Record,play and visualize raw audio data in Android
通过 AudioRecord 和 AudioTrack 处理音频裸(raw)数据。
一些必要的理论
音频数据是模拟信号，转换模拟信号为数字数据的过程称之为抽样(sampling)
Wikipedia : In signal processing, sampling is the reduction of a continuous signal to a discrete signal. A common example is the conversion of a sound wave (a continuous signal) to a sequence of samples (a discrete-time signal). A sample is a value or set of values at a point in time and/or space
抽样的速度称为 sampling rate (不知标准译法，还是取原样好了)。通常音频录制的频率为 44100Hz : 即每秒从模拟信号中取44100个样本。最常见的模拟信号抽样方法为 Pulse-code modulation (PCM)。PCM 通常用 16位数据保存(signed short)。
由于 Android 手机的音频硬件差异，设备上的对立体声(stereo) 捕获及播放等高级功能支持程度也不一。不过，好消息是，所有设备都支持单声道 PCM。
录制裸数据
录音需要用到 AudioRecord 对象，初始化参数有：来源(source)，通道(channel config)，编码格式(encoding)及缓冲大小(buffersize)。 缓冲大小的单位为 byte , 表示一次可以获取的数据，这个参数可以使用 getMinBufferSize() 方法来计算：
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int bufferSize = AudioRecord.getMinBufferSize(SAMPLE_RATE,
AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT);
AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.DEFAULT,
44100,
AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT,
bufferSize);
有了 AudioRecord 对象就可以开始录音了，数据需要直接拉取，就像一个持续 IO ，因此这个操作要在一个独立的线程上完成。当完成了录音操作后，需要调用 stop 以释放 native 资源。
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final int SAMPLE_RATE = 44100; // The sampling rate
boolean mShouldContinue; // Indicates if recording / playback should stop
void recordAudio() {
new Thread(new Runnable() {
@Override
public void run() {
android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_AUDIO);
// buffer size in bytes
int bufferSize = AudioRecord.getMinBufferSize(SAMPLE_RATE,
AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT);
if (bufferSize == AudioRecord.ERROR || bufferSize == AudioRecord.ERROR_BAD_VALUE) {
bufferSize = SAMPLE_RATE * 2;
}
short[] audioBuffer = new short[bufferSize / 2];
AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.DEFAULT,
SAMPLE_RATE,
AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT,
bufferSize);
if (record.getState() != AudioRecord.STATE_INITIALIZED) {
Log.e(LOG_TAG, "Audio Record can't initialize!");
return;
}
record.startRecording();
Log.v(LOG_TAG, "Start recording");
long shortsRead = 0;
while (mShouldContinue) {
int numberOfShort = record.read(audioBuffer, 0, audioBuffer.length);
shortsRead += numberOfShort;
// Do something with the audioBuffer
}
record.stop();
record.release();
Log.v(LOG_TAG, String.format("Recording stopped. Samples read: %d", shortsRead));
}
}).start();
}
每获取一次数据，线程就会阻塞直至下一块数据填满缓冲。本例中，线程会阻塞约41毫秒。因此在 pool 循环中执行的代码，需要在41毫秒内结束，否则会错过一些数据。（译注：是错过数据，还是数据的时序不对？）如果你要保存数据，最好的办法是用生产消费者模式。
播放裸数据
AudioTracker 的构造函数和 AudioRecord 很相像。AudioTracker 有两种方式获取数据：static / stream 。前者把内容全加载进内存，播放小音频文件通常都用这种方法，而后者则通常用于大文件以提升性能。
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int mBufferSize = AudioTrack.getMinBufferSize(SAMPLE_RATE, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT);
if (mBufferSize == AudioTrack.ERROR || mBufferSize == AudioTrack.ERROR_BAD_VALUE) {
// For some readon we couldn't obtain a buffer size
mBufferSize = SAMPLE_RATE * CHANNELS * 2;
}
AudioTrack mAudioTrack = new AudioTrack(
AudioManager.STREAM_MUSIC,
SAMPLE_RATE,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT,
mBufferSize,
AudioTrack.MODE_STREAM);
同样，播放时，需要另起线程：
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ShortBuffer mSamples; // the samples to play
int mNumSamples; // number of samples to play
void playAudio() {
new Thread(new Runnable() {
@Override
public void run() {
int bufferSize = AudioTrack.getMinBufferSize(SAMPLE_RATE, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT);
if (bufferSize == AudioTrack.ERROR || bufferSize == AudioTrack.ERROR_BAD_VALUE) {
bufferSize = SAMPLE_RATE * 2;
}
AudioTrack audioTrack = new AudioTrack(
AudioManager.STREAM_MUSIC,
SAMPLE_RATE,
AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT,
bufferSize,
AudioTrack.MODE_STREAM);
audioTrack.play();
Log.v(LOG_TAG, "Audio streaming started");
short[] buffer = new short[bufferSize];
mSamples.rewind();
int limit = mNumSamples;
int totalWritten = 0;
while (mSamples.position() < limit && mShouldContinue) {
int numSamplesLeft = limit - mSamples.position();
int samplesToWrite;
if (numSamplesLeft >= buffer.length) {
mSamples.get(buffer);
samplesToWrite = buffer.length;
} else {
for (int i = numSamplesLeft; i < buffer.length; i++) {
buffer[i] = 0;
}
mSamples.get(buffer, 0, numSamplesLeft);
samplesToWrite = numSamplesLeft;
}
totalWritten += samplesToWrite;
audioTrack.write(buffer, 0, samplesToWrite);
}
if (!mShouldContinue) {
audioTrack.release();
}
Log.v(LOG_TAG, "Audio streaming finished. Samples written: " + totalWritten);
}
}).start();
}