2679 Commits

Author SHA1 Message Date
qwu16
ae82df718c Add codec name H265 to support H265 in WebRTC
Bug: webrtc:13485
Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40773}
2023-09-20 09:25:32 +00:00
Harald Alvestrand
e14d122a7b Remove deprecated SendRtp and SendRtcp functions
and delete remaining usages.

Bug: webrtc:15410
Change-Id: I912bedca80a5a446a3f770211d164a5eb0af02bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40766}
2023-09-19 10:59:26 +00:00
Danil Chapovalov
090699a01b Delete deprecated RtpSource timestamp_ms constructor and accessor
Bug: webrtc:13756
Change-Id: Ic43cf82451785b4fbe184fce466e77b16b2c781a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319581
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40765}
2023-09-19 09:00:03 +00:00
philipel
31718d7ce2 Reland "Add option to disable quality scaling for AV1."
This reverts commit 83102d39077f82f2d4539c160c659dcf789a5fdb.

Reason for revert: reland with fix

Original change's description:
> Revert "Add option to disable quality scaling for AV1."
>
> This reverts commit 446dbc66fde7e9d5e684d3f71e357c2076a91740.
>
> Reason for revert: downstream break
>
> Original change's description:
> > Add option to disable quality scaling for AV1.
> >
> > The main goal of this change is to disable the quality scaler when multiple spatial layers are used.
> >
> > Bug: b/295129711
> > Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40709}
>
> Bug: b/295129711
> Change-Id: Iaeb13951d1b839bc0426120436035843bb3ee98f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320081
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Owners-Override: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40742}

Bug: b/295129711
Change-Id: Iab4846c2cd6074f50a3ebe9551432d449243b5d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40743}
2023-09-13 15:19:36 +00:00
Philip Eliasson
83102d3907 Revert "Add option to disable quality scaling for AV1."
This reverts commit 446dbc66fde7e9d5e684d3f71e357c2076a91740.

Reason for revert: downstream break

Original change's description:
> Add option to disable quality scaling for AV1.
>
> The main goal of this change is to disable the quality scaler when multiple spatial layers are used.
>
> Bug: b/295129711
> Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40709}

Bug: b/295129711
Change-Id: Iaeb13951d1b839bc0426120436035843bb3ee98f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320081
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Owners-Override: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40742}
2023-09-13 12:21:31 +00:00
philipel
46bbf7ec48 Add ScalabilityModeStringToEnum helper function.
Bug: none
Change-Id: Iea602c88afbfe1f8f8e94b353eda96d62b651bd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319882
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40732}
2023-09-11 17:04:27 +00:00
Danil Chapovalov
564e8d395d Mark RtpSource timestamp_ms constructor and accessor deprecated
Bug: webrtc:13757
Change-Id: Ica680dfc0b7420e0d168b4854a07bd73e367218e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319281
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40718}
2023-09-08 07:38:27 +00:00
Danil Chapovalov
541756ff6b Discourage structs in api
Structs make api harder to evolve:
deprecated unused properties,
change how data is represented.

Classes with accessors allow more graduated and safer api evolution.

Bug: None
Change-Id: I8ebd5e072d51cf7f5800666cfdac523d0f9a937f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40714}
2023-09-07 10:41:49 +00:00
philipel
446dbc66fd Add option to disable quality scaling for AV1.
The main goal of this change is to disable the quality scaler when multiple spatial layers are used.

Bug: b/295129711
Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40709}
2023-09-06 12:37:22 +00:00
Harald Alvestrand
96e1882860 Convert AsyncDnsResolver to use absl::AnyInvocable
Bug: webrtc:12598
Change-Id: I0950231d6de7cf53116a573dcd97a3cf5514946c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40670}
2023-08-31 08:50:40 +00:00
Philipp Hancke
df3683e9a7 Remove public GenerateKeyFrame(list-of-rids) API from RtpSender
since the spec and implementation took a different route

BUG=chromium:1354101

Change-Id: I6beda0db89b9e771ad2a7b51ba739bc46e18a331
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40665}
2023-08-30 14:54:17 +00:00
Tony Herre
55b593fb6b Remove EncodedFrame::MissingFrame and start removing Decode() param
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.

Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
2023-08-30 10:38:35 +00:00
Qiu Jianlin
44943c8064 Add H265 codec name and profile/tier/level utils.
This adds H265 codec name and profile/tier/level handling needed for
H265 SDP negotiation.

Bug: webrtc:13485
Change-Id: I838b910042ce36f8ae3979c41a73ee46935c57d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315900
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#40661}
2023-08-30 08:49:09 +00:00
Philipp Hancke
82c56ca794 Request keyframe via setParameters
after the W3C changes in approach documented here:
  https://github.com/w3c/webrtc-extensions/pull/167

chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/4643591

Note that this does not follow the route taken by the W3C API but
still considers this flag a part of the encodingParameters.

BUG=chromium:1354101

Change-Id: If0f0ec09bebddea1f01dd8afbe4747c21afe6793
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286741
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40656}
2023-08-29 13:53:33 +00:00
Danil Chapovalov
f53597140f In RtpSource represent time with Timestamp type instead of int64_t
Bug: webrtc:13757
Change-Id: I5d7da9c9aee489e4b57d361de174c59713cb2b14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40650}
2023-08-29 10:05:03 +00:00
Tony Herre
5f14f9e6ed Remove VCMEncodedFrame from webrtc::EncodedFrame inheritance
Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by
- moving getters for EncodedImage fields up to EncodedImage
- copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame
- Removing EncodedFrame's inheritance of VCMEncodedFrame

We leave VCMEncodedFrame as part of the (near) deprecated
VideoCodingModule code. The only place which needs to accept either is
in the generic decoder.

Bug: webrtc:9378, b:296992877
Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40639}
2023-08-28 11:46:48 +00:00
Harald Alvestrand
4d25a77fd3 Deprecate AsyncResolver config fields and remove internal usage.
Bug: webrtc:12598
Change-Id: Ic43cbcd13e4de44b02351c89da12844606368623
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317604
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40627}
2023-08-25 14:02:27 +00:00
Tommi
66bf3f472c Make PendingTaskSafetyFlag compatible with component builds
Bug: chromium:1470992
Change-Id: I06cec9cda36c9de75b970eaf709f9ed3b9f466b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40618}
2023-08-24 14:24:49 +00:00
Florent Castelli
43a5dd86c2 Implement codec selection api
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.

Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
2023-08-24 13:18:04 +00:00
Danil Chapovalov
7084e1b6d9 In VideoPlayoutDelay delete access to integer representation of min/max values
Bug: webrtc:13756
Change-Id: I1a81c25e5e3fab68a44e94a5ab93e8184c824683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316864
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40612}
2023-08-23 16:14:26 +00:00
Arthur Sonzogni
47faf32287 Add rtc_common_public_deps
When built for chromium, some webrtc implementations are overridden and
are implemented by chrome's "//base". For instance webrtc::Location is
implemented by base::Location. So far so good, the affected targets are
correctly defined in GN to depend on base.

The problem: Most targets in webrtc do not declare correctly their
public_deps. When a public header of a target includes one from its
dependency, the dependency must be a public_deps. The public_deps
instruct GN to forward the capability to use code from the dependency
toward the dependent.

Unfortunately, it is not possible to fix the `public_deps` in webrtc,
because its is disallowed via a presubmit. See:
https://webrtc-review.googlesource.com/c/src/+/30262

WebRTC developers decided not to use `public_deps`, because GN config
are "translated" toward different kind of downstream build system who do
not really support the `public` dependencies concept. Instead WebRTC is
using some "common" configuration applied to all of its targets.

This patch add `rtc_common_public_deps` argument, to let embedders
add the dependencies WebRTC depends on.

Bug: chromium:1467773
Change-Id: I7de43372414a09886fcb07905451e6339c8ecc64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316660
Commit-Queue: Arthur Sonzogni <arthursonzogni@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40595}
2023-08-22 11:32:06 +00:00
Danil Chapovalov
06717773a5 Move EncodedImage::playout_delay_ to private section of the class
Remove code where integer -1 as delay is used to represent unset value.

Bug: webrtc:13756
Change-Id: I16a01e12c25a09ce21a971c9edabf47af5936662
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316923
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40592}
2023-08-22 08:24:37 +00:00
Tony Herre
960882047d Add mock of GetCaptureTimeIdentifier to MockTransformableVideoFrame
Bug: webrtc:14878
Change-Id: I2dffad0932aee4d2ba37c8d57a3c28330e3cc294
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316880
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40585}
2023-08-21 14:42:25 +00:00
Danil Chapovalov
b0b03a87b7 Mark api video timing classes with RTC_EXPORT
Bug: None
Change-Id: Icf99dcdef7278b6051f040c51583a5e164e8f22e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316921
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40584}
2023-08-21 12:55:08 +00:00
Artem Titov
1997837d16 Add stream label to test video source for better debugablity and testability
Bug: b/294812400
Change-Id: I830515b797100ca2dc0e68dd3b79d5a1bb4068da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316221
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40581}
2023-08-21 08:25:38 +00:00
Danil Chapovalov
c146b5f77b Represent unset VideoPlayoutDelay with nullopt rather than special value
Remove support for setting one limit without another limit
because related rtp header extension doesn't support such values.

Start morphing VideoPlayouDelay into a class and stricter type: add accessors returning TimeDelta

Bug: webrtc:13756
Change-Id: If0dd02620528dc870b015beeff3a8103e04022ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40570}
2023-08-18 13:17:50 +00:00
Tony Herre
392e4714e7 Remove deprecated TransformableAudioFrameInterface::getHeader() method
Fixed: chromium:1456628
Change-Id: I12ea08070578de846f042c64f2808b55de1603a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315960
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40555}
2023-08-15 16:31:02 +00:00
Sergey Silkin
8efd93dd76 Encoder type agnostic resolution based fallback
WebRTC-Video-EncoderFallbackSettings/resolution_threshold_px:X sets resolution threshold to switch from primary to fallback encoder. When the field trial is present, VP8-specific resolution based fallback settings, provided by WebRTC-VP8-Forced-Fallback-Encoder-v2, are ignored.

Bug: none
Change-Id: I8f2e28438547f3896c7fc288ed6634720328f3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40526}
2023-08-08 14:18:31 +00:00
Harald Alvestrand
34d82df2ba Use ArrayView versions of SendRtp and SendRtcp
This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.

Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}
2023-08-07 08:28:48 +00:00
Philipp Hancke
e2e04513e7 stats: implement fecSsrc on inbound-rtp
which is present if a fec mechanism like FlexFEC is negotiated

spec change:
  https://github.com/w3c/webrtc-stats/pull/765

BUG=webrtc:15250

Change-Id: I7d71d49fab0153d734f22831e6684d2acfc647fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314981
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40514}
2023-08-04 12:54:48 +00:00
Jonas Oreland
b17806a4cf Add StunDictionary
This patch adds a StunDictionary.
The dictionary has a reader and a writer.
A writer can update a reader by creating a delta.
The delta is applied by the reader, and the ack is applied by the
writer.

Using this mechanism, two ice agents can (in the future) communicate
properties w/o manually needing to add new code.

The delta and delta-ack attributes has been allocated at IANA.

Bug: webrtc:15392
Change-Id: Icdbaf157004258b26ffa0c1f922e083b1ed23899
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40513}
2023-08-04 12:08:44 +00:00
Harald Alvestrand
b38d9d2b6f Add ArrayView versions of SendRtp and SendRtcp
This is part of the long term plan to stop using pointer + length
to pass around buffers.

Bug: webrtc:14870
Change-Id: Ibaf5258fd326b56132b9b5a8a6b1563a763ef2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40512}
2023-08-04 11:20:53 +00:00
Mirko Bonadei
cad3aed5fc Add setters to NetworkEmulationManager::SimulatedNetworkNode::Builder.
Bug: b/294494713
Change-Id: I89130a4742da5f04680aa38721afcd7f91fb30ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314980
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40511}
2023-08-04 10:38:59 +00:00
Philipp Hancke
9b82b2f8d6 stats: implement RTX ssrc on inbound-rtp/outbound-rtp
spec change:
  https://github.com/w3c/webrtc-stats/pull/765

BUG=webrtc:15096

Change-Id: I7c72193c23460330b6bb612a9568641d187ee638
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312362
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40510}
2023-08-04 08:41:52 +00:00
Philipp Hancke
55b89a8068 Rename cipher_suite to crypto_suite
and replace "cs" in the appropriate places.

This is the terminology used by
https://www.rfc-editor.org/rfc/rfc4568#section-10.3.2.1
and
https://www.iana.org/assignments/sdp-security-descriptions/sdp-security-descriptions.xhtml

BUG=None

Change-Id: I45f2c52eb266c0f94bdd710a9b941142b9411827
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314483
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40502}
2023-08-02 11:45:24 +00:00
Chunbo Hua
5eb521955a Correct typo from valee to value for color space definitions
Bug: None
Change-Id: I7854669de1216385e188bc53ee0cbd26c002c680
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312741
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40489}
2023-07-31 06:28:45 +00:00
Philipp Hancke
ea06be2682 candidate: do not log full IP addresses for related address
since this may contain sensitive data, just like the address.

BUG=None

Change-Id: I3faa1512a15467cd5cc4bcbcaebadb736f1bae07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313040
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40473}
2023-07-25 17:19:25 +00:00
Harald Alvestrand
00f11224fd Remove extra usage of video-content-type header extension
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.

Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
2023-07-22 21:47:08 +00:00
Danil Chapovalov
630c40d716 Update RtpSenderVideo::SendVideo/SendEncodedImage to take Timestamp/TimeDelta types
Bug: webrtc:13757
Change-Id: I2f21b14ecf003c5cb0c4c92d0c6b9b6f11c35f71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311945
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40450}
2023-07-21 10:36:49 +00:00
Joachim Reiersen
e9e03a9160 Fix inaccurate contentType in RTCInbound/OutboundRtpStreamStats
The existing equality check did not always work since content_type
is sometimes overloaded with extra internal information such as simulcast layer index. Fix by using the videocontenttypehelpers::IsScreenshare helper method.

Bug: webrtc:15381
Change-Id: I2fe84e7f036ea2c223e4fa6dd58af1c4c0bcfbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312261
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40448}
2023-07-19 21:57:10 +00:00
Angelo Tadres
775470214a Removing enum used for removed UMA metrics WebRTC.PeerConnection.Simulcast.ApplyLocalDescription and WebRTC.PeerConnection.Simulcast.ApplyRemoteDescription
This is pending work from this CL already merged: https://webrtc-review.googlesource.com/c/src/+/311640

Bug: chromium:1447193
Change-Id: I9b2ffb60d65f87f0497b099b6253bf122ff1d873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311740
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40428}
2023-07-13 09:00:52 +00:00
Jianhui Dai
32a8169a65 Use common VideoFrameTypeToString helper
This CL cleans up all local conversions, in favor of the common helper
function.

Bug: webrtc:15210
Change-Id: Id77e1c6b1151a2542d92e220e91d5b11285479b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311060
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40420}
2023-07-12 00:28:47 +00:00
Henrik Boström
a6c4f12fad Set noparent in api/stats/OWNERS.
This avoids the risk of a higher level owner approving something getting
added to the getStats() API without +1s from stats owners.

Bug: None
Change-Id: Iedd7133d0e943d1db6977dc0e1d406e5b545e31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311543
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40412}
2023-07-10 12:25:04 +00:00
Tony Herre
b4062e5611 Add a setter for RTPTimestamp on TransformableFrameInterface
Move the SetRTPTimestamp method from TransformableAudioFrameInterface
to the base class, so that RTPTimestamps can also be modified on encoded
video frames.

Bug: webrtc:14709
Change-Id: I355be527c2be201c9201e04c431394c962237140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310781
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40378}
2023-06-29 13:42:15 +00:00
Tony Herre
58ee9dff08 Deprecate encoded audio frame GetHeader
Bug: chromium:1456628
Change-Id: Ifc7d1aa1153c0593c673381f153e5793b94c98c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40359}
2023-06-27 13:13:34 +00:00
Tony Herre
fc68f1f7d9 Stop using TransformableAudioFrameInterface::GetHeader() within webrtc
Instead switch to specific getters, or methods only defined on specific implementations rather than part of the public API.

Once uses are removed from Chromium, I'll mark GetHeader() deprecated
and eventually remove it.

Bug: chromium:1456628
Change-Id: I19b80489b3a0322c201e24994494cfbb742ee13e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309780
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40344}
2023-06-26 10:07:50 +00:00
Philipp Hancke
17e8a5cc7d stats: implement flexfec fecBytesReceived stats for FlexFEC
specified in https://github.com/w3c/webrtc-stats/pull/762
and take FlexFEC into account for receive statistics.

BUG=webrtc:15250

Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40325}
2023-06-21 13:04:31 +00:00
Hans Wennborg
d1780836f4 Fix incorrect use of scoped enumerations in format strings
Scoped enums do not get automatically promoted to their underlying type,
so these uses have undefined behavior and Clang recently started warning
about it.

Bug: chromium:1456289
Change-Id: I9cf4e5a68378930a3bf7d8ac7b0a21eaf0d12670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309520
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Hans Wennborg <hans@chromium.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40321}
2023-06-21 09:16:11 +00:00
Philipp Hancke
17ec0569c6 ICE: adjust priority of non-relay candidates
Introduces a field trial
  WebRTC-IncreaseIceCandidatePriorityHostSrflx
that adjusts the priority of non-relay candidates such that the STUN priority attribute calculated as
  (prflx-type-preference << 24) | (priority & 0x00FFFFFF)
as described in
  https://www.rfc-editor.org/rfc/rfc5245#section-7.1.2.1
will satisfy the condition that the STUN priority of server-reflexive candidates will always be higher than the STUN priority of relay candidates.

Previously this was not the case because the TURN relay preference was added to the local_preference for relay candidates, making it higher than the local_preference of srflx candidates gathered from the same interface.
This led to cases where the resulting candidate pair priority of a srflx-relay pair was higher than the candidate pair priority of a srflx-srflx pair, i.e. using a TURN server in cases that work using a direct P2P connection.

Whether the field trial is active can be observed by checking that
  priority-of-srflx-candidate&0x00FFFFFF
is greater than
  priority-of-relay-candidate&0x00FFFFFF

BUG=webrtc:13195,webrtc:5813,webrtc:15020

Change-Id: Ib91708fbe7310b6454f93158a45c9d77da009091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292700
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40311}
2023-06-20 08:02:27 +00:00
Tony Herre
097a4decc2 Make all encodedaudioframes inherit from TransformableAudioFrameI'face
Make outgoing encoded audio frames inherit from the same Audio interface
that incoming frames inherit from, to align them and make it possible to
eg clone frames regardless of their direction.

Also begin removing GetHeader() from the Audio interface, replacing it
with getters for the specific values we actually need to propagate in
the API: sequence number and CSRCs. This makes it much easier to treat
incoming and outgoing frames the same, even if they don't have full
RtpHeaders prepared at the point of the transform.

Bug: chromium:1453226
Change-Id: Ib5b39b30dea8a378b3b26efb1589dfd64741d201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308141
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40309}
2023-06-19 18:54:47 +00:00