Voice Detection in Android Application

Description


My application record sound from phone microphone. I am using android standard classes (android.media.AudioRecord) to do that. Application have 2 buttons "Start" and "Stop" when I press start button application start record and when I press stop application stops recording and give me back buffer, with voice data in .wav format. All work fine.

Problem


I want to change my application in this way, when application start work it begin to analyze sound coming from microphone, if user keep silent application continue to analyze income sound from microphone, if user start to talk application begin to Record sound coming from microphone and then user finish talking application stops recording and give me back same buffer, with voice data in .wav format.

Question


  1. How I can detect that user start to talk
  2. How I can detect that user stop to talk

Answers


Just add this code to your application and you will detect when user start to speak and when he stops.

public void onCreate(Bundle savedInstanceState) 
    {
        super.onCreate(savedInstanceState);
        setContentView(R.layout.main);

    // Get the minimum buffer size required for the successful creation of an AudioRecord object. 
    int bufferSizeInBytes = AudioRecord.getMinBufferSize( RECORDER_SAMPLERATE,
                                                          RECORDER_CHANNELS,
                                                          RECORDER_AUDIO_ENCODING
                                                         ); 
    // Initialize Audio Recorder.
    AudioRecord audioRecorder = new AudioRecord( MediaRecorder.AudioSource.MIC,
                                                 RECORDER_SAMPLERATE,
                                                 RECORDER_CHANNELS,
                                                 RECORDER_AUDIO_ENCODING,
                                                 bufferSizeInBytes
                                                );
    // Start Recording.
    audioRecorder.startRecording();

    int numberOfReadBytes   = 0; 
    byte audioBuffer[]      = new  byte[bufferSizeInBytes];
    boolean recording       = false;
    float tempFloatBuffer[] = new float[3];
    int tempIndex           = 0;
    int totalReadBytes      = 0;
    byte totalByteBuffer[]  = new byte[60 * 44100 * 2];


    // While data come from microphone. 
    while( true )
    {
        float totalAbsValue = 0.0f;
        short sample        = 0; 

        numberOfReadBytes = audioRecorder.read( audioBuffer, 0, bufferSizeInBytes );

        // Analyze Sound.
        for( int i=0; i<bufferSizeInBytes; i+=2 ) 
        {
            sample = (short)( (audioBuffer[i]) | audioBuffer[i + 1] << 8 );
            totalAbsValue += Math.abs( sample ) / (numberOfReadBytes/2);
        }

        // Analyze temp buffer.
        tempFloatBuffer[tempIndex%3] = totalAbsValue;
        float temp                   = 0.0f;
        for( int i=0; i<3; ++i )
            temp += tempFloatBuffer[i];

        if( (temp >=0 && temp <= 350) && recording == false )
        {
            Log.i("TAG", "1");
            tempIndex++;
            continue;
        }

        if( temp > 350 && recording == false )
        {
            Log.i("TAG", "2");
            recording = true;
        }

        if( (temp >= 0 && temp <= 350) && recording == true )
        {
            Log.i("TAG", "Save audio to file.");

            // Save audio to file.
            String filepath = Environment.getExternalStorageDirectory().getPath();
            File file = new File(filepath,"AudioRecorder");
            if( !file.exists() )
                file.mkdirs();

            String fn = file.getAbsolutePath() + "/" + System.currentTimeMillis() + ".wav";

            long totalAudioLen  = 0;
            long totalDataLen   = totalAudioLen + 36;
            long longSampleRate = RECORDER_SAMPLERATE;
            int channels        = 1;
            long byteRate       = RECORDER_BPP * RECORDER_SAMPLERATE * channels/8;
            totalAudioLen       = totalReadBytes;
            totalDataLen        = totalAudioLen + 36;
            byte finalBuffer[]  = new byte[totalReadBytes + 44];

            finalBuffer[0] = 'R';  // RIFF/WAVE header
            finalBuffer[1] = 'I';
            finalBuffer[2] = 'F';
            finalBuffer[3] = 'F';
            finalBuffer[4] = (byte) (totalDataLen & 0xff);
            finalBuffer[5] = (byte) ((totalDataLen >> 8) & 0xff);
            finalBuffer[6] = (byte) ((totalDataLen >> 16) & 0xff);
            finalBuffer[7] = (byte) ((totalDataLen >> 24) & 0xff);
            finalBuffer[8] = 'W';
            finalBuffer[9] = 'A';
            finalBuffer[10] = 'V';
            finalBuffer[11] = 'E';
            finalBuffer[12] = 'f';  // 'fmt ' chunk
            finalBuffer[13] = 'm';
            finalBuffer[14] = 't';
            finalBuffer[15] = ' ';
            finalBuffer[16] = 16;  // 4 bytes: size of 'fmt ' chunk
            finalBuffer[17] = 0;
            finalBuffer[18] = 0;
            finalBuffer[19] = 0;
            finalBuffer[20] = 1;  // format = 1
            finalBuffer[21] = 0;
            finalBuffer[22] = (byte) channels;
            finalBuffer[23] = 0;
            finalBuffer[24] = (byte) (longSampleRate & 0xff);
            finalBuffer[25] = (byte) ((longSampleRate >> 8) & 0xff);
            finalBuffer[26] = (byte) ((longSampleRate >> 16) & 0xff);
            finalBuffer[27] = (byte) ((longSampleRate >> 24) & 0xff);
            finalBuffer[28] = (byte) (byteRate & 0xff);
            finalBuffer[29] = (byte) ((byteRate >> 8) & 0xff);
            finalBuffer[30] = (byte) ((byteRate >> 16) & 0xff);
            finalBuffer[31] = (byte) ((byteRate >> 24) & 0xff);
            finalBuffer[32] = (byte) (2 * 16 / 8);  // block align
            finalBuffer[33] = 0;
            finalBuffer[34] = RECORDER_BPP;  // bits per sample
            finalBuffer[35] = 0;
            finalBuffer[36] = 'd';
            finalBuffer[37] = 'a';
            finalBuffer[38] = 't';
            finalBuffer[39] = 'a';
            finalBuffer[40] = (byte) (totalAudioLen & 0xff);
            finalBuffer[41] = (byte) ((totalAudioLen >> 8) & 0xff);
            finalBuffer[42] = (byte) ((totalAudioLen >> 16) & 0xff);
            finalBuffer[43] = (byte) ((totalAudioLen >> 24) & 0xff);

            for( int i=0; i<totalReadBytes; ++i )
                finalBuffer[44+i] = totalByteBuffer[i];

            FileOutputStream out;
            try {
                out = new FileOutputStream(fn);
                 try {
                        out.write(finalBuffer);
                        out.close();
                    } catch (IOException e) {
                        // TODO Auto-generated catch block
                        e.printStackTrace();
                    }

            } catch (FileNotFoundException e1) {
                // TODO Auto-generated catch block
                e1.printStackTrace();
            }

            //*/
            tempIndex++;
            break;
        }

        // -> Recording sound here.
        Log.i( "TAG", "Recording Sound." );
        for( int i=0; i<numberOfReadBytes; i++ )
            totalByteBuffer[totalReadBytes + i] = audioBuffer[i];
        totalReadBytes += numberOfReadBytes;
        //*/

        tempIndex++;

    }
}

Check this link.


I've answered a similar question before, an approach is to use threads on recording and the speech power analyzing process on the recorded bytes, there's a sample code for your reference: http://musicg.googlecode.com/files/musicg_android_demo.zip


it is better to used private static final int RECORDER_SAMPLERAT=8000; it works for me.I think it will help for you


when i replaced totalAbsValue += Math.abs( sample ) / (numberOfReadBytes/2) by totalAbsValue += (float)Math.abs( sample ) / ((float)numberOfReadBytes/(float)2) then it work fine.


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