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enum Mode { // Adaptive mode intended for use if an analog volume control is available // on the capture device. It will require the user to provide coupling // between the OS mixer controls and AGC through the |stream_analog_level()| // functions. // // It consists of an analog gain prescription for the audio device and a // digital compression stage. kAdaptiveAnalog, // Adaptive mode intended for situations in which an analog volume control // is unavailable. It operates in a similar fashion to the adaptive analog // mode, but with scaling instead applied in the digital domain. As with // the analog mode, it additionally uses a digital compression stage. kAdaptiveDigital, // Fixed mode which enables only the digital compression stage also used by // the two adaptive modes. // // It is distinguished from the adaptive modes by considering only a // short time-window of the input signal. It applies a fixed gain through // most of the input level range, and compresses (gradually reduces gain // with increasing level) the input signal at higher levels. This mode is // preferred on embedded devices where the capture signal level is // predictable, so that a known gain can be applied. kFixedDigital }; 以下摘自于
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AGC parameters:
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1. Target level DBFS
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According to the description of webrtc:
Change of this parameter will set the target peak |level| (or envelope) of the AGC in dBFs (decibels from digital full-scale).
The convention is to use positive values.
For instance, passing in a value of 3 corresponds to -3 dBFs, or a target level 3 dB below full-scale.
Value range: limited to [0, 31].
TODO(ajm): use a negative value here instead, if/when VoE will similarly update its interface.
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2. Compression gain DB
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Sets the maximum |gain| the digital compression stage may apply, in dB. A higher number corresponds to greater compression, while a value of 0 will leave the signal uncompressed.
Value range: limited to [0, 90]
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3. Enable limiter or not
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When enabled, the compression stage will hard limit the signal to the target level. Otherwise, the signal will be compressed but not limited above the target level.
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4. Analog level limits
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Sets the |minimum| and |maximum| analog levels of the audio capture device.
Must be set if and only if an analog mode is used.
Value range: limited to [0, 65535].
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5. Stream saturated or not
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Returns true if the AGC has detected a saturation event (period where the signal reaches digital full-scale) in the current frame and the analog level cannot be reduced.
This could be used as an indicator to reduce or disable analog mic gain at the audio HAL.
以上摘自于
由于不同版本的代码差距较大，首先声明下面使用例子来自于下面的commit
使用自动增益控制放大的音量比较有限，所以使用了固定增益模式
int agcTest2(char *filename, char *outfilename, int mode){ //init agc void *agcInst = WebRtcAgc_Create(); int minLevel = 0; int maxLevel = 255; int agcMode = kAgcModeFixedDigital; int fs = 16000; int status = 0; status = WebRtcAgc_Init(agcInst, minLevel, maxLevel, agcMode, fs); if(status != 0) { printf("failed in WebRtcAgc_Init\n"); return -1; } WebRtcAgcConfig agcConfig; agcConfig.compressionGaindB = 20;//在Fixed模式下，越大声音越大 agcConfig.limiterEnable = 1; agcConfig.targetLevelDbfs = 3; //dbfs表示相对于full scale的下降值，0表示full scale，越小声音越大 status = WebRtcAgc_set_config(agcInst, agcConfig); if(status != 0) { printf("failed in WebRtcAgc_set_config\n"); return -1; } //alloc FILE *infp=fopen(filename,"r"); int nBands = 1; int frameSize = 160;//10ms对应于160个short short **pData = (short**)malloc(nBands*sizeof(short*)); pData[0] = (short*)malloc(frameSize*sizeof(short)); short **pOutData = (short**)malloc(nBands*sizeof(short*)); pOutData[0] = (short*)malloc(frameSize*sizeof(short)); //process FILE *outfp = fopen(outfilename,"w"); int len = frameSize; int micLevelIn = 0; int micLevelOut = 0; while(len > 0) { memset(pData[0], 0, frameSize*sizeof(short)); len = fread(pData[0], sizeof(short), frameSize, infp); int inMicLevel = micLevelOut; int outMicLevel = 0; uint8_t saturationWarning; status = WebRtcAgc_Process(agcInst, pData, nBands, frameSize, pOutData, inMicLevel, &outMicLevel, 0, &saturationWarning); if (status != 0) { printf("failed in WebRtcAgc_Process\n"); return -1; } if (saturationWarning == 1) { printf("saturationWarning\n"); } micLevelIn = outMicLevel; //write file len = fwrite(pOutData[0], sizeof(short), len, outfp); } fclose(infp); fclose(outfp); WebRtcAgc_Free(agcInst); free(pData[0]); free(pData); free(pOutData[0]); free(pOutData); 代码使用参考：webrtc/modules/audio_processing/gain_control_impl.cc
经过google邮件和论坛考证，确定没有公布出相关的参考文献和原理说明文档。
想不通为什么代码都开源了，好歹说几个参考文献啊！源码也没怎么看，大致看了下注释，所以我在使用过程中，参数的含义和使用基本上靠猜和试验，后续有时间读几篇ns的文献再来细聊，有什么错误麻烦指正！
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