All the builder machines have been upgraded to Mac-14.
Change-Id: Ia6fe055e21bcf483d08debc85109b36dedf18c5b
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357864
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42683}
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}
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}
Prior to https://crrev.com/c/5740609, NewContents never flushed cache
so the second NewContents() would always produce the same contents
post-yapf as as pre-yapf. Flush cache on second NewContents() call to
get updated file contents. Also fix the formatting a bit.
Bug: b/333744051
Change-Id: Ic627dd72675d7d3694b1978635ae047b38f06596
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357960
Auto-Submit: Gavin Mak <gavinmak@google.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42677}
this is flipping
WebRTC-PermuteTlsClientHello
to a killswitch in the SSLStreamAdapter used for DTLS.
BUG=webrtc:42225803
Change-Id: I942851c474ec5e723c5b6c9f6206e7eafbe80ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42676}
cog workspaces don't have a git directory and can't run "git diff".
Replace it with python's difflib instead.
Bug: b/333744051
Change-Id: I5bd8ccd873a0db55f0bbadf165180b3f2aa42903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357900
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42674}
wanted_fps_ seems redundant with target_capture_fps_.
The problem with wanted_fps_ is that it lowers the capture fps but does not decimate frames so that a 30 fps stream played at 5 fps is played slowly instead of played at the normal speed with dropped frames.
Change-Id: I1440953f9909ad1d4a102a0671fe933d95498a1f
Bug: b/355120692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42670}
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}
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}
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}
iOS 15.5 is not tested anymore and we start to test on iOS 18.0.
Change-Id: Ia7340d25f6cf8480763ea689db267c0c9a843319
Bug: b/353975341
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42659}
Skip CQ because it is currently broken and changes to config.star are not picked up by CQ anyway.
Bug: None
Change-Id: I3fb6c1fd8db6466b6f058f10d1232cc1624e0472
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357541
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42658}
This is a reland of commit 09f03be54804e81f626c26e8fde8c86cc952545f
Use max_num_layers instead of encoder_config.number_of_streams when calculation stream resolutions in EncoderStreamFactory::GetStreamResolutions().
Original change's description:
> Pass true stream resolutions to GetSimulcastConfig()
>
> Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
>
> Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
> * GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
> * GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
> * GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
> * GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
> * GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
> * GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
> * GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
> * GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
>
> Bug: webrtc:351644568, b/352504711
> Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42651}
Bug: webrtc:351644568, b/352504711
Change-Id: Ib3fd859257b61c2a5d695b8b8f45c95495117c0e
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42654}
This reverts commit 09f03be54804e81f626c26e8fde8c86cc952545f.
Reason for revert: breaks downstream projects
Original change's description:
> Pass true stream resolutions to GetSimulcastConfig()
>
> Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
>
> Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
> * GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
> * GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
> * GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
> * GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
> * GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
> * GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
> * GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
> * GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
>
> Bug: webrtc:351644568, b/352504711
> Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42651}
Bug: webrtc:351644568, b/352504711
Change-Id: I7aadbe49419b7ac610db4db99284fdcdce9deff5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42653}
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}
Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
* GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
* GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
* GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
* GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
* GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
* GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
* GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
* GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
Bug: webrtc:351644568, b/352504711
Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42651}
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}
Merge GetAndSetRtpSendParametersScaleResolutionDownByVP8, GetAndSetRtpSendParametersScaleResolutionDownByVP8WithOddResolution, GetAndSetRtpSendParametersScaleResolutionDownByH264 and GetAndSetRtpSendParametersScaleResolutionDownByH264WithOddResolution into one parameterized test.
PS. Not sure why we need separate tests for VP8 and H264. Underlaying code paths are codec agnostic as I can see.
Bug: webrtc:351644568, b/352504711
Change-Id: Ic95c59c1bfacdba8a42de8ecca9cab42014842e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42646}
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}
QualityConvergenceController is a layer between VideoStreamEncoder
and QualityConvergenceMonitor that takes care of the simulcast
logic.
Bug: chromium:328598314
Change-Id: Iad8a9d9138e69a60fd508a7ef038220947888f0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356420
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42642}
All calls in code under test were migrated to AudioEncoderFactory::Create and thus there is no longer need to propagate older function.
Bug: webrtc:343086059
Change-Id: I9e0ea4024759deb22c0d284e0e4bac7322a08f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357181
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42638}
