嵌入式软件中的回声消除（AEC） [英] Acoustic Echo Cancellation (AEC) in embedded software

问题描述

我正在嵌入式设备上执行VoIP项目。我使用具有低级音频编解码器的32位MCU构建了示例。现在，我发现我的设备上存在回声问题，即我可以听到扬声器说的话。我进行了一些研究，发现大多数应用程序使用具有回声消除功能的DSP编解码器。但是，是否可以使用我的32位MCU在软件中进行声学回声消除？

I am doing a VoIP project on embedded device. I have built a sample using a 32bits MCU with a low grade audio codec. Now I found that there is echo issue on my device, that is I can hear what I said from the speaker. I have do some research and found that most appliaction use a DSP codec with acoustic echo cancellation feature. However, is it possible that I do the acoustic echo cancellation in the software, using my 32bits MCU?

您能否为该算法或甚至源代码：P辩护？做回声消除？我知道在MCU上不可能使用复杂的方法，但是也欢迎使用简单的算法。

Can you adive the algorithm, or even source code:P, for doing acoustic echo cancellation? I know sophisticated method is not possible on a MCU, whereas a simple algorithm is also welcomed.

谢谢

[跟进]：我已经尝试了一些AEC代码，但是它们在我的MCU中无法正常工作，可能是MCU功率的限制。我发现实施这些代码后，我的设备变为非实时的（但是VoIP需要实时响应）。最后，我增加了AEC芯片，从而实现了模拟硬件解决方案，因为我不想在另一个DSP芯片中再次编写代码。

[Follow up] : I have tried some AEC code but they can not work well in my MCU, probably it is the limit of the MCU power. I found that my device become non-real-time when implemented these codes (but a VoIP need a real-time respond). At last I implemented a analog hardware solution by adding an AEC chips, because I do not want to write the code again in another DSP chip.

推荐答案

我曾经有段时间使用回声消除功能。我写了一个软件电话，用户可以根据自己的喜好切换音频输入和输出设备。我尝试了Speex回声消除库，以及在网上找到的其他几个开源库。没有一个对我有效。我尝试了不同的扬声器/麦克风配置，并且回声始终以某种形式或方式存在。

I had a heck of a time with echo cancellation. I wrote a softphone, and the user can switch their audio input and output devices around to suit their fancy. I tried the Speex echo cancellation library, and several other open source libs I found online. None worked well for me. I tried different speaker/mike configuration and the echo was always there in some form or fashion.

我相信创建适用于所有人的AEC代码将非常困难。可能的扬声器配置/房间大小/背景噪音等最终，我坐下来，用这种算法为我的软件电话编写了自己的回声消除模块。

I believe it would be very hard to create AEC code that would work for all possible speaker configurations / room sizes / background noises..etc. Finally I sat down and wrote my own echo cancellation module for my softphone with this algorithm.

虽然有些粗糙，但是效果很好并且可靠。

It's somewhat crude, but it has worked well and is reliable.

变量1：
记录下您正在与之交谈的人说话时的平均振幅。 （不考虑静默时间）

variable1: Keep a record of what the average amplitude is of when the person to whom you're talking is speaking. (Don't factor quiet-time)

变量2：
记录输入（麦克风）的平均幅度，但仅当存在再次是声音-不要把安静的时间都考虑在内。

variable2: Keep a record of what the average amplitude is on the input (mike), but only when there is voice- again- don't factor quiet time.

一旦有音频播放，就切断话筒。并假设正在听的人没有说话，则在播放最后一个可听音频帧后，在150-300毫秒内打开麦克风。

As soon as there's audio to play- cut the mike. And assuming the person listening is not talking, turn the mike on 150-300ms after the last audible audio frame comes in to be played.

如果来自麦克风的音频（ （在播放过程中放下的音频）大于OH-（变量2 * 1.5），开始发送音频输入帧达指定的持续时间，每次输入幅度达到（variable2 * 1.5）时重置该持续时间。

If the audio from the microphones (that you're dropping during playback) is greater than oh- say (variable2 * 1.5), start sending the audio input frames for a specified duration, resetting that duration every time the input amplitude reaches (variable2 * 1.5).

这样，说话的人就会知道自己被打扰了，停下来看看别人在说什么。如果说话的人没有太多的背景噪音，他们可能会听到大部分（即使不是全部）干扰。

That way the person talking will know they are being interrupted, and stop to see what the person is saying. If the person talking doesn't have too noisy of a background, they will probably hear most if not all of the interruption.

就像我说的那样，不是最优雅的，但它不占用大量资源（CPU，内存），并且实际上工作得很好。我对我的声音感到非常满意。

Like I said, not the most graceful, but it doesn't use a lot of resources (CPU, memory) and it actually works pretty darn well. I am very pleased with how mine sounds.

要实现它，我做了一些功能。

To implement it, I just made a few functions.

在接收到的音频帧上，我调用一个函数，该函数称为：

On a received audio frame, I call a function I called:

void audioin( AEC *ec, short *frame ) {
    unsigned int tas=0; /* Total sum of all audio in frame (absolute value) */
    int i=0;
    for (;i<160;i++)
        tas+=ABS(frame[i]);
    tas/=160; /* 320 byte frames muLaw */
    if (tas>300) { /* I assume this is audiable */
        lockecho(ec);
        ec->lastaudibleframe=GetTickCount64();
        unlockecho(ec);
    }
    return;
}

在发送帧之前，我会做：

and before sending a frame, I do:

#define ECHO_THRESHOLD 300 /* Time to keep suppression alive after last audible frame */
#define ONE_MINUTE 3000 /* 3000 20ms samples */
#define AVG_PERIOD 250 /* 250 20ms samples */
#define ABS(x) (x>0?x:-x)


char removeecho( AEC *ec, short *aecinput ) {
    int tas=0; /* Average absolute amplitude in this signal */
    int i=0;
    unsigned long long *tot=0;
    unsigned int *ctr=0;
    unsigned short *avg=0;
    char suppressframe=0;
    lockecho(ec);
    if (ec->lastaudibleframe+ECHO_THRESHOLD > GetTickCount64() ) {
        /* If we're still within the threshold for echo (speaker state is ON) */
        tot=&ec->t_aiws;
        ctr=&ec->c_aiws;
        avg=&ec->aiws;
    } else {
        /* If we're outside the threshold for echo (speaker state is OFF) */
        tot=&ec->t_aiwos;
        ctr=&ec->c_aiwos;
        avg=&ec->aiwos;
    }
    for (;i<160;i++) {
        tas+=ABS(aecinput[i]);
    }
    tas/=160;
    if (tas>200) {
        (*tot)+=tas;
        (*avg)=(unsigned short)((*tot)/( (*ctr)?(*ctr):1));
        (*ctr)++;
        if ((*ctr)>AVG_PERIOD) {
            (*tot)=(*avg);
            (*ctr)=0;
        }
    }
    if ( (avg==&ec->aiws) ) {
        tas-=ec->aiwos;
        if (tas<0) {
            tas=0;
        }
        if ( ((unsigned short) tas > (ec->aiws*1.5)) && ((unsigned short)tas>=ec->aiwos) && (ec->aiwos!=0) ) {
            suppressframe=0;
        } else {
            suppressframe=1;
        }
    }
    if (suppressframe) { /* Silence frame */
        memset(aecinput, 0, 320);
    }
    unlockecho(ec);
    return suppressframe;
}

如果需要，它将使框架静音。我将所有变量（如计时器）和幅度平均值保留在AEC结构中，这是我从调用返回的

Which will silence the frame if it needs to. I keep all my variables, like the timers, and amplitude averages in the AEC struct, which I return from a call to

AEC *initecho( void ) {
    AEC *ec=0;
    ec=(AEC *)malloc(sizeof(AEC));
    memset(ec, 0, sizeof(AEC));
    ec->aiws=200; /* Just a default guess as to what the average amplitude would be */
    return ec;
}





typedef struct aec {
    unsigned long long lastaudibleframe; /* time stamp of last audible frame */
    unsigned short aiws; /* Average mike input when speaker is playing */
    unsigned short aiwos; /*Average mike input when speaker ISNT playing */
    unsigned long long t_aiws, t_aiwos; /* Internal running total (sum of PCM) */
    unsigned int c_aiws, c_aiwos; /* Internal counters for number of frames for     averaging */
    unsigned long lockthreadid; /* Thread ID with lock */
    int stlc; /* Same thread lock-count */
} AEC;

您可以根据需要进行调整并付诸实践，但是就像我说的那样。实际上听起来不错。我唯一的问题是背景噪音是否很大。但是对我来说，如果他们拿起USB听筒或使用头戴式耳机，他们可以关闭回声消除功能，而不必担心...但是，尽管PC扬声器带有麦克风，但我对此感到非常满意。

You can adapt as you need to and play with the idea, but like I said. It actually sounds pretty dang good. The only problem I have is if they have a lot of background noise. But for me, if they pick up their USB handset or are using a headset, they can turn echo cancellation off, and not worry about it...but though PC speakers with a mike...I'm pretty happy with it.

我希望它会有所帮助，或者为您提供一些基础...

I hope it helps, or gives you something to build on...

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