（二）Audio子系统之new AudioRecord()（Android4.4）

在上一篇文章《（一）Audio子系统之AudioRecord.getMinBufferSize》中已经介绍了AudioRecord如何获取最小缓冲区大小，接下来，继续分析AudioRecorder方法中的new AudioRecorder的实现，本文基于Android4.4，Android5.1请戳这里

注：本篇文章仅作为笔记使用，如其中有误或与Android5.1版本的分析不同，以Android5.1版本的为准

函数原型：

public AudioRecord(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat,int bufferSizeInBytes) throws IllegalArgumentException

　　作用：

　　　　创建AudioRecord对象，建立录音通道

　　参数：

　　　　audioSource：录制源，这里设置MediaRecorder.AudioSource.Mic，其他请见MediaRecorder.AudioSource录制源定义，比如MediaRecorder.AudioSource.FM_TUNER等；

　　　　sampleRateInHz：默认采样率，单位Hz，这里设置为44100，44100Hz是当前唯一能保证在所有设备上工作的采样率；

　　　　channelConfig： 描述音频通道设置，这里设置为AudioFormat.CHANNEL_CONFIGURATION_MONO，CHANNEL_CONFIGURATION_MONO保证能在所有设备上工作；

　　　　audioFormat：音频数据保证支持此格式，这里设置为AudioFormat.ENCODING_16BIT；

　　　　bufferSizeInBytes：在录制过程中，音频数据写入缓冲区的总数（字节），即getMinVufferSize()获取到的值。

　　异常：

　　　　当参数设置不正确或不支持的参数时，将会抛出IllegalArgumentException

接下来进入系统分析具体实现

base\media\java\android\media\AudioRecord.java

public AudioRecord(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat,
         int bufferSizeInBytes)
 throws IllegalArgumentException {
     mRecordingState = RECORDSTATE_STOPPED;

     // remember which looper is associated with the AudioRecord instanciation
     if ((mInitializationLooper = Looper.myLooper()) == null) {
         mInitializationLooper = Looper.getMainLooper();
     }

     audioParamCheck(audioSource, sampleRateInHz, channelConfig, audioFormat);

     audioBuffSizeCheck(bufferSizeInBytes);

     // native initialization
     int[] session = new int[1];
     session[0] = 0;
     //TODO: update native initialization when information about hardware init failure
     //      due to capture device already open is available.
     int initResult = native_setup( new WeakReference<AudioRecord>(this), mRecordSource, mSampleRate, mChannelMask, mAudioFormat, mNativeBufferSizeInBytes, session);
     if (initResult != SUCCESS) {
         loge("Error code "+initResult+" when initializing native AudioRecord object.");
         return; // with mState == STATE_UNINITIALIZED
     }

     mSessionId = session[0];

     mState = STATE_INITIALIZED;
}

首先标记录音状态为停止，然后调用audioParamCheck函数判断参数是否非法，再调用audioBuffSizeCheck函数检查buffer大小，一切正常，开始调用native_setup方法进行注册，并把自己的WeakReference传过去。

base\core\jni\android_media_AudioRecord.cpp

static int
android_media_AudioRecord_setup(JNIEnv *env, jobject thiz, jobject weak_this,
        jint source, jint sampleRateInHertz, jint channelMask,
                // Java channel masks map directly to the native definition
        jint audioFormat, jint buffSizeInBytes, jintArray jSession)
{
    if (!audio_is_input_channel(channelMask)) {
        ALOGE("Error creating AudioRecord: channel mask %#x is not valid.", channelMask);
        return AUDIORECORD_ERROR_SETUP_INVALIDCHANNELMASK;
    }
    uint32_t nbChannels = popcount(channelMask);

    // compare the format against the Java constants
    if ((audioFormat != ENCODING_PCM_16BIT)
        && (audioFormat != ENCODING_PCM_8BIT)) {
        ALOGE("Error creating AudioRecord: unsupported audio format.");
        return AUDIORECORD_ERROR_SETUP_INVALIDFORMAT;
    }

    int bytesPerSample = audioFormat == ENCODING_PCM_16BIT ? 2 : 1;
    audio_format_t format = audioFormat == ENCODING_PCM_16BIT ?
            AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_8_BIT;

    if (buffSizeInBytes == 0) {
         ALOGE("Error creating AudioRecord: frameCount is 0.");
        return AUDIORECORD_ERROR_SETUP_ZEROFRAMECOUNT;
    }
    int frameSize = nbChannels * bytesPerSample;
    size_t frameCount = buffSizeInBytes / frameSize;

    if ((uint32_t(source) >= AUDIO_SOURCE_CNT) && (uint32_t(source) != AUDIO_SOURCE_HOTWORD)) {
        ALOGE("Error creating AudioRecord: unknown source.");
        return AUDIORECORD_ERROR_SETUP_INVALIDSOURCE;
    }

    jclass clazz = env->GetObjectClass(thiz);
    if (clazz == NULL) {
        ALOGE("Can't find %s when setting up callback.", kClassPathName);
        return AUDIORECORD_ERROR_SETUP_NATIVEINITFAILED;
    }

    if (jSession == NULL) {
        ALOGE("Error creating AudioRecord: invalid session ID pointer");
        return AUDIORECORD_ERROR;
    }

    jint* nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
    if (nSession == NULL) {
        ALOGE("Error creating AudioRecord: Error retrieving session id pointer");
        return AUDIORECORD_ERROR;
    }
    int sessionId = nSession[0];
    env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
    nSession = NULL;

    // create an uninitialized AudioRecord object
    sp<AudioRecord> lpRecorder = new AudioRecord();

    // create the callback information:
    // this data will be passed with every AudioRecord callback
    audiorecord_callback_cookie *lpCallbackData = new audiorecord_callback_cookie;
    lpCallbackData->audioRecord_class = (jclass)env->NewGlobalRef(clazz);
    // we use a weak reference so the AudioRecord object can be garbage collected.
    lpCallbackData->audioRecord_ref = env->NewGlobalRef(weak_this);
    lpCallbackData->busy = false;

    lpRecorder->set((audio_source_t) source,
        sampleRateInHertz,
        format,        // word length, PCM
        channelMask,
        frameCount,
        recorderCallback,// callback_t
        lpCallbackData,// void* user
        0,             // notificationFrames,
        true,          // threadCanCallJava
        sessionId);

    if (lpRecorder->initCheck() != NO_ERROR) {
        ALOGE("Error creating AudioRecord instance: initialization check failed.");
        goto native_init_failure;
    }

    nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
    if (nSession == NULL) {
        ALOGE("Error creating AudioRecord: Error retrieving session id pointer");
        goto native_init_failure;
    }
    // read the audio session ID back from AudioRecord in case a new session was created during set()
    nSession[0] = lpRecorder->getSessionId();
    env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
    nSession = NULL;

    {   // scope for the lock
        Mutex::Autolock l(sLock);
        sAudioRecordCallBackCookies.add(lpCallbackData);
    }
    // save our newly created C++ AudioRecord in the "nativeRecorderInJavaObj" field
    // of the Java object
    setAudioRecord(env, thiz, lpRecorder);

    // save our newly created callback information in the "nativeCallbackCookie" field
    // of the Java object (in mNativeCallbackCookie) so we can free the memory in finalize()
    env->SetIntField(thiz, javaAudioRecordFields.nativeCallbackCookie, (int)lpCallbackData);

    return AUDIORECORD_SUCCESS;

    // failure:
native_init_failure:
    env->DeleteGlobalRef(lpCallbackData->audioRecord_class);
    env->DeleteGlobalRef(lpCallbackData->audioRecord_ref);
    delete lpCallbackData;
    env->SetIntField(thiz, javaAudioRecordFields.nativeCallbackCookie, 0);

    return AUDIORECORD_ERROR_SETUP_NATIVEINITFAILED;
}

static JNINativeMethod gMethods[] = {
    // name,               signature,  funcPtr
    {"native_start",         "(II)I",    (void *)android_media_AudioRecord_start},
    {"native_stop",          "()V",    (void *)android_media_AudioRecord_stop},
    {"native_setup",         "(Ljava/lang/Object;IIIII[I)I",
                                       (void *)android_media_AudioRecord_setup},
    {"native_finalize",      "()V",    (void *)android_media_AudioRecord_finalize},
    {"native_release",       "()V",    (void *)android_media_AudioRecord_release},
    {"native_read_in_byte_array",
                             "([BII)I", (void *)android_media_AudioRecord_readInByteArray},
    {"native_read_in_short_array",
                             "([SII)I", (void *)android_media_AudioRecord_readInShortArray},
    {"native_read_in_direct_buffer","(Ljava/lang/Object;I)I",
                                       (void *)android_media_AudioRecord_readInDirectBuffer},
    {"native_set_marker_pos","(I)I",   (void *)android_media_AudioRecord_set_marker_pos},
    {"native_get_marker_pos","()I",    (void *)android_media_AudioRecord_get_marker_pos},
    {"native_set_pos_update_period",
                             "(I)I",   (void *)android_media_AudioRecord_set_pos_update_period},
    {"native_get_pos_update_period",
                             "()I",    (void *)android_media_AudioRecord_get_pos_update_period},
    {"native_get_min_buff_size",
                             "(III)I",   (void *)android_media_AudioRecord_get_min_buff_size},
};

在JNI中继续调用AudioRecoder的set函数，同时把callback回调绑定起来，其中set函数的定义在frameworks\av\include\media\AudioRecord.h

status_t    set(audio_source_t inputSource,
                uint32_t sampleRate,
                audio_format_t format,
                audio_channel_mask_t channelMask,
                int frameCount      = 0,
                callback_t cbf = NULL,
                void* user = NULL,
                int notificationFrames = 0,
                bool threadCanCallJava = false,
                int sessionId = 0,
                transfer_type transferType = TRANSFER_DEFAULT,
                audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);

这里设置了transfer_type为默认

frameworks\av\media\libmedia\AudioRecord.cpp

status_t AudioRecord::set(
        audio_source_t inputSource,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        int frameCountInt,
        callback_t cbf,
        void* user,
        int notificationFrames,
        bool threadCanCallJava,
        int sessionId,
        transfer_type transferType,
        audio_input_flags_t flags)
{
    switch (transferType) {    case TRANSFER_DEFAULT: 

        if (cbf == NULL || threadCanCallJava) {transferType = TRANSFER_SYNC; 

        } else {
            transferType = TRANSFER_CALLBACK; 

        }
        break;
    case TRANSFER_CALLBACK:
        if (cbf == NULL) {
            ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
            return BAD_VALUE;
        }
        break;
    case TRANSFER_OBTAIN:
    case TRANSFER_SYNC:
        break;
    default:
        ALOGE("Invalid transfer type %d", transferType);
        return BAD_VALUE;
    }
    mTransfer = transferType;

    // FIXME "int" here is legacy and will be replaced by size_t later
    if (frameCountInt < 0) {
        ALOGE("Invalid frame count %d", frameCountInt);
        return BAD_VALUE;
    }
    size_t frameCount = frameCountInt;

    ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
            frameCount);

    AutoMutex lock(mLock);

    if (mAudioRecord != 0) {
        ALOGE("Track already in use");
        return INVALID_OPERATION;
    }

    if (inputSource == AUDIO_SOURCE_DEFAULT) {
        inputSource = AUDIO_SOURCE_MIC;
    }
    mInputSource = inputSource;

    if (sampleRate == 0) {
        ALOGE("Invalid sample rate %u", sampleRate);
        return BAD_VALUE;
    }
    mSampleRate = sampleRate;

    // these below should probably come from the audioFlinger too...
    if (format == AUDIO_FORMAT_DEFAULT) {
        format = AUDIO_FORMAT_PCM_16_BIT;
    }

    // validate parameters
    if (!audio_is_valid_format(format)) {
        ALOGE("Invalid format %d", format);
        return BAD_VALUE;
    }
    // Temporary restriction: AudioFlinger currently supports 16-bit PCM only
    if (format != AUDIO_FORMAT_PCM_16_BIT) {
        ALOGE("Format %d is not supported", format);
        return BAD_VALUE;
    }
    mFormat = format;

    if (!audio_is_input_channel(channelMask)) {
        ALOGE("Invalid channel mask %#x", channelMask);
        return BAD_VALUE;
    }
    mChannelMask = channelMask;
    uint32_t channelCount = popcount(channelMask);
    mChannelCount = channelCount;

    // Assumes audio_is_linear_pcm(format), else sizeof(uint8_t)
    mFrameSize = channelCount * audio_bytes_per_sample(format);

    // validate framecount
    size_t minFrameCount = 0;
    status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
            sampleRate, format, channelMask);
    if (status != NO_ERROR) {
        ALOGE("getMinFrameCount() failed; status %d", status);
        return status;
    }
    ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);

    if (frameCount == 0) {
        frameCount = minFrameCount;
    } else if (frameCount < minFrameCount) {
        ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
        return BAD_VALUE;
    }
    mFrameCount = frameCount;

    mNotificationFramesReq = notificationFrames;
    mNotificationFramesAct = 0;

    if (sessionId == 0 ) {
        mSessionId = AudioSystem::newAudioSessionId();
    } else {
        mSessionId = sessionId;
    }
    ALOGV("set(): mSessionId %d", mSessionId);

    mFlags = flags;

    // create the IAudioRecord      status = openRecord_l(0 /*epoch*/); 

    if (status) {
        return status;
    }

    if (cbf != NULL) {
        mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
        mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
    }

    mStatus = NO_ERROR;

    // Update buffer size in case it has been limited by AudioFlinger during track creation
    mFrameCount = mCblk->frameCount_;

    mActive = false;
    mCbf = cbf;
    mRefreshRemaining = true;
    mUserData = user;
    // TODO: add audio hardware input latency here
    mLatency = (1000*mFrameCount) / sampleRate;
    mMarkerPosition = 0;
    mMarkerReached = false;
    mNewPosition = 0;
    mUpdatePeriod = 0;
    AudioSystem::acquireAudioSessionId(mSessionId);
    mSequence = 1;
    mObservedSequence = mSequence;
    mInOverrun = false;

    return NO_ERROR;
}

首先对参数进行二次处理，可以看到SOURCE_DEFAULT的源其实也指向了MIC，FORMAT_DEFAULT其实就是PCM_16_BIT，然后调用getMinFrameCount获取最小frame数量，最后调用openRecord_l()函数创建IAudioRecord对象，并开启一个AudioRecordThread线程，这里着重分析openRecord_l函数

status_t AudioRecord::openRecord_l(size_t epoch)
{
    status_t status;
    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
    if (audioFlinger == 0) {
        ALOGE("Could not get audioflinger");
        return NO_INIT;
    }

    IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
    pid_t tid = -1;

    // Client can only express a preference for FAST.  Server will perform additional tests.
    // The only supported use case for FAST is callback transfer mode.
    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
        if ((mTransfer != TRANSFER_CALLBACK) || (mAudioRecordThread == 0)) {
            ALOGW("AUDIO_INPUT_FLAG_FAST denied by client");
            // once denied, do not request again if IAudioRecord is re-created
            mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
        } else {
            trackFlags |= IAudioFlinger::TRACK_FAST;
            tid = mAudioRecordThread->getTid();
        }
    }

    mNotificationFramesAct = mNotificationFramesReq;

    if (!(mFlags & AUDIO_INPUT_FLAG_FAST)) {
        // Make sure that application is notified with sufficient margin before overrun
        if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) {
            mNotificationFramesAct = mFrameCount/2;
        }
    }

    audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat,
            mChannelMask, mSessionId);
    if (input == 0) {
        ALOGE("Could not get audio input for record source %d", mInputSource);
        return BAD_VALUE;
    }

    int originalSessionId = mSessionId;
    sp<IAudioRecord> record = audioFlinger->openRecord(input,
                                                       mSampleRate, mFormat,
                                                       mChannelMask,
                                                       mFrameCount,
                                                       &trackFlags,
                                                       tid,
                                                       &mSessionId,
                                                       &status);
    ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId,
            "session ID changed from %d to %d", originalSessionId, mSessionId);

    if (record == 0 || status != NO_ERROR) {
        ALOGE("AudioFlinger could not create record track, status: %d", status);
        AudioSystem::releaseInput(input);
        return status;
    }
    sp<IMemory> iMem = record->getCblk();
    if (iMem == 0) {
        ALOGE("Could not get control block");
        return NO_INIT;
    }
    void *iMemPointer = iMem->pointer();
    if (iMemPointer == NULL) {
        ALOGE("Could not get control block pointer");
        return NO_INIT;
    }
    if (mAudioRecord != 0) {
        mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
        mDeathNotifier.clear();
    }
    mInput = input;
    mAudioRecord = record;
    mCblkMemory = iMem;
    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
    mCblk = cblk;
    // FIXME missing fast track frameCount logic
    mAwaitBoost = false;
    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
        if (trackFlags & IAudioFlinger::TRACK_FAST) {
            ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", mFrameCount);
            mAwaitBoost = true;
            // double-buffering is not required for fast tracks, due to tighter scheduling
            if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount) {
                mNotificationFramesAct = mFrameCount;
            }
        } else {
            ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", mFrameCount);
            // once denied, do not request again if IAudioRecord is re-created
            mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
            if (mNotificationFramesAct == 0 || mNotificationFramesAct > mFrameCount/2) {
                mNotificationFramesAct = mFrameCount/2;
            }
        }
    }

    // starting address of buffers in shared memory
    void *buffers = (char*)cblk + sizeof(audio_track_cblk_t);

    // update proxy
    mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
    mProxy->setEpoch(epoch);
    mProxy->setMinimum(mNotificationFramesAct);

    mDeathNotifier = new DeathNotifier(this);
    mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);

    return NO_ERROR;
}

先分析下这个函数的主要工作：

1.AudioSystem::get_audio_flinger()获取AudioFlinger服务

    2.AudioSystem::getInput获取输入流句柄

3.audioFlinger->openRecord获取IAudioRecord对象

4.通过IMemory共享内存，获取录音数据

5.通过AudioRecordClientProxy客户端更新录音数据

接下来再具体分析下这5步

1.首先获取AudioFlinger服务AudioSystem::get_audio_flinger()

frameworks\av\media\libmedia\AudioSystem.cpp

const sp<IAudioFlinger>& AudioSystem::get_audio_flinger()
{
    Mutex::Autolock _l(gLock);
    if (gAudioFlinger == 0) {
        sp<IServiceManager> sm = defaultServiceManager();
        sp<IBinder> binder;
        do {
            binder = sm->getService(String16("media.audio_flinger"));
            if (binder != 0)
                break;
            ALOGW("AudioFlinger not published, waiting...");
            usleep(500000); // 0.5 s
        } while (true);
        if (gAudioFlingerClient == NULL) {
            gAudioFlingerClient = new AudioFlingerClient();
        } else {
            if (gAudioErrorCallback) {
                gAudioErrorCallback(NO_ERROR);
            }
        }
        binder->linkToDeath(gAudioFlingerClient);
        gAudioFlinger = interface_cast<IAudioFlinger>(binder);
        gAudioFlinger->registerClient(gAudioFlingerClient);
    }
    ALOGE_IF(gAudioFlinger==0, "no AudioFlinger!?");

    return gAudioFlinger;
}

然后获取input通道设备AudioSystem::getInput()

frameworks\av\media\libmedia\AudioSystem.cpp

audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
                                    uint32_t samplingRate,
                                    audio_format_t format,
                                    audio_channel_mask_t channelMask,
                                    int sessionId)
{
    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
    if (aps == 0) return 0;
    return aps->getInput(inputSource, samplingRate, format, channelMask, sessionId);
}

获取policy服务AudioSystem::get_audio_policy_service()

const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
{
    gLock.lock();
    if (gAudioPolicyService == 0) {
        sp<IServiceManager> sm = defaultServiceManager();
        sp<IBinder> binder;
        do {
            binder = sm->getService(String16("media.audio_policy"));
            if (binder != 0)
                break;
            ALOGW("AudioPolicyService not published, waiting...");
            usleep(500000); // 0.5 s
        } while (true);
        if (gAudioPolicyServiceClient == NULL) {
            gAudioPolicyServiceClient = new AudioPolicyServiceClient();
        }
        binder->linkToDeath(gAudioPolicyServiceClient);
        gAudioPolicyService = interface_cast<IAudioPolicyService>(binder);
        gLock.unlock();
    } else {
        gLock.unlock();
    }
    return gAudioPolicyService;
}

然后调用函数AudioPolicyService::getInput()

frameworks\av\services\audioflinger\AudioPolicyService.cpp

audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,
                                    uint32_t samplingRate,
                                    audio_format_t format,
                                    audio_channel_mask_t channelMask,
                                    int audioSession)
{
    if (mpAudioPolicy == NULL) {
        return 0;
    }
    // already checked by client, but double-check in case the client wrapper is bypassed
    if (inputSource >= AUDIO_SOURCE_CNT && inputSource != AUDIO_SOURCE_HOTWORD) {
        return 0;
    }

    if ((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) {
        return 0;
    }

    Mutex::Autolock _l(mLock);
    // the audio_in_acoustics_t parameter is ignored by get_input()
    audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
                                                   format, channelMask, (audio_in_acoustics_t) 0);

    if (input == 0) {
        return input;
    }
    // create audio pre processors according to input source
    audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ?
                                    AUDIO_SOURCE_VOICE_RECOGNITION : inputSource;

    ssize_t index = mInputSources.indexOfKey(aliasSource);
    if (index < 0) {
        return input;
    }
    ssize_t idx = mInputs.indexOfKey(input);
    InputDesc *inputDesc;
    if (idx < 0) {
        inputDesc = new InputDesc(audioSession);
        mInputs.add(input, inputDesc);
    } else {
        inputDesc = mInputs.valueAt(idx);
    }

    Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
    for (size_t i = 0; i < effects.size(); i++) {
        EffectDesc *effect = effects[i];
        sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
        status_t status = fx->initCheck();
        if (status != NO_ERROR && status != ALREADY_EXISTS) {
            ALOGW("Failed to create Fx %s on input %d", effect->mName, input);
            // fx goes out of scope and strong ref on AudioEffect is released
            continue;
        }
        for (size_t j = 0; j < effect->mParams.size(); j++) {
            fx->setParameter(effect->mParams[j]);
        }
        inputDesc->mEffects.add(fx);
    }
    setPreProcessorEnabled(inputDesc, true);
    return input;
}

继续调用mpAudioPolicy->get_input()，这里struct audio_policy *mpAudioPolicy;该结构体在#include <hardware/audio_policy.h>，也就是说接下来将进入HAL层

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.get_device_connection_state = ap_get_device_connection_state;
    lap->policy.set_phone_state = ap_set_phone_state;
    lap->policy.set_ringer_mode = ap_set_ringer_mode;
    lap->policy.set_force_use = ap_set_force_use;
    lap->policy.get_force_use = ap_get_force_use;
    lap->policy.set_can_mute_enforced_audible =
        ap_set_can_mute_enforced_audible;
    lap->policy.init_check = ap_init_check;
    lap->policy.get_output = ap_get_output;
    lap->policy.start_output = ap_start_output;
    lap->policy.stop_output = ap_stop_output;
    lap->policy.release_output = ap_release_output;
    lap->policy.get_input = ap_get_input;
    lap->policy.start_input = ap_start_input;
    lap->policy.stop_input = ap_stop_input;
    lap->policy.release_input = ap_release_input;
    lap->policy.init_stream_volume = ap_init_stream_volume;
    lap->policy.set_stream_volume_index = ap_set_stream_volume_index;
    lap->policy.get_stream_volume_index = ap_get_stream_volume_index;
    lap->policy.set_stream_volume_index_for_device = ap_set_stream_volume_index_for_device;
    lap->policy.get_stream_volume_index_for_device = ap_get_stream_volume_index_for_device;
    lap->policy.get_strategy_for_stream = ap_get_strategy_for_stream;
    lap->policy.get_devices_for_stream = ap_get_devices_for_stream;
    lap->policy.get_output_for_effect = ap_get_output_for_effect;
    lap->policy.register_effect = ap_register_effect;
    lap->policy.unregister_effect = ap_unregister_effect;
    lap->policy.set_effect_enabled = ap_set_effect_enabled;
    lap->policy.is_stream_active = ap_is_stream_active;
    lap->policy.is_stream_active_remotely = ap_is_stream_active_remotely;
    lap->policy.is_source_active = ap_is_source_active;
    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;
}

ap_get_input()函数实现如下

static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,
                                      uint32_t sampling_rate,
                                      audio_format_t format,
                                      audio_channel_mask_t channelMask,
                                      audio_in_acoustics_t acoustics)
{
    struct legacy_audio_policy *lap = to_lap(pol);
    return lap->apm->getInput((int) inputSource, sampling_rate, (int) format, channelMask,
                              (AudioSystem::audio_in_acoustics)acoustics);
}

先看legacy_audio_policy结构体

struct legacy_audio_policy {
    struct audio_policy policy;

    void *service;
    struct audio_policy_service_ops *aps_ops;
    AudioPolicyCompatClient *service_client;
    AudioPolicyInterface *apm;
};

接着调用getInput()

hardware\libhardware_legacy\audio\AudioPolicyManagerBase.cpp

audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
                                    uint32_t samplingRate,
                                    uint32_t format,
                                    uint32_t channelMask,
                                    AudioSystem::audio_in_acoustics acoustics)
{
    audio_io_handle_t input = 0;
    audio_devices_t device = getDeviceForInputSource(inputSource);

    ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
          inputSource, samplingRate, format, channelMask, acoustics);

    if (device == AUDIO_DEVICE_NONE) {
        ALOGW("getInput() could not find device for inputSource %d", inputSource);
        return 0;
    }

    // adapt channel selection to input source
    switch(inputSource) {
    case AUDIO_SOURCE_VOICE_UPLINK:
        channelMask = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
        break;
    case AUDIO_SOURCE_VOICE_DOWNLINK:
        channelMask = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
        break;
    case AUDIO_SOURCE_VOICE_CALL:
        channelMask = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
        break;
    default:
        break;
    }

    IOProfile *profile = getInputProfile(device,
                                         samplingRate,
                                         format,
                                         channelMask);
    if (profile == NULL) {
        ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d,"
                "channelMask %04x",
                device, samplingRate, format, channelMask);
        return 0;
    }

    if (profile->mModule->mHandle == 0) {
        ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
        return 0;
    }

    AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);

    inputDesc->mInputSource = inputSource;
    inputDesc->mDevice = device;
    inputDesc->mSamplingRate = samplingRate;
    inputDesc->mFormat = (audio_format_t)format;
    inputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
    inputDesc->mRefCount = 0;
    input = mpClientInterface->openInput(profile->mModule->mHandle,
                                    &inputDesc->mDevice,
                                    &inputDesc->mSamplingRate,
                                    &inputDesc->mFormat,
                                    &inputDesc->mChannelMask);

    // only accept input with the exact requested set of parameters
    if (input == 0 ||
        (samplingRate != inputDesc->mSamplingRate) ||
        (format != inputDesc->mFormat) ||
        (channelMask != inputDesc->mChannelMask)) {
        ALOGV("getInput() failed opening input: samplingRate %d, format %d, channelMask %d",
                samplingRate, format, channelMask);
        if (input != 0) {
            mpClientInterface->closeInput(input);
        }
        delete inputDesc;
        return 0;
    }
    mInputs.add(input, inputDesc);
    return input;
}

在这个函数中调用getDeviceForInputSource获取输入设备，再调用getInputProfile获取配置信息，最后打开输入设备

mpClientInterface->openInputhardware\libhardware_legacy\audio\AudioPolicyCompatClient.cpp

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);
}

其中：struct audio_policy_service_ops* mServiceOps;，该结构体为Audio output Control functions，再继续，又回到Frameworks了

frameworks\av\services\audioflinger\AudioPolicyService.cpp

namespace {
    struct audio_policy_service_ops aps_ops = {
        open_output           : aps_open_output,
        open_duplicate_output : aps_open_dup_output,
        close_output          : aps_close_output,
        suspend_output        : aps_suspend_output,
        restore_output        : aps_restore_output,
        open_input            : aps_open_input,
        close_input           : aps_close_input,
        set_stream_volume     : aps_set_stream_volume,
        set_stream_output     : aps_set_stream_output,
        set_parameters        : aps_set_parameters,
        get_parameters        : aps_get_parameters,
        start_tone            : aps_start_tone,
        stop_tone             : aps_stop_tone,
        set_voice_volume      : aps_set_voice_volume,
        move_effects          : aps_move_effects,
        load_hw_module        : aps_load_hw_module,
        open_output_on_module : aps_open_output_on_module,
        open_input_on_module  : aps_open_input_on_module,
    };
}; // namespace <unnamed>

也就是说对应到aps_open_input_on_module函数中

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);
}

这里又需要调用Flinger的openInput函数

frameworks\av\services\audioflinger\AudioFlinger.cpp

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
        sp<NBAIO_Sink> teeSink;
        enum {
            TEE_SINK_NO,    // don't copy input
            TEE_SINK_NEW,   // copy input using a new pipe
            TEE_SINK_OLD,   // copy input using an existing pipe
        } kind;
        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
                                        popcount(inStream->common.get_channels(&inStream->common)));
        if (!mTeeSinkInputEnabled) {
            kind = TEE_SINK_NO;
        } else if (format == Format_Invalid) {
            kind = TEE_SINK_NO;
        } else if (mRecordTeeSink == 0) {
            kind = TEE_SINK_NEW;
        } else if (mRecordTeeSink->getStrongCount() != 1) {
            kind = TEE_SINK_NO;
        } else if (format == mRecordTeeSink->format()) {
            kind = TEE_SINK_OLD;
        } else {
            kind = TEE_SINK_NEW;
        }
        switch (kind) {
        case TEE_SINK_NEW: {
            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
            size_t numCounterOffers = 0;
            const NBAIO_Format offers[1] = {format};
            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
            ALOG_ASSERT(index == 0);
            PipeReader *pipeReader = new PipeReader(*pipe);
            numCounterOffers = 0;
            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
            ALOG_ASSERT(index == 0);
            mRecordTeeSink = pipe;
            mRecordTeeSource = pipeReader;
            teeSink = pipe;
            }
            break;
        case TEE_SINK_OLD:
            teeSink = mRecordTeeSink;
            break;
        case TEE_SINK_NO:
        default:
            break;
        }
#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;
}

找到一个可用的device：inHwDev = findSuitableHwDev_l

AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
        audio_module_handle_t module,
        audio_devices_t devices)
{
    // if module is 0, the request comes from an old policy manager and we should load
    // well known modules
    if (module == 0) {
        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
            loadHwModule_l(audio_interfaces[i]);
        }
        // then try to find a module supporting the requested device.
        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
            audio_hw_device_t *dev = audioHwDevice->hwDevice();
            if ((dev->get_supported_devices != NULL) &&
                    (dev->get_supported_devices(dev) & devices) == devices)
                return audioHwDevice;
        }
    } else {
        // check a match for the requested module handle
        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
        if (audioHwDevice != NULL) {
            return audioHwDevice;
        }
    }

    return NULL;
}

然后打开输入流：inHwHal->open_input_stream，这里又进入HAL

hardware\libhardware_legacy\audio\audio_hw_hal.cpp

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;
    in->stream.common.set_sample_rate = in_set_sample_rate;
    in->stream.common.get_buffer_size = in_get_buffer_size;
    in->stream.common.get_channels = in_get_channels;
    in->stream.common.get_format = in_get_format;
    in->stream.common.set_format = in_set_format;
    in->stream.common.standby = in_standby;
    in->stream.common.dump = in_dump;
    in->stream.common.set_parameters = in_set_parameters;
    in->stream.common.get_parameters = in_get_parameters;
    in->stream.common.add_audio_effect = in_add_audio_effect;
    in->stream.common.remove_audio_effect = in_remove_audio_effect;
    in->stream.set_gain = in_set_gain;
    in->stream.read = in_read;
    in->stream.get_input_frames_lost = in_get_input_frames_lost;

    *stream_in = &in->stream;
    return 0;

err_open:
    free(in);
    *stream_in = NULL;
    return ret;
}

对于openInputStream函数来说，申明在hardware\libhardware_legacy\include\hardware_legacy\AudioHardwareInterface.h中

但是在libhardware_legacy\audio\AudioHardwareGeneric.cpp与libhardware_legacy\audio\AudioHardwareStub.cpp中都继承了该接口，并实现了他

AudioHardwareStub.cpp：

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;
}

AudioHardwareGeneric.cpp：

AudioStreamIn* AudioHardwareGeneric::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;
    }

    AutoMutex lock(mLock);

    // only one input stream allowed
    if (mInput) {
        if (status) {
            *status = INVALID_OPERATION;
        }
        return 0;
    }

    // create new output stream
    AudioStreamInGeneric* in = new AudioStreamInGeneric();
    status_t lStatus = in->set(this, mFd, devices, format, channels, sampleRate, acoustics);
    if (status) {
        *status = lStatus;
    }
    if (lStatus == NO_ERROR) {
        mInput = in;
    } else {
        delete in;
    }
    return mInput;
}