音频回声消除测试类

这里演示 webRTC 的 3A算法测试类。

 

【01】准备工作

仅编译 example：

cd src

set DEPOT_TOOLS_WIN_TOOLCHAIN=0

gn gen out/Default

ninja -C out/Default examples

在 src/example 下创建自己的测试目录 audio/aec，创建 test.cc ：

该类只创建了一个 APM 音频处理对象：

#include "iostream"

#include "modules/audio_processing/include/audio_processing.h"

int main(){

    std::cout << "Start ->" << std::endl;

    rtc::scoped_refptr<webrtc::AudioProcessing> apm =  webrtc::AudioProcessingBuilder().Create();

    std::cout << "End ->" << std::endl;

    return 0;

}

然后修改 example/BUILD.gn ，加入上面可执行测试类：

直接追加在  rtc_executable("peerconnection_server") 后面即可：

# 音频处理测试类

rtc_executable("audio_aec_test") {

  testonly = true

  sources = [ "audio/aec/test.cpp" ]

  deps = [

    # 音频处理的模块

    "../modules/audio_processing",

    "../modules/audio_processing:api",

  ]

}

同时在分组 example 里面追加可执行类的依赖：

deps += [

  ":peerconnection_server",

  ":stunserver",

  ":turnserver",

  ":audio_aec_test"    # 追加在这三个可执行类后面即可

]

【01】测试类

//

// Created by tangyufan on 10/10/2022.

//

#include "iostream"

#include "modules/audio_processing/include/audio_processing.h"

#include "api/audio/echo_canceller3_config.h"

#include "api/audio/echo_control.h"

// 引入头文件参考 peerconnection

// 测试用例参考 audio_processing_unittest.cc

int main(int argc, char **argv){

    std::cout << "Start ->" << std::endl;

    // 采样率和声道数

    int kSampleRateHz = 8000;

    int kNumChannels = 1;

    // fread 每次读取10ms的数据

    float frame_step = 10;

    int size = kSampleRateHz * frame_step / 1000.0;

    int delay = 100;

    int analog_level = 60;

    FILE *far_file = fopen("C:\\workspace\\webRTC\\webRTC-code\\src\\data\\audio_processing\\far.wav", "rb");

    FILE *near_file = fopen("C:\\workspace\\webRTC\\webRTC-code\\src\\data\\audio_processing\\near.wav", "rb");

    FILE *aec_out_file = fopen("C:\\workspace\\webRTC\\webRTC-code\\src\\data\\audio_processing\\out.raw", "wb");

    rtc::scoped_refptr<webrtc::AudioProcessing> apm = webrtc::AudioProcessingBuilder().Create();

    // 只开启 aec

    webrtc::AudioProcessing::Config config;

    config.echo_canceller.enabled = true;

    config.echo_canceller.mobile_mode = false;

    config.high_pass_filter.enabled = false;

    config.gain_controller1.enabled = false;

    config.gain_controller2.enabled = false;

    config.noise_suppression.enabled = false;

    apm->ApplyConfig(config);

    apm->set_stream_delay_ms(delay);

    apm->set_stream_analog_level(analog_level);

    webrtc::StreamConfig inStreamConfig  = webrtc::StreamConfig(kSampleRateHz, kNumChannels);

    webrtc::StreamConfig outStreamConfig = webrtc::StreamConfig(kSampleRateHz, kNumChannels);

    int16_t *far_frame = (int16_t*)malloc(1024);

    int16_t *near_frame = (int16_t*)malloc(1024);

    int16_t *out_frame = (int16_t*)malloc(1024);

    std::cout << "Do parse ->" << std::endl;

    int read_count = 0;

    while (true){

        read_count = fread((char*)far_frame,sizeof(int16_t),size,far_file);

        if(read_count != size){

            fseek(near_file, read_count * sizeof(int16_t), SEEK_CUR);

        }

        apm->ProcessReverseStream(far_frame,inStreamConfig,outStreamConfig,NULL);

        read_count = fread((char*)near_frame,sizeof(int16_t),size,near_file);

        if (read_count != size) {

            std::cout << "Do parse break ->" << std::endl;

            break;

        }

        apm->ProcessStream(near_frame,inStreamConfig,outStreamConfig,out_frame);

        fwrite (out_frame,sizeof(int16_t),size,aec_out_file);

    }

    fclose(far_file);

    fclose(near_file);

    fclose(aec_out_file);

    free(far_frame);

    free(near_frame);

    free(out_frame);

    std::cout << "End ->" << std::endl;

    return 0;

}