If WebRTC-AutomaticAnimationDetectionScreenshare experiment is enabled,
content type is screenshare and degradation preference is BALANCED,
then input resolution is restricted if update_rect of the incoming frames
is the same for considerable amount of time and is big enough.
This entails treating BALANCED degradation preference for screenshare as
MAINTAIN_RESOLUTION in adaptation logic.
Bug: webrtc:11058
Change-Id: I903dddf53fcbd7c8eac6c5b1447225b15fd8fe5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161097
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30002}
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}
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}
Together with RtpDepacketizer refactoring that would reduce
number of memcpy while handling an rtp packet
Bug: webrtc:11152
Change-Id: I6f4e09c93af5e2a9314967a15eac8ced57ec712e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161087
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@{#29985}
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.
Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
There are edge cases where the caching of encoder info will cause
issues. For instance if a sub-encoder fails en Encode call and falls
back to some other implementation, or if the fps targets shift due to
SetRates() triggering new layers to be enabled.
This CL forces a complete rebuild on every call to GetEncoderInfo().
It also adds new logging of when the info changes, as debugging issues
can be very time consuming if we can't tell that happened.
Bug: webrtc:11000
Change-Id: I7ec7962a589ccba0e188e60a11f851c9de874fab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29938}
Removed old code, which took care of VP8 case, but actually interferes with
VP9 case.
Now regardless of codec bw_limited_resolution is calculated based on signals
from the bitrate allocator.
Bug: webrtc:11015
Change-Id: Ic99dbb504ab2d1a4b5f15ca93193a1af05ae5924
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160651
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29937}
This change makes it easier to propagate more information from the sink
to the video adapter, for example alignment requirements.
Bug: None
Change-Id: I536248d59f871c103a18a48615b6c5e61f61697b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29827}
Some of the field trial default values are changed as well.
Now available bitrate estimation will be decreasing when RTT is more than 3 seconds.
Unless different parameters for the field trial are specified.
Bug: None
Change-Id: Icd1923fc2e2e7766a7f645016c5432a52537145f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158840
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Nikita Zetilov <zetilovn@google.com>
Cr-Commit-Position: refs/heads/master@{#29823}
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}
Instead of taking TaskQueue from outside create one internally.
Detach MultiStreamTests from test::CallTest since that inheritance
only used for constants and (now unneeded) task_queue object.
Bug: webrtc:10933
Change-Id: I7e30ddcf6faaa134ebcd9d53b578b40fdedf2a3c
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-Commit-Position: refs/heads/master@{#29744}
task_queue_ outlives the BandwidthStatsTest object, but Posted task
captures |this|. Blocking call in the destructor is a simple way to avoid
that race
(should work as long as posted task doesn't call virtual functions from |this|).
Bug: webrtc:10933
Change-Id: Id30badb711480af5ee737b96b9224c1a73e730ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158898
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29707}
BandwidthEndToEndTest failed when I tested it with the new task-queue
based paced sender. This turned out to be issues with this test.
Problems fixed by this CL:
1. Send-side BWE not set up correctly. Caused probing to fail.
2. Test waited for non-zero pacer delay, but the new pacer will not
generate any delay in this scenario.
3. Race condition during shutdown of test.
1) Is just a matter of configiuring the right header extension.
2) Set up test with high encoder bitrate to trigger pacer delay.
3) TaskQueue outlives the Call instances used in test, so make sure
they are not referenced from lambda during teardown.
Bug: webrtc:10809
Change-Id: I6393975691dfa05eb5b25d9283e476062e23a876
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158722
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29669}
The encoder can drop all frames for extended periods if it has produced over budget. After 2 seconds without any encoded frames, the video send stream times out and deallocates the stream.
Ideally the send stream should keep track if frames are captured instead of encoded, but keeping the stream alive using OnDroppedFrame can work as a proxy for that.
Bug: webrtc:11062
Change-Id: Id7ec1ff333427643453c4a36d1db03ca826cd9ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29662}
For the automatic content type detection we need to know if the update
rect is trusted or just not available.
Currently we only care if it's not empty, so in case of no update rect
available, full frame resolution was set as a changed region.
This CL makes the update_rect field optional but should be a no-op in the
current code, as absence of update_rect is treated as a full update via
a new getter method |update_rect_or_full_frame()|.
Bug: webrtc:11058
Change-Id: I913545b71ac2fc845861549ac34eb1b630012109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158673
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29654}
Add new histograms
WebRTC.Video.DecodeTimePerFrameInMs.[codec].[resolution].[decoder]
These histograms are more explicit than the existing histogram
WebRTC.VideoDecodTimeMs, since they allow to see performance per
codec/resolution/decoder and also contain per frame statistics instead
of an average decode time.
There's a killswitch, WebRTC-DecodeTimeHistogramsKillSwitch, that can be
used to disable the histograms.
Bug: chromium:1007526
Change-Id: I9f75127b4bc5341e9f406c64ed91164564290b26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157881
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29572}
this change is intentionally noop.
Goal is to minimize change that would replace the
SingleThreadedTaskQueueForTest with a regular task queue.
Bug: webrtc:10933
Change-Id: I6da768941af048de3716af13e41b8f0f1ccd4cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157892
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29569}
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}
that brings RtpPacketReceived closer to the packet buffer
to allow strore original packets rather than VCMPacket in it.
Bug: webrtc:10979
Change-Id: Ia0fc0abf3551a843b19b0ee66ca0f20cae014479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157164
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29516}
This is a reland of 96f3de094566f32d842be6dd0906f1d13b8c8825
Downstream test is fixed, this is a pure reland.
TBR=danilchap@webrtc.org,srte@webrtc.org
Original change's description:
> Use just a lookup map of RTP modules in PacketRouter
>
> Since SSRCs of RTP modules are now set at construction time, we can
> use just a simple unordered map from SSRC to module in packet router.
>
> Bug: webrtc:11036
> Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29510}
Bug: webrtc:11036
Change-Id: I0731339dfd0781cc7f2f7ca78ac903539f25ff9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29514}
This reverts commit 96f3de094566f32d842be6dd0906f1d13b8c8825.
Reason for revert: Downstream test is borked.
Original change's description:
> Use just a lookup map of RTP modules in PacketRouter
>
> Since SSRCs of RTP modules are now set at construction time, we can
> use just a simple unordered map from SSRC to module in packet router.
>
> Bug: webrtc:11036
> Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29510}
TBR=danilchap@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I31330fd68ab809ff3951573791e9a79b81599958
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29511}
Since SSRCs of RTP modules are now set at construction time, we can
use just a simple unordered map from SSRC to module in packet router.
Bug: webrtc:11036
Change-Id: I0b3527f17c9ee2df9253c778e5b9e3651a70b355
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155965
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29510}
