Commit Graph

9 Commits

Author SHA1 Message Date
553fd3220b Use unordered_map in RtpDemuxer
The RtpDemuxer uses a number of maps, all of which can be made
unordered as they map SSRCs/MIDs/payload types/RSIDs etc - all of which
have no inherent ordering.

In busy media servers, the std::map operations can use ~0.5% CPU. After
this commit has landed, it will be evaluated and if it doesn't live up
to expected savings, it will be reverted.

Bug: webrtc:12689
Change-Id: I99e21c6b1ddb21dd9d47b0f9a891df5a2c3df59a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216243
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33846}
2021-04-27 11:38:01 +00:00
15e078c574 Fix unsignalled ssrc race in WebRtcVideoChannel.
BaseChannel adds and removes receive streams on the worker thread
(UpdateRemoteStreams_w) and then posts a task to the network thread to
update the demuxer criteria. Until this happens, OnRtpPacket() keeps
forwarding "recently removed" ssrc packets to the WebRtcVideoChannel.
Furthermore WebRtcVideoChannel::OnPacketReceived() posts task from the
network thread to the worker thread, so even if the demuxer criteria was
instantly updated we would still have an issue of in-flight packets for
old ssrcs arriving late on the worker thread inside WebRtcVideoChannel.

The wrong ssrc could also arrive when the demuxer goes from forwarding
all packets to a single m= section to forwarding to different m=
sections. In this case we get packets with an ssrc for a recently
created m= section and the ssrc was never intended for our channel.

This is a problem because when WebRtcVideoChannel sees an unknown ssrc
it treats it as an unsignalled stream, creating and destroying default
streams which can be very expensive and introduce large delays when lots
of packets are queued up.

This CL addresses the issue with callbacks for when a demuxer criteria
update is pending and when it has completed. During this window of time,
WebRtcVideoChannel will drop packets for unknown ssrcs.

This approach fixes the race without introducing any new locks and
packets belonging to ssrcs that were not removed continue to be
forwarded even if a demuxer criteria update is pending. This should make
a=inactive for 50p receive streams a glitch-free experience.

Bug: webrtc:12258, chromium:1069603
Change-Id: I30d85f53d84e7eddf7d21380fb608631863aad21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214964
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33757}
2021-04-16 09:33:42 +00:00
e091fd21d6 Remove lock from RtpStreamReceiverController.
The demuxer variable is now being used from the same thread consistently
so it's safe to replace the lock with a sequence checker.

Down the line, we may move construction+use of the
RtpStreamReceiverController class in Call, over to the network thread.
This should be possible without further modifications to
RtpStreamReceiverController.

Bug: webrtc:11993, webrtc:11567
Change-Id: Iee8c31ddf9b26b39393f40b5b1d25343b0233ae3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202245
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33016}
2021-01-18 09:10:14 +00:00
8fddf1f7b3 Delete callbacks from RtpDemuxer on ssrc binding
It was only used by RtcpDemuxer that is now deleted

Bug: None
Change-Id: Ief2c285e293cde3ed7576b194d2df137d6cbad38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178902
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31760}
2020-07-17 15:41:39 +00:00
49a5e3ec3c Logging of demuxer conflicts and sink bindings.
Bug: webrtc:10139
Change-Id: I7e49069c636a2aa4abc76dd0ee7cacaaa06007fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175114
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31286}
2020-05-17 07:43:41 +00:00
884adca3a0 Log details when RtpDemuxer fails to deliver a packet
Bug: None
Change-Id: Ie9dc5c3c545073d2e43b464d2585cb945eb346fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131360
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27634}
2019-04-16 00:47:53 +00:00
ed09dc6f56 Don't check MIDs when demuxing RTP packets in Call
The MID header extension is handled by the RtpTransport
which lives above Call and takes care of demuxing to SSRC.

Bug: webrtc:4050
Change-Id: I27135e296ae9c7b15e926ba17547c26c75684ad6
Reviewed-on: https://webrtc-review.googlesource.com/65025
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22682}
2018-03-29 20:36:08 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00