This is a reland of commit ccc87ea3c625e43ab138e00ba2ef1a2d99756199
Downstream project has been updated.
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: Ibc7aeb518ed0bd7f1d725f140132c99e5a89bcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308880
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40305}
This reverts commit ccc87ea3c625e43ab138e00ba2ef1a2d99756199.
Reason for revert: Breaks downstream project
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: I05db80ba9f29460239de82cea9d95136e4c708e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308860
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40298}
This is a reland of commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I123d1134a212f65cfbc90ecec9013d0aafebd9ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308721
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40294}
while not really covered by
https://www.rfc-editor.org/rfc/rfc5576.html#section-4.2
and using the same SSRC for RTX and primary payload may work
since payload type demuxing *could* be used is not a good idea.
This also applies to flexfec's FEC-FR.
For the nonstandard SIM ssrc-group duplicates make no sense.
This rejects duplicates for unknown ssrc-groups as well.
BUG=chromium:1454860
Change-Id: I3e86101dbd5d6c4099f2fdb7b4a52d5cd0809c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308820
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40292}
There are now multiple ways to configure VP9 L1Tx:
- Legacy API: configure legacy SVC and disable encodings, this gets
interpreted as disabling spatial layers (non-standard API hack).
- Standard API: configure scalability_mode. This can be done either
with a single encoding or multiple encodings. As long as only one
encoding is active we get a single L1Tx ssrc, same as legacy API.
Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks
bitrates was only applied in the legacy API code path, not the standard
API code path, despite both code paths configuring L1Tx.
The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if
`number_of_streams == 1`. This is true in legacy code path but not
standard code path. The fix is to look at
`numberOfSimulcastStreams == 1` instead, which is set to the correct
value regardless of code path used.
This CL adds comments documenting the difference between
`number_of_streams` and `numberOfSimulcastStreams` to reduce the risk
of more mistakes like this in the future.
Bug: chromium:1455039, b:279161263
Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40287}
This reverts commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc.
Reason for revert: Breaks downstream projects
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I57778cccc3a13eb9f955f6ece054dee0ff5a7e92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308720
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40278}
This allows simplifying code in the codebase to be able to remove a lot
of templated code and special casing for either AudioCodec and VideoCodec.
Code simplifications will come in later changes.
Bug: webrtc:15214
Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40276}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
Part 1 of the migration towards merging the types.
Any method that could belong to the Codec type was moved, the others
are deprecated.
Alternatives to the AudioCodec and VideoCodec constructors are introduced
to allow creating objects of an indefinite type without having to
reference the old classes.
Bug: webrtc:15214
Change-Id: I20e1aa32962821cad98e9a92c2ec86f8f75e5dd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40213}
This is required for ReportTransportStats since iterating over the
transceiver list from the network thread is not safe.
Bug: chromium:1446274, webrtc:12692
Change-Id: I7c514df9f029112c4b1da85826af91217850fb26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40197}
after measurements have shown this is quite rare. Rollout is guarded by
WebRTC-PreventBundleHeaderExtensionIdCollision
which acts as a killswitch.
BUG=webrtc:14782,chromium:1447758
Change-Id: Ib314c2c8099c05ace761710fdf0e01a77fc89f76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306223
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40177}
There are no common functions between MediaSendChannelInterface
and MediaReceiveChannelInterface except media_type().
This allows us to remove the common superclass for the two interfaces,
making for a simpler class structure.
Bug: webrtc:13931
Change-Id: I82a12ca31f0dc62d7bd97bdda34ca37e59a5fd55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40154}
since BUNDLE is not meaningful for those cases.
This matches Firefox behavior.
BUG=chromium:1444615
Change-Id: Id841b7e30a1c920efd977caebc71ab25d084577a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40151}
Whether a metric is to be exposed to JavaScript or not is a blink
implementation detail that the WebRTC repository does not need to be
concerned with.
This CL removes unused code and paves the way for the possibility of
making the one and only RTCStatsMember class be absl::optional<>-based
in the future.
Bug: webrtc:15162
Change-Id: I578715f48b8fcc3534b72b4c700fd6567f8d553e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40139}
This is a refactoring that should have no production impact.
It has been activated for 2 weeks before, but was rolled back
because of a performance impact - this has now been fixed.
Intended to be submitted May 24 - after the 115 branch cut.
Bug: webrtc:13931
Change-Id: I745558cc3062cb4ea0a4d6f537702efc96eb7574
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305221
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40131}
Track stats are roughly equal in size as the RTP stream stats which
are the largest objects making up the majority of the RTCStatsReport
size and scales with meeting size. Deleting track/stream reduces the
size in approximately half which should reduce performance overhead
and unblock code simplifications.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4517530
# Relevant bots already passed
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: Ib7bdb84c10459b42b829228d11876498e5227312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40129}
And fix bug that prevented it from passing.
Bug: webrtc:13931
Change-Id: I6cbc8e3aad704f6f7e33362efb7ec589ca6e6568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306184
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40112}
This makes the handling somewhat more uniform, and is the same
for both video and audio channels.
Bug: webrtc:13931
Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40107}
by not starting the receive stream whenever it is creating.
Instead, this is controlled by the direction of the media content.
BUG=webrtc:11013
Change-Id: Iaaa0ac0aa9f90a4be776a1348f53a0f9c2b84d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40064}
If the caller calls RegisterObserver() on the network thread while the
state is not kOpen but there are queued received data, those received
data will be immediately delivered to the observer before the state is
transitioned to kOpen, which may break the observer's assertions and
cause problems.
The problem turns out to be that, when SctpDataChannel::RegisterObserver
calls DeliverQueuedReceivedData(), the data will be passed to the
observer without checking the |state_| first, meanwhile
SctpDataChannel::UpdateState does effectively check the state and
null-check |observer_| before delivering the received data. This CL
fixes this by simply making DeliverQueuedReceivedData() also check
`state_ == kOpen`. In case the state transitions to kOpen after
RegisterObserver() is called, the first DeliverQueuedReceivedData()
call will be no-op, while the second DeliverQueuedReceivedData() call
will do the work.
Bug: chromium:1442696
Change-Id: If25ce6a038d704939b1a8ae73d7ced110448b050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304687
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40036}
This is especially needed for datachannels that get created in
response to an OPEN message and RegisterObserver() is called from
within the OnDataChannel callback. More details in the associated bug.
Bug: webrtc:15165
Change-Id: I833db6c3c503623d482808dc5a02f03b9821a5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304721
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40032}
which avoids an infinitely growing SDP if the remote end rejects
the datachannel section. This will reactivate the m-line even if
all datachannels are closed.
BUG=chromium:1442604
Change-Id: If60f93b406271163df692d96102baab701923602
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40029}
The web compat requirement that was the reason for keeping
is now solved in Chromium and its stats bindings.
BUG=webrtc:9674
Change-Id: Ifb722769414b2bcc5f4d36d7dff87a875336e039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303860
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40024}
This reduces dependency on the struct RTCPReportBlock and would allow to
delete it in favor of class ReportBlockData
Bug: None
Change-Id: I93874c4f54cf62af0c16ae26e2231b8fb49f195d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304161
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39985}
* retransmittedBytesReceived
* retransmittedPacketsReceived
added to the specification in
https://github.com/w3c/webrtc-stats/pull/735
BUG=webrtc:15096
Change-Id: I6770e5d8d09ac1c2693c918fd943b0ab257ec7ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295260
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39959}
which gets assigned from a uint32_t VideoReceiverInfo::frames_received so should remain an unsigned type
BUG=None
Change-Id: I1db6a3f96c4ff49eee72dcce54eb6fff346c128c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302342
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39958}
This is in preparation of using the state that SourceTracker manages
for more things than only getContributingSources. Audio levels reported
via getStats(), aren't consistent with levels reported via getCS.
Since more operations will be derived from the ST owned data, moving
the management of it away from the audio thread, reduces the potential
of contention.
Bug: webrtc:14029, webrtc:7517, webrtc:15119
Change-Id: I553f7e473316a1c61eeb43ded905a18242a04424
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302280
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39943}
Tsan bots detected races since callbacks are being made on the network
thread but tests checked the state from the signaling thread.
Bug: none
Change-Id: If854e44159c56c0d12616e0b62ad92018291ed30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302281
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39928}
This removes frequent output (typically 3 times in a row):
"RED codec red is missing an associated payload type."
Bug: none
Change-Id: Ie7e0f344209cb01f9730960a6fec9d5987eaadfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301720
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39885}
This reverts commit 298313534df2420e079ffc6fc9c6019d01d29a88.
Changes from the original commit:
* Call OnTransportClosed() from TeardownDataChannelTransport_n()
(same as before the original commit)
* Not call OnTransportClosed() from OnTransportChanged() when its
called with nullptr (also preserving the behaviour from before
the original commit).
Original change's description:
> Revert "Add param to DCC::SetupDataChannelTransport_n, simplify DCC* setup code."
>
> This reverts commit 2ec6a6c57830e06f601607c1b9473ad821b57e07.
>
> Reason for revert: It breaks WPT tests (e.g. https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/1361972/overview) blocking the roll into Chromium.
>
> Original change's description:
> > Add param to DCC::SetupDataChannelTransport_n, simplify DCC* setup code.
> >
> > * DCC = DataChannelController.
> >
> > * Consolidate steps to set the mid and transport name. They're now
> > set at the same time and without a separate PostTask.
> > * Transport sink is now consistently set in DCC
> > * Order of notifications for setting up the transport is now the same
> > regardless of the first time the transport is being set or if it's
> > being replaced.
> > * Made set_data_channel_transport() private.
> >
> > Bug: webrtc:11547
> > Change-Id: I39e89c6e269e6f06d55981d7944678bf23c8817a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300562
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39859}
>
> Bug: webrtc:11547
> Change-Id: I0d8d7453b71be80fbf1b7eba7d161336e29de091
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301360
> Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39864}
Bug: webrtc:11547
Change-Id: I8ebbc3d3a12786dff2096350a77e03e98466ff00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301702
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39884}
This reverts commit 2ec6a6c57830e06f601607c1b9473ad821b57e07.
Reason for revert: It breaks WPT tests (e.g. https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/1361972/overview) blocking the roll into Chromium.
Original change's description:
> Add param to DCC::SetupDataChannelTransport_n, simplify DCC* setup code.
>
> * DCC = DataChannelController.
>
> * Consolidate steps to set the mid and transport name. They're now
> set at the same time and without a separate PostTask.
> * Transport sink is now consistently set in DCC
> * Order of notifications for setting up the transport is now the same
> regardless of the first time the transport is being set or if it's
> being replaced.
> * Made set_data_channel_transport() private.
>
> Bug: webrtc:11547
> Change-Id: I39e89c6e269e6f06d55981d7944678bf23c8817a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300562
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39859}
Bug: webrtc:11547
Change-Id: I0d8d7453b71be80fbf1b7eba7d161336e29de091
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301360
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39864}
* DCC = DataChannelController.
* Consolidate steps to set the mid and transport name. They're now
set at the same time and without a separate PostTask.
* Transport sink is now consistently set in DCC
* Order of notifications for setting up the transport is now the same
regardless of the first time the transport is being set or if it's
being replaced.
* Made set_data_channel_transport() private.
Bug: webrtc:11547
Change-Id: I39e89c6e269e6f06d55981d7944678bf23c8817a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300562
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39859}
