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}
ChangeResolution
Instead a warning is logged.
This effect tests only, and allows us to write screen share tests that may try to trigger the capturers to change resolution.
Bug: none
Change-Id: I4740fc4ed0bcf75e1c9df332fa610c24ed14973a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245981
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35668}
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}
Update RtpPacketInfos internals to use rtc::make_ref_counted, and a
Data class with no virtual methods.
Bug: webrtc:13464, webrtc:12701
Change-Id: I03f6bee69a9f060dcf287284fc779268d5eb433e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244505
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35660}
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}
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}
Field telemetry has shown the combination of min_fps = 0 and max_fps >
0 is unused in the wild. Therefore it's safe to turn the
WebRTC-ZeroHertzScreenshare field trial default on unless the field
trial is disabled.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: Iea701218aa178b569333087b004106ffe2e85133
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244086
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35621}
