Commit graph

10 commits

Author SHA1 Message Date
Alex Narest
bcf91808a2 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.

Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
2018-06-27 10:33:40 +00:00
Sebastian Jansson
1d430f727d Interface for packet feedback availability in SSCC
This CL adds a function to the SendSideCongestionController interface
for reporting per packet feedback availability.

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: Ifcb6837bb80c5fcfc1f12f4f93ec38cc2903118f
Reviewed-on: https://webrtc-review.googlesource.com/63205
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22565}
2018-03-22 13:38:49 +00:00
Sebastian Jansson
a03585f611 Removing SetTransportOverhead from SSCC Interface.
This is a follow up on an earlier CL removing the usage of
SetTransportOverhead.

Bug: webrtc:8415
Change-Id: I8d9572c06f3ae1e8cacbe7b9bd57a9b65f371c0e
Reviewed-on: https://webrtc-review.googlesource.com/61502
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22427}
2018-03-14 14:27:35 +00:00
Sebastian Jansson
68ee4653ef Moving SetPacingFactor and allocation limits to SSCC.
This CL adds methods to the SendSideCongestionController (SSCC)
interface for configuring pacing factor and allocation based data rate limits.
This means that old SSCC implement the same interface as the new, task
queue based SSCC. This also allows merging the max total allocated
bit rate into SetAllocatedSendBitrateLimits.

This is done in preparation for an upcoming CL where the SSCC version
is controlled by a field trial.

Bug: webrtc:8415
Change-Id: I4d5446a3bedd5b0c725dbd009fb75815fd661eff
Reviewed-on: https://webrtc-review.googlesource.com/61320
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@{#22408}
2018-03-13 16:58:21 +00:00
Sebastian Jansson
a06e919b9f Removing interface to access pacer via SSCC.
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}
2018-03-07 19:01:50 +00:00
philipel
db4fa4b944 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 3.
Trigger on total bitrate change.

Bug: webrtc:8955
Change-Id: I2373a1b7f139c7ea748a9641593e714d6895c8f6
Reviewed-on: https://webrtc-review.googlesource.com/59323
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22323}
2018-03-07 10:31:35 +00:00
Sebastian Jansson
45087cd23f Moved retransmission rate limiter to Call class.
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}
2018-03-01 17:22:28 +00:00
Sebastian Jansson
0f9d9a9a12 Removed unused DeRegisterNetworkObserver.
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}
2018-03-01 12:52:28 +00:00
Sebastian Jansson
98e0111ea5 Implementing interface with new SSCC
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}
2018-02-28 17:30:29 +00:00
Sebastian Jansson
dfde334be0 Adding SendSideCongestionControllerInterface.
This prepares for a later CL providing two implementations of
SendSideCongestionController.

Bug: webrtc:8415
Change-Id: I890dbe4b88bf609921558e03aac66b42629857c8
Reviewed-on: https://webrtc-review.googlesource.com/56700
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22151}
2018-02-22 14:19:53 +00:00