628 Commits

Author SHA1 Message Date
Jeremy Leconte
e25b15e22b Update ownership of PCLF documentation.
Change-Id: I87fe1b4c0f72cc164e94ae63e7f544c0d15f39ab
Bug: b/362910201, b/362909489
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362340
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#43006}
2024-09-11 13:03:08 +00:00
Artem Titov
3652dd30db Review documentation and update review date
Bug: b/362492031, b/362492070, b/362492356, b/364207037, b/364831690
Change-Id: Ic889c731b98f8876c0ee31c0bda91a5a18b3add1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362100
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42981}
2024-09-09 11:21:55 +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
Dor Hen
1921fa5ea1 Apply include-cleaner to api/test/[^/]*
e.g all files in the api/test folder not including subdirectories

Bug: webrtc:42226242
Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42773}
2024-08-13 15:28:34 +00:00
Dor Hen
8aa4e7c453 Apply include-cleaner to api/test/(video|pclf|network_emulation)/.*
Bug: webrtc:42226242
Change-Id: I28dde76246f9c10e61cd6f294278edd364513267
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42770}
2024-08-13 10:25:20 +00:00
Emil Vardar
59c7b2277b Log received frames QP value.
spatial_layers_qp holds the QP values for all the spatial layers. However, receiver only sees one of the spatial layers at a given time. It is of interest to know the shown spatial layers QP. This could also be used to indicate if upper layers are frequently dropped or not.

Bug: None
Change-Id: I462ea11e3447f8ffd11f4a6f2ccbf361102c762f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358863
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42759}
2024-08-12 08:58:25 +00:00
Mirko Bonadei
b1b6129944 Revert "Each spatial layer can only have 1 QP value."
This reverts commit 962b3935e44053641764beb0bd095540fe8cbd64.

Reason for revert: Breaks downstream tests.

Original change's description:
> Each spatial layer can only have 1 QP value.
>
> As the code is now, it looks like it accepts that a spatial layer can have more than 1 QP value. These QP values according to the code are summed. However, to my best knowledge this cannot be the case and makes the code hard to read. Therefore, updating it with a check such that it checks that each spatial layer only have 1 QP value and would be easier for a future code reader.
>
> Bug: webrtc:357636606
> Change-Id: I650cac724811a1ddc7ab8933c1e1ac5fe844b61c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358101
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
> Cr-Commit-Position: refs/heads/main@{#42736}

Bug: webrtc:357636606
Change-Id: I60a2d4e1285f961f2ed2ea4c1d2e5942ea68b365
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358721
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42737}
2024-08-07 08:42:20 +00:00
Emil Vardar
962b3935e4 Each spatial layer can only have 1 QP value.
As the code is now, it looks like it accepts that a spatial layer can have more than 1 QP value. These QP values according to the code are summed. However, to my best knowledge this cannot be the case and makes the code hard to read. Therefore, updating it with a check such that it checks that each spatial layer only have 1 QP value and would be easier for a future code reader.

Bug: webrtc:357636606
Change-Id: I650cac724811a1ddc7ab8933c1e1ac5fe844b61c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358101
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42736}
2024-08-07 07:59:22 +00:00
Florent Castelli
0012bfa128 Change DataChannelInit::priority to integer and forward to SCTP transport
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}
2024-07-30 15:07:25 +00:00
Artem Titov
eb3da2b1ec Extract video writing into separate target
Bug: None
Change-Id: I3af192606eb623e21a4d648fb69bb62c14ab8b0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42541}
2024-06-26 12:47:15 +00:00
Dor Hen
aefed55c25 [iwyu][1\n] Applying to api/[a-s]*
First batch of applying iwyu to the repo.
Done with:
> ./tools_webrtc/iwyu/apply-iwyu api
> git add api/[a-s]*
> python3 gn_autodeps.py ~/local/webrtc/src out/Default

Last step is a custom script I wrote to automatically apply new required
dependencies for target in gn, which saved tons of time manually going
over the files and fixing.
If this is something that interest others, I can submit it as well.

Bug: webrtc:42226242
Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42512}
2024-06-19 06:19:20 +00:00
Florent Castelli
99c519b3fd Mass removal of absl_deps in all BUILD.gn files
Bug: webrtc:341803749
Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 15:09:46 +00:00
Per K
5566b91356 Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
This reverts commit ff2dd50fd88e07affc4b070ce535935409f6673a.

Reason for revert: Temporary fix for downstream breakage in patch 2

Original change's description:
> Revert "Replace usage of link_capacity_kbps with DataRate link_capacity"
>
> This reverts commit 6186c0226e41dabbfc5cc8527e6b350b62f39f02.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > Replace usage of link_capacity_kbps with DataRate link_capacity
> >
> > 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}
>
> Bug: webrtc:14525
> Change-Id: I09ede3e89d065061cb4133bff96dbf242ff70832
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350621
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42309}

Bug: webrtc:14525
Change-Id: Ie35cd97a158d008a80ed007b27d2c6b1a9affff9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350541
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42320}
2024-05-16 10:39:10 +00:00
Jeremy Leconte
f9da667dca [DVQA] Sum the encoded_image_size for simulcast scenario.
This aligns actual_encode_bitrate with target_encode_bitrate

Change-Id: I8c460a5883e5eacbee8e67b5b6625f65792efb77
Bug: b/340790240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42310}
2024-05-15 11:23:45 +00:00
Mirko Bonadei
ff2dd50fd8 Revert "Replace usage of link_capacity_kbps with DataRate link_capacity"
This reverts commit 6186c0226e41dabbfc5cc8527e6b350b62f39f02.

Reason for revert: Breaks downstream project.

Original change's description:
> Replace usage of link_capacity_kbps with DataRate link_capacity
>
> 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}

Bug: webrtc:14525
Change-Id: I09ede3e89d065061cb4133bff96dbf242ff70832
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350621
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42309}
2024-05-15 11:09:33 +00:00
Per K
6186c0226e Replace usage of link_capacity_kbps with DataRate link_capacity
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}
2024-05-15 08:44:20 +00:00
Sergey Sukhanov
26a082ce36 Introduce a mode that lets NetworkEmulationManager ignore DTLS handshake sizes.
Bug: b/169531206
Change-Id: I02c19385ff7078944f7509ecc07358b4315f7b08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350181
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42261}
2024-05-08 13:20:20 +00:00
Per K
569849e885 Move call/simulated_network to test/network
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}
2024-04-29 09:55:06 +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
Per K
29abba982c Cleanup WebRTC-SendPacketsOnWorkerThread
Experiment has been concluded and cleaned up.

Bug: webrtc:14502
Change-Id: I7f892538dc676056ca2e8969a1ef81ffa3d40014
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347645
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42095}
2024-04-17 11:20:58 +00:00
Victor Boivie
cdecc4e6df Expose bufferedAmountLowThreshold
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}
2024-03-21 19:59:39 +00:00
Jeremy Leconte
1a37aa197e Fix frame not found error when encoder is paused.
The problem occurs when a frame is sent again because the encoder was paused but the frame has already been received by all participants:
https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=2322

Change-Id: If8890986301c44a472db9bc4750d23761c150669
Bug: b/328175783
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343560
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41931}
2024-03-19 17:16:19 +00:00
Danil Chapovalov
dcc95081e1 Cleanup QualityAnalyzingVideoEncoderFactory::CreateVideoEncoder
And thus require Environment to be propagated to this test helper

Bug: webrtc:15860
Change-Id: Ia4796d7a6a8e6f5dcb947899617df43e991419e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343181
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41910}
2024-03-15 15:24:54 +00:00
Victor Boivie
fea41f540c pc: Include SCTP queued bytes in buffered_amount
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}
2024-03-13 15:44:17 +00:00
Per K
0fa90887c5 Deprecate VideoFrame::timestamp() and set_timestamp
Instead, add rtp_timestamp and set_rtp_timestamp.

Bug: webrtc:13756
Change-Id: Ic4266394003e0d49e525d71f4d830f5e518299cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342781
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41894}
2024-03-13 11:08:37 +00:00
Harald Alvestrand
afaae4e38a Remove remaining .cc files from rtc_media_base
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}
2024-03-12 14:09:38 +00:00
Danil Chapovalov
329f0ead43 Provide Environment when creating VideoEncoder in test code
Bug: webrtc:15860
Change-Id: I8c79ff58619716842e02f33e78a0529c631494e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342280
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41884}
2024-03-12 11:09:31 +00:00
Victor Boivie
cd54fd8606 sctp: Pass webrtc::Environment to DcSctpTransport
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}
2024-03-08 09:45:12 +00:00
Philipp Hancke
bbff58d935 Introduce "well-known" SdpVideoFormat codecs
describing video codecs with their parameters as static members of SdpVideoFormat:
  static const SdpVideoFormat VP8();
  static const SdpVideoFormat H264();
  static const SdpVideoFormat VP9Profile0();
  static const SdpVideoFormat VP9Profile1();
  static const SdpVideoFormat VP9Profile2();
  static const SdpVideoFormat VP9Profile3();
  static const SdpVideoFormat AV1Profile0();
  static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.

BUG=webrtc:15703

Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
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@{#41833}
2024-02-28 06:57:10 +00:00
Harald Alvestrand
3bddaed569 rtc_p2p: Split turn port and basic port allocator
This completes the breakup of the rtc_p2p target.
Remaining cleanup is to delete the rtc_p2p target and make clients
depend on the base targets.

Bug: webrtc:15796
Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-07 10:30:59 +00:00
Harald Alvestrand
8bb54c1c42 Penultimate split-up of rtc_p2p build target
This takes the rest of the .cc files out of the rtc_p2p build
target, leaving only one entangled target to clean up.

Bug: webrtc:15796
Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 17:52:39 +00:00
Philipp Hancke
bda5cc63ce Clean up use of WebRTC-UseStandardBytesStats trial in tests
BUG=webrtc:10525

Change-Id: Ia0ec88d5b561ec98af540f849182805d49a327e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337520
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41663}
2024-02-05 09:24:01 +00:00
Danil Chapovalov
62cee88e4b Propagate Environment through QualityAnalyzingVideoDecoderFactory
Bug: webrtc:15791
Change-Id: I9eddf7bf9fb66ee70495e9bc3c810126e2015287
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336800
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41630}
2024-01-29 20:11:46 +00:00
Harald Alvestrand
a310d78662 Refactor a lot of the p2p:rtc_p2p target
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.

One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.

Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 18:28:27 +00:00
Henrik Boström
7209548090 Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb

Previously reverted due to an importer issue (b/320646178) and later a
dependency on RTCStatsMember<T>::ValueToString().

In this reland, we add Attribute::ToString() but we don't delete the
RTCStatsMember<T> stringifier methods, allowing downstream to migrate
before they are deleted.

Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}

Bug: webrtc:15164
Change-Id: I281ccf5b23d8f194b5ce00186a32846c757b46fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41575}
2024-01-19 14:42:10 +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
Mirko Bonadei
4d706a9fd1 Revert "Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson.""
This reverts commit 55cdc29b9d7259d17ccc281855dd21adc51ca957.

Reason for revert: Breaks downstream project.

Original change's description:
> Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
>
> This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb
>
> Original change's description:
> > [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
> >
> > Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> > Attribute::ToString().
> >
> > The difference between "ToString" and "ToJson" is that the "ToJson"
> > version converts 64-bit integers and doubles to floating points with no
> > more than ~15 digits of precision as to not exceed JSON's precision
> > limitations. So only in edge cases of really large numbers or numbers
> > with a silly number of digits will the two methods produce different
> > results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> > as opposed to "{foo:123}".
> >
> > Going forward we see no reason to maintain two different string
> > converted paths that are this similar, so we only implement one
> > Attribute::ToString() method which does what "ToJson" did.
> >
> > In the next CL we can delete RTCStatsMember<T>.
> >
> > Bug: webrtc:15164
> > Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Evan Shrubsole <eshr@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41544}
>
> Bug: webrtc:15164
> Change-Id: If34509ebf3d7c0291442ae11596e7c2d3978fb64
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335240
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41566}

Bug: webrtc:15164
Change-Id: I5819811237a6dbd85a8c738ca0180039fc705909
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335280
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41567}
2024-01-19 08:03:48 +00:00
Henrik Boström
55cdc29b9d Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb

Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}

Bug: webrtc:15164
Change-Id: If34509ebf3d7c0291442ae11596e7c2d3978fb64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335240
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41566}
2024-01-19 07:21:51 +00:00
Henrik Boström
df0b363cf0 Reland "[Stats] Add value_or() and migrate from ValueOrDefault()."
This is a reland of commit 9e4a97bb02663604b02e219b9d501a8dd91b5614

Original change's description:
> [Stats] Add value_or() and migrate from ValueOrDefault().
>
> Yet another prerequisite for replacing RTCStatsMember<T> with
> absl::optional<T>, but this looks like the last one.
>
> Bug: webrtc:15164
> Change-Id: I2cde51e8c8c951f71b48ccd45e07146091a99616
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334647
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41541}

Bug: webrtc:15164
Change-Id: I5fdba499383e5d9efe0a1dcef6bf6c2e0a812857
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335102
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41564}
2024-01-18 22:07:18 +00:00
Mirko Bonadei
111e381822 Revert "[Stats] Add value_or() and migrate from ValueOrDefault()."
This reverts commit 9e4a97bb02663604b02e219b9d501a8dd91b5614.

Reason for revert: Breaks downstream project

Original change's description:
> [Stats] Add value_or() and migrate from ValueOrDefault().
>
> Yet another prerequisite for replacing RTCStatsMember<T> with
> absl::optional<T>, but this looks like the last one.
>
> Bug: webrtc:15164
> Change-Id: I2cde51e8c8c951f71b48ccd45e07146091a99616
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334647
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41541}

Bug: webrtc:15164
Change-Id: I89af6470c82d07981d8d064aa6ff8b50fae42b12
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334801
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41548}
2024-01-17 14:38:25 +00:00
Mirko Bonadei
1fee69cfff Revert "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
This reverts commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb.

Reason for revert: Breaks downstream project.

Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}

Bug: webrtc:15164
Change-Id: I187d7dff6f330a4a440279e6c32d88eb6ddefac8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334820
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41546}
2024-01-17 14:06:34 +00:00
Henrik Boström
54be7084e0 [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
Attribute::ToString().

The difference between "ToString" and "ToJson" is that the "ToJson"
version converts 64-bit integers and doubles to floating points with no
more than ~15 digits of precision as to not exceed JSON's precision
limitations. So only in edge cases of really large numbers or numbers
with a silly number of digits will the two methods produce different
results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
as opposed to "{foo:123}".

Going forward we see no reason to maintain two different string
converted paths that are this similar, so we only implement one
Attribute::ToString() method which does what "ToJson" did.

In the next CL we can delete RTCStatsMember<T>.

Bug: webrtc:15164
Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41544}
2024-01-17 12:36:46 +00:00
Henrik Boström
9e4a97bb02 [Stats] Add value_or() and migrate from ValueOrDefault().
Yet another prerequisite for replacing RTCStatsMember<T> with
absl::optional<T>, but this looks like the last one.

Bug: webrtc:15164
Change-Id: I2cde51e8c8c951f71b48ccd45e07146091a99616
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334647
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41541}
2024-01-17 10:35:14 +00:00
Danil Chapovalov
972546ed50 Delete TestPeerFactory constructor that uses rtc::TaskQueue
Bug: webrtc:14169
Change-Id: Id1414ecc19c8ff9da826688684003560d9a3139d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334642
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41536}
2024-01-16 17:29:58 +00:00
Danil Chapovalov
e052eee7a3 Deprecate rtc::TaskQueue variant of AudioProcessing::CreateAndAttachAecDump
Bug: webrtc:14169
Change-Id: I63f40ec18b72cba89eb0b9b298f448ce7f7c4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334201
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41528}
2024-01-15 13:36:35 +00:00
Dor Hen
764ac7ec0a Allowing to set PCF options via peer configurer
Bug: webrtc:15752
Change-Id: I408cf2e118d09504d59a09ef4c2767ab89982db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332645
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41462}
2024-01-02 10:59:05 +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
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
Danil Chapovalov
4fd1cc70da Add EnableMediaWithDefaultsAndTimeController
To replace CreateTimeControllerBasedCallFactory

Update webrtc tests to use this new function

Bug: webrtc:15574
Change-Id: I2b74cd930ecc4f72dd1e7aa853764ca298b66ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325527
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41076}
2023-11-03 13:26:32 +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