Commit Graph

2210 Commits

Author SHA1 Message Date
b5671da00d [PCLF] Remove previously replaced configuration options
VideoSimulcastConfig::encoding_params and VideoConfig::min/max_encode_bitrate_bps
were replaced by VideoConfig::encoding_params.
All usage of the previous options has been updated to the new option.

Bug: webrtc:11607
Change-Id: I52cd9efa6e640929485da7aa1e61d15a1a693b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261949
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36865}
2022-05-12 14:26:53 +00:00
8c6d88ff5c api: Add MockDtmfSender
Similar mocks are used internally, there should only be one in WebRTC.

Bug: None
Change-Id: Ic5163ae0c554c602344a0d25d17c3f0d46fc2e95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261955
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36864}
2022-05-12 13:11:15 +00:00
1f49157b41 stats: implement transport iceState
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-selectedcandidatepairid

BUG=webrtc:14022

Change-Id: I206bff7048d2df3e3aff0af55072097f49d54e8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261720
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36840}
2022-05-10 13:55:21 +00:00
adaf511221 Extend UnitBase multiplication to support size_t
Bug: None
Change-Id: I8dcb85cdb2819df54d4cb0cae59b77d7d629123a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260941
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36828}
2022-05-10 07:41:30 +00:00
14d01508be Move VP8 SupportsScalabilityMode utility to its own build target
Intended to let Vp8TemporalLayersFactory (an api/ target) reuse
this function, without depending on the codec implementation, and
without introducing a dependency cycle with the webrtc_vp8 build
target.

Bug: webrtc:11607
Change-Id: I671422e994e1005da8c7d768e8dd8ff795553e51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36816}
2022-05-09 13:25:25 +00:00
ea1e6f44f8 Delete rtc_base/format_macros.h
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.

Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
2022-05-09 12:03:21 +00:00
42a829e623 Delete implicit conversion from rtc::scoped_refptr<T> to T*
Bug: webrtc:13464
Change-Id: I24c742c11a4ea5c4e307e170ee4fbd4e81cf1814
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260325
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36808}
2022-05-09 09:06:44 +00:00
3b0481389f Update SupportsScalabilityMode functions to use enum ScalabilityMode.
And add missing values to ScalabilityMode.

Bug: webrtc:11607
Change-Id: I892ac35a3528db11b0901d26902699ecfe8f49a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260982
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36792}
2022-05-06 07:29:20 +00:00
8615bf0582 Move FrameBuffer3 to api/
The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.

Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
2022-05-05 14:34:48 +00:00
26d12fcc71 Remove rtc_base:rtc_base_approved
It's now empty, let's remove it!

Bug: webrtc:9838
Change-Id: I4b3310e882ea95fdf47903f9ad31e2efb35703f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36774}
2022-05-05 09:43:31 +00:00
95b1a3497c stats: implement iceLocalUsernameFragment
https://www.w3.org/TR/webrtc-stats/#dom-rtctransportstats-icelocalusernamefragment

BUG=webrtc:14022

Change-Id: If56ebe66d83f4e535c2245f2ca3848469914679f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261243
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36772}
2022-05-05 08:08:48 +00:00
cc1b9b060d stats: implement iceRole
https://www.w3.org/TR/webrtc-stats/#dom-rtctransportstats-icerole

BUG=webrtc:14022

Change-Id: I88de2c843a2042ce99076d55ce41be22589e2d92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261201
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36766}
2022-05-05 05:05:40 +00:00
44be579b4a Make all VideoReceiveStream2Test use simulated time
Adds matchers to webrtc::VideoFrame to help with the tests.

Bug: webrtc:14003
Change-Id: I62fc1c577bb76b21a96741ba829f6dcd53a308c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36755}
2022-05-04 11:27:16 +00:00
eeffb6aaf7 Remove deprecated audio option residual_echo_detector
The feature is now enabled in other ways. See PSA or linked Monorail issue for details.
https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ

Bug: webrtc:11539
Change-Id: I0f5816baf2bfa1508a1c85ddbd7b775417434c62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260107
Auto-Submit: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36742}
2022-05-03 07:07:58 +00:00
a16a6a6341 stats: implement inbound-rtp totalProcessingDelay for video
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay

BUG=webrtc:13984

Change-Id: Ifd821bd8553add46218f09a11366096d62f5d09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259768
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36732}
2022-05-02 10:56:22 +00:00
a63b6b7d40 [PCLF] Allow configuring RtpEncodingParameters with singlecast
With the encoding parameters in the SimulcastConfig objects, it wasn't
possible to configure explicit encoding parameters when using singlecast,
required for example to use the spec standard SVC API.

Bug: webrtc:11607
Change-Id: I92b1446e772e2ecec93379dc91a3da159b8bc209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260002
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36731}
2022-05-02 10:30:14 +00:00
69c1df2f44 stats: add dtlsRole to transport
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-dtlsrole

BUG=webrtc:13978

Change-Id: Ib158427d2df0307884381bdd46c411f60f56a371
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259761
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36730}
2022-05-02 10:13:54 +00:00
79d566b0cf New enum ScalabilityMode.
Used instead of string representation in lower-levels of encoder configuration, to avoid string comparisons (with risk of misspelling) in lots of places.

Bug: webrtc:11607
Change-Id: I4d51c2265aac297c29976d2aa601d8ffb33b7326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259870
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36706}
2022-04-29 12:16:42 +00:00
9432768024 Prepare for deletion of implicit conversion from rtc::scoped_refptr<T> to T*
Bug: webrtc:13464
Change-Id: I4c7095d3a1c7c1a9ab609f5f1595545f6cad18db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249087
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36693}
2022-04-28 12:58:56 +00:00
e890e36c99 PCLF: Reserve vector before pushing.
Bug: None
Change-Id: I961f555085032330028b426e46a2c4ac576a2b03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260283
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36686}
2022-04-28 09:49:34 +00:00
d425f506ad Switch VideoReceiveStream2 internals to Time units
Change-Id: Ifcee6372120e968499acbdf3bf2c0d002d1c4724
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259777
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Johannes Kron <kron@google.com>
Cr-Commit-Position: refs/heads/main@{#36685}
2022-04-28 09:38:54 +00:00
df209e797b Avoid more usage of implicit conversion from scoped_refptr<T> to T*
Update api/, call/, examples/ and rtc_tools/.

Bug: webrtc:13464
Change-Id: I7b0008cca68c579e89b45527a45300d1e67c3483
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260000
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36673}
2022-04-27 13:18:30 +00:00
a92d051e0f [PCLF] Introduce API to safely mutate ConfigurableParams in TestPeer
Bug: b/213863770
Change-Id: I90b7b5cd55ac5a8ebee5d790205a4fa6700dfff4
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260117
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36668}
2022-04-27 11:00:11 +00:00
e2c780ea24 Reland "Add canonical mock for MediaStreamInterface"
This reverts commit fc61750523120d0d7d7b02e4f6af7fb607a6b824.

Reason for revert: Fixed downstream project.

Original change's description:
> Revert "Add canonical mock for MediaStreamInterface"
>
> This reverts commit e217217bf3782dbd3ea27be33d5ebd26ae0500ca.
>
> Reason for revert: break upstream project (name collision)
>
> Original change's description:
> > Add canonical mock for MediaStreamInterface
> >
> > Needed to let upstream APIs integrate changes more easily.
> >
> > Bug: webrtc:13980
> > Change-Id: I6cd46f75d56597c10e08d0d66e16089516f5129c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259821
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36613}
>
> Bug: webrtc:13980
> Change-Id: I18b91327225e0f844af5dd86c9b4ca8d6301d03e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259860
> 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@{#36614}

Bug: webrtc:13980
Change-Id: I84dbe5ec754389d30e5d22f9f9553fd9e9ee5bd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260115
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@{#36660}
2022-04-26 17:23:45 +00:00
c3e6e3a3e8 Remove dependency on rtc_base_approved from most targets
Bug: webrtc:9838
Change-Id: Ibd0199803597eff48ca139a5cecdc3209c62c5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259873
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36643}
2022-04-25 12:15:30 +00:00
a30aef3dea Move event_tracer out of rtc_base_approved
Bug: webrtc:9838
Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-24 14:47:40 +00:00
ed4aadc0a2 Move copy_on_write_buffer out of rtc_base_approved
Bug: webrtc:9838
Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-24 09:51:50 +00:00
ceb7b36d3a Move byte_buffer out of rtc_base_approved
Bug: webrtc:9838
Change-Id: Ic7e912cba1218f1eed794cb8c393ac148106b16c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259310
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36634}
2022-04-23 22:47:39 +00:00
a876a92d31 Move byte_order out of rtc_base_approved
Bug: webrtc:9838
Change-Id: Ieb2fbaad8831439ca04fc5249e295c8839f6890e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259309
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36633}
2022-04-23 19:04:29 +00:00
f9c5984a1d Move buffer out of rtc_base_approved
Bug: webrtc:9838
Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-22 21:19:28 +00:00
a3b5c4e027 test: replace media_type with kind
media_kind is the old name (that is kept around since we can't deprecate)

BUG=None

Change-Id: I445441a54bb4ff408502d1aba6834cdac874324b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259766
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36625}
2022-04-22 14:53:08 +00:00
fc61750523 Revert "Add canonical mock for MediaStreamInterface"
This reverts commit e217217bf3782dbd3ea27be33d5ebd26ae0500ca.

Reason for revert: break upstream project (name collision)

Original change's description:
> Add canonical mock for MediaStreamInterface
>
> Needed to let upstream APIs integrate changes more easily.
>
> Bug: webrtc:13980
> Change-Id: I6cd46f75d56597c10e08d0d66e16089516f5129c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259821
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36613}

Bug: webrtc:13980
Change-Id: I18b91327225e0f844af5dd86c9b4ca8d6301d03e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259860
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@{#36614}
2022-04-22 10:01:47 +00:00
e217217bf3 Add canonical mock for MediaStreamInterface
Needed to let upstream APIs integrate changes more easily.

Bug: webrtc:13980
Change-Id: I6cd46f75d56597c10e08d0d66e16089516f5129c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259821
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36613}
2022-04-22 09:15:39 +00:00
2f7ad28a6d Change stream.AddTrack/RemoveTrack to take a scoped_refptr argument
This better reflects the ownership passing of AddTrack, and is more
consistent for RemoveTrack.

Bug: webrtc:13980
Change-Id: Ide5baccf15fc687a4e092f8831ce8c0fea46604e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259740
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36603}
2022-04-21 12:32:17 +00:00
50dc7301a0 [PCLF] Remove VideoSubscription::Resolution
Bug: b/213863770
Change-Id: I65cf68a8d5101f3ce416b7163c062c78e8a1ef87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259506
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36600}
2022-04-21 09:39:15 +00:00
7017a13597 [PCLF] Move Resolution from VideoSubscription to the fixture.
Move VideoSubscription::Resolution to the fixture class and rename it to
the VideoResolution. It should be then integrated into other video
related classes.

Bug: b/213863770
Change-Id: Ifd391f840ef8de43bbac66d23df3ecf7258b3943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259523
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36594}
2022-04-20 20:00:24 +00:00
83962d9182 [PCLF] Add GetMaxResolution among resolutions
Bug: b/213863770
Change-Id: I5f90cc5345be8630a2ededf93e1648d4c9bb1be2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259504
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36593}
2022-04-20 18:10:14 +00:00
098c4ea2ca Add generated comfort noise counter.
Currently only implemented for codec internal CNG (Opus).

Bug: webrtc:13322
Change-Id: I00622f2967f066dba64a792e26081038ae0cb0d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259200
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36590}
2022-04-20 14:25:03 +00:00
afb246b5a9 Update pc/ to not use implicit conversion from scoped_refptr<T> to T*.
Bug: webrtc:13464
Change-Id: I768646af8ded6338ef51486b8d69db1ad71e9a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259500
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36588}
2022-04-20 13:18:33 +00:00
6c7f98472e WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 16/inf
This cl/ adds the feature actually injecting a FieldTrialsView into
PeerConnectionFactory, or into a PeerConnection or both.

The field trials used for a PeerConnection is those specified in
PeerConnectionDependencies. Otherwise will those from
PeerConnectionFactoryDependencies be used (and until we're finished with
this conversion, the global string fallback is used as last resort).

Note that it is currently not possible to create 2 FieldTrials
objects concurrently...due to global string,
so this cl/ is mostly (but entirely) for show, i.e one _can_
realistically inject them into a PeerConnectionFactory.


Bug: webrtc:10335
Change-Id: Id2e60525f48a1f8293c1dd0be771e3ed03790963
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258134
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36578}
2022-04-20 06:35:27 +00:00
71337f387e Move random out of rtc_base_approved
Bug: webrtc:9838
Change-Id: I64a5ef18c19d446139354d04aa6cb2a76d18aad0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258762
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36572}
2022-04-19 14:00:47 +00:00
0d510529e7 [PCLF] Introduce API to subscribe to particular streams
Bug: b/213863770
Change-Id: If858686cd265ad48b4ea8be246651eff65fad4f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258981
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36570}
2022-04-19 12:18:20 +00:00
45a0599978 Remove platform_thread from //rtc_base:rtc_base_approved public_deps
While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.

Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36566}
2022-04-18 23:12:52 +00:00
21f12d592a Add rtc_offer_answer_options to peer_connection_quality_test_params.
And use it to generate SDP offers.

Bug: b/203195868
Change-Id: I6f04c92dcef42e2d406d954c2e2ee6e845bcbac1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258795
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36551}
2022-04-14 12:42:39 +00:00
35c5cc8a6f Enable DSCP by default.
DSCP is controlled by the spec-compliant API
RTCRtpEncodingParameters.networkPriority[1]. It already has a default
value that is the same as when DSCP is disabled.
- If you want non-default DSCP default values, you need to set
  networkPriority and shouldn't need to set a non-standard googDscp flag
  for it to have an effect.
- If you want the default DSCP value, you wouldn't change
  networkPriority and so you don't care if enable_dscp is true... you'll
  get the default regardless.

Drive-by: This CL also adds crbug references to other goog flags.

[1] https://w3c.github.io/webrtc-priority/#dom-rtcrtpencodingparameters-networkpriority

Bug: chromium:1315574
Change-Id: I15a0470fa04f55e2534cee0d240eeb03446c2de6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36550}
2022-04-14 09:35:29 +00:00
6daa3048fc Added OnResolutionChange to EncoderSelectorInterface.
Bug: webrtc:12406
Change-Id: I0160636d93ad0a33caf7ae7443cefe321a191406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258442
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36517}
2022-04-11 10:04:57 +00:00
cf7f7f9fa0 AEC3: Add hysteresis period before entering stereo processing
Even if playout audio is only very briefly stereo, the AEC will enter stereo processing mode. To save CPU and improve AEC performance, this CL adds a hysteresis period before treating playout as stereo.

The feature is enabled by default in the AEC3 config.

Bug: chromium:1295710
Change-Id: I29116ab2e7823e25a02aa3b66a1c619f1d966d9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258479
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36503}
2022-04-08 17:01:08 +00:00
fa07b43074 AEC3: Add fallback to mono processing if no stereo is detected for some time
If playout audio is temporarily stereo, the AEC will currently enter stereo processing mode indefinitely. To save CPU and improve AEC performance, this CL adds support for falling back to mono after a period of no stereo.

The feature is enabled by default in the AEC3 config.

Bug: chromium:1295710
Change-Id: I690b5b22f8407f950bf41f3bcaa9ca0138452157
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258421
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36502}
2022-04-08 16:43:14 +00:00
64cdcc0792 AEC3: Add multichannel configuration and multichannel detection
The features have two safety fallbacks:
- multichannel config has a killswitch WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch
- stereo detection has a killswitch WebRTC-Aec3StereoContentDetectionKillSwitch

Both features are enabled by default in the AEC3 config.

Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I340cdc9140dacd4ca22d0911eb9f732b6cf8b226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258129
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36482}
2022-04-07 14:58:02 +00:00
00112748e1 rename functions to be moved to libyuv
Bug: webrtc:13826
Change-Id: I0d694cbe35a272fbe5da9dc6e74c88a976458df8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257441
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Frank Barchard <fbarchard@google.com>
Cr-Commit-Position: refs/heads/main@{#36468}
2022-04-06 21:48:43 +00:00