95e7dbb7c7
This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e. > > > > Reason for revert: Broke chromium tests. > > Original change's description: > > > Replace BundleFilter with RtpDemuxer in RtpTransport. > > > > > > BundleFilter is replaced by RtpDemuxer in RtpTransport for payload > > > type-based demuxing. RtpTransport will support MID-based demuxing later. > > > > > > Each BaseChannel has its own RTP demuxing criteria and when connecting > > > to the RtpTransport, BaseChannel will register itself as a demuxer sink. > > > > > > The inheritance model is changed. New inheritance chain: > > > DtlsSrtpTransport->SrtpTransport->RtpTranpsort > > > > > > NOTE: > > > When RTCP packets are received, Call::DeliverRtcp will be called for > > > multiple times (webrtc:9035) which is an existing issue. With this CL, > > > it will become more of a problem and should be fixed. > > > > > > Bug: webrtc:8587 > > > Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0 > > > Reviewed-on: https://webrtc-review.googlesource.com/61360 > > > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#22613} > > > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > > > Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:8587 > > Reviewed-on: https://webrtc-review.googlesource.com/64860 > > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22614} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
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