# 桃夭终端 ## 支持版本 * SDK 11 * WebRTC m120 * libmediasoupclient m120 ## C++终端 * [libmediasoupclient源码](https://github.com/versatica/libmediasoupclient) * [libmediasoupclient文档](https://mediasoup.org/documentation/v3/libmediasoupclient) * [libmediasoupclient接口](https://mediasoup.org/documentation/v3/libmediasoupclient/api) ## 项目配置 可以自己编译`WebRTC`依赖或者下载已有依赖,项目导入以后拷贝`libmediasoupclient`源码还有`WebRTC`头文件和库文件到`deps`目录。 [WebRTC](https://pan.baidu.com/s/1E_DXv32D9ODyj5J-o-ji_g?pwd=hudc) > 注意删除目录`build`目录和`third_party`目录中除了`abseil-cpp`以外的所有依赖(当然不删也没关系就是文件太多编译器会变慢) * https://gitee.com/openharmony-sig/ohos_webrtc/blob/master/doc/webrtc_build.md ## 鸿蒙编译 ``` # WebRTC版本:m120 # libmediasoupclient版本:m120 # armeabi-v7a gn gen ./out/armeabi-v7a --args='target_os="ohos" target_cpu="arm" is_clang=true is_debug=false use_rtti=true rtc_use_h264=true rtc_use_h265=true rtc_libvpx_build_vp9=true is_component_build=false rtc_include_tests=false libyuv_include_tests=false rtc_build_examples=false treat_warnings_as_errors=false ohos_sdk_native_root="/data/dev/ohos-sdk/linux/native"' ninja -C ./out/armeabi-v7a -j 32 # arm64-v8a gn gen ./out/arm64-v8a --args='target_os="ohos" target_cpu="arm64" is_clang=true is_debug=false use_rtti=true rtc_use_h264=true rtc_use_h265=true rtc_libvpx_build_vp9=true is_component_build=false rtc_include_tests=false libyuv_include_tests=false rtc_build_examples=false treat_warnings_as_errors=false ohos_sdk_native_root="/data/dev/ohos-sdk/linux/native"' ninja -C ./out/arm64-v8a -j 32 ``` ## openharmony-sig/ohos_webrtc * https://gitee.com/openharmony-sig/ohos_webrtc * https://gitee.com/openharmony-sig/ohos_webrtc/tree/master/doc ## openharmony-tpc/chromium_third_party_webrtc * https://gitee.com/openharmony-tpc * https://gitee.com/openharmony-tpc/chromium_third_party_webrtc * https://gitee.com/openharmony-tpc/chromium_third_party_ohos_prebuilts ## 参考项目 * https://gitee.com/han_jin_fei/oh_web-rtc ## 源码修改 ``` vim modules/audio_device/audio_device_impl.cc vim modules/audio_device/ohos/ohaudio_recorder_wrapper.cc OH_AudioStreamBuilder_SetLatencyMode(builder, AUDIOSTREAM_LATENCY_MODE_NORMAL); ```