Android AudioPolicyService服务启动过程

转至:https://www.cnblogs.com/senior-engineer/p/5089559.html

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。

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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

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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

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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。

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static const char *getServiceName() { return "media.audio_policy"; }

AudioPolicyService注册名为media.audio_policy的服务。

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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);

    }

}

    1. 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

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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

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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】

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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。

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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

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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

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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。

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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。

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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

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hardware\libhardware_legacy\audio\ audio_policy_hal.cpp

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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。

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audio_policy与AudioPolicyService、AudioPolicyCompatClient之间的关系如下:

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AudioPolicyClient创建

hardware\libhardware_legacy\audio\ AudioPolicyCompatClient.h

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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

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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。

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hardware\libhardware_legacy\audio\ AudioPolicyManagerBase.cpp

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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);

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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。

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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;

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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:

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a2dp定义:

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usb定义:

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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。

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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。

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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:

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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:

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StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES];

initializeVolumeCurves()函数就是初始化该数组元素:

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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。

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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】

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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】

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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】

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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

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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

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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

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audio_module_handle_t AudioFlinger::loadHwModule(const char *name)

{

    if (!settingsAllowed()) {

        return 0;

    }

    Mutex::Autolock _l(mLock);

    return loadHwModule_l(name);

}

\

 

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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。

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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

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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

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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:

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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);

}

 

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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);

}

\


 

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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。

\

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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。

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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

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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);

}

 

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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);

}

\

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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。

\

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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;

}

 

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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。

\

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