Parse the estimation parameters from the field trial string.
BUG=webrtc:6690
Review-Url: https://codereview.webrtc.org/2489323002
Cr-Commit-Position: refs/heads/master@{#15126}
Rename kFullSendSideEstimator -> kSendSideEstimator and add new class SendSideBweSender (controlled by kSendSideEstimator) that actually uses the send side BWE.
Move the mock to logging/rtc_event_log/mock.
Allow congestion_controller, remote_bitrate_estimator and audio to depend on loggging/rtc_event_log
BUG=webrtc:6526
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2431093003
Cr-Commit-Position: refs/heads/master@{#14772}
but remove the #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE so that it always builds.
BUG=webrtc:6497
Review-Url: https://codereview.webrtc.org/2398123002
Cr-Commit-Position: refs/heads/master@{#14564}
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)
Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.
Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.
BUG=webrtc:5177
Review URL: https://codereview.webrtc.org/1457023002
Cr-Commit-Position: refs/heads/master@{#10965}
Removed Rand(int low, int high) since that function outputs results that are non-random and/or outside the interval if low is negative.
Added new Uniform(uint32_t, uint32_t) function to replace Rand(int low, int high).
Changed various unit tests to use the new functions.
BUG=
Review URL: https://codereview.webrtc.org/1413053002
Cr-Commit-Position: refs/heads/master@{#10541}
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.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
-- Made plot logging from MetricRecorder and from RateCounterFilter/PacketReceiver standard to fit python and shell plotting scripts likewise.
-- RateCounterFilter is initialized with algorithm name.
-- Removed spare commas and duplicated flow ids from RateCounterFilter name.
-- Added optional plot_delay and plot_loss in MetricRecorder.
-- PacketReceiver can plot directly plot delay if there is no metric_recorder_.
-- Added comments to plot scripts.
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1253473004 .
Cr-Commit-Position: refs/heads/master@{#9636}
---- Dynamic receiving rate.
---- Dynamic packet-loss.
---- Dynamic objective function.
---- Dynamic available capacity.
---- Dynamic available capacity per flow.
---- Average delay Histogram with standard deviation or 5th/95th percentiles.
---- Average bitrate Histogram with error bars.
---- Optimal average bitrate dashed line.
---- Average packet-loss Histogram.
---- Total objective function Histogram.
Added media Pause/Resume methods to Video and TcpSender.
Modified LinkedSet: computing GlobalPacketLossRatio even if packet's sequence_number overflows.
Added small randomization to frame send times, modified bwe_test_framework_unittest accordingly.
Taking offset time into account for plotting.
Added nada_unittests.
Added bwe_unittests.
Added a RateCounter to BweReceiver (replaced ReceivingRate)
Added LossAccount.
Fixed NadaBweReceiver issue: using sender_timestamp instead of creation_time.
Fixed memory leaks.
Fixed int division rounding issues.
Supporting plots on bandwidth Estimators:
Logging received packet information on on SubClassesBweReceiver::ReceivePacket
Updating RateCounter, updating packet loss account and relieving LinkedSet when necessary.
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1202253003 .
Cr-Commit-Position: refs/heads/master@{#9585}
This improves self-fairness and competing for resources with TCP flows.
BUG=4711
Review URL: https://codereview.webrtc.org/1151603008
Cr-Commit-Position: refs/heads/master@{#9545}
A modified operation mode was added, holding:
--- Stricter conditions for AcceleratedRampUp.
--- Smoother GradualRateUpdate adjustments.
--- New AcceleratedRampDown update mode.
This mode reduces significantly the delay for bitrates around its minimum bound.
Several NADA unittests and a few simulations were added.
Fixed LinkedSet bug.
Fixed IsNewerSequenceNumber/IsNewerTimestamp bug.
BUG=4550
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54399004
Cr-Commit-Position: refs/heads/master@{#9340}
Add a propagation delay to tests and make the run-time configurable for the fairness tests.
Handle losses in-between feedback messages.
BUG=4549
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49819004
Cr-Commit-Position: refs/heads/master@{#9099}
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes it possible to build more flexible simulations, and makes it easier to implement bi-directional simulations. This also removes support for generating baseline files and comparing against a baseline, which hasn't turned out to be particuarly useful.
BUG=4173
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35069004
Cr-Commit-Position: refs/heads/master@{#8311}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8311 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes it possible to build more flexible simulations, and makes it easier to implement bi-directional simulations. This also removes support for generating baseline files and comparing against a baseline, which hasn't turned out to be particuarly useful.
BUG=4173
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34989004
Cr-Commit-Position: refs/heads/master@{#8297}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8297 4adac7df-926f-26a2-2b94-8c16560cd09d
The new format instantiates the RemoteBitrateEstimator at the send-side and feeds back all packet arrival timestamps and sequence numbers to the sender, where inter-arrival deltas are calculated.
Next step will be to make feedback packets part of regular packets and send them over the network. This also requires bi-directional simulations.
BUG=4173
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37109004
Cr-Commit-Position: refs/heads/master@{#8264}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8264 4adac7df-926f-26a2-2b94-8c16560cd09d
