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。