编译查阅
webRTC源码编译(Android,Linux)-CSDN博客
编译相关问题
webRTC源码编译问题记录_webrtc m105-CSDN博客
下面的相关操作,
代表你已经gclient sync
具体的步骤和问题请查看前面两篇内容
开始编译
gn gen out/AudioProcess
改写参数
# out/AudioProcess/args.gn
target_os = "android"
target_cpu = "arm64"
is_debug = false
rtc_use_h264 = false
use_rtti = true
use_custom_libcxx = false
rtc_include_tests = false
rtc_build_examples = false
rtc_enable_protobuf = false
rtc_enable_static_analysis = false
rtc_build_tools = false
rtc_build_audio_processing = true
开始编译
gn gen out/Debug
nin