PacketBuffer is not designed to store wide range of the rtp sequence numbers
Bug: webrtc:15508
Change-Id: I62b19ba2896a667d795a41c38a60f55ee3f60566
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321845
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@google.com>
Cr-Commit-Position: refs/heads/main@{#40839}
Traditionally, we'd back off to 85% of the measured throughput in response to an overuse. However, this backoff doesn't appear to be sufficient to drain the queues in some low-bitrate scenarios, and the problem has gotten a bit worse with the RobustThroughputEstimator. (The new estimate looks more stable. The old estimator had more variation, the lowest points were lower, causing backoffs to lower rates.)
With this change, we back off to 0.85*thoughput-5kbps. The difference is negligible except at low bitrates.
Bug: webrtc:13402,b/298636540
Change-Id: I53328953c056b8ad77f6c7561d6017f171b2dfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321701
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40827}
To avoid name collision with Timestamp type,
To avoid confusion with capture time represented as Timestamp
Bug: webrtc:9378
Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40796}
NSApplicationActivateIgnoringOtherApps is about to be deprecated.
The default behavior is good enough.
Tested on Chrome using https://wicg.github.io/conditional-focus/demo/
Bug: webrtc:15511
Change-Id: I1f59aea3d4e7c4942d17ee5c4f1b6c2d398016ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321080
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40795}
A follow up cl/ is to remove passing upper link capacity from goog_cc to loss_based_bwe_v2.
Bug: webrtc:12707
Change-Id: I45af8ca6e8ba185700d0b7eb57004d2b61edeb9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40780}
This reverts commit 0f87b3853554ee5d4e92e487a5165b57771b6742.
This is not needed with the macOS 14 SDK, which has the fix, and which
was landed in https://crrev.com/c/4875713.
Bug: chromium:1484363, chromium:1431897
Change-Id: I1e019ce71b90333d5d1333a3cf8bb510a3dbd212
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320820
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Auto-Submit: Avi Drissman <avi@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40777}
When the specified device was not found in GetDevicesInfo,
SetPlayoutDevice/SetRecordingDevice will never return a (-1) error.
Bug: None
Change-Id: I9ac71cf72f7876c1c54ee593f184aa4007dba22f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320500
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40768}
The purpose is to not allow an initial low link capacity estimate to reduce the current estimate.
Only delay overuse detection , low probe results or a loss event can
reduce the estimate.
Bug: webrtc:14392
Change-Id: Ib1618347f2c7681e3bd65d85ee687dec3cd67c97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320380
Reviewed-by: Diep Bui <diepbp@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40751}
The sequence number is generally not used for the estimation,
but may be used as a tie-breaker when ordering packet feedbacks.
Bug: b/299667054
Change-Id: I52a5145c889c8f6924838667cc267b1cd9565f7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320240
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40749}
In change https://webrtc-review.googlesource.com/c/src/+/319961, I changed a error. Also the same code will be added for video to enable Glass 2 Glass metric for Android. To me it make sense to add this method, and then change the audio code and video code to use it.
Bug: None
Change-Id: Id5d38c3bb8266213a93e67ceb82e88d65f29de53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320080
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#40745}
Update RtpPacketizerH264::PacketizeStapA to use
single_packet_reduction_len when all fragments end up in one H.264
packet.
Previous code was using first_packet_reduction_len +
last_packet_reduction_len for this case, which can cause an occasional
RTC_CHECK crash in RtpPacketizerH264::NextAggregatePacket due to
exceeding the available payload capacity of an RTP packet.
Bug: webrtc:15477
Change-Id: Iba1564a6a29618bef22f19d82aba938420994b23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319645
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40737}
rtc::strcpyn second param should be the size of the destination buffer,
not the size of the source string. The result is that the final character
(usually a trailing directory path separator) is lost during the copy.
This has been masked because FormFileName helpfully adds a trailing path
separator if one is missing.
BUG=webrtc:15441
Change-Id: I992e69cad86a7e8bc2057ec629063f34c75fe75f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317502
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40736}
Only has an effect on Windows versions higher than 2104 (10.0.20348.0).
Bug: webrtc:15451
Change-Id: I3ca48c88a6c2b9b87d43805fcb2ade444cd90480
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318060
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40721}
This is a reland of commit a22c2a0c581cbe3f612f7a7d9fb9840186cc1e06
after systems depending on this have been fixed.
Original change's description:
> rtp sender: don't send BYE on deactivating streams
>
> as this breaks RTCP assumptions about SSRCs being no longer
> active as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.6
>
> This should not be sent in reaction to temporarily disabling
> a stream via RTCRtpParameters.active as this does not mean that
> the participant is leaving the session as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7
> and does not indicate end of participation as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.1
> which stipulates BYE should be the last packet sent from this SSRC.
>
> BUG=webrtc:11082
>
> Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38059}
Bug: webrtc:11082
Change-Id: Iad8b503b3101d1e684a4da2d1547b879e77b85dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293861
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40716}
These two functions contain complicated logic that will be used as
a fallback in Chromium if the new macOS picker code does not work
as intended.
Bug: chromium:1478172
Change-Id: I5f2878c5a8da38d59aa42ec1358398e3c921b65c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319260
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40711}
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}
Support the Pipewire videotransform meta via the already existing shared
infrastructure. This is needed for mobile devices which often have a 90
degree rotated camera - which is likely the reason there is already
support in the shared code paths.
Bug: webrtc:15464
Change-Id: I15223055d8675502ae326d270ebd2debbcfbfa50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318641
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40708}
This can be helpful in various situations, such as debugging with an
unrestricted Pipewire socket or for downstream projects like
B2G/Capyloon. Additionally it will help once we move from the camera
portal to the more generic device portal.
Original patch by Fabrice Desré <fabrice@desre.org>
Bug: webrtc:15464
Change-Id: Iae6802f242d68244bca85947cb15ef3eee923ab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318642
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40706}
BitrateTracker uses same implementation as RateStatistics, but provides api using Timestamp and DataRate types instead of plain numbers
Bug: webrtc:13756
Change-Id: Ie37fa58ede7590f870ec4376a64e7cf2c94431d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318841
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40697}
Combine all parameters into single struct so that it is easier to add and remove optional parameters
Use Timestamp type instad of plain int to represent capture time
Use rtc::ArrayView instead of pointer+size to represent payload
Merge passing audio level into send function.
Bug: webrtc:13757, webrtc:14870
Change-Id: I0386b710eb99b864334d61235add9abcde9bc69d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317442
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40688}
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}
The build target that CreateFromCGImage() belongs to, desktop_capture_obj
is not visible externally. A utility header is created to make it accessible.
Bug: chromium:1471931
Change-Id: Ie40f39114d277dc4b62fe2ce95a6b0c7b61a3943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318123
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Johannes Kron <kron@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40659}
Experiments has not showed significant metric changes. However, simulations has showed that RobustThroughputEstimator better follow the actually receive rate better. Especially during bursts of sent packets. Code is also simpler.
Bug: webrtc:13402 chromium:1411666
Change-Id: I38c309f74e8e1322602196354545b3a465866263
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318040
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40653}
