2617 Commits

Author SHA1 Message Date
Björn Terelius
78a57efb29 Revert "FrameCadenceAdapter: align video encoding to metronome"
This reverts commit b39c2a8464c48306a495f14beccf431b91e51efd.

Reason for revert: Breaks downstream build

Original change's description:
> FrameCadenceAdapter: align video encoding to metronome
>
> This CL aligns the video encoding tasks to metronome tick which
> similar with the metronome decoding.
>
> Design doc: https://docs.google.com/document/d/18PvEgS-DehClK6twCSCATOlX-j9acmXd-3vjb0tR9-Y
>
> Bug: b/304158952
> Change-Id: I262bd4a5097fdaeed559b9d7391a059ae86e2d63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327460
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
> Cr-Commit-Position: refs/heads/main@{#41469}

Bug: b/304158952
Change-Id: I6f7a3d45cc24b63bc1fe92a93bf5c8d5058f32a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333482
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41471}
2024-01-04 20:02:49 +00:00
Zhaoliang Ma
b39c2a8464 FrameCadenceAdapter: align video encoding to metronome
This CL aligns the video encoding tasks to metronome tick which
similar with the metronome decoding.

Design doc: https://docs.google.com/document/d/18PvEgS-DehClK6twCSCATOlX-j9acmXd-3vjb0tR9-Y

Bug: b/304158952
Change-Id: I262bd4a5097fdaeed559b9d7391a059ae86e2d63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327460
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/main@{#41469}
2024-01-04 04:14:12 +00:00
Philipp Hancke
de17252e8e Reland "Unify access to SDP codec parameters"
This is a reland of commit 63d03f586bb668f72113b61030ec0930aa192010

Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
>   std::map<std::string, std:string>
> with three aliases,
>   cricket::CodecParameterMap
>   SdpAudioFormat::Parameters
>   SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}

Bug: None
Change-Id: I5f8f45688df232eb37b12fa3e56a893a1c754e17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331402
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@{#41467}
2024-01-03 12:03:11 +00:00
Tommi
267f9bdd53 Update LegacyStatsCollector to conform with Wc++11-narrowing
Bug: none
Change-Id: Ida6a1af5c324473a55ea4f3b143862ea016ff50a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332240
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Alexander Kornienko <alexfh@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41432}
2023-12-21 14:58:48 +00:00
Christoffer Dewerin
771b524606 Revert "Delete pc/peerconnection build target"
This reverts commit 18a42e3272a6a25a23042fd39e67de02def8cafb.

Reason for revert: Breaks downstream project.

Original change's description:
> Delete pc/peerconnection build target
>
> It is not useful any more.
>
> Bug: webrtc:13634, b/238176207
> Change-Id: I3dd4ebca355bb828c6c3c30392333d9fe03a478c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267821
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41427}

Bug: webrtc:13634, b/238176207
Change-Id: Ib53e0b0cc81ac218e3c19e4c652ffe0b19155c22
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332220
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Christoffer Dewerin <jansson@google.com>
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#41430}
2023-12-21 12:40:44 +00:00
Harald Alvestrand
18a42e3272 Delete pc/peerconnection build target
It is not useful any more.

Bug: webrtc:13634, b/238176207
Change-Id: I3dd4ebca355bb828c6c3c30392333d9fe03a478c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267821
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41427}
2023-12-21 09:32:50 +00:00
Philipp Hancke
c5d921899b Do no return media-playout stats unless there is an audio receiver
which avoids those stats on datachannel-only or video-only connections.
Note that a receiver always exists, regardless of the transceiver direction.

BUG=None

Change-Id: I1ef33a8446fafe2978ac603e658e67d51d7af904
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330441
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Fredrik Hernqvist <fhernqvist@google.com>
Cr-Commit-Position: refs/heads/main@{#41423}
2023-12-20 16:14:05 +00:00
Judith Hemp
e56055220b Remove expired histograms WebRTC.PeerConnection.Simulcast.NumberOfSendEncodings
Bug: chromium:1508060
Change-Id: I4a66e53d0c59c320e1ca3cb5a7afa3caf1275064
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331840
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Judith Hemp <hempjudith@google.com>
Cr-Commit-Position: refs/heads/main@{#41412}
2023-12-19 09:12:18 +00:00
Marco Paniconi
52da14c44f Re-enable SvcTestAV1/SvcTest.ScalabilityModeSupported/L2T2_DD
Issue has been fixed in latest libaom code rolled into webrtc.

Bug: webrtc:15722
Change-Id: I5e00e202e929703a9af05422884cfb5d0829964b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331862
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41407}
2023-12-18 20:13:15 +00:00
Harald Alvestrand
8f59f54120 Revise the pc:libjingle_peerconnection target
This adds the absolutely required files for this target that is used
by Chrome and others in order to link in all needed libraries, and
removes the dependency on peerconnection.

Bug: webrtc:13634
Change-Id: Ia66f5f627680ce15bcac941998ca1b6da4edb6ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331621
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41395}
2023-12-15 16:21:56 +00:00
Tommi
3ba809d6a6 Reduce locking in DtlsTransport
Access to `internal_dtls_transport_` only occurs on the network thread
and doesn't require locking. Access to `info_` still requires a lock
but writing to it only occurs on the network thread. If reading from
the network thread is needed, that could be done without requiring
the lock.

The scope of holding the lock is much smaller now.

Bug: none
Change-Id: Ic284df04196dfcf8b77c66a48e484ca6893de050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325283
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41387}
2023-12-14 21:01:04 +00:00
Philipp Hancke
6f0f158af0 sdp: make msid support parsing more robust
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}
2023-12-14 14:50:31 +00:00
Mirko Bonadei
6c9c958c69 Revert "Unify access to SDP codec parameters"
This reverts commit 63d03f586bb668f72113b61030ec0930aa192010.

Reason for revert: Breaks downstream project (not backwards compatible API change)

Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
>   std::map<std::string, std:string>
> with three aliases,
>   cricket::CodecParameterMap
>   SdpAudioFormat::Parameters
>   SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}

Bug: None
Change-Id: I841735d98533d3b66850b9cfcf7ee0a99ddde078
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41377}
2023-12-13 16:28:44 +00:00
Philipp Hancke
63d03f586b Unify access to SDP codec parameters
which come from the a=fmtp:<pt> lines in the SDP and were used as either
  std::map<std::string, std:string>
with three aliases,
  cricket::CodecParameterMap
  SdpAudioFormat::Parameters
  SdpVideoFormat::Parameters

Use webrtc::CodecParameterMap in all places.

BUG=None

Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41375}
2023-12-13 14:22:15 +00:00
Philipp Hancke
601ac2eea8 Reject offer content with no common codecs
instead of throwing an error when trying to pick a send codec.

BUG=webrtc:15145,webrtc:4957

Change-Id: I056b145c093348576e1aeaf5def50d5414f2de70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330122
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41360}
2023-12-12 10:04:59 +00:00
Qiu Jianlin
b3488d08db Add SDP negotiation support for HEVC.
This adds neccessary checks for SDP negotiation with HEVC.

Test: Manually apply the CL on Chromium and enable HEVC HW encoder,
and add HEVC profiles in rtc video decoder/encoder factory, H265 is
negotiated in SDP with correct FMTP lines added.

Bug: webrtc:13485
Change-Id: I5557b20b646cc96c5acb578521204fe10df0dcf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330202
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#41357}
2023-12-12 02:09:11 +00:00
Philipp Hancke
cdd92da549 sdp: backfill default codec parameters for H264 and VP9
as preparation for H265 work.

BUG=webrtc:15703

Change-Id: Ib6e0afa5ccbb8172a70d4e4eb876639559070fd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329981
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41350}
2023-12-11 10:48:11 +00:00
Mirko Bonadei
ae86daf830 Skip SvcTestAV1/SvcTest.ScalabilityModeSupported/L2T2_DD.
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}
2023-12-11 07:56:25 +00:00
Per K
86b1cf776e Allow configuring pacer burst through RtcConfiguration
This allow exernal applications to control how many packets can be sent relative current BWE.

This is a partial revert of https://webrtc-review.googlesource.com/c/src/+/311102

Bug: chromium:1354491
Change-Id: Ia236aaacc468ddac12341efa555041bb2dfdde62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330580
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41343}
2023-12-08 15:35:09 +00:00
Emil Lundmark
5e3eb52497 Revert "Enable DD and VLA header extensions by default for Simulcast/SVC"
This reverts commit 33c7edd58ad0edc71939b9372fff3ab563c1f4a7.

Reason for revert: Breaks downstream project

Original change's description:
> Enable DD and VLA header extensions by default for Simulcast/SVC
>
> When Simulcast (more than one encoding) or SVC (a scalability mode
> other than the default L1T1) is used, enable the AV1 Dependency
> Descriptor and the video-layer-allocations RTP header extensions by
> default.
>
> The RTP header extensions API can be used to disable them if needed.
>
> BUG=webrtc:15378
>
> Change-Id: I587ac32c9d681461496a136f6950b007e72da86d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326100
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41332}

Bug: webrtc:15378
Change-Id: I6b5f71f321d30a510db3bd180deaa57732f9349b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41341}
2023-12-08 11:29:46 +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
Philipp Hancke
33c7edd58a Enable DD and VLA header extensions by default for Simulcast/SVC
When Simulcast (more than one encoding) or SVC (a scalability mode
other than the default L1T1) is used, enable the AV1 Dependency
Descriptor and the video-layer-allocations RTP header extensions by
default.

The RTP header extensions API can be used to disable them if needed.

BUG=webrtc:15378

Change-Id: I587ac32c9d681461496a136f6950b007e72da86d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41332}
2023-12-07 08:55:42 +00:00
Henrik Boström
65bee96054 Delete old "metronome" name, API users should use "decode_metronome".
Now that Chromium has migrated to the new name[1], "decode_metronome",
we can delete the variable with the old name, "metronome".

[1] https://chromium-review.googlesource.com/c/chromium/src/+/5093942

Bug: webrtc:15704
Change-Id: I50fef88a692d83e37af10956b2e12389fa601662
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330300
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41331}
2023-12-07 08:04:32 +00:00
Danil Chapovalov
539bca9ebb Cleanup ConnectionContext dependency on field trials
ConntectionContext now keeps and expose field trials as part of the
Environment, and do not need to be aware about field trials specifically

Bug: webrtc:15656
Change-Id: Ib78694a65a9ca7c8bf273eeaf9334323ddb841c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329420
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41328}
2023-12-06 18:01:06 +00:00
Tommi
d6601ce66b Remove PeerConnection::GetRtpTransport
This function isn't used anymore.

Bug: webrtc:9987
Change-Id: I37f1c86cc4802950347db302e8a9207b9dd370bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330261
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41327}
2023-12-06 15:52:37 +00:00
Danil Chapovalov
3d9c3687a4 Delete CallFactoryInterface as no longer needed
Replace CallFactory class with a factory function

Bug: webrtc:15574
Change-Id: Ib1d8cff8d7550da3af01693a7bc117a7bd342258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330000
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41321}
2023-12-05 15:44:43 +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
Danil Chapovalov
c93f4f98a5 Revert^2 "Delete deprecated SetMediaEngineDefaults"
This reverts commit c176175f010a17491a0986a8c2fc67bd48e67315.

Reason for revert: chromium is updated not to depend on the deleted target. (chromium import succeed before the revert)

Original change's description:
> Revert "Delete deprecated SetMediaEngineDefaults"
>
> This reverts commit 1682a7f41135d9529917c0f8e5b6a57fbb47220a.
>
> Reason for revert: Breaks chromium import: https://chromium-review.googlesource.com/c/chromium/src/+/5083877?tab=checks
>
> Original change's description:
> > Delete deprecated SetMediaEngineDefaults
> >
> > Bug: webrtc:15574
> > Change-Id: Ie60973e020ca91ca93ca46159d53d4a89d1757fe
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326004
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41304}
>
> Bug: webrtc:15574
> Change-Id: Id09c8e1682831032e84a83187c6905a84e68d736
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329842
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41312}

Bug: webrtc:15574
Change-Id: Id376c76dbaa069e3cf178b45be7823c1aa9e3789
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329843
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41314}
2023-12-04 19:40:20 +00:00
Ilya Nikolaevskiy
c176175f01 Revert "Delete deprecated SetMediaEngineDefaults"
This reverts commit 1682a7f41135d9529917c0f8e5b6a57fbb47220a.

Reason for revert: Breaks chromium import: https://chromium-review.googlesource.com/c/chromium/src/+/5083877?tab=checks

Original change's description:
> Delete deprecated SetMediaEngineDefaults
>
> Bug: webrtc:15574
> Change-Id: Ie60973e020ca91ca93ca46159d53d4a89d1757fe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326004
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41304}

Bug: webrtc:15574
Change-Id: Id09c8e1682831032e84a83187c6905a84e68d736
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329842
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41312}
2023-12-04 17:05:41 +00:00
Danil Chapovalov
fe66dda733 Delete deprecated call_factory and media_engine
from PeerConnectionFactoryDependencies

Bug: webrtc:15574
Change-Id: Id0ead8086ddd41f6792e2a3c224d8705cd797d49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326003
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41309}
2023-12-04 14:42:44 +00:00
Danil Chapovalov
1682a7f411 Delete deprecated SetMediaEngineDefaults
Bug: webrtc:15574
Change-Id: Ie60973e020ca91ca93ca46159d53d4a89d1757fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326004
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41304}
2023-12-04 11:18:07 +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
Danil Chapovalov
530b243a1d Move TaskQueueFactory ownership into Environment
Now that it is used through the environment both for creating MediaEngine and for creating Calls.

Bug: webrtc:15656
Change-Id: Ib95ee46fe08d9d1ed1ef96bd67189e98052599ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329202
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41286}
2023-11-30 14:07:04 +00:00
Tony Herre
a5c8ee1672 Revert "Make Codec::Matches also consider packetization"
This reverts commit 1ae700a9233ed647e1b4080c0fcb48f61a0cca0a.

Reason for revert: Potential root cause of crbug.com/1504351

Original change's description:
> Make Codec::Matches also consider packetization
>
> If it's not considered it can lead to payload IDs erroneously being
> reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.
>
> Bug: webrtc:15473
> Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41153}

Bug: webrtc:15473 chromium:1504351
Change-Id: I87fb671d76c3b17beb65124603cc040bb9bf4fa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329201
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41285}
2023-11-30 14:06:01 +00:00
Danil Chapovalov
3bdb49b483 Create PeerConnection specific environment
Bug: webrtc:15656
Change-Id: I11616e3470798b43cb07a776f5d58669d629e24d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41283}
2023-11-30 09:54:24 +00:00
Harald Alvestrand
13834cfacd Add callback-based interface to IceTransportInternal GatheringState
This allows both the signal and the callback to be used.

Bug: webrtc:11943
Change-Id: I89460126d415520295c7e7d4ee440156a6e9e5ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329140
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41282}
2023-11-30 09:25:37 +00:00
Danil Chapovalov
49c35d377b In PeerConnection postpone RtcEventLog destruction
This is done as a preparation to move RtcEventLog ownership into Environment where destruction happens later, when all users of the Environment are deleted.

Bug: webrtc:15656
Change-Id: I2a72c74f1fabb1e25c5200aa47a5d61e4b3d9cd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41272}
2023-11-29 11:18:31 +00:00
Harald Alvestrand
09b2fb65da Replace RTCTransportInternal::SignalCandidatePairChange sigslot
and add a callback instead.

Bug: webrtc:12598
Change-Id: I41ee044fc45f15bbf9fc31ba9067cef2a5071faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329060
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41263}
2023-11-28 22:29:24 +00:00
Danil Chapovalov
680f103baa Use Environemnt in MedaFactory::CreateMediaEngine
to propagate field trials and task queue factory

Bug: webrtc:15656
Change-Id: I2d19e169d2ff1cc871899a0e96b1733333fdc604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41257}
2023-11-28 10:30:15 +00:00
Per K
b202bc1db2 Per default set PacingController burst interval to 40ms
PacingController per default use a burst interval of 40ms. The behaviour can still be overriden by  using the method SetSendBurstInterval.

Bug: chromium:1354491
Change-Id: Ie3513109e88e9832dff47380c482ed6d943a2f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311102
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41254}
2023-11-28 07:53:50 +00:00
Danil Chapovalov
7eaa9dc170 Use Environment to keep peer connection factory field trials in ConnectionContext
Bug: webrtc:15656
Change-Id: Ice52fcb9ba54a5d0034b59233ceae4f9cefbceae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41252}
2023-11-27 16:46:27 +00:00
Harald Alvestrand
abc5066bd9 Replace IcetransportInternal::SignalCandidatesRemoved sigslot
with an one-user callback.

Bug: webrtc:11943
Change-Id: Ia61c7811f0058fa7238d47ef13fadfd547f052ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328900
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41250}
2023-11-27 15:06:04 +00:00
Harald Alvestrand
50a238fbd4 Replace IceTransportInternal::SignalCandidateError
with a callback function.

Bug: webrtc:11943
Change-Id: Ieed740a36f86be6dd45d6a495cc4fd023ea98477
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328862
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41249}
2023-11-27 15:04:10 +00:00
Danil Chapovalov
9fdceb80b5 Add environment_construction poison
This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation.

This poison extends and thus replaces default task queue poison.

Bug: webrtc:15656
Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41247}
2023-11-27 11:44:50 +00:00
Philipp Hancke
d0f0f38f72 Remove most usage of MediaContentDescription::as_audio()/as_video()
and unify algorithms a bit more.

BUG=webrtc:15214

Change-Id: Ie9903f3e56d25b1dc026367e8ae6817275faa07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328442
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41244}
2023-11-27 09:35:39 +00:00
Philipp Hancke
0322493aed Refactor MediaSession to unify audio/video codec handling
since the offer/answer rules do not depend on the media type for
the most part. Also make use of recently introduced Codec types.

BUG=webrtc:15214

Change-Id: Ieae27247a8910c3fcaa9609dca0297985907f86a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327740
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41221}
2023-11-23 14:27:54 +00:00
Per K
14630a7e37 Use rtc::ReceivedPacket in Stun and TurnServer
StunServer is updated to ensure registring for receiving packet from the socket is happening on the same thread as where the packets are recevied.

Bug: webrtc:15368, webrtc:11943
Change-Id: I94cc3a47278d5489de7f170c8d43015d1551c437
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328120
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41219}
2023-11-23 10:40:56 +00:00
Per K
7dd6ea234d Disable AV1 L3 and S3 tests
Bug: webrtc:15666
Change-Id: I56d6f28b3e71dc6564cc35265ce2b0ca7e13c40d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328320
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41214}
2023-11-22 15:38:36 +00:00
Tomas Gunnarsson
3a15ba6fbf Reland^2 "Reland: Remove unsupported configuration value, allow_codec_switching"
This reverts commit 117d847901ea231cd86ca152b359b88619b9de20.

Reason for revert: Downstream error has been corrected.

Original change's description:
> Revert "Reland: Remove unsupported configuration value, `allow_codec_switching`"
>
> This reverts commit 23501a2aa656b94e26d4c67b8b9393258551560f.
>
> Reason for revert: Breaks downstream features
>
> Original change's description:
> > Reland: Remove unsupported configuration value, `allow_codec_switching`
> >
> > This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.
> >
> > Reason for revert: Relanding once downstream issues have been addressed
> >
> > Original change's description:
> > > Revert "Remove unsupported configuration value, `allow_codec_switching`"
> > >
> > > This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
> > >
> > > Reason for revert: breaks downstream
> > >
> > > Original change's description:
> > > > Remove unsupported configuration value, `allow_codec_switching`
> > > >
> > > > Bug: webrtc:11341
> > > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > > > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#40995}
> > >
> > > Bug: webrtc:11341
> > > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> > > Owners-Override: Philip Eliasson <philipel@webrtc.org>
> > > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#40998}
> >
> > Bug: webrtc:11341
> > Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41032}
>
> Bug: webrtc:11341
> Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41161}

Bug: webrtc:11341
Change-Id: I4a5390a3b8c5e665b742fc564709847ad8853ba9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328160
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41213}
2023-11-22 13:22:08 +00:00
Harald Alvestrand
572502c2ab Deprecate char* functions on ByteBufferReader
Bug: webrtc:15661, webrtc:15665
Change-Id: Ia35b0092c219a89b5eba08d2e1a91be6e47dc746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328000
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41210}
2023-11-22 11:46:25 +00:00