(1)在官方网站下下载webrtc。
(2)下载depot_tools,并配进环境变量,比如:
在.bashrc添加
export PATH=/root/webrtc_all/depot_tools/:$PATH
(3)编译:
gn gen out/linux/
ninja -C out/linux/
以上可以完成webrtc的原生操作。
添加自己程序的方法:
(4)在src/video下面新建test.cc,内容如下:
#include <iostream>
using namespace std;
int main()
{
cout<<"hahaha"<<endl;
return 0;
}
(5)在src/video的build.gn中,添加:
rtc_executable("test") {
testonly = true
sources = [
"test.cc",
]
deps = [
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
这一步的deps中,可以添加需要的库,比如可以参考video_loopback的deps,可以写出视频采集编码发送的实例。
(6)在src的build.gn中,添加video:test
deps += [
":rtc_unittests",
":video_engine_tests",
":webrtc_nonparallel_tests",
":webrtc_perf_tests",
"common_audio:common_audio_unittests",
"common_video:common_video_unittests",
"media:rtc_media_unittests",
"modules:modules_tests",
"modules:modules_unittests",
"modules/audio_coding:audio_coding_tests",
"modules/audio_processing:audio_processing_tests",
"modules/remote_bitrate_estimator:bwe_simulations_tests",
"modules/rtp_rtcp:test_packet_masks_metrics",
"modules/video_capture:video_capture_internal_impl",
"ortc:ortc_unittests",
"pc:peerconnection_unittests",
"pc:rtc_pc_unittests",
"rtc_base:rtc_base_tests_utils",
"stats:rtc_stats_unittests",
"system_wrappers:system_wrappers_unittests",
"test",
"video:screenshare_loopback",
"video:sv_loopback",
"video:video_loopback",
"voice_engine:voice_engine_unittests",
"video:test",
]
(7)继续编译
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ninja -C out/linux/
(8)在out/linux下,查看结果:
test,可以运行。