Commit graph

3024 commits

Author SHA1 Message Date
Per K
b4c1f2f6fc Remove DegradedCall - To be submitted after 2024-07-01
Bug: webrtc:343801362
Change-Id: Icae19ab2f4c87521483d25ae8d44c849c5f8ed2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353140
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42892}
2024-08-30 08:08:39 +00:00
Fanny Linderborg
2f91bdceee Declare corruption detection URI in RtpExtension
R=sprang@webrtc.org

Bug: webrtc:358039777
Change-Id: I9c66794b8a622bef5505f3a4a7252a0e7a989813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42887}
2024-08-29 19:41:16 +00:00
Jakob Ivarsson
04cc4ce2f2 Deprecate NetEq::GetDecoderFormat and remove implementation.
Bug: None
Change-Id: I9c90b41ee528984d1a3cd1632565c6dc1598e4d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360920
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42881}
2024-08-29 10:47:29 +00:00
Danil Chapovalov
a99bf7fa84 Delete deprecated AudioDecoderOpus::MakeAudioDecoder
Bug: webrtc:356878416
Change-Id: I2dc830c46fb5eece3b93a0354fd1e8a323a5e2ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360841
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42880}
2024-08-29 08:55:27 +00:00
Ho Cheung
f2487c0d4f [audio] Adjust the order of some definitions in audio_processing
Moving defines before they are used with
unique_ptr allows to compile this file with
-std=c++2b flag.

Bug: webrtc:339074792
Change-Id: Ie7c37ab724800aea4545b72b4d2a779e12a2026a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360860
Auto-Submit: Ho Cheung <hocheung@chromium.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42879}
2024-08-29 07:08:12 +00:00
Danil Chapovalov
4c862e781b Implement Create instead of MakeAudioDecoder in AudioDecoderFactory template
Bug: webrtc:356878416
Change-Id: Iae9369eb6a2ae09a707854a18d909eed453bbfd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359960
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42874}
2024-08-28 14:18:22 +00:00
Jakob Ivarsson
b6046aece2 Add NetEq API to get info about the current decoder.
This is currenly tracked in both AcmReceiver and NetEq. Adding this API
enables us to have it in just one place.

Bug: None
Change-Id: Ia537f87f36b0aedf19c00a57bd6cec4425a49df1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360743
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42872}
2024-08-28 12:50:50 +00:00
Johannes Kron
0b91688817 Mark EncodedImage::{Set, Is}AtTargetQuality() as deprecated
The "at target quality" attribute is no longer set to the encoded
image in VideoStreamEncoder, see
https://webrtc-review.googlesource.com/c/src/+/359640

Mark the attribute as deprecated to avoid new dependencies and prepare
for deletion.

Bug: chromium:359410061
Change-Id: Id5a98ec9d2068099cb75a70b849bbf1c77feabb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359660
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42848}
2024-08-26 14:48:29 +00:00
Peter Kasting
b92345615e [jumbo] Add begin()/end() to EncodedImage[BufferInterface].
This allows these types to meet the requirements of e.g.
std::ranges::range, which is necessary for them to work with the
std::span range constructor, or the "non-legacy" constructor for
Chromium's base::span.

Bug: none
Change-Id: Ia51c17690c785e0714c36d237094877129e0cbaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358844
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42838}
2024-08-23 19:28:27 +00:00
Philipp Hancke
b31ade36ff stun/turn: suppress icecandidateerror for incompatible address family
Suppresses the ice candidate error callback when the STUN/TURN server
address family is not compatible with the local candidate address family.

This is similar to not pairing between candidates that have different
incompatible address families as described in
https://datatracker.ietf.org/doc/html/rfc5245#section-5.7.1

The spec actually says to emit the 701 error if *no* host candidate is able to reach the server:
  https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectioniceerrorevent-errorcode

Also use the same (spec) error code for STUN and TURN, see
https://github.com/webrtc/samples/issues/1215 (error 600 for TURN)
https://github.com/webrtc/samples/issues/1227 (error 701 with AF mismatch)

Drive-by: misc logging fixes

BUG=webrtc:359404135

Change-Id: I99574b7b2b79986a52ab38a7fa58ea1bebab954c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358961
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42830}
2024-08-22 21:33:45 +00:00
Christoffer Dewerin
a6fad74043 Add missing optional deps
Bug: webrtc:42226242
Change-Id: If09580139acb52b11ac4827f68aba46929cc5755
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360080
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Auto-Submit: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42813}
2024-08-20 14:50:36 +00:00
Philipp Hancke
2cfedb277a Remove vestiges of GTURN
and update some usage to use the "correct" stun attribute names

BUG=webrtc:42229250

Change-Id: If0c34d1d9b399766d7073661ea2a5515100256a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359440
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42810}
2024-08-19 16:05:37 +00:00
Harald Alvestrand
b1ffa9bd4e Export SdpTypeFromString
Needed to get rid of a form of CreateSessionDescription that is due
for deprecation.

Bug: None
Change-Id: I9717b7ded1e28cf803de4bebc852c2f255851918
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359941
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42808}
2024-08-19 13:34:49 +00:00
Dor Hen
52e46247bf Apply include-cleaner to api/voip
Bug: webrtc:42226242
Change-Id: I54f58eca55bed5db08020129514fb187b9a05d58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359882
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@{#42806}
2024-08-19 12:45:52 +00:00
Dor Hen
d7d940ea59 Apply include-cleaner to api/video_codecs
Bug: webrtc:42226242
Change-Id: I7292811f782ec9a6710c84b1fc36e42ae7ea2c17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359840
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42802}
2024-08-19 11:23:44 +00:00
Dor Hen
9fb83a18e3 Apply include-cleaner to api/video
Bug: webrtc:42226242
Change-Id: I023f058f3b5e2747bd02f01a191a91636c85f12d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359820
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@{#42801}
2024-08-19 11:22:41 +00:00
Dor Hen
de972c1126 Apply include-cleaner to api/units
Bug: webrtc:42226242
Change-Id: Ic646947aee054f597d1fb069c7a6d0bbaadeb1dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359562
Commit-Queue: Dor Hen <dorhen@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42796}
2024-08-18 06:55:23 +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
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
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
Danil Chapovalov
ea233be459 Remove passing FieldTrialsView into AudioDecoderFactory as unused
Plan is to pass field trials using Environment when creating individual AudioDecoder rather than providing single set of field trials for the factory.
Current implementation is not used, and doesn't pass field trials when actually creating an AudioDecoder

Bug: webrtc:356878416
Change-Id: I0f79f09f7a6aa63e20fbdd783e90e8d026158330
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359221
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42762}
2024-08-12 14:42:37 +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
Dor Hen
73195da677 Apply include-cleaner to api/test/metrics
Bug: webrtc:42226242
Change-Id: I3134d7a3885a86f0bd02ee5e12bb85b2e2d6ee8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42760}
2024-08-12 12:11:35 +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
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
Dor Hen
ec5a968070 Apply include-cleaner to api/transport
Bug: webrtc:42226242
Change-Id: I41fe8dcb2f68b1f05c70058fee527f4cae67c51b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358502
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@{#42742}
2024-08-07 16:09:45 +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
Mirko Bonadei
a8dd3a36fa Add missing dependency.
Bug: None
Change-Id: I864ee21709f1e7e4622c4f4717eab2de64cf64b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358640
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42735}
2024-08-07 07:00:06 +00:00
Dor Hen
5ac2c74022 Apply include-cleaner to api/task_queue
Bug: webrtc:42226242
Change-Id: Ieee3dddf72eb8ecea699be87f05a59036670f3d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42729}
2024-08-06 13:32:04 +00:00
Lionel Koenig Gélas
b4462510c3 Pass receive_time through frame transformer
Bug: webrtc:344347965
Change-Id: Iee5ae13487f57f2b0c98dd6fb6a14286ff317fbb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358100
Reviewed-by: Tony Herre <herre@google.com>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42717}
2024-08-02 07:01:33 +00:00
Florent Castelli
c427c1723d Make PriorityValue constructor explicit
Bug: webrtc:42225365
Change-Id: I8d0e6cf0f2d6f5677cb10e4b5ea32121dab733bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358301
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42715}
2024-08-01 18:18:04 +00:00
Jeremy Leconte
53291d49b2 Add 'SkipNextFrame' to the FrameGeneratorInterface.
Also fix PRESUBMIT.py following https://webrtc-review.googlesource.com/c/src/+/358160.

Change-Id: I00682209607a184448255cf5ad8fd213fda7f4af
Bug: b/355120692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358320
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42711}
2024-08-01 14:38:52 +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
Danil Chapovalov
cbb13bba86 Delete deprecated CreateAudioEncoderFactory with unused field trials parameter
Field trials are passed during AudioEncoder construction through Environment parameter
All known users were migrated to the same named function without parameters.

Bug: webrtc:343086059
Change-Id: I79e2edae22ab43f98a386430da82b41d1c71e426
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358061
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42693}
2024-07-30 12:32:24 +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
e2f02c2df0 Delete AudioEncoderFactory::MakeAudioEncoder
Make AudioEncoderFactory::Create pure virtual.

To finalize migrating AudioEncoderFactory to new interface for creating AudioEncoder and thus guarantee AudioEncoders always have an Environment at construction.

Bug: webrtc:343086059
Change-Id: I1d607082437c15201c8a75dd7a3925fe0f75b70f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355800
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42686}
2024-07-29 16:00:28 +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
Tony Herre
5079e8a30a Allow supplying a custom NetworkControllerInterfaceFactory per-Call in PeerConnectionDependencies
This requires making CallConfig move-only so it can hold a unique_ptr to
the factory, but as discussed with Danil, that seems fine.

Bug: chromium:355610792
Change-Id: Ie52e33faaa4a2af748daeb25f5327b7a532936e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42679}
2024-07-29 07:17:14 +00:00
Danil Chapovalov
161956b89d Cleanup deprecated accessors in VideoFrame
Bug: None
Change-Id: I3f8f428f04e86c38d5cf6d481709b7bcdfbd495c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357781
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42671}
2024-07-24 13:49:19 +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
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
Sergey Silkin
7a6053ae62 Rename minimum_qp to min_qp
For better consistency with the rest codebase (it is min_/max_ for all params in video_encoder.h; only qp is for some reason prefixed with minimum_).

Also fixed constant names in libaom AV1 encoder wrapper (moved min from suffix to prefix, minimum -> min_).

Bug: chromium:328598314
Change-Id: I6d8521a3abff3a0595a5241c02ef4746eb4694df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42604}
2024-07-08 15:37:23 +00:00
Tommi
41ffc51e45 Remove unused AudioprocFloat with input_aecdump param support
There's some test code associated with this code path that can
be deleted, so this is a first step towards removing it. From what
I can tell, this is never used.

Bug: none
Change-Id: Idfb8a6c58b929c2eedd0cfc7bdc72f5b3862f5bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356481
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42600}
2024-07-08 08:20:53 +00:00
Tommi
55c3600781 Remove <ostream> dependencies
Some dependencies still exist but are a bit more complex to remove.
This CL removes either unused or easily replaced with ToString()
instances of ostream usage. In one case, moving the operator<<
implementation to the one test file that requires it.

Bug: webrtc:8982
Change-Id: Ia5c840b12a42893494af401317a3daf2fe50ba9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356240
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42582}
2024-07-03 12:27:55 +00:00
Johannes Kron
216cce5f49 Add minimum_qp to VideoEncoder::EncoderInfo
The minimum QP field will be used to signal what the QP value will be
once the encoder reach its target video quality. This will be used
in the generalized QP convergence detection.

Bug: chromium:328598314
Change-Id: I82299cd921e3c091e651218d1e3f337875176567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Cr-Commit-Position: refs/heads/main@{#42559}
2024-06-28 10:48:22 +00:00
Johannes Kron
7235dd0e2b Add is_steady_state_refresh_frame_ to EncodedImage
The field is_steady_state_refresh_frame_ can be used to determine
if the encoded video frame is a repeated frame that should be considered
for QP convergence detection.

Bug: chromium:328598314
Change-Id: Iffba0f9f70af8b41b9bde25cf40b08b77dad8021
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355702
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42550}
2024-06-27 11:57:11 +00:00