Removing functions that has been removed from
RtpTransportControllerSendInterface from
MockRtpTransportControllerSend.
Deleted functions: GetPacerModule, GetModule,
SetTransportOverhead and AvailableBandwidth.
Bug: webrtc:8415
Change-Id: I24d460bd18d57966e3b333ce0c234c3e3dc19a9a
Reviewed-on: https://webrtc-review.googlesource.com/62762
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22489}
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}
Added dependencies used by MockRtpTransportControllerSend to its header
file. The mocked interface can't be used properly without those.
This prepares for later CLs utilizing the mock.
Bug: None
Change-Id: I5f8ca04032ad09810240f7c034cc628d700dbedb
Reviewed-on: https://webrtc-review.googlesource.com/56181
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22154}
Moving usage of bitrate configuration related interface from Call
interface to the corresponding methods in the RtpSendTransportController
interface.
SetBitrateConfig was replaced with SetSdpBitrateParameters
SetBitrateConfigMask was replaced with SetClientBitratePreferences
OnNetworkRouteChanged was replaced with OnNetworkRouteChanged
This makes it more clear that RtpSendTransportController owns bitrate
configuration and fits a longer term ambition to reduce the scope of
the Call class.
Bug: webrtc:8415
Change-Id: I6d04eaad22a54ecd5ed60096e01689b0c67e9c65
Reviewed-on: https://webrtc-review.googlesource.com/54365
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@{#22131}
