Android Audio代码分析1 - AudioTrack使用示例

计划从接口的使用，开始分析Audio相关源码。
此处的代码为Android中自带的测试代码。
由于本人惰性，不打算将所有函数全部细说。主要函数，会拿来细细品味；本人认为非主要的函数，将一笔带过。
主要非主要，是从本人当前项目的需要来看的。


*****************************************源码*************************************************
public void testWriteByte() throws Exception { // constants for test final String TEST_NAME = "testWriteByte"; final int TEST_SR = 22050; final int TEST_CONF = AudioFormat.CHANNEL_OUT_MONO; final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT; final int TEST_MODE = AudioTrack.MODE_STREAM; final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC; //-------- initialization -------------- int minBuffSize = AudioTrack.getMinBufferSize(TEST_SR, TEST_CONF, TEST_FORMAT); AudioTrack track = new AudioTrack(TEST_STREAM_TYPE, TEST_SR, TEST_CONF, TEST_FORMAT, 2*minBuffSize, TEST_MODE); byte data[] = new byte[minBuffSize]; //-------- test -------------- assumeTrue(TEST_NAME, track.getState() == AudioTrack.STATE_INITIALIZED); assertTrue(TEST_NAME, track.write(data, 0, data.length) == data.length); //-------- tear down -------------- track.release(); }

***********************************************************************************************
源码路径：
frameworks\base\media\tests\mediaframeworktest\src\com\android\mediaframeworktest\functional\MediaAudioTrackTest.java


###########################################说明##############################################################
1、TEST_NAME就不作说明了。
2、TEST_SR，是函数AudioTrack.getMinBufferSize的第一个参数。
   关于该参数的注释为：
     the sample rate expressed in Hertz. 也就是以赫兹为单位的采样率。
   函数AudioTrack.getMinBufferSize将会细品，此处就不再累述。
3、TEST_CONF，是函数AudioTrack.getMinBufferSize的第二个参数。
   关于该参数的注释为：
     describes the configuration of the audio channels. 
     *   See {@link AudioFormat#CHANNEL_OUT_MONO} and
     *   {@link AudioFormat#CHANNEL_OUT_STEREO}
   我们看到，其赋值为AudioFormat.CHANNEL_OUT_MONO。那就先说说AudioFormat。
   类AudioFormat的英文注释如下：
/**
 * The AudioFormat class is used to access a number of audio format and
 * channel configuration constants. They are for instance used
 * in {@link AudioTrack} and {@link AudioRecord}.
 * 
 */
   看了下其内容，主要包括各种track和record的channel的定义，和一些格式定义。
   我们此处预备创建一个AudioTrack，可用的Channel类型如下：
public static final int CHANNEL_OUT_FRONT_LEFT = 0x4; public static final int CHANNEL_OUT_FRONT_RIGHT = 0x8; public static final int CHANNEL_OUT_FRONT_CENTER = 0x10; public static final int CHANNEL_OUT_LOW_FREQUENCY = 0x20; public static final int CHANNEL_OUT_BACK_LEFT = 0x40; public static final int CHANNEL_OUT_BACK_RIGHT = 0x80; public static final int CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100; public static final int CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200; public static final int CHANNEL_OUT_BACK_CENTER = 0x400; public static final int CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT; public static final int CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT); public static final int CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT); public static final int CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER); public static final int CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT); public static final int CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER);



   从以下注释可知，此处的Channel定义，应该与include/media/AudioSystem.h中的保持一致。
   // Channel mask definitions must be kept in sync with native values in include/media/AudioSystem.h


4、TEST_FORMAT，是函数AudioTrack.getMinBufferSize的第三个参数。
   关于该参数的注释为：
   the format in which the audio data is represented. 
     *   See {@link AudioFormat#ENCODING_PCM_16BIT} and 
     *   {@link AudioFormat#ENCODING_PCM_8BIT}
   其赋值为AudioFormat.ENCODING_PCM_16BIT。还在类AudioFormat中。
   可用的类型如下：
    /** Audio data format: PCM 16 bit per sample */
    public static final int ENCODING_PCM_16BIT = 2; // accessed by native code
    /** Audio data format: PCM 8 bit per sample */
    public static final int ENCODING_PCM_8BIT = 3;  // accessed by native code


5、TEST_MODE，是AudioTrack的构造函数的第六个参数。
   该参数的注释如下：
   streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}
   其赋值为AudioTrack.MODE_STREAM。是类AudioTrack中定义的常量。
   可用的类型有以下两种：
    /**
     * Creation mode where audio data is transferred from Java to the native layer
     * only once before the audio starts playing.
     */
    public static final int MODE_STATIC = 0;
    /**
     * Creation mode where audio data is streamed from Java to the native layer
     * as the audio is playing.
     */
    public static final int MODE_STREAM = 1;


   看了下类AudioTrack的注释，其中大部分内容都是说MODE_STATIC与MODE_STREAM的差别的。
   注释如下：
/**
 * The AudioTrack class manages and plays a single audio resource for Java applications.
 * It allows to stream PCM audio buffers to the audio hardware for playback. This is
 * achieved by "pushing" the data to the AudioTrack object using one of the
 *  {@link #write(byte[], int, int)} and {@link #write(short[], int, int)} methods.
 *  
 * <p>An AudioTrack instance can operate under two modes: static or streaming.<br>
 * In Streaming mode, the application writes a continuous stream of data to the AudioTrack, using
 * one of the write() methods. These are blocking and return when the data has been transferred
 * from the Java layer to the native layer and queued for playback. The streaming mode
 *  is most useful when playing blocks of audio data that for instance are:
 * <ul>
 *   <li>too big to fit in memory because of the duration of the sound to play,</li>
 *   <li>too big to fit in memory because of the characteristics of the audio data
 *         (high sampling rate, bits per sample ...)</li>
 *   <li>received or generated while previously queued audio is playing.</li>
 * </ul>
 * The static mode is to be chosen when dealing with short sounds that fit in memory and
 * that need to be played with the smallest latency possible. AudioTrack instances in static mode
 * can play the sound without the need to transfer the audio data from Java to native layer
 * each time the sound is to be played. The static mode will therefore be preferred for UI and
 * game sounds that are played often, and with the smallest overhead possible.
 * 
 * <p>Upon creation, an AudioTrack object initializes its associated audio buffer.
 * The size of this buffer, specified during the construction, determines how long an AudioTrack
 * can play before running out of data.<br>
 * For an AudioTrack using the static mode, this size is the maximum size of the sound that can
 * be played from it.<br>
 * For the streaming mode, data will be written to the hardware in chunks of
 * sizes inferior to the total buffer size.
 */
   主要内容是说：
     MODE_STREAM是采用流的方式。也就是说，随着文件的播放，不停地有数据从Java层传到Native层。
       这中模式适合比较大的，并且对延迟没有要求的音频文件。
     MODE_STATIC是一次将数据从Java层传到Native层。
       这种模式时候数据量小（应为要存在内存中，要考虑内存消耗），并且对延迟有要求的音频。
   详细说明，可以仔细阅读英文注释。


6、TEST_STREAM_TYPE，是类AudioTrack构造函数中的第一个参数。
   该参数的注释如下：
    the type of the audio stream. See
     *   {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM},
     *   {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC} and
     *   {@link AudioManager#STREAM_ALARM}
   赋值的类型为 AudioManager.STREAM_MUSIC，是类AudioManager中定义的常量。
   可用的有以下十种类型：
/** The audio stream for phone calls */ public static final int STREAM_VOICE_CALL = AudioSystem.STREAM_VOICE_CALL; /** The audio stream for system sounds */ public static final int STREAM_SYSTEM = AudioSystem.STREAM_SYSTEM; /** The audio stream for the phone ring */ public static final int STREAM_RING = AudioSystem.STREAM_RING; /** The audio stream for music playback */ public static final int STREAM_MUSIC = AudioSystem.STREAM_MUSIC; /** The audio stream for alarms */ public static final int STREAM_ALARM = AudioSystem.STREAM_ALARM; /** The audio stream for notifications */ public static final int STREAM_NOTIFICATION = AudioSystem.STREAM_NOTIFICATION; /** @hide The audio stream for phone calls when connected to bluetooth */ public static final int STREAM_BLUETOOTH_SCO = AudioSystem.STREAM_BLUETOOTH_SCO; /** @hide The audio stream for enforced system sounds in certain countries (e.g camera in Japan) */ public static final int STREAM_SYSTEM_ENFORCED = AudioSystem.STREAM_SYSTEM_ENFORCED; /** The audio stream for DTMF Tones */ public static final int STREAM_DTMF = AudioSystem.STREAM_DTMF; /** @hide The audio stream for text to speech (TTS) */ public static final int STREAM_TTS = AudioSystem.STREAM_TTS;



   类AudioManager的注释如下：
   AudioManager provides access to volume and ringer mode control.


   各种类型的赋值都是从类AudioSystem中而来，类AudioSystem中的相关定义如下：
/* FIXME: Need to finalize this and correlate with native layer */ /* * If these are modified, please also update Settings.System.VOLUME_SETTINGS * and attrs.xml */ /* The audio stream for phone calls */ public static final int STREAM_VOICE_CALL = 0; /* The audio stream for system sounds */ public static final int STREAM_SYSTEM = 1; /* The audio stream for the phone ring and message alerts */ public static final int STREAM_RING = 2; /* The audio stream for music playback */ public static final int STREAM_MUSIC = 3; /* The audio stream for alarms */ public static final int STREAM_ALARM = 4; /* The audio stream for notifications */ public static final int STREAM_NOTIFICATION = 5; /* @hide The audio stream for phone calls when connected on bluetooth */ public static final int STREAM_BLUETOOTH_SCO = 6; /* @hide The audio stream for enforced system sounds in certain countries (e.g camera in Japan) */ public static final int STREAM_SYSTEM_ENFORCED = 7; /* @hide The audio stream for DTMF tones */ public static final int STREAM_DTMF = 8; /* @hide The audio stream for text to speech (TTS) */ public static final int STREAM_TTS = 9;

   从注释中可知，需要将此处的定义与Native层的正确关联。
   并且，如果这些内容改变，需要更新Settings.System.VOLUME_SETTINGS和attrs.xml。


7、下面是代码：
     int minBuffSize = AudioTrack.getMinBufferSize(TEST_SR, TEST_CONF, TEST_FORMAT);
   从函数的名字可知，是获取最小Buffer的大小。也就是说，如果想让我正常工作，至少要给我这些Buffer。
   提该要求的依据有采样率、Channel数量和样本大小（8BIT还是16BIT）。


8、接下来就是创建一个AudioTrack对象：
      AudioTrack track = new AudioTrack(TEST_STREAM_TYPE, TEST_SR, TEST_CONF, TEST_FORMAT, 
                2*minBuffSize, TEST_MODE);
   参数中，TEST_SR, TEST_CONF, TEST_FORMAT和函数AudioTrack.getMinBufferSize的相同。
   TEST_STREAM_TYPE是流动类型。
   minBuffSize是上面请求到的最小的Buffer Size。不过此处为何会乘以个2？？？
   看了下类AudioTrack的构造函数中的注释：
     * @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read
     *   from for playback. If using the AudioTrack in streaming mode, you can write data into
     *   this buffer in smaller chunks than this size. If using the AudioTrack in static mode,
     *   this is the maximum size of the sound that will be played for this instance.
     *   See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size
     *   for the successful creation of an AudioTrack instance in streaming mode. Using values
     *   smaller than getMinBufferSize() will result in an initialization failure.
   还是不明白。
   再看下函数getMinBufferSize的注释：
     * @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed,
     *   or {@link #ERROR} if the implementation was unable to query the hardware for its output 
     *     properties, 
     *   or the minimum buffer size expressed in bytes.
   函数getMinBufferSize的返回值是以byte为单位，AudioTrack构造函数中的参数也是以byte为单位，况且接下来的语句：
      byte data[] = new byte[minBuffSize];
   创建的buffer的大小也是minBuffSize。
   究竟为何乘个2？？？
   AudioTrack的构造函数中会做Buffer size check：
     audioBuffSizeCheck(bufferSizeInBytes);


   函数audioBuffSizeCheck的注释如下：
    // Convenience method for the contructor's audio buffer size check.
    // preconditions:
    //    mChannelCount is valid
    //    mAudioFormat is valid
    // postcondition:
    //    mNativeBufferSizeInBytes is valid (multiple of frame size, positive)
   需要保证buffer size为正数，并且是frame的整数倍。
   frame是个嘛概念？看看函数audioBuffSizeCheck的实现：
private void audioBuffSizeCheck(int audioBufferSize) { // NB: this section is only valid with PCM data. // To update when supporting compressed formats int frameSizeInBytes = mChannelCount * (mAudioFormat == AudioFormat.ENCODING_PCM_8BIT ? 1 : 2); if ((audioBufferSize % frameSizeInBytes != 0) || (audioBufferSize < 1)) { throw (new IllegalArgumentException("Invalid audio buffer size.")); } mNativeBufferSizeInBytes = audioBufferSize; }

   我们的Channel为AudioFormat.CHANNEL_OUT_MONO，所以mChannelCount为1，mAudioFormat为2，所以frameSizeInBytes等于2。
   如果audioBufferSize不是2（frameSizeInBytes）的整数倍，将会抛出异常！！！
   纳炉嚎啕！！！


9、创建Buffer：
     byte data[] = new byte[minBuffSize];
   可以，Java层中Buffer大小仍然为minBuffSize。
   乘以2的，是传给Native层的：
// native initialization int initResult = native_setup(new WeakReference<AudioTrack>(this), mStreamType, mSampleRate, mChannels, mAudioFormat, mNativeBufferSizeInBytes, mDataLoadMode, session);

   也就是说，要保证Native中，buffer的大小为frame的整数倍。


10、接下来是状态判断：
      assumeTrue(TEST_NAME, track.getState() == AudioTrack.STATE_INITIALIZED);
    Android中Media操作时，涉及到一个状态问题。
    也就是说，从一个状态，只能迁移到特定的一个或多个状态。即，需要在特定的状态下操作才有效，否则将导致错误。
    函数getState的注释：
    /**
     * Returns the state of the AudioTrack instance. This is useful after the
     * AudioTrack instance has been created to check if it was initialized
     * properly. This ensures that the appropriate hardware resources have been
     * acquired.
     * @see #STATE_INITIALIZED
     * @see #STATE_NO_STATIC_DATA
     * @see #STATE_UNINITIALIZED
     */


11、下面开始写数据：
    assertTrue(TEST_NAME,
                track.write(data, 0, data.length) == data.length);
    write函数将会细品，此处不再累述。
    其注释如下，可以先对其有个大致了解：
    /**
     * Writes the audio data to the audio hardware for playback.
     * @param audioData the array that holds the data to play.
     * @param offsetInBytes the offset expressed in bytes in audioData where the data to play 
     *    starts.
     * @param sizeInBytes the number of bytes to read in audioData after the offset.
     * @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION}
     *    if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if
     *    the parameters don't resolve to valid data and indexes.
     */


12、最后一步操作：
      track.release();


    其实现：
/** * Releases the native AudioTrack resources. */ 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. } native_release(); mState = STATE_UNINITIALIZED; }

    先调用自己的stop函数，然后再调到native层中的native_release函数。


    stop函数的实现：
/** * Stops playing the audio data. * @throws IllegalStateException */ public void stop() throws IllegalStateException { if (mState != STATE_INITIALIZED) { throw(new IllegalStateException("stop() called on uninitialized AudioTrack.")); } // stop playing synchronized(mPlayStateLock) { native_stop(); mPlayState = PLAYSTATE_STOPPED; } }

    先判断状态，然后调到native层的native_stop函数。


    如果从Java层调到native层？是通过JNI机制。
    就不在此介绍JNI机制了。


    上面提到的两个native中的函数，都是在文件：frameworks\base\core\jni\android_media_AudioTrack.cpp
    中进行关联的：
static JNINativeMethod gMethods[] = { // name, signature, funcPtr {"native_start", "()V", (void *)android_media_AudioTrack_start}, {"native_stop", "()V", (void *)android_media_AudioTrack_stop}, {"native_pause", "()V", (void *)android_media_AudioTrack_pause}, {"native_flush", "()V", (void *)android_media_AudioTrack_flush}, {"native_setup", "(Ljava/lang/Object;IIIIII[I)I", (void *)android_media_AudioTrack_native_setup}, {"native_finalize", "()V", (void *)android_media_AudioTrack_native_finalize}, {"native_release", "()V", (void *)android_media_AudioTrack_native_release}, {"native_write_byte", "([BIII)I", (void *)android_media_AudioTrack_native_write}, {"native_write_short", "([SIII)I", (void *)android_media_AudioTrack_native_write_short}, {"native_setVolume", "(FF)V", (void *)android_media_AudioTrack_set_volume}, {"native_get_native_frame_count", "()I", (void *)android_media_AudioTrack_get_native_frame_count}, {"native_set_playback_rate", "(I)I", (void *)android_media_AudioTrack_set_playback_rate}, {"native_get_playback_rate", "()I", (void *)android_media_AudioTrack_get_playback_rate}, {"native_set_marker_pos","(I)I", (void *)android_media_AudioTrack_set_marker_pos}, {"native_get_marker_pos","()I", (void *)android_media_AudioTrack_get_marker_pos}, {"native_set_pos_update_period", "(I)I", (void *)android_media_AudioTrack_set_pos_update_period}, {"native_get_pos_update_period", "()I", (void *)android_media_AudioTrack_get_pos_update_period}, {"native_set_position", "(I)I", (void *)android_media_AudioTrack_set_position}, {"native_get_position", "()I", (void *)android_media_AudioTrack_get_position}, {"native_set_loop", "(III)I", (void *)android_media_AudioTrack_set_loop}, {"native_reload_static", "()I", (void *)android_media_AudioTrack_reload}, {"native_get_output_sample_rate", "(I)I", (void *)android_media_AudioTrack_get_output_sample_rate}, {"native_get_min_buff_size", "(III)I", (void *)android_media_AudioTrack_get_min_buff_size}, {"native_setAuxEffectSendLevel", "(F)V", (void *)android_media_AudioTrack_setAuxEffectSendLevel}, {"native_attachAuxEffect", "(I)I", (void *)android_media_AudioTrack_attachAuxEffect}, };



    native_stop对应的函数为android_media_AudioTrack_stop：
static void android_media_AudioTrack_stop(JNIEnv *env, jobject thiz) { AudioTrack *lpTrack = (AudioTrack *)env->GetIntField( thiz, javaAudioTrackFields.nativeTrackInJavaObj); if (lpTrack == NULL ) { jniThrowException(env, "java/lang/IllegalStateException", "Unable to retrieve AudioTrack pointer for stop()"); return; } lpTrack->stop(); }



    native_release对应的函数为android_media_AudioTrack_native_release：
static void android_media_AudioTrack_native_release(JNIEnv *env, jobject thiz) { // do everything a call to finalize would android_media_AudioTrack_native_finalize(env, thiz); // + reset the native resources in the Java object so any attempt to access // them after a call to release fails. env->SetIntField(thiz, javaAudioTrackFields.nativeTrackInJavaObj, 0); env->SetIntField(thiz, javaAudioTrackFields.jniData, 0); }



    函数android_media_AudioTrack_native_finalize的实现：
static void android_media_AudioTrack_native_finalize(JNIEnv *env, jobject thiz) { //LOGV("android_media_AudioTrack_native_finalize jobject: %x\n", (int)thiz); // delete the AudioTrack object AudioTrack *lpTrack = (AudioTrack *)env->GetIntField( thiz, javaAudioTrackFields.nativeTrackInJavaObj); if (lpTrack) { //LOGV("deleting lpTrack: %x\n", (int)lpTrack); lpTrack->stop(); delete lpTrack; } // delete the JNI data AudioTrackJniStorage* pJniStorage = (AudioTrackJniStorage *)env->GetIntField( thiz, javaAudioTrackFields.jniData); if (pJniStorage) { // delete global refs created in native_setup env->DeleteGlobalRef(pJniStorage->mCallbackData.audioTrack_class); env->DeleteGlobalRef(pJniStorage->mCallbackData.audioTrack_ref); //LOGV("deleting pJniStorage: %x\n", (int)pJniStorage); delete pJniStorage; } }



    函数android_media_AudioTrack_stop和android_media_AudioTrack_native_finalize都调用了函数env->GetIntField：
    AudioTrack *lpTrack = (AudioTrack *)env->GetIntField(
        thiz, javaAudioTrackFields.nativeTrackInJavaObj);
    看意思应该是获取Java侧保存的native的Track对象。
    既然此处是Get，那就应该有地方去Set。
    不错，上面的函数android_media_AudioTrack_native_release中就有去Set：
      env->SetIntField(thiz, javaAudioTrackFields.nativeTrackInJavaObj, 0);
    不过，此处是将其清0。
    真正Set的地方在哪儿？一个字，搜！


    且慢，先看看javaAudioTrackFields是个嘛东东：
struct fields_t { // these fields provide access from C++ to the... jclass audioTrackClass; //... AudioTrack class jmethodID postNativeEventInJava; //... event post callback method int PCM16; //... format constants int PCM8; //... format constants int STREAM_VOICE_CALL; //... stream type constants int STREAM_SYSTEM; //... stream type constants int STREAM_RING; //... stream type constants int STREAM_MUSIC; //... stream type constants int STREAM_ALARM; //... stream type constants int STREAM_NOTIFICATION; //... stream type constants int STREAM_BLUETOOTH_SCO; //... stream type constants int STREAM_DTMF; //... stream type constants int MODE_STREAM; //... memory mode int MODE_STATIC; //... memory mode jfieldID nativeTrackInJavaObj; // stores in Java the native AudioTrack object jfieldID jniData; // stores in Java additional resources used by the native AudioTrack }; static fields_t javaAudioTrackFields;

    原来是为了提供一个从C++访问...的区域，此处...应该是Java，是不是以后也扩展到其他语言？
    其中nativeTrackInJavaObj是保存在Java侧的native的AudioTrack对象。


    继续刚才的话题，搜！
    原来在函数android_media_AudioTrack_native_setup中调用了函数 env->SetIntField来实现set的。
    文件路径：frameworks\base\core\jni\android_media_AudioTrack.cpp
    与刚才从Java层调到native中的入口相同，也就是说函数android_media_AudioTrack_native_setup应该也是从Java层调过来的。
    找了下对应表，果然，对应的是native_setup函数。


    函数android_media_AudioTrack_native_setup的内容就先不细嚼了，大致处理如下：
      参数及状态检查
      创建一个AudioTrack对象。
      调用AudioTrack对象的一些初始化和设置函数。
      最后将AudioTrack对象通过env->SetIntField函数保存到Java层。
    与此类似处理的还有一个AudioTrackJniStorage对象。


总结一下使用示例：
1、首先根据采用率，样本大小，声道数获取一个最小需要的buffersize。
2、根据流的类型，模式（stream或static），1中获取的最小buffersize（为了native中的buffer size是frame的整数倍，此处乘了个2），以及采用率，样本大小，声道数来创建
一个AudioTrack。此处的AudioTrack是Java中的类，其构造函数最终会调到native中，并创建一个native中的AudioTrack类，并通过函数env->SetIntField将其保存到Java的
AudioTrack对象中。
3、调用AudioTrack对象的write函数，此处直接掉的是Java的AudioTrack对象，函数write中应该会调到native中的AudioTrack对象。信不信由你，反正我是信了。
4、调用release函数，停止播放，并释放资源。