Adding configuration of new GoogCcNetworkController to initializer, this
makes sure that it is properly initialized from the start. To achieve
this SendSideCongestionController waits until it has received the
necessary information to construct the object. This information should
be provided in the constructor for SendSideCongestionController in the
future.
Bug: webrtc:8415
Change-Id: Icc09b8b246bae9f9704b80855fc4caa3450b34fc
Reviewed-on: https://webrtc-review.googlesource.com/58099
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22404}
This adds the ability to the pacer to apply a congestion window by
tracking sent data. This makes it more reliable when the congestion
window is small enough to be filled at a high rate as there are less
thread context switches that might affect the timing and performance.
Outstanding data is not reduced by the pacer as it has no information
about acknowledged packet feedback. This is by design as the pacer would
also need to keep track of on which connection packets were sent or
received, requiring a larger, more complex, change to the pacer.
Bug: webrtc:8415
Change-Id: I4ecd303e835552ced042cd21186da910288a8258
Reviewed-on: https://webrtc-review.googlesource.com/51764
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22371}
This ensures that SendSideCongestionController is always initialized
with starting bitrate and bitrate limits.
Bug: webrtc:8415
Change-Id: If3b75e935dda755f9e0f40af1021f97ff150c9e9
Reviewed-on: https://webrtc-review.googlesource.com/59224
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22370}
The GetTransportFeedbackVector() method is only used in tests, and they
can access the class directly anyway. Keeping it is adding code bloat
and is also making upcoming refactoring more difficult.
Bug: webrtc:8975
Change-Id: I8323addb3c1461dd73b30353c8d9fe9410471c15
Reviewed-on: https://webrtc-review.googlesource.com/60860
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22349}
SSCC was accessing the pacer just to report values back to
RtpTransportControllerSend which already owns the pacer.
This CL moves those access methods.
To make RtpTransportControllerSend simpler, Call is made
responsible to keep track of network status used only as a
condition for report the pacer queuing delay.
Bug: webrtc:8415
Change-Id: I306bc9fcd3d8dcc7a637d51f2629ececebd48cad
Reviewed-on: https://webrtc-review.googlesource.com/60483
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22331}
Small changes to the unit tests for send side congestion controller.
Mostly adding some extra waits and expectation verifications. This
prepares for an upcoming CL.
Bug: webrtc:8415
Change-Id: Id3086a485eda99732d01192cac9a91141158ab45
Reviewed-on: https://webrtc-review.googlesource.com/59223
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22270}
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}
This is a reland of e6cefdf9c572cdce55ff0497ad6e516c76132ee8.
Original change's description:
> Moved congestion controller to goog_cc folder.
>
> Bug: webrtc:8415
> Change-Id: I2070da0cacf1dbfc4b6a89285af3e68fd03497ab
> Reviewed-on: https://webrtc-review.googlesource.com/43841
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21928}
Bug: webrtc:8415
Change-Id: Ib5cf8641466655d64ac80f720561817f4cab49a9
Reviewed-on: https://webrtc-review.googlesource.com/53062
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22244}
The functionality in network_control is by design not rtp specific and
should live outside the rtp directory in congestion_controller.
Bug: webrtc:8415
Change-Id: I68778f1a630c8e93950040bc56ee864c201b5990
Reviewed-on: https://webrtc-review.googlesource.com/55383
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22240}
Makes the new task queue based congestion controller implement the
SendSideCongestionControllerInterface.
Bug: webrtc:8415
Change-Id: I3dfe11c2eb200bc8d85c83edf78d7fdd0129bbff
Reviewed-on: https://webrtc-review.googlesource.com/56781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22235}
This CL activates unit tests for the task queue based send side
congestion controller that will replace the current one in the future.
To be able to have the tests run side by side with the tests of the old
congestion controller, the old tests have been prefixed with "Legacy".
This CL also contains some minor fixes to the new congestion controller
code.
Bug: webrtc:8415
Change-Id: I5e7474d42f17fcbfef402e26f638846fa3424695
Reviewed-on: https://webrtc-review.googlesource.com/55381
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22229}
This makes it easier to have the new and the old send side congestion
controller side by side. This namespace is only temporary. As soon the
new task queue based congestion controller is fully functional, the old
will be deprecated and removed together with the temporary namespace.
Bug: webrtc:8415
Change-Id: Ie817511345c91cab2ebca68f038075875c7e6529
Reviewed-on: https://webrtc-review.googlesource.com/56720
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22221}
To be able to safely move over to the new code, the revised
code is added alongside the old code. Most of the files added
in this CL are more or less direct copies of the previous code.
This new version of send side congestion controller will be
activated under a field trial in a followup CL.
Bug: webrtc:8415
Change-Id: I034e583cf891a8f6357119739a1517cc0a4abe88
Reviewed-on: https://webrtc-review.googlesource.com/53322
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22212}
