since the tie breaker is owned by the allocator now.
BUG=webrtc:42224914
Change-Id: I76bd5ae714fb2a6df38e014991242f390ae87e6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42371}
with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target.
BUG=webrtc:339300437
Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42362}
reduced-size RTCP, i.e. not prefixing RTCP packets with either a sender report or receiver report has been implemented for a long time but only for video.
This CL adds it for audio as well. This reduces the size of audio NACKs (16 bytes, typically one NACK per packet) sent by not prefixing it with a receiver report (32 bytes).
Other packets are not affected as e.g. transport-cc feedback does not add a RR even though that is technically required.
The effect on NACK can be tested by running Chromium with
--disable-webrtc-encryption --force-fieldtrials=WebRTC-FakeNetworkReceiveConfig/loss_percent:5/
against this fiddle negotiating audio nack:
https://jsfiddle.net/fippo/8ubtLnfx/1/
BUG=webrtc:340041654
Change-Id: I06fb94742ff1b6f9a464c404bfc53913f23498d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350269
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42330}
Intended use is to convert between different representations of "codec".
Bug: webrtc:42226302
Change-Id: If6d985ad17c2ff6018c77c7858e602b9eefa9297
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350562
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42319}
Replace usage of BuiltInNetworkBehaviorConfig.link_capacity_kbps in tests with DataRate link_capacity.
Bug: webrtc:14525
Change-Id: Id1c1b8d20eb2be5e9d1461704bb7c37c61c491f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350300
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42306}
This is a reland of commit 47bfe39ecfe45b2f94c616ace97949003d9e87b4
Original change's description:
> Split digest methods from ssl target into digest target
>
> in an attempt to break up the monolithic ssl target.
>
> BUG=None
>
> Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#42249}
Bug: webrtc:339300437
Change-Id: I31bb79bbc6cc55a2634176f95ec67de195974e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42304}
in an attempt to break up the monolithic ssl target.
BUG=None
Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42249}
This provides a way to tell the SDP generator to use a specific list
of codecs, rather than trying to compute what list to send.
Preparatory to making codec decisions per-transceiver.
Bug: webrtc:42226302
Change-Id: I1b7d4e55ed7a0546394b74820b4e51434ef86ad9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349620
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#42247}
this removes the reliance on the no-longer-spec a=msid-semantic lines
in case the offer did not signal any msid. Endpoints not supporting
msid should silently ignore the resulting a=msid: line. This also changes behavior such that a "legacy" offer without msid-semantic
line will be responded to with both msid-semantic and msid for any tracks present.
Plan-B ssrc-specific msid attributes are not signalled in that case.
See https://datatracker.ietf.org/doc/html/rfc8829#section-5.3.1
which includes it in the answer depending on the transceiver direction
but not if and only if the offer signalled a msid.
This also avoids recreating the stream and changing the SSRC
which could happen if the answer object was serialized to SDP
(which most unit tests do not do)
BUG=chromium:328522463
Change-Id: Id2f890b7756721d7c50460359950826d392483ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346741
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42237}
mids get backfilled starting with 0 which means they are always
present in the answer (even though JSEP says otherwise) and may
even be backfilled in a manner compatible with their usage in a
BUNDLE group. Those cases are ok-ish but should be documented by
tests.
BUG=None
Change-Id: I69f0475c279da5022109a56f0006169dbc2de147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349380
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42234}
These are aliases for cricket::Codec.
Also remove internal usage
Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
Old target and call/simulated.h exist but refer to new target in test/network.
Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}
This is the first step in implementing custom codecs in SDP.
Bug: none
Change-Id: I7789478208a769eaefd58b410ae6f488c604594d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348662
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#42171}
The integration_test_helpers.h file was too long and had too many
big functions inline.
This CL takes some of the largest and puts them in the .cc file.
Bug: None
Change-Id: Ibaaf9675ca8b5efa29878b4883b21f14104451a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349020
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42169}
This adds Perfetto support to WebRTC with a GN flag rtc_use_perfetto.
The configuration of perfetto depends on whether or not webrtc is
build within Chrome or not. When in Chrome, WebRTC will depend on
//third_party/perfetto:libperfetto. When building standalone, specific includes required for Perfetto are exposed with the library webrtc_libperfetto.
The perfetto trace API is exposed with a header export in
trace_event.h which is used instead of the legacy API.
The addition of Perfetto means there are 4 compilation modes for
tracing in WebRTC,
1. No tracing implementation.
2. Legacy tracing (AddTraceEvent/GetCategoryEnabled).
3.a. Perfetto statically linked (webrtc_libperfetto).
3.b. Perfetto in Chrome (Chrome's libperfetto).
This CL removes the tracing expectations from
rtc_stats_integrationtest.cc because those directly used the old API.
Integration into Chrome is a follow up CL which depends on
https://chromium-review.googlesource.com/c/chromium/src/+/5471691.
Tested: Ran Chrome with Perfetto and traces appear. WebRTC Unit test tracing working: https://ui.perfetto.dev/#!?s=04ea2613ea36b814394639a1ec4b60be5b5097527f1a485995ecc13469885468
Bug: webrtc:15917
Change-Id: I537d79dc247c2b759689910c621087286a4d8fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mikhail Khokhlov <khokhlov@google.com>
Cr-Commit-Position: refs/heads/main@{#42166}
the rollout has happened a while ago with no issues requiring the use
of the killswitch
BUG=chromium:40066610
Change-Id: I2c8148976a1da219ebbfbe6908224b6384348194
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348823
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42164}
These are required by the Perfetto API and the missing argument prevents
the use of Perfetto.
Bug: webrtc:15917
Change-Id: Ie40c0344dc9d8cd40f7c751b133d150b975a33c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347702
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#42147}
which was using the audio type to generate the id. Safe change
since the id is supposed to be random.
BUG=webrtc:12529
Change-Id: I9909c6d320f6f9239f0466599eba1f0eacf00adf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347683
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42142}
Earlier attempts have shown that this signal is multiply listened to.
Bug: webrtc:11943
Change-Id: I382df9a554925d214872d788c5d7a36f2f7c7b7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348661
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42139}
This reverts commit dc43cb24bd8ee85d6a8224c5928ceaf90de729b6.
Reason for revert: Converted the wrong signal, should have been GatheringState.
Original change's description:
> Convert P2PTransportChannel Candidate Pair Change to CallbackList
>
> Earlier attempts have shown that this signal is multiply listened to.
>
> Bug: webrtc:11943
> Change-Id: If9130a7f4c70714b5afda5aca0469b66c8e2612f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347981
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42124}
Bug: webrtc:11943
Change-Id: I73d5d815ced8d7aef4df765c9cf54d7637c6769d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42129}
Earlier attempts have shown that this signal is multiply listened to.
Bug: webrtc:11943
Change-Id: If9130a7f4c70714b5afda5aca0469b66c8e2612f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347981
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42124}
Instead, PeerConnectionFactoryDependencies.network_controller_factory is
used if it exists.
Bug: webrtc:8415
Change-Id: I37d5cc7325072bf1d87993e53949f1b97c277f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347860
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42120}
following the audio changes. Note that RTT-related fields require
DLRR and are not implemented yet.
BUG=webrtc:12529
Change-Id: I3f9449fbe876a1b282a32f2bcebe1cf3e10989bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346580
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42069}
To pass field trials to EncoderStreamFactory in FakeVideoSendStream and thus reduce dependency on the global field trial.
Bug: webrtc:10335
Change-Id: Iad32881c2d9158fe1d77f1b71f8d606374ea111e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42023}
This changes the libvpx VP9 encoder to generate the scalability mode based on the current encoding parameters when using layer activation.
Tested: Ran with L3T3_KEY reduced to L2T3_KEY and L1T3 due to bandwidth or layer activation. Added unit tests.
Bug: webrtc:15892
Change-Id: Iaedca4ea5fc3a692996666ceaf0d6aa03fb058a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344760
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42007}
This code looked a bit weird before this CL - probably because of old
refactorings.
In JsepTransport constructor, there is a DCHECK assuring that the RTP
DTLS transport is always present, so it can be passed directly to the
SctpTransport constructor, which avoids having the SetDtlsTransport
method in it.
Also, in the SctpTransport constructor, there was code that would set
the SCTP transport state to `kConnecting` if the DTLS transport was
present, but that was dead code, as it was always `nullptr` inside the
constructor before this CL. With this CL, it's always present, and the
SCTP Transport's state will initially always be `kConnecting` now. Which
is a step to deprecating the `kNew` state that doesn't exist in
https://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate.
One test case was modified, as it didn't test the reality. The test
created a SctpTransport, registered an observer, and added the DTLS
transport, and expected to receive a "statechange" from `kNew` (which is
not a state that exists in the spec) to `kConnecting`. If the test had
tested the opposite ordering - adding the DTLS transport first, and then
adding an observer, it wouldn't have experienced this. And since in
reality (with the implementation of JsepTransport before and
after this CL), it always adds the DTLS transport before any observer is
registered. So it wouldn't ever be fired, outside of tests.
Bug: webrtc:15897
Change-Id: I6ac24e0a331b686eb400fcf388ece50f2ad46a32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345420
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41987}
The timeout was not long enough in debug mode on slower machines.
Bug: chromium:40072842
Change-Id: Id82399cd7211abf5dd2e03ffa2ee4bd49f8c492f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344680
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41971}
After we're done sending all the messages, if the channel was in closing
state, then we start closing the association at the SCTP level, which
allows transitioning to the closed state.
Bug: chromium:40072842
Change-Id: I81b26b4137593b8feeb4bd9a2563cdfd67e1049e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344421
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41962}
This CL removes the send buffers (but not the receive buffer) from
SctpDataChannel and increases the send buffer in DcSctpSocket instead.
The reasons are:
1) Simplify the code. This additional buffering was strictly needed
before we migrated away from usrsctp, as that send buffer was very
limited in size (by design). But with the migration to dcSCTP, it's
no longer needed, so it just adds complexity.
2) Make `RTCDataChannel::bufferedAmount` correct. Before this CL, it
represented just the data buffered in SctpDataChannel, and not the
data accepted by the SCTP socket, but not yet put on the wire. This
makes it hard for clients to know when a message has ever been sent.
3) Better handle draining data on data channel close. While this is not
implemented in dcSCTP, having a single buffer makes this easier to
add.
While most of this CL is straightforward, the handling of bufferedAmount
in the signaling thread (in RTCDataChannel in Blink), is a bit special.
The number returned by `RTCDataChannel::bufferedAmount` is not what the
true value is inside the SCTP socket, but an eventual consistent view
of that value. When a message is sent, the value is incremented and:
- Before this change: When a message was put on the SCTP socket, the
view's value was decremented. Which made the view reflect what was
buffered outside the SCTP socket, and that buffering is now gone.
- After this change: SctpDataChannel will track what RTCDataChannel
will think it is, and provide updates to that number as we are
notified that it's reduced - by setting a "low threshold" callback
trigger.
A bonus with the new behavior is that it will be eventually consistent
and auto-heal also in error conditions - when messages are dropped due
to errors (bad input, bad state, etc). Previously, the bufferedAmount
value could drift away from the correct value on errors.
Note that a big chunk of unit tests were removed with this CL, as those
tested how the buffering behaved. Now, there is no buffering, so the
removed test cases represent a simpler interface.
This CL has been extensively tested with data channel benchmarks that
use the bufferedAmount thresholds (in Javascript).
Bug: chromium:40072842
Change-Id: I1a6a4af6b6e1116832f5028f989ce9f44683d229
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343361
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41945}
This code was extracted to make the next following CL easier to review.
This CL simply exposes the getters, setters and callbacks to set the
buffered amount low threshold on a specific SCTP stream. It will be
used in a follow-up CL, but is just boilerplate.
Bug: chromium:40072842
Change-Id: Iccd72208b369ddc252cc5886f6446b9c2ceeb0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41943}
TRACE_EVENT is already scoped!
#rtc_fixit
Tested: Compiled the patch in Chromium and confirmed the Proxy events are still present. I can send the resulting trace to reviewers if desired.
Bug: webrtc:15867
Change-Id: I5717a85c0ee25e8e20123afa08064c9b6666ba96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41916}
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.
Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.
This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.
This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.
Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
Before this CL, the StreamId class represented either a valid SCTP
stream ID, or "nothing", which means that it was a wrapped
absl::optional. Since created data channels don't have a SCTP stream ID
until it's known whether this peer will use odd or even numbers, the
"nothing" value was used for that state.
This unfortunately made it a bit hard to work with objects of this type,
as one always had to check if it contained a value. And even if a caller
would check this, and then pass the StreamId to a different function,
that function would have to do the check itself (often as a RTC_DCHECK)
since the passed StreamId always could have that state.
This CL simply extracts the "absl::optional" part of it, forcing holders
to wrap it in an optional type - when it can be "nothing". But allowing
the other code to just pass StreamId that can't be "nothing". That
simplifies the code a bit, potentially removing some bugs.
Bug: chromium:41221056
Change-Id: I93104cdd5d2f5fc1dbeb9d9dfc4cf361f11a9d68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342440
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41880}
This reverts commit ed8390d21a7b15091d01bc8e843193d0a6efd23a.
Reason for revert: Fix has landed in chrome, ready to reland.
Original change's description:
> Revert "Deprecate old constructors and set_type() in Candidate and Port"
>
> This reverts commit aaa6851d53741179a591d79fc82c4dd6651a7ba5.
>
> Reason for revert: breaks chromium webrtc import
>
> Original change's description:
> > Deprecate old constructors and set_type() in Candidate and Port
> >
> > * Deprecates constructors that use string based `type`
> > * Deprecates string based type functions in favor of enum based.
> > * Restrict possible values of Candidate::type. Ensure a valid value
> > is assigned at construction.
> > * Make Port constructors protected to limit their use to subclasses.
> > - The reason for this is to make sure that use of SharedSocket()
> > is controlled (it adds a bit of complexity).
> > * Simplify construction of Port (remove Construct() etc)
> >
> > Bug: webrtc:15846
> > Change-Id: If24ed674e175642efa49da37fd2bc847dd14f613
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339860
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41865}
>
> Bug: webrtc:15846
> Change-Id: Ic8b7cba97f8fb207ef51a88900e704658ade28b7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342140
> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#41867}
Bug: webrtc:15846
Change-Id: I3d52643bbb537d1c072643528828d26eb18fea94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342200
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41875}
The DcSctpTransport will soon use field trials to conditionally enable
some options.
And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.
Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
