MT6737 Android N 平台 Audio系统学习----AudioTrack
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一、 框架
从上图可知Android播放声音有MediaPlayer和AudioTrack两种方法，这两种方法区别是比较大的。
MediaPlayer：可以播放多种格式的声音文件，如mp3、wav、ogg、aac等等，其会在framework层创建了对应的音频解码器。
AudioTrack：只能播放已经解码的PCM流，播放文件的格式也只支持wav格式，因为wav格式音频大部分为PCM流，其不创建解码器。
MediaPlayer和AudioTrack联系：由上图可知，MediaPlayer在framework层还是会创建AudioTrack，把解码后的PCM数据流传给AudioTrack，AudioTrack再传给AudioFlinger进行混音，然后才传给硬件播放，所以是MediaPlayer包含了AudioTrack。
二、 程序详解（根据MT6737 Android N平台为例）
1、代码相关代码路径及框图
frameworks/base/media/java/android/media/AudioTrack.java
frameworks/base/core/jni/android_media_AudioTrack.cpp
frameworks/av/media/libmedia/AudioTrack.cpp
2、调用AudioTrack api写个播放音乐示例
package com.audiotrack;import java.io.File;import java.io.FileInputStream;import java.io.FileNotFoundException;import java.io.FileOutputStream;import java.io.IOException;import android.app.Activity;import android.media.AudioFormat;import android.media.AudioManager;import android.media.AudioRecord;import android.media.AudioTrack;import android.media.MediaRecorder;import android.os.Bundle;import android.os.Environment;import android.os.Handler;import android.os.Message;import android.view.View;import android.view.View.OnClickListener;import android.widget.Button;import android.widget.RadioButton;import android.widget.RadioGroup;import android.widget.Toast;import android.widget.RadioGroup.OnCheckedChangeListener;public class ActivityMain extends Activity{ /** * btn 录音 */ private Button btnRecord; /** * btn 停止录音 */ private Button btnStopRecord; /** * btn 播放 */ private Button btnPlay; /** * btn 停止播放 */ private Button btnStopPlay; /** * btn 即时播放 */ private Button btnInstantPlay; /** * btn 停止即时播放 */ private Button btnStopInstantPlay; /** * rGrp 声道 */ private RadioGroup rGrpTrack; /** * rGrp 采样率 */ private RadioGroup rGrpFP; /** * rGrp 采样精度，编码率 */ private RadioGroup rGrpPCM; /** * rbtn 单声道 */ private RadioButton rbtnSingle; /** * rbtn 双声道 */ private RadioButton rbtnDouble; /** * rbtn 采样率 44100hz */ private RadioButton rbtn44100; /** * rbtn 采样率 11025hz */ private RadioButton rbtn11025; /** * rbtn 采样率 22050hz */ private RadioButton rbtn22050; /** * rbtn 采样精度 pcm8 */ private RadioButton rbtnPcm8; /** * rbtn 采样精度 pcm16 */ private RadioButton rbtnPcm16; private int recBufSize = 0; private int playBufSize = 0; /** * 采样率（默认44100，每秒44100个点） */ private int sampleRateInHz = 44100; /** * 声道（默认单声道） */ private int channelConfig = AudioFormat.CHANNEL_CONFIGURATION_MONO; /** * 编码率（默认ENCODING_PCM_16BIT） */ private int encodingBitrate = AudioFormat.ENCODING_PCM_16BIT; private AudioRecord audioRecord; private AudioTrack audioTrack; /** * 是否录音 */ private boolean blnRecord = false; /** * 是否播放 */ private boolean blnPlay = false; /** * 即时播放 */ private boolean blnInstantPlay = false; /** * 录音线程 */ private Thread threadRecord; /** * 播放录音线程 */ private ThreadAudioTrack threadAudioTrack; /** * 文件夹 */ private static final String AUDIO_RECORDER_FOLDER = "audioRecorder"; /** * 临时文件名称 */ private static final String AUDIO_RECORDER_TEMP_FILE = "record_temp.raw"; /** * 文件名称 */ private static final String AUDIO_RECORDER_FILE = "session.wav"; private static final int RECORDER_BPP = 16; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); btnRecord = (Button)findViewById(R.id.btn_record); btnStopRecord = (Button)findViewById(R.id.btn_stop_record); btnPlay = (Button)findViewById(R.id.btn_play); btnStopPlay = (Button)findViewById(R.id.btn_stop_play); btnInstantPlay = (Button)findViewById(R.id.btn_instantplay); btnStopInstantPlay = (Button)findViewById(R.id.btn_stop_instantplay); rGrpTrack = (RadioGroup)findViewById(R.id.rgrp_track); rGrpFP = (RadioGroup)findViewById(R.id.rgrp_fp); rGrpPCM = (RadioGroup)findViewById(R.id.rgrp_pcm); rbtnSingle = (RadioButton)findViewById(R.id.rbtn_single); rbtnDouble = (RadioButton)findViewById(R.id.rbtn_double); rbtn44100 = (RadioButton)findViewById(R.id.rbtn_fp_44100); rbtn11025 = (RadioButton)findViewById(R.id.rbtn_fp_11025); rbtn22050 = (RadioButton)findViewById(R.id.rbtn_fp_22050); rbtnPcm8 = (RadioButton)findViewById(R.id.rbtn_pcm_8bit); rbtnPcm16 = (RadioButton)findViewById(R.id.rbtn_pcm_16bit); btnStopRecord.setEnabled(false); btnPlay.setEnabled(false); btnStopPlay.setEnabled(false); btnStopInstantPlay.setEnabled(false); //声道 rGrpTrack.setOnCheckedChangeListener(new OnCheckedChangeListener() { @Override public void onCheckedChanged(RadioGroup group, int checkedId) { if(checkedId==rbtnSingle.getId()) //单声道 { channelConfig = AudioFormat.CHANNEL_CONFIGURATION_MONO; } else if(checkedId==rbtnDouble.getId()) //双声道 { channelConfig = AudioFormat.CHANNEL_CONFIGURATION_STEREO; } } }); rGrpFP.setOnCheckedChangeListener(new OnCheckedChangeListener() { @Override public void onCheckedChanged(RadioGroup group, int checkedId) { if(checkedId==rbtn44100.getId()) { sampleRateInHz = 44100; } else if(checkedId==rbtn11025.getId()) { sampleRateInHz = 11025; } else if(checkedId==rbtn22050.getId()) { sampleRateInHz = 22050; } if(blnInstantPlay||blnPlay) { audioTrack.setPlaybackRate(sampleRateInHz); } } }); rGrpPCM.setOnCheckedChangeListener(new OnCheckedChangeListener() { @Override public void onCheckedChanged(RadioGroup group, int checkedId) { if(checkedId==rbtnPcm8.getId()) { encodingBitrate = AudioFormat.ENCODING_PCM_8BIT; } else if(checkedId==rbtnPcm16.getId()) { encodingBitrate = AudioFormat.ENCODING_PCM_16BIT; } } }); //录音 btnRecord.setOnClickListener(new OnClickListener() { @Override public void onClick(View v) { btnRecord.setEnabled(false); btnInstantPlay.setEnabled(false); btnStopRecord.setEnabled(true); startRecord(); } }); //停止录音 btnStopRecord.setOnClickListener(new OnClickListener() { @Override public void onClick(View v) { btnStopRecord.setEnabled(false); btnPlay.setEnabled(true); stopRecord(); } }); //播放 btnPlay.setOnClickListener(new OnClickListener() { @Override public void onClick(View v) { btnStopPlay.setEnabled(true); btnPlay.setEnabled(false); btnInstantPlay.setEnabled(false); threadAudioTrack = new ThreadAudioTrack(); threadAudioTrack.init(); threadAudioTrack.start(); } }); //停止播放 btnStopPlay.setOnClickListener(new OnClickListener() { @Override public void onClick(View v) { threadAudioTrack.free(); threadAudioTrack = null; btnRecord.setEnabled(true); btnInstantPlay.setEnabled(true); btnStopRecord.setEnabled(false); btnPlay.setEnabled(false); btnStopPlay.setEnabled(false); } }); //开始即时播放 btnInstantPlay.setOnClickListener(new OnClickListener() { @Override public void onClick(View v) { btnRecord.setEnabled(false); btnInstantPlay.setEnabled(false); btnStopInstantPlay.setEnabled(true); btnPlay.setEnabled(false); recBufSize = AudioRecord.getMinBufferSize(sampleRateInHz, channelConfig, encodingBitrate); playBufSize = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, encodingBitrate); audioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, sampleRateInHz, channelConfig, encodingBitrate, recBufSize); audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, encodingBitrate, playBufSize, AudioTrack.MODE_STREAM); blnInstantPlay = true; new ThreadInstantPlay().start(); } }); //停止即时播放 btnStopInstantPlay.setOnClickListener(new OnClickListener() { @Override public void onClick(View v) { blnInstantPlay = false; btnStopInstantPlay.setEnabled(false); btnRecord.setEnabled(true); btnInstantPlay.setEnabled(true); } }); } /** * 开始录音 */ private void startRecord() { //根据采样率、声道、采样精度 决定frame的大小 recBufSize = AudioRecord.getMinBufferSize(sampleRateInHz, channelConfig, encodingBitrate); audioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, sampleRateInHz, channelConfig, encodingBitrate, recBufSize); audioRecord.startRecording(); blnRecord = true; threadRecord = new Thread(new Runnable() { @Override public void run() { writeAudioDataToFile(); } }); threadRecord.start(); } /** * 停止录音 */ private void stopRecord() { if(audioRecord!=null) { blnRecord = false; audioRecord.stop(); audioRecord.release(); audioRecord = null; threadRecord = null; } copyWaveFile(getTempFilename(),getFileName()); deleteTempFile(); } /** * 删除临时文件 */ private void deleteTempFile() { File file = new File(getTempFilename()); file.delete(); } /** * 将语音信息 */ private void writeAudioDataToFile() { byte[] bs = new byte[recBufSize]; String fileName = getTempFilename(); FileOutputStream fos = null; try { fos = new FileOutputStream(fileName); } catch (FileNotFoundException e) { e.printStackTrace(); } int line = 0; if(fos!=null) { while(blnRecord) { line = audioRecord.read(bs, 0, recBufSize); if(line!=AudioRecord.ERROR_INVALID_OPERATION) { try { fos.write(bs); } catch (IOException e) { e.printStackTrace(); } } } } } /** * 获取临时文件名称 * * @return String */ private String getTempFilename() { String filePath = Environment.getExternalStorageDirectory().getPath(); File file = new File(filePath, AUDIO_RECORDER_FOLDER); if(!file.exists()) { file.mkdirs(); } File tmpFile = new File(filePath, AUDIO_RECORDER_TEMP_FILE); if(tmpFile.exists()) { tmpFile.delete(); } return (file.getPath() + "/" + AUDIO_RECORDER_TEMP_FILE); } /** * 获取文件名称 * * @return String */ private String getFileName() { String filePath = Environment.getExternalStorageDirectory().getAbsolutePath(); File file = new File(filePath, AUDIO_RECORDER_FOLDER); if(file.exists()) { file.delete(); } return (file.getAbsolutePath() + "/" + AUDIO_RECORDER_FILE); } /** * 复制文件 * * @param inFilename * @param outFilename */ private void copyWaveFile(String inFilename,String outFilename) { FileInputStream fis = null; FileOutputStream fos = null; long totalAudioLen = 0; long totalDataLen = totalAudioLen + 36; long sampleRate = sampleRateInHz; int channels = 2; long byteRate = RECORDER_BPP * sampleRateInHz * channels/8; byte[] data = new byte[recBufSize]; try { fis = new FileInputStream(inFilename); fos = new FileOutputStream(outFilename); totalAudioLen = fis.getChannel().size(); totalDataLen = totalAudioLen + 36; writeWaveFileHeader(fos, totalAudioLen, totalDataLen, sampleRate, channels, byteRate); while(fis.read(data)!=-1) { fos.write(data); } fis.close(); fos.close(); } catch(FileNotFoundException e) { e.printStackTrace(); } catch (IOException e) { e.printStackTrace(); } } /** * @param fos * @param totalAudioLen * @param totalDataLen * @param longSampleRate * @param channels * @param byteRate * @throws IOException */ private void writeWaveFileHeader(FileOutputStream fos, long totalAudioLen, long totalDataLen, long sampleRate, int channels, long byteRate) throws IOException { byte[] header = new byte[44]; header[0] = 'R'; // RIFF/WAVE header header[1] = 'I'; header[2] = 'F'; header[3] = 'F'; header[4] = (byte) (totalDataLen & 0xff); header[5] = (byte) ((totalDataLen >> 8) & 0xff); header[6] = (byte) ((totalDataLen >> 16) & 0xff); header[7] = (byte) ((totalDataLen >> 24) & 0xff); header[8] = 'W'; header[9] = 'A'; header[10] = 'V'; header[11] = 'E'; header[12] = 'f'; // 'fmt ' chunk header[13] = 'm'; header[14] = 't'; header[15] = ' '; header[16] = 16; // 4 bytes: size of 'fmt ' chunk header[17] = 0; header[18] = 0; header[19] = 0; header[20] = 1; // format = 1 header[21] = 0; header[22] = (byte) channels; header[23] = 0; header[24] = (byte) (sampleRate & 0xff); header[25] = (byte) ((sampleRate >> 8) & 0xff); header[26] = (byte) ((sampleRate >> 16) & 0xff); header[27] = (byte) ((sampleRate >> 24) & 0xff); header[28] = (byte) (byteRate & 0xff); header[29] = (byte) ((byteRate >> 8) & 0xff); header[30] = (byte) ((byteRate >> 16) & 0xff); header[31] = (byte) ((byteRate >> 24) & 0xff); header[32] = (byte) (2 * 16 / 8); // block align header[33] = 0; header[34] = RECORDER_BPP; // bits per sample header[35] = 0; header[36] = 'd'; header[37] = 'a'; header[38] = 't'; header[39] = 'a'; header[40] = (byte) (totalAudioLen & 0xff); header[41] = (byte) ((totalAudioLen >> 8) & 0xff); header[42] = (byte) ((totalAudioLen >> 16) & 0xff); header[43] = (byte) ((totalAudioLen >> 24) & 0xff); fos.write(header, 0, 44); } /** * 播放录音线程 */ class ThreadAudioTrack extends Thread { byte[] bs; File file; FileInputStream fis; /** * 初始化AudioTrack */ public void init() { file = new File("/sdcard/" + AUDIO_RECORDER_FOLDER +"/", AUDIO_RECORDER_FILE); try { file.createNewFile(); fis = new FileInputStream(file); blnPlay = true; playBufSize = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, encodingBitrate); audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, encodingBitrate, playBufSize, AudioTrack.MODE_STREAM); bs = new byte[playBufSize]; } catch (IOException e) { e.printStackTrace(); } } /** * 释放AudioTrack */ public void free() { blnPlay = false; } @Override public void run() { audioTrack.play(); while(blnPlay) { try { int line = fis.read(bs, 0, recBufSize); if(line==-1) { blnPlay = false; handler.sendMessage(new Message());//发送空消息体 return; } byte[] tmpBuf = new byte[line]; System.arraycopy(bs, 0, tmpBuf, 0, line);// fis.read(bs);// tmpBuf = bs.clone(); audioTrack.write(tmpBuf, 0, tmpBuf.length); } catch (IOException e) { e.printStackTrace(); } } audioTrack.stop(); audioTrack = null; try { fis.close(); } catch (IOException e) { e.printStackTrace(); } } } /** * * 即时播放线程 * */ class ThreadInstantPlay extends Thread { @Override public void run() { byte[] bsBuffer = new byte[recBufSize]; audioRecord.startRecording(); audioTrack.play(); while(blnInstantPlay) { int line = audioRecord.read(bsBuffer, 0, recBufSize); byte[] tmpBuf = new byte[line]; System.arraycopy(bsBuffer, 0, tmpBuf, 0, line); audioTrack.write(tmpBuf, 0, tmpBuf.length); } audioTrack.stop(); audioRecord.stop(); } } @Override protected void onDestroy() { super.onDestroy(); android.os.Process.killProcess(android.os.Process.myPid()); } Handler handler = new Handler() { @Override public void handleMessage(Message msg) { super.handleMessage(msg); Toast.makeText(ActivityMain.this, "播放结束！", 2000).show(); btnPlay.setEnabled(true); btnInstantPlay.setEnabled(true); btnStopPlay.setEnabled(false); } };}3、分析AudioTrack（java空间）
AudioTrack.java文件主要为应用层提供API接口，将函数进行封装。下面对部分函数进行分析。
3.1、先看下例子中调用的AudioTrack构造函数的代码
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,// 音频流类型 sampleRateInHz, //采样率 channelConfig, //声道数 encodingBitrate, //采样精度 playBufSize, //分配音频数据缓冲区的最小size AudioTrack.MODE_STREAM);//数据加载模式3.1.1、AudioTrack函数各参数分析
3.1.1.1、音频流的类型
追踪代码到实现定义这个参数文件在AudioTrack.java->AudioManager.java->AudioSytem.java
/* The default audio stream */ public static final int STREAM_DEFAULT = -1; /* The audio stream for phone calls */ public static final int STREAM_VOICE_CALL = 0;//通话声 /* The audio stream for system sounds */ public static final int STREAM_SYSTEM = 1;//系统声音 /* The audio stream for the phone ring and message alerts */ public static final int STREAM_RING = 2;//铃声 /* The audio stream for music playback */ public static final int STREAM_MUSIC = 3;//音乐声 /* The audio stream for alarms */ public static final int STREAM_ALARM = 4;//警告声 /* The audio stream for notifications */ public static final int STREAM_NOTIFICATION = 5;//通知音频流 /* @hide The audio stream for phone calls when connected on bluetooth */ public static final int STREAM_BLUETOOTH_SCO = 6;//从注释上看时使用蓝牙耳机通话的音频流 /* @hide The audio stream for enforced system sounds in certain countries (e.g camera in Japan) */ public static final int STREAM_SYSTEM_ENFORCED = 7;//一些国家强制使用的音频流，比如日本要求任何模式下拍照都要有声音，防止偷拍 /* @hide The audio stream for DTMF tones */ public static final int STREAM_DTMF = 8;//DTMF音频流 /* @hide The audio stream for text to speech (TTS) */ public static final int STREAM_TTS = 9;//文件到语言的音频流，即机器说话 #####3.1.1.2、采样率 AudioFormat.java中可以看出采样率范围4000<sampleRateInHz <192000。 public static final int SAMPLE_RATE_HZ_MIN = 4000; /* Maximum value for sample rate*/ public static final int SAMPLE_RATE_HZ_MAX = 192000; /* Sample rate will be a route-dependent value.*/ public static final int SAMPLE_RATE_UNSPECIFIED = 0; if ((sampleRateInHz < AudioFormat.SAMPLE_RATE_HZ_MIN || sampleRateInHz > AudioFormat.SAMPLE_RATE_HZ_MAX) && sampleRateInHz != AudioFormat.SAMPLE_RATE_UNSPECIFIED) { throw new IllegalArgumentException(sampleRateInHz + "Hz is not a supported sample rate."); } mSampleRate = sampleRateInHz;#####3.1.1.3、声道 AudioFormat.java有定义声道类型： @Deprecated public static final int CHANNEL_CONFIGURATION_INVALID = 0; @Deprecated public static final int CHANNEL_CONFIGURATION_DEFAULT = 1; @Deprecated public static final int CHANNEL_CONFIGURATION_MONO = 2;//单声道 @Deprecated public static final int CHANNEL_CONFIGURATION_STEREO = 3;//双声道#####3.1.1.4、采样精度 AudioFormat.java有定义采样精度： /** Invalid audio data format */ public static final int ENCODING_INVALID = 0; /** Default audio data format */ public static final int ENCODING_DEFAULT = 1; /** Audio data format: PCM 16 bit per sample. Guaranteed to be supported by devices. */ public static final int ENCODING_PCM_16BIT = 2;//一个采样点16比特，相当于2个字节 /** Audio data format: PCM 8 bit per sample. Not guaranteed to be supported by devices. */ public static final int ENCODING_PCM_8BIT = 3;//一个采样点8比特，相当于1个字节 /** Audio data format: single-precision floating-point per sample */ public static final int ENCODING_PCM_FLOAT = 4;3.1.1.5、缓冲区playBufSize
playBufSize 是根据音频数据的特性来确定所要分配的缓冲区的最小size。
playBufSize = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, encodingBitrate);3.1.1.6、数据加载模式
AudioTrack数据加载模式有两种MODE_STATIC 和 MODE_STREAM。
STREAM的意思是由用户在应用程序通过write方式把数据一次一次得写到audiotrack中。这个和我们在socket中发送数据一样，应用层从某个地方获取数据，例如通过编解码得到PCM数据，然后write到audiotrack。这种方式的坏处就是总是在JAVA层和Native层交互，效率损失较大。
STATIC的意思是一开始创建的时候，就把音频数据放到一个固定的buffer，然后直接传给audiotrack，后续就不用一次次得write了。AudioTrack会自己播放这个buffer中的数据。这种方法对于铃声等内存占用较小，延时要求较高的声音来说很适用。
/** * 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; /** @hide */ @IntDef({ MODE_STATIC, MODE_STREAM })3.1.2、AudioTrack函数的实现
3.1.2.1、AudioTrack函数在AudioTrack.java中实现
AudioTrack函数实现在AudioTrack.java中，下面对此实现函数进行分析。
public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,int mode, int sessionId) throws IllegalArgumentException { super(attributes); // mState already == STATE_UNINITIALIZED if (format == null) { throw new IllegalArgumentException("Illegal null AudioFormat"); } // remember which looper is associated with the AudioTrack instantiation Looper looper; if ((looper = Looper.myLooper()) == null) { looper = Looper.getMainLooper(); } int rate = format.getSampleRate(); if (rate == AudioFormat.SAMPLE_RATE_UNSPECIFIED) { rate = 0; } 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);//检查参数是否合法 mStreamType = AudioSystem.STREAM_DEFAULT; audioBuffSizeCheck(bufferSizeInBytes);//检查缓冲区大小 mInitializationLooper = looper;//记录创建实例时所在的looper /* 创建AudioTrack的时候需要指定一个audio session。如果这个session已经被其他的播放器或者audio track使用，新创建的AudioTrack就会和它们共享AudioEffect。如果是第一次使用该session，AudioEffect就会将session与新创建的AudioTrack对象关联起来，以后其他的播放器或audio track再使用该session时，就会与该AudioTrack共享AudioEffect。如果没有指定一个session，就会为该AudioTrack创建一个自己的session，该session可以被别人通过getAudioSessionId得到。别人得到了该session，并且使用该session创建播放器或者audio track，就会与你共享AudioEffect。 */ //检查sessionId的合法性 if (sessionId < 0) { throw new IllegalArgumentException("Invalid audio session ID: "+sessionId); } int[] sampleRate = new int[] {mSampleRate}; int[] session = new int[1];//new一个变量，用来保存sessionId session[0] = sessionId; // native initialization //调用native层的native_setup，将上层设置的采样率、声道、采样精度、数据加载模式、buffersize传给native层 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]; //根据加载模式，设置状态 if (mDataLoadMode == MODE_STATIC) { mState = STATE_NO_STATIC_DATA; } else { mState = STATE_INITIALIZED; } }3.1.2.2、native_setup函数调用
native_setup函数通过JNI调用到native层函数，通过下图JNI函数对照表可知native层对应函数是android_media_AudioTrack.cpp中的android_media_AudioTrack_setup。 
下面进入android_media_AudioTrack_setup函数进行分析。
static jint android_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; //检查参数合法 if (jSession == NULL) { ALOGE("Error creating AudioTrack: invalid session ID pointer"); return (jint) AUDIO_JAVA_ERROR; } jint* nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL); if (nSession == NULL) { ALOGE("Error creating AudioTrack: Error retrieving session id pointer"); return (jint) AUDIO_JAVA_ERROR; } audio_session_t sessionId = (audio_session_t) nSession[0]; 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. 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; } 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);//check采样精度 if (format == AUDIO_FORMAT_INVALID) { ALOGE("Error creating AudioTrack: unsupported audio format %d.", audioFormat); return (jint) AUDIOTRACK_ERROR_SETUP_INVALIDFORMAT; } // compute the frame count计算帧数 //帧数 = buffSizeInBytes / (channel数 * 每个channel数据量) 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; } //创建一个native AudioTrack对象 // create the native AudioTrack object lpTrack = new AudioTrack(); // read the AudioAttributes values paa = (audio_attributes_t *) calloc(1, sizeof(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这些数据将通过每个AudioTrack回调 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; //初始化不同加载模式下的native AudioTrack对象 // initialize the native AudioTrack object 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 共享内存，STREAM模式下为空，实际使用的共享内存在AudioFlinger中创建 true,// thread can call Java内部线程可以调用JNI函数 sessionId,// audio session ID AudioTrack::TRANSFER_SYNC, NULL, // default offloadInfo -1, -1, // default uid, pid values paa); break; case MODE_STATIC: //如果是static模式，需要先创建共享内存 // 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 // need to be initialized from the Native AudioTrack // these are directly returned from getters: // mSampleRate // mAudioFormat // mStreamType // mChannelConfiguration // mChannelCount // mState (?) // mPlayState (?) // these may be used internally (Java AudioTrack.audioParamCheck(): // mChannelMask // mChannelIndexMask // mDataLoadMode // 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; } nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL); if (nSession == NULL) { ALOGE("Error creating AudioTrack: Error retrieving session id pointer"); goto native_init_failure; } // read the audio session ID back from AudioTrack in case we create a new session nSession[0] = lpTrack->getSessionId(); env->ReleasePrimitiveArrayCritical(jSession, nSession, 0); nSession = NULL; { const jint elements[1] = { (jint) lpTrack->getSampleRate() }; env->SetIntArrayRegion(jSampleRate, 0, 1, elements); } { // scope for the lock Mutex::Autolock l(sLock); sAudioTrackCallBackCookies.add(&lpJniStorage->mCallbackData); } // save our newly created C++ AudioTrack in the "nativeTrackInJavaObj" field // of the Java object (in mNativeTrackInJavaObj) setAudioTrack(env, thiz, lpTrack); // save the JNI resources so we can free them later //ALOGV("storing lpJniStorage: %x\n", (long)lpJniStorage); env->SetLongField(thiz, javaAudioTrackFields.jniData, (jlong)lpJniStorage); // since we had audio attributes, the stream type was derived from them during the // 把JNI层中new出来的AudioTrack对象指针保存到Java对象的一个变量中，这样就把JNI层的AudioTrack对象和Java层的AudioTrack对象关联起来了，这是Android的常用技法。 // creation of the native AudioTrack: push the same value to the Java object env->SetIntField(thiz, javaAudioTrackFields.fieldStreamType, (jint) lpTrack->streamType()); if (paa != NULL) { // audio attributes were copied in AudioTrack creation free(paa); paa = NULL; } return (jint) AUDIO_JAVA_SUCCESS; // failures: native_init_failure: if (paa != NULL) { free(paa); } if (nSession != NULL) { env->ReleasePrimitiveArrayCritical(jSession, nSession, 0); } env->DeleteGlobalRef(lpJniStorage->mCallbackData.audioTrack_class); env->DeleteGlobalRef(lpJniStorage->mCallbackData.audioTrack_ref); delete lpJniStorage; env->SetLongField(thiz, javaAudioTrackFields.jniData, 0); // lpTrack goes out of scope, so reference count drops to zero return (jint) AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED;}3.1.2.3、分析AudioTrackJniStorage
AudioTrack函数位于AudioTrack.cpp中，现在进行分析。
class AudioTrackJniStorage { public: sp<MemoryHeapBase> mMemHeap; sp<MemoryBase> mMemBase; audiotrack_callback_cookie mCallbackData; sp<JNIDeviceCallback> mDeviceCallback; AudioTrackJniStorage() { mCallbackData.audioTrack_class = 0; mCallbackData.audioTrack_ref = 0; } ~AudioTrackJniStorage() { mMemBase.clear(); mMemHeap.clear(); } //分配一块指定大小的匿名共享内存 bool allocSharedMem(int sizeInBytes) { //创建一个匿名共享内存 mMemHeap = new MemoryHeapBase(sizeInBytes, 0, "AudioTrack Heap Base"); if (mMemHeap->getHeapID() < 0) { return false; } mMemBase = new MemoryBase(mMemHeap, 0, sizeInBytes); //注意用法，先弄一个HeapBase，再把HeapBase传入到MemoryBase中去 return true; }};MemoryHeapBase和MemoryBase主要干了下面两件事：（1）分配了一块共享内存，这样两个进程可以共享这块内存。（2）基于Binder通信，这样使用这两个类的进程就可以交互了。MemoryHeapBase::MemoryHeapBase(size_t size, uint32_t flags, char const * name) : mFD(-1), mSize(0), mBase(MAP_FAILED), mFlags(flags), mDevice(0), mNeedUnmap(false), mOffset(0) { //获取内存页大小 const size_t pagesize = getpagesize(); //字节对齐 size = ((size + pagesize-1) & ~(pagesize-1)); /* 创建共享内存，打开/dev/ashmem设备，得到一个文件描述符 */ int fd = ashmem_create_region(name == NULL ? "MemoryHeapBase" : name, size); ALOGE_IF(fd<0, "error creating ashmem region: %s", strerror(errno)); if (fd >= 0) { //通过mmap将匿名共享内存映射到当前进程地址空间 if (mapfd(fd, size) == NO_ERROR) { if (flags & READ_ONLY) { ashmem_set_prot_region(fd, PROT_READ); } } } } 3.2、分析audio.play()
3.2.1、AudioTrack.java中实现
public void play() throws IllegalStateException { if (mState != STATE_INITIALIZED) { throw new IllegalStateException("play() called on uninitialized AudioTrack."); } baseStart(); synchronized(mPlayStateLock) { native_start();//调用JNI接口android_media_AudioTrack.cpp的android_media_AudioTrack_start mPlayState = PLAYSTATE_PLAYING; } }3.2.2、native层native_start
`{"native_start", "()V", (void *)android_media_AudioTrack_start},`从JNI对照表可知native_start对应android_media_AudioTrack.cpp中android_media_AudioTrack_start函数。
static void android_media_AudioTrack_start(JNIEnv *env, jobject thiz) { sp<AudioTrack> lpTrack = getAudioTrack(env, thiz); if (lpTrack == NULL) { jniThrowException(env, "java/lang/IllegalStateException", "Unable to retrieve AudioTrack pointer for start()"); return; } lpTrack->start();//调用AudioTrack的start方法 }3.3、分析audio.write()
3.3.1、分析AudioTrack中write
java层write函数有两个，一个是用来写PCM16数据的，它对应的一个采样点的数据量是两个字节。另一个是用来写PCM8数据的，它对应的一个采样点的数据量是一个字节。
public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts, @WriteMode int writeMode) { if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) { return ERROR_INVALID_OPERATION; } if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); return ERROR_BAD_VALUE; } if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0) || (offsetInShorts + sizeInShorts < 0) // detect integer overflow || (offsetInShorts + sizeInShorts > audioData.length)) { return ERROR_BAD_VALUE; } int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat, writeMode == WRITE_BLOCKING);//通过JNI调用android_media_AudioTrack.cpp中函数(void *)android_media_AudioTrack_writeArray<jshortArray> if ((mDataLoadMode == MODE_STATIC) && (mState == STATE_NO_STATIC_DATA) && (ret > 0)) { // benign race with respect to other APIs that read mState mState = STATE_INITIALIZED; } return ret; } public int write(@NonNull float[] audioData, int offsetInFloats, int sizeInFloats, @WriteMode int writeMode) { if (mState == STATE_UNINITIALIZED) { Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); return ERROR_INVALID_OPERATION; } if (mAudioFormat != AudioFormat.ENCODING_PCM_FLOAT) { Log.e(TAG, "AudioTrack.write(float[] ...) requires format ENCODING_PCM_FLOAT"); return ERROR_INVALID_OPERATION; } if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); return ERROR_BAD_VALUE; } if ( (audioData == null) || (offsetInFloats < 0 ) || (sizeInFloats < 0) || (offsetInFloats + sizeInFloats < 0) // detect integer overflow || (offsetInFloats + sizeInFloats > audioData.length)) { Log.e(TAG, "AudioTrack.write() called with invalid array, offset, or size"); return ERROR_BAD_VALUE; } int ret = native_write_float(audioData, offsetInFloats, sizeInFloats, mAudioFormat, writeMode == WRITE_BLOCKING);//通过JNI调用android_media_AudioTrack.cpp中函数(void *)android_media_AudioTrack_writeArray<jfloatArray> if ((mDataLoadMode == MODE_STATIC) && (mState == STATE_NO_STATIC_DATA) && (ret > 0)) { // benign race with respect to other APIs that read mState mState = STATE_INITIALIZED; } return ret; }3.3.2、分析(void *)android_media_AudioTrack_writeArray
template <typename T> static jint android_media_AudioTrack_writeArray(JNIEnv *env, jobject thiz, T javaAudioData, jint offsetInSamples, jint sizeInSamples, jint javaAudioFormat, jboolean isWriteBlocking) { //ALOGV("android_media_AudioTrack_writeArray(offset=%d, sizeInSamples=%d) called", // offsetInSamples, sizeInSamples); sp<AudioTrack> lpTrack = getAudioTrack(env, thiz); if (lpTrack == NULL) { jniThrowException(env, "java/lang/IllegalStateException", "Unable to retrieve AudioTrack pointer for write()"); return (jint)AUDIO_JAVA_INVALID_OPERATION; } if (javaAudioData == NULL) { ALOGE("NULL java array of audio data to play"); return (jint)AUDIO_JAVA_BAD_VALUE; } // NOTE: We may use GetPrimitiveArrayCritical() when the JNI implementation changes in such // a way that it becomes much more efficient. When doing so, we will have to prevent the // AudioSystem callback to be called while in critical section (in case of media server // process crash for instance) // get the pointer for the audio data from the java array auto cAudioData = envGetArrayElements(env, javaAudioData, NULL); if (cAudioData == NULL) { ALOGE("Error retrieving source of audio data to play"); return (jint)AUDIO_JAVA_BAD_VALUE; // out of memory or no data to load } jint samplesWritten = writeToTrack(lpTrack, javaAudioFormat, cAudioData, offsetInSamples, sizeInSamples, isWriteBlocking == JNI_TRUE /* blocking */); envReleaseArrayElements(env, javaAudioData, cAudioData, 0); //ALOGV("write wrote %d (tried %d) samples in the native AudioTrack with offset %d", // (int)samplesWritten, (int)(sizeInSamples), (int)offsetInSamples); return samplesWritten; } 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); //如果是STATIC模式，sharedBuffer（）返回不为空。如果是STREAM模式，sharedBuffer（）返回为空。 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 if ((size_t)sizeInBytes > track->sharedBuffer()->size()) { sizeInBytes = track->sharedBuffer()->size(); } //在STATIC模式下，直接把数据memcpy到共享内存，记住在这种模式下要先调用write，后调用play。 memcpy(track->sharedBuffer()->pointer(), data + offsetInSamples, sizeInBytes); written = sizeInBytes; } if (written >= 0) { return written / sizeof(T); } return interpretWriteSizeError(written);}3.4、分析audio.release()
当数据都write完后，需要调用stop停止播放，或者直接调用release来释放相关资源。由于release和stop有一定的相关性，这里只分析release调用。
3.4.1、分析AudioTrack.java中的release（）
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();//通过JNI调用android_media_AudioTrack.cpp中函数android_media_AudioTrack_release mState = STATE_UNINITIALIZED; } @Override protected void finalize() { baseRelease(); native_finalize(); } public void stop() throws IllegalStateException { if (mState != STATE_INITIALIZED) { throw new IllegalStateException("stop() called on uninitialized AudioTrack."); } // stop playing synchronized(mPlayStateLock) { native_stop();//通过JNI调用android_media_AudioTrack.cpp中函数android_media_AudioTrack_stop mPlayState = PLAYSTATE_STOPPED; mAvSyncHeader = null; mAvSyncBytesRemaining = 0; } }3.4.1、分析android_media_AudioTrack_stop和android_media_AudioTrack_release
#define CALLBACK_COND_WAIT_TIMEOUT_MS 1000 static 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; } } static void android_media_AudioTrack_stop(JNIEnv *env, jobject thiz) { sp<AudioTrack> lpTrack = getAudioTrack(env, thiz); if (lpTrack == NULL) { jniThrowException(env, "java/lang/IllegalStateException", "Unable to retrieve AudioTrack pointer for stop()"); return; } lpTrack->stop(); }3.5AudioTrack（Java空间）的分析总结：
AudioTrack在JNI层使用了Native的AudioTrack对象，总结一下调用Native对象的流程：
· new一个AudioTrack，使用无参的构造函数。
· 调用set函数，把Java层的参数传进去，另外还设置了一个audiocallback回调函数。
· 调用了AudioTrack的start函数。
· 调用AudioTrack的write函数。
· 工作完毕后，调用stop。
· 最后就是Native对象的delete。
4、分析AudioTrack（Native空间）
native层AudioTrack代码在AudioTrack.cpp中，下面来进行分析。
4.1、AudioTrack构造函数
在上面android_media_AudioTrack.cpp中有lpTrack = new AudioTrack()这么一句，其创建了一个native AudioTrack对象。下面看看AudioTrack构造函数。
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, ""); }4.2、AudioTrack set调用
status_t AudioTrack::set( audio_stream_type_t streamType,//音频流加载模式 uint32_t sampleRate,//采样率 audio_format_t format,//采样格式 audio_channel_mask_t channelMask,//输出声道 size_t frameCount,//帧数 audio_output_flags_t flags,//输出标志位 callback_t cbf,//callback function(会调函数)，如果不是null，这个函数会被定期调用来提供新的数据和通知标记，位置更新等。 void* user,//为cbf参数 int32_t notificationFrames, const sp<IMemory>& sharedBuffer,//共享内存 bool threadCanCallJava, audio_session_t sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid, const audio_attributes_t* pAttributes, bool doNotReconnect, float maxRequiredSpeed){ ALOGD("set(): %p, streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " "flags #%x, notificationFrames %u, sessionId %d, transferType %d, uid %d, pid %d", this, streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, sessionId, transferType, uid, pid); /// M: ALPS02266941: For Game Detection @{ #ifdef MTK_GAS_SERVICE_SUPPORT sp<GameDetectionThread> gameDetectionThread = new GameDetectionThread; gameDetectionThread->run("GameDetectionThread"); #endif /// @} #ifdef MTK_AOSP_ENHANCEMENT if(AUDIO_STREAM_ENFORCED_AUDIBLE == streamType) { flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); ALOGD("Camera not support fast !!"); } #endif mThreadCanCallJava = threadCanCallJava;/* 为AudioTrack设置音频参数信息,transfer_type参数用于指定音频数据的传输方式，Android定义了4种音频数据传输方式：enum transfer_type { TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters TRANSFER_CALLBACK, // callback EVENT_MORE_DATA TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() TRANSFER_SYNC, // synchronous write() TRANSFER_SHARED, // shared memory};*///设置音频数据类型 switch (transferType) { case TRANSFER_DEFAULT: if (sharedBuffer != 0) { transferType = TRANSFER_SHARED; } else if (cbf == NULL || threadCanCallJava) { transferType = TRANSFER_SYNC; } else { transferType = TRANSFER_CALLBACK; } break; case TRANSFER_CALLBACK: if (cbf == NULL || sharedBuffer != 0) { ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); return BAD_VALUE; } break; case TRANSFER_OBTAIN: case TRANSFER_SYNC: if (sharedBuffer != 0) { ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); return BAD_VALUE; } break; case TRANSFER_SHARED: if (sharedBuffer == 0) { ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); return BAD_VALUE; } break; default: ALOGE("Invalid transfer type %d", transferType); return BAD_VALUE; } mSharedBuffer = sharedBuffer; mTransfer = transferType; mDoNotReconnect = doNotReconnect; ALOGD_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %zu", sharedBuffer->pointer(), sharedBuffer->size()); ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); #ifdef MTK_AOSP_ENHANCEMENT SLOGD("audiotrack %p set Type %d, rate %d, fmt %d, chn %d, fcnt %zu, flags %04x", this, streamType, sampleRate, format, channelMask, frameCount, flags); #endif // invariant that mAudioTrack != 0 is true only after set() returns successfully if (mAudioTrack != 0) { ALOGE("Track already in use"); return INVALID_OPERATION; }//音频数据流加载模式设置 // handle default values first. if (streamType == AUDIO_STREAM_DEFAULT) { streamType = AUDIO_STREAM_MUSIC; } if (pAttributes == NULL) { if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) { ALOGE("Invalid stream type %d", streamType); return BAD_VALUE; } mStreamType = streamType; } else { // stream type shouldn't be looked at, this track has audio attributes memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); mStreamType = AUDIO_STREAM_DEFAULT; if ((mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } if ((mAttributes.flags & AUDIO_FLAG_LOW_LATENCY) != 0) { flags = (audio_output_flags_t) (flags | AUDIO_OUTPUT_FLAG_FAST); } } //音频格式设置 // these below should probably come from the audioFlinger too... if (format == AUDIO_FORMAT_DEFAULT) { format = AUDIO_FORMAT_PCM_16_BIT; } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through? mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO; } // validate parameters if (!audio_is_valid_format(format)) { ALOGE("Invalid format %#x", format); return BAD_VALUE; } mFormat = format; if (!audio_is_output_channel(channelMask)) { ALOGE("Invalid channel mask %#x", channelMask); return BAD_VALUE; } mChannelMask = channelMask; uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); mChannelCount = channelCount; // force direct flag if format is not linear PCM // or offload was requested if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) || !audio_is_linear_pcm(format)) { ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) ? "Offload request, forcing to Direct Output" : "Not linear PCM, forcing to Direct Output"); flags = (audio_output_flags_t) // FIXME why can't we allow direct AND fast? ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); } // force direct flag if HW A/V sync requested if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); } if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { if (audio_has_proportional_frames(format)) { mFrameSize = channelCount * audio_bytes_per_sample(format); } else { mFrameSize = sizeof(uint8_t); } #ifdef MTK_AOSP_ENHANCEMENT // AudioSystem Get HDMI capability // if Channel count is multi-channel if(channelMask != AUDIO_CHANNEL_OUT_STEREO && channelMask != AUDIO_CHANNEL_OUT_MONO) { // Check if HDMI is connected if(AudioSystem::getDeviceConnectionState(AUDIO_DEVICE_OUT_AUX_DIGITAL,"") == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { int hdim_channelCount, hdmi_maxSampleRate, hdmi_bitwidth; hdim_channelCount = channelCount; // need to get AF HDMI capability. if (AudioSystem::getHDMICapability(&hdim_channelCount, &hdmi_bitwidth, &hdmi_maxSampleRate ) != NO_ERROR) { return BAD_VALUE; } ALOGD("GetHDMICapability hdim_channelCount %d, maxSampleRate%d, bitwidth %d", hdim_channelCount, hdmi_maxSampleRate, hdmi_bitwidth); // if AF channel count > 2 if (hdim_channelCount > 2) { // track is Multi-channel and HDMI support Multi-channel. // raise for direct output. flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT) ; ALOGD("flags %d", flags); } } else { ALOGD("receive multi-channel content and not detecting aux digital"); } } #endif } else { ALOG_ASSERT(audio_has_proportional_frames(format)); mFrameSize = channelCount * audio_bytes_per_sample(format); // createTrack will return an error if PCM format is not supported by server, // so no need to check for specific PCM formats here } // sampling rate must be specified for direct outputs if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { return BAD_VALUE; } mSampleRate = sampleRate; mOriginalSampleRate = sampleRate; mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT; // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX); // Make copy of input parameter offloadInfo so that in the future: // (a) createTrack_l doesn't need it as an input parameter // (b) we can support re-creation of offloaded tracks if (offloadInfo != NULL) { mOffloadInfoCopy = *offloadInfo; mOffloadInfo = &mOffloadInfoCopy; } else { mOffloadInfo = NULL; } //AudioTrack初始化 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; mSendLevel = 0.0f; // mFrameCount is initialized in createTrack_l mReqFrameCount = frameCount; if (notificationFrames >= 0) { mNotificationFramesReq = notificationFrames; mNotificationsPerBufferReq = 0; } else { if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { ALOGE("notificationFrames=%d not permitted for non-fast track", notificationFrames); return BAD_VALUE; } if (frameCount > 0) { ALOGE("notificationFrames=%d not permitted with non-zero frameCount=%zu", notificationFrames, frameCount); return BAD_VALUE; } mNotificationFramesReq = 0; const uint32_t minNotificationsPerBuffer = 1; const uint32_t maxNotificationsPerBuffer = 8; mNotificationsPerBufferReq = min(maxNotificationsPerBuffer, max((uint32_t) -notificationFrames, minNotificationsPerBuffer)); ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames, "notificationFrames=%d clamped to the range -%u to -%u", notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer); } mNotificationFramesAct = 0; if (sessionId == AUDIO_SESSION_ALLOCATE) { mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); } else { mSessionId = sessionId; } int callingpid = IPCThreadState::self()->getCallingPid(); int mypid = getpid(); if (uid == -1 || (callingpid != mypid)) { mClientUid = IPCThreadState::self()->getCallingUid(); } else { mClientUid = uid; } if (pid == -1 || (callingpid != mypid)) { mClientPid = callingpid; } else { mClientPid = pid; } mAuxEffectId = 0; mOrigFlags = mFlags = flags; mCbf = cbf; //cbf是JNI层传入的回调函数audioCallback,如果用户设置了回调函数，则启动AudioTrackThread线程来提供音频数据 if (cbf != NULL) { ALOGD("set: Create AudioTrackThread"); 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() } //调用createTrack_l // create the IAudioTrack status_t status = createTrack_l(); if (status != NO_ERROR) { ALOGD("set: createTrack_l fail! status = %d", status); if (mAudioTrackThread != 0) { mAudioTrackThread->requestExit(); // see comment in AudioTrack.h mAudioTrackThread->requestExitAndWait(); mAudioTrackThread.clear(); } return status; } mStatus = NO_ERROR; mUserData = user; mLoopCount = 0; mLoopStart = 0; mLoopEnd = 0; mLoopCountNotified = 0; mMarkerPosition = 0; mMarkerReached = false; mNewPosition = 0; mUpdatePeriod = 0; mPosition = 0; mReleased = 0; mStartUs = 0; AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); mSequence = 1; mObservedSequence = mSequence; mInUnderrun = false; mPreviousTimestampValid = false; mTimestampStartupGlitchReported = false; mRetrogradeMotionReported = false; mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID; mUnderrunCountOffset = 0; mFramesWritten = 0; mFramesWrittenServerOffset = 0; return NO_ERROR;}4.3、AudioTrackThread线程
AudioTrack支持两种数据输入方式：
（1） Push方式：用户主动write，MediaPlayerService通常采用此方式；
（2） Pull方式： AudioTrackThread线程通过audioCallback回调函数主动从用户那里获取数据，ToneGenerator就是采用这种方式；
bool AudioTrack::AudioTrackThread::threadLoop() { { AutoMutex _l(mMyLock); ALOGV("AudioTrackThread::threadLoop, mPaused = %d, mIgnoreNextPausedInt = %d, mPausedInt = %d", mPaused, mIgnoreNextPausedInt, mPausedInt); 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()) { ALOGD("AudioTrackThread::threadLoop exitPending"); return false; } //调用创建当前AudioTrackThread线程的AudioTrack的processAudioBuffer函数 nsecs_t ns = mReceiver.processAudioBuffer(); ALOGV("processAudioBuffer() returned %" PRId64, ns); 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; }}4.3、createTrack_l
音频播放需要AudioTrack写入音频数据，同时需要AudioFlinger完成混音，因此需要在AudioTrack与AudioFlinger之间建立数据通道，而AudioTrack与AudioFlinger又分属不同的进程空间，Android系统采用Binder通信方式来搭建它们之间的桥梁。
status_t AudioTrack::createTrack_l() { //得到AudioFlinger的代理对象 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); if (audioFlinger == 0) { ALOGE("Could not get audioflinger"); return NO_INIT; } #ifdef MTK_AOSP_ENHANCEMENT // forbit fast track for voice feature String8 ret = AudioSystem::getParameters(String8("VoiceFeatureOn")); if (String8("VoiceFeatureOn=1") == ret) { mFlags = (audio_output_flags_t)(mFlags & ~AUDIO_OUTPUT_FLAG_FAST); ALOGV("remove direct flag flags for voice feature %d", mFlags); } // AudioSystem Get HDMI capability // if Channel count is multi-channel ALOGV("mChannelMask 0x%x", mChannelMask); if (mChannelMask != AUDIO_CHANNEL_OUT_STEREO && mChannelMask != AUDIO_CHANNEL_OUT_MONO) { // Check if HDMI is connected if( AudioSystem::getDeviceConnectionState(AUDIO_DEVICE_OUT_AUX_DIGITAL, "") == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { int hdim_channelCount, hdmi_maxSampleRate, hdmi_bitwidth; hdim_channelCount = mChannelCount; // need to get AF HDMI capability. if (AudioSystem::getHDMICapability(&hdim_channelCount, &hdmi_bitwidth, &hdmi_maxSampleRate) != NO_ERROR) { return BAD_VALUE; } ALOGD("GetHDMICapability hdim_channelCount %d, maxSampleRate%d, bitwidth %d", hdim_channelCount, hdmi_maxSampleRate, hdmi_bitwidth); // if AF channel count > 2 if (hdim_channelCount > 2) { // track is Multi-channel and HDMI support Multi-channel. // raise for direct output. mFlags = (audio_output_flags_t)(mFlags | AUDIO_OUTPUT_FLAG_DIRECT); ALOGD("flags %d", mFlags); } else { mFlags = (audio_output_flags_t)(mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); ALOGD("remove direct flag flags %d", mFlags); } } else { ALOGD("receive multi-channel content and not detecting aux digital"); } } #endif if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); } audio_io_handle_t output; audio_stream_type_t streamType = mStreamType; audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted. // After fast request is denied, we will request again if IAudioTrack is re-created. status_t status; status = AudioSystem::getOutputForAttr(attr, &output, mSessionId, &streamType, mClientUid, mSampleRate, mFormat, mChannelMask, mFlags, mSelectedDeviceId, mOffloadInfo); if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x," " channel mask %#x, flags %#x", mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); return BAD_VALUE; } { // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, // we must release it ourselves if anything goes wrong. // Not all of these values are needed under all conditions, but it is easier to get them all status = AudioSystem::getLatency(output, &mAfLatency); if (status != NO_ERROR) { ALOGE("getLatency(%d) failed status %d", output, status); goto release; } ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency); //获取音频帧数 status = AudioSystem::getFrameCount(output, &mAfFrameCount); if (status != NO_ERROR) { ALOGE("getFrameCount(output=%d) status %d", output, status); goto release; } // TODO consider making this a member variable if there are other uses for it later size_t afFrameCountHAL; status = AudioSystem::getFrameCountHAL(output, &afFrameCountHAL); if (status != NO_ERROR) { ALOGE("getFrameCountHAL(output=%d) status %d", output, status); goto release; } ALOG_ASSERT(afFrameCountHAL > 0); status = AudioSystem::getSamplingRate(output, &mAfSampleRate); if (status != NO_ERROR) { ALOGE("getSamplingRate(output=%d) status %d", output, status); goto release; } if (mSampleRate == 0) { mSampleRate = mAfSampleRate; mOriginalSampleRate = mAfSampleRate; } #ifdef MTK_AOSP_ENHANCEMENT ALOGV("mAfLatency %d, mAfFrameCount %zu, mAfSampleRate %d", mAfLatency,mAfFrameCount, mAfSampleRate); if((mAfFrameCount <= 0) || (mAfSampleRate <= 0)) { ALOGE("Get audioflinger parameter error afFrameCount-%zu, afSampleRate-%u", mAfFrameCount, mAfSampleRate); return NO_INIT; } #endif // Client decides whether the track is TIMED (see below), but can only express a preference // for FAST. Server will perform additional tests. #ifndef MTK_AUDIO 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); } } #else 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; 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); } } #endif mNotificationFramesAct = mNotificationFramesReq; #ifdef MTK_AOSP_ENHANCEMENT if (!strcmp(mAttributes.tags, "addr=CrossMount-Mic")) { mReqFrameCount = 4094; ALOGD("found mic mount"); } #endif size_t frameCount = mReqFrameCount; ALOGD("mReqFrameCount %zu", 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; pid_t tid = -1; if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { trackFlags |= IAudioFlinger::TRACK_FAST; if (mAudioTrackThread != 0 && !mThreadCanCallJava) { tid = mAudioTrackThread->getTid(); } } if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { trackFlags |= IAudioFlinger::TRACK_OFFLOAD; } if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { trackFlags |= IAudioFlinger::TRACK_DIRECT; } #ifdef MTK_CROSSMOUNT_SUPPORT if (mFlags & AUDIO_OUTPUT_FLAG_TO_REMOTE_SUBMIX) { trackFlags |= IAudioFlinger::TRACK_REMOTE; } #endif #ifdef MTK_AOSP_ENHANCEMENT if(!strcmp(mAttributes.tags,"BootAnimationAudioTrack")) { ALOGD("mAttributes.tags = BootAnimationAudioTrack"); trackFlags |= IAudioFlinger::TRACK_BOOT; } #endif 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; //向AudioFlinger发送createTrack请求，在stream模式下sharedBuffer为空， output为AudioSystem::getOutput得到一个值，代表AF中的线程索引号该函数返回IAudioTrack（实际类型是BpAudioTrack）对象，后续AF和AT的交互就是围绕IAudioTrack进行的 。 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. //在STREAM模式下，没有在AudioTrack端创建共享内存，但前面提到了AudioTrack和AudioFligner的数据交互是通过共享内存完成的，这块共享内存最终由AudioFligner的createTrack创建。下面这个调用会取出AudioFligner创建的共享内存 // FIXME compare to AudioRecord sp<IMemory> iMem = track->getCblk(); if (iMem == 0) { ALOGE("Could not get control block"); return NO_INIT; } 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 if (mAudioTrack != 0) { IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); mDeathNotifier.clear(); } //将创建的Track代理对象、匿名共享内存代理对象保存到AudioTrack的成员变量中 mAudioTrack = track; mCblkMemory = iMem; IPCThreadState::self()->flushCommands(); //IMemory的pointer在此处将返回共享内存的首地址，类型为void*，static_cast直接把这个void*类型转成audio_track_cblk_t，表明这块内存的首部中存在audio_track_cblk_t这个对象 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); mCblk = cblk; ALOGD("createTrack_l: %p, mCblk = %p", this, mCblk); // 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) {//stream模式 //buffers指向数据空间，它的起始位置是共享内存的首部加上audio_track_cblk_t的大小 buffers = cblk + 1; } else {//static模式 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; ALOGD("createTrack_l: %p, mLatency = %d, mAfLatency = %d, frameCount = %zu, mSampleRate = %d", this, mLatency, mAfLatency, 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 if (mSharedBuffer == 0) { mStaticProxy.clear(); mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); } else { 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); } #ifdef MTK_AOSP_ENHANCEMENT gAudioTrackCenter.addTrack((intptr_t)mCblk, frameCount, mSampleRate, (void*)this, mAfFrameCount, mAfSampleRate, mReleased.value()); #endif return NO_ERROR; } release: AudioSystem::releaseOutput(output, streamType, mSessionId); if (status == NO_ERROR) { status = NO_INIT; } return status; }IAudioTrack建立了AudioTrack和AudioFligner之间的关系，在static模式下，用于存放音频数据的匿名共享内存在AudioTrack这边创建，在stream播放模式下，匿名共享内存却是在AudioFlinger这边创建。这两种播放模式下创建的匿名共享内存是有区别的，stream模式下的匿名共享内存头部会创建一个audio_track_cblk_t对象，用于协调生产者AudioTrack和消费者AudioFlinger之间的步调。createTrack就是在AudioFlinger中创建一个Track对象。
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