Qt重写webrtc的demo peerconnection

整个demo为:

可以选择多个编码方式:

复制代码
cmake_minimum_required(VERSION 3.5)

project(untitled LANGUAGES CXX)
set(CMAKE_CXX_STANDARD 20)
set(CMAKE_INCLUDE_CURRENT_DIR ON)

set(CMAKE_AUTOUIC ON)
set(CMAKE_AUTOMOC ON)
set(CMAKE_AUTORCC ON)

set(CMAKE_CXX_STANDARD 20)
set(CMAKE_CXX_STANDARD_REQUIRED ON)
include_directories(
        "D:/webrtc/webrtc-checkout/src"
        "D:/webrtc/webrtc-checkout/src/out/release/obj"
        "D:/webrtc/webrtc-checkout/src/third_party/abseil-cpp"
        "D:/webrtc/webrtc-checkout/src/third_party/libyuv/include"
        "D:/webrtc/webrtc-checkout/src/third_party/jsoncpp/source/include"
        "D:/webrtc/webrtc-checkout/src/third_party/jsoncpp/generated"        
)
include_directories(./)
# QtCreator supports the following variables for Android, which are identical to qmake Android variables.
# Check http://doc.qt.io/qt-5/deployment-android.html for more information.
# They need to be set before the find_package(Qt5 ...) call.

#if(ANDROID)
#    set(ANDROID_PACKAGE_SOURCE_DIR "${CMAKE_CURRENT_SOURCE_DIR}/android")
#    if (ANDROID_ABI STREQUAL "armeabi-v7a")
#        set(ANDROID_EXTRA_LIBS
#            ${CMAKE_CURRENT_SOURCE_DIR}/path/to/libcrypto.so
#            ${CMAKE_CURRENT_SOURCE_DIR}/path/to/libssl.so)
#    endif()
#endif()
add_definitions(
        -D_ITERATOR_DEBUG_LEVEL=2
        -DUSE_AURA=1
        -DWEBRTC_USE_H264
        -D_HAS_EXCEPTIONS=0
        -D__STD_C
        -D_CRT_RAND_S
        -D_CRT_SECURE_NO_DEPRECATE
        -D_SCL_SECURE_NO_DEPRECATE
        -D_ATL_NO_OPENGL
        -D_WINDOWS
        -DCERT_CHAIN_PARA_HAS_EXTRA_FIELDS
        -DPSAPI_VERSION=2
        -DWIN32
        -D_SECURE_ATL
        -DWINUWP
        -D__WRL_NO_DEFAULT_LIB__
        -DWINAPI_FAMILY=WINAPI_FAMILY_PC_APP
        -DWIN10=_WIN32_WINNT_WIN10
        -DWIN32_LEAN_AND_MEAN
        -DNOMINMAX
        -D_UNICODE
        -DUNICODE
        -DNTDDI_VERSION=NTDDI_WIN10_RS2
        -D_WIN32_WINNT=0x0A00
        -DWINVER=0x0A00
        -DNDEBUG
        -DNVALGRIND
        -DDYNAMIC_ANNOTATIONS_ENABLED=0
        -DWEBRTC_ENABLE_PROTOBUF=0
        -DWEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
        -DRTC_ENABLE_VP9
        -DHAVE_SCTP
        -DWEBRTC_LIBRARY_IMPL
        -DWEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0
        -DWEBRTC_WIN
        -DABSL_ALLOCATOR_NOTHROW=1
        -DQT_DEPRECATED_WARNINGS
        -DQT_NO_KEYWORDS
        -DHAVE_SCTP
        -DWEBRTC_VIDEO_CAPTURE_WINRT)
find_package(Qt5 COMPONENTS Widgets Network REQUIRED)
include_directories("D:/webrtc/webrtc-checkout/src/third_party/jsoncpp/source/install/include")
include_directories("D:/webrtc/webrtc-checkout/src/third_party/libyuv/include")
link_directories("D:/webrtc/webrtc-checkout/src/out/release/obj" "D:/webrtc/webrtc-checkout/src/third_party/jsoncpp/source/install/lib")
include_directories("D:/webrtc/webrtc-checkout/src/third_party/abseil-cpp/install/include")
add_executable(untitled
    main.cpp conductor.cc defaults.cc peer_connection_client.cc  test_video_capturer.cc vcm_capturer.cc
    mainwindow.cpp D:/webrtc/webrtc-checkout/src/rtc_base/strings/json.cc
    mainwindow.h
    mainwindow.ui
  )
  file(GLOB_RECURSE MY_FILES "D:/webrtc/webrtc-checkout/src/third_party/abseil-cpp/install/lib/*.lib")
  message(${MY_FILES})
target_link_libraries(untitled
        PRIVATE
        WS2_32 secur32.lib winmm.lib dmoguids.lib wmcodecdspuuid.lib msdmo.lib Strmiids.lib Iphlpapi.lib ${MY_FILES} )
target_link_libraries(untitled PRIVATE Qt5::Widgets Qt5::Network jsoncpp.lib D:/webrtc/webrtc-checkout/src/out/release/obj/webrtc.lib)

其中h264在cmake中要加上-DWEBRTC_USE_H264,编译时的参数为

复制代码
gn gen out/release --ide=vs --args="is_debug=true use_custom_libcxx=false rtc_enable_protobuf=false target_cpu=\"x64\" enable_iterator_debugging=true symbol_level=2 is_clang=true rtc_include_tests=true rtc_use_h264=true ffmpeg_branding=\"Chrome\" proprietary_codecs=true"

需要打开webrtc选项。这个demo的下载链接为

https://download.csdn.net/download/qq_42805085/90245215

相关推荐
Fisher3Star2 天前
mediasoup WebRtcTransport核心机制解析
webrtc
小小前端--可笑可笑2 天前
【Web 流媒体三部曲之一】直播、点播与 WebRTC 是什么?
前端·webrtc
hz567892 天前
实时音视频SDK选型指南:TRTC、WebRTC与音视频PaaS能力对比
安全·音视频·webrtc·实时音视频·信息与通信·paas
Fisher3Star3 天前
WebRTC回声消除定位方法
webrtc
Fisher3Star3 天前
WebRTC音频模块替换方案
webrtc
Fisher3Star3 天前
WebRTC Android音频播放三方案解析
webrtc
Fisher3Star4 天前
mediasoup如何基于RTCP更新媒体流score
webrtc
hz567896 天前
2026 年 RTC 音视频 SDK 解析:技术架构、主流厂商与选型指南
架构·云计算·音视频·webrtc·实时音视频·信息与通信
Fisher3Star6 天前
mediasoup demo 遇到问题
webrtc
福大大架构师每日一题6 天前
pion/webrtc v4.2.13:SCTP统计信息曝光、DataChannel并发与关闭竞态修复、测试稳定性提升、依赖升级一次看懂
webrtc