在上一篇文章《（三）Audio子系統之AudioRecord.startRecording》中已經介紹了AudioRecord如何開始錄製音訊，接下來，繼續分析AudioRecord方法中的read的實現

 

　　

 

　　函式原型：

 

　　　　public int read(byte[] audioData, int offsetInBytes, int sizeInBytes)

 

       作用：

 

　　　　從音訊硬體錄製緩衝區讀取資料，直接複製到指定緩衝區。

如果audioBuffer不是直接的緩衝區，此方法總是返回0

 

　　引數：

 

　　　　audioData：寫入的音訊錄製資料

 

　　　　offsetInBytes：audioData的起始偏移值，單位byte

 

　　　　sizeInBytes：讀取的最大位元組數

 

　　返回值：

 

　　　　讀入緩衝區的總byte數，如果物件屬性沒有初始化，則返回ERROR_INVALID_OPERATION，如果引數不能解析成有效的資料或索引，則返回ERROR_BAD_VALUE。 讀取的總byte數不會超過sizeInBytes

 

 

 

接下來進入系統分析具體實現

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

    public int read(byte[] audioData, int offsetInBytes, int sizeInBytes) {
        if (mState != STATE_INITIALIZED) {
            return ERROR_INVALID_OPERATION;
        }

        if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
                || (offsetInBytes + sizeInBytes < 0)  // detect integer overflow
                || (offsetInBytes + sizeInBytes > audioData.length)) {
            return ERROR_BAD_VALUE;
        }

        return native_read_in_byte_array(audioData, offsetInBytes, sizeInBytes);
    }

這裡我們只分析獲取byte[]型別的資料

frameworks\base\core\jni\android_media_AudioRecord.cpp

static jint android_media_AudioRecord_readInByteArray(JNIEnv *env,  jobject thiz,
                                                        jbyteArray javaAudioData,
                                                        jint offsetInBytes, jint sizeInBytes) {
    jbyte* recordBuff = NULL;
    // get the audio recorder from which we'll read new audio samples
    sp<AudioRecord> lpRecorder = getAudioRecord(env, thiz);
    if (lpRecorder == NULL) {
        ALOGE("Unable to retrieve AudioRecord object, can't record");
        return 0;
    }

    if (!javaAudioData) {
        ALOGE("Invalid Java array to store recorded audio, can't record");
        return 0;
    }

    recordBuff = (jbyte *)env->GetByteArrayElements(javaAudioData, NULL);

    if (recordBuff == NULL) {
        ALOGE("Error retrieving destination for recorded audio data, can't record");
        return 0;
    }

    // read the new audio data from the native AudioRecord object
    ssize_t recorderBuffSize = lpRecorder->frameCount()*lpRecorder->frameSize();

    ssize_t readSize = lpRecorder->read(recordBuff + offsetInBytes,
                                        sizeInBytes > (jint)recorderBuffSize ?
                                            (jint)recorderBuffSize : sizeInBytes );
    env->ReleaseByteArrayElements(javaAudioData, recordBuff, 0);

    if (readSize < 0) {
        readSize = (jint)AUDIO_JAVA_INVALID_OPERATION;
    }
    return (jint) readSize;
}

這裡根據AudioRecord.cpp中的mFrameCount*mFrameSize 重新計算出Audio緩衝區大小，並與傳過來的size比較，選擇較小的那個數，然後再呼叫lpRecorder->read函式，需要注意的是mFrameCount這個值在呼叫openRecord_l函式的時候更新過，我在實際測試的時候發現mFrameCount重新計算過後為3072，而之前是2048，所以這裡傳過去的buffSize依然是4092 frameworks\av\media\libmedia\AudioRecord.cpp

ssize_t AudioRecord::read(void* buffer, size_t userSize)
{
    if (mTransfer != TRANSFER_SYNC) {
        return INVALID_OPERATION;
    }

    if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
        // sanity-check. user is most-likely passing an error code, and it would
        // make the return value ambiguous (actualSize vs error).
        ALOGE("AudioRecord::read(buffer=%p, size=%zu (%zu)", buffer, userSize, userSize);
        return BAD_VALUE;
    }

    ssize_t read = 0;
    Buffer audioBuffer;

    while (userSize >= mFrameSize) {
        audioBuffer.frameCount = userSize / mFrameSize;

        status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever);
        if (err < 0) {
            if (read > 0) {
                break;
            }
            return ssize_t(err);
        }

        size_t bytesRead = audioBuffer.size;
        memcpy(buffer, audioBuffer.i8, bytesRead);
        buffer = ((char *) buffer) + bytesRead;
        userSize -= bytesRead;
		
        read += bytesRead;

        releaseBuffer(&audioBuffer);
    }

    return read;
}

這個mFrameSize是通過channelCount*取樣精度所佔位元組計算得出的，所以每次通過obtainBuffer獲取共享記憶體中的資料，然後通過memcpy把資料拷貝到應用層的buffer中，直到把整個userSize都拷貝到buffer中為止。

這裡就詳細分析下obtainBuffer函式

status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
        struct timespec *elapsed, size_t *nonContig)
{
    // previous and new IAudioRecord sequence numbers are used to detect track re-creation
    uint32_t oldSequence = 0;
    uint32_t newSequence;

    Proxy::Buffer buffer;
    status_t status = NO_ERROR;

    static const int32_t kMaxTries = 5;
    int32_t tryCounter = kMaxTries;

    do {
        // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
        // keep them from going away if another thread re-creates the track during obtainBuffer()
        sp<AudioRecordClientProxy> proxy;
        sp<IMemory> iMem;
        sp<IMemory> bufferMem;
        {
            // start of lock scope
            AutoMutex lock(mLock);

            newSequence = mSequence;
            // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
            if (status == DEAD_OBJECT) {
                // re-create track, unless someone else has already done so
                if (newSequence == oldSequence) {

                    status = restoreRecord_l("obtainBuffer");
                    if (status != NO_ERROR) {
                        buffer.mFrameCount = 0;
                        buffer.mRaw = NULL;
                        buffer.mNonContig = 0;
                        break;
                    }
                }
            }
            oldSequence = newSequence;

            // Keep the extra references
            proxy = mProxy;
            iMem = mCblkMemory;
            bufferMem = mBufferMemory;

            // Non-blocking if track is stopped
            if (!mActive) {
                requested = &ClientProxy::kNonBlocking;
            }

        }   // end of lock scope

        buffer.mFrameCount = audioBuffer->frameCount;
        // FIXME starts the requested timeout and elapsed over from scratch
        status = proxy->obtainBuffer(&buffer, requested, elapsed);

    } while ((status == DEAD_OBJECT) && (tryCounter-- > 0));

    audioBuffer->frameCount = buffer.mFrameCount;
    audioBuffer->size = buffer.mFrameCount * mFrameSize;
    audioBuffer->raw = buffer.mRaw;

    if (nonContig != NULL) {
        *nonContig = buffer.mNonContig;
    }
    return status;
}

在這個函式中的主要工作如下：

    1.獲取AudioRecordClientProxy代理，mCblkMemory與mBufferMemory，他們是在構建AudioRecord物件的時候，通過AF端獲取的，即RecordThread::RecordTrack->getCblk()與RecordThread::RecordTrack->getBuffers()

    2.在start中，AudioRecordThread.resume之前，mActive已經標記為true了，所以requested還是&ClientProxy::kForever；

    3.呼叫proxy->obtainBuffer繼續獲取資料；

    3.更新audioBuffer的frameCount，size以及pcm資料；

這裡繼續分析第3步：ClientProxy::obtainBuffer

frameworks\av\media\libmedia\AudioTrackShared.cpp

status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *requested,
        struct timespec *elapsed)
{
    LOG_ALWAYS_FATAL_IF(buffer == NULL || buffer->mFrameCount == 0);
    struct timespec total;          // total elapsed time spent waiting
    total.tv_sec = 0;
    total.tv_nsec = 0;
    bool measure = elapsed != NULL; // whether to measure total elapsed time spent waiting

    status_t status;
    enum {
        TIMEOUT_ZERO,       // requested == NULL || *requested == 0
        TIMEOUT_INFINITE,   // *requested == infinity
        TIMEOUT_FINITE,     // 0 < *requested < infinity
        TIMEOUT_CONTINUE,   // additional chances after TIMEOUT_FINITE
    } timeout;
    if (requested == NULL) {
        timeout = TIMEOUT_ZERO;
    } else if (requested->tv_sec == 0 && requested->tv_nsec == 0) {
        timeout = TIMEOUT_ZERO;
    } else if (requested->tv_sec == INT_MAX) {
        timeout = TIMEOUT_INFINITE;
    } else {
        timeout = TIMEOUT_FINITE;
        if (requested->tv_sec > 0 || requested->tv_nsec >= MEASURE_NS) {
            measure = true;
        }
    }
    struct timespec before;
    bool beforeIsValid = false;
    audio_track_cblk_t* cblk = mCblk;
    bool ignoreInitialPendingInterrupt = true;
    // check for shared memory corruption
    if (mIsShutdown) {
        status = NO_INIT;
        goto end;
    }
    for (;;) {
        int32_t flags = android_atomic_and(~CBLK_INTERRUPT, &cblk->mFlags);
        // check for track invalidation by server, or server death detection
        if (flags & CBLK_INVALID) {
            ALOGV("Track invalidated");
            status = DEAD_OBJECT;
            goto end;
        }
        // check for obtainBuffer interrupted by client
        if (!ignoreInitialPendingInterrupt && (flags & CBLK_INTERRUPT)) {
            ALOGV("obtainBuffer() interrupted by client");
            status = -EINTR;
            goto end;
        }
        ignoreInitialPendingInterrupt = false;
        // compute number of frames available to write (AudioTrack) or read (AudioRecord)
        int32_t front;
        int32_t rear;

        if (mIsOut) {
            // The barrier following the read of mFront is probably redundant.
            // We're about to perform a conditional branch based on 'filled',
            // which will force the processor to observe the read of mFront
            // prior to allowing data writes starting at mRaw.
            // However, the processor may support speculative execution,
            // and be unable to undo speculative writes into shared memory.
            // The barrier will prevent such speculative execution.
            front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
            rear = cblk->u.mStreaming.mRear;
        } else {
            // On the other hand, this barrier is required.
            rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
            front = cblk->u.mStreaming.mFront;
        }
        ssize_t filled = rear - front;

        // pipe should not be overfull
        if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
            if (mIsOut) {
                ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); "
                        "shutting down", filled, mFrameCount);
                mIsShutdown = true;
                status = NO_INIT;
                goto end;
            }
            // for input, sync up on overrun
            filled = 0;
            cblk->u.mStreaming.mFront = rear;
            (void) android_atomic_or(CBLK_OVERRUN, &cblk->mFlags);
        }
        // don't allow filling pipe beyond the nominal size
        size_t avail = mIsOut ? mFrameCount - filled : filled;
        if (avail > 0) {
            // 'avail' may be non-contiguous, so return only the first contiguous chunk
            size_t part1;
            if (mIsOut) {
                rear &= mFrameCountP2 - 1;
                part1 = mFrameCountP2 - rear;
            } else {
                front &= mFrameCountP2 - 1;
                part1 = mFrameCountP2 - front;
            }
            if (part1 > avail) {
                part1 = avail;
            }
            if (part1 > buffer->mFrameCount) {
                part1 = buffer->mFrameCount;
            }
            buffer->mFrameCount = part1;
            buffer->mRaw = part1 > 0 ?
                    &((char *) mBuffers)[(mIsOut ? rear : front) * mFrameSize] : NULL;
            buffer->mNonContig = avail - part1;
            mUnreleased = part1;
            status = NO_ERROR;

            break;
        }
        struct timespec remaining;
        const struct timespec *ts;

        switch (timeout) {
        case TIMEOUT_ZERO:
            status = WOULD_BLOCK;
            goto end;
        case TIMEOUT_INFINITE:
            ts = NULL;
            break;
        case TIMEOUT_FINITE:
            timeout = TIMEOUT_CONTINUE;
            if (MAX_SEC == 0) {
                ts = requested;
                break;
            }
            // fall through
        case TIMEOUT_CONTINUE:
            // FIXME we do not retry if requested < 10ms? needs documentation on this state machine
            if (!measure || requested->tv_sec < total.tv_sec ||
                    (requested->tv_sec == total.tv_sec && requested->tv_nsec <= total.tv_nsec)) {
                status = TIMED_OUT;
                goto end;
            }
            remaining.tv_sec = requested->tv_sec - total.tv_sec;
            if ((remaining.tv_nsec = requested->tv_nsec - total.tv_nsec) < 0) {
                remaining.tv_nsec += 1000000000;
                remaining.tv_sec++;
            }
            if (0 < MAX_SEC && MAX_SEC < remaining.tv_sec) {
                remaining.tv_sec = MAX_SEC;
                remaining.tv_nsec = 0;
            }
            ts = &remaining;
            break;
        default:
            LOG_ALWAYS_FATAL("obtainBuffer() timeout=%d", timeout);
            ts = NULL;
            break;
        }

        int32_t old = android_atomic_and(~CBLK_FUTEX_WAKE, &cblk->mFutex);

        if (!(old & CBLK_FUTEX_WAKE)) {
            if (measure && !beforeIsValid) {
                clock_gettime(CLOCK_MONOTONIC, &before);
                beforeIsValid = true;
            }
            errno = 0;

            (void) syscall(__NR_futex, &cblk->mFutex,
                    mClientInServer ? FUTEX_WAIT_PRIVATE : FUTEX_WAIT, old & ~CBLK_FUTEX_WAKE, ts);

            // update total elapsed time spent waiting
            if (measure) {
                struct timespec after;
                clock_gettime(CLOCK_MONOTONIC, &after);
                total.tv_sec += after.tv_sec - before.tv_sec;
                long deltaNs = after.tv_nsec - before.tv_nsec;
                if (deltaNs < 0) {
                    deltaNs += 1000000000;
                    total.tv_sec--;
                }
                if ((total.tv_nsec += deltaNs) >= 1000000000) {
                    total.tv_nsec -= 1000000000;
                    total.tv_sec++;
                }
                before = after;
                beforeIsValid = true;
            }

            switch (errno) {
            case 0:            // normal wakeup by server, or by binderDied()
            case EWOULDBLOCK:  // benign race condition with server
            case EINTR:        // wait was interrupted by signal or other spurious wakeup
            case ETIMEDOUT:    // time-out expired
                // FIXME these error/non-0 status are being dropped
                break;
            default:
                status = errno;
                ALOGE("%s unexpected error %s", __func__, strerror(status));
                goto end;
            }
        }

    }

end:
    if (status != NO_ERROR) {
        buffer->mFrameCount = 0;
        buffer->mRaw = NULL;
        buffer->mNonContig = 0;
        mUnreleased = 0;
    }
    if (elapsed != NULL) {
        *elapsed = total;
    }
    if (requested == NULL) {
        requested = &kNonBlocking;
    }
    if (measure) {
        ALOGV("requested %ld.%03ld elapsed %ld.%03ld",
              requested->tv_sec, requested->tv_nsec / 1000000,
              total.tv_sec, total.tv_nsec / 1000000);
    }
    return status;
}

這個函式的主要工作如下：

    1.從ClientProxy::kForever的定義可知，這個tv_sec是INT_MAX，所以timeout為TIMEOUT_INFINITE；

    2.從cblk中獲取到rear以及front，之前分析過了，這兩個指標是在RecordThread執行緒中維護的，他一邊在讀取pcm資料，一直在更新最新資料指標的位置；

    3.如果發現獲取到的資料小於mFrameCount，或者沒有獲取到資料，那麼也就是表示應用讀的太快了，其實應該說RecordThread讀的太慢了導致，這時候也需要設定為OVERRUN，則更新cblk的指標，重置filled為0；

    4.如果獲取到了資料，則把錄音資料放到mRaw中，把獲取到的資料大小放到mFrameCount中。

 

總結：

    到這裡，整個獲取pcm資料的流程就結束了，到這裡我們應該能理解整個Audio系統中對AudioBuffer的管理策略了，即通過RecordThread執行緒把資料從硬體層讀取到IMemory中，然後應用層在去IMemory中去讀取。

 

由於作者內功有限，若文章中存在錯誤或不足的地方，還請給位大佬指出，不勝感激！