AudioTrack.cpp 源码分析

AudioTrack. java 进行音频播放的相关处理， 它对应的JNI 中间层为android_media_AudioTrack.cpp ，然后到达C层AudioStack.cpp

这部分根据他们之间的对应关系来分析android_media_AudioTrack.cpp  的源码。

一个典型的AudioTracK 播放音频的实例如下：

    //1. 根据音频特性来确定索要分配的缓冲区大小    int bufSize=AudioTrack.getMinBufferSize（8000, //采样率    AudioFormat.CHANNEL_COFIGURATION_STEREO,//声道数    AudioFormat.ENCODING_PCM_16BIT //采样精度，一个采样点16bit,两个字节    );    //2.创建 AudioTrack      AudioTrack track=new AudioTrack(              AudioManager.STREAM_MUSIC, // 音频流类型              8000,AudioFormat.CHANNEL_COFIGURATION_STEREO,              AudioFormat.ENCODING_PCM_16BIT,bufSize,            AudioTrack.MODE_STREAM //数据加载模式, 非静态加载    );    //3.开始播放    track.play();    ......    //4,写入播放数据    track.write(audiodata,0,length);    ......    //5.停止播放， 释放资源    track.stop(); // 停止播放    track.release();// 释放底层资源

一 获得缓冲区最小值

     在AudioTrack.java 中

 static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {        int channelCount = 0;        switch(channelConfig) {        case AudioFormat.CHANNEL_OUT_MONO:        case AudioFormat.CHANNEL_CONFIGURATION_MONO:  // 单声道            channelCount = 1;            break;        case AudioFormat.CHANNEL_OUT_STEREO:        case AudioFormat.CHANNEL_CONFIGURATION_STEREO: // 双声道            channelCount = 2;            break;        default:......        int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);        if (size <= 0) {            loge("getMinBufferSize(): error querying hardware");            return ERROR;        }        else {            return size;        }

channelCount 对应着声道数

在android_media_AudioTrack.cpp 中，

static jint android_media_AudioTrack_get_min_buff_size(JNIEnv *env,  jobject thiz,    jint sampleRateInHertz, jint channelCount, jint audioFormat) {    size_t frameCount;    const status_t status = AudioTrack::getMinFrameCount(&frameCount, AUDIO_STREAM_DEFAULT,            sampleRateInHertz);    if (status != NO_ERROR) {        ALOGE("AudioTrack::getMinFrameCount() for sample rate %d failed with status %d",                sampleRateInHertz, status);        return -1;    }    const audio_format_t format = audioFormatToNative(audioFormat);    if (audio_has_proportional_frames(format)) {        const size_t bytesPerSample = audio_bytes_per_sample(format);        return frameCount * channelCount * bytesPerSample;    } else {        return frameCount;    }}

二  建立AudioTrack. 对象

在AudioTrack中

  public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,            int bufferSizeInBytes, int mode)    throws IllegalArgumentException {        this(streamType, sampleRateInHz, channelConfig, audioFormat,                bufferSizeInBytes, mode, AudioManager.AUDIO_SESSION_ID_GENERATE);    }

    public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,            int bufferSizeInBytes, int mode, int sessionId)    throws IllegalArgumentException {        // mState already == STATE_UNINITIALIZED        this((new AudioAttributes.Builder())                    .setLegacyStreamType(streamType)                    .build(),                (new AudioFormat.Builder())                    .setChannelMask(channelConfig)                    .setEncoding(audioFormat)                    .setSampleRate(sampleRateInHz)                    .build(),                bufferSizeInBytes,                mode, sessionId);    }

   public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,            int mode, int sessionId)                    throws IllegalArgumentException {        super(attributes);        // mState already == STATE_UNINITIALIZED        int channelIndexMask = 0;        if ((format.getPropertySetMask()                & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {            channelIndexMask = format.getChannelIndexMask();        }        int channelMask = 0;        if ((format.getPropertySetMask()                & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {            channelMask = format.getChannelMask();        } else if (channelIndexMask == 0) { // if no masks at all, use stereo            channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT                    | AudioFormat.CHANNEL_OUT_FRONT_RIGHT;        }        int encoding = AudioFormat.ENCODING_DEFAULT;        if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {            encoding = format.getEncoding();        }        audioParamCheck(rate, channelMask, channelIndexMask, encoding, mode);        mStreamType = AudioSystem.STREAM_DEFAULT;        audioBuffSizeCheck(bufferSizeInBytes);        mInitializationLooper = looper;        if (sessionId < 0) {            throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);        }        int[] sampleRate = new int[] {mSampleRate};        int[] session = new int[1];        session[0] = sessionId;        // AudioTrack 在JNI 的初始化        int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes,                sampleRate, mChannelMask, mChannelIndexMask, mAudioFormat,                mNativeBufferSizeInBytes, mDataLoadMode, session, 0 /*nativeTrackInJavaObj*/);        if (initResult != SUCCESS) {            loge("Error code "+initResult+" when initializing AudioTrack.");            return; // with mState == STATE_UNINITIALIZED        }        mSampleRate = sampleRate[0];        mSessionId = session[0];        if (mDataLoadMode == MODE_STATIC) {            mState = STATE_NO_STATIC_DATA;        } else {            mState = STATE_INITIALIZED;        }    }

重点看一下JNI层的初始化

       int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes,
                sampleRate, mChannelMask, mChannelIndexMask, mAudioFormat,
                mNativeBufferSizeInBytes, mDataLoadMode, session, 0 /*nativeTrackInJavaObj*/);

JNI的初始化

// ----------------------------------------------------------------------------int register_android_media_AudioTrack(JNIEnv *env){    // must be first    int res = RegisterMethodsOrDie(env, kClassPathName, gMethods, NELEM(gMethods));    javaAudioTrackFields.nativeTrackInJavaObj = NULL;    javaAudioTrackFields.postNativeEventInJava = NULL;    // Get the AudioTrack class    // audioTrackClass 对应于AudioTrack.java 对象    jclass audioTrackClass = FindClassOrDie(env, kClassPathName);    // Get the postEvent method    // postNativeEventInJava对应于AudioTrack.java 的postEventFromNative 方法    javaAudioTrackFields.postNativeEventInJava = GetStaticMethodIDOrDie(env,            audioTrackClass, JAVA_POSTEVENT_CALLBACK_NAME,            "(Ljava/lang/Object;IIILjava/lang/Object;)V");    // Get the variables fields    //      nativeTrackInJavaObj    // nativeTrackInJavaObj对应于AudioTrack.java 的变量mNativeTrackInJavaObj    javaAudioTrackFields.nativeTrackInJavaObj = GetFieldIDOrDie(env,            audioTrackClass, JAVA_NATIVETRACKINJAVAOBJ_FIELD_NAME, "J");    //      jniData    // nativeTrackInJavaObj对应于AudioTrack.java 的变量mJniData    javaAudioTrackFields.jniData = GetFieldIDOrDie(env,            audioTrackClass, JAVA_JNIDATA_FIELD_NAME, "J");    //      fieldStreamType    // nativeTrackInJavaObj对应于AudioTrack.java 的变量mStreamType    javaAudioTrackFields.fieldStreamType = GetFieldIDOrDie(env,            audioTrackClass, JAVA_STREAMTYPE_FIELD_NAME, "I");    env->DeleteLocalRef(audioTrackClass);    // Get the AudioAttributes class and fields    // audioAttrClass 对应于AudioAttributes.java 对象    jclass audioAttrClass = FindClassOrDie(env, kAudioAttributesClassPathName);    //fieldUsage 对应于AudioAttributes.java  的变量mUsage    javaAudioAttrFields.fieldUsage = GetFieldIDOrDie(env, audioAttrClass, "mUsage", "I");  // 得到class:audioAttrClass 的对应域的ID    //fieldUsage 对应于AudioAttributes.java  的变量mContentType    javaAudioAttrFields.fieldContentType = GetFieldIDOrDie(env,            audioAttrClass, "mContentType", "I");    //fieldUsage 对应于AudioAttributes.java  的变量mFlags    javaAudioAttrFields.fieldFlags = GetFieldIDOrDie(env, audioAttrClass, "mFlags", "I");    //fieldUsage 对应于AudioAttributes.java  的变量mFormattedTags    javaAudioAttrFields.fieldFormattedTags = GetFieldIDOrDie(env,            audioAttrClass, "mFormattedTags", "Ljava/lang/String;");    env->DeleteLocalRef(audioAttrClass);    // initialize PlaybackParams field info    gPlaybackParamsFields.init(env);    return res;}

AudioTrack 初始化。

static jintandroid_media_AudioTrack_setup(JNIEnv *env, jobject thiz, jobject weak_this, jobject jaa,        jintArray jSampleRate, jint channelPositionMask, jint channelIndexMask,        jint audioFormat, jint buffSizeInBytes, jint memoryMode, jintArray jSession,        jlong nativeAudioTrack) {    ALOGV("sampleRates=%p, channel mask=%x, index mask=%x, audioFormat(Java)=%d, buffSize=%d"        "nativeAudioTrack=0x%llX",        jSampleRate, channelPositionMask, channelIndexMask, audioFormat, buffSizeInBytes,        nativeAudioTrack);    sp<AudioTrack> lpTrack = 0;    // 获得指向数组jSession内容的指针    jint* nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);    if (nSession == NULL) {        ALOGE("Error creating AudioTrack: Error retrieving session id pointer");        return (jint) AUDIO_JAVA_ERROR;    }    //获取数组的第一个元素    audio_session_t sessionId = (audio_session_t) nSession[0];    //释放指向数组的指针    env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);    nSession = NULL;    AudioTrackJniStorage* lpJniStorage = NULL;    audio_attributes_t *paa = NULL;    jclass clazz = env->GetObjectClass(thiz);    if (clazz == NULL) {        ALOGE("Can't find %s when setting up callback.", kClassPathName);        return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;    }    // if we pass in an existing *Native* AudioTrack, we don't need to create/initialize one.    //第一次初始化    if (nativeAudioTrack == 0) {        if (jaa == 0) {            ALOGE("Error creating AudioTrack: invalid audio attributes");            return (jint) AUDIO_JAVA_ERROR;        }        if (jSampleRate == 0) {            ALOGE("Error creating AudioTrack: invalid sample rates");            return (jint) AUDIO_JAVA_ERROR;        }        // 获得采样率        int* sampleRates = env->GetIntArrayElements(jSampleRate, NULL);        int sampleRateInHertz = sampleRates[0];        env->ReleaseIntArrayElements(jSampleRate, sampleRates, JNI_ABORT);        // Invalid channel representations are caught by !audio_is_output_channel() below.        audio_channel_mask_t nativeChannelMask = nativeChannelMaskFromJavaChannelMasks(                channelPositionMask, channelIndexMask);        if (!audio_is_output_channel(nativeChannelMask)) {            ALOGE("Error creating AudioTrack: invalid native channel mask %#x.", nativeChannelMask);            return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK;        }        uint32_t channelCount = audio_channel_count_from_out_mask(nativeChannelMask);        // check the format.        // This function was called from Java, so we compare the format against the Java constants        audio_format_t format = audioFormatToNative(audioFormat);        if (format == AUDIO_FORMAT_INVALID) {            ALOGE("Error creating AudioTrack: unsupported audio format %d.", audioFormat);            return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDFORMAT;        }        // compute the frame count        size_t frameCount;        if (audio_is_linear_pcm(format)) {            const size_t bytesPerSample = audio_bytes_per_sample(format);            frameCount = buffSizeInBytes / (channelCount * bytesPerSample);        } else {            frameCount = buffSizeInBytes;        }        // 1.建立AudioTrack        lpTrack = new AudioTrack();        // read the AudioAttributes values        paa = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));        /* 2.        *jaa 对应Java层的AudioAttributes,        *将AudioAttributes 属性赋予audio_attributes_t        */        const jstring jtags =                (jstring) env->GetObjectField(jaa, javaAudioAttrFields.fieldFormattedTags);        const char* tags = env->GetStringUTFChars(jtags, NULL);        // copying array size -1, char array for tags was calloc'd, no need to NULL-terminate it        strncpy(paa->tags, tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);        env->ReleaseStringUTFChars(jtags, tags);        paa->usage = (audio_usage_t) env->GetIntField(jaa, javaAudioAttrFields.fieldUsage);        paa->content_type =                (audio_content_type_t) env->GetIntField(jaa, javaAudioAttrFields.fieldContentType);        paa->flags = env->GetIntField(jaa, javaAudioAttrFields.fieldFlags);        ALOGV("AudioTrack_setup for usage=%d content=%d flags=0x%#x tags=%s",                paa->usage, paa->content_type, paa->flags, paa->tags);        // initialize the callback information:        // this data will be passed with every AudioTrack callback        lpJniStorage = new AudioTrackJniStorage();        lpJniStorage->mCallbackData.audioTrack_class = (jclass)env->NewGlobalRef(clazz);        // we use a weak reference so the AudioTrack object can be garbage collected.        lpJniStorage->mCallbackData.audioTrack_ref = env->NewGlobalRef(weak_this);        lpJniStorage->mCallbackData.busy = false;        // initialize the native AudioTrack object        //3. 根据memoryMode来进行AudioTrack 的配置        status_t status = NO_ERROR;        switch (memoryMode) {        case MODE_STREAM:            status = lpTrack->set(                    AUDIO_STREAM_DEFAULT,// stream type, but more info conveyed in paa (last argument)                    sampleRateInHertz,                    format,// word length, PCM                    nativeChannelMask,                    frameCount,                    AUDIO_OUTPUT_FLAG_NONE,                    audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user)                    0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack                    0,// shared mem                    true,// thread can call Java                    sessionId,// audio session ID                    AudioTrack::TRANSFER_SYNC,                    NULL,                         // default offloadInfo                    -1, -1,                       // default uid, pid values                    paa);            break;        case MODE_STATIC:            // AudioTrack is using shared memory            if (!lpJniStorage->allocSharedMem(buffSizeInBytes)) {                ALOGE("Error creating AudioTrack in static mode: error creating mem heap base");                goto native_init_failure;            }            status = lpTrack->set(                    AUDIO_STREAM_DEFAULT,// stream type, but more info conveyed in paa (last argument)                    sampleRateInHertz,                    format,// word length, PCM                    nativeChannelMask,                    frameCount,                    AUDIO_OUTPUT_FLAG_NONE,                    audioCallback, &(lpJniStorage->mCallbackData),//callback, callback data (user));                    0,// notificationFrames == 0 since not using EVENT_MORE_DATA to feed the AudioTrack                    lpJniStorage->mMemBase,// shared mem                    true,// thread can call Java                    sessionId,// audio session ID                    AudioTrack::TRANSFER_SHARED,                    NULL,                         // default offloadInfo                    -1, -1,                       // default uid, pid values                    paa);            break;        default:            ALOGE("Unknown mode %d", memoryMode);            goto native_init_failure;        }        if (status != NO_ERROR) {            ALOGE("Error %d initializing AudioTrack", status);            goto native_init_failure;        }    } else {  // end if (nativeAudioTrack == 0)        lpTrack = (AudioTrack*)nativeAudioTrack;        // TODO: We need to find out which members of the Java AudioTrack might

这里主要说明一下

1. 初始化AudioTrack

AudioTrack::AudioTrack()    : mStatus(NO_INIT),      mState(STATE_STOPPED),      mPreviousPriority(ANDROID_PRIORITY_NORMAL),      mPreviousSchedulingGroup(SP_DEFAULT),      mPausedPosition(0),      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE){    mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;    mAttributes.usage = AUDIO_USAGE_UNKNOWN;    mAttributes.flags = 0x0;    strcpy(mAttributes.tags, "");}

这里初始化了AudioStrack 的状态，暂停位置为0，和audio_attributes_t，audio_attributes_t 对应于java层的AudioAttributes。

typedef struct {    audio_content_type_t content_type;    audio_usage_t        usage;    audio_source_t       source;    audio_flags_mask_t   flags;    char                 tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE]; /* UTF8 */} audio_attributes_t;对应java 层的AudioAttributes

2. 下面要将java层的AudioAttributes 映射到C层。audio_attributes_fields_t 保存了AudioAttributes 的域名在C层的对应ID(在

register_android_media_AudioTrack方法中初始化）

struct audio_attributes_fields_t {    jfieldID  fieldUsage;        // AudioAttributes.mUsage    jfieldID  fieldContentType;  // AudioAttributes.mContentType    jfieldID  fieldFlags;        // AudioAttributes.mFlags    jfieldID  fieldFormattedTags;// AudioAttributes.mFormattedTags};

static audio_attributes_fields_t javaAudioAttrFields

AudioTrackJniStorage 是实现一种共享内存机制， 当静态播放时， 用共享内存保存数据。

动态播放时不用。

3. 配置AudioTrack

  配置AudipoTrack的函数定义为

            status_t    set(audio_stream_type_t streamType,                            uint32_t sampleRate,                            audio_format_t format,                            audio_channel_mask_t channelMask,                            size_t frameCount   = 0,                            audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,                            callback_t cbf      = NULL,                            void* user          = NULL,                            int32_t notificationFrames = 0,                            const sp<IMemory>& sharedBuffer = 0,                            bool threadCanCallJava = false,                            audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,                            transfer_type transferType = TRANSFER_DEFAULT,                            const audio_offload_info_t *offloadInfo = NULL,                            int uid = -1,                            pid_t pid = -1,                            const audio_attributes_t* pAttributes = NULL,                            bool doNotReconnect = false,                            float maxRequiredSpeed = 1.0f);

status_t AudioTrack::set 重要的是调用了createTrack_l 函数， 得到了指针sp<IAudioTrack> mAudioTrack。

然后生成mProxy 对象管理共享内存（AudioTrack 写入数据， AudioFlinger 读取数据）

// must be called with mLock heldstatus_t AudioTrack::createTrack_l(){    /*    *得到AudioFlinger 的Binder 代理端BpAudioFliger    */    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();    /*    * audio_io_handle是一个int 类型，通过typedef 定义，这个值的来历非常复杂，    * 涉及AudioFlinger 和AudioPolicyService，这个值主要被AudioFliger 使用，用    * 来标识内部的工作线程索引号. AudioFlinger 会根据情况创建几个工作线程，    * 下面的AudioSystem::getOutputforAttr 根据流类型等其他参数最终选择一个合适的    * 工作线程，并返回它在AF 中的索引号。    */    audio_io_handle_t output;    audio_stream_type_t streamType = mStreamType;    audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL;......    // Client can only express a preference for FAST.  Server will perform additional tests.    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {        bool useCaseAllowed =            // either of these use cases:            // use case 1: shared buffer            (mSharedBuffer != 0) ||            // use case 2: callback transfer mode            (mTransfer == TRANSFER_CALLBACK) ||            // use case 3: obtain/release mode            (mTransfer == TRANSFER_OBTAIN) ||            // use case 4: synchronous write            ((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava);        // sample rates must also match        bool fastAllowed = useCaseAllowed && (mSampleRate == mAfSampleRate);        if (!fastAllowed) {            ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, "                "track %u Hz, output %u Hz",                mTransfer, mSampleRate, mAfSampleRate);            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);        }    }    mNotificationFramesAct = mNotificationFramesReq;    size_t frameCount = mReqFrameCount;    if (!audio_has_proportional_frames(mFormat)) {        if (mSharedBuffer != 0) {            // Same comment as below about ignoring frameCount parameter for set()            frameCount = mSharedBuffer->size();        } else if (frameCount == 0) {            frameCount = mAfFrameCount;        }        if (mNotificationFramesAct != frameCount) {            mNotificationFramesAct = frameCount;        }    } else if (mSharedBuffer != 0) {        // FIXME: Ensure client side memory buffers need        // not have additional alignment beyond sample        // (e.g. 16 bit stereo accessed as 32 bit frame).        size_t alignment = audio_bytes_per_sample(mFormat);        if (alignment & 1) {            // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).            alignment = 1;        }        if (mChannelCount > 1) {            // More than 2 channels does not require stronger alignment than stereo            alignment <<= 1;        }        if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {            ALOGE("Invalid buffer alignment: address %p, channel count %u",                    mSharedBuffer->pointer(), mChannelCount);            status = BAD_VALUE;            goto release;        }        // When initializing a shared buffer AudioTrack via constructors,        // there's no frameCount parameter.        // But when initializing a shared buffer AudioTrack via set(),        // there _is_ a frameCount parameter.  We silently ignore it.        frameCount = mSharedBuffer->size() / mFrameSize;    } else {        size_t minFrameCount = 0;        // For fast tracks the frame count calculations and checks are mostly done by server,        // but we try to respect the application's request for notifications per buffer.        if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {            if (mNotificationsPerBufferReq > 0) {                // Avoid possible arithmetic overflow during multiplication.                // mNotificationsPerBuffer is clamped to a small integer earlier, so it is unlikely.                if (mNotificationsPerBufferReq > SIZE_MAX / afFrameCountHAL) {                    ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu",                            mNotificationsPerBufferReq, afFrameCountHAL);                } else {                    minFrameCount = afFrameCountHAL * mNotificationsPerBufferReq;                }            }        } else {            // for normal tracks precompute the frame count based on speed.            const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :                            max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);            minFrameCount = calculateMinFrameCount(                    mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,                    speed /*, 0 mNotificationsPerBufferReq*/);        }        if (frameCount < minFrameCount) {            frameCount = minFrameCount;        }    }    IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;    if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {        trackFlags |= IAudioFlinger::TRACK_OFFLOAD;    }    if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {        trackFlags |= IAudioFlinger::TRACK_DIRECT;    }    size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,                                // but we will still need the original value also    audio_session_t originalSessionId = mSessionId;    /*1.    * 向AF发送CreatTrack请求，该函数返回IAudioTrack(实际类型是BpAudioTrack)对象，后续    * AF与AT 的交互通过track 进行    * 在介绍AudioFlinger.cpp 时在介绍    */    sp<IAudioTrack> track = audioFlinger->createTrack(streamType,                                                      mSampleRate,                                                      mFormat,                                                      mChannelMask,                                                      &temp,                                                      &trackFlags,                                                      mSharedBuffer,                                                      output,                                                      mClientPid,                                                      tid,                                                      &mSessionId,                                                      mClientUid,                                                      &status);    ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,            "session ID changed from %d to %d", originalSessionId, mSessionId);    if (status != NO_ERROR) {        ALOGE("AudioFlinger could not create track, status: %d", status);        goto release;    }    ALOG_ASSERT(track != 0);    // AudioFlinger now owns the reference to the I/O handle,    // so we are no longer responsible for releasing it.    // FIXME compare to AudioRecord    // 获得AT 端的共享内存    sp<IMemory> iMem = track->getCblk();    if (iMem == 0) {        ALOGE("Could not get control block");        return NO_INIT;    }    // 获得共享内存的首地址    void *iMemPointer = iMem->pointer();    if (iMemPointer == NULL) {        ALOGE("Could not get control block pointer");        return NO_INIT;    }    // invariant that mAudioTrack != 0 is true only after set() returns successfully    //清理原来的mAudioTrack    if (mAudioTrack != 0) {        IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);        mDeathNotifier.clear();    }    mAudioTrack = track;    mCblkMemory = iMem;    IPCThreadState::self()->flushCommands();    /*    *将类型void* 类型转成audio_track_cdlk_t ,表明这块内存的首部中存在audio_track_cdlk_t这个对象。    */    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);    mCblk = cblk;    // note that temp is the (possibly revised) value of frameCount    if (temp < frameCount || (frameCount == 0 && temp == 0)) {        // In current design, AudioTrack client checks and ensures frame count validity before        // passing it to AudioFlinger so AudioFlinger should not return a different value except        // for fast track as it uses a special method of assigning frame count.        ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);    }    frameCount = temp;    mAwaitBoost = false;    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {        if (trackFlags & IAudioFlinger::TRACK_FAST) {            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount);            if (!mThreadCanCallJava) {                mAwaitBoost = true;            }        } else {            ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);        }    }    // Make sure that application is notified with sufficient margin before underrun.    // The client can divide the AudioTrack buffer into sub-buffers,    // and expresses its desire to server as the notification frame count.    if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {        size_t maxNotificationFrames;        if (trackFlags & IAudioFlinger::TRACK_FAST) {            // notify every HAL buffer, regardless of the size of the track buffer            maxNotificationFrames = afFrameCountHAL;        } else {            // For normal tracks, use at least double-buffering if no sample rate conversion,            // or at least triple-buffering if there is sample rate conversion            const int nBuffering = mOriginalSampleRate == mAfSampleRate ? 2 : 3;            maxNotificationFrames = frameCount / nBuffering;        }        if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) {            if (mNotificationFramesAct == 0) {                ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu",                    maxNotificationFrames, frameCount);            } else {                ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu",                    mNotificationFramesAct, maxNotificationFrames, frameCount);            }            mNotificationFramesAct = (uint32_t) maxNotificationFrames;        }    }    // We retain a copy of the I/O handle, but don't own the reference    mOutput = output;    mRefreshRemaining = true;    // Starting address of buffers in shared memory.  If there is a shared buffer, buffers    // is the value of pointer() for the shared buffer, otherwise buffers points    // immediately after the control block.  This address is for the mapping within client    // address space.  AudioFlinger::TrackBase::mBuffer is for the server address space.    void* buffers;    if (mSharedBuffer == 0) {        buffers = cblk + 1;    } else {        buffers = mSharedBuffer->pointer();        if (buffers == NULL) {            ALOGE("Could not get buffer pointer");            return NO_INIT;        }    }    mAudioTrack->attachAuxEffect(mAuxEffectId);    // FIXME doesn't take into account speed or future sample rate changes (until restoreTrack)    // FIXME don't believe this lie    mLatency = mAfLatency + (1000*frameCount) / mSampleRate;    mFrameCount = frameCount;    // If IAudioTrack is re-created, don't let the requested frameCount    // decrease.  This can confuse clients that cache frameCount().    if (frameCount > mReqFrameCount) {        mReqFrameCount = frameCount;    }    // reset server position to 0 as we have new cblk.    mServer = 0;    // update proxy    //2.生成共享内存的客户端， 
    // 动态播放 用AudioTrackClientProxy, 用来管理共享内存，AudioTrack 向其写数据， AF获得数据播放    if (mSharedBuffer == 0) {        mStaticProxy.clear();        mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);    } else {    //model_Stream 用StaticAudioTrackClientProxy        mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);        mProxy = mStaticProxy;    }    mProxy->setVolumeLR(gain_minifloat_pack(            gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),            gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));    mProxy->setSendLevel(mSendLevel);    const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);    const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);    const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);    mProxy->setSampleRate(effectiveSampleRate);    AudioPlaybackRate playbackRateTemp = mPlaybackRate;    playbackRateTemp.mSpeed = effectiveSpeed;    playbackRateTemp.mPitch = effectivePitch;    mProxy->setPlaybackRate(playbackRateTemp);    mProxy->setMinimum(mNotificationFramesAct);    mDeathNotifier = new DeathNotifier(this);    IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);    if (mDeviceCallback != 0) {        AudioSystem::addAudioDeviceCallback(mDeviceCallback, mOutput);    }    return NO_ERROR;    }release:    AudioSystem::releaseOutput(output, streamType, mSessionId);    if (status == NO_ERROR) {        status = NO_INIT;    }    return status;}

但是这里要介绍一个函数参数callback_t。

call_back_t 在AudioTrack.cpp 中是一个函数指针,

typedef void (*callback_t)(int event, void* user, void *info);

给此指针赋值为：audioCallback,

static void audioCallback(int event, void* user, void *info) {    audiotrack_callback_cookie *callbackInfo = (audiotrack_callback_cookie *)user;    {        Mutex::Autolock l(sLock);        if (sAudioTrackCallBackCookies.indexOf(callbackInfo) < 0) {            return;        }        callbackInfo->busy = true;    }    switch (event) {    case AudioTrack::EVENT_MARKER: {        JNIEnv *env = AndroidRuntime::getJNIEnv();        if (user != NULL && env != NULL) {            env->CallStaticVoidMethod(                callbackInfo->audioTrack_class,                javaAudioTrackFields.postNativeEventInJava,                callbackInfo->audioTrack_ref, event, 0,0, NULL);            if (env->ExceptionCheck()) {                env->ExceptionDescribe();                env->ExceptionClear();            }        }        } break;    case AudioTrack::EVENT_NEW_POS: {        JNIEnv *env = AndroidRuntime::getJNIEnv();        if (user != NULL && env != NULL) {            env->CallStaticVoidMethod(                callbackInfo->audioTrack_class,                javaAudioTrackFields.postNativeEventInJava,                callbackInfo->audioTrack_ref, event, 0,0, NULL);            if (env->ExceptionCheck()) {                env->ExceptionDescribe();                env->ExceptionClear();            }        }        } break;    }    {        Mutex::Autolock l(sLock);        callbackInfo->busy = false;        callbackInfo->cond.broadcast();    }}

此函数的核心就是env->CallStaticVoidMethod函数， 这是一种JNI 调用java 层静态方法的方式。

函数原型 void CallStaticVoidMethod(jclass clazz, jmethodID methodID, ...)    C++ 格式

            void        (*CallStaticVoidMethod)(JNIEnv*, jclass, jmethodID, ...)     C格式

这里用的是C++ 格式，一会我们会看到

callbackInfo->audioTrack_class 对应于class AudioTrack.java

    javaAudioTrackFields.postNativeEventInJava = GetStaticMethodIDOrDie(env,            audioTrackClass, JAVA_POSTEVENT_CALLBACK_NAME,            "(Ljava/lang/Object;IIILjava/lang/Object;)V")

#define JAVA_POSTEVENT_CALLBACK_NAME "postEventFromNative"

从以上代码 看到javaAudioRecordFields.postNativeEventInJava 对应法方法ID为

audioTrack.postEventFromNative(Object audiotrack_ref,int what, int arg1, int arg2, Object obj) 方法的ID.

所以env->CallStaticVoidMethod会将参数提交AudioStrack.java的postEventFromNative 方法处理。

那么这个回调在AudioTrack.cpp 中是如何触发的呢？

    if (cbf != NULL) {        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);        mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);        // thread begins in paused state, and will not reference us until start()    }

AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)    : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),      mIgnoreNextPausedInt(false){}AudioTrack::AudioTrackThread::~AudioTrackThread(){}bool AudioTrack::AudioTrackThread::threadLoop(){    {        AutoMutex _l(mMyLock);        if (mPaused) {            mMyCond.wait(mMyLock);            // caller will check for exitPending()            return true;        }        if (mIgnoreNextPausedInt) {            mIgnoreNextPausedInt = false;            mPausedInt = false;        }        if (mPausedInt) {            if (mPausedNs > 0) {                (void) mMyCond.waitRelative(mMyLock, mPausedNs);            } else {                mMyCond.wait(mMyLock);            }            mPausedInt = false;            return true;        }    }    if (exitPending()) {        return false;    }    nsecs_t ns = mReceiver.processAudioBuffer();    switch (ns) {    case 0:        return true;    case NS_INACTIVE:        pauseInternal();        return true;    case NS_NEVER:        return false;    case NS_WHENEVER:        // Event driven: call wake() when callback notifications conditions change.        ns = INT64_MAX;        // fall through    default:        LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);        pauseInternal(ns);        return true;    }

当线程调用mReceiver.processAudioBuffer()方法时会触发回调。

但是processAudioBuffer 是干什么用的，

三 play

    AudioTrack.play对应于JNI 的native_start() 方法，调用AT的start方法。

status_t AudioTrack::start(){    AutoMutex lock(mLock);    if (mState == STATE_ACTIVE) {        return INVALID_OPERATION;    }    mInUnderrun = true;   //更新播放状态    State previousState = mState;    if (previousState == STATE_PAUSED_STOPPING) {        mState = STATE_STOPPING;    } else {        mState = STATE_ACTIVE;    }    // 更新服务器播放的帧（frames):Modulo<uint32_t>  mServer    //mProxy->getPosition()获得    (void) updateAndGetPosition_l();    if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {        // reset current position as seen by client to 0        mPosition = 0;        mPreviousTimestampValid = false;        mTimestampStartupGlitchReported = false;        mRetrogradeMotionReported = false;        mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;        // read last server side position change via timestamp.        ExtendedTimestamp ets;        if (mProxy->getTimestamp(&ets) == OK &&                ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {            // Server side has consumed something, but is it finished consuming?            // It is possible since flush and stop are asynchronous that the server            // is still active at this point.            ALOGV("start: server read:%lld  cumulative flushed:%lld  client written:%lld",                    (long long)(mFramesWrittenServerOffset                            + ets.mPosition[ExtendedTimestamp::LOCATION_SERVER]),                    (long long)ets.mFlushed,                    (long long)mFramesWritten);            mFramesWrittenServerOffset = -ets.mPosition[ExtendedTimestamp::LOCATION_SERVER];        }        mFramesWritten = 0;        mProxy->clearTimestamp(); // need new server push for valid timestamp        mMarkerReached = false;        // For offloaded tracks, we don't know if the hardware counters are really zero here,        // since the flush is asynchronous and stop may not fully drain.        // We save the time when the track is started to later verify whether        // the counters are realistic (i.e. start from zero after this time).        mStartUs = getNowUs();        // force refresh of remaining frames by processAudioBuffer() as last        // write before stop could be partial.        mRefreshRemaining = true;    }    mNewPosition = mPosition + mUpdatePeriod;    int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);    status_t status = NO_ERROR;    if (!(flags & CBLK_INVALID)) {        // 调用playBackThread:Thrack.start()        status = mAudioTrack->start();        if (status == DEAD_OBJECT) {            flags |= CBLK_INVALID;        }    }    if (flags & CBLK_INVALID) {        status = restoreTrack_l("start");    }    // resume or pause the callback thread as needed.    sp<AudioTrackThread> t = mAudioTrackThread;    if (status == NO_ERROR) {        if (t != 0) {            if (previousState == STATE_STOPPING) {                mProxy->interrupt();            } else {                t->resume();            }        } else {            mPreviousPriority = getpriority(PRIO_PROCESS, 0);            get_sched_policy(0, &mPreviousSchedulingGroup);            androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);        }    } else {        ALOGE("start() status %d", status);        mState = previousState;        if (t != 0) {            if (previousState != STATE_STOPPING) {                t->pause();            }        } else {            setpriority(PRIO_PROCESS, 0, mPreviousPriority);            set_sched_policy(0, mPreviousSchedulingGroup);        }    }    return status;}

四  write

    android_media_AudioTrack.cpp 的android_media_AudioTrack_write_native_bytes 函数调用了writeToTrack函数。

template <typename T>static jint writeToTrack(const sp<AudioTrack>& track, jint audioFormat, const T *data,                         jint offsetInSamples, jint sizeInSamples, bool blocking) {    // give the data to the native AudioTrack object (the data starts at the offset)    ssize_t written = 0;    // regular write() or copy the data to the AudioTrack's shared memory?    size_t sizeInBytes = sizeInSamples * sizeof(T);    //MODE_STREAM, AudioTrack.cpp 不存在shareBuffer    if (track->sharedBuffer() == 0) {        written = track->write(data + offsetInSamples, sizeInBytes, blocking);        // for compatibility with earlier behavior of write(), return 0 in this case        if (written == (ssize_t) WOULD_BLOCK) {            written = 0;        }    } else {        // writing to shared memory, check for capacity        // ,MODE_STATIC, AudioTrack.cpp 存在shareBuffer, 则直接用memcpy 函数拷贝即可。        if ((size_t)sizeInBytes > track->sharedBuffer()->size()) {            sizeInBytes = track->sharedBuffer()->size();        }        memcpy(track->sharedBuffer()->pointer(), data + offsetInSamples, sizeInBytes);        written = sizeInBytes;    }    if (written >= 0) {        return written / sizeof(T);    }    return interpretWriteSizeError(written);}

下面值讨论一下MODE_STREAM格式的写入。

AudioTrack 与AudioFlinger 公用一块内存，这里写入数据就是AudioTrack.cpp 向内存中写入数据的过程。

这块内存的写入和读出 由AudioTrackShared.cpp 控制。AudioTrack的AudioTrackClientProxy或者StaticAudioTrackServerProxy 对象

就是在AudioTrackShared.cpp 中实现。

ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking){    if (mTransfer != TRANSFER_SYNC) {        return INVALID_OPERATION;    }    if (isDirect()) {        AutoMutex lock(mLock);        int32_t flags = android_atomic_and(                            ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),                            &mCblk->mFlags);        if (flags & CBLK_INVALID) {            return DEAD_OBJECT;        }    }    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("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize);        return BAD_VALUE;    }    size_t written = 0;    Buffer audioBuffer;    while (userSize >= mFrameSize) {        audioBuffer.frameCount = userSize / mFrameSize;        /* 申请共享内存， 用sp<AudioTrackClientProxy> mProxy  申请        *在AudioTrackShared.cpp 中实现        */        status_t err = obtainBuffer(&audioBuffer,                blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);        if (err < 0) {            if (written > 0) {                break;            }            return ssize_t(err);        }        size_t toWrite = audioBuffer.size;        // 将数据拷贝进共享内存        memcpy(audioBuffer.i8, buffer, toWrite);        buffer = ((const char *) buffer) + toWrite;        userSize -= toWrite;        written += toWrite;        // 更新AudioTrackShared.cpp 中的读写位置        releaseBuffer(&audioBuffer);    }    if (written > 0) {        mFramesWritten += written / mFrameSize;    }    return written;}

5 stop

 

void AudioTrack::stop(){    AutoMutex lock(mLock);    if (isOffloaded_l()) {        mState = STATE_STOPPING;    } else {        mState = STATE_STOPPED;        mReleased = 0;    }    //共享内存客户端处理    mProxy->interrupt();    //Track的处理    mAudioTrack->stop();    // Note: legacy handling - stop does not clear playback marker    // and periodic update counter, but flush does for streaming tracks.    if (mSharedBuffer != 0) {        // clear buffer position and loop count.        mStaticProxy->setBufferPositionAndLoop(0 /* position */,                0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);    }    // PlayBack    sp<AudioTrackThread> t = mAudioTrackThread;    if (t != 0) {        if (!isOffloaded_l()) {            t->pause();        }    } else {        setpriority(PRIO_PROCESS, 0, mPreviousPriority);        set_sched_policy(0, mPreviousSchedulingGroup);    }}

6. release 方法

    public void release() {        // even though native_release() stops the native AudioTrack, we need to stop        // AudioTrack subclasses too.        try {            stop();        } catch(IllegalStateException ise) {            // don't raise an exception, we're releasing the resources.        }        baseRelease();        native_release();        mState = STATE_UNINITIALIZED;    }

#define CALLBACK_COND_WAIT_TIMEOUT_MS 1000static void android_media_AudioTrack_release(JNIEnv *env,  jobject thiz) {    sp<AudioTrack> lpTrack = setAudioTrack(env, thiz, 0);    if (lpTrack == NULL) {        return;    }    //ALOGV("deleting lpTrack: %x\n", (int)lpTrack);    // delete the JNI data    AudioTrackJniStorage* pJniStorage = (AudioTrackJniStorage *)env->GetLongField(        thiz, javaAudioTrackFields.jniData);    // reset the native resources in the Java object so any attempt to access    // them after a call to release fails.    env->SetLongField(thiz, javaAudioTrackFields.jniData, 0);    if (pJniStorage) {        Mutex::Autolock l(sLock);        audiotrack_callback_cookie *lpCookie = &pJniStorage->mCallbackData;        //ALOGV("deleting pJniStorage: %x\n", (int)pJniStorage);        while (lpCookie->busy) {            if (lpCookie->cond.waitRelative(sLock,                                            milliseconds(CALLBACK_COND_WAIT_TIMEOUT_MS)) !=                                                    NO_ERROR) {                break;            }        }        sAudioTrackCallBackCookies.remove(lpCookie);        // delete global refs created in native_setup        env->DeleteGlobalRef(lpCookie->audioTrack_class);        env->DeleteGlobalRef(lpCookie->audioTrack_ref);        delete pJniStorage;    }}