- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
This replaces webrtc::VideoSendStream::DegradationPreference with
webrtc::DegradationPreference, and adds "DISABLED".
It's still not wired up from RtpSenderInterface::SetParameters to the
underlying video engine; that would be the next step.
Bug: webrtc:8830
Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864
Reviewed-on: https://webrtc-review.googlesource.com/77024
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23276}
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
Include dropped frames by the encoder in the frame drop percentage.
To react faster at low framerates:
- Use ExpFilter instead of MovingAverage to filter QP values.
- Reduce sampling interval while waiting for minimum number of needed frames (when not in fast rampup mode).
A separate slower ExpFilter is used for upscaling.
Bug: webrtc:9169
Change-Id: If7ff6c3bd4201fda2da67125889838fe96ce7061
Reviewed-on: https://webrtc-review.googlesource.com/70761
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23014}
This is a reland of fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org
Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.
Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22905}
This is a reland of dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8
Original change's description:
> Storing frame if encoder is paused.
>
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
>
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
>
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}
Bug: webrtc:8415
Change-Id: I0ea7d4d679e7845907cfbe9a120f128ff2180e4b
Reviewed-on: https://webrtc-review.googlesource.com/68580
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22810}
This reverts commit dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8.
Reason for revert: breaks downstream projects
Original change's description:
> Storing frame if encoder is paused.
>
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
>
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
>
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}
TBR=sprang@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8415
Change-Id: I4449eca65a64e2bc2fb25d866e5775e9a085cee9
Reviewed-on: https://webrtc-review.googlesource.com/68280
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22788}
Adds a pending frame to VideoStreamEncoder that is used to store frames
that are not sent because the encoder is paused. If the encoder is
resumed within 200 ms, the pending frame will be encoded and sent. This
ensures that resuming a stream instantly starts sending frames if it is
possible.
This also protects against a race between submitting the first frame
and enabling the encoder that caused flakiness in end to end tests
when using the task queue based congestion controller.
Bug: webrtc:8415
Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
Reviewed-on: https://webrtc-review.googlesource.com/67141
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22781}
This CL replaces ConfigureEncoderTask with a slightly simpler struct
that is defined within the scope of the function using it. This makes
it more clear that it is only used once and slightly reduces the amount
of code.
Bug: None
Change-Id: I181a3da90540aa514048ff77fc9a9e9cf19d8f34
Reviewed-on: https://webrtc-review.googlesource.com/63026
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22514}
Intended to make it easier to wire up cpu-adaptation experiments.
To setup the circular references between OveruseFrameDetector and
VideoStreamEncoder, let the AdaptationObserverInterface pointer be
an argument to StartCheckForOveruse.
Bug: webrtc:8504
Change-Id: Ifcf7655ec65e637819d77f507552cb22a6aa5f0f
Reviewed-on: https://webrtc-review.googlesource.com/33340
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22006}
Implementation of the EncodedImageCallback and
AdaptationObserverInterface belongs with the implementation, not the
public interface.
Bug: webrtc:8830
Change-Id: I901ae79ede252599c5fe22284cf649274c5b1422
Reviewed-on: https://webrtc-review.googlesource.com/51862
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22000}
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.
Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
And wire it up to methods on RTCConfiguration, via MediaConfig::Video.
Bug: webrtc:8504
Change-Id: I30805ee20c11d1d2fe552eb81f16d514db0ba4a8
Reviewed-on: https://webrtc-review.googlesource.com/39786
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21670}
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.
Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
In balanced adaptation mode, a 1280x720 feed would only ever be reduced in
resolution twice, and would never have its framerate reduced (due to an
interaction with MinFps()).
This change removes the hard limits entirely, instead relying only on
kMinFramerateFps and VideoEncoder::ScalingSettings::min_pixels_per_frame.
Deleted SinkWantsFromOveruseDetector test because it duplicates other tests.
Fixed DoesntAdaptDownPastMinFramerate; it wasn't testing what it claimed to
because it wasn't updating the fake clock correctly, meaning FPS was detected as
0, meaning framerate adaptation was never triggered.
Bug: webrtc:8068, b/38207842
Change-Id: If99d0e74c1334879c1b0c3117eb079f5f2139851
Reviewed-on: https://webrtc-review.googlesource.com/31644
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21312}
This callback was used only by the PrintSamplesToFile feature of
video_quality_test, which looks like it has been broken for some time
(due to mixup of capture time and ntp time).
Bug: webrtc:8504
Change-Id: I7d2b55405caeffda582ae0d6fb0e7dfdfce4c5a9
Reviewed-on: https://webrtc-review.googlesource.com/31420
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21211}
Add fine grained dropped video frames counters on sending side
4 new counters added to SendStatisticsProxy and reported to UMA and logs.
Bug: webrtc:8355
Change-Id: I1f9bdfea9cbf17cf38b3cb2f55d406ffdb06614f
Reviewed-on: https://webrtc-review.googlesource.com/14580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20421}
system_wrapper/Atomic32 has been deprecated (which is already just a
wrapper of std::atomic) in favor of platform-independent std::atomic
from C++11. This CL replaces all use of Atomic32 in video/
Bug: webrtc:8428
Change-Id: If4dab4909df06944c009e7b70141f58daef7be10
Reviewed-on: https://webrtc-review.googlesource.com/14720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Yuwei Huang <yuweih@google.com>
Cr-Commit-Position: refs/heads/master@{#20417}
TBR=sprang@webrtc.org
This is a reland of af721b72cc1bdc5d945629ad78fbea701b6f82b9
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
Bug: webrtc:8375
Change-Id: I06ea90ae8646ba11ddd8ddceb82ea82d75ae2109
Reviewed-on: https://webrtc-review.googlesource.com/11320
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20308}
This reverts commit af721b72cc1bdc5d945629ad78fbea701b6f82b9.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
TBR=asapersson@webrtc.org,sprang@webrtc.org
Change-Id: Ic914f03ff7065ac410ae06b6f82b56a935399b66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8375
Reviewed-on: https://webrtc-review.googlesource.com/8480
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20248}
Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
Store sent frame info in map to solve potential issue where sent framerate statistics could be
incorrect.
Bug: webrtc:8375
Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
Reviewed-on: https://webrtc-review.googlesource.com/7880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20225}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
