This implementation will be replaced by a faster one and sparse will be removed.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1530913002
Cr-Commit-Position: refs/heads/master@{#11099}
Before this change, UpdateEstimate would repeatedly decrease bitrate
even though there's no fresh corresponding RTCP loss report, triggering
multiple reactions to a single indication of high packet loss.
BUG=webrtc:5101
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1417723005
Cr-Commit-Position: refs/heads/master@{#10374}
This removes the TRFC rate control which does not introduce any help in the
computation of the sending rate.
BUG=5083
Review URL: https://codereview.webrtc.org/1383813003
Cr-Commit-Position: refs/heads/master@{#10299}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
Modify bitrate controller to update bitrate based on process call and not
only whenever a RTCP receiver block is received.
Additionally:
Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
Did not touch decrease logic, however since it can be triggered more often it
may decrease much faster and closer to the original written cap of once every
300ms + rtt.
Note:
rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5794 4adac7df-926f-26a2-2b94-8c16560cd09d
This triggered an occasional TSAN failure in
CallTest.ReceivesPliAndRecoversWithNack e.g.:
http://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan/builds/1444/steps/memory%20test%3A%20video_engine_tests/logs/stdio
I managed to reproduce this locally and verified that reverting this CL
corrected it.
> Modify bitrate controller to update bitrate based on process call and not
> only whenever a RTCP receiver block is received.
>
> Additionally:
> Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
>
> Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
>
> Did not touch decrease logic, however since it can be triggered more often it
> may decrease much faster and closer to the original written cap of once every
> 300ms + rtt.
>
> Note:
> rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
> bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
>
> BUG=3065
> R=stefan@webrtc.org, mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/10529004TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5785 4adac7df-926f-26a2-2b94-8c16560cd09d
only whenever a RTCP receiver block is received.
Additionally:
Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
Did not touch decrease logic, however since it can be triggered more often it
may decrease much faster and closer to the original written cap of once every
300ms + rtt.
Note:
rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5775 4adac7df-926f-26a2-2b94-8c16560cd09d
- Move condition of 0 bps as max meaning 1gbps from SendSideBandwidthEstimation to BitrateController.
- Remove condition on bitrate=0 meaning bandwidth estimation off as that could only happen when no observers existed
and in which case the estimation would be ignored.
- Add MaybeTriggerOnNetworkChanged which only runs rate allocation if any of the dependent variables has changed
thus allowing to remove many of the bool returns that try to indicate if the estimation has changed which would not
be aware if the observers have changed.
- SendSideBandwidthEstimation now has a UpdateBitrate and has clear code paths to which calls update bitrate.
- Changes in enforce_min_bitrate so the 10kbps min is set from the BitrateController and not from the outside this keep valid as observers are changed.
R=henrik.lundin@webrtc.org, stefan@webrtc.org
BUG=3065
Review URL: https://webrtc-codereview.appspot.com/10189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5752 4adac7df-926f-26a2-2b94-8c16560cd09d
The new test is based upon the exisiting rampup test, but also adds
a low-rate period. The main purpose of the test is to verify the
SuspendBelowMinBitrate functionality, which must be enabled for the
test to pass.
The CL also adds a change to the min bitrate in the send-side bandwidth
estimator when SuspendBelowMinBitrate is enabled.
An anonymous namespace is added around the StreamObserver classes
in the test to avoid silent linker conflicts that could happen
otherwise.
Note: this CL depends on https://webrtc-codereview.appspot.com/9049004/
BUG=2636
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5646 4adac7df-926f-26a2-2b94-8c16560cd09d
