Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
--- Background ---
The webrtc::VideoSendStream::StreamStats are converted into
VideoSenderInfo objects which turn into "outbound-rtp" stats objects in
getStats() (or "ssrc" objects in legacy getStats()).
StreamStats are created for each type of substream: RTP media streams,
RTX streams and FlexFEC streams - each with individual packet counters.
The RTX stream is responsible for retransmissions of a referenced media
stream and the FlexFEC stream is responsible for FEC of a referenced
media stream. RTX/FEC streams do not show up as separate objects in
getStats(). Only the media streams become "outbound-rtp" objects, but
their packet and byte counters have to include the RTX and FEC counters.
--- Overview of this CL ---
This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes
StreamStats of all kinds as input, and outputs media-only StreamStats
- incorporating the RTX and FEC counters into the relevant media
StreamStats.
The merged StreamStats objects is a smaller set of objects than the
non-merged counterparts, but when aggregating all packet counters
together we end up with exact same packet and count as before.
Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates
the StreamStats into a single VideoSenderInfo (single "outbound-rtp"),
this CL should not have any observable side-effects. Prior to this CL:
aggregate StreamStats. After this CL: merge StreamStats and then
aggregate them.
However, when simulcast stats are implemented (WIP CL:
https://webrtc-review.googlesource.com/c/src/+/168120) each RTP media
stream should turn into an individual "outbound-rtp" object. We will
then no longer aggregate all StreamStats into a single "info". This CL
unblocks simulcast stats by providing StreamStats objects that could be
turned into individual VideoSenderInfos.
--- The Changes ---
1. Methods added to RtpConfig to be able to easily tell the relationship
between RTP, RTX and FEC ssrcs.
2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that
replaces the booleans (is_rtx, is_flexfec).
3. "referenced_media_ssrc" is added to StreamStats, making it possible
to tell which kRtx/kFlexFec stream stats need to be merged with which
kMedia StreamStats.
4. MergeInfoAboutOutboundRtpSubstreams() added and used.
Bug: webrtc:11439
Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30869}
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201.
Issue 43201 didn't do the job properly.
1. The audio rtcp report interval is not properly hooked up.
2. We don't need to propagate audio rtcp interval into video send stream or vice versa.
3. We don't need to propagate rtcp report interval to any receiving streams.
Bug: webrtc:8789
Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f
Reviewed-on: https://webrtc-review.googlesource.com/c/110105
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25610}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
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}
