Audio笔记之AudioTrack

        播放声音可以用MediaPlayer和AudioTrack，两者都提供了java API供应用开发者使用。虽然都可以播放声音，但两者还是有很大的区别的。其中最大的区别是MediaPlayer可以播放多种格式的声音文件，例如MP3，AAC，WAV，OGG，MIDI等。MediaPlayer会在framework层创建对应的音频解码器。而AudioTrack只能播放已经解码的PCM流，如果是文件的话只支持wav格式的音频文件，因为wav格式的音频文件大部分都是PCM流。AudioTrack不创建解码器，所以只能播放不需要解码的wav文件。当然两者之间还是有紧密的联系，MediaPlayer在framework层还是会创建AudioTrack，把解码后的PCM数流传递给AudioTrack，AudioTrack再传递给AudioFlinger进行混音，然后才传递给硬件播放,所以是MediaPlayer包含了AudioTrack。使用AudioTrack播放音乐示例：

/*cts/tests/tests/media/src/android/media/cts*/

    public voidtestSetStereoVolumeMax()  throwsException {

        final String TEST_NAME= "testSetStereoVolumeMax";

        final int TEST_SR =22050;

        final int TEST_CONF =AudioFormat.CHANNEL_CONFIGURATION_STEREO;

        final int TEST_FORMAT= AudioFormat.ENCODING_PCM_16BIT;

        final int TEST_MODE =AudioTrack.MODE_STREAM;

        final intTEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;


        // --------initialization --------------

        /*Step1.*/
       //根据Hal buff size计算需要为AudioTrack分配的buff size,
       //后续会将这些buff分配给track的mblck，用来进行数据传输
        int  minBuffSize = AudioTrack.getMinBufferSize(TEST_SR, TEST_CONF, TEST_FORMAT);

        /*Step 2.*/
        AudioTrack  track = newAudioTrack(TEST_STREAM_TYPE, TEST_SR, TEST_CONF,
                                                                         TEST_FORMAT, 2 * minBuffSize,TEST_MODE);

        byte data[] = newbyte[minBuffSize];

        // -------- test--------------

        track.write(data, OFFSET_DEFAULT, data.length);

        track.write(data, OFFSET_DEFAULT, data.length);

        track.play();

        float maxVol =AudioTrack.getMaxVolume();

        assertTrue(TEST_NAME, track.setStereoVolume(maxVol, maxVol) == AudioTrack.SUCCESS);

        // -------- tear down--------------

        track.release();

    }

  //默认sessionId设置为0 
   public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
            int bufferSizeInBytes, int mode)
    throws IllegalArgumentException {
        this(streamType, sampleRateInHz, channelConfig, audioFormat,
                bufferSizeInBytes, mode, 0 /*session*/);
    }

public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
            int bufferSizeInBytes, int mode, int sessionId)
    throws IllegalArgumentException {
        // mState already == STATE_UNINITIALIZED

        // remember which looper is associated with the AudioTrack instantiation
        Looper looper;
        if ((looper = Looper.myLooper()) == null) {
            looper = Looper.getMainLooper();
        }
       //保留创建该AudioTrack的loop，后续可以进行message传递
        mInitializationLooper = looper;
/** 
     * 参数检查 
     * 1.检查streamType是合法，并赋值给mStreamType 
     * 2.检查sampleRateInHz是否在4000到48000之间，并赋值给mSampleRate 
     * 3.设置mChannels，将声道转换为 CHANNEL_OUT_MONO(单声道)
            或CHANNEL_OUT_STEREO(双声道)
     * 4.设置mAudioFormat：  
     *      ENCODING_PCM_16BIT、ENCODING_DEFAULT ---> ENCODING_PCM_16BIT 
     *      ENCODING_PCM_8BIT ---> ENCODING_PCM_8BIT 
     * 5.设置mDataLoadMode： MODE_STREAM 或MODE_STATIC 
     */  
        audioParamCheck(streamType, sampleRateInHz, channelConfig, audioFormat, mode);

    / *根据采样精度计算每帧字节大小和缓冲区帧数=缓冲区/每帧大小 ，
      *且缓冲区帧数必须是每帧大小的整数倍
     */  
        audioBuffSizeCheck(bufferSizeInBytes);

        if (sessionId < 0) {
            throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
        }

        int[] session = new int[1];
        session[0] = sessionId;
        // native initialization
        int initResult = native_setup(new WeakReference<AudioTrack>(this),
                mStreamType, mSampleRate, mChannels, mAudioFormat,
                mNativeBufferSizeInBytes, mDataLoadMode, session);
        if (initResult != SUCCESS) {
            loge("Error code "+initResult+" when initializing AudioTrack.");
            return; // with mState == STATE_UNINITIALIZED
        }
       //更新为native层返回的sessionId
        mSessionId = session[0];

        if (mDataLoadMode == MODE_STATIC) {
            mState = STATE_NO_STATIC_DATA;
        } else {
            mState = STATE_INITIALIZED;
        }
    }

static jint
android_media_AudioTrack_native_setup(JNIEnv *env, jobject thiz, jobject weak_this,
        jint streamType, jint sampleRateInHertz, jint javaChannelMask,
        jint audioFormat, jint buffSizeInBytes, jint memoryMode, jintArray jSession)
{
    ALOGV("sampleRate=%d, audioFormat(from Java)=%d, channel mask=%x, buffSize=%d",
        sampleRateInHertz, audioFormat, javaChannelMask, buffSizeInBytes);
    uint32_t afSampleRate;
    size_t afFrameCount;
    //通过AudioSystem从AudioPolicyService中读取对应音频流类型的帧数
    //需要强调的是，getOutputxxxx函数其实很复杂，以getOutputFrameCount为例
    //具体过程如下：
    //1、通过streamType找到strategy，
    //2、然后通过strategy找到device，
    //3、最后通过device找到output
    //4、返回output的FrameCount(硬件缓冲区包含的帧数)
    if (AudioSystem::getOutputFrameCount(&afFrameCount, (audio_stream_type_t) streamType) != NO_ERROR) {
        ALOGE("Error creating AudioTrack: Could not get AudioSystem frame count.");
        return (jint) AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM;
    }
    if (AudioSystem::getOutputSamplingRate(&afSampleRate, (audio_stream_type_t) streamType) != NO_ERROR) {
        ALOGE("Error creating AudioTrack: Could not get AudioSystem sampling rate.");
        return (jint) AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM;
    }

    // Java channel masks don't map directly to the native definition, but it's a simple shift
    // to skip the two deprecated channel configurations "default" and "mono".
    // native层没有default和mono类型的channel定义，需要转化
    uint32_t nativeChannelMask = ((uint32_t)javaChannelMask) >> 2;

    if (!audio_is_output_channel(nativeChannelMask)) {
        ALOGE("Error creating AudioTrack: invalid channel mask %#x.", javaChannelMask);
        return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK;
    }
    //计算bit 1的数目，每个bit代表一个声道
    int nbChannels = popcount(nativeChannelMask);

    // check the stream type
    audio_stream_type_t atStreamType;
    switch (streamType) {
    case AUDIO_STREAM_VOICE_CALL:
    case AUDIO_STREAM_SYSTEM:
    case AUDIO_STREAM_RING:
    case AUDIO_STREAM_MUSIC:
    case AUDIO_STREAM_ALARM:
    case AUDIO_STREAM_NOTIFICATION:
    case AUDIO_STREAM_BLUETOOTH_SCO:
    case AUDIO_STREAM_DTMF:
        atStreamType = (audio_stream_type_t) streamType;
        break;
    default:
        ALOGE("Error creating AudioTrack: unknown stream type.");
        return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDSTREAMTYPE;
    }

    // check the format.
    // This function was called from Java, so we compare the format against the Java constants
    if ((audioFormat != ENCODING_PCM_16BIT) && (audioFormat != ENCODING_PCM_8BIT)) {
        ALOGE("Error creating AudioTrack: unsupported audio format.");
        return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDFORMAT;
    }

    // for the moment 8bitPCM in MODE_STATIC is not supported natively in the AudioTrack C++ class
    // so we declare everything as 16bitPCM, the 8->16bit conversion for MODE_STATIC will be handled
    // in android_media_AudioTrack_native_write_byte()
    // 扩展精度8BIT->16BIT和缓冲区*2，通过将精度设置为16BIT
    //因为在static模式下，底层只支持16BIT，
    //后续写数据的过程中也要做对应的处理