5daeff9c1f
This reverts commit dfbced6504720d2c0807d7b92798eb80ba3f8be9. Reason for revert: Crashes when making a video call. #9 0x00000001043dd8d8 in webrtc::RTPVideoHeaderH264& absl::variant_internal::TypedThrowBadVariantAccess<webrtc::RTPVideoHeaderH264&>() at /third_party/absl/types/internal/variant.h:315 #10 0x00000001043dd8ac in absl::variant_internal::VariantAccessResultImpl<2ul, absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&&&>::type absl::variant_internal::VariantCoreAccess::CheckedAccess<2ul, absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&>(absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&&&) at /third_party/absl/types/internal/variant.h:597 #11 0x00000001043db778 in webrtc::RTPVideoHeaderH264& absl::get<webrtc::RTPVideoHeaderH264, webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>(absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&) at /third_party/absl/types/variant.h:299 #12 0x0000000104558bcc in webrtc::RtpPacketizer::Create(webrtc::VideoCodecType, unsigned long, unsigned long, webrtc::RTPVideoHeader const*, webrtc::FrameType) at webrtc/modules/rtp_rtcp/source/rtp_format.cc:30 Original change's description: > Remove RTPVideoHeader::h264() accessors. > > Bug: none > Change-Id: I043bcaf358575688b223bc3631506e148b47fd58 > Reviewed-on: https://webrtc-review.googlesource.com/88220 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23971} TBR=danilchap@webrtc.org,stefan@webrtc.org,philipel@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: none Change-Id: If99bcabdfe3cae7094f24e407bbe2f47233e46e3 Reviewed-on: https://webrtc-review.googlesource.com/88820 Commit-Queue: JT Teh <jtteh@webrtc.org> Reviewed-by: Zeke Chin <tkchin@webrtc.org> Reviewed-by: JT Teh <jtteh@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23993}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct