音頻自動增益與靜音檢測算法附完整C代碼

前面分享過一個算法《音頻增益響度分析 ReplayGain 附完整C代碼示例》

主要用于評估一定長度音頻的音量強度，

而分析之后，很多類似的需求，肯定是做音頻增益，提高音量諸如此類做法。

不過在項目實測的時候，其實真的很難定標準，

到底在什么樣的環境下，要增大音量，還是降低。

在通訊行業一般的做法就是采用靜音檢測，

一旦檢測為靜音或者噪音，則不做處理，反之通過一定的策略進行處理。

這里就涉及到兩個算法，一個是靜音檢測，一個是音頻增益。

增益其實沒什么好說的，類似于數據歸一化拉伸的做法。

靜音檢測?在WebRTC中?是采用計算GMM (Gaussian Mixture Model,高斯混合模型)進行特征提取的。

在很長一段時間里面，音頻特征?有3個主要的方法，

GMM? ,Spectrogram (聲譜圖), MFCC?即 Mel-Frequency Cepstrum(Mel頻率倒譜)

恕我直言,GMM?提取的特征，其魯棒性不如后兩者。

也不多做介紹，感興趣的同學，翻翻維基百科 ,補補課。

當然在實際使用算法時，會由此延伸出來一些小技巧。

例如，用靜音檢測?來做音頻裁剪，或者搭配音頻增益做一些音頻增強之類的操作。

自動增益在WebRTC?源代碼文件是:analog_agc.c?和?digital_agc.c

靜音檢測?源代碼文件是: webrtc_vad.c

這個命名，有一定的歷史原因了。

經過梳理后，

增益算法為 agc.c agc.h

靜音檢測為 vad.c vad.h

增益算法的完整示例代碼：

#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
//采用https://github.com/mackron/dr_libs/blob/master/dr_wav.h 解碼
#define DR_WAV_IMPLEMENTATION
#include "dr_wav.h"
#include "agc.h"#ifndef nullptr
#define nullptr 0
#endif#ifndef MIN
#define  MIN(A, B)        ((A) < (B) ? (A) : (B))
#endif//寫wav文件
void wavWrite_int16(char *filename, int16_t *buffer, size_t sampleRate, size_t totalSampleCount) {drwav_data_format format = {};format.container = drwav_container_riff;     // <-- drwav_container_riff = normal WAV files, drwav_container_w64 = Sony Wave64.format.format = DR_WAVE_FORMAT_PCM;          // <-- Any of the DR_WAVE_FORMAT_* codes.format.channels = 1;format.sampleRate = (drwav_uint32) sampleRate;format.bitsPerSample = 16;drwav *pWav = drwav_open_file_write(filename, &format);if (pWav) {drwav_uint64 samplesWritten = drwav_write(pWav, totalSampleCount, buffer);drwav_uninit(pWav);if (samplesWritten != totalSampleCount) {fprintf(stderr, "ERROR\n");exit(1);}}
}//讀取wav文件
int16_t *wavRead_int16(char *filename, uint32_t *sampleRate, uint64_t *totalSampleCount) {unsigned int channels;int16_t *buffer = drwav_open_and_read_file_s16(filename, &channels, sampleRate, totalSampleCount);if (buffer == nullptr) {printf("讀取wav文件失敗.");}//僅僅處理單通道音頻if (channels != 1) {drwav_free(buffer);buffer = nullptr;*sampleRate = 0;*totalSampleCount = 0;}return buffer;
}//分割路徑函數
void splitpath(const char *path, char *drv, char *dir, char *name, char *ext) {const char *end;const char *p;const char *s;if (path[0] && path[1] == ':') {if (drv) {*drv++ = *path++;*drv++ = *path++;*drv = '\0';}} else if (drv)*drv = '\0';for (end = path; *end && *end != ':';)end++;for (p = end; p > path && *--p != '\\' && *p != '/';)if (*p == '.') {end = p;break;}if (ext)for (s = end; (*ext = *s++);)ext++;for (p = end; p > path;)if (*--p == '\\' || *p == '/') {p++;break;}if (name) {for (s = p; s < end;)*name++ = *s++;*name = '\0';}if (dir) {for (s = path; s < p;)*dir++ = *s++;*dir = '\0';}
}int agcProcess(int16_t *buffer, uint32_t sampleRate, size_t samplesCount, int16_t agcMode) {if (buffer == nullptr) return -1;if (samplesCount == 0) return -1;WebRtcAgcConfig agcConfig;agcConfig.compressionGaindB = 9; // default 9 dBagcConfig.limiterEnable = 1; // default kAgcTrue (on)agcConfig.targetLevelDbfs = 3; // default 3 (-3 dBOv)int minLevel = 0;int maxLevel = 255;size_t samples = MIN(160, sampleRate / 100);if (samples == 0) return -1;const int maxSamples = 320;int16_t *input = buffer;size_t nTotal = (samplesCount / samples);void *agcInst = WebRtcAgc_Create();if (agcInst == NULL) return -1;int status = WebRtcAgc_Init(agcInst, minLevel, maxLevel, agcMode, sampleRate);if (status != 0) {printf("WebRtcAgc_Init fail\n");WebRtcAgc_Free(agcInst);return -1;}status = WebRtcAgc_set_config(agcInst, agcConfig);if (status != 0) {printf("WebRtcAgc_set_config fail\n");WebRtcAgc_Free(agcInst);return -1;}size_t num_bands = 1;int inMicLevel, outMicLevel = -1;int16_t out_buffer[maxSamples];int16_t *out16 = out_buffer;uint8_t saturationWarning = 1;                 //是否有溢出發生，增益放大以后的最大值超過了65536int16_t echo = 0;                                 //增益放大是否考慮回聲影響for (int i = 0; i < nTotal; i++) {inMicLevel = 0;int nAgcRet = WebRtcAgc_Process(agcInst, (const int16_t *const *) &input, num_bands, samples,(int16_t *const *) &out16, inMicLevel, &outMicLevel, echo,&saturationWarning);if (nAgcRet != 0) {printf("failed in WebRtcAgc_Process\n");WebRtcAgc_Free(agcInst);return -1;}memcpy(input, out_buffer, samples * sizeof(int16_t));input += samples;}WebRtcAgc_Free(agcInst);return 1;
}void auto_gain(char *in_file, char *out_file) {//音頻采樣率uint32_t sampleRate = 0;//總音頻采樣數uint64_t inSampleCount = 0;int16_t *inBuffer = wavRead_int16(in_file, &sampleRate, &inSampleCount);//如果加載成功if (inBuffer != nullptr) {//  kAgcModeAdaptiveAnalog  模擬音量調節//  kAgcModeAdaptiveDigital 自適應增益//  kAgcModeFixedDigital 固定增益
        agcProcess(inBuffer, sampleRate, inSampleCount, kAgcModeAdaptiveDigital);wavWrite_int16(out_file, inBuffer, sampleRate, inSampleCount);free(inBuffer);}
}int main(int argc, char *argv[]) {printf("WebRTC Automatic Gain Control\n");printf("博客:http://cpuimage.cnblogs.com/\n");printf("音頻自動增益\n");if (argc < 2)return -1;char *in_file = argv[1];char drive[3];char dir[256];char fname[256];char ext[256];char out_file[1024];splitpath(in_file, drive, dir, fname, ext);sprintf(out_file, "%s%s%s_out%s", drive, dir, fname, ext);auto_gain(in_file, out_file);printf("按任意鍵退出程序 \n");getchar();return 0;
}

?靜音檢測完整示例代碼：

#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
//采用https://github.com/mackron/dr_libs/blob/master/dr_wav.h 解碼
#define DR_WAV_IMPLEMENTATION#include "dr_wav.h"
#include "vad.h"#ifndef nullptr
#define nullptr 0
#endif#ifndef MIN
#define  MIN(A, B)        ((A) < (B) ? (A) : (B))
#endif#ifndef MAX
#define  MAX(A, B)        ((A) > (B) ? (A) : (B))
#endif//讀取wav文件
int16_t *wavRead_int16(char *filename, uint32_t *sampleRate, uint64_t *totalSampleCount) {unsigned int channels;int16_t *buffer = drwav_open_and_read_file_s16(filename, &channels, sampleRate, totalSampleCount);if (buffer == nullptr) {printf("讀取wav文件失敗.");}//僅僅處理單通道音頻if (channels != 1) {drwav_free(buffer);buffer = nullptr;*sampleRate = 0;*totalSampleCount = 0;}return buffer;
}int vadProcess(int16_t *buffer, uint32_t sampleRate, size_t samplesCount, int16_t vad_mode, int per_ms_frames) {if (buffer == nullptr) return -1;if (samplesCount == 0) return -1;// kValidRates : 8000, 16000, 32000, 48000// 10, 20 or 30 ms framesper_ms_frames = MAX(MIN(30, per_ms_frames), 10);size_t samples = sampleRate * per_ms_frames / 1000;if (samples == 0) return -1;int16_t *input = buffer;size_t nTotal = (samplesCount / samples);void *vadInst = WebRtcVad_Create();if (vadInst == NULL) return -1;int status = WebRtcVad_Init(vadInst);if (status != 0) {printf("WebRtcVad_Init fail\n");WebRtcVad_Free(vadInst);return -1;}status = WebRtcVad_set_mode(vadInst, vad_mode);if (status != 0) {printf("WebRtcVad_set_mode fail\n");WebRtcVad_Free(vadInst);return -1;}printf("Activity ： \n");for (int i = 0; i < nTotal; i++) {int nVadRet = WebRtcVad_Process(vadInst, sampleRate, input, samples);if (nVadRet == -1) {printf("failed in WebRtcVad_Process\n");WebRtcVad_Free(vadInst);return -1;} else {// output resultprintf(" %d \t", nVadRet);}input += samples;}printf("\n");WebRtcVad_Free(vadInst);return 1;
}void vad(char *in_file) {//音頻采樣率uint32_t sampleRate = 0;//總音頻采樣數uint64_t inSampleCount = 0;int16_t *inBuffer = wavRead_int16(in_file, &sampleRate, &inSampleCount);//如果加載成功if (inBuffer != nullptr) {//    Aggressiveness mode (0, 1, 2, or 3)int16_t mode = 1;int per_ms = 30;vadProcess(inBuffer, sampleRate, inSampleCount, mode, per_ms);free(inBuffer);}
}int main(int argc, char *argv[]) {printf("WebRTC Voice Activity Detector\n");printf("博客:http://cpuimage.cnblogs.com/\n");printf("靜音檢測\n");if (argc < 2)return -1;char *in_file = argv[1];vad(in_file);printf("按任意鍵退出程序 \n");getchar();return 0;
}