在一般的VOIP软件或视频会议系统中，假设我们只有A和B两个人在通话，首先，A的声音传给B，B然后用喇叭放出来，而这时B的MIC呢则会采集到喇叭放出来的声音，然后传回给A，如果这个传输的过程中时延足够大，A就会听到一个和自己刚才说过的话一样的声音，这就是回声，声学回声消除器的作用就是在B端对B采集到的声音进行处理，把采集到声音包含的A的声音去掉再传给A，这样，A就不会听到自己说过的话了。
声学回声消除的原理我就不说了，这在网上有很多文档，网上缺少的是实现，所以，我在这把一个开源的声学回声消除器介绍一下，希望对有些有人用，如果有人知道怎么把这消除器用的基于实时流的VOIP软件中，希望能一起分享一下。
这个声学回声消除器是一个著名的音频编解码器speex中的一部分，1.1.9版本后的回声消除器才起作用，以前版本的都不行，我用的也是这个版本，测试表明，用同一个模拟文件，它有效果比INTEL IPP库4.1版中的声学回声消除器的还要好。
先说编译。首先，从www.speex.org上下载speex1.1.9的源代码，解压，打开speex/win32/libspeex中的libspeex.dsw，这个工作区里有两个工程，一个是libspeex，另一个是libspeex_dynamic。然后，将libspeex中的mdf.c文件添加到工程libspeex中，编译即可。
以下是我根据文档封装的一个类，里面有一个测试程序：
//file name: speexEC.h
#ifndef SPEEX_EC_H#define SPEEX_EC_H#include <stdio.h>#include <stdlib.h>#include "speex/speex_echo.h"#include "speex/speex_preprocess.h" class CSpeexEC{public:CSpeexEC();~CSpeexEC();void Init(int frame_size=160, int filter_length=1280, int sampling_rate=8000); void DoAEC(short *mic, short *ref, short *out);protected:void Reset();private:bool m_bHasInit;SpeexEchoState* m_pState; SpeexPreprocessState* m_pPreprocessorState;int m_nFrameSize;int m_nFilterLen;int m_nSampleRate;float* m_pfNoise;};#endif
//fine name:speexEC.cpp
#include "SpeexEC.h"CSpeexEC::CSpeexEC(){m_bHasInit = false;m_pState = NULL;m_pPreprocessorState = NULL;m_nFrameSize = 160;m_nFilterLen = 160*8;m_nSampleRate = 8000;m_pfNoise = NULL;}CSpeexEC::~CSpeexEC(){Reset();}void CSpeexEC::Init(int frame_size, int filter_length, int sampling_rate){Reset(); if (frame_size<=0 || filter_length<=0 || sampling_rate<=0){ m_nFrameSize =160; m_nFilterLen = 160*8; m_nSampleRate = 8000;}else{ m_nFrameSize =frame_size; m_nFilterLen = filter_length; m_nSampleRate = sampling_rate;}m_pState = speex_echo_state_init(m_nFrameSize, m_nFilterLen);m_pPreprocessorState = speex_preprocess_state_init(m_nFrameSize, m_nSampleRate);m_pfNoise = new float[m_nFrameSize+1];m_bHasInit = true;}void CSpeexEC::Reset(){if (m_pState != NULL){ speex_echo_state_destroy(m_pState); m_pState = NULL;}if (m_pPreprocessorState != NULL){ speex_preprocess_state_destroy(m_pPreprocessorState); m_pPreprocessorState = NULL;}if (m_pfNoise != NULL){ delete []m_pfNoise; m_pfNoise = NULL;}m_bHasInit = false;}void CSpeexEC:DoAEC(short* mic, short* ref, short* out){if (!m_bHasInit) return;speex_echo_cancel(m_pState, mic, ref, out, m_pfNoise); speex_preprocess(m_pPreprocessorState, (__int16 *)out, m_pfNoise); }
可以看出，这个回声消除器类很简单，只要初始化一下就可以调用了。但是，要注意的是，传给回声消除器的两个声音信号，必须同步得非常的好，就是说，在B端，接收到A说的话以后，要把这些话音数据传给回声消除器做参考，然后再传给声卡，声卡再放出来，这有一段延时，这时，B再采集，然后传给回声消除器，与那个参考数据比较，从采集到的数据中把频域和参考数据相同的部分消除掉。如果传给消除器的两个信号同步得不好，即两个信号找不到频域相同的部分，就没有办法进行消除了。
测试程序：
#define NN 160void main(){FILE* ref_fd, *mic_fd, *out_fd;short ref[NN], mic[NN], out[NN];ref_fd = fopen ("ref.pcm", "rb"); //打开参考文件，即要消除的声音mic_fd = fopen ("mic.pcm", "rb");//打开mic采集到的声音文件，包含回声在里面out_fd = fopen ("echo.pcm", "wb");//消除了回声以后的文件CSpeexEC ec;ec.Init();while (fread(mic, 1, NN*2, mic_fd)) { fread(ref, 1, NN*2, ref_fd); ec.DoAEC(mic, ref, out); fwrite(out, 1, NN*2, out_fd); } fclose(ref_fd); fclose(mic_fd); fclose(out_fd);}
以上的程序是用文件来模拟回声和MIC，但在实时流中是大不一样的，在一般的VOIP软件中，接收对方的声音并传到声卡中播放是在一个线程中进行的，而采集本地的声音并传送到对方又是在另一个线程中进行的，而声学回声消除器在对采集到的声音进行回声消除的同时，还需要播放线程中的数据作为参考，而要同步这两个线程中的数据是非常困难的，因为稍稍有些不同步，声学回声消除器中的自适应滤波器就会发散，不但消除不了回声，还会破坏原始采集到的声音，使被破坏的声音难以分辨。我做过好多尝试，始终无法用软件来实现对这两个线程中的数据进行同步，导致实现失败，希望有经验的网友们一起分享一下这方面的经验。
示例代码：
Sample code
This section shows sample code for encoding and decoding speech using the Speex API. The commands can be used to encode and decode a file by calling:
% sampleenc in_file.sw | sampledec out_file.sw
where both files are raw (no header) files encoded at 16 bits per sample (in the machine natural endianness).
sampleenc.c
sampleenc takes a raw 16 bits/sample file, encodes it and outputs a Speex stream to stdout. Note that the packing used is NOT compatible with that of speexenc/speexdec.
#include <speex/speex.h>#include <stdio.h>/*The frame size in hardcoded for this sample code but it doesn't have to be*/#define FRAME_SIZE 160int main(int argc, char **argv){char *inFile;FILE *fin;short in[FRAME_SIZE];float input[FRAME_SIZE];char cbits[200];int nbBytes;/*Holds the state of the encoder*/void *state;/*Holds bits so they can be read and written to by the Speex routines*/SpeexBits bits;int i, tmp;/*Create a new encoder state in narrowband mode*/state = speex_encoder_init(&speex_nb_mode);/*Set the quality to 8 (15 kbps)*/tmp=8;speex_encoder_ctl(state, SPEEX_SET_QUALITY, &tmp);inFile = argv[1];fin = fopen(inFile, "r");/*Initialization of the structure that holds the bits*/speex_bits_init(&bits);while (1){/*Read a 16 bits/sample audio frame*/fread(in, sizeof(short), FRAME_SIZE, fin);if (feof(fin))break;/*Copy the 16 bits values to float so Speex can work on them*/for (i=0;i<FRAME_SIZE;i++)input[i]=in[i];/*Flush all the bits in the struct so we can encode a new frame*/speex_bits_reset(&bits);/*Encode the frame*/speex_encode(state, input, &bits);/*Copy the bits to an array of char that can be written*/nbBytes = speex_bits_write(&bits, cbits, 200);/*Write the size of the frame first. This is what sampledec expects butit's likely to be different in your own application*/fwrite(&nbBytes, sizeof(int), 1, stdout);/*Write the compressed data*/fwrite(cbits, 1, nbBytes, stdout);}/*Destroy the encoder state*/speex_encoder_destroy(state);/*Destroy the bit-packing struct*/speex_bits_destroy(&bits);fclose(fin);return 0;}
sampledec.c
sampledec reads a Speex stream from stdin, decodes it and outputs it to a raw 16 bits/sample file. Note that the packing used is NOT compatible with that of speexenc/speexdec.
#include <speex/speex.h>#include <stdio.h>/*The frame size in hardcoded for this sample code but it doesn't have to be*/#define FRAME_SIZE 160int main(int argc, char **argv){char *outFile;FILE *fout;/*Holds the audio that will be written to file (16 bits per sample)*/short out[FRAME_SIZE];/*Speex handle samples as float, so we need an array of floats*/float output[FRAME_SIZE];char cbits[200];int nbBytes;/*Holds the state of the decoder*/void *state;/*Holds bits so they can be read and written to by the Speex routines*/SpeexBits bits;int i, tmp;/*Create a new decoder state in narrowband mode*/state = speex_decoder_init(&speex_nb_mode);/*Set the perceptual enhancement on*/tmp=1;speex_decoder_ctl(state, SPEEX_SET_ENH, &tmp);outFile = argv[1];fout = fopen(outFile, "w");/*Initialization of the structure that holds the bits*/speex_bits_init(&bits);while (1){/*Read the size encoded by sampleenc, this part will likely bedifferent in your application*/fread(&nbBytes, sizeof(int), 1, stdin);fprintf (stderr, "nbBytes: %d/n", nbBytes);if (feof(stdin))break;/*Read the "packet" encoded by sampleenc*/fread(cbits, 1, nbBytes, stdin);/*Copy the data into the bit-stream struct*/speex_bits_read_from(&bits, cbits, nbBytes);/*Decode the data*/speex_decode(state, &bits, output);/*Copy from float to short (16 bits) for output*/for (i=0;i<FRAME_SIZE;i++)out[i]=output[i];/*Write the decoded audio to file*/fwrite(out, sizeof(short), FRAME_SIZE, fout);}/*Destroy the decoder state*/speex_decoder_destroy(state);/*Destroy the bit-stream truct*/speex_bits_destroy(&bits);fclose(fout);return 0;}