目前音频处理模块已经有单独打包编译的项目：
1.编译官方模块
下载：
http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-0.3.1.tar.xz
目前当前官方模块仅有linux版本：
解压
xz -d webrtc-audio-processing-0.3.1.tar.xz
tar xf webrtc-audio-processing-0.3.1.tar
重命名：
mv webrtc-audio-processing-0.3.1 webrtc-audio-processing
编译：
cd webrtc-audio-processing
./configure --enable-debug --prefix=`pwd`/debug && make -j && make install
最终会在 webrtc-audio-processing 下 debug 目录中保存所有编译结果，里面的头文件和库文件在我们自己的项目中进行引入使用。
头文件目录：
webrtc-audio-processing/debug/include/webrtc_audio_processing
库文件.a/.so文件目录：
webrtc-audio-processing/debug/lib
2.创建测试项目
项目源码：https://e.coding.net/TangYuFan/speech-processing/WebRTC-AudioProcessing-main.git
wangzhongyu1997/WebRTC-AudioProcessing (github.com)
参考链接：
WebRTC AudioProcessing 调试方法 - 知乎 (zhihu.com)
项目根目录：
CMakeLists.txt：编写编译配置生成文件。
data：测试语音wav等格式文件。
out：创建空目录，在这下面执行 cmake.. 生成 Makefile 之后再执行 make 进行编译，最终得到的测试项目可执行文件 webtrc-audio-process
src：项目源码目录，测试类 webrtc_audio_processing.cc。
webrtc-audio-processing：就是第一步编译官方模块的目录。
测试类源码 webrtc_audio_processing.cc：
/* 09/2017
shichaog
This is the main test File. Include noise suppresion, AEC, VAD.
*/
#include <string>
#include <iostream>
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
#define EXPECT_OP(op, val1, val2) \
do { \
if (!((val1) op (val2))) { \
fprintf(stderr, "Check failed: %s %s %s\n", #val1, #op, #val2); \
exit(1); \
} \
} while (0)
#define EXPECT_EQ(val1, val2) EXPECT_OP(==, val1, val2)
#define EXPECT_NE(val1, val2) EXPECT_OP(!=, val1, val2)
#define EXPECT_GT(val1, val2) EXPECT_OP(>, val1, val2)
#define EXPECT_LT(val1, val2) EXPECT_OP(<, val1, val2)
int usage() {
std::cout <<
"Usage: webrtc-audio-process -anc|-agc|-aec value input.wav output.wav [delay_ms echo_in.wav]"
<< std::endl;
return 1;
}
bool ReadFrame(FILE* file, webrtc::AudioFrame* frame) {
// The files always contain stereo audio.
size_t frame_size = frame->samples_per_channel_;
size_t read_count = fread(frame->data_,
sizeof(int16_t),
frame_size,
file);
if (read_count != frame_size) {
// Check that the file really ended.
EXPECT_NE(0, feof(file));
return false; // This is expected.
}
return true;
}
bool WriteFrame(FILE* file, webrtc::AudioFrame* frame) {
// The files always contain stereo audio.
size_t frame_size = frame->samples_per_channel_;
size_t read_count = fwrite (frame->data_,
sizeof(int16_t),
frame_size,
file);
if (read_count != frame_size) {
return false; // This is expected.
}
return true;
}
int main(int argc, char **argv) {
if (argc != 5 && argc != 6) {
return usage();
}
bool is_echo_cancel = false;
int level = -1;
level = atoi(argv[2]);
int delay_ms = 200;
// Usage example, omitting error checking:
webrtc::AudioProcessing* apm = webrtc::AudioProcessing::Create();
webrtc::Config config;
apm->level_estimator()->Enable(true);
config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(true));
apm->echo_cancellation()->Enable(true);
apm->echo_cancellation()->enable_metrics(true);
apm->echo_cancellation()->enable_delay_logging(true);
apm->set_stream_delay_ms(delay_ms);
// apm->echo_cancellation()->enable_drift_compensation(true);
apm->SetExtraOptions(config);
apm->high_pass_filter()->Enable(true);
if (std::string(argv[1]) == "-anc") {
std::cout << "ANC: level " << level << std::endl;
apm->noise_suppression()->Enable(true);
switch (level) {
case 0:
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kLow);
break;
case 1:
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kModerate);
break;
case 2:
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
break;
case 3:
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kVeryHigh);
break;
default:
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kVeryHigh);
}
apm->voice_detection()->Enable(true);
} else if (true || std::string(argv[1]) == "-agc") {
std::cout << "AGC: model " << level << std::endl;
apm->gain_control()->Enable(true);
apm->gain_control()->set_analog_level_limits(0, 255);
switch (level) {
case 0:
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
break;
case 1:
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
break;
case 2:
apm->gain_control()->set_mode(webrtc::GainControl::kFixedDigital);
break;
default:
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
}
} else if (std::string(argv[1]) == "-aec") {
std::cout << "AEC: level " << level << std::endl;
switch (level) {
case 0:
apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kLowSuppression);
break;
case 1:
apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kModerateSuppression);
break;
case 2:
apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kHighSuppression);
}
apm->echo_cancellation()->enable_delay_logging(true);
} else if(std::string(argv[1]) == "-vad"){
std::cout << "AEC: level " << level << std::endl;
apm->voice_detection()->Enable(true);
switch(level){
case 0:
apm->voice_detection()->set_likelihood(webrtc::VoiceDetection::kVeryLowLikelihood);
apm->voice_detection()->set_frame_size_ms(10);
}
} else {
delete apm;
return usage();
}
webrtc::AudioFrame far_frame;
webrtc::AudioFrame near_frame;
float frame_step = 10; // ms
int near_read_bytes;
far_frame.sample_rate_hz_ = 16000;
far_frame.samples_per_channel_ = far_frame.sample_rate_hz_ * frame_step / 1000.0;
far_frame.num_channels_ = 1;
near_frame.sample_rate_hz_ = 16000;
near_frame.samples_per_channel_ = near_frame.sample_rate_hz_ * frame_step / 1000.0;
near_frame.num_channels_ = 1;
size_t size = near_frame.samples_per_channel_;
FILE *far_file = fopen(argv[3], "rb");
FILE *near_file = fopen(argv[4], "rb");
FILE *aec_out_file = fopen(argv[5], "wb");
int drift_samples = 0;
size_t read_count = 0;
while(true){
read_count = fread(far_frame.data_, sizeof(int16_t), size, far_file);
if(read_count != size){
fseek(near_file, read_count * sizeof(int16_t), SEEK_CUR);
}
apm->ProcessReverseStream(&far_frame);
read_count = fread(near_frame.data_, sizeof(int16_t), size, near_file);
near_read_bytes += read_count * sizeof(int16_t);
if (read_count != size) {
break; // This is expected.
}
apm->set_stream_delay_ms(delay_ms);
//std::cout << "!!!!!!!!!!drift_samples"<< drift_samples << std::endl;
// apm->echo_cancellation()->set_stream_drift_samples(drift_samples);
int err = apm->ProcessStream(&near_frame);
int vad_flag = (int)apm->voice_detection()->stream_has_voice();
WriteFrame(aec_out_file, &near_frame);
}
fclose(far_file);
fclose(near_file);
fclose(aec_out_file);
delete apm;
std::cout<<"over.";
return 0;
}编译配置 CMakeLists.txt：
cmake_minimum_required(VERSION 3.0)
project(webrtc_examples)
SET(CMAKE_BUILD_TYPE "Debug")
set (webrtc_examples_VERSION_MAJAR 1)
set (webrtc_examples_VERSION_MINOR 0)
add_definitions(-std=c++11 -static
-DWEBRTC_POSIX=1
-DWEBRTC_AUDIO_PROCESSING_ONLY_BUILD=1)
include_directories (webrtc-audio-processing/debug/include/webrtc_audio_processing)
link_directories(webrtc-audio-processing/debug/lib)
link_libraries(webrtc_audio_processing)
add_executable(webrtc-audio-process src/webrtc_audio_processing.cc)
测试类编译结果：
3.项目测试
测试参数：
