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}
Start using RobustThoughputEstimator in DelayBasedBwe test in preparation for making it default.
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: I83cfa1fc15486982b18cc22fbd0752ff59c1c1b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40644}
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}
This is to ensure that a bad NetworkState estimate can not decrease BWE
unless an delay BW overuse has been detected.
Bug: webrtc:10489
Change-Id: Ic3a516345999eeba814200c2e295a19b347b2eb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317800
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@google.com>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40628}
Remove unused includes, including a TODO that is now irrelevant
Add missing includes
Remove definitinon for constexpr class constants as not needed since c++17 to avoid adding include for RTPExtensionType
Bug: webrtc:10198
Change-Id: I5f0ed15c5a9020d8b2e58bdfa213bb38eb59a840
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317443
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40611}
When GenericFrameDescriptor or DependencyDescriptor RTP extensions are used, we may receive multiple consecutive StapA packets where only the first packet has is_first_packet_in_frame set. The previous code assumed that all StapA had is_first_packet_in_frame = true. Per discussion on the attached bug, removing the DCHECK is OK.
Bug: webrtc:15155
Change-Id: I6348740eac7d70bca2b7541721aaa7e2b5e5a970
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316941
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40608}
Expose new function MaybeCreateFrameDumpingEncoderWrapper that
wraps another passed encoder and dumps its encoded frames out
into a unique IVF file into the directory specified by the
"WebRTC-EncoderDataDumpDirectory" field trial. If the passed
encoder is nullptr, or the field trial is not setup, the function
just returns the passed encoder. The directory specified by the
field trial parameter should be delimited by ';'.
The new function is wired up in VideoStreamEncoder.
Bug: b/296242528
Change-Id: I6143adf899f78fcc03d4239a86c68dcbab483f1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317200
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40600}
Goal is to be able to get an improved overview of the distribution
of the total delay.
Bug: None
Change-Id: I0dced53eafd1fb09941590f3706480066c52419b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40597}
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}
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}
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}
