This is an operation specific subclass of SdpOfferAnswerHandler that in
this first step, takes over the implementation details that before this
CL were implemented in SdpOfferAnswerHandler::DoSetRemoteDescription.
This CL does not change the behavior of the implementation but it does
break up DoSetRemoteDescription into smaller methods and moves the state
related to the SRD operation, into a class that in upcoming steps can
be passed around asynchronously as needed, which will allow us to avoid
blocking threads.
Bug: webrtc:13540
Change-Id: Id2002d2390a4a13725f5967df5b82064b37c7490
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244980
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35669}
This changes SetLocalDescription/SetRemoteDescription to just resetting
the internal cache timestamp for the legacy stats handler instead of
performing a full update, which can be costly.
Bug: webrtc:13557
Change-Id: I93971dbd7abf33c0d0f2836f9c17ba4550f41a00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245645
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35661}
In preparation for switching the default from kPlanB to kUnifiedPlan,
which could cause subtle bugs for those not prepared for it, we change
the default to kNotSpecified. The only purpose of kNotSpecified is to
crash, forcing any dependencies to explicitly set their sdp_semantics
value.
Tests are updated to explicitly set sdp_semantics when necessary, and
where the test does not care we update to kUnifiedPlan.
If this change lands without getting reverted we can let it sit for a
few weeks, after which we should change the default to kUnifiedPlan and
delete kNotSpecified.
Bug: webrtc:11121
Change-Id: I19b669b0735d78e269e19eaae86c2d7d95a91141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242968
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35651}
If a call to BaseChannel::SetPayloadTypeDemuxingEnabled does not result
in a demuxer criteria configuration change, we can avoid an Invoke
and speed up negotiations.
Bug: webrtc:11993
Change-Id: I80c894d7cc2a733ab84c1f4978f0c58c09a123a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35649}
This part is specific to unified plan and doesn't need most of
the state related to the remote description (and doesn't return an
error).
Bug: none
Change-Id: I0de66bdb2e925072a6d9010e4444e75d4574ae04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245102
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35642}
This is just a step to reduce the size of ApplyRemoteDescription to make
refactoring it easier (and ultimately support async operations).
Bug: none
Change-Id: Idb950c35f585a887d6640278b6edfdd0c7cec3fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245101
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35641}
With these changes, we now often have 0 invokes and at most 1 when
calling SetLocalContent on a channel. Before we had at least 1 and
typically 2.
Summary of changes.
* Updating RtpExtension::DeduplicateHeaderExtensions to return a sorted
vector (+test) for easy detection of changes.
* Before updating the transport on the network thread, detect if
actual changes to the demuxer criteria or changes to the rtp header
extensions have been made.
* Consolidate both transport updates to a single call instead of two.
* Added DCHECK guards to catch regressions in number of invokes.
A possible upcoming improvement is to update the transport
asynchronously.
Bug: webrtc:13536
Change-Id: I71ef7b181635a796ffa1e3a02a0f661d28a4870c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35638}
This is a bit of refactoring to clear the way for some more upcoming
changes and fix little oddities here and there that are basically
artifacts of many small incremental changes throughout the years.
* Remove the CryptoOptions member variable and instead only keep around
the filter for rtp header extensions.
* Remove several member methods that only forwarded calls to
media_channel() and effectively reduced readability.
* Consolidated quite a bit of code related to UpdateRemoteStreams_w
and the copy/pasted code in the Video/Voice classes around calling it.
* UpdateRemoteStreams_w now returns an error when it's encountered.
Before, an error would still be returned in those cases but all
operations were unnecessarily performed.
Bug: none
Change-Id: I85a37b9e8f00584aa794abef11abfe89dec5d0a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244098
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35637}
This is a slight refactoring while doing some other changes, so not
strictly necessary, but the error param is always supplied in practice
so it made sense to update the tests to reflect that, test that error
values are reported in (at least) some cases and remove the additional
code that checks for whether or not error information is requested.
Bug: none
Change-Id: Ia5739a18ea2beb6970eabf9d809c24dfa43466b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244097
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35632}
Make the jsep_transport.cc code conform to WebRTC code style.
Bug: None
Change-Id: I9142c1e83d4006988e206aeb0bbe52dfffafda60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35631}
Improve consistency between using DCHECK checkers and compile time.
For virtual methods, we were sometimes using both and in other cases
we could be using compile time checks but were using runtime.
Added annotation for last_send_params_, last_recv_params_ in
audio/video channel classes.
Also removing redundant logging for when registration with the
transport fails. This is already being logged in the demuxer.
Bug: webrtc:12230
Change-Id: I48e156c9996dec26a990151301dabc06673541d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244095
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35630}
Remove unnecessary optimization from BaseChannel,
previous_demuxer_criteria_, that I'm not seeing as providing value.
Previously it was used to avoid a thread hop if a reconfiguration
wasn't needed, but the way that was done, wasn't thread safe. So after
addressing that issue, the variable more represents increased complexity
in the code than runtime efficiency.
Bug: webrtc:11993, webrtc:12230
Change-Id: Ic8e3e1d2f57e669a168cc7b5cf5d407133976e3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244093
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35628}
This also removes all internal usage of RemoveTrack, and changes
the replacement function to RemoveTrackOrError rather than RemoveTrackNew.
Bug: webrtc:9534
Change-Id: Idf7bb17495686de77c70428dcbfb12278328ce59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244094
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35624}
BaseChannel::content_name() is consistently used as the mid throughout
the code and the mid is stored in the demuxer criteria. Furthermore
there's a chance that the two variables might not be in sync if the
mid needs to be truncated, so it's better to use one source of truth.
Bug: webrtc:11993, webrtc:12230
Change-Id: Ia98443d8ee65fd0795651981acab27c29428ba0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244092
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35622}
Make sure previous_demuxer_criteria_ is only accessed on the network
thread and add annotation.
Bug: webrtc:11993, webrtc:12230
Change-Id: I4700fe41c947a3b1cce9649642dcd38ed62f873d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244087
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35618}
Because of this (seemingly simple) change, I had to change the return
type of transport_name from `const std::string&` to `absl::string_view`
to handle the case when there's no transport assigned.
That in turn caused an avalanche of required updates.
Bug: webrtc:12230, webrtc:11993
Change-Id: I16ec6c6a5fc2f5f7c7de572355a3c6ca924bb9d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244084
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35617}
Add TODO for accessing `previous_demuxer_criteria_`, currently accessed
from two threads (unsafe).
Changed RtpDemuxerCriteria to be a class, all members private with
accessor methods instead of direct variable access. Moving forward
this can allow for things like checking for thread/sequence and state
consistency.
Bug: webrtc:12517, webrtc:11993
Change-Id: I21c1b3067e988494ce6f4c6c85c62165801883bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244083
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35616}
This simplifies the work that happens on the worker thread in
preparation of avoiding having to go to the worker at all.
Bug: webrtc:11993
Change-Id: I13f063bdecce8efdb978ef1976c819019f30e020
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244082
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35610}
This makes things slightly simpler for the time being as surrounding
code is being refactored. This also removes a PostTask which has the
effect of shrinking the window between the Pending/Complete
notifications slightly since there's no additional async task
for the 'complete' step.
Bug: webrtc:11993
Change-Id: Ia86779b21c6f87301f37d763f89ace722e06e563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244081
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35609}
This CL continues on https://webrtc-review.googlesource.com/c/src/+/242361
to enable progression of RequestRefreshFrame calls in into
the stored VideoTrackSourceInterface.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I558e7a616b3369192c52f7211bbe41a4376b48c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242365
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35567}
Under zero-hertz mode, provided that a frame arrived to the
VideoStreamEncoder, the receiver may experience up to a second
between incoming frames. This results in key frame requests getting
serviced with that delay, which is undesired.
What's worse is also the fact that if no frame ever arrived to the
VideoStreamEncoder, it will not service the keyframe requests at all
until the first frame comes.
This change introduces VideoSourceInterface::RequestRefreshFrame
which results in a refresh frame being sent from complying sources.
The method is used under zero-hertz mode from the VideoStreamEncoder
when frames didn't arrive to it yet (with changes to the zero-hertz
adapter).
With this change, when the frame adapter has received at least one
frame, it will conditionally repeat the last frame in response to the
key frame request.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I6f97813b3a938747357d45e5dda54f759129b44d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242361
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35562}
The matcher layer tolerates MIDs only up to 16, but the parser
polices a limit of 32 (due to users using this in ways that do
not create a matcher).
Tolerate the MID instead of crashing.
Bug: webrtc:12517
Change-Id: I67ac4a7fa53c918b271b5a3020f497c9d60ec6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240521
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35513}
If we reference a remote codec that was offered without rtx we need
to change the reference of our local rtx codec in a reoffer.
This only happens with stopped transceivers.
BUG=chromium:1201441
Change-Id: I60dc4908ba7ba2c0249a2a05b03a7b35af504e18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238382
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35487}
So applications don't need to create and inject their own instance of
BasicPortAllocator, just to change these settings.
Bug: webrtc:13145
Change-Id: I08ac8658b4c0ef87019fa579be9195a8a6b50feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239643
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35476}
This patch adds reporting of relay protocol,
i.e how a client connect to the turn server.
This is added in the old stats api...cause there
are clients still using it.
Bug: none
Change-Id: Iac7fe3e3de0ba42d5897c304ebbae368edf498fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35469}
the maximum used in practice is multiopus with
6 or 8 channels. 24 is the maximum number of channels
supported in the audio decoder.
BUG=chromium:1265806
Change-Id: Iba8e3185a1f235b846fed9c154e66fb3983664ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/main@{#35440}
This was the behaviour before https://webrtc-review.googlesource.com/c/src/+/218605,
and is currently relied upon by Chrome to mute received audio tracks
by default, until they should be played out.
Bug: chromium:1272566
Change-Id: I8a288a287e7c01392f4af1db5b083e8d7ee7b2a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238665
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35401}
This is a partial revert of commit f9e502d935fa9ca8d9cc7540680021b93d285f28.
Reason for revert: Functionality turns out to be needed by some partners for some months more.
Original change's description:
> Remove enable_dtls_srtp option
>
> This is part of the removal of support for SDES.
>
> Bug: webrtc:11066
> Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35262}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11066, chromium:1271469
Change-Id: I79a90f025e53816789b391bc52a0e896b9be87e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238170
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35378}
they should be assertions and just decrease the readability of the actual test.
BUG=None
Change-Id: Ib6b23246fe03c8058e721cd63e887c0cd92a32e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237806
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35358}
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}
This makes relay candidates identifiable in scenarios
where the TURN server is behind another entity and the peer
sees a different ip address:
client -> turn -> relay address -> third party relay + address -> peer
In those cases, the relay candidate will become peer-reflexive
since the peer sends the third party relay's address in the xor-mapped
address and it is currently not easily possible to determine this is a
relay candidate anymore.
BUG=webrtc:13392
Change-Id: I6787339d0abdc735f8a43f636a676cccd8cadcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237561
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35346}
this could happen if setCodecPreferences is used to prefer
red over opus as it is done for red+opus.
BUG=webrtc:13287
Change-Id: I3d61cd8f1a364572bc531a75dcc239c3919138cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237800
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35344}
which is the maxium length allowed by one-byte header extensions
BUG=webrtc:12517
Change-Id: I003105d3566a34b5b7affb84ffe69b7705973ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35333}
which was not checking this candidate
BUG=webrtc:7063
Change-Id: Ic2b9a27cfa842bcacd434d171a919515aafb2312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35332}
