Commit Graph

417 Commits

Author SHA1 Message Date
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
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
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
435fb9ad06 Remove screen_share_config from the VideoConfig.
After the migration of the pc framework tests (https://webrtc-review.googlesource.com/c/src/+/174023), having "absl::optional<ScreenShareConfig> screen_share_config" field in VideoConfig became redundant. Replaced it with VideoTrackInterface::ContentHint content_hint field.

Bug: webrtc:11534
Change-Id: Ibf4b1c8daed95ef02111fe952171f11e290905d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174702
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31187}
2020-05-08 08:56:13 +00:00
c064467b32 Pass frame generator to the AddVideoConfig method in the pc framework tests.
Bug: webrtc:11534
Change-Id: Id68feca50611f412897ddef3d43b811a224b200f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174023
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31167}
2020-05-05 17:20:25 +00:00
1148fd5cef Define MockFrameTransformer in test/.
Add MockFrameTransformer to test/, and remove definitions from unit test
files.

Bug: webrtc:11380
Change-Id: Ia709883e8d000852e3f71e7bfb87877072e22aeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174001
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31151}
2020-05-04 13:45:22 +00:00
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
ce0a11d5f9 Unify AdaptationReason and AdaptReason enums.
Moves the unified AdaptationReason to the api/ folder.

Bug: webrtc:11392
Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31084}
2020-04-16 13:33:49 +00:00
3c9bcc1f7a Reland of the test portion of:
https://webrtc-review.googlesource.com/c/src/+/172847

------------ original description --------------

Preparation for ReceiveStatisticsProxy lock reduction.

Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).

Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.

One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.

Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.

Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.

Bug: webrtc:11489
Change-Id: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
2020-04-15 16:09:44 +00:00
16cc9efd54 Revert "Preparation for ReceiveStatisticsProxy lock reduction."
This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.

Reason for revert: Speculative revert: breaks downstream project

Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
> 
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
> 
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
> 
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
> 
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
> 
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
> 
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}

TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org

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

Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
2020-04-07 19:50:20 +00:00
24eed2735b Preparation for ReceiveStatisticsProxy lock reduction.
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).

Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.

One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.

Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.

Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.

Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
2020-04-06 14:34:38 +00:00
06d3559b79 Replace std::string::find() == 0 with absl::StartsWith (part 2).
This CL has been generated using clang-tidy [1] except for changes to
BUILD.gn files.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/abseil-string-find-startswith.html

Bug: None
Change-Id: Ibf75601065a53bde28623b8eef57bec067235640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172586
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30984}
2020-04-02 14:38:30 +00:00
78964c1e0a Transform encoded frames in RtpVideoStreamReceiver.
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: If4ffcfe5761492a2ae5513ec46deb9f837e8aee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169130
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30755}
2020-03-11 09:46:57 +00:00
62057627ef [Adaptation] Rename and move resource adaptation module/processor stuff.
Rename:
- call/adaptation/resource_adaptation_module_interface.[h/cc] -->
  call/adaptation/resource_adaptation_processor_interface.[h/cc]
- call/adaptation/resource_adaptation_processor.[h/cc] -->
  call/adaptation/new_resource_adaptation_processor_poc.[h/cc]

Move + Rename:
- video/overuse_frame_detector_resource_adaptation_module.[h/cc] -->
  video/adaptation/resource_adaptation_processor.[h/cc]

Move:
- video/encode_usage_resource.[h/cc] --> video/adaptation/...
- video/overuse_frame_detector.[h/cc] --> video/adaptation/...
- video/quality_scaler_resource.[h/cc] --> video/adaptation/...

Unittests are also moved. In order to avoid a circular dependency,
VideoStreamEncoder::kDefaultLastFrameInfo[Width/Height] is moved and
renamed to kDefaultInputPixels[Width/Height] in
video/adaptation/resource_adaptation_processor.[h/cc].

Bug: webrtc:11222
Change-Id: Icf920e8a7362002b1c63c42b2d9e2e63c990b532
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170117
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30754}
2020-03-11 09:30:16 +00:00
efbec9a304 [Overuse] Initial version of VideoStreamAdapter (Restrictor moved).
This CL simply moves the VideoSourceRestrictor from being an inner class
of OveruseFrameDetectorResourceAdaptationModule to a new class,
VideoStreamAdapter.

In follow-up CLs, the responsibility of determining what the next step
for adapting up or down should also be moved to the VideoStreamAdapter.

The end-goal is that the VideoStreamAdapter takes care of "can adapt?"
and "do adapt!" type of logic so that a multi-stream aware adaptation
module can decide which stream (adapter) to adapt, and the adapter can
take care of the nitty gritty details of doing so.

In this CL the "can?"/"do!" part is realized but not the logic for
determining what the next step up or down is, and the class interface
needs improvement.

This CL also sets up the video/adaptation/ subdirectory and moves the
AdaptationCounters class here. Other adaptation-related classes (e.g.
the module and its resources) should move into this directory as well
in the future.

Bug: webrtc:11393
Change-Id: I2c12c1281eca854c62791abb65f0aca47a119726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169542
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30705}
2020-03-06 12:20:01 +00:00
33be9dfe7a Replace AdaptCount with a single counter.
There is still a counter for the active counts for the
scaling, but these will be removed at a later date.

BUG=webrtc:11392

Change-Id: Ie9bcf3f744a0bbac601f0da61197f4bac1e9f879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169447
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30701}
2020-03-06 08:43:47 +00:00
412a31bbf8 Insert frame transformer between Depacketizer and Decoder.
Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.

The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130.

This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30654}
2020-03-02 08:33:44 +00:00
aa6fbc156e Support injecting new Resources for overuse
* This replaces the video stream methods for forcing adaptation
with a mock resource that triggers overuse.
* Resources can now be injected to the Module using the AddResource
function.
* Resources now have tests for adding and removing callbacks.
* Quality/EncoderUse% resources are tracked in the Resource list of
the adaptation module.
* The adaptation module ties all resources to a reason to keep stats
working as expected.

BUG=webrtc:11377

Change-Id: I1f5902f7416dc41b4915c0072e6f0da2bb3bb2b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168948
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30610}
2020-02-25 16:17:42 +00:00
9526c557be Refactoring mock_transport to be used separately
Bug: webrtc:11251
Change-Id: I0a494c34c8d5c458b4d9b1b3616ae360d04df0d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168980
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30584}
2020-02-21 17:02:52 +00:00
9b05803e19 Implement injectable EncoderSelectorInterface and wire it up in the VideoStreamEncoder.
The EncoderSelectorInterface is meant to replace the "WebRTC-NetworkCondition-EncoderSwitch" field trial, so the field trial will be ignored if an EncoderSelectorInterface object has been injected.

Bug: webrtc:11341
Change-Id: I5371fac9c9ad8e38223a81dd1e7bfefb2bb458cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168193
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30490}
2020-02-10 12:12:47 +00:00
d4c3c3a454 Move video_replay under rtc_tools/.
As pointed out in [1], RTC public tools should live in rtc_tools.

[1] - https://webrtc-review.googlesource.com/c/src/+/168320/2#message-1f40103105ecb077aeec153c5270575138349a50

Bug: chromium:942546
Change-Id: Ic827d9b31ade9a32bf4ef24d020ef8c81d2c9a5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168308
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30486}
2020-02-07 17:57:30 +00:00
f12231d742 Add wildcard visibility to video_replay to make it buildable in Chromium.
Bug: chromium:942546
Change-Id: Ib798b58e854a2471ab1bb94725cb0ee2b04b84da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168320
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Max Moroz <mmoroz@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30477}
2020-02-06 21:41:31 +00:00
065348503c [Overuse] Move EncodeUsageResource/QualityScalerResource to own files.
This CL changes EncodeUsageResource and QualityScalerResource from
private inner classes of OveruseFrameDetectorResourceAdaptationModule to
standalone classes, moving them into separate files.

This CL does not intend to change any lines of code, only move them.
Except for removing an unused method quality_scaler().

Bug: webrtc:11222
Change-Id: I86bf7eb78c80031888c403ac43c2bdf9b24eaea6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168198
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30472}
2020-02-06 14:08:39 +00:00
a9e1026304 Make video_replay buildable from Chromium.
Bug: chromium:942546
Change-Id: Ic127e74b75ccb1fa65b317711d20344d0caee5fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168280
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30467}
2020-02-06 10:55:22 +00:00
33aaa35d54 Fix video_replay to build and actually work
Add it to default build target, so it won't get broken accidentally
again. Fix configuration issue with field trials (new parameter was
added recently, but wasn't set by video_replay)

Bug: webrtc:11287
Change-Id: I9c18746d899acd7ac68c1b9b3a646b862c41897a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166900
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30345}
2020-01-22 13:16:28 +00:00
64f1f3f04e Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED
Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 15:20:35 +00:00
07b17df771 Move DegradationPreference logic to the encoder queue.
This moves SetHasInputVideoAndDegradationPreference() to the encoder
queue. OveruseFrameDetectorResourceAdaptationModule is now entirely
single-threaded, including its inner class VideoSourceRestrictor.

VideoStreamEncoder now protects the module with RTC_GUARDED_BY. This
ensures it is safely used, even without a SequenceChecker inside of the
module. The module's |encoder_queue_| is removed.

The one task queue reference that is needed - passing down the current
task queue to StartCheckForOveruse() - is replaced by a TaskQueueBase*
(instead of rtc::TaskQueue*), enabling obtaining the current queue with
TaskQueueBase::Current(). (There is no rtc::TaskQueue::Current().)

Furthermore, the only uses of VideoSourceSinkController that isn't on
the encoder queue are documented, with a TODO saying if these are moved
the VideoSourceSinkController could also be made single-threaded.
However since this requires introducing a delay to
VideoStreamEncoder::SetSource() and VideoStreamEncoder::Stop(),
arguably a more risky change, if this is to be attempted that should be
in a separate CL.

Bug: webrtc:11222
Change-Id: I448ca5125708d5f66b95b0b180d6d24cc356dfa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165783
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30263}
2020-01-15 11:58:04 +00:00
ce0ea49001 VideoStreamEncoder configuring source/sink with VideoSourceController.
This is part of the work for making VideoStreamEncoder responsible for
configuring its source/sink and limiting the responsibility of
OveruseFrameDetectorResourceAdaptationModule to only output relevant
VideoSourceRestrictions.

BEFORE THIS CL

Prior to this CL, OveruseFrameDetector was responsible for performing
AddOrUpdateSink() on the source, which it did using its nested class
VideoSourceProxy.

AddOrUpdateSink() could happen for both adaptation and non-adaptation
related reasons. For example:
- Adaptation related: AdaptUp() or AdaptDown() happens, causing updated
  VideoSourceRestrictions.
- Non-adaptation related: VideoStreamEncoder asks the module to
  reconfigure the source/sink for it, such as with
  SetMaxFramerateAndAlignment() or SetWantsRotationApplied().

AFTER THIS CL

AddOrUpdateSink() is performed by VideoSourceController, which is owned
by VideoStreamEncoder. Any reconfiguration has to go through the
VideoStreamEncoder. This means that:
- Non-adaptation related settings happen between VideoStreamEncoder and
  VideoSourceController directly (without going through the adaptation
  module).
- Adaptation related changes can be expressed in terms of
  VideoSourceRestrictions. OveruseFrameDetectorResourceAdaptationModule
  only has to output the restrictions and not know or care about other
  source/sink settings.

For now, VideoSourceController has to know about DegradationPreference.
In a future CL, the DegradationPreference logic should move back to
the adaptation module. The VideoSourceRestrictions are fully capable of
expressing all possible source/sink values without the "modifier" that
is the degradation preference.

Bug: webrtc:11222
Change-Id: I0f058c4700ca108e2d9f212e38b61f6f728aa419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162802
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30228}
2020-01-13 11:14:04 +00:00
539f9b376e Use a TaskQueue for decoding in VideoStreamDecoderImpl.
Long term goal is to use the VideoStreamDecoder in the VideoReceiveStream so
that we can stop using legacy VideoCodingModule components and classes. This CL is
one of several in preparation for that.

Bug: webrtc:7408, webrtc:9378
Change-Id: Ifd7e4c3c7d38dbb7c4b0636aaad318c571a29158
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164525
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30211}
2020-01-10 14:31:22 +00:00
382cc6d8a6 Add incomplete ResourceAdaptationModuleInterface.
This interface will be improved upon iteratively to aid reviewability.
The initial version only handles starting and stopping the module; input
and output of the module is still implementation-specific.

TBR=sprang@webrtc.org

Bug: webrtc:11222
Change-Id: Ie307cfe3d3211c84346c035f2c0e9a632f58221b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162580
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30167}
2020-01-07 13:24:42 +00:00
b08882b625 Refactor out VideoStreamEncoder's overuse logic to separate module.
This CL puts the VideoStreamEncoder's current adaptation logic inside
the new class OveruseFrameDetectorResourceAdaptationModule. The
intention is not to change any behavior, only to move code.

Future CLs should step by step decrease the coupling between
OveruseFrameDetectorResourceAdaptationModule, VideoStreamEncoder and
the VideoStreamEncoder's QualityScaler by introducing more abstract
interfaces. This is not done in this CL because it is large enough as
it is, but the long term goal is to make it possible to replace the
existing overuse module with a different implementation.

This CL relies on existing tests exercising the VideoStreamEncoder, but
part of making overuse logic modular should include testing each module
separately as well as continued integration testing of the
VideoStreamEncoder.

Bug: webrtc:11222
Change-Id: I316a174adfd00d60cdd224a23a5f616efd235d13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161953
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30163}
2020-01-07 09:55:54 +00:00
b57fe17e7c Migrate video tests and tool to VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I1e7868ca88b162db8615cb4903bd89d3daac4827
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161452
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30085}
2019-12-13 11:41:04 +00:00
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00
486cc55a02 TimeController: Rename Sleep to AdvanceTime.
This change renames TimeController's Sleep method to AdvanceTime, unifying
the same name with the same semantic as for downstream projects.

Bug: webrtc:11154
Change-Id: Id79bcf0eafcd0b47a76407ba220479d84df5a736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161092
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29989}
2019-12-03 16:08:54 +00:00
269ac81a86 VideoReceiveStream: Enable encoded frame sink.
This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames
can flow to sinks installed in VideoTrackSourceInterface.

Bug: chromium:1013590
Change-Id: I0779932c251a2159880a39b2d42d5ce439cc88e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161090
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29988}
2019-12-03 15:55:04 +00:00
0197887d71 Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in MultiStreamTester
Bug: webrtc:10933
Change-Id: I61ae0726fb197e5a779e036b5b1390c29ca96aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159714
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29829}
2019-11-19 10:52:12 +00:00
aa3f5da8dc Fork VCMPacket for PacketBuffer into own struct
it is easier to reduce and eliminate it when it is not bound to legacy video code

Bug: webrtc:10979
Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-18 15:48:07 +00:00
1242d9cc48 Reland Cleanup MultiStreamTester
Instead of taking TaskQueue from outside create one internally.
Detach MultiStreamTests from test::CallTest since that inheritance
only used for constants and for task_queue object.

Unlike original cleanup
keep using DEPRECATED_SingleThreadedTaskQueueForTesting for now.

Bug: webrtc:10933
Change-Id: Ife9143bfda0ebefd56a9199622296e64b14a7b20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159034
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#29744}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159280
Cr-Commit-Position: refs/heads/master@{#29782}
2019-11-13 08:53:22 +00:00
e644a03195 Add field trial for rampup in quality based on available bandwidth.
Bug: none
Change-Id: I32e1ea6fb2f2e20fc631e09b02c8f3a11b6c9fac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158888
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29751}
2019-11-11 10:13:28 +00:00
9cd53b4910 Avoid DEPRECATED_SingleThreadedTaskQueueForTesting::CancelTask in VideoAnalyzer
Bug: webrtc:10933
Change-Id: Iba24100b092df7306ee77f6592ad5469c541099a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157901
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29559}
2019-10-21 12:51:57 +00:00
82a3f0ad7f Replace SingleThreadedTaskQueueForTesting::SendTask usage with ::webrtc::SendTask
Bug: webrtc:10933
Change-Id: I60738434b46e77b4644173ad168bc0efa58459b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156001
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29551}
2019-10-21 08:45:02 +00:00
86d053c2db Use source_sets in component builds and static_library in release builds.
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).

Source sets always pass all the object files to the linker.

On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.

See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set

Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:17:18 +00:00
80f53b785b Extend WebRTC-Video-MinVideoBitrate to experiment per-codec
The experiment was extended to support per-codec minimum bitrates
for the following codecs:
 * VP8
 * VP9
 * H.264

The old semantic meaning for the field trial is retained, in that
specifying "br:" applies a minimum bitrate to all codecs. If "br:"
is not specified, the per-codec minimum config is consulted.

Bug: webrtc:11024
Change-Id: I89630262c7710771d5e25d039fe35f0bd217b58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156171
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29450}
2019-10-11 15:34:33 +00:00
7c06777ab0 Cleanup includes in modules/include/module_common_types.h
Add missing includes to files that were transactivly depending on removed includes.

Bug: None
Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29396}
2019-10-07 16:06:26 +00:00
44db436e87 Propagate task queue to create test::DirectTransport by TaskQueueBase interface
actual task queue implementation for these tests is intentionally unchanged for now.

while at it, change return type of created transports to unique_ptr to note passing ownership.

Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
2019-09-30 03:23:07 +00:00
01dd88505c Moves contents of bitrate_controller to goog_cc
This CL moves send_side_bandwidth_estimation.cc/h and
loss_based_bandwidth_estimation.cc/h from modules/bitrate_controller
to modules/congestion_controller/goog_cc.

Bug: webrtc:9883
Change-Id: Ibb2c2ba3762007e7e5114f39042ee96431b73776
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154346
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29297}
2019-09-25 08:43:24 +00:00
738bfa7bab Remove api/bitrate_constraints.h.
Bug: webrtc:8733
Change-Id: Iaeb26e07d399f25dc18b0c4af38ed400577a5d3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29217}
2019-09-18 06:37:58 +00:00
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
eaaaf41298 Introduce api/crypto/BUILD.gn.
No-Try: True
Bug: webrtc:8733
Change-Id: I8679735be1e5069e371a9f1115a54e897e09964b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152622
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29186}
2019-09-13 17:21:47 +00:00
65f17ca6b4 Move MediaTransportInterface out of the libjingle_peerconnection_api target
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.

Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-13 10:49:56 +00:00