Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.orgTBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
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}
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1441673002
Cr-Commit-Position: refs/heads/master@{#10630}
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1405023016
Cr-Commit-Position: refs/heads/master@{#10594}
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}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.
BUG=webrtc:4836
Review URL: https://codereview.webrtc.org/1368943002
Cr-Commit-Position: refs/heads/master@{#10087}
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}
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
Depends on https://codereview.webrtc.org/1345433002/
BUG=chromium:468375
(in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1342543004
Cr-Commit-Position: refs/heads/master@{#9954}
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}
When using send-side bandwidth estimation, the inter-packet delay is
reported back to the sender using RTCP TransportFeedback messages.
Theis data needs to be translated into a format which the bandwidth
estimator (now instantiated on the send side) can use, including looking
up the local absolute send time from the send time history.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1329083005
Cr-Commit-Position: refs/heads/master@{#9929}
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
BUG=chromium:468375 (in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1316363005
Cr-Commit-Position: refs/heads/master@{#9913}
For use when send-side bandwidth estimation is enabled.
Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1290813008
Cr-Commit-Position: refs/heads/master@{#9898}
This will be used for the send side bitrate estimation. Storing various
meta-data about packets that can be retreived when arrival time feeback
arrives.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1288033008
Cr-Commit-Position: refs/heads/master@{#9859}
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
PacketSender can now log Pause/Resume events into a MetricRecorder. Solved estimate error and optimal bitrate issue for test 5.7 (multiple short TCP flows).
Added Sending Estimate logging and plotting.
Fixed plotting issue on plot_dynamics.py
Now lines with the same color (in different boxes) correspond to the same flow.
Adjusting plot_dynamics.py font size according to number of variables.
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1270543002 .
Cr-Commit-Position: refs/heads/master@{#9664}
Computing all metrics using constant extra memory.
PlotHistogram methods are executed in constant time.
-- Previously throughput and delay were using O(num_packets) extra memory and their associated PlotHistograms took linear time complexity.
Added MetricRecorder unittests
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1257683006 .
Cr-Commit-Position: refs/heads/master@{#9658}
These small packets are common for H.264 where the first packet of an IDR
contains the parameter sets.
BUG=4806
Review URL: https://codereview.webrtc.org/1221943002
Cr-Commit-Position: refs/heads/master@{#9639}
-- 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}
Supports:
- bwe_plot.sh
---- Histogram plots
---- Baseline histograms
---- Vertical error lines
---- Dashed horizontal lines
---- Different colors for different algorithms
---- Legends read as input
---- Extra horizontal space in case there are TCP flows plotted
---- Multiple windows
---- Auto vertical scale, except for latency plots which are kept constant for all plots
- plot_dynamics.py
---- Dynamic plots
---- Different colors for different flows and algorithms
---- Dashed line plot for available capacity
---- Throughput, latency and loss on separated boxes
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1237903003 .
Cr-Commit-Position: refs/heads/master@{#9614}
Using a right-sided (absolute value), truncated gaussian distribution originally with zero mean.
Currently truncated at x = 3 * std_dev.
Added expected value computation.
Modified jitter unittests accordingly.
BUG=webrtc:4848
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1237303002 .
Cr-Commit-Position: refs/heads/master@{#9587}
---- 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}
