Packet-loss computation and plot were added to BweReceiver class.
Objective function and plot were added to PacketReceiver class.
BUG=4550
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56459004
Cr-Commit-Position: refs/heads/master@{#9391}
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 pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
The implementation of this proposal is in progress.
More unittest will be added.
Sender side is being implemented.
Some constants need to be tuned.
BUG=4550
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43299004
Cr-Commit-Position: refs/heads/master@{#9146}
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}
Note that the timeout should depend on the smoothed RTT, but for now is hard coded to 1000 ms.
This solves issues where a full cwnd gets lost.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51739004
Cr-Commit-Position: refs/heads/master@{#9051}
The problem was that only ACKed packets were subtracted from in_flight_, but lost packets were never removed, which caused TCP to stop sending eventually.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43239004
Cr-Commit-Position: refs/heads/master@{#9041}
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
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 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
Changes differing from https://webrtc-codereview.appspot.com/37859004:
* I put the include_tests==1 stuff of audio_coding.gypi in its
own audio_coding_tests.gypi file, including the Android and isolate
targets which were incorrectly located in the previous CL
* I moved the bwe utilities in remote_bitrate_estimator.gypi
into include_tests==1 since they depend on test.gyp after I
cleaned up the duplicated inclusion of rtp_file_reader.cc
R=stefan@webrtc.orgTBR=tina.legrand@webrtc.org
TESTED=Passing gyp and compile using:
webrtc/build/gyp_webrtc -Dinclude_tests=1
webrtc/build/gyp_webrtc -Dinclude_tests=0
I also setup a Chromium checkout with my checkout mounted in
third_party/webrtc and ran build/gyp_chromium successfully.
BUG=4185
Review URL: https://webrtc-codereview.appspot.com/33159004
Cr-Commit-Position: refs/heads/master@{#8205}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
This required to move the AGC tools source files
into webrtc/tools and create a new agc_test_utils target.
Since audio_codec_speed_tests.gypi referenced sources above,
the best approach I could come up with was to add an audio_coding.gypi
file at a higher level and move the targets in there (+ the includes from
modules.gyp which is an improvement IMO).
I also added a PRESUBMIT.py check to prevent new source
entries being added with <(webrtc_root) in the path.
BUG=4185
R=andrew@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37859004
Cr-Commit-Position: refs/heads/master@{#8197}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
