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 AudioProcessingwebrtc::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)