The new type PriorityValue is a strong 16-bit integer matching RFC 8831
requirements that can be built from a Priority enum.
The value is now propagated and used by the SCTP transport, but enabling
the feature still requires a field trial for now.
Bug: webrtc:42225365
Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42695}
This requires making CallConfig move-only so it can hold a unique_ptr to
the factory, but as discussed with Danil, that seems fine.
Bug: chromium:355610792
Change-Id: Ie52e33faaa4a2af748daeb25f5327b7a532936e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42679}
Allow API users to access the NetworkControllerInterface instance that a
given PC ended up with, to allow integrators who have provided a
PeerConnectionFactoryDependencies.network_controller_factory to
associate a created instance of their custom network controller with the
PC using it.
Eg for the RTCRtpTransport Chromium implementation as in crrev.com/c/5607744.
Bug: chromium:345101934
Change-Id: Ia712ca4f45b90d5078f4e8e5977622d3e9f9aa6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353980
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42506}
since it contains helpers mostly related to cryptographically secure random numbers and strings.
BUG=webrtc:339300437
Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
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}
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}
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}
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}
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}
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 is a reland of commit 1cce1d7ddcbde3a3648007b5a131bd0c2638724b
after updating the WPT that broke on Mac.
Original change's description:
> Make setCodecPreferences only look at receive codecs
>
> which is what is noted in JSEP:
> https://www.rfc-editor.org/rfc/rfc8829.html#name-setcodecpreferences
>
> Some W3C spec modifications are required since the W3C specification
> currently takes into account send codecs as well.
>
> Spec issue:
> https://github.com/w3c/webrtc-pc/issues/2888
> Spec PR:
> https://github.com/w3c/webrtc-pc/pull/2926
>
> setCodecPreferences continues to modify the codecs in an offer.
>
> Also rename RtpSender::SetCodecPreferences to RtpSender::SetSendCodecs for consistent semantics.
>
> BUG=webrtc:15396
>
> Change-Id: I1e8fbe77cb2670575578a777ed1336567a1e4031
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41719}
Bug: webrtc:15396
Change-Id: I0c7b17f00de02286f176b500460e17980b83b35b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339541
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41807}
Goal is to make PeerConnectionInterface methods pure virtual.
This is a split of cl https://webrtc-review.googlesource.com/c/src/+/340143 in order to be able to fix Chromium test RTCPeerConnectionHandlerTest.OnRenegotiationNeeded
Bug: none
Change-Id: I5eac4d9a96c1b594c9e2b3505ef2466046065dc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340481
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41798}
This reverts commit 1cce1d7ddcbde3a3648007b5a131bd0c2638724b.
Reason for revert: Breaks WPTs
Original change's description:
> Make setCodecPreferences only look at receive codecs
>
> which is what is noted in JSEP:
> https://www.rfc-editor.org/rfc/rfc8829.html#name-setcodecpreferences
>
> Some W3C spec modifications are required since the W3C specification
> currently takes into account send codecs as well.
>
> Spec issue:
> https://github.com/w3c/webrtc-pc/issues/2888
> Spec PR:
> https://github.com/w3c/webrtc-pc/pull/2926
>
> setCodecPreferences continues to modify the codecs in an offer.
>
> Also rename RtpSender::SetCodecPreferences to RtpSender::SetSendCodecs for consistent semantics.
>
> BUG=webrtc:15396
>
> Change-Id: I1e8fbe77cb2670575578a777ed1336567a1e4031
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41719}
Bug: webrtc:15396
Change-Id: I7b545e91f820c3affc39841c6e93939eac75c363
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41725}
Using the Api, BWE components are recreated and new settings can be
applied. Initially, the only configuration available is allowing BWE probes without media".
Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration.
Integration test in https://webrtc-review.googlesource.com/c/src/+/337322
Bug: webrtc:14928
Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41687}
This removes several references across the code base that depended on
the global string constants.
Bug: none
Change-Id: I007bd4b195c35261039f655f1a8f52e632c3691f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335320
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41578}
Same method, different name. Unblocks replacing RTCStatsMember<T> with
absl::optional<T>.
Bug: webrtc:15164
Change-Id: I251dd44d3b0f9576b3b68915fe0406d1b3381e5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41573}
This will be done by splitting the use of variables values prior to performing std:move
Bug: webrtc:15771
Change-Id: Ia88e733c3a4edf729e440295ae271d3cd9926ec5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41532}
by also taking into account any a=msid: line in addition to
msid-semantic. Also document issues with msid-semantic generation and unify support determination by removing the msid_supported flag.
BUG=webrtc:10421
Change-Id: Icea554ebd1998f2b526846457029eff6854a772a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329760
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@{#41386}
This is temporary and should be re-enabled as soon as the test is
fixed.
Bug: webrtc:15722
Change-Id: I9d262c9931a19bc9c33f7f93e9e275d39fab403c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330561
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41348}
In preparation for experimentally supporting different types of
metronomes and metronome use cases we'd like to rename for clarity.
This is the first step, which introduces the new name and prefers it if
it is set, but keeps the old name for backwards compat reasons.
Once Chromium has migrated to the new name, we can delete the old name.
Bug: webrtc:15704
Change-Id: I23077bf2415ebb2b2338320c9a14e3bd17d3abb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330020
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41319}
To be submitted after downstream usage has been removed, but no earlier than December 1, 2023.
Bug: webrtc:12598
Change-Id: Id9acbac591c48c0c5883fe8f06cf6a68471b70f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323004
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41290}
This test drives the new tools_webrtc/remove_extra_namespace.py tool.
Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
Add test helper to inject fake media engine for those tests.
Bug: webrtc:15574
Change-Id: Iae4282d2d3b9804548ccadf58797f39508f07c6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325880
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41075}
Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults
Bug: webrtc:15574
Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41059}
This moves steps from the sdp code for pc state over to the PC class
and slightly simplifies the contract between the two classes.
Moving forward it's easier to consolidate those steps in the PC
class with other grouped operations e.g. during teardown.
Also removing GetDataMid() method in favor of the sctp_mid() property.
Bug: none
Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40981}
Speculative fix. Writing the test for it is more complex.
Bug: chromium:1483874
Change-Id: Icfaf1524b0499c609010753e0b6f3cadbd0e98f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321480
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40820}
