Fanny Linderborg
c03edf6096
Add missing includes and remove unused includes
...
Unused includes in header files are not removed.
Bug: webrtc:358039777
Change-Id: I4586971cd33ff76cac2f869bcdfb063c31e9a7a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42826}
2024-08-22 08:53:12 +00:00
Denis Genestier
5e70fd3d8f
fix of a compilation error in Visual Studio 2022 due to a warning C4244.
...
Bug: webrtc:353294048
Change-Id: I8c40a9df06464b9ffd7112fd7d070dee2d0e5ca4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360140
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42825}
2024-08-22 08:51:50 +00:00
Philipp Hancke
13b327b05f
srtp: demonstrate wraparound with loss decryption failure
...
by encryption a packet with sequence number 65535 followed
by a packet with sequence number 1. The second packet is encrypted
with a SRTP ROC of 1 as described in
https://datatracker.ietf.org/doc/html/rfc3711#section-3.3.1
The packets are (received and) decrypted in a different order,
the packet with sequence number 1 (and ROC=1) is decrypted first.
Since the ROC is maintained locally the decrypting session assumes
it to be 0.
Why is that a problem? The RFC recommends estimating the ROC with +-1 which, as demonstrated by the test, libSRTP does not.
But this is a rare problem that requires a random in a high range combined with packet loss/reordering which turns into no-a-problem if you choose carefully as done by packet_sequencer.cc which restricts the initial sequence number in the range 0..32767 which means you do not run into this issue in production.
See also Q6 in libsrtp's historical documentation at
https://srtp.sourceforge.net/historical/faq.html
BUG=webrtc:353565743
Change-Id: I9bd72b198c946937aeb25c229005a0c682447f53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42798}
2024-08-19 05:17:18 +00:00
Danil Chapovalov
24823c502b
Add AudioDecoderOpus::MakeAudioDecoder overload taking Environment
...
Mark old overload deprecated.
This allows to migrate both calls through AudioDecoderFactory and direct calls to AudioDecpderOpus trait.
Bug: webrtc:356878416
Change-Id: I1502aee5b18aac43a8258e77b770c8e73a056f92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359741
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42793}
2024-08-16 15:10:30 +00:00
Danil Chapovalov
e0fe4200eb
Provide Environment to consturct AudioDecoder in tests
...
Bug: webrtc:356878416
Change-Id: Id2803736d06445b536f2ced02509eaaaf8fd804c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359361
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42792}
2024-08-16 14:34:37 +00:00
Fanny Linderborg
aa9e557c81
Add header extension reader/writer for automatic corruption detection
...
R=sprang@webrtc.org
Bug: b/358039777
Change-Id: I84f447edf0524d4ac6c55cfd96cffe6abb77aaa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359760
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42791}
2024-08-16 14:14:58 +00:00
Emil Vardar
97ba6afb37
Propagate the base QP value from frame header to Decoded callback in dav1d decoder.
...
Current version of the dav1d decoder does not propagate any QP value to the Decoded callback. This CL updates this such that the base QP gets propagated from the frame header.
Bug: None
Change-Id: Ib7624b7e27d2c973f1821df5688cbb444e4847a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359740
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42790}
2024-08-16 12:02:20 +00:00
Philipp Hancke
952c19511f
Document when the dependency descriptor can be negotiated but not sent
...
This can happen when VP8 simulcast is negotiated while two-byte header
extensions are not negotiated via extmap-allow-mixed. For VP8 the
DD extension would be 23 bytes long which exceeds the maximum size
of 15 bytes for a one-byte header extension.
To test, revert
f04b52b4a7https://webrtc-review.googlesource.com/c/src/+/359623
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42786}
2024-08-15 16:38:54 +00:00
Jan Grulich
b4aba7834e
PipeWire camera: filter out devices with no capabilities
...
Filter out devices that do not support any format supported by WebRTC.
This will for example be IR cameras that show as duplicated in the list
of cameras, but support only GRAY8 format and for that reason do not
work at all.
Bug: webrtc:42225999
Change-Id: Ic2905bc66b55c3f48b49ff4097167f10d17ad656
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358864
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42785}
2024-08-15 09:45:35 +00:00
Danil Chapovalov
759f8d80f0
Delete expired and unused field trial WebRTC-Audio-OpusPlcUsePrevDecodedSamples
...
Bug: b/143582588, webrtc:42221607
Change-Id: I49f477ab785801c8ef7143ab8b8654dd7379dfbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359560
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42783}
2024-08-14 17:21:39 +00:00
Danil Chapovalov
eb26634e6a
Cleanup NetEqControllerFactory interface
...
Finalize change started in https://webrtc-review.googlesource.com/c/src/+/359243
Remove fallback to old interface and unneeded clock member in the config struct.
Bug: None
Change-Id: I4c2b65a09dd1c8a0d44ee76320b095516e2c61fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359561
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42782}
2024-08-14 16:43:57 +00:00
Danil Chapovalov
ce807810be
Change AudioDecoderFactory api to provide Environment to construct AudioDecoders
...
Bug: webrtc:356878416
Change-Id: Id910bef48138b1b659938b1c1a6d23b5634967f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359540
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42781}
2024-08-14 16:02:30 +00:00
Danil Chapovalov
2bc77cebf2
Propagate field trials into NetEq DelayManager
...
Bug: webrtc:42220378
Change-Id: Idf261b0966fb76a68ec610544c705f0aa0f026bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359243
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42779}
2024-08-14 11:03:29 +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
Per K
0478d3e15c
Ensure loss spikes are ignored
...
If all packets are dropped for a period of time, an observation window will have the same length as the period when packets are dropped.
If later, no packets are lost, there is no point in loss based bwe backing down.
Therefore, ignore the observation with most loss and least loss when calculating an instant upper bound.
Bug: webrtc:42222865
Change-Id: I1d0125d6c76e68018b2aec1ecaa9b65729963136
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356380
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@google.com>
Cr-Commit-Position: refs/heads/main@{#42772}
2024-08-13 14:47:45 +00:00
Danil Chapovalov
9b1c0c8245
Propagate Environment to DecoderDatabase::DecoderInfo
...
to make it available for creating AudioDecoders
Bug: webrtc:356878416
Change-Id: Ibd24a55df70985dfe02d924da037618f13661032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359241
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42771}
2024-08-13 13:05:27 +00:00
Danil Chapovalov
96370309a0
Propagate field trials into audio NackTracker
...
Bug: webrtc:42220378
Change-Id: Ibf831e15b5931925a9efa9099178f71b1a23c147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359280
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42769}
2024-08-13 10:04:05 +00:00
Danil Chapovalov
defe1358a5
Pass Environment into NetEqImpl
...
To propagate field trials in addition to clock
Bug: webrtc:356878416
Change-Id: Idefc4848ec4af30c8aed0f93b7fadfc3181bddb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358980
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42761}
2024-08-12 13:33:38 +00:00
Hanna Silen
cc5c549fac
Deprecate TransientSuppression
...
APM transient suppression config has no impact after
https://webrtc-review.googlesource.com/c/src/+/355880 .
Bug: webrtc:7494, webrtc:13663, webrtc:357281131
Change-Id: I5017995aad4f89108b7de46e58df1cd391f61734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358865
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42752}
2024-08-09 14:03:24 +00:00
Danil Chapovalov
0c4c4e6070
Provide Environment to create NetEq in tests
...
Bug: webrtc:356878416
Change-Id: I1d01c61ad6822fb018873fed949154f72b90a11b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358920
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42748}
2024-08-08 13:45:59 +00:00
Jakob Ivarsson
723ea45075
Cleanup NetEq decision logic field trial.
...
Bug: webrtc:42223518
Change-Id: I2e5064109b9f0358e8e590e4aae6ebeb9d16bc3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358861
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42746}
2024-08-08 11:48:11 +00:00
Mirko Bonadei
07e83f244b
Roll chromium_revision ba1ae79f58..6f9b3224db (1319128:1338914)
...
Change log: ba1ae79f58..6f9b3224dbba1ae79f58..6f9b3224dbaa6dbe6d6a..f092508cc25bce81deee..a0e571350294d7b86a83..19268125549e33110a5d..a603c11cfaa1b47952f3..7605e1a3e391945cadc2..49b13d983276788faa4e..1420017c7fhttps://chromium.googlesource.com/catapult.git/+log/022cd349fe..2feeb46c60
* src/third_party/dav1d/libdav1d: 92f592ed10..2355eeb8f2f1c7c96958..75932421dad941d9677b..67060f141b6ede1ccc9e..fb9afbafc773720c7c99..ec378cecaf1d17ea141d..ff233bdd4c163e29159e..9408c6fd4ahttps://aomedia.googlesource.com/aom.git/+log/afedaf9da5..d8ef6563e0
* src/third_party/libc++/src: 09b99fd8ab..7993b2fda2bac941ca44..eb6567388edcc1ffafb5..6db1ccfa71253d6365e3..428f3104fa09a4f3ec84..478e5ab3ecf9098afffa..26643fc4e96144b62bec..6dcd83d60fa8fe86b922..1487e03fbbba1ae79f58..6f9b3224dbba1ae79f58..6f9b3224dbhttps://webrtc-review.googlesource.com/c/src/+/358880
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42744}
2024-08-08 09:20:02 +00:00
Xinyu Ma
675986ec5f
Pass Environment into UlpfecGenerator
...
To make it available for FEC to use field trials in follow ups
Bug: webrtc:355577231
Change-Id: I4a6260a38e50a70dae27db28401b08bf0160aaec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358680
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42740}
2024-08-07 12:39:20 +00:00
Danil Chapovalov
e1dbddfbcf
Introduce NetEqFactory::Create taking Environment instead of the Clock
...
To propagate field trials into the NetEq and further towards Audio Decoders
Bug: webrtc:356878416
Change-Id: Ia7cf18451aef70441ca958bf652f492138c6051a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358620
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42739}
2024-08-07 10:54:38 +00:00
Xinyu Ma
075349f039
Pass Environment into FlexfecSender
...
To make it available for FEC to use field trials in follow ups
Bug: webrtc:355577231
Change-Id: Ie0b7761915696e6ee7453df3d0531b0f7ad30ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358240
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42732}
2024-08-06 17:10:12 +00:00
Danil Chapovalov
3b0424bc41
Delete deprecated AcmReceiver contstructor
...
Bug: webrtc:356878416
Change-Id: Ic7e444e7f35c6927722a61f2f9ba6042cf10002f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358600
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42731}
2024-08-06 15:41:13 +00:00
Danil Chapovalov
33582ea42f
Pass Environment instead of just clock to AcmReceiver at construction
...
Passing Environment would allow to propage field trials with it further to NetEq and AudioDecoders
Bug: webrtc:356878416
Change-Id: Ic68420df3b157ed341146207a2c45cb49e59a931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358501
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42728}
2024-08-06 08:28:23 +00:00
Florent Castelli
916bf2f260
Remove usage of old copy of rtp_packet_sender.h
...
Bug: chromium:345101934
Change-Id: I9123dbd39f5d1e34dd1874b840ab6f34f34849a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357863
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@{#42725}
2024-08-05 13:09:18 +00:00
Hanna Silen
90c430cda4
AudioProcessingImpl: Remove the use of transient suppressor
...
Remove the use of transient suppression, i.e.:
- Transient suppressor submodule (ignore the config),
- WebRTC-TransientSuppressorForcedOff fieldtrial,
- Voice activity detection submodule (use AGC2/AGC VAD instead),
- Submodule overrides, and
- WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR macro.
Bug: webrtc:7494, webrtc:13663, webrtc:357281131
Change-Id: I7edb46c7ff048992ac5a10473800405bad268895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355880
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42724}
2024-08-05 12:38:37 +00:00
Jan Grulich
9e755f0e19
PipeWire camera: Annotate functions with PipeWire calls to avoid CFI
...
Similar to PipeWire implementation of desktop capture, we have to avoid
CFI check for calls of dlopened PipeWire library. This avoid crashing
PipeWire camera backend when "is_official_build=true" option is used as
this turns on "is_cfi=true" enabling control flow integrity.
Bug: chromium:354776214
Change-Id: I7a9fc1c2d77c4ee0e8fe0586369b7246e0bb9180
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358103
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42706}
2024-07-31 18:11:40 +00:00
Björn Terelius
8d7642a9f7
Remove unused QpFastFilterLow method
...
Bug: None
Change-Id: I63665a3fc9afd57aec8f0f7d2a2a2e631452f6c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358080
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42704}
2024-07-31 10:40:42 +00:00
Danil Chapovalov
05309c5236
Delete AudioEncoderOpus constructor that doesn't provide Environment
...
Bug: webrtc:343086059
Change-Id: I55573eff8a13c504c7e14f370398bba1a6eae906
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358060
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42692}
2024-07-30 11:40:34 +00:00
Danil Chapovalov
c2160b14b1
Delete expired field trial Audio-OpusAvoidNoisePumpingDuringDtx
...
Bug: webrtc:42222522, chromium:40174928
Change-Id: I2391b3078e5fff93edca3c3e6e568560b2a1c1cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357742
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42691}
2024-07-30 09:43:52 +00:00
Danil Chapovalov
1932b44aa2
Provide Environment for AudioEncoderOpus in tests when created using the trait
...
To allow delete old signature of the AudioEncoderOpus::MakeAudioEncoder function and thus guarantee Opus AudioEncoder always has an Environment
Bug: webrtc:343086059
Change-Id: Ib660678aeb5a549dddd1dffa3d8c28b2ec6b9d0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356981
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42690}
2024-07-30 09:29:11 +00:00
Artem Titov
e02a200f5e
[numpy] Fix users of NumPy APIs that are removed in NumPy 2.0.
...
This change migrates users of APIs removed in NumPy 2.0 to their
recommended replacements
(https://numpy.org/devdocs/numpy_2_0_migration_guide.html ).
Bug: None
Change-Id: I5c275ed3f39863d42b5c34df0723933f7a8b94a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358020
Reviewed-by: Lionel Koenig <lionelk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42684}
2024-07-29 12:46:53 +00:00
Florent Castelli
5b9d4adfc8
Move rtp_packet_sender.h to api/
...
Old copy of the header and some previous usage is kept around
for compatibility with downstream projects for now.
Bug: chromium:345101934
Change-Id: Icbe42fb8450d3a4115799438d209da4eda127bab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357441
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42681}
2024-07-29 11:40:45 +00:00
Sergey Silkin
12f9d5ce60
Revert "Update support for missing HIGH profiles and 1080p"
...
This reverts commit 46b43e007296737751aea10685f92ddf4df63e0d.
Reason for revert: chromium:354143228
Original change's description:
> Update support for missing HIGH profiles and 1080p
>
> The High and ConstrainedHigh profiles are missing from the decoder capabilities. Also level 3.1 doesn't allow 1080p
>
> Bug: webrtc:347724928
> Change-Id: I3f33468327d2aaf352fc80f69d2ee31481bafcb5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355001
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42528}
Bug: webrtc:347724928
Change-Id: I4d55b2982aca2e94ec983473336c4fa2a72d842f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357861
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42675}
2024-07-26 09:32:40 +00:00
Abby Yeh
35f10a083d
Add listener to detect mute speech event, and callback function to handle the event
...
Bug: webrtc:343347289
Change-Id: I56b1433b0dd8220f95d7d72fb04b4f92fe4a905e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355761
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Abby Yeh <abbyyeh@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42667}
2024-07-23 13:01:39 +00:00
memetao
7fe62f25d1
Reland "Fix 'Image will be cropped if WindowCapturerWinGdi used'"
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This is a reland of commit 844225a76a98aa3be5aca09c19ab72a5e7b6c38a
Original change's description:
> Fix 'Image will be cropped if WindowCapturerWinGdi used'
>
> Bug: webrtc:15719
> Change-Id: I7daf8ee5b90fbe9f1246f1d99211ffa0d8a19f73
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330780
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Commit-Queue: Alexander Cooper <alcooper@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#41503}
Bug: webrtc:15719
Change-Id: Idbb2f4dcc8811d3b2b763a49adc7a57535b3d1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334380
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42666}
2024-07-23 10:28:10 +00:00
Danil Chapovalov
f90a3ad3b3
Reenable disabled passing tests
...
Libvpx was adjusted to support scenarios test verifies, but WebRTC tests were forgotten.
Bug: webrtc:42223649
Change-Id: I19a10c939d844d00dd564bc0a16fe21844cc7cfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357680
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42665}
2024-07-23 07:14:13 +00:00
Danil Chapovalov
ac15a137ac
In RtpVideoStreamReceiver do not rely on RTP sequence number unwrap to be stable
...
Currently this class assumed that if the same RTP sequence number is unwrapped again result would be the same.
That might not be true when several packets were inserted in between these two calls and unwrapper changed its state
This CL propose instead to unwrap once, and save the result in the intermediate struct.
To minimize the change and the risk, only redundant unwrapping is replaced to use unwrapped sequence number
Bug: webrtc:353565743
Change-Id: I8a18c8c206a0e16010951cabcf81dd9cb1588eda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357660
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42662}
2024-07-22 15:42:12 +00:00
Philipp Hancke
3753c8190e
h264: fix first_packet_in_frame logic for multislice in a single rtp packet
...
a frame must be (or should be) first when it contains either SPS (but not just PPS),
is an IDR or is a slice with first_mb_in_slice == 0.
Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit
into a single RTP packet which can happen with small 320x196 frames
BUG=webrtc:352379280,webrtc:346608838
Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42652}
2024-07-19 08:49:24 +00:00
Sergio Garcia Murillo
45e5e385f3
Use ArrayView on H264 bitstream parsing
...
No-Try: true
Bug: webrtc:42225170
Change-Id: I4682f400054fee5c86ea24bebf6d703fb90074da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42650}
2024-07-19 08:16:11 +00:00
Sergey Silkin
4dedf5efae
Use EncoderStreamFactory::CreateEncoderStreams() instead of GetSimulcastConfig()
...
In preparation for upcoming changes in GetSimulcastConfig(), which will require a vector of stream resolutions instead of just the max resolution as an input, switch tests to use CreateEncoderStreams() instead of calling GetSimulcastConfig() directly.
Bug: webrtc:351644568, b/352504711
Change-Id: I541dd54a21a8b75028cff07a250f858a47898223
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357400
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42648}
2024-07-18 16:35:10 +00:00
Danil Chapovalov
faf5b0308c
Delete forwarding rtp_rtcp/time_util.h as unused
...
All known users are updated to use ntp_time_util.h directly
Bug: webrtc:343076000
Change-Id: I7229b9e5dd72d83bfd98ba4050ae7583d792575b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357300
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42645}
2024-07-17 08:53:00 +00:00
Sergey Silkin
3f9589ae64
Remove max_qp argument from GetSimulcastConfig()
...
This is a cleanup of simulcast.cc. max_qp is not needed to decide simulcast config. Move setting of max QP in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams(), where it can be set per stream.
Bug: webrtc:351644568, b/352504711
Change-Id: Ia0e3e9d90032383574dc8867b30d362e9c5df7e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357102
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42634}
2024-07-15 10:23:10 +00:00
Sergey Silkin
55d328dc25
Add ssilkin@webrtc to OWNERS in video/
...
Bug: none
Change-Id: Ie5b5e339634c07d260cc3e10312f97aad63fa552
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42633}
2024-07-15 09:11:54 +00:00
Sergey Silkin
e9810a8adb
Use GetTemporalLayerSum
...
Bug: b/337757868
Change-Id: Ieff4c22425bab06c12419d64db7a2eef69cc54d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355962
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42630}
2024-07-12 15:25:28 +00:00
Sergey Silkin
c0a32fe01b
Remove bitrate_priority argument from GetSimulcastConfig()
...
This is a cleanup of simulcast.cc. bitrate_priority is not needed to decide simulcast config. Move setting of bitrate priority in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams().
Bug: webrtc:351644568
Change-Id: I002d728ccf8d141fe4bbb32b390129ce57c830cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357101
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42629}
2024-07-12 15:02:59 +00:00
Sergey Silkin
3d20dce85f
Delete unused YUV files
...
Bug: none
Change-Id: I2b7794d76be0461271218a55f25021a065a318bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357000
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42625}
2024-07-11 20:26:16 +00:00
