These methods were only used for testing.
Change-Id: Icbb6a3cc59cbc0b5e1f42efcb86a7203704b92d8
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256362
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36293}
- Use NtpTime instead of pair of uint32_t to represent ntp time
- Increase precision estimate with NtpTime precision instead of ms precision
- Hide helper structs as private types
- Modernize interface to prefer return values over output parameters
- embed LinearRegression helper into the only user: UpdateParameters
Bug: webrtc:13757
Change-Id: I0a62a03e2869b2ae1eacaa15253accc43ba0a598
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36232}
convert almost all of video/ (and the collateral)
Bug: webrtc:10335
Change-Id: Ic94e05937f54d11ee8a635b6b66fd146962d9f11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36192}
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.
Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
Additionally,
* Moved to its own GN target.
* Added unittests.
* Removed unused variable `_zeroWallClock`.
* Renamed variables to match style guide.
* Moved fields _dTS and _wrapArounds to variables.
Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36147}
* Uses DataSize to represent incoming and outgoing bytes.
* Puts units into doubles as they enter the Kalman filter
* Moved to its own GN target.
Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36143}
This test method previously didn't account for retransmissions that can
fail for valid reasons e.g. retransmitting too soon or overusing the
retransmission bandwidth budget.
Now, we keep periodically requesting retransmissions until the packets
we seek have been received.
Bug: webrtc:13480
Change-Id: Ica620049b2c6c123117f5fa270dae0a08b320948
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252283
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36061}
Ensures that frames are decoded instantly when in low-latency render
mode. This also tests the max queue size behaviour. Adds a new test
suite for FrameBufferProxy that sets the appropriate field trials.
* Fixes FrameDecodeTiming to never use negative wait times for decode
timestamps.
R=kron@webrtc.org
Change-Id: I06cbec52e1e866e21aa964b24c4fd0163c26961b
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35999}
Wires up DecodeSynchronizer in Call if there is a Metronome injected
into the PeerConnectionFactoryDependencies.
Change-Id: I362cd12648bfa0c32e73111fcd0f3296fca2b275
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251341
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35996}
Adds new class DecodeSynchronizer that will coalesce the decoding
of received streams on the metronome. This feature is experimental and
is backed by a field trial WebRTC-FrameBuffer3.
This experiment now has 3 arms to it,
"WebRTC-FrameBuffer3/arm:FrameBuffer2/": Default, uses old frame buffer.
"WebRTC-FrameBuffer3/arm:FrameBuffer3/": Uses new frame buffer.
"WebRTC-FrameBuffer3/arm:SyncDecoding/": Uses new frame buffer with
frame scheduled on the metronome.
The SyncDecoding arm will not work until it is wired up in the follow-up
CL.
This change also makes the following modifications,
* Adds FakeMetronome utilities for tests using a metronome.
* Makes FrameDecodeScheduler an interface. The default implementation is
TaskQueueFrameDecodeScheduler.
* FrameDecodeScheduler now has a Stop() method, which must be called
before destruction.
TBR=philipel@webrtc.org
Change-Id: I58a306bb883604b0be3eb2a04b3d07dbdf185c71
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250665
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <holmer@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35988}
The scalability mode could be something invalid set by user, in this
case, |num_spatial_layers| should not be updated.
Bug: chromium:1292923
Change-Id: I78e1a6f12cf6d165597205608e4c124117a3d01b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/main@{#35985}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
This emulates behaviour from frame buffer 2, but does not handle stats.
In contrast to frame buffer 2, all work happens on the same task queue.
FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind
a field trial WebRTC-FrameBuffer3.
This separates frame scheduling behaviour into a few components,
VideoReceiveStreamTimeoutTracker
* Handles the stream timeouts.
FrameDecodeScheduler
* Manages the scheduling and cancelling of frames being sent to the
decoder.
FrameDecodeTiming
* Handles the timing and ordering of frames to be decoded.
Other changes
* Adds CurrentSize() method to FrameBuffer3
* Move timing to a separate library
* Does a thread check for Receive statistics as this is now
on the worker thread.
* Adds `FlushImmediate` method to RunLoop so that
video_receive_stream2_unittest can pass when scheduling is happening
on the worker thread.
Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721
Bug: webrtc:13343
Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35847}
This is a delegate that is used by video_receive_stream2 to handle frame
buffer tasks like threading, and stats. This will be used in a follow up
to use FrameBuffer3 as a strategy selected by field trial.
Unit-tests will be used in follow-up CLs containing Frame Buffer 3, and
are expected to work with both Frame buffer proxy versions.
Change-Id: I524279343d60a348d044d9085d618f12d7bf3a23
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241605
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35803}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
Some of the state that's managed in VideoStreamEncoder, is updated
and accessed on the encoder queue, but a method that's used for testing
only (GetAdaptationResources()), represents a race between PostTask
operations that update state and checking for said state on the worker
thread.
This CL removes Wait() operations related to adaptation resources from
the common path and puts one in the test path instead.
Bug: webrtc:13612
Change-Id: Ie3e018e815e24951bc0634ed70de17eaf336a508
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249220
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35797}
This patch adds a method for setting the keyframe request method
to VideoReceiveStream.
This code exists in the version that Mozilla is shipping, with a review
https://phabricator.services.mozilla.com/D105773 .
Bug: webrtc:13486
Change-Id: I7cc19dec95d6523368d73395319854bd8c2166f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240140
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35793}
Use cases of TaskQueue or TaskQueueBase that are considered high
precision are updated to make use of PostDelayedHighPrecisionTask
(see go/postdelayedtask-precision-in-webrtc) instead of PostDelayedTask.
The cases here are the ones covered by that document, plus some
testing-only uses. The FrameBuffer2 and DataTracker use cases will
be covered by separate CLs because FrameBuffer2 uses
RepeatingTaskHandle and DataTracker uses dcsctp::Timer.
This protects these use cases against regressions when PostDelayedTask
gets its precision lowered.
This CL also adds TaskQueue::PostDelayedHighPrecisionTask which calls
TaskQueueBase::PostDelayedHighPrecisionTask (same pattern as for
PostDelayedTask).
Bug: webrtc:13604
Change-Id: I7dcab59cbe4d274d27b734ceb4fc06daa12ffd0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248864
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35781}
Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance.
Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`.
Bug: webrtc:13573
Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35763}
Currently if encoder initialization fails WebRTC doesn't send any video.
This CL adds functionality that changes encoder type in such case and
restores the video. If encoder selector is available we switch to
encoder it recommends. Otherwise, VP8 is used as the default fallback
encoder.
Bug: webrtc:13572
Change-Id: Ifcdf707a575711f5ff81f9451caf30140c9171dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35761}
This CL updates both the static GOF pattern with the correct flags for
temporal_up_switch, as well the flexible mode logic to base the flag
on dependency descriptors instead use reference buffers.
Bug: webrtc:13576
Change-Id: I578f744bec51d1f3531da5f4a89d12f05a16a6c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247187
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35741}
the new spelling is more standard and more compact, in particular doesn't need extra include and thus dependency
Bug: None
Change-Id: Iaea69d2154e4d9eff2468514f5734cb3fe016ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35709}
Field telemetry has shown the combination of min_fps = 0 and max_fps >
0 is unused in the wild. Therefore it's safe to turn the
WebRTC-ZeroHertzScreenshare field trial default on unless the field
trial is disabled.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: Iea701218aa178b569333087b004106ffe2e85133
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244086
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35621}
The frame cadence adapter ignores key frame processing that happens
before the point where zero-hertz mode is activated, which leads to
no refresh frame requests if the key frame request comes too early.
Fix this to register a pending refresh frame request that gets
serviced when zero-hertz mode is activated. The CL changes the
FrameCadenceAdapterInterface::ProcessKeyFrameRequest from returning
whether to request a refresh frame into the frame cadence adapter
actively doing so itself via a new Callback::RequestRefreshFrame
API.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I53c2dbf6468e883eb2a2e81498e7134b1b35c336
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242963
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35598}
Timestamps are currently dead-reckoned for repeated frames in
zero-hertz mode. This leads to an ever increasing
totalPacketSendDelay metric in chrome://webrtc-internals which is
bad.
Fix this by tracking the origin timestamp of the first delay and
measuring time's progression since then. A unit test was added
which fails with the previous version.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I8627b91424f9bc56305b1dbd6a4c0624b6b3669d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242863
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35595}
The QP value of encoded key frames is normally very large. However,
the zero-hertz mode is retaining quality convergence info, leading
to only a single short repeat on key frame request when idle
repeating.
Fix this by resetting quality convergence information on key frame
requests, ensuring zero-hertz mode goes back to idle repeating
only when quality has converged again.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: Ia1ad686cc98007f01c8aaef9162011837575938c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242862
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35594}
