New ctor added without optional and media specific fields.
Bug: webrtc:10739, b/246753278
Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38124}
The Chromium RTCVideoEncoder unfortunately doesn't set if the
result is at target quality, and the definition of the threshold
is buried in libvpx_vp8_encoder.h.
This change
* Updates VideoStreamEncoder to postprocess an incoming EncodedImage
by interpreting the incoming QP information instead.
* Updates the related VideoStreamEncoder test to simulate an encoder
producing images around the QP threshold.
* Updates the steady state VP8 screencast QP threshold to a central
include file.
* Moves this and previously existing EncodedImage post-processing to a
new method AugmentEncodedImage.
Bug: b/245029833
Change-Id: I69ae29ffe501e84f28908f7d9a8cfd066ba82b43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38091}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
* Migrate callback to one-time absl::AnyInvocable.
* Clean tests to use MockFunction.
* Use main thread instead of helper function in unittests.
* Fix some spelling mistakes.
Bug: None
Change-Id: I6145f5f5e2748dfa5278898cfdfd762c1840ff8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274170
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38016}
The new frame buffer is already the default. Sync decoding can now be
inferred by the presence of a metronome rather than using the field
trial.
Tests have been updated to use the DecodeSynchronizer rather than the
field trial.
Bug: webrtc:14003
Change-Id: I33b457feaf4eac1500f3bf828680e445ae4d93cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274163
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38011}
The frame cadence adapter would previously crash on encountering
unconfigured layer updates. Fix this to silently ignore such updates.
Fixed: webrtc:14417
Change-Id: I524aa196f6aed10ffb8cd4ddcb4b053c71dfabba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273325
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37980}
This moves the ownership away from VideoReceiveStream2 and closer to
VCMDecoderDataBase. That facilitates unregistration (upcoming change)
without recreating receive streams.
Bug: none
Change-Id: I812175134730a0ffbf7077fd149c8489481c73d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37866}
This reverts commit 4f79b1d2e5f8754237657904dd1e6aa766fb6282.
Reason for revert: Still used in one project. I'll make a fix for that and then reland this.
Original change's description:
> cleanup obsolete sps-pps-idr field trial
>
> which has been superseeded by the equivalent nonstandard sdp fmtp
> sps-pps-idr-in-keyframe
> parameter.
>
> Bug: webrtc:11769
> Change-Id: I02667a165dd3f86b4685530c43f19531ec654737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271121
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37839}
Bug: webrtc:11769
Change-Id: I11e097e00813b7b232e01b236510cbf1b2850843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272560
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37862}
Structs with user-declared constructors are not aggregates in C++20.
This fixes a C++20 compile error.
Bug: chromium:1284275
Change-Id: Iaeab959fc72ac6bf395af57d10808baee2db533f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272522
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37854}
Now Configure(), Decode() and Release() calls to the decoders should
all happen on the decoder thread. Added thread checkers to verify.
Bug: None
Change-Id: I2a1cf2cf7f3c3c7c50e382d82a3638e916ed9c34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272368
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37840}
which has been superseeded by the equivalent nonstandard sdp fmtp
sps-pps-idr-in-keyframe
parameter.
Bug: webrtc:11769
Change-Id: I02667a165dd3f86b4685530c43f19531ec654737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271121
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37839}
This way we're sure instantiation, configuration and decode calls all
happen on the decoder queue - making thread checking easier in the
actual decoder classes.
Bug: None
Change-Id: Ia98f47009f26b34eb8dad2ee0b4ddcde082d1994
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272022
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37825}
Encapsulate the results in a struct and move the post back to the packet
sequence in the original method. This allows the calls to
ReceiveStatisticsProxy::OnPreDecode to be moved to the main thread
always.
Bug: webrtc:11993
Change-Id: I37516444efeabdb330868733bccb1e842ea4d54f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271285
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37823}
This was only set to nullptr in non-test environments and was thusly
unused. With this change, the stats callbacks are gaurenteed to only
come from the VideoStreamBufferController and so the thread checks can
be removed.
Bug: webrtc:14003
Change-Id: Iaf0e77aa7c45a317e38ae27739edcefd3123d832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37816}
Moves FrameBuffer2 to its own GN target to reduce the binary size of the
video target.
Bug: None
Change-Id: I40e86a1eabc0c9e8e6fada3dcdb4e3a043c61c6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271286
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37803}
It no longer has to interact with the decode queue, that will only
happen in VideoReceieveStream2. This moves some members in
VideoReceieveStream2 to the packet sequence which removes a few
post-tasks.
Bug: webrtc:14003, webrtc:11993
Change-Id: I4641b593b1a2f68e017c384b73ee4e75d06cf559
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271700
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37802}
This makes the class concrete, and the former FrameBuffer3Proxy is the implementation.
Bug: webrtc:14003
Change-Id: Ife825b9f4efc7b79d9be8b4afb03904da819958a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265868
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37793}
A follow-up change will combine the setters for ulpfec and red payload
types, since they're entwined.
Bug: webrtc:11993
Change-Id: Ifea7fe9f4ebc7ac88a62db6cd6748f4d3c20db4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271482
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37785}
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
Resolves an issue where, in Chrome, WebRTC event logs do not capture outgoing packets for video receivers because no reference to the event log was passed to the video receiver.
Bug: webrtc:14338
Change-Id: Ia33ce6f2d69a0e341530648b10a08516dc53abf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271080
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37746}
This behaviour has been fixed with the introduction of FrameBuffer3
Bug: webrtc:14033, webrtc:13343, webrtc:9974
Change-Id: Iba81c169706336e85194ed141324466e44a2c859
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265867
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37742}
The data that's used to report the histograms is owned by UlpfecReceiver
and moving the reporting there, simplifies things as configuration
changes happen in RtpVideoStreamReceiver2 (which currently require all
receive streams to be deleted+reconstructed).
Additional updates:
* Consistently using `Clock` for timestamps. Before there was
a mix of Clock and rtc::TimeMillis.
* Update code to use Timestamp and TimeDelta.
Bug: none
Change-Id: I89ca28ec7067a49d6b357315ae733b04e7c5a2e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271027
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37729}
We only need to see which bitrates have been configured, no need to
wait for failed frame. This should also reduce test durations somewhat.
Bug: None
Change-Id: Ie081310f9f80e21039c78d8c80510769cb400c3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270747
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37711}
This replaces use of RunLoop and SimulatedClock. As a related change,
units like TimeDelta and Frequency are used as needed.
Bug: None
Change-Id: I892ee38641f2fd37d4bd1b0fb425bfb5d4706ac1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270626
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37708}
This allows for `config_.rtp.nack.rtp_history_ms` to be modified
without deleting and recreating video receive streams.
Bug: webrtc:11993
Change-Id: I8ba132b22fe0e6de03d1c42fc38a570cbe138817
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269301
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37701}
...to allow for turning on/off loss notifications for video receive
streams without tearing down and recreating the whole stream.
Bug: webrtc:11993
Change-Id: Ia961bd343ce816ffe3414f11e3a58bb3c235307c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269252
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37688}
follow-up from https://webrtc-review.googlesource.com/c/src/+/262810
* replace Time::Millis(0) and TimeDelta::Millis(0) with ::Zero()
* drop unnecessary webrtc namespace from some TimeDeltas
* make TimeDelta do the unit conversion for stats
BUG=webrtc:13756
Change-Id: Ic60625ae0fc7959a47a6be9f5051851feaf76373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37664}
