add webrtc aec code.

This commit is contained in:
luocai 2024-09-05 12:24:05 +08:00
parent 76d9e6c8cb
commit 78bb8b9fc7
7 changed files with 122 additions and 6 deletions

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@ -13,7 +13,9 @@
"${workspaceFolder}/build/_deps/kylin-src/Universal",
"${workspaceFolder}/build/_deps/kylin-src/HttpProxy"
],
"defines": [],
"defines": [
"WEBRTC_POSIX"
],
"compilerPath": "/usr/bin/gcc",
"cStandard": "c17",
"cppStandard": "gnu++17",

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@ -29,6 +29,7 @@ FetchContent_Declare(Kylin
)
FetchContent_MakeAvailable(Kylin)
add_subdirectory(VocieProcess)
add_subdirectory(GatePass)
add_subdirectory(Record)
add_subdirectory(VoucherVerifyServer)

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@ -9,6 +9,7 @@ add_executable(Record main.cpp
Recorder.cpp
SpeexDsp.h SpeexDsp.cpp
Utility.h Utility.cpp
WebRtcAecm.h WebRtcAecm.cpp
WebRTCPublisher.h WebRTCPublisher.cpp
)
@ -35,6 +36,7 @@ target_link_directories(Record
)
target_link_libraries(Record
PRIVATE VocieProcess
PRIVATE asound
PRIVATE easymedia
PRIVATE drm

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@ -1,4 +1,7 @@
#include "BoostLog.h"
#include "SpeexDsp.h"
#include "Utility.h"
#include "WebRtcAecm.h"
#include "main.h"
#include <memory>
@ -14,18 +17,54 @@ void EchoRecordTask::setChannels(int channels) {
}
}
// underrun occurred pcm播放饥饿
// 回采信号提前于mic信号时间差<80ms
// ./Record --echo --vqe=false --channels=2
// ./Record --echo --vqe=true --channels=2
// ./Record --echo --vqe=false --channels=1
void EchoRecordTask::run() {
RkAudio::Format format;
format.channels = m_channels;
format.period = 64;
format.period = 10;
m_speex = std::make_shared<SpeexDsp>();
m_speex->start(format.sampleRate, m_channels, format.period);
m_farendBuffer.resize(m_channels * sizeof(int16_t) * format.sampleRate / 1000 * format.period);
m_nearendBuffer.resize(m_channels * sizeof(int16_t) * format.sampleRate / 1000 * format.period);
m_webRtcAecm = std::make_shared<WebRtcAecm>();
m_webRtcAecm->start(format.sampleRate, format.channels, format.period);
m_output = std::make_shared<RkAudio::Output>();
if (!m_output->open(sizeof(uint16_t), format.sampleRate, format.channels, format.period, m_vqeEnabled)) {
if (!m_output->open(sizeof(uint16_t), format.sampleRate, 2, format.period, m_vqeEnabled)) {
LOG(error) << "audio output open failed.";
return;
}
m_outBuffer.resize(m_channels * sizeof(int16_t) * format.sampleRate / 1000 * format.period);
m_input = std::make_shared<RkAudio::Input>();
m_input->setDataCallback([this](const RkAudio::Frame &frame) { m_output->write(frame.data, frame.byteSize); });
m_input->setDataCallback([this](const RkAudio::Frame &frame) {
memcpy(m_nearendBuffer.data(), frame.data, frame.byteSize);
m_webRtcAecm->echoPlayback(reinterpret_cast<const int16_t *>(m_farendBuffer.data()), m_farendBuffer.size() / 2);
m_webRtcAecm->echoCancellation(reinterpret_cast<int16_t *>(frame.data), reinterpret_cast<int16_t *>(m_nearendBuffer.data()),
reinterpret_cast<int16_t *>(m_outBuffer.data()), frame.frameSize);
// m_speex->echoPlayback(reinterpret_cast<const int16_t *>(frame.data));
// m_speex->echoPlayback(reinterpret_cast<const int16_t *>(m_buffer.data()));
// m_speex->echoCapture(reinterpret_cast<const int16_t *>(frame.data), reinterpret_cast<int16_t *>(m_buffer.data()));
if (m_channels == 2) {
m_output->write(frame.data, frame.byteSize);
} else if (m_channels == 1) {
auto filledData = duplicate(m_outBuffer.data(), m_outBuffer.size());
m_output->write(filledData.data(), filledData.size());
}
memcpy(m_farendBuffer.data(), m_outBuffer.data(), m_outBuffer.size());
// m_output->write(reinterpret_cast<const uint8_t *>(m_buffer.data()), m_buffer.size());
});
m_input->open(format, m_vqeEnabled);
}

50
Record/WebRtcAecm.cpp Normal file
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@ -0,0 +1,50 @@
#include "WebRtcAecm.h"
#include "BoostLog.h"
#include "modules/audio_processing/aecm/echo_control_mobile.h"
WebRtcAecm::~WebRtcAecm() {
close();
}
void WebRtcAecm::close() {
if (m_hanlde != nullptr) {
webrtc::WebRtcAecm_Free(m_hanlde);
m_hanlde = nullptr;
}
}
void WebRtcAecm::echoPlayback(const int16_t *play, int samples) {
if (webrtc::WebRtcAecm_BufferFarend(m_hanlde, play, samples) != 0) {
LOG(warning) << "failed to buffer farend signal.";
}
}
void WebRtcAecm::echoCancellation(const int16_t *nearendNoisy, const int16_t *nearendClean, int16_t *out, int samples) {
if (webrtc::WebRtcAecm_Process(m_hanlde, nearendNoisy, nearendClean, out, samples, 60) != 0) {
LOG(warning) << "WebRtcAecm_Process() failed.";
}
}
bool WebRtcAecm::start(int sampleRate, int channels, int period) {
close();
m_hanlde = webrtc::WebRtcAecm_Create();
if (m_hanlde == nullptr) {
LOG(error) << "WebRtcAecm_Create() failed.";
return false;
}
if (webrtc::WebRtcAecm_Init(m_hanlde, sampleRate) != 0) {
LOG(error) << "WebRtcAecm_Init() failed.";
close();
return false;
}
webrtc::AecmConfig config;
config.cngMode = webrtc::AecmTrue; // 启用舒适噪声生成
config.echoMode = 3; // 回声消除模式,范围是 0-4
if (WebRtcAecm_set_config(m_hanlde, config) != 0) {
LOG(error) << "WebRtcAecm_set_config() failed.";
close();
return false;
}
return true;
}

18
Record/WebRtcAecm.h Normal file
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@ -0,0 +1,18 @@
#ifndef __WEBRTCAECM_H__
#define __WEBRTCAECM_H__
#include <cstdint>
class WebRtcAecm {
public:
~WebRtcAecm();
bool start(int sampleRate, int channels, int period);
void close();
void echoPlayback(const int16_t *play, int samples);
void echoCancellation(const int16_t *nearendNoisy, const int16_t *nearendClean, int16_t *out, int samples);
private:
void *m_hanlde = nullptr;
};
#endif // __WEBRTCAECM_H__

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@ -6,7 +6,7 @@
#include <memory>
class SpeexDsp;
class WebRtcAecm;
class Task {
public:
virtual void run() = 0;
@ -50,7 +50,11 @@ private:
std::shared_ptr<RkAudio::Output> m_output;
std::shared_ptr<RkAudio::Input> m_input;
std::shared_ptr<SpeexDsp> m_speex;
std::vector<uint8_t> m_buffer;
std::shared_ptr<WebRtcAecm> m_webRtcAecm;
std::vector<uint8_t> m_outBuffer;
std::vector<uint8_t> m_nearendBuffer;
std::vector<uint8_t> m_farendBuffer;
};
#endif // __MAIN_H__