AudioRecord和AudioTrack延迟 [英] AudioRecord and AudioTrack latency

问题描述

我试图开发类似的IRIG为Android的aplication，所以第一步是捕捉麦克风输入，并在同一时间玩。

I'm trying to develop an aplication like iRig for android, so the first step is to capture the mic input and play it at the same time.

我有，但问题是，我得到一些延迟，使这个不可用的，如果我开始处理缓冲区恐怕会完全无法使用。

I have it, but the problem is that i get some latency that makes this unusable, and if I start processing the buffer i'm afraid it will get totally unusable.

我用audiorecord和audiotrack是这样的：

I use audiorecord and audiotrack like this:

    new Thread(new Runnable() {
        public void run() {
            while(mRunning){
                mRecorder.read(mBuffer, 0, mBufferSize);
                //Todo: Apply filters here into the buffer and then play it modified
                mPlayer.write(mBuffer, 0, mBufferSize);         
                //Log.v("MY AMP","ARA");
            }

而inicialization是这样的：

And the inicialization this way:

// ==================== INITIALIZE ========================= //
public void initialize(){

    mBufferSize = AudioRecord.getMinBufferSize(mHz, 
                AudioFormat.CHANNEL_CONFIGURATION_MONO, 
                AudioFormat.ENCODING_PCM_16BIT);

    mBufferSize2 = AudioTrack.getMinBufferSize(mHz, 
                AudioFormat.CHANNEL_CONFIGURATION_MONO, 
                AudioFormat.ENCODING_PCM_16BIT);

    mBuffer = new byte[mBufferSize];

    Log.v("MY AMP","Buffer size:" + mBufferSize);

    mRecorder = new AudioRecord(MediaRecorder.AudioSource.MIC, 
                mHz,
                AudioFormat.CHANNEL_CONFIGURATION_MONO,
                AudioFormat.ENCODING_PCM_16BIT, 
                mBufferSize);

    mPlayer = new AudioTrack(AudioManager.STREAM_MUSIC,
                mHz,
                AudioFormat.CHANNEL_CONFIGURATION_MONO,
                AudioFormat.ENCODING_PCM_16BIT,
                mBufferSize2, 
                AudioTrack.MODE_STREAM);    

}

你知道如何获得更快的响应？ 谢谢！

do you know how to get a faster response? Thanks!

推荐答案

Android的AudioTrack \ AudioRecord类具有很高的延迟，由于最小缓冲区大小。 之所以这些缓冲区大小是尽量减少降低时GC的根据谷歌（这在我看来是一个错误的决定，您可以优化自己的内存管理）的发生。

Android's AudioTrack\AudioRecord classes have high latency due to minimum buffer sizes. The reason for those buffer sizes is to minimize drops when GC's occur according to Google (which is a wrong decision in my opinion, you can optimize your own memory management).

您想要做的是使用OpenSL，可从2.3。它包含了音频流本地API。 下面是一些文档： <一href="http://mobilepearls.com/labs/native-android-api/opensles/index.html">http://mobilepearls.com/labs/native-android-api/opensles/index.html

What you want to do is use OpenSL, which is available from 2.3. It contains native APIs for streaming audio. Here's some docs: http://mobilepearls.com/labs/native-android-api/opensles/index.html

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