的Android AudioRecord MP3编码AudioFormat.CHANNEL_IN_STEREO

Question

我似乎这个问题被卡住，

我试图让 https://github.com/yhirano/SimpleLameLibForAndroid 对channelConfig AudioFormat.CHANNEL_IN_STEREO模式下工作。

下面的代码完美地工作，如果我用channelConfig = AudioFormat.CHANNEL_IN_MONO调用它，但不能用STEREO调用它。

我已经打得四处

short[] buffer = new short[mSampleRate * (16/8) * nChannels * 5] 
byte[] mp3buffer = new byte[(int) (7200 + buffer.length * 2 * 1.25)]; 

BU似乎无法得到它的工作。我的意思是它的工作原理，但录制的声音非常慢。听这个例子https://dl.dropboxusercontent.com/u/1465252/1381762795295.mp3

似乎还有另一个类似的问题在Lame encoded mp3 audio slowed down - Android没有解决方案。

任何人都可以帮忙吗？

下面是代码：

new Mp3Audio(MediaRecorder.AudioSource.MIC, 44100, AudioFormat.CHANNEL_IN_STEREO, A udioFormat.ENCODING_PCM_16BIT, 128); 


public Mp3Audio(int audioSource, int sampleRate, int channelConfig, int audioFormat, int bitRate) { 
    if (sampleRate <= 0) { 
     throw new InvalidParameterException(
       "Invalid sample rate specified."); 
    } 

    mSampleRate = sampleRate; 
    mBitRate = bitRate; 
    if (channelConfig == AudioFormat.CHANNEL_IN_MONO) { 
     nChannels = 1; 
    } else { 
     nChannels = 2; 
    } 
    builder = new Builder(mSampleRate, nChannels, mSampleRate, mBitRate); 
    //builder = new Builder(44100, 1, 44100, 128); 

    builder.quality(6); 

    mEncoder = builder.create(); 
    cAmplitude = 0; 
    payloadSize = 0; 
    aFormat = audioFormat; 
    aSource = audioSource; 
    mChannelConfig = channelConfig; 


} 
    public void start() { 
final int minBufferSize = AudioRecord.getMinBufferSize(mSampleRate, mChannelConfig, aFormat) * mBufferSizeFactor;  
      if (minBufferSize < 0) { 
       AppHelper.Log(tag, "MSG_ERROR_GET_MIN_BUFFERSIZE"); 
       return; 
      } 
      AppHelper.Log(tag, "minBufferSize: " +  AppHelper.humanReadableByteCount(minBufferSize, true)); 
      aRecorder = new AudioRecord(
        aSource, 
        mSampleRate, 
        mChannelConfig, 
        aFormat, 
        minBufferSize); 




      short[] buffer = new short[mSampleRate * (16/8) * nChannels * 5]; // SampleRate[Hz] * 16bit * Mono * 5sec 
      AppHelper.Log(tag, "buffer: " + AppHelper.humanReadableByteCount(buffer.length, true)); 
      byte[] mp3buffer = new byte[(int) (7200 + buffer.length * 2 * 1.25)]; 
      AppHelper.Log(tag, "mp3buffer: " + AppHelper.humanReadableByteCount(mp3buffer.length, true)); 

...... .......

Answer 1

为了给你一个指针，你需要调用lame_encode_buffer_interleaved（）如果你使用2个频道（.stereo）进行录制。

我花了几天的数字出来，这是你可以使用代码：

if (lame_get_num_channels(glf) == 2) 
    { 
     result = lame_encode_buffer_interleaved(glf, j_buffer_l, samples/2, j_mp3buf, mp3buf_size); 
    } 
    else 
    { 
     result = lame_encode_buffer(glf, j_buffer_l, j_buffer_r, samples, j_mp3buf, mp3buf_size); 
    }