The vcm::VideoReceiver class is used by both VideoReceiveStream and
the legacy api VideoCodingModule. They have different requirements,
since the latter uses the old jitterbuffer and runs the code on a
ProcessThread.
By making a copy and trimming it down to what's actually used by
VideoReceiveStream, we can drop the dependency on the old
jitterbuffer, without breaking the legacy api. This should also make
it easier to do follow-up refactorings to trim down the class further,
and ultimately remove it.
Bug: webrtc:7408
Change-Id: Iec8a167fe5d0425114b0b67a5b4c2fd5fc4fa150
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151910
Commit-Queue: 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@{#29108}
The |frames_dropped| statistics contain not only frames that are dropped
but also frames that are in internal queues. This CL changes that so
that |frames_dropped| only contains frames that are dropped.
Bug: chromium:990317
Change-Id: If222568501b277a75bc514661c4f8f861b56aaed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150111
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28968}
It's called from VideoReceiver::Process, which we no longer use.
Bug: webrtc:7408
Change-Id: I2ce5b7a07437e110d20d04aa159dddf245504abe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133189
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27657}
It configures features of the old jitter buffer which we don't use.
Bug: webrtc:7408
Change-Id: I0f0e9c20b76fb5589f12bc696022d7d2ffef0ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133168
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27656}
VCMReceiveStatisticsCallback originates in the old jitter buffer, and
is no longer used.
VCMFrameTypeCallback originates in VideoReceiver::RequestKeyFrame,
which is called from OncomingPacket, Process, Decode(uint16_t
maxWaitTimeMs), all of which are unused by VideoReceiveStream.
So delete the code to wire them up via VideoStreamDecoder.
Bug: webrtc:7408
Change-Id: I173bc94eb32f2641f943c125083db038c3bcaeb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128870
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27277}
Only interesting call deleted in cl
https://codereview.webrtc.org/2704183002.
Move call to QualitySample (used for bad call detection) to
OnRenderedFrame
Bug: webrtc:7408
Change-Id: I0e9ae2ed62fe19a282377cb840e38bd2aae8f3e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128768
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27243}
It's routed to the old jitter buffer, which is instantiated but
unused.
Bug: webrtc:7408
Change-Id: I8fe2d8a2bfa6c2119945cc294d17982af4f830cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127541
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27100}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
- 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}
This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend.
Bug: webrtc:9517
Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7
Reviewed-on: https://webrtc-review.googlesource.com/88241
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23936}
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 class receives data about video frames from ReceiveStatisticsProxy,
calculates spatial and temporal quality metrics and outputs them to UMA
stats. It is all done in a separate class because it will be further
extended to calculate aggregated quality metrics in the future.
Bug: webrtc:9295
Change-Id: Ie36db83e10c0e8da0b9baa392651cb9a67a54a80
Reviewed-on: https://webrtc-review.googlesource.com/78220
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23387}
Both of these implementations are added as observers by
VideoReceiveStream, which is also an observer.
Instead, VideoReceiveStream now forward the necessary information
when the stats are updated as is already done for |frame_buffer_|.
Fewer observers registered, simplifies the code a little bit,
reduces synchronization when registering/unregistering observers
and reduces the set of existing implementations.
(I'm changing call_stats.* a little bit, so this is a by-the-way change)
Bug: none
Change-Id: I2d40b80a60e5ca632293c5118ac3b1a35d48ea40
Reviewed-on: https://webrtc-review.googlesource.com/64524
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22594}
The effect of this is one less lock being grabbed when updating rtt stats.
(Using NoTry=True since the mac_asan and mac_rel bots are acting up)
Bug: None
Change-Id: I7dac4031573aa54550068a7039e5c2d3054cb50b
NoTry: true
Reviewed-on: https://webrtc-review.googlesource.com/64522
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22593}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
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}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/video/video_stream_decoder.cc (Browse further)
