96 Commits

Author SHA1 Message Date
Harald Alvestrand
4aeab708bb Reland "Fix codec collision on reoffer after munged codec on offer."
This reverts commit 20bd702ebeb13a709832463fe5aadd623b7dc71b.

Reason for revert: Fixed test to not fail when AV1 is missing

Original change's description:
> Revert "Fix codec collision on reoffer after munged codec on offer."
>
> This reverts commit b9ddaa154b91b5d1cbe38bf38fce544a87e00d1a.
>
> Reason for revert: Downstream failure.
>
> Original change's description:
> > Fix codec collision on reoffer after munged codec on offer.
> >
> > Bug: chromium:395077842
> > Change-Id: I7665e593fa0f6883150363cb75103facd62f4fea
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377141
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43889}
>
> Bug: chromium:395077842
> Change-Id: I10184a0d521add123838861a5c5e7929864537bb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43901}

Bug: chromium:395077842
Change-Id: I3ba5cacebc7eb608edffea2de54cf1c1e9355a81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377281
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43907}
2025-02-17 11:46:42 -08:00
Harald Alvestrand
20bd702ebe Revert "Fix codec collision on reoffer after munged codec on offer."
This reverts commit b9ddaa154b91b5d1cbe38bf38fce544a87e00d1a.

Reason for revert: Downstream failure.

Original change's description:
> Fix codec collision on reoffer after munged codec on offer.
>
> Bug: chromium:395077842
> Change-Id: I7665e593fa0f6883150363cb75103facd62f4fea
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377141
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43889}

Bug: chromium:395077842
Change-Id: I10184a0d521add123838861a5c5e7929864537bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43901}
2025-02-17 04:40:30 -08:00
Harald Alvestrand
b9ddaa154b Fix codec collision on reoffer after munged codec on offer.
Bug: chromium:395077842
Change-Id: I7665e593fa0f6883150363cb75103facd62f4fea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377141
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43889}
2025-02-13 21:43:26 -08:00
Danil Chapovalov
3164c2a4eb Restructure PeerConnection tests not to create PortAllocator directly
Instead rely on PeerConnectionFactory to create it.

Bug: webrtc:42232556
Change-Id: I24c9842ef027604b840188306db3e29756e1925f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43849}
2025-02-05 04:08:12 -08:00
Evan Shrubsole
fa73a2ed79 Convert timeouts in integration_test_helpers to TimeDelta
Bug: webrtc:42223979
Change-Id: Ia77b34c5c30a32fcb520359b993ff0b976be378c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374880
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43771}
2025-01-20 02:58:26 -08:00
Evan Shrubsole
3e8e4784ac Replace gunit.h macros with WaitUntil in pc/
Bug: webrtc:381524905
Change-Id: I15946ab73aaef2e830d6801451636e717708adbf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373704
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43680}
2025-01-08 05:34:50 -08:00
Jonas Oreland
ac40185001 DTLS 1.3 - patch 2
- add DTLS1.3 ciphers (without KeyType)
- remove code in dtls_transport.cc that tries to parse DTLS packet
- cleanup some test
- start on test for packet loss during dtls handshake (more to come!)

After this patch is submitted, it is possible
to set max version = dtls1.3 and it will active
but DON'T do it yet.

BUG=webrtc:383141571

Change-Id: I6f9a120c53415ccee7a560ea83bd0c2636702997
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371300
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43595}
2024-12-18 02:26:22 -08:00
Danil Chapovalov
acf26ce00a Refactor PC tests to use non-global field trials
In particular that avoids lifetime issues with the field trials passed into peerconnection, as now PC takes field trials object by unique_ptr and thus fully manages its lifetime.

Bug: webrtc:42220378
Change-Id: Ia863e9703b5c76ae1866d0ff995b83286c0b947e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371480
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43576}
2024-12-16 05:29:01 -08:00
Philipp Hancke
316d93b415 test: do not use SDP munging to enable corruption detection
BUG=webrtc:358039777

Change-Id: Ibe3fc1f230185b542ee6312596a31d94c3c9156e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370713
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43561}
2024-12-13 09:15:51 -08:00
Harald Alvestrand
882b32d00f Reland "Use PayloadTypePicker for video PT assignment"
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.

Reason for revert: Revised codec matching to fix issue.

Changes also back out some changes that should not have been
included (using PayloadTypePicker for codec list merging).

Original change's description:
> Revert "Use PayloadTypePicker for video PT assignment"
>
> This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
>
> Reason for revert: Broke internal tests.
>
> Original change's description:
> > Use PayloadTypePicker for video PT assignment
> >
> > This includes changes that change the order of codecs.
> > It is preparatory to doing late assignment of video PTs.
> >
> > Bug: webrtc:360058654
> > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43489}
>
> Bug: webrtc:360058654
> Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43490}

Bug: webrtc:360058654
Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43554}
2024-12-12 16:37:30 -08:00
Jonas Oreland
575d323671 Fix dcsctp handling of dtls restart
dtls_transport will when detecting a new fingerprint
(e.g by usage of pranswer) signal DtlsTransportState::kNew.
When this happen, the dtls crypto state is lost, and
sctp should reconnect, srtp does this automatically
in current code base.

The existing behavior in dcsctp is that it will detect
peer sending an init, and reconnect. But any messages sent
between the dtls restart and the message arriving from the
peer will be lost.

This patch changes so that this case is gracefully handled by
a) letting dcsctp_transport listen to dtls state
this is big part of patch and involves changing the type of
the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this
into a separate patch...

b) if a dtls restart happens, delete and restart socket.

Testcase that fails before patch and works after is attached.
Bonus: And include-what-you-use on patch

Bug: b/375327137
Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43546}
2024-12-12 02:47:01 -08:00
Harald Alvestrand
e046787a5a Revert "Use PayloadTypePicker for video PT assignment"
This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.

Reason for revert: Broke internal tests.

Original change's description:
> Use PayloadTypePicker for video PT assignment
>
> This includes changes that change the order of codecs.
> It is preparatory to doing late assignment of video PTs.
>
> Bug: webrtc:360058654
> Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43489}

Bug: webrtc:360058654
Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43490}
2024-12-03 22:24:21 +00:00
Harald Alvestrand
e5048949b0 Use PayloadTypePicker for video PT assignment
This includes changes that change the order of codecs.
It is preparatory to doing late assignment of video PTs.

Bug: webrtc:360058654
Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43489}
2024-12-03 18:18:28 +00:00
Harald Alvestrand
0c6d31919e Enable RFC 8888 feedback if negotiated
This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated.

This should eliminate the need for the "force" flag in the field trial.

Bug: webrtc:42225697
Change-Id: Iec7a894c244a417a8499200861550a33f89966a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43398}
2024-11-14 06:27:45 +00:00
Emil Vardar
416cb498cc Rename corruption related metrics according to WebRTC's Statistics API.
See https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalcorruptionprobability for more details.

Bug: webrtc:358039777
Change-Id: I34236b9423864008486a9f9949f46397ff8b9f92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367960
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43379}
2024-11-08 11:57:59 +00:00
Jakob Ivarsson
68f4e27794 Add RtpSender OnFirstPacketSent callback.
It works in the same way as the first packet received callback and can be used for latency measurements.

One important detail is that RTCP and probe packets are excluded from triggering the callback.

Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
2024-10-25 16:17:04 +00:00
Harald Alvestrand
b7abaee819 Revert "Use Payload Type suggester for all codec merging"
This reverts commit 0bac2aae596771db020f01a57fee4828081fbc38.

Reason for revert: Suspected breakages downstream

Original change's description:
> Use Payload Type suggester for all codec merging
>
> Bug: webrtc:360058654
> Change-Id: Id475762253c427c1800c2352a60fc0121c2dc388
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364783
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43267}

Bug: webrtc:360058654, b/375132036
Change-Id: Ieda626270193e7e6c93903b3c03a691b2bf0c1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43290}
2024-10-23 11:37:18 +00:00
Emil Vardar
a2205e3943 Propagate the corruption_score metric to RTCInboundRtpStreamStats.
Bug: webrtc:358039777
Change-Id: I7e956188a5ef913cbe1647d00ca02b5a46a99b3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362083
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43281}
2024-10-22 12:53:14 +00:00
Harald Alvestrand
0bac2aae59 Use Payload Type suggester for all codec merging
Bug: webrtc:360058654
Change-Id: Id475762253c427c1800c2352a60fc0121c2dc388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364783
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43267}
2024-10-18 16:58:42 +00:00
Danil Chapovalov
9c21f6386f Replace AudioProcessingBuilderForTesting with the BuiltinAudioProcessingFactory
Bug: webrtc:369904700
Change-Id: Ie96dc1a9c052cb5340b10bf834d95f88f0a96a14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365801
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43247}
2024-10-16 10:55:38 +00:00
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Jonas Oreland
a49abbb3b6 Extend testing of prAnswer
- Modify munger to take (mutable)
  std::unique_ptr<SessionDescriptionInterface> rather than
  cricket::SessionDescription (that latter is embedded in the former)

- For all pranswer test cases, do a final SetRemoteDescription(kAnswer) and
check that signaling_state == stable

Add new test cases:
1) A test case that only applies it as prAnswer on caller (callee is stable)
2) A test case that "scramble" sdb between prAnswer and Anser.

Bug: None
Change-Id: Ifedd92ade01ae781a2e59d0569133c486c7093fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42891}
2024-08-30 08:06:47 +00:00
Harald Alvestrand
90e0829c59 Add test for PR-Answer functionality
Bug: None
Change-Id: I29bf1e40d47361917eb6f52424df23f7697bde0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360721
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42859}
2024-08-27 08:17:32 +00:00
Philipp Hancke
4158678b46 Split "helpers" from SSL target to "crypto_random" and rename
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}
2024-06-07 06:41:51 +00:00
Harald Alvestrand
1a3120f3fd Move some integration test functions to the .cc file
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}
2024-04-25 07:25:42 +00:00
Florent Castelli
f4673f97ed Move webrtc::AudioDeviceModule include to api/ folder
Bug: webrtc:15874
Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-22 08:56:31 +00:00
Florent Castelli
0afde7614d Move webrtc::AudioProcessing include to api/ folder
Bug: webrtc:15874
Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-20 07:02:50 +00:00
Florent Castelli
15e46aa358 pc: Increase timeout for EndToEndCallWithSctpDataChannelFullBuffer
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}
2024-03-27 11:09:05 +00:00
Tommi
0a7fc84887 Use is_* getters when checking the Candidate type
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}
2024-01-19 15:37:32 +00:00
Henrik Boström
ed1d084d0a [Stats] Replace all uses of is_defined() with has_value().
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}
2024-01-19 12:26:56 +00:00
Danil Chapovalov
151003d341 Deprecate RtcEventLogFactory constructor taking unused parameter
Bug: webrtc:15656
Change-Id: I22ed4cca4c0ce7ebf9c533ed7434617bf0a0f4a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41338}
2023-12-07 21:46:56 +00:00
Henrik Boström
f887e07234 Rename "metronome" to "decode_metronome".
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}
2023-12-05 15:00:54 +00:00
Harald Alvestrand
24510d43dc Delete deprecated AsyncResolver and related classes
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}
2023-11-30 15:36:55 +00:00
Harald Alvestrand
a6544377bc Remove not-needed webrtc:: prefixes in pc/
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}
2023-11-13 13:23:04 +00:00
Danil Chapovalov
166111da62 Migrate PeerConnectionIntegrationWrapper to EnableMedia api
Bug: webrtc:15574
Change-Id: I164916b6ba9d29519660b119ed38580c478ea7f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325528
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41064}
2023-11-02 13:33:18 +00:00
Danil Chapovalov
554f7db01c Add EnableMediaWithDefaults to replace SetMediaEngineDefaults
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}
2023-11-01 11:47:59 +00:00
Philipp Hancke
f16e139357 Generalize ssrc-group check to apply to groups other than SIM
BUG=chromium:1477075

Change-Id: I20f094dee11ea26a180471ce52d78d916f922f29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322440
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40888}
2023-10-09 05:59:48 +00:00
Tommi
3756e29b15 Remove another ctor from BasicPortAllocator
This constructor isn't used in production. Removing it further
made the construction state of the class simpler, allowed for removal
of the separate Init() method and making more members const.

Bug: none
Change-Id: Ibc8516a01ce7e385207251d841d21bb7b72c9d9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318281
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40678}
2023-09-01 11:55:43 +00:00
Harald Alvestrand
96e1882860 Convert AsyncDnsResolver to use absl::AnyInvocable
Bug: webrtc:12598
Change-Id: I0950231d6de7cf53116a573dcd97a3cf5514946c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40670}
2023-08-31 08:50:40 +00:00
Harald Alvestrand
4d25a77fd3 Deprecate AsyncResolver config fields and remove internal usage.
Bug: webrtc:12598
Change-Id: Ic43cbcd13e4de44b02351c89da12844606368623
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317604
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40627}
2023-08-25 14:02:27 +00:00
Henrik Boström
9f3ea9d934 Increase concealement threshold for debug bots.
Internal bots are flaking, 0.96666666666666667 concealement observed.

Bug: b/294020344
Change-Id: I65ff8d1dcfe52ba4c8024736cf203005d5c1e4f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314541
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40508}
2023-08-03 14:08:37 +00:00
Johannes Kron
4133797557 Remove expired histograms WebRTC.PeerConnection.SrtpCryptoSuite
Fixed: chromium:1448119
Change-Id: Ibf903626f78860e2fb33e5f58b37276c106fdcbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308380
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40254}
2023-06-09 14:48:38 +00:00
Henrik Boström
4e231eedbd Delete deprecated 'track' and 'stream' metrics from WebRTC.
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}
2023-05-24 12:26:56 +00:00
Tommi
c848268ab1 Use SequenceChecker(SequenceChecker::kDetached) in a few places.
This CL is partly a test to see if there's an impact on binary size:
- Not a big difference for binaries (decrease): -776b to -4Kb
- For libraries (libwebrtc.a) it actually increases the size: +40Kb

Secondarily this CL is basically to introduce this pattern to the
code base. In terms of LOC, this makes things slightly more compact.

From:

  class Foo {
   public:
     Foo() {
       checker_.Detach();
     }
   private:
    SequenceChecker checker_;
  };

To:

  class Foo {
   public:
     Foo() = default;
   private:
    SequenceChecker checker_{SequenceChecker::kDetached};
  };

Bug: none
Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39664}
2023-03-24 07:44:18 +00:00
Harald Alvestrand
041ecb87f5 New PeerConnectionFactory::CreateVideoTrack with refcounted source
Bug: webrtc:15017
Change-Id: I04c794d8959583bb4cc5c3898f4175783ec49f16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249363
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39635}
2023-03-22 09:10:27 +00:00
Philipp Hancke
1f98b466b8 stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats
to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.

The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.

BUG=webrtc:14973

Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
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@{#39497}
2023-03-07 14:27:47 +00:00
Henrik Boström
323e9df6d1 Remove dependencies on 'track' stats from PC integration tests.
This unblocks the deletion of this deprecated stats object.

Bug: webrtc:14175
Change-Id: I850c028fc9556a36191909afa3d635a7e6b65b69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288582
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38983}
2023-01-03 11:40:28 +00:00
Markus Handell
15a82c93d0 Metronome: complete API migration.
This CL finalizes the Metronome refactor undertaken in
crbug.com/1381982 and enables it again in call.cc.

Fixed: chromium:1381982
Change-Id: I1642103e9c8a3f2a1f12d7635a1b27310802c1c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282920
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38605}
2022-11-10 13:42:30 +00:00
Markus Handell
be400e465b Metronome: disable & refactor for single-threaded operation.
The Chromium implementation unfortunately has a rare deadlock.
Rather than patching that up, we're changing the metronome
implementation to be able to use a single-threaded environment
instead.

The metronome functionality is disabled in VideoReceiveStream2
construction inside call.cc.

The new design does not have listener registration or
deresigstration and instead accepts and invokes callbacks, on
the same sequence that requested the callback. This allows
the clients to use features such as WeakPtrFactories or
ScopedThreadSafety for cancellation.

The CL will be followed up with cleanup CLs that removes
registration APIs once downstream consumers have adapted.

Bug: chromium:1381982
Change-Id: I43732d1971e2276c39b431a04365cd2fc3c55c25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38582}
2022-11-08 12:23:40 +00:00
Henrik Boström
15166b2fa4 [ModernStats] Mark obsolete stats as [[deprecated]].
This includes the stats dictionaries that have been made obsolete in
the spec and whose IDs are prefixed "DEPRECATED_":
- RTCMediaStreamTrackStats
- RTCMediaStreamStats

There is an ongoing experiment to unship these stats dictionaries in
Chrome (https://crbug.com/1374215). Marking then as [[deprecated]] helps
alert other dependencies that these classes are deprecated.

In the meantime, the "DEPRECATED_RTCMediaStreamTrackStats" prefix makes
it possible to use the deprecated classes.

# Unrelated infra failures
NOTRY=True

Bug: webrtc:14175, webrtc:14419
Change-Id: I498d370294058a628278e1e5b027cd12e24ad31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38439}
2022-10-19 09:58:37 +00:00