The purpose is to make the fixture reusable in downstream
projects. The CL adds the following things to API:
- api/test/video_quality_test_fixture.h
- api/test/create_video_quality_test_fixture.h
The following things are moved to API:
- call/bitrate_constraints.h (api/bitrate_constraints.h)
- call/simulated_network.h (api/test/simulated_network.h)
- call/media_type.h (api/mediatypes.h)
These are required by the params struct passed to the
fixture. I didn't attempt to split the params struct into
an internal-only and public version in this CL, and as
a result we need to pull in the above things. They are
quite harmless though, so I think it's worth it in order
to avoid splitting up the test config struct.
This CL doesn't solve all the problems we need to
implement downstream tests; we probably need to upstream
tracing variants of FakeNetworkPipe for instance, but
that will come later. This puts in place the basic
structure for now.
Bug: None
Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911
Reviewed-on: https://webrtc-review.googlesource.com/69601
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23714}
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}
Replaces both BitrateConstraintsMask and
PeerConnectionInterface::BitrateParameters. The latter is kept
temporarily for backwards compatibility.
Bug: None
Change-Id: Ibe1d043f2a76e56ff67809774e9c0f5e0ec9e00f
Reviewed-on: https://webrtc-review.googlesource.com/74020
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23148}
Moving ownership of worker task queue in Call to
RtpTransportControllerSend. This CL also ensures that the task queue
is not destroyed until the process thread running
SendSideCongestionController is stopped.
The worker queue should be owned by RtpTransportControllerSend since
it is mainly used for rtp and transport related tasks such as bitrate
allocation and signaling network state.
Bug: webrtc:9232
Change-Id: I211edf1a3b9f9b2572875d5584cb788cb2449ef9
Reviewed-on: https://webrtc-review.googlesource.com/63023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23119}
This CL adds reporting of per packet feedback availability from Call
via RtpTransportControllerSend to SendSideCongestionController.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: I20b3dbb4a027c46476bc2d2bc875374bff05609a
Reviewed-on: https://webrtc-review.googlesource.com/63220
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22566}
Removing the Synchronous call AvailableBandwidth from the
RtpTransportControllerSend interface. The bandwidth estimate is
provided trough a new interface that communicates with a struct
making it easier to add parameters in the future.
This prepares for removing locking behavior in
SendSideCongestionController that exists just to support this feature.
To keep backwards compatibility with the old
SendSideCongestionController, the struct TargetTransferRate
is constructed in RtpTransportControllerSend. This step can be
removed in the future when the old SendSideCongestionController
is deprecated.
Bug: webrtc:8415
Change-Id: I06f64a89848157de412901c989650d1ecf35246b
Reviewed-on: https://webrtc-review.googlesource.com/60800
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22387}
Ownership of the retransmission rate limiter for video is moved
from send side congestion controller to Call. This is to reduce the
interface on the rtp transport controller send.
Bug: webrtc:8415
Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769
Reviewed-on: https://webrtc-review.googlesource.com/58745
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22254}
DeRegisterNetworkObserver is not used, since
RtpTransportControllerSend owns the thread on which
SendSideCongestionController runs it would not be safe to allow it to be
called from outside. Deregistration should be done by destroying
RtpTransportControllerSend.
In the future, the RegisterObserver functions should be removed as well,
in favor of providing the observer in the constructor. This requires
breaking a circular dependency between RtpTransportControllerSend and
Call.
Bug: webrtc:8415
Change-Id: Ifeb4c5d4a41e4d8419994b3146980bdaaf9cd6a9
Reviewed-on: https://webrtc-review.googlesource.com/58098
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22249}
SetTransportOverhead was used by send streams to signal the packet
overhead that they received from Call. However, call receives the value
from OnNetworkRouteChanged in WebRtcVideoChannel and
WebRtcVoiceMediaChannel which is already propagated to
RtpTransportControllerSend. By skipping the round trip, the interface on
the rtp transport controller can be reduced.
Bug: None
Change-Id: I759b1207aab214bbc2b993106f6ff7cc24e177f7
Reviewed-on: https://webrtc-review.googlesource.com/57182
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22226}
Moving the module process thread responsible for running the pacer
and the send side congestion controller to RtpTransportControllerSend
since it already owns the pacer and the congestion controller. They
are also moved to a common thread rather than using two separate
threads.
As part of the move, the remote bitrate estimator has been moved to the
common process thread in the Call class. Previously it was run on the
removed pacer thread.
Bug: webrtc:8415
Change-Id: I4322eef30d8b97b9611f33af7e560703b710d232
Reviewed-on: https://webrtc-review.googlesource.com/55700
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22166}
Since rtp transport controller send owns the congestion controller it
also should own the bitrate configuration logic, this way it can
initialize the send side congestion controller with the bitrate
configuration.
Bug: webrtc:8415
Change-Id: Ifaa16139ca477cb1c80bf4aa24f17652af997553
Reviewed-on: https://webrtc-review.googlesource.com/54303
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22127}
This CL removes direct access to SendSideCongestionController (SSCC) via
the RtpTransportControllerSend interface and replaces all usages with
calls on RtpTransportControllerSend which will in turn calls SSCC. This
prepares for later refactor of RtpTransportControllerSend.
Bug: webrtc:8415
Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5
Reviewed-on: https://webrtc-review.googlesource.com/53860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22044}
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}
