gistfile1.java

import java.io.OutputStream;
import java.util.concurrent.atomic.AtomicBoolean;

import android.media.AudioFormat;
import android.media.AudioRecord;
import android.media.MediaRecorder;
import android.os.Handler;
import android.util.Log;

public class MyAudioRecord {
    private static final String TAG = MyAudioRecord.class.getName();

    private static final int SAMPLE_RATE = QueuePlayer.SAMPLE_RATE;
    private static final int CHANNEL_CONFIG = AudioFormat.CHANNEL_IN_MONO;
    private static final int AUDIO_FORMAT = AudioFormat.ENCODING_PCM_16BIT; // 看系統. 本來我們實驗的 Hero 最低只能 16 bit, 不能 8 bit
    private static final int MIN_BUFFER_SIZE = AudioRecord.getMinBufferSize(
            SAMPLE_RATE, CHANNEL_CONFIG, AUDIO_FORMAT); // on hero : 4096 bytes
    private static final int AUDIO_RECORD_BUFFER_SIZE = MIN_BUFFER_SIZE * 10; // 看原廠 AudioRecordTest 中有 * 10
    private static final int OUR_16BIT_BUFFER_SIZE = MIN_BUFFER_SIZE * 2;
    //private static final int OUR_8BIT_BUFFER_SIZE = OUR_16BIT_BUFFER_SIZE / 2;

    private Thread t;
    private AtomicBoolean shouldStop = new AtomicBoolean(false);

    // non-blocking, 叫了就開始錄
    // maxMillis 是最大錄製的秒數 (粗略的, 防止爆掉用的)
    // 外面記得要在 stopListener 裏面做事, 包含關閉 output stream
    public MyAudioRecord(final OutputStream os, final long maxMillis, final Handler handler, final Runnable stopListener) {
        Log.v(TAG, "SAMPLE_RATE: " + SAMPLE_RATE + " CHANNEL_CONFIG: "
                + CHANNEL_CONFIG + " AUDIO_FORMAT: " + AUDIO_FORMAT);
        Log.v(TAG, "MIN_BUFFER_SIZE: " + MIN_BUFFER_SIZE
                + " AUDIO_RECORD_BUFFER_SIZE: " + AUDIO_RECORD_BUFFER_SIZE
                + " OUR_16BIT_BUFFER_SIZE: " + OUR_16BIT_BUFFER_SIZE
                );
                //+ " OUR_8BIT_BUFFER_SIZE: " + OUR_8BIT_BUFFER_SIZE);

        final byte[] our16BitBuffer = new byte[OUR_16BIT_BUFFER_SIZE];
        //final byte[] our8BitBuffer = new byte[OUR_8BIT_BUFFER_SIZE];

        final AudioRecord ar = new AudioRecord(MediaRecorder.AudioSource.DEFAULT, SAMPLE_RATE,
                CHANNEL_CONFIG, AUDIO_FORMAT, AUDIO_RECORD_BUFFER_SIZE);

        t = new Thread() {
            public void run() {
                Log.v(TAG, "run() >>>>");
                try {
                    long stopNano = System.nanoTime() + maxMillis * 1000 * 1000;
                    ar.startRecording();
                    while (true) {
                        if (shouldStop.get() == true || System.nanoTime() >= stopNano) {
                            ar.stop();
                            ar.release();
                            handler.post(stopListener); // TODO: 沒檢查傳回值, 放失敗就算了(?)
                            break;
                        }

                        int bytesRead = ar.read(our16BitBuffer, 0, our16BitBuffer.length);
                        //Log.v(TAG, "bytesRead: " + bytesRead);

                        os.write(our16BitBuffer, 0, bytesRead);



//                        // convert 16 to 8
//                        int j = 0;
//                        for (int i = 1; i < bytesRead; i += 2, j += 1) {
//                            our8BitBuffer[j] = our16BitBuffer[i];
//                            our8BitBuffer[j] ^= (1 << 7);
//                        }
//
//                        os.write(our8BitBuffer, 0, j);
                    }
                } catch (Exception e) {
                    e.printStackTrace();
                }
                Log.v(TAG, "run() <<<<");
            }
        };
        t.start();
    }

    // non-blocking, return 時不見得真的停了, 該在傳進來的 listener 中作.
    public void stop() {
        shouldStop.set(true);
        // 在此就不 t.join 了
    }

}