both input and output device indication to forward to audio // pre processing modules thread = new RecordThread( this , input, reqSamplingRate, reqChannels, id, primaryOutputDevice_l(), *pDevices #ifdef TEE_SINK , teeSink #endif ); mRecordThreads.add(id, thread); ALOGV( "openInput() created record thread: ID %d thread %p" , id, thread); if (pSamplingRate != NULL) { *pSamplingRate = reqSamplingRate; } if (pFormat != NULL) { *pFormat = config.format; } if (pChannelMask != NULL) { *pChannelMask = reqChannels; } // notify client processes of the new input creation thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); return id; } return 0 ; } |
打开音频输入流过程其实就是创建AudioStreamIn对象及RecordThread线程过程。首先通过抽象的音频接口设备audio_hw_device_t来创建输出流对象legacy_stream_in。
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 | static int adev_open_input_stream(struct audio_hw_device *dev, audio_io_handle_t handle, audio_devices_t devices, struct audio_config *config, struct audio_stream_in **stream_in) { struct legacy_audio_device *ladev = to_ladev(dev); status_t status; struct legacy_stream_in *in; int ret; in = (struct legacy_stream_in *)calloc( 1 , sizeof(*in)); if (!in) return -ENOMEM; devices = convert_audio_device(devices, HAL_API_REV_2_0, HAL_API_REV_1_0); in->legacy_in = ladev->hwif->openInputStream(devices, ( int *) &config->format, &config->channel_mask, &config->sample_rate, &status, (AudioSystem::audio_in_acoustics) 0 ); if (!in->legacy_in) { ret = status; goto err_open; } in->stream.common.get_sample_rate = in_get_sample_rate; … *stream_in = &in->stream; return 0 ; err_open: free(in); *stream_in = NULL; return ret; } |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 | AudioStreamIn* AudioHardwareStub::openInputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) { // check for valid input source if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) { return 0 ; } AudioStreamInStub* in = new AudioStreamInStub(); status_t lStatus = in->set(format, channels, sampleRate, acoustics); if (status) { *status = lStatus; } if (lStatus == NO_ERROR) return in; delete in; return 0 ; } |
打开音频输入创建了以下legacy_stream_in对象:
打开音频输入后,在AudioFlinger与AudioPolicyService中的表现形式如下:
当AudioPolicyManagerBase构造时,它会根据用户提供的audio_policy.conf来分析系统中有哪些audio接口(primary,a2dp以及usb),然后通过AudioFlinger::loadHwModule加载各audio接口对应的库文件,并依次打开其中的output(openOutput)和input(openInput):
->打开音频输出时创建一个audio_stream_out通道,并创建AudioStreamOut对象以及新建PlaybackThread播放线程。
-> 打开音频输入时创建一个audio_stream_in通道,并创建AudioStreamIn对象以及创建RecordThread录音线程。