6272 Commits

Author SHA1 Message Date
Rasmus Brandt
bc3a41e0d7 Move deprecated VCMDecodingState to modules/video_coding/deprecated/
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.

No functional changes are intended.

Bug: webrtc:14876
Change-Id: Ie6820a820f22635fe7a970db70b9c28d37499848
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296443
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39518}
2023-03-09 19:44:00 +00:00
Danil Chapovalov
b40aedf911 Delete RTPHeader::payload_type_frequency as unused
payload type frequency is not communicated inside an RTP packet and
thus do not belong to the RTPHeader

Bug: None
Change-Id: Ic3e48f1b0507a96ddc697503145f7c8785962926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296763
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39515}
2023-03-09 16:32:22 +00:00
Henrik Boström
f8bc1169a8 Test that VideoCodecInitializer propagates VP9 resolution alignment.
The GetSvcConfig() call here[1] can result in resolution alignment due
to [2], which gets propagated to the output VideoCodec due to [3].

This CL adds test coverage for this part.
It also removes some comments that are no longer true and updates
VideoCodecInitializerTest's SetUpFor() to make number of simulcast
streams explicit.

[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/video_codec_initializer.cc;l=251;drc=e4a304ed4da869ab6131a06b3e8b7e985f50229d
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/codecs/vp9/svc_config.cc;l=112;drc=31acc7339cf658ce82c7ec490ba38d67170920ed
[3] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/video_codec_initializer.cc;l=285;drc=e4a304ed4da869ab6131a06b3e8b7e985f50229d

Bug: webrtc:14884
Change-Id: Id22e36aebab573f53d15dca688642d32c8c4be7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296762
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39514}
2023-03-09 13:40:12 +00:00
Rasmus Brandt
5a54800957 Move deprecated VCMFrameBuffer to modules/video_coding/deprecated/
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.

No functional changes are intended.

Bug: webrtc:14876
Change-Id: Id1c7fbb969a63eee96fd88c376371aa7eafd0919
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296440
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39512}
2023-03-09 11:05:20 +00:00
Danil Chapovalov
6a0997de01 Skip calling rtcp callback on packets containing invalid rtcp message
Low-Coverage-Reason: added code handles invalid rtcp packet scenarios which are covered by rtcp_receiver_fuzzer
Bug: webrtc:5260
Change-Id: Ia6ba0b736f5732fa6516edb2a5ed1b96079eb4ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296580
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39503}
2023-03-08 14:19:54 +00:00
Per K
ffdecdce2c Add possibility to start probe without waiting for media packet
Bug: webrtc:14928
Change-Id: If3c7fe39979e7a4d45b01dd59debd8ac69de3d01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39496}
2023-03-07 14:21:32 +00:00
Sergey Silkin
a5f32e445c Set frame capture timestamp
Unlike SW encoder wrappers, Android HW encoder wrapper uses frame capture timestamp instead of RTP timestamp: https://source.corp.google.com/piper///depot/google3/third_party/webrtc/files/stable/webrtc/sdk/android/src/jni/video_frame.cc;rcl=514125748;l=309

Bug: b/261160916, webrtc:14852
Change-Id: Ief76abae659f7ba890371901cc9b505526ac4f97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296500
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39495}
2023-03-07 13:56:14 +00:00
Sergey Silkin
1c1382be0f Dump codec output to ivf/y4m
Bug: b/261160916, webrtc:14852
Change-Id: I19de2210aa03b56752db5ce8b6fd94498123d6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39490}
2023-03-07 08:33:39 +00:00
Sergey Silkin
9259b5f72c Add rate adaptation tests
Bug: b/261160916, webrtc:14852
Change-Id: I58b3647218c961dcf0305c3902f79adb448b73e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295866
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39489}
2023-03-06 18:33:16 +00:00
Danil Chapovalov
812186d060 Unassign personal TODOs in modules/rtp_rtcp
1st TODO is removed because there are no plans to use RtcpSender for multiple media streams, RtcpTransceiver designed for that instead
2nd TODO is removed because benefits of not producing same report blocks is unclear, while there is risk to break RTT calculation for remote receivers

TODOs in RtcpTransceiver are reassigned to general bug about it instead, in particular to note low priority, same as that issues has.

No-Try: true
Bug: webrtc:10198
Change-Id: Ibcc3691265459c0732019ceba7903e27d57d543a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296360
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39488}
2023-03-06 17:24:50 +00:00
Olov Brändström
3ca732d12d Prevent warnings from timestamp aligner used in AudioDeviceBuffer
The logcat is spammed with warnings from timestamp aligner.
This CL make a workaround to use the timestamp aligner to get capture times,
without generating the warnings.

Bug: webrtc:14970
Change-Id: Idab4b298e0484a57841a214db9440f9ac6faaa4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296324
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#39486}
2023-03-06 15:47:51 +00:00
Rasmus Brandt
78e1388ea7 Move deprecated VCMSessionInfo to modules/video_coding/deprecated/.
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.

No functional changes are intended.

Bug: webrtc:14876
Change-Id: Ieafdb2640b12c254edfac04e98f86f9170c5a71a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295870
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39483}
2023-03-06 14:10:45 +00:00
Danil Chapovalov
8eb7aef196 Relax expectation on the libaom rate controller
Bug: b/271819773
Change-Id: I580b6fde352d1f23773fd394b0ee1543724b828f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296323
Reviewed-by: Jerome Jiang <jianj@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39482}
2023-03-06 13:58:51 +00:00
Tove Petersson
1e2d951762 Add a clone method to the audio frame transformer API.
This will clone an encoded audio frame into a sender frame.

Bug: webrtc:14949
Change-Id: Ie62d9f5ec457541b335bde8f2f6e9b6d24704cf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39480}
2023-03-06 08:22:25 +00:00
Markus Handell
5145d90660 NackRequester::ClearUpTo: avoid PostTasks.
Under the combined network/worker thread project, tasks
are unnecessarily posted to the same thread. Avoid this
by posting only if invoked on a diffferent sequence.

TESTED=presubmit + local Meet calls.

Bug: chromium:1373439
Change-Id: I2ca15b2c725f412ca8a0b8132d6b221f9f6b6032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295868
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39477}
2023-03-03 22:57:32 +00:00
Markus Handell
c35a3f13c7 RtpSenderEgress::SendPacket: avoid PostTasks.
Under the combined network/worker thread project, tasks
are unnecessarily posted to the same thread. Avoid this
by posting only if invoked on a diffferent sequence.

TESTED=presubmit + local Meet calls.

Bug: webrtc:137439
Change-Id: I9befde0583b214ebe014617695c2eb9f047de8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295869
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39472}
2023-03-03 16:08:07 +00:00
Rasmus Brandt
65a6ecab33 Rename InterFrameDelay -> InterFrameDelayVariationCalculator.
This class name better reflects the nomenclature defined by RFC5481: https://datatracker.ietf.org/doc/html/rfc5481#section-1.

Some code style improvements were performed. No functional changes are intended.

Bug: webrtc:14905
Change-Id: I84b9deb7b2ac7f1a07ae00670eaff9656a50c2cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295661
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39466}
2023-03-03 11:49:37 +00:00
Evan Shrubsole
02bdf66f95 Reland "Launch WebRTC-SendPacketsOnWorkerThread""
This reverts commit a09b30dd8a18f809c4a245d7ecd5848a00ccfe0e.

Reland OK: Internal test fixed.

Bug: webrtc:14502, b/254640777
Change-Id: I4838111169b10099a8b14e18170307b342e45033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295864
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39460}
2023-03-02 17:11:03 +00:00
Tony Herre
2311f93909 Remove uses of TransformableVideoFrame::GetMetadata and deprecate it
Chromium uses have been migrated to Metadata(), so we should be clear.
Other projects can easily migrate similarly.

Bug: chromium:1420245
Change-Id: I150654812676dabd5c957cff00d40d4c95eaf5d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295481
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39455}
2023-03-02 13:38:48 +00:00
Rasmus Brandt
34d339f12b Move deprecated VCMPacket to modules/video_coding/deprecated/
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.

No functional changes are intended.

Bug: webrtc:14876
Change-Id: Ib11fe46f35ab0efba35c6a9a2482b4f7c016226c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295821
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39451}
2023-03-02 12:43:51 +00:00
Alan Zhao
6cf8b486eb Fix missing libc++ includes in webrtc
Several files refer to symbols declared in headers not explicitly
included. This compiles now because libc++ tranitively includes these
headers via other libc++ headers; however, these transitive includes are
not guaranteed to exist and in Chrome, will no longer exist once libc++
is compiled with modules.

Bug: chromium:543704
Change-Id: I638bb02df3d050a48345248e80aebd2dd60956c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295762
Auto-Submit: Alan Zhao <ayzhao@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39448}
2023-03-02 10:14:51 +00:00
Emil Lundmark
4e86aa0870 Remove mentions of already deleted field trials
- WebRTC-Audio-Agc2ForceExtraSaturationMargin
- WebRTC-Audio-Agc2ForceInitialSaturationMargin
- WebRTC-Audio-BitrateAdaptation
- WebRTC-Audio-TransientSuppressorVadMode
- WebRTC-FrameBuffer3
- WebRTC-IntelVP8
- WebRTC-UseActiveIceController

Bug: None
Change-Id: I3545727c09f761867f2f4c2bb5c400012ce146d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295723
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39444}
2023-03-01 15:53:37 +00:00
Markus Handell
c0f8813870 Implement support for Chrome task origin tracing. #3.6/4
This CL migrates the task queue paced sender unit test
to the new TaskQueueBase interface.

Bug: chromium:1416199
Change-Id: Id0568bb9a08bf43b92e33fdf45fe75a57e5a7a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39436}
2023-03-01 11:53:08 +00:00
henrika
d2ee133c59 Avoids initial kFrameDropped burst for WGC
Bug: chromium:1412584
Change-Id: I6bfdcec98dfae0f99bfce51ace15795a044eb7d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295504
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39435}
2023-03-01 11:21:53 +00:00
Danil Chapovalov
0f43da2248 Cleanup RtcpReceiver::NTP function
Replace it with GetSenderReportStats that returns result instead of
filling lots of output parameters
On the way replace pair of uint32_t with dedicated NtpTime type

Bug: None
Change-Id: I5b821b8d000d63ebd8cdc1b9897a86429d97b19b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295560
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39431}
2023-03-01 10:44:29 +00:00
Andrey Logvin
a09b30dd8a Revert "Launch WebRTC-SendPacketsOnWorkerThread"
This reverts commit 8d33105015183d02978ecefcedef241247af3802.

Reason for revert: Speculative revert, may have caused breakage in post submit tests. E.g. https://ci.chromium.org/p/webrtc/builders/ci/Linux32%20Debug/32343 (waterfall https://ci.chromium.org/p/webrtc/g/ci/console?limit=200)

Original change's description:
> Launch WebRTC-SendPacketsOnWorkerThread
>
> Bug: webrtc:14502, b/254640777
> Change-Id: I61269443b5ce87ba0c5354f863c731292c86dbce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293581
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39425}

Bug: webrtc:14502, b/254640777
Change-Id: Iec5d373fb7a73bc07d8cc4af4ca03a0f60331eda
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295662
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Auto-Submit: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39429}
2023-03-01 09:14:32 +00:00
Sergey Silkin
fddc9131a5 Aggregate and log video codec metrics
Bug: b/261160916, webrtc:14852
Change-Id: Idcb7e96b12ca38af49b9b1f10d1e23cc7faac92b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293945
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39427}
2023-03-01 08:27:54 +00:00
Michael Horowitz
b27efd487d Add option to configure AV1 EncoderInfo resolution_bitrate_limits.
bug: webrtc:14931
Change-Id: I8ade2a888d29f76a0f690fc3541b45b7304ad4d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294600
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39426}
2023-02-28 20:48:33 +00:00
Evan Shrubsole
8d33105015 Launch WebRTC-SendPacketsOnWorkerThread
Bug: webrtc:14502, b/254640777
Change-Id: I61269443b5ce87ba0c5354f863c731292c86dbce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293581
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39425}
2023-02-28 18:03:59 +00:00
Tony Herre
a6135bcd43 Remove deprecated TransformableVideoFrame::GetAdditionalData
It was marked deprecated on Feb 9th, ~3 weeks ago.

Bug: chromium:1414370
Change-Id: I251b91984ca9a958e221f6eaf01c63b05c5a7a48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295506
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39422}
2023-02-28 16:23:52 +00:00
Tove Petersson
1fccaa4485 Reland "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
This reverts commit 8bf321062973939ef35f529640f5e69852e89a7e.

Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9())

Original change's description:
> Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
>
> This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084.
>
> Reason for revert: Breaks upstream project
>
> Original change's description:
> > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
> >
> > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
> >
> > Also default-initialized VideoFrameMetadata::ssrc_ to 0.
> >
> > Bug: webrtc:14708
> > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> > Commit-Queue: Tove Petersson <tovep@google.com>
> > Reviewed-by: Tony Herre <herre@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39411}
>
> Bug: webrtc:14708
> Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39413}

Bug: webrtc:14708
Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39418}
2023-02-28 15:44:21 +00:00
Danil Chapovalov
9f397217e1 Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats
Bug: None
Change-Id: I8d5ed723ce29231f805e6819156a16ba275f8e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295321
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39415}
2023-02-28 13:55:27 +00:00
Andrey Logvin
8bf3210629 Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084.

Reason for revert: Breaks upstream project

Original change's description:
> operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
>
> Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
>
> Also default-initialized VideoFrameMetadata::ssrc_ to 0.
>
> Bug: webrtc:14708
> Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> Commit-Queue: Tove Petersson <tovep@google.com>
> Reviewed-by: Tony Herre <herre@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39411}

Bug: webrtc:14708
Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39413}
2023-02-28 11:50:42 +00:00
Tove Petersson
437bf78ed9 operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.

Also default-initialized VideoFrameMetadata::ssrc_ to 0.

Bug: webrtc:14708
Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39411}
2023-02-28 08:32:09 +00:00
Tony Herre
6d262c504a Add TransformableVideoFrameInterface::Metadata()
Add a method to TransformableVideoFrameInterface which returns a new
instance of VideoFrameMetadata which the caller can move and use as
they like.
This will replace the existing GetMetadata which returns a dangerous const ref to a field which might change if someone calls SetMetadata
etc. That method will be deprecated as soon as we've migrated Chromium
usages.

Bug: webrtc:14708
Change-Id: Id7c15f33d6ec28c4a975ce250cdc791d7a3087bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39403}
2023-02-27 15:38:32 +00:00
Alessio Bazzica
ba3a1e2c26 Add RtpPacket test for the abs-capture-time extension
Clarify when the RTP header extension can be set depending on the
value of the `extmap-allow-mixed` option and on whether the header
extension ID is for one-byte or two-bytes extensions.

Bug: b/270541827
Change-Id: I4b939f6862d1f19cbfea11518a1cc1507beb2362
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294920
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39399}
2023-02-27 12:16:18 +00:00
henrika
274408feab Switch WGC to ScreenCaptureFrameQueue
* Avoids alloc/dealloc for each captured frame.
* Reduces time to capture first frame.
* Improves performance in terms of max FPS.

Bug: chromium:1412584
Change-Id: Ie16519ad788165c9553451ecea5adff12cd15eea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293582
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39384}
2023-02-24 09:39:00 +00:00
Erik Språng
ff1cf61cf3 Fix potentially bad rate control with libaom av1 encoder.
This can happen when the encoder uses real presentation timestamps that
originate with the input frames. By using those, the encoder can bypass
webrtc frame dropping logic and may severely over/under-shoot if the
timestamps are very precise. In practice, this seems rather common on
Chrome on Windows.

Bug: aomedia:3391
Change-Id: I2be5eed4fabc86dac8a6c7bfdd068c2dcb5a3743
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294740
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39382}
2023-02-23 18:54:57 +00:00
Diep Bui
5ece09b2db Init delay_detector_state to be kNormal
Bug: webrtc:14933
Change-Id: If667aac639ebd23b4aa2bb857d7db12c5cfefcae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294700
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39379}
2023-02-23 11:59:54 +00:00
Tony Herre
4c49190ac9 Add unittest for RtpSenderVideoFrameTransformerDelegate
Bug: webrtc:14708
Change-Id: I7926b3cfa6530e02eb13c31fecbc9e2e73f78f71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293744
Reviewed-by: Tove Petersson <tovep@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39375}
2023-02-22 20:17:35 +00:00
Palak Agarwal
a09f21b207 Introduce capture_time_identifier in webrtc::EncodedImage
This CL propagates capture_time_identifier introduced in
webrtc::VideoFrame and propagates it to EncodedImage. For use cases
involving EncodedTransforms, this identifier is further propagated to
TransformableVideoSenderFrame.

VideoEncoder::Encode function is overriden by each encoder. Each of
these overriden functions needs to be changed so that they can handle
this new identifier and propagate its value in the created EncodedImage.

Change-Id: I5bea4c5a3fe714f1198e497a4bcb5fd059afe516
Bug: webrtc:14878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291800
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39374}
2023-02-22 17:08:53 +00:00
Diep Bui
14e2779a6c Do not use acknowledge bitrate for backing off in alr.
Without the flag enabled: https://screenshot.googleplex.com/BZ6fqsNHQAUxbyU

With the flag enabled: https://screenshot.googleplex.com/4GKrfvAdUpsQDx9

Bug: webrtc:12707
Change-Id: Ia1a9761aeaedc57cff6a2d1eca3c61519b9dd26e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293660
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39372}
2023-02-22 14:38:23 +00:00
Philipp Hancke
b660b7a89c Enable multithreaded OpenH264 encoding behind field trial
This uses the field trial introduced is crbug.com/1406331 and
extends the usage to OpenH264. This simplifies experimentation
whether this change improves performance without requiring
multi-slice encoding.

BUG=webrtc:14368

Change-Id: I0031e59059f7113dd5453234869c957d46f311bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39371}
2023-02-22 14:26:33 +00:00
Zen Xu
b3c5bdb85a Allow video frame gaps in packet buffer for H.264
With LTR and SVC etc., H.264 should be able to skip lost frames, and continue to play from the new frames. With DependencyDescriptor, it is allowed to reference the previous frames, even there is a gap in the middle. However, we found there is a special logic for H.264 in packet_buffer.cc, which requires no gap for H.264.

We should allow gaps if the packet has GenericDescriptorInfo (either GenericDescriptor or DependencyDescriptor header extension).

Bug: webrtc:14887
Change-Id: Id66726bab33229bd883f257136ff2e8523fb44c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294062
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39370}
2023-02-22 13:51:10 +00:00
Danil Chapovalov
6aba07e5fe Account for mid and rrsid when reserving extra space for an rtx packet
Bug: webrtc:11031
Change-Id: I44405d0d15e885307b3134b1b88dcb74b96381fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294400
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39368}
2023-02-22 12:52:43 +00:00
Jesús de Vicente Peña
e7478182ac Penalization of large delays on the initial phase.
Bug: webrtc:14919
Change-Id: Iba00b2782b7e7c3dbd345a94aba541fad8c979ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294289
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39364}
2023-02-22 07:11:58 +00:00
Björn Terelius
f2c67e2253 Update documentation links in modules/desktop_capture/linux/wayland/screencast_portal.(cc|h)
Bug: webrtc:14917
Change-Id: I40e8f011b7263675aab99c452cda8f89ad137cc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294283
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39361}
2023-02-21 20:12:16 +00:00
Evan Shrubsole
262b2d8509 Delete WebRTC-SlackedTaskQueuePacer experiment
R=hbos@webrtc.org, sprang@webrtc.org

Bug: webrtc:14913
Change-Id: I1ea9d5bda798ea01fa9ec2a9b8d96cb50ccb9ec2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39356}
2023-02-21 13:11:49 +00:00
Michael Olbrich
cd5c62362f Build video capture implementation for chromium
It will be used to support cameras via xdg desktop portal / pipewire in
chromium. This includes exporting additional classes that will be used
by chromium.


Bug: webrtc:13177
Change-Id: I7524ffb47ed2eb7af1de4d7fd741fbb15277a0a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264553
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39350}
2023-02-20 15:29:43 +00:00
Henrik Boström
fbd0ddb32e Introduce WebRTC-VideoEncoderSettings/encoder_thread_limit:X.
As requested by a CEF hosted application (https://crbug.com/1406331)
who want to be able to limit the number of threads in a controlled
environment, this CL adds a flag to control the max limit per encoder.

For plumbing-reasons, this is placed in VideoEncoder::Settings but
with a note that this is considered an experimental API with limited
support. For now only LibvpxVp8Encoder uses it and there are no plans
to roll this out.

I have manually confirmed this is working with printf debugging,
--force-fieldtrials=WebRTC-VideoEncoderSettings/encoder_thread_limit:2
and https://jsfiddle.net/henbos/2bd6m7Lt/

Bug: chromium:1406331
Change-Id: Ib02bd83e2071034874843d3aaa0d3b0adc5bbf46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293960
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39349}
2023-02-20 14:01:32 +00:00