31 Commits

Author SHA1 Message Date
Artem Titov
ab30d72b72 Use backticks not vertical bars to denote variables in comments for /video
Bug: webrtc:12338
Change-Id: I47958800407482894ff6f17c1887dce907fdf35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227030
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34585}
2021-07-28 13:22:27 +00:00
Markus Handell
06a2bf09a4 NackModule2: Rename to NackRequester.
The alternative new name proposed, NackTracker, is already in
use in audio_coding.

Fixed: webrtc:11594
Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34539}
2021-07-23 08:30:33 +00:00
Markus Handell
0e62f7aa98 NackModule2: coalesce repeating tasks.
NackModule2 creates repeating tasks, but as there are
many modules (one per receiver) these tasks execute out
of phase with each other, multipliying the amount of wakeups
caused.

Fix this by creating a single wakeup source that serves all
NackModule2 instances in a call.

Bug: webrtc:12989
Change-Id: Ia9c84307eb57349679e42b673474feb2cb43f08e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226464
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34527}
2021-07-22 12:11:13 +00:00
Markus Handell
eb61b7f620 ModuleRtcRtcpImpl2: remove Module inheritance.
This change achieves an Idle Wakeup savings of 200 Hz.

ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is
enabled in RtcpSender, which it never is. Hence the Module
inheritance could be removed. The change removes all known
dependencies of the module inheritance, and any related mentions
of ProcessThread.

Fixed: webrtc:11581
Change-Id: I440942f07187fdb9ac18186dab088633969b340e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34358}
2021-06-22 14:51:04 +00:00
Tommi
d350006b70 Add rtp_config() accessor to ReceiveStream.
This is a consistent way to get to common config parameters for
all receive streams and avoids storing a copy of the extension
headers inside of Call. This is needed to get rid of the need of
keeping config and copies in sync, which currently is part of why
we repeatedly delete and recreate audio receive streams on config
changes.

Bug: webrtc:11993
Change-Id: Ia356b6cac1425c8c6766abd2e52fdeb73c4a4b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222040
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34285}
2021-06-14 17:57:57 +00:00
Tommi
376cf07ea2 Add packet_sequence_checker_ to RtpVideoStreamReceiver2.
Specifying guards for functions and member variables. Also updating
a few places for VideoReceiveStream2 accordingly.

Bug: webrtc:11993
Change-Id: I2d13b009ec9853c6b2d90b08af555ecdd2b1ced6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220765
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34174}
2021-05-31 18:49:13 +00:00
Tommi
90738ddb4e Split VideoReceiveStream2 init into worker / network steps.
This is in preparation for actually doing this initialization
differently in the Call class. This CL takes the registration
steps that are inherently network thread associated and makes
them separate from the ctor/dtor.

Inject Call* instead of worker_thread(), which will simplify upcoming
work that needs to access the network_thread() as well.

This is related to:
https://webrtc-review.googlesource.com/c/src/+/220608
https://webrtc-review.googlesource.com/c/src/+/220609

Bug: webrtc:11993
Change-Id: I72769fd61de84967d9a645750c40d01660a2716b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220764
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34172}
2021-05-31 17:10:23 +00:00
philipel
2182096e66 RtpFrameReferenceFinder return frames directly instead of via callback.
Bug: webrtc:12579
Change-Id: I41263f70a6f3dc60167e41f8b015a7d3b0dc3dd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219633
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34136}
2021-05-26 15:47:03 +00:00
Markus Handell
588f9b3705 VideoReceiveStream2: AV1 encoded sink support.
This change adds support for emitting encoded frames
for recording when the decoder can't easily read out
encoded width and height as is the case for AV1 streams,
in which case the information is buried in OBUs. Downstream
project relies on resolution information being present for key
frames. With the change, VideoReceiveStream2 infers the
resolution from decoded frames, and supplies it in the
RecordableEncodedFrames.

Bug: chromium:1191972
Change-Id: I07beda6526206c80a732976e8e19d3581489b8fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214126
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33662}
2021-04-08 20:07:22 +00:00
philipel
ca18809ee5 Move RtpFrameObject and EncodedFrame out of video_coding namespace.
Bug: webrtc:12579
Change-Id: Ib7ecd624eb5c54abb77fe08440a014aa1e963865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33542}
2021-03-23 14:22:47 +00:00
philipel
6a6715042a Move RtpFrameReferenceFinder out of video_coding namespace.
Namespace used because of copy-pasting an old pattern, should never have been used in the first place. Removing it now to make followup refactoring prettier.

Bug: webrtc:12579
Change-Id: I00a80958401cfa368769dc0a1d8bbdd76aaa4ef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212603
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33536}
2021-03-23 08:48:37 +00:00
Johannes Kron
16359f65c4 Delay creation of decoders until they are needed
Before this CL, WebRTC created a decoder for each negotiated codec
profile. This quickly consumed all available HW decoder resources
on some platforms. This CL adds a field trial,
WebRTC-PreStreamDecoders, that makes it possible to set how many
decoders that should be created up front, from 0 to ALL. If the
field trial is set to 1, we only create a decoder for the
preferred codec. The other decoders are only created when they are
needed (i.e., if we receive the corresponding payload type).

Bug: webrtc:12462
Change-Id: I087571b540f6796d32d34923f9c7f8e89b0959c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208284
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33300}
2021-02-19 12:08:49 +00:00
Tomas Gunnarsson
8408c9938c Remove 'secondary sink' concept from webrtc::VideoReceiveStream.
In practice, support for multiple sinks is not needed and supporting
the API that allows for dynamically adding/removing sinks at runtime,
adds to the complexity of the implementation.

This CL removes that Add/Remove methods for secondary sinks as well
as vectors of callback pointers (which were either of size 0 or 1).
Instead, an optional callback pointer is added to the config struct
for VideoReceiveStream, that an implementation can consider to be
const and there's not a need to do thread synchronization for that
pointer for every network packet.

As part of webrtc:11993, this simplifies the work towards keeping
the processing of network packets on the network thread. The secondary
sinks, currently operate on the worker thread.

Bug: webrtc:11993
Change-Id: I10c473e57d3809527a1b689f4352e903a4c78168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207421
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33272}
2021-02-15 18:08:17 +00:00
Artem Titov
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
Mirko Bonadei
20e4c80fbe Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447

This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:29:36 +00:00
Johannes Kron
111e981466 Signaling for low-latency renderer algorithm
This feature is active if and only if the RTP header extension
playout-delay is used with min playout delay=0 and max playout delay>0.

In this case, a maximum composition delay will be calculated and attached
to the video frame as a signal to use the low-latency renderer algorithm,
which is landed in a separate CL in Chromium.

The maximum composition delay is specified in number of frames and is
calculated based on the max playout delay.

The feature can be completetly disabled by specifying the field trial
WebRTC-LowLatencyRenderer/enabled:false/

Bug: chromium:1138888
Change-Id: I05f461982d0632bd6e09e5d7ec1a8985dccdc61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190141
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32493}
2020-10-26 15:03:56 +00:00
Mirko Bonadei
0abd518abd Revert "Introduce RTC_NO_UNIQUE_ADDRESS."
This reverts commit f5e261aaf65cdf2eb903cdf40d651846be44f447.

Reason for revert: Breaks downstream projects.

Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
2020-10-07 07:37:01 +00:00
Mirko Bonadei
f5e261aaf6 Introduce RTC_NO_UNIQUE_ADDRESS.
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.

The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.

Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
2020-09-30 09:52:49 +00:00
Ivo Creusen
bef7b058f5 Make AV sync robust to failures to set a desired minimum delay
Setting a minimum delay can fail in some cases. It is important that the
AV sync code is aware of failures and can act accordingly to recover and
prevent sync delays that keep increasing indefinitely.

Bug: webrtc:11805
Change-Id: I0deed951dc6c6d0905536a949af875e0a6d9f7fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32062}
2020-09-09 15:44:47 +00:00
Danil Chapovalov
2549f174b5 Remove RTPFragmentationHeader creation and propagation through webrtc
Bug: webrtc:6471
Change-Id: I5cb1e10088aaecb5981888082b87ae9957bbaaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31955}
2020-08-17 16:37:33 +00:00
Niels Möller
2a70703eb8 Delete MediaTransportInterface and DatagramTransportInterface
Bug: webrtc:9719
Change-Id: Ic9936a78ab42f4a9bb4cc3265f0a2cf36946558f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31536}
2020-06-17 08:41:14 +00:00
Tommi
63673fe2cc Remove locks and dependency on ProcessThread+Module from NackModule2.
Change-Id: I39975e7812d7722fd231ac57e261fd6add9de000
Bug: webrtc:11594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175341
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31367}
2020-05-27 14:20:34 +00:00
Tommi
d3807da009 Fork NackModule and RtpVideoStreamReceiver
Bug: webrtc:11595
Change-Id: I4d14c0bf9c32e09d1624099a256f2778afebd4df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175901
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31337}
2020-05-22 17:07:16 +00:00
Tommi
a98cea863d Remove the PendingTaskSafetyFlag::Pointer type add ScopedTaskSafety.
ScopedTaskSafety simplifies usage of PendingTaskSafetyFlag,
so this CL also includes ToQueuedTask support for ScopedTaskSafety
and test updates.

This is following up on feedback in the following CL:
https://webrtc-review.googlesource.com/c/src/+/174262

Change-Id: Idd38dfc1914b24a05fdc4ad256b409dcf1795fc0
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174740
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31241}
2020-05-13 14:17:39 +00:00
Tommi
822a874463 Switch CallStats to TQ interface + callbacks on the worker thread.
Bug: webrtc:11489
Change-Id: I08c4cd42dfa28d88ed9f0aa8c8b2cfb606bf00df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174240
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31203}
2020-05-10 23:24:35 +00:00
Tommi
ad84d0254a Remove locking from RtpStreamsSynchronizer.
Remove dependency on ProcessThread.

Instead RtpStreamsSynchronizer uses the worker thread
and makes callbacks on the same thread. That in turn
simplifies locking for VideoReceiveStream2, which we'll
take advantage of later.

Bug: webrtc:11489
Change-Id: Id9a5a7977771b92e420a09cc472cfb43de5627cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174221
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31200}
2020-05-10 18:11:44 +00:00
Tommi
d7e08c8cf8 Move processing of frame meta data for OnFrame/OnRenderedFrame to the worker thread
Bug: webrtc:11489
Change-Id: I9f88fec0aef449fd8923c5eec81cddf9ee42316b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174220
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31199}
2020-05-10 11:47:52 +00:00
Tommi
6a871d3487 Revert "Remove playout delay lock."
This reverts commit c623495fd1ff90aada0eb625af91ec17843fefd0.

Reason for revert: Need to look into failure in remoting_unittests in Chrome (Webrtc/ConnectionTest.SecondCaptureFailed/0). It looks like the order FrameBuffer2 calls into VCMTiming while receiving frames and updating playout delay values, needs to be synchronized better.

Original change's description:
> Remove playout delay lock.
> Now update the playout delay and related stats on the worker thread.
> 
> This was previously reviewed here:
> https://webrtc-review.googlesource.com/c/src/+/172929/
> 
> With the exception of reducing unnecessarily broad
> lock scope in one function in rtp_rtcp_impl.cc
> and added comments in rtp_streams_synchronizer.h
> 
> Bug: webrtc:11489
> Change-Id: I77807b5da2accfe774255d9409542d358f288993
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174200
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31193}

TBR=tommi@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11489
Change-Id: I9149025d2fc10686314e6d4e89d1b92125650c36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174757
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31197}
2020-05-09 21:30:40 +00:00
Tommi
c623495fd1 Remove playout delay lock.
Now update the playout delay and related stats on the worker thread.

This was previously reviewed here:
https://webrtc-review.googlesource.com/c/src/+/172929/

With the exception of reducing unnecessarily broad
lock scope in one function in rtp_rtcp_impl.cc
and added comments in rtp_streams_synchronizer.h

Bug: webrtc:11489
Change-Id: I77807b5da2accfe774255d9409542d358f288993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174200
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31193}
2020-05-08 19:02:36 +00:00
Tommi
553c869c58 Start consolidating management/querying of stats on the Call thread.
Call is instantiated on what we traditionally call the 'worker thread'
in PeerConnection terms. Call statistics are however gathered, processed
and reported in a number of different ways, which results in a lot of
locking, which is also unpredictable due to the those actions themselves
contending with other parts of the system.

Designating the worker thread as the general owner of the stats, helps
us keeps things regular and avoids loading unrelated task queues/threads
with reporting things like histograms or locking up due to a call to
GetStats().

This is a reland of remaining changes from https://webrtc-review.googlesource.com/c/src/+/172847:
This applies the changes from the above CL to the forked files and
switches call.cc over to using the forked implementation.

Bug: webrtc:11489
Change-Id: I93ad560500806ddd0e6df1448b1bcf5a1aae7583
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174000
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31186}
2020-05-08 07:24:39 +00:00
Tommi
74fc574cbc Fork a few VideoReceiveStream related classes.
We'll need to deprecate the previous classes due to being used externally
as an API.

Bug: webrtc:11489
Change-Id: I64de29c8adae304d0b7628e24dd0abc5be6387ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31136}
2020-04-27 09:25:47 +00:00