The feauture was added in https://webrtc-review.googlesource.com/c/src/+/291119 in order to ensure RTX packet is part of BWE even before the RTP stream is known.
However, it cause an issue if media is signaled with an SSRC that has this RTX SSRC.
Since BWE is now notified about received packets before demuxing to the correct receive stream, it is not necessary to demux RTX packets before the media SSRC is known.
Note that WebRTC require at least one negotiated SSRC/MID before RTCP feedback can be sent. Ie, for BWE to work, at least one media SSRC must be known after this cl. It can either be unsignaled or signaled.
BWE tested with BweRampupWithInitialProbeTest.
Bug: webrtc:14795, webrtc:14817, b/320258158
Change-Id: Icf2c67bedc352720bf846b9ee38d509346af36f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41785}
MediaType::Any will be used by packets that can not be demuxed by
RtpTransport.
Bug: webrtc:14928
Change-Id: Ib759e65c7eede29defdad8073fd1ed6be814ab81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299280
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39710}
This reverts commit 897ea04db5.
Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200
Original change's description:
> Delete PacketReceiver::DeliverPacket from all implementations
>
> And fix tests that still depend on extensions to be known by the receiver.
>
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
>
> Bug: webrtc:7135,webrtc:14795
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39184}
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39189}
And fix tests that still depend on extensions to be known by the receiver.
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Bug: webrtc:7135,webrtc:14795
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39184}
With this cl, a packet is only parsed once in RtpTransport::DemuxPacket and the metadata is reused.
Extensions are still identified twice- one for demuxing based on mid. The second time in Channel::OnReceivedPacket in order to use extensions specific to that mid.
Bug: webrtc:7135, webrtc:14795
Change-Id: I50e3814af92ca4378f148876b20a54bcfac1e146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290540
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39058}
DeliverRtpPacket use a parsed RTP packet as argument where the RTP extensions are supposed to be known.
This method is implemented in webrt::Call and temporary used by the exising method Call::DeliverRtp, but the idea is to instead avoid extra packet parsing by forwarding a RtpPacketReceived from RtpTransport::DemuxRtpPacket via WebrtcVideoChannel::OnPacketReceived and WebrtcVoiceChannel.
DeliverRtcpPacket is also implemented in Call and is directly used in PeerConnection::InitializeRtcpCallback.
Bug: webrtc:14795, webrtc:7135
Change-Id: Ib6ffe8e1229ac07fa459ee2fc9a0af8455a23bac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290401
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39015}
This is currently unused and since we ultimately don't want the delivery
of packets to be async at this stage (but rather stay on the network
thread), we don't need it.
Bug: webrtc:11993
Change-Id: I6809026b6901c8ecfacd961e98ddf79aaa16d0bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220601
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34152}
This change includes a basic first implementation of the method in Call.
The implementation assumes being called on the network thread
and simply posts to the worker thread which is what currently happens
inside the BaseChannel class (but I'm that moving out of there).
Bug: webrtc:11993
Change-Id: Id94934808f655071ea9dc87ab20d2a4d8ca7e61e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202255
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33033}
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/91840, which needed
transitional methods while updating downstream code. This cl completes
the deletion, and can be landed after downstream code is updated.
Bug: webtrc:9584
Change-Id: I4d3654748973a4757a8d79bb93f524c630a0eca3
Reviewed-on: https://webrtc-review.googlesource.com/93285
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24329}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
