webrtc3A算法

使用ubuntu18.04

选择webrtc_audio_processing v0.3

下载地址

https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/tree/master

git clone 完

编译

# Initialise into the build/ directory, for a prefixed install into the
# install/ directory
meson . build -Dprefix=$PWD/install

# Run the actual build
ninja -C build

# Install locally
ninja -C build install

里面会提醒你需要安装的东西，

meson 0.45会报错

meson 需要0.63的，需要下载python3.8然后pip 安装

然后

~/.local/bin/meson . build -Dprefix=$PWD/install

编译完webrtc会有以下文件

就可以把include头文件和.so文件去测试

测试demo

#include <iostream>
#include <vector>
#include <sndfile.hh>
#include "modules/audio_processing/include/audio_processing.h"
#include "api/scoped_refptr.h"

// 读取 WAV 文件
bool ReadWavFile(const std::string& filename, std::vector<float>& audio_data, int& sample_rate) {
    SndfileHandle file(filename);
    if (file.error()) {
        std::cerr << "Error opening file: " << filename << std::endl;
        return false;
    }
    sample_rate = file.samplerate();
    audio_data.resize(file.frames() * file.channels());
    file.read(audio_data.data(), audio_data.size());
    return true;
}

// 写入 WAV 文件
bool WriteWavFile(const std::string& filename, const std::vector<float>& audio_data, int sample_rate, int channels) {
    SndfileHandle file(filename, SFM_WRITE, SF_FORMAT_WAV | SF_FORMAT_PCM_16, channels, sample_rate);
    if (file.error()) {
        std::cerr << "Error creating file: " << filename << std::endl;
        return false;
    }
    file.write(audio_data.data(), audio_data.size());
    return true;
}

int main(int argc, char* argv[]) {
    if (argc != 4 || !argv[1] || !argv[2] || !argv[3]) {
        printf("Usage: EXE nearIn.wav farIn.wav output.wav\n");
        return 0;
    }
    
    const char* nearInFile = argv[1];
    const char* farInFile = argv[2];
    const char* outputFile = argv[3];
    
    std::vector<float> near_data, far_data;
    int sample_rate;
    
    // 读取输入文件
    if (!ReadWavFile(nearInFile, near_data, sample_rate) || !ReadWavFile(farInFile, far_data, sample_rate)) {
        return -1;
    }
    
    // 打印输入文件信息
    std::cout << "Sample rate: " << sample_rate << std::endl;
    std::cout << "Near data (first 10 samples): ";
    for (int i = 0; i < 10 && i < near_data.size(); ++i) {
        std::cout << near_data[i] << " ";
    }
    std::cout << std::endl;
    
    // 创建 AudioProcessing 实例
    rtc::scoped_refptr<webrtc::AudioProcessing> apm = webrtc::AudioProcessingBuilder().Create();
    
    // 配置 AudioProcessing
    webrtc::AudioProcessing::Config config;
    config.echo_canceller.enabled = true;  // 禁用回声消除
    config.noise_suppression.enabled = true; // 禁用噪声抑制
    config.gain_controller1.enabled = true; // 禁用自动增益控制
    apm->ApplyConfig(config);
    
    // 设置流配置
    webrtc::StreamConfig stream_config(sample_rate, 1); // 单声道
    apm->Initialize(); // 确保 AudioProcessing 初始化 TODO:之前这步没做，一直没声音
    
    // 处理音频
    std::vector<float> output_data(near_data.size());
    
    // 将数据包装成 float* 数组
    float* near_channel[1] = { near_data.data() };  // 输入信号
    float* output_channel[1] = { output_data.data() }; // 输出信号
    
    size_t num_frames = near_data.size() / stream_config.num_channels();
    
    // 处理音频帧
    for (size_t i = 0; i < num_frames; i += stream_config.num_frames()) {
        // 处理当前帧
        size_t frames_to_process = std::min(stream_config.num_frames(), num_frames - i);//todo :使用 std::min 确保不会超过实际的音频帧数。
        apm->ProcessStream(near_channel, stream_config, stream_config, output_channel);
        
        // 移动指针
        near_channel[0] += stream_config.num_channels() * frames_to_process;//更改点
        output_channel[0] += stream_config.num_channels() * frames_to_process;
    }
    
    // 打印输出数据
    std::cout << "Output data (first 10 samples): ";
    for (int i = 0; i < 10 && i < output_data.size(); ++i) {
        std::cout << output_data[i] << " ";
    }
    std::cout << std::endl;
    
    // 写入输出文件
    if (!WriteWavFile(outputFile, output_data, sample_rate, 1)) {
        return -1;
    }
    
    std::cout << "3A processing completed. Output saved to " << outputFile << std::endl;
    return 0;
}

原图

webrtc_3a默认处理后图(有效果了)