{"push_id":2483999544,"size":1,"distinct_size":1,"ref":"refs/heads/master","head":"365381fdf1cd83e6dc7ce1debaa8a97ede6f4e21","before":"c6416fbe167df739b6551109d48eb435e26a89f9","commits":[{"sha":"365381fdf1cd83e6dc7ce1debaa8a97ede6f4e21","author":{"name":"Zhi Huang","email":"64514bc03b49e67cfb75da60b665f8d8aefa50ca@webrtc.org"},"message":"Replace BundleFilter with RtpDemuxer in RtpTransport.\n\nBundleFilter is replaced by RtpDemuxer in RtpTransport for payload\ntype-based demuxing. RtpTransport will support MID-based demuxing later.\n\nEach BaseChannel has its own RTP demuxing criteria and when connecting\nto the RtpTransport, BaseChannel will register itself as a demuxer sink.\n\nThe inheritance model is changed. New inheritance chain:\nDtlsSrtpTransport->SrtpTransport->RtpTranpsort\n\nThe JsepTransport2 is renamed to JsepTransport.\n\nNOTE:\nWhen RTCP packets are received, Call::DeliverRtcp will be called for\nmultiple times (webrtc:9035) which is an existing issue. With this CL,\nit will become more of a problem and should be fixed.\n\nBug: webrtc:8587\nChange-Id: Ibd880e7b744bd912336a691309950bc18e42cf62\nReviewed-on: https://webrtc-review.googlesource.com/65786\nCommit-Queue: Zhi Huang <zhihuang@webrtc.org>\nReviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>\nReviewed-by: Benjamin Wright <benwright@webrtc.org>\nCr-Commit-Position: refs/heads/master@{#22867}","distinct":true,"url":"https://api.github.com/repos/notedit/webrtc-clone/commits/365381fdf1cd83e6dc7ce1debaa8a97ede6f4e21"}]}
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