instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.
Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue
Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
Fix call in FrameBuffer3Proxy to OnDecodableFrameTimeout to match with
FrameBuffer2Proxy and how OnEncodedFrame is called.
Bug: none
Change-Id: I6f5f18bd9a425aa55f5826654b24c2c0ee5af5f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269302
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37610}
rtc::TaskQueue is a simple wrapper over TaskQueueBase and adds no
extra features when task queue is used without passing ownership.
Reducing usage of the internal rtc::TaskQueue wrapper gives users more flexibility how TaskQueueBase* is stored.
Bug: webrtc:14169
Change-Id: If5c8827544c843502c7dfcef775ac558de79ec3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268189
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37549}
SetFeedbackParameters no longer recreates the embedded streams for:
- transport cc flag
- rtcp status
Bug: none
Change-Id: If6117a1ae760ca9a02f06bbfa2b46c6e0f448cfc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268281
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37526}
The decode time per frame and codec profile histograms were added
temporarily to make it possible to get an overview of the decode
time distributions. This fine grained information is not needed
longer and the histograms can be deleted.
Bug: chromium:1007526
Change-Id: Ie59627a88813e0710700cf0e13eedd6627010266
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266496
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37316}
This is the first CL out of three to make the low-latency stream signaling
explicit. At the moment this is done by setting the render time to 0.
There's a dependency between Chromium and WebRTC which is why this is
split into three CLs to not break any existing functionality.
Bug: chromium:1327251
Change-Id: Ie6b268746d587a99334485db77181fb2c6e9b567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264502
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37225}
When a large queue of frames builds up due to a lost frame, the decode
delay can sometimes become quite large. In this case the stream may
signal as timed out when in fact it is not. Instead, the delay should
be capped at the timeout limit.
Bug: webrtc:14168
Change-Id: I5b4e8851b2c6d7d27a698627dc1633931d7fc00e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265404
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37199}
Setting the transport cc flag was only possible post creation for
audio receive streams, while video receive streams need to be recreated.
This CL moves the setter for transport_cc() to where the getter is and
adds boiler plate implementations for the video streams. For audio
streams this splits "SetUseTransportCcAndNackHistory" into two methods,
SetTransportCc and SetNackHistory.
Bug: none
Change-Id: Idbec8217aef10ee77907cebaecdc27b4b0fb18e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264443
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37038}
There are two cases that can be confusing for applications developers
which may result in the playout delay not being set as intended.
First, it is not well defined which min playout delay should be used
when multiple are set. This changes adds a warning to alert application
developers that they are setting multiple playout delays.
Second, if the playout delay header extension is used, developers must
be careful that the max playout delay is always larger than the min
playout delay, otherwise the behaviour is undefined. This change logs an
error when this case is detected.
Bug: None
Change-Id: I8477d48ef64636da080792362fa898e42f038bef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263202
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36977}
Putting these classes in a sub folder increases
structure and clarifies that they are used as
helper classes. Affected classes in this change:
* CodecTimer
* InterFrameDelay
* RttFilter
VCMTiming will be moved in a separate CL.
Additional changes:
* Remove VCM prefix from class names.
* Introduce granular BUILD.gn targets.
* Update some includes.
Bug: webrtc:14111
Change-Id: Ia75128aa955a819033b97d4784cb61904de7230b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262960
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36975}
GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36839}
Instead offer accessors for the specific config values from the struct
that are needed at different times. The remote_ssrc and rtx_ssrc
properties maybe accessed from any thread, other properties have
stricter requiremets.
Bug: webrtc:11993
Change-Id: I3ff8527b13452c773fae1b2574f1e3fd2583b481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261319
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36823}
This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.
Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36820}
The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.
Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
There is no active use of it, and the fields are enabled by default in
the uses of it.
Change-Id: Ibfdb3f1befca886cb4b2f4b2ae4d6235aafafe3d
Fixed: webrtc:13998
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256262
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36655}
This patch takes a stab at modules/video_coding,
but reaches only about half.
Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
* 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}
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}
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}
Fix multiple use-after-move issues in VideoReceiveStream2, as found by
clang-tidy:
video/video_receive_stream2.cc:259:
'config' used after it was moved
video/video_receive_stream2.cc:199:
move occurred here
Bug: chromium:1122844
Change-Id: I6367dc835f002718a5353c3e0b64c2a154e79925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241201
Auto-Submit: Maksim Ivanov <emaxx@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35530}
This was a remenant leftover from a previous design, which was no longer
valid after the switch to TaskQueues. ReturnReason::kStopped was not
used at all, and so Timeout or FrameFound can be inferred from whether
the frame is null or not.
Bug: webrtc:13343, webrtc:13346
Change-Id: Ib0f847b1e1192e32ea11208e48f5a3892703521e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239651
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35490}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
The experiment WebRTC-PreStreamDecoders (aka Lazy decoder creation) has
investigated the benefit of only creating a subset of all decoders
during negotiation and the remaining decoders on demand.
This CL changes the default value to only create one decoder during
negotiation. This frees up hardware resources and reduces the SDP
negotiation time.
Bug: chromium:1202042
Change-Id: I6e2206839162aa857fcc948ccd53d0ff91cbdeaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231643
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34959}
Setting the rtp header extensions on the packet delivery thread
(currently worker, soon to be network), is now possible without
taking the hit of deleting and recreating the receive stream (and
rtp receiver and related state).
Bug: webrtc:11993
Change-Id: I9bbe306844a25d85d79cd216092ead66eaf68960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223741
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34953}
Instead offer getters for the sync_group and rtp struct. Both are
a part of the config but expose much less of the config, which has
mutable parts.
Bug: none
Change-Id: Icc8007246e9776a5d20f30cda1a2df3fb7252ffc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229980
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34838}
The config struct is big and in order to control access to its state,
some of which isn't always const, we need to limit raw unlocked access
to it from other classes.
Bug: none
Change-Id: I4513c41486e79ef6c5cfd6376122ab338ad94642
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229921
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34835}
from runtime check in proxy classes that picks decoder (VCMDecoderDataBase)
to a DCHECK in the VideoDecoder::Settings
Bug: None
Change-Id: Ic8c2e971486a3a7eb247f9d03815aba5ca5a7bad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228644
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34761}
