准备
简单
两台带有麦克风和摄像头的电脑
复杂
没有麦克风和摄像头的单台电脑需要安装OBS软件(虚拟摄像头)
代码方面需要禁用音频
void Conductor::AddTracks() {
if (!peer_connection_->GetSenders().empty()) {
return; // Already added tracks.
}
// rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
// peer_connection_factory_->CreateAudioTrack(
// kAudioLabel,
// peer_connection_factory_->CreateAudioSource(cricket::AudioOptions())
// .get()));
// auto result_or_error = peer_connection_->AddTrack(audio_track, {kStreamId});
// if (!result_or_error.ok()) {
// RTC_LOG(LS_ERROR) << "Failed to add audio track to PeerConnection: "
// << result_or_error.error().message();
// }
rtc::scoped_refptr<CapturerTrackSource> video_device =
CapturerTrackSource::Create();
if (video_device) {
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track_(
peer_connection_factory_->CreateVideoTrack(video_device, kVideoLabel));
main_wnd_->StartLocalRenderer(video_track_.get());
auto result_or_error =
peer_connection_->AddTrack(video_track_, {kStreamId});
if (!result_or_error.ok()) {
RTC_LOG(LS_ERROR) << "Failed to add video track to PeerConnection: "
<< result_or_error.error().message();
}
} else {
RTC_LOG(LS_ERROR) << "OpenVideoCaptureDevice failed";
}
main_wnd_->SwitchToStreamingUI();
}
OBS版本:OBS-Studio-29.1.3-Full-Installer-x64.exe
配置参考
手把手!教你用新版OBS虚拟摄像头 - 哔哩哔哩 (bilibili.com)
可以使用自动配置向导,选择最后一项“我只使用虚拟摄像机”即可
回到主界面,在下方的“来源”栏下点击“+”加号,添加一个“媒体源”
点击“浏览”,选择刚才录制好的视频文件
选择好文件后,一定记得勾选“循环”!
单击主界面右下角“控件”一栏里的“启动虚拟摄像机”
WebRTC服务器配置
src\examples\peerconnection\client\flag_defs.h
ABSL_FLAG(std::string, server, "localhost", "The server to connect to.");
定义了连接WebRTC服务器IP地址
修改为指定的服务器IP地址
src\examples\peerconnection\client\defaults.cc
const uint16_t kDefaultServerPort = 8888;
定义了连接WebRTC服务器端口
操作
任意一方可以在中途按ESC键退出音视频通信
运行失败按照如下修改代码
依次运行exe文件peerconnection_server和peerconnection_client,可以发现server窗口中出现了新的连接字样,但是client并没有跳转画面,依旧卡在初始界面
方法一
运行PeerConnectionClient,使用本地ip连接PeerConnectionServer时,PeerConnectionClient的onConnect没有回调。经过分析,是由于main.cc定义的线程没有运行
需要在peer_connection_client.cc中的方法PeerConnectionClient::DoConnect()添加逻辑,在方法最后添加rtc::Thread::Current()->Start();
void PeerConnectionClient::DoConnect() {
control_socket_.reset(CreateClientSocket(server_address_.ipaddr().family()));
hanging_get_.reset(CreateClientSocket(server_address_.ipaddr().family()));
InitSocketSignals();
char buffer[1024];
snprintf(buffer, sizeof(buffer), "GET /sign_in?%s HTTP/1.0\r\n\r\n",
client_name_.c_str());
onconnect_data_ = buffer;
bool ret = ConnectControlSocket();
if (ret)
state_ = SIGNING_IN;
if (!ret) {
callback_->OnServerConnectionFailure();
}
rtc::Thread::Current()->Start();
}
另外还有个线程死等的问题,要修改peer_connection_client.cc中的Wait方法
windows运行webrtc的PeerConnectionClient连接问题修复_peerconnectionclient 空白_vsix_ling的博客-CSDN博客
方法二
/*
* Copyright 2012 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// clang-format off
// clang formating would change include order.
#include <windows.h>
#include <shellapi.h> // must come after windows.h
// clang-format on
#include <string>
#include <vector>
#include "absl/flags/parse.h"
#include "examples/peerconnection/client/conductor.h"
#include "examples/peerconnection/client/flag_defs.h"
#include "examples/peerconnection/client/main_wnd.h"
#include "examples/peerconnection/client/peer_connection_client.h"
#include "rtc_base/checks.h"
#include "rtc_base/ssl_adapter.h"
#include "rtc_base/string_utils.h" // For ToUtf8
#include "rtc_base/win32_socket_init.h"
#include "system_wrappers/include/field_trial.h"
#include "test/field_trial.h"
#include "rtc_base/logging.h"
#include "rtc_base/log_sinks.h"
namespace {
// A helper class to translate Windows command line arguments into UTF8,
// which then allows us to just pass them to the flags system.
// This encapsulates all the work of getting the command line and translating
// it to an array of 8-bit strings; all you have to do is create one of these,
// and then call argc() and argv().
class WindowsCommandLineArguments {
public:
WindowsCommandLineArguments();
WindowsCommandLineArguments(const WindowsCommandLineArguments&) = delete;
WindowsCommandLineArguments& operator=(WindowsCommandLineArguments&) = delete;
int argc() { return argv_.size(); }
char** argv() { return argv_.data(); }
private:
// Owned argument strings.
std::vector<std::string> args_;
// Pointers, to get layout compatible with char** argv.
std::vector<char*> argv_;
};
WindowsCommandLineArguments::WindowsCommandLineArguments() {
// start by getting the command line.
LPCWSTR command_line = ::GetCommandLineW();
// now, convert it to a list of wide char strings.
int argc;
LPWSTR* wide_argv = ::CommandLineToArgvW(command_line, &argc);
// iterate over the returned wide strings;
for (int i = 0; i < argc; ++i) {
args_.push_back(rtc::ToUtf8(wide_argv[i], wcslen(wide_argv[i])));
// make sure the argv array points to the string data.
argv_.push_back(const_cast<char*>(args_.back().c_str()));
}
LocalFree(wide_argv);
}
} // namespace
class CustomSocketServer : public rtc::PhysicalSocketServer {
public:
bool Wait(webrtc::TimeDelta max_wait_duration, bool process_io) override {
if (!process_io)
return true;
return rtc::PhysicalSocketServer::Wait(webrtc::TimeDelta::Zero(),
process_io);
}
};
int PASCAL wWinMain(HINSTANCE instance,
HINSTANCE prev_instance,
wchar_t* cmd_line,
int cmd_show) {
rtc::WinsockInitializer winsock_init;
//rtc::PhysicalSocketServer ss;
CustomSocketServer ss;
rtc::AutoSocketServerThread main_thread(&ss);
rtc::FileRotatingLogSink* pLog = new rtc::FileRotatingLogSink("D:/", "webrtc.log", 1024*1024*10, 10);
pLog->Init();
rtc::LogMessage::AddLogToStream(pLog, rtc::LS_VERBOSE);
WindowsCommandLineArguments win_args;
int argc = win_args.argc();
char** argv = win_args.argv();
absl::ParseCommandLine(argc, argv);
// InitFieldTrialsFromString stores the char*, so the char array must outlive
// the application.
const std::string forced_field_trials =
absl::GetFlag(FLAGS_force_fieldtrials);
webrtc::field_trial::InitFieldTrialsFromString(forced_field_trials.c_str());
// Abort if the user specifies a port that is outside the allowed
// range [1, 65535].
if ((absl::GetFlag(FLAGS_port) < 1) || (absl::GetFlag(FLAGS_port) > 65535)) {
printf("Error: %i is not a valid port.\n", absl::GetFlag(FLAGS_port));
return -1;
}
const std::string server = absl::GetFlag(FLAGS_server);
MainWnd wnd(server.c_str(), absl::GetFlag(FLAGS_port),
absl::GetFlag(FLAGS_autoconnect), absl::GetFlag(FLAGS_autocall));
if (!wnd.Create()) {
RTC_DCHECK_NOTREACHED();
return -1;
}
rtc::InitializeSSL();
PeerConnectionClient client;
auto conductor = rtc::make_ref_counted<Conductor>(&client, &wnd);
main_thread.Start();
// Main loop.
MSG msg;
BOOL gm;
while ((gm = ::GetMessage(&msg, NULL, 0, 0)) != 0 && gm != -1) {
if (!wnd.PreTranslateMessage(&msg)) {
::TranslateMessage(&msg);
::DispatchMessage(&msg);
}
}
if (conductor->connection_active() || client.is_connected()) {
while ((conductor->connection_active() || client.is_connected()) &&
(gm = ::GetMessage(&msg, NULL, 0, 0)) != 0 && gm != -1) {
if (!wnd.PreTranslateMessage(&msg)) {
::TranslateMessage(&msg);
::DispatchMessage(&msg);
}
}
}
rtc::CleanupSSL();
return 0;
}
注意事项
默认启动后是 localhost ,不知道为什么我必须改成 127.0.0.1 才可以连接服务器成功。 翻阅资料得出结果是因为 windows 把 localhost 解析为 ipv6 地址 ::1 而不是 127.0.0.1。查了hosts 把ipv6已经屏蔽了,127.0.0.1 localhost 也有,就是不能正常解析
退出崩溃解决
去掉RTC_NORETURN提示错误
function declared 'noreturn' should not return [-Werror,-Winvalid-noreturn]
修改代码如下
RTC_NORETURN void WriteFatalLog(absl::string_view output) {
#if defined(WEBRTC_ANDROID)
std::string output_str(output);
__android_log_print(ANDROID_LOG_ERROR, RTC_LOG_TAG_ANDROID, "%s\n",
output_str.c_str());
#endif
fflush(stdout);
fwrite(output.data(), output.size(), 1, stderr);
fflush(stderr);
#if defined(WEBRTC_WIN)
//DebugBreak();
#endif
exit(0);
//abort();
}
外网扩展
Windows下 WebRTC Demo运行: PeerConnection_window.rtcpeerconnection文档_chenxiemin的博客-CSDN博客
参考
fix.patch (1.2 KB) - webrtc - Web-based real-time communication - Monorail (chromium.org)
13498 - the peerconnection_client on windows not displaying the video on win10 - webrtc (chromium.org)
【Bug修复记录】WebRTC源码demo无法运行_陈成不姓丞的博客-CSDN博客