轉載地址:https://blog.csdn.net/yangwen123/article/details/39497375
AudioPolicyService是策略的制定者,比如什麼時候打開音頻接口設備、某種Stream類型的音頻對應什麼設備等等。而AudioFlinger則是策略的執行者,例如具體如何與音頻設備通信,如何維護現有系統中的音頻設備,以及多個音頻流的混音如何處理等等都得由它來完成。AudioPolicyService根據用戶配置來指導AudioFlinger加載設備接口,起到路由功能。
AudioPolicyService啓動過程
AudioPolicyService服務運行在mediaserver進程中,隨着mediaserver進程啓動而啓動。
frameworks\av\media\mediaserver\ Main_mediaserver.cpp
- int main(int argc, char** argv)
- {
- sp<ProcessState> proc(ProcessState::self());
- sp<IServiceManager> 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繼承了模板類BinderService,該類用於註冊native service。
frameworks\native\include\binder\ BinderService.h
- template<typename SERVICE>
- class BinderService
- {
- public:
- static status_t publish(bool allowIsolated = false) {
- sp<IServiceManager> sm(defaultServiceManager());
- return sm->addService(String16(SERVICE::getServiceName()), new SERVICE(), allowIsolated);
- }
- static void instantiate() { publish(); }
- };
BinderService是一個模板類,該類的publish函數就是完成向ServiceManager註冊服務。
- static const char *getServiceName() { return "media.audio_policy"; }
AudioPolicyService註冊名爲media.audio_policy的服務。
- 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 */
- /* 加載audio_policy.default.so庫得到audio_policy_module模塊 */
- rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);
- if (rc)
- return;
- /* 通過audio_policy_module模塊打開audio_policy_device設備 */
- rc = audio_policy_dev_open(module, &mpAudioPolicyDev);
- ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc));
- if (rc)
- return;
- //通過audio_policy_device設備創建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);
- // 讀取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);
- }
- }
- 創建AudioCommandThread (ApmTone、ApmAudio、ApmOutput)
- 加載legacy_ap_module
- 打開legacy_ap_device
- 創建legacy_audio_policy
- 讀取audio_effects.conf
創建AudioCommandThread線程
在AudioPolicyService對象構造過程中,分別創建了ApmTone、ApmAudio、ApmOutput三個AudioCommandThread線程:
1、 ApmTone用於播放tone音;
2、 ApmAudio用於執行audio命令;
3、ApmOutput用於執行輸出命令;
在第一次強引用AudioCommandThread線程對象時,AudioCommandThread的onFirstRef函數被回調,在此啓動線程
- void AudioPolicyService::AudioCommandThread::onFirstRef()
- {
- run(mName.string(), ANDROID_PRIORITY_AUDIO);
- }
這裏採用異步方式來執行audio command,當需要執行上表中的命令時,首先將命令投遞到AudioCommandThread的mAudioCommands命令向量表中,然後通過mWaitWorkCV.signal()喚醒AudioCommandThread線程,被喚醒的AudioCommandThread線程執行完command後,又通過mWaitWorkCV.waitRelative(mLock, waitTime)睡眠等待命令到來。
加載audio_policy_module模塊
audio_policy硬件抽象層動態庫位於/system/lib/hw/目錄下,命名爲:audio_policy.$(TARGET_BOARD_PLATFORM).so。audiopolicy的硬件抽象層定義在hardware\libhardware_legacy\audio\audio_policy_hal.cpp中,AUDIO_POLICY_HARDWARE_MODULE_ID硬件抽象模塊定義如下:
hardware\libhardware_legacy\audio\ audio_policy_hal.cpp【audio_policy.scx15.so】
- 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繼承於audio_policy_module。
關於hw_get_module函數加載硬件抽象層模塊的過程請參考Android硬件抽象Hardware庫加載過程源碼分析。
打開audio_policy_device設備
hardware\libhardware\include\hardware\ audio_policy.h
- 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);
- }
通過legacy_ap_module模塊的open方法來打開一個legacy_ap_device設備。
hardware\libhardware_legacy\audio\ audio_policy_hal.cpp
- 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;
- }
打開得到一個legacy_ap_device設備,通過該抽象設備可以創建一個audio_policy對象。
創建audio_policy對象
在打開legacy_ap_device設備時,該設備的create_audio_policy成員初始化爲create_legacy_ap函數指針,我們通過legacy_ap_device設備可以創建一個legacy_audio_policy對象。
- rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this,
- &mpAudioPolicy);
這裏通過audio_policy_device設備創建audio策略對象
hardware\libhardware_legacy\audio\ audio_policy_hal.cpp
- 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實現在audio_policy_hal.cpp中,audio_policy_service_ops實現在AudioPolicyService.cpp中。create_audio_policy()函數就是創建並初始化一個legacy_audio_policy對象。
audio_policy與AudioPolicyService、AudioPolicyCompatClient之間的關係如下:
AudioPolicyClient創建
hardware\libhardware_legacy\audio\ AudioPolicyCompatClient.h
- AudioPolicyCompatClient(struct audio_policy_service_ops *serviceOps,void *service) :
- mServiceOps(serviceOps) , mService(service) {}
AudioPolicyCompatClient是對audio_policy_service_ops的封裝類,對外提供audio_policy_service_ops數據結構中定義的接口。
AudioPolicyManager創建
- extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
- {
- ALOGI("SPRD policy manager created.");
- return new AudioPolicyManagerSPRD(clientInterface);
- }
使用AudioPolicyClientInterface對象來構造AudioPolicyManagerSPRD對象,AudioPolicyManagerSPRD繼承於AudioPolicyManagerBase,而AudioPolicyManagerBase又繼承於AudioPolicyInterface。
hardware\libhardware_legacy\audio\ AudioPolicyManagerBase.cpp
- AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
- :
- #ifdef AUDIO_POLICY_TEST
- Thread(false),
- #endif //AUDIO_POLICY_TEST
- //變量初始化
- 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對象,這樣音頻管理器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("");
- /**
- * 優先加載/vendor/etc/audio_policy.conf配置文件,如果該配置文件不存在,則
- * 加載/system/etc/audio_policy.conf配置文件,如果該文件還是不存在,則通過
- * 函數defaultAudioPolicyConfig()來設置默認音頻接口
- */
- 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();
- }
- }
- //設置各種音頻流對應的音量調節點,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++) {
- //通過名稱打開對應的音頻接口硬件抽象庫
- 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];
- //打開mAttachedOutputDevices對應的輸出
- if ((outProfile->mSupportedDevices & mAttachedOutputDevices) &&
- ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
- //將輸出IOProfile封裝爲AudioOutputDescriptor對象
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
- //設置當前音頻接口的默認輸出設備
- outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice & outProfile->mSupportedDevices);
- //打開輸出,在AudioFlinger中創建PlaybackThread線程,並返回該線程的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 {
- //設置可以使用的輸出設備爲mAttachedOutputDevices
- mAvailableOutputDevices =(audio_devices_t)(mAvailableOutputDevices | (outProfile->mSupportedDevices & mAttachedOutputDevices));
- if (mPrimaryOutput == 0 && outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
- mPrimaryOutput = output;
- }
- //將輸出描述符對象AudioOutputDescriptor及創建的PlaybackThread線程id以鍵值對形式保存
- addOutput(output, outputDesc);
- //設置默認輸出設備
- 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對象構造過程中主要完成以下幾個步驟:
1、 loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE)加載audio_policy.conf配置文件;
2、 initializeVolumeCurves()初始化各種音頻流對應的音量調節點;
3、 加載audio policy硬件抽象庫:mpClientInterface->loadHwModule(mHwModules[i]->mName)
4、 打開attached_output_devices輸出:
mpClientInterface->openOutput();
5、 保存輸出設備描述符對象:addOutput(output, outputDesc);
讀取audio_policy.conf文件
Android爲每種音頻接口定義了對應的硬件抽象層,且編譯爲單獨的so庫。
每種音頻接口定義了不同的輸入輸出,一個接口可以具有多個輸入或者輸出,每個輸入輸出有可以支持不同的音頻設備。通過讀取audio_policy.conf文件可以獲取系統支持的音頻接口參數。
audio_policy.conf文件定義了兩種音頻配置信息:
1、 當前系統支持的音頻輸入輸出設備及默認輸入輸出設備;
這些信息時通過global_configuration配置項來設置,在global_configuration中定義了三種音頻設備信息:
attached_output_devices:已連接的輸出設備;
default_output_device:默認輸出設備;
attached_input_devices:已連接的輸入設備;
1、 系統支持的音頻接口信息;
audio_policy.conf定義了系統支持的所有音頻接口參數信息,比如primary、a2dp、usb等,對於primary定義如下:
a2dp定義:
usb定義:
每種音頻接口包含輸入輸出,每種輸入輸出又包含多種輸入輸出配置,每種輸入輸出配置又支持多種音頻設備。AudioPolicyManagerBase首先加載/vendor/etc/audio_policy.conf,如果該文件不存在,則加/system/etc/audio_policy.conf。
- 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("", "");
- //讀取配置文件
- 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;
- }
通過loadGlobalConfig(root)函數來讀取這些全局配置信息。
- 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同時定義了多個audio 接口,每一個audio 接口包含若干output和input,而每個output和input又同時支持多種輸入輸出模式,每種輸入輸出模式又支持若干種設備。
通過loadHwModules ()函數來加載系統配置的所有audio 接口:
- 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);
- //加載音頻接口
- loadHwModule(node);
- node = node->next;
- }
- }
由於audio_policy.conf可以定義多個音頻接口,因此該函數循環調用loadHwModule()來解析每個音頻接口參數信息。Android定義HwModule類來描述每一個audio 接口參數,定義IOProfile類來描述輸入輸出模式配置。
到此就將audio_policy.conf文件中音頻接口配置信息解析到了AudioPolicyManagerBase的成員變量mHwModules、mAttachedOutputDevices、mDefaultOutputDevice、mAvailableInputDevices中。
初始化音量調節點
音量調節點設置在Android4.1與Android4.4中的實現完全不同,在Android4.1中是通過VolumeManager服務來管理,通過devicevolume.xml文件來配置,但Android4.4取消了VolumeManager服務,將音量控制放到AudioPolicyManagerBase中。在AudioPolicyManagerBase中定義了音量調節對應的音頻流描述符數組:
- StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES];
initializeVolumeCurves()函數就是初始化該數組元素:
- 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數組定義了不同音頻設備下不同音頻流對應的音量調節檔位,定義如下:
數組元素爲音量調節檔位,每種模式下的音量調節都包含4個檔位,定義如下:
加載audio_module模塊
AudioPolicyManager通過讀取audio_policy.conf配置文件,可以知道系統當前支持那些音頻接口以及attached的輸入輸出設備、默認輸出設備。接下來就需要加載這些音頻接口的硬件抽象庫。
這三中音頻接口硬件抽象定義如下:
/vendor/sprd/open-source/libs/audio/audio_hw.c 【audio.primary.scx15.so】
- 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】
- 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】
- 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提供了加載音頻接口硬件抽象庫的接口函數,通過前面的介紹,我們知道,AudioPolicyCompatClient通過代理audio_policy_service_ops實現AudioPolicyClientInterface接口。
hardware\libhardware_legacy\audio\ AudioPolicyCompatClient.cpp
- audio_module_handle_t AudioPolicyCompatClient::loadHwModule(const char *moduleName)
- {
- return mServiceOps->load_hw_module(mService, moduleName);
- }
AudioPolicyCompatClient將音頻模塊加載工作交給audio_policy_service_ops
frameworks\av\services\audioflinger\ AudioPolicyService.cpp
- static audio_module_handle_t aps_load_hw_module(void *service,const char *name)
- {
- sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == 0) {
- ALOGW("%s: could not get AudioFlinger", __func__);
- return 0;
- }
- return af->loadHwModule(name);
- }
AudioPolicyService又將其轉交給AudioFlinger
frameworks\av\services\audioflinger\ AudioFlinger.cpp
- audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
- {
- if (!settingsAllowed()) {
- return 0;
- }
- Mutex::Autolock _l(mLock);
- return loadHwModule_l(name);
- }
- 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;
- //加載音頻接口對應的so庫,得到對應的音頻接口設備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;
- }
函數首先加載系統定義的音頻接口對應的so庫,並打開該音頻接口的抽象硬件設備audio_hw_device_t,爲每個音頻接口設備生成獨一無二的ID號,同時將打開的音頻接口設備封裝爲AudioHwDevice對象,將系統中所有的音頻接口設備保存到AudioFlinger的成員變量mAudioHwDevs中。
函數load_audio_interface根據音頻接口名稱來打開抽象的音頻接口設備audio_hw_device_t。
- static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
- {
- const hw_module_t *mod;
- int rc;
- //根據名字加載audio_module模塊
- 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;
- }
- //打開audio_device設備
- 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
- 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
- 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;
- }
打開音頻接口設備過程其實就是構造並初始化legacy_audio_device對象過程,legacy_audio_device數據結構關係如下:
legacy_adev_open函數就是創建並初始化一個legacy_audio_device對象:
到此就加載完系統定義的所有音頻接口,並生成相應的數據對象,如下圖所示:
打開音頻輸出
AudioPolicyService加載完所有音頻接口後,就知道了系統支持的所有音頻接口參數,可以爲音頻輸出提供決策。
爲了能正常播放音頻數據,需要創建抽象的音頻輸出接口對象,打開音頻輸出過程如下:
- 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);
- }
- 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<IAudioFlinger> 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);
- }
- 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;
- }
- //創建一個音頻輸出流對象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);
- //從音頻接口列表mAudioHwDevs中查找出對應的音頻接口,如果找不到,則重新加載音頻接口動態庫
- outHwDev = findSuitableHwDev_l(module, *pDevices);
- if (outHwDev == NULL)
- return 0;
- //取出module對應的audio_hw_device_t設備
- audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
- //爲音頻輸出流生成一個獨一無二的id號
- audio_io_handle_t id = nextUniqueId();
- mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
- //打開音頻輸出流
- 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來封裝音頻輸出流audio_stream_out_t
- AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
- //根據flag標誌位,創建不同類型的線程
- 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);
- }
- //將創建的線程及id以鍵值對的形式保存在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;
- }
打開音頻輸出流過程其實就是創建AudioStreamOut對象及PlaybackThread線程過程。首先通過抽象的音頻接口設備audio_hw_device_t來創建輸出流對象legacy_stream_out。
- 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;
- //分配一個legacy_stream_out對象
- 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);
- //創建AudioStreamOut對象
- 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;
- }
- //初始化成員變量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;
- }
由於legacy_audio_device的成員變量hwif的類型爲AudioHardwareInterface,因此通過調用AudioHardwareInterface的接口openOutputStream()來創建AudioStreamOut對象。
- 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;
- }
打開音頻輸出後,在AudioFlinger與AudioPolicyService中的表現形式如下:
打開音頻輸入
- 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);
- }
- 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<IAudioFlinger> 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);
- }
- 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;
- }
打開音頻輸入流過程其實就是創建AudioStreamIn對象及RecordThread線程過程。首先通過抽象的音頻接口設備audio_hw_device_t來創建輸出流對象legacy_stream_in。
- 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;
- }
- 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;
- }
打開音頻輸入創建了以下legacy_stream_in對象:
打開音頻輸入後,在AudioFlinger與AudioPolicyService中的表現形式如下:
當AudioPolicyManagerBase構造時,它會根據用戶提供的audio_policy.conf來分析系統中有哪些audio接口(primary,a2dp以及usb),然後通過AudioFlinger::loadHwModule加載各audio接口對應的庫文件,並依次打開其中的output(openOutput)和input(openInput):
->打開音頻輸出時創建一個audio_stream_out通道,並創建AudioStreamOut對象以及新建PlaybackThread播放線程。
-> 打開音頻輸入時創建一個audio_stream_in通道,並創建AudioStreamIn對象以及創建RecordThread錄音線程。