由于工作需要，硬件3A中的AEC效果实在太差，后面使用SpeexDSP的软3A，效果依旧不是很好，猜测是内部时延估计和时延补偿做的并不是很好，于是采用了webrtc的3A算法，这里记录一下3A移植过程。
|版本声明：山河君，未经博主允许，禁止转载
在linux下，webrtc 3A是比较好移植的，原因是Pulse Audio是支持webrtc 3A插件的，也就是说，不需要我们自己翻墙下载配置webrtc的环境以及编译链路，Pulse Audio已经帮我们做好了这一步，剩下的就是交叉编译的工作。
对应的gitlab地址：Pulse Audio webrtc-audio-processing
先选择好版本，截至到当前博客时间，最新版本是1.3.1，网上之前也有介绍的，基本都是0.3版本的
1.0版本和1.0之前的版本最大的区别就是编译器的不同，之前是通过脚本配置，1.0后都是使用meson 进行配置，所以需要下载meson 以及ninja（用于编译webrtc）
meson和代码下载之后，就需要配置交叉编译链路，meson对应交叉编译器环境需要我们自己写meson配置文件，在源文件目录下，打开cross_file.txt，内容如下：
[binaries]
c ='/usr/bin/aarch64-linux-gnu-gcc'
cpp ='/usr/bin/aarch64-linux-gnu-g++'
ar ='/usr/bin/aarch64-linux-gnu-ar'
strip ='/usr/bin/aarch64-linux-gnu-strip'
pkgconfig ='/usr/bin/aarch64-linux-gnu-pkg-config'[host_machine]
system ='linux'
cpu_family ='aarch64'
cpu ='armv8-a'
endian ='little'[paths]
prefix ='/home/aaron/workplace/webrtc-audio-processing/build'
在源文件目录下创建编译缓存目录以及安装目录
meson . build -Dprefix=$PWD/install --cross-file=cross_file.txt
ninja -C build
ninja -C build install
最后在install目录下可以看到编译好的文件
编译好的完整demo下载：demo下载，如果没有积分的话就自己编译，这里只是少了demo的脚本
#include<stdio.h>#include<stdlib.h>#include<alsa/asoundlib.h>#include<pthread.h>#include<errno.h>#include<mutex>#include"modules/audio_processing/include/audio_processing.h"constexprint sample_rate_hz =16000;// 假设采样率为16kHzconstexpr size_t num_channels =1;// 假设单声道constexpr size_t frame_size = sample_rate_hz /100;// 10ms 帧大小#defineCHANNELS2#defineSAMPLE_RATE16000#definePERIOD_SIZE160#defineBUFFER_SIZE(PERIOD_SIZE *2)bool gRun =true;typedefstruct{
snd_pcm_t *playback_handle;
FILE *pcm_file;int8_t m_pPlayoutBuffer[PERIOD_SIZE *2*2];int m_nPlayoutBufferSizeIn20MS = PERIOD_SIZE *2*2;int m_nPlayoutFramesLeft =0;
snd_pcm_t *capture_handle;
FILE *pcm_out_file;int8_t m_pRecordBuffer[PERIOD_SIZE *2*2];int m_nRecordBufferSizeIn20MS = PERIOD_SIZE *2*2;int m_nRecordingFramesLeft = PERIOD_SIZE;
FILE *pcm_3a_file;
FILE *pcm_ref_file;int8_t m_pRefBuffer[PERIOD_SIZE *2];int8_t m_p3ABuffer[PERIOD_SIZE *2];// in byteint8_t m_pNearBuffer[PERIOD_SIZE *2];// in byte
rtc::scoped_refptr<webrtc::AudioProcessing> apm;} audio_data_t;voidmonoTurnStereo(constint16_t*pSrc,int16_t*pDts, size_t size){for(int j =0; j < size; j++){
pDts[2* j]= pSrc[j];
pDts[2* j +1]= pSrc[j];}}voidstereoTurnMono(constunsignedchar*pSrc,unsignedchar*pDts, size_t size){int nLeftCount =0;for(size_t i =0; i < size; i +=4){
pDts[nLeftCount]= pSrc[i];
pDts[nLeftCount +1]= pSrc[i +1];
nLeftCount +=2;}}int32_terrorRecovery(int32_t nRet, snd_pcm_t *pDeviceHandle){int st =snd_pcm_state(pDeviceHandle);printf("Trying to recover from %s error: %s nRet:(%d) (state:%d)\n",((snd_pcm_stream(pDeviceHandle)==
SND_PCM_STREAM_CAPTURE)?"capture":"playout"),snd_strerror(nRet),
nRet,
st);int res =snd_pcm_recover(pDeviceHandle, nRet,1);if(0== res){printf("Recovery - snd_pcm_recover OK\n");if((nRet ==-EPIPE || nRet ==-ESTRPIPE)&&snd_pcm_stream(pDeviceHandle)== SND_PCM_STREAM_CAPTURE){// For capture streams we also have to repeat the explicit start()// to get data flowing again.int nRet =snd_pcm_start(pDeviceHandle);if(nRet !=0){printf("Recovery - snd_pcm_start error: %d\n", nRet);return-1;}}if((nRet ==-EPIPE || nRet ==-ESTRPIPE)&&snd_pcm_stream(pDeviceHandle)== SND_PCM_STREAM_PLAYBACK){// For capture streams we also have to repeat the explicit start() to get// data flowing again.
snd_pcm_state_t state =snd_pcm_state(pDeviceHandle);if(state != SND_PCM_STATE_PREPARED){snd_pcm_prepare(pDeviceHandle);}int nRet =snd_pcm_start(pDeviceHandle);if(nRet !=0){printf("Recovery - snd_pcm_start error: %s\n",snd_strerror(nRet));return-1;}}return-EPIPE == nRet ?1:0;}else{printf("Unrecoverable alsa stream error: %d\n", res);}return res;}void*playback_thread(void*arg){printf("playback_thread\n");
audio_data_t *data =(audio_data_t *)arg;constint policy = SCHED_FIFO;constint min_prio =sched_get_priority_min(policy);constint max_prio =sched_get_priority_max(policy);
sched_param param;constint top_prio = max_prio -1;constint low_prio = min_prio +1;
param.sched_priority = top_prio;pthread_setschedparam(pthread_self(), policy,¶m);while(gRun){int nRet;
snd_pcm_sframes_t sndFrames;
snd_pcm_sframes_t sndAvailFrames;
sndAvailFrames =snd_pcm_avail_update(data->playback_handle);if(sndAvailFrames <0){printf("playout snd_pcm_avail_update error: %s\n",snd_strerror(sndAvailFrames));errorRecovery(sndAvailFrames, data->playback_handle);continue;}elseif(sndAvailFrames ==0){
nRet =snd_pcm_wait(data->playback_handle,2);// if (nRet == 0)// printf("playout snd_pcm_wait timeout\n");continue;}if(data->m_nPlayoutFramesLeft <=0){
size_t frames =fread(data->m_pRefBuffer,2, PERIOD_SIZE, data->pcm_file);if(frames ==0|| frames != PERIOD_SIZE){// 文件播放完毕，重新开始fseek(data->pcm_file,0,SEEK_SET);continue;}monoTurnStereo((int16_t*)data->m_pRefBuffer,(int16_t*)data->m_pPlayoutBuffer, PERIOD_SIZE);
data->m_nPlayoutFramesLeft = frames;}if((uint32_t)(sndAvailFrames)> data->m_nPlayoutFramesLeft){
sndAvailFrames =(uint32_t)data->m_nPlayoutFramesLeft;}int size =snd_pcm_frames_to_bytes(data->playback_handle, data->m_nPlayoutFramesLeft);
sndFrames =snd_pcm_writei(data->playback_handle,&data->m_pPlayoutBuffer[data->m_nPlayoutBufferSizeIn20MS - size], sndAvailFrames);if(sndFrames <0){printf("playout snd_pcm_writei error: %s\n",snd_strerror(sndFrames));
data->m_nPlayoutFramesLeft =0;errorRecovery(sndFrames, data->playback_handle);continue;}else{fwrite(&data->m_pRefBuffer[PERIOD_SIZE *2- data->m_nPlayoutFramesLeft *2],1, sndFrames *2, data->pcm_ref_file);
data->m_nPlayoutFramesLeft -= sndFrames;}}returnNULL;}void*capture_thread(void*arg){printf("capture_thread\n");
audio_data_t *data =(audio_data_t *)arg;constint policy = SCHED_FIFO;constint min_prio =sched_get_priority_min(policy);constint max_prio =sched_get_priority_max(policy);
sched_param param;constint top_prio = max_prio -1;constint low_prio = min_prio +1;
param.sched_priority = top_prio;pthread_setschedparam(pthread_self(), policy,¶m);while(gRun){int nRet;
snd_pcm_sframes_t sndFrames;
snd_pcm_sframes_t sndAvailFrames;int8_t buffer[data->m_nRecordBufferSizeIn20MS];
sndAvailFrames =snd_pcm_avail_update(data->capture_handle);if(sndAvailFrames <0){printf("capture snd_pcm_avail_update error: %s\n",snd_strerror(sndAvailFrames));errorRecovery(sndAvailFrames, data->capture_handle);continue;}elseif(sndAvailFrames ==0){continue;}if((uint32_t)(sndAvailFrames)> data->m_nRecordingFramesLeft)
sndAvailFrames = data->m_nRecordingFramesLeft;
sndFrames =snd_pcm_readi(data->capture_handle, buffer, sndAvailFrames);if(sndFrames <0){printf("capture snd_pcm_readi error: %s\n",snd_strerror(sndFrames));errorRecovery(sndFrames, data->capture_handle);continue;}elseif(sndFrames >0){int nLeftSize =snd_pcm_frames_to_bytes(data->capture_handle, data->m_nRecordingFramesLeft);int size =snd_pcm_frames_to_bytes(data->capture_handle, sndFrames);memcpy(&data->m_pRecordBuffer[data->m_nRecordBufferSizeIn20MS - nLeftSize], buffer, size);
data->m_nRecordingFramesLeft -= sndFrames;}if(!data->m_nRecordingFramesLeft){
data->m_nRecordingFramesLeft = PERIOD_SIZE;stereoTurnMono((unsignedchar*)data->m_pRecordBuffer,(unsignedchar*)data->m_pNearBuffer, PERIOD_SIZE *2*2);fwrite(data->m_pNearBuffer,1, PERIOD_SIZE *2, data->pcm_out_file);
webrtc::StreamConfig stream_config(sample_rate_hz, num_channels);if(data->apm->ProcessReverseStream((int16_t*)data->m_pRefBuffer, stream_config, stream_config,(int16_t*)data->m_pRefBuffer)!= webrtc::AudioProcessing::kNoError){printf("ProcessReverseStream fail\n");}if(data->apm->ProcessStream((int16_t*)data->m_pNearBuffer, stream_config, stream_config,(int16_t*)data->m_p3ABuffer)!= webrtc::AudioProcessing::kNoError){printf("ProcessStream fail\n");}fwrite(data->m_p3ABuffer,1, PERIOD_SIZE *2, data->pcm_3a_file);}}returnNULL;}intsetup_pcm_device(snd_pcm_t **handle,constchar*device, snd_pcm_stream_t stream){
snd_pcm_hw_params_t *params;int err;if((err =snd_pcm_open(handle, device, stream, SND_PCM_NONBLOCK))<0){printf("无法打开 PCM 设备 %s: %s\n", device,snd_strerror(err));return err;}snd_pcm_hw_params_alloca(¶ms);snd_pcm_hw_params_any(*handle, params);snd_pcm_hw_params_set_access(*handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);snd_pcm_hw_params_set_format(*handle, params, SND_PCM_FORMAT_S16_LE);snd_pcm_hw_params_set_channels(*handle, params, CHANNELS);snd_pcm_hw_params_set_rate(*handle, params, SAMPLE_RATE,0);snd_pcm_hw_params_set_period_size(*handle, params, PERIOD_SIZE,0);snd_pcm_hw_params_set_buffer_size(*handle, params, PERIOD_SIZE *4);if((err =snd_pcm_hw_params(*handle, params))<0){printf("设置 PCM 参数失败: %s\n",snd_strerror(err));snd_pcm_close(*handle);return err;}return0;}intmain(int argc,char*argv[]){
audio_data_t data;if((data.pcm_file =fopen("./far.pcm","rb"))==NULL||(data.pcm_3a_file =fopen("./3a.pcm","wb"))==NULL||(data.pcm_out_file =fopen("./near.pcm","wb"))==NULL||(data.pcm_ref_file =fopen("./ref.pcm","wb"))==NULL){printf("fail to open file\n");return-1;}// 3a
data.apm = webrtc::AudioProcessingBuilder().Create();
webrtc::AudioProcessing::Config config;// 噪声抑制配置
config.noise_suppression.enabled =true;
config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh;// 增益控制配置
config.gain_controller1.enabled =true;
config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital;
config.gain_controller1.target_level_dbfs =3;// 目标输出电平// 回声抑制配置
config.echo_canceller.enabled =true;
config.echo_canceller.mobile_mode =false;// 如果是移动设备可以设置为 true// 语音活动检测（可选）
config.voice_detection.enabled =true;// 应用配置
data.apm->ApplyConfig(config);//if(setup_pcm_device(&data.playback_handle,"hw:0,0", SND_PCM_STREAM_PLAYBACK)<0){fclose(data.pcm_file);return-1;}if(setup_pcm_device(&data.capture_handle,"hw:0,0", SND_PCM_STREAM_CAPTURE)<0){snd_pcm_close(data.playback_handle);fclose(data.pcm_file);return-1;}
pthread_t play_thread, record_thread;pthread_create(&play_thread,NULL, playback_thread,&data);int nRet =snd_pcm_prepare(data.playback_handle);if(nRet <0){printf("playout snd_pcm_prepare failed (%s)\n",snd_strerror(nRet));}pthread_create(&record_thread,NULL, capture_thread,&data);
nRet =snd_pcm_prepare(data.capture_handle);if(nRet <0){printf("capture snd_pcm_prepare failed:%s \n",snd_strerror(nRet));}
nRet =snd_pcm_start(data.capture_handle);if(nRet <0){printf("capture snd_pcm_start err:%s\n",snd_strerror(nRet));
nRet =snd_pcm_start(data.capture_handle);if(nRet <0){printf("capture snd_pcm_start 2nd try err:%s\n",snd_strerror(nRet));returnfalse;}}getchar();
gRun =false;pthread_join(play_thread,NULL);pthread_join(record_thread,NULL);snd_pcm_close(data.playback_handle);snd_pcm_close(data.capture_handle);fclose(data.pcm_file);fclose(data.pcm_out_file);fclose(data.pcm_3a_file);fclose(data.pcm_ref_file);printf("end................... \n");return0;}
远端参考信号
近端采集信号
回声消除后的信号
webrtc不愧是音视频领域的顶尖，值得我们学习的东西太多了。实际上demo里对于设备的读写，也是从webrtc中摘录出来的。
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