Android 单独抽取 WebRtc-AGC（音频增益）模块

Android 单独抽取 WebRtc-AGC 模块，封装好JNI层，并且ndk-build出so库。

项目地址：https://github.com/wangzhengcheng1994/android-webrtc-agc
(欢迎star，谢谢)

先看下效果图：

AGC前：

Android 单独抽取 WebRtc-AGC（音频增益）模块_第1张图片

AGC后：

Android 单独抽取 WebRtc-AGC（音频增益）模块_第2张图片

其实也可以用来衰减：

Android 单独抽取 WebRtc-AGC（音频增益）模块_第3张图片

Android层调用（部分代码）：

 try{            AgcUtils agcUtils = new AgcUtils();            agcUtils.setAgcConfig(3,1,20).prepare();            FileInputStream fInt = new FileInputStream(Environment.getExternalStorageDirectory().getAbsolutePath() + "/agc-input-test.pcm");            FileOutputStream fOut = new FileOutputStream(Environment.getExternalStorageDirectory().getAbsolutePath() +"/agc-out-test.pcm");            byte[] buffer = new byte[160];            int bytes;            while((bytes = fInt.read(buffer)) != -1){                short[] data = new short[80];                short[] outData = new short[80];                short[] processData = new short[80];                ByteBuffer.wrap(buffer).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(data);                                 int agcProcessStatus = agcUtils.agcProcess(data,0,80,outData,0,micOutLevel,0,0);                Log.e(TAG, "return value" + agcProcessStatus);                fOut.write(shortArrayToByteArry(outData));            }            fInt.close();            fOut.close();        }catch (Exception e){            e.printStackTrace();        }

agcUtils 是对native方法的一个封装，其核心处理方法agcProcess（short[] inNear,int num_bands,int samples,short[] out,int inMicLevel,int outMicLevel,int echo,int saturationWarning），在webRtc源码的头文件中如下：

/* * This function processes a 10/20ms frame and adjusts (normalizes) the gain * both analog and digitally. The gain adjustments are done only during * active periods of speech. The input speech length can be either 10ms or * 20ms and the output is of the same length. The length of the speech * vectors must be given in samples (80/160 when FS=8000, and 160/320 when * FS=16000 or FS=32000). The echo parameter can be used to ensure the AGC will * not adjust upward in the presence of echo. * * This function should be called after processing the near-end microphone * signal, in any case after any echo cancellation. * * Input: *      - agcInst           : AGC instance *      - inNear            : Near-end input speech vector (10 or 20 ms) for *                            L band *      - inNear_H          : Near-end input speech vector (10 or 20 ms) for *                            H band *      - samples           : Number of samples in input/output vector *      - inMicLevel        : Current microphone volume level *      - echo              : Set to 0 if the signal passed to add_mic is *                            almost certainly free of echo; otherwise set *                            to 1. If you have no information regarding echo *                            set to 0. * * Output: *      - outMicLevel       : Adjusted microphone volume level *      - out               : Gain-adjusted near-end speech vector (L band) *                          : May be the same vector as the input. *      - out_H             : Gain-adjusted near-end speech vector (H band) *      - saturationWarning : A returned value of 1 indicates a saturation event *                            has occurred and the volume cannot be further *                            reduced. Otherwise will be set to 0. * * Return value: *                          :  0 - Normal operation. *                          : -1 - Error */int WebRtcAgc_Process(void* agcInst,                      const int16_t* inNear,                      const int16_t* inNear_H,                      int16_t samples,                      int16_t* out,                      int16_t* out_H,                      int32_t inMicLevel,                      int32_t* outMicLevel,                      int16_t echo,                      uint8_t* saturationWarning);

参数比较多，具体传参可参考上面的android层代码。

**agcUtils.setAgcConfig(3,1,20)**此方法在头文件中定义如下：

/* * This function sets the config parameters (targetLevelDbfs, * compressionGaindB and limiterEnable). * * Input: *      - agcInst           : AGC instance *      - config            : config struct * * Output: * * Return value: *                          :  0 - Normal operation. *                          : -1 - Error */int WebRtcAgc_set_config(void* agcInst, WebRtcAgc_config_t config);**解释下WebRtcAgc_config_t中的变量含义**typedef struct{    int16_t targetLevelDbfs;   // default 3 (-3 dBOv)    int16_t compressionGaindB; // default 9 dB    uint8_t limiterEnable;     // default kAgcTrue (on)} WebRtcAgc_config_t;

上面的效果图，有增益有衰减，其实现就是通过WebRtcAgc_set_config方法实现的

compressionGaindB （上面设置的是20），在kAgcModeFixedDigital模式下，越大声音越大；
targetLevelDbfs （上面设置的是3），0表示full scale，越小声音越大，越大声音越小，（笔者测试发现最大只能设置到30）
limiterEnable（上面设置的是1），默认设置1就行，是一个开关。

WebRtc源码中的NS模块位于 src\modules\audio_processing\agc，另外和agc同级目录下，还有一个agc2目录，没有做深层研究。AGC最外层的头文件是gain_control.h

原创：VampireHunter

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