Theil and Sen's estimator essentially looks at the line through every pair of points and selects the median slope. This is robust to corruption of up to 29% of the data points.
Wire up new estimator to field trial experiment. Add unit and integration tests. Results are promising.
BUG=webrtc:6728
Review-Url: https://codereview.webrtc.org/2512693002
Cr-Commit-Position: refs/heads/master@{#15508}
These functions are identical. BoringSSL added these APIs, then OpenSSL
1.1.0 added similar ones but with slightly longer names. We're
standardizing on the OpenSSL names to avoid API skew.
BUG=none
Review-Url: https://codereview.webrtc.org/2550423004
Cr-Commit-Position: refs/heads/master@{#15455}
When implementing ISOLATED_OUTDIR feature in WebRTC, I found two issues,
1. pathutils and flags are not accessible in testsupport. But both of them are
useful for the feature. Pathname can help to combine path with filename, while
a flag is needed to handle command line parameter.
2. rtc::File cannot accept an rtc::Pathname, which is a little bit inconvenient.
After investigating bug webrtc:3806, flags, pathutils and urlencode are
removed from rtc_base_approved because of the including of common.h. So I
replaced common.h with checks.h, and ASSERT with RTC_DCHECK. flags,
pathutils and urlencode pairs now can be placed into rtc_base_approved to
unblock file.h to include pathutils.h.
Please kindly let me know if you have other concerns about this change.
BUG=webrtc:3806, webrtc:6732
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng;master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:linux_android_rel_ng
Review-Url: https://codereview.webrtc.org/2533213005
Cr-Commit-Position: refs/heads/master@{#15451}
It was incorrectly using a older version of the macro, which
wasn't discovered since the code wasn't built in WebRTC until now.
I moved webrtc/base/sigslottester.h from rtc_unittests into
rtc_base_test_utils instead to make it more usable.
BUG=webrtc:6821
Review-Url: https://codereview.webrtc.org/2551813002
Cr-Commit-Position: refs/heads/master@{#15422}
The actual time used in production code should honor the epoch time.
BUG=webrtc:6737
Review-Url: https://codereview.webrtc.org/2526433002
Cr-Commit-Position: refs/heads/master@{#15282}
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.
It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
Reason for revert:
Breaks downstream projects:
error: undefined reference to 'rtc::ExpFilter::kValueUndefined'
error: undefined reference to 'rtc::ExpFilter::Apply(float, float)'
error: undefined reference to 'rtc::ExpFilter::Reset(float)'
rror: undefined reference to 'rtc::ExpFilter::UpdateBase(float)'
Original issue's description:
> Move smoothing filter to common audio.
>
> This will make the smoothing filter a basic tool that is going to be used by both voice engine and ANA.
>
> BUG=webrtc:6443
>
> Committed: https://crrev.com/a82395bf7cd15b7396456df06fe952ede8db0c39
> Cr-Commit-Position: refs/heads/master@{#15146}
TBR=minyue@webrtc.org,solenberg@webrtc.org,perkj@webrtc.org,tommi@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2510373002
Cr-Commit-Position: refs/heads/master@{#15147}
This will make the smoothing filter a basic tool that is going to be used by both voice engine and ANA.
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2484153002
Cr-Commit-Position: refs/heads/master@{#15146}
When the selected connection becomes not receiving and there are many connections,
If we use a round-robin fashion to ping all connections, none of the connections will
be in receiving state for sufficient long time to ensure switching connections promptly.
Triggered check will help in this situation to some extent but it may still fail to switch promptly when there are a lot of connections.
With this CL, if the selected connection is weak, once we find a writable connection on a network we start to ping it with a higher priority to keep it in receiving state.
Plus, if the selected connection is weak, we choose a shorter ping interval (900ms) for all writable connections.
BUG=b/32022719
Review-Url: https://codereview.webrtc.org/2369963004
Cr-Commit-Position: refs/heads/master@{#14991}
This is intended to make SequencedTaskChecker work for native dispatch queues
on iOS and macOS. These labels can be compared by their pointers to determine
if a task is running on the same queue.
BUG=webrtc:6643
Review-Url: https://codereview.webrtc.org/2464383002
Cr-Commit-Position: refs/heads/master@{#14900}
With this change, instead of
RTC_DCHECK_GE(unsigned_var, 17u);
we can simply write
RTC_DCHECK_GE(unsigned_var, 17);
or even
RTC_DCHECK_GE(unsigned_var, -17); // Always true.
and the mathematically sensible thing will happen.
Perhaps more importantly, we can replace checks like
// index is size_t, num_channels is int.
RTC_DCHECK(num_channels >= 0
&& index < static_cast<size_t>(num_channels));
or, even worse, just
// Surely num_channels isn't negative. That would be absurd!
RTC_DCHECK_LT(index, static_cast<size_t>(num_channels));
with simply
RTC_DCHECK_LT(index, num_channels);
In short, you no longer have to keep track of the signedness of the arguments, because the sensible thing will happen.
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2459793002
Cr-Commit-Position: refs/heads/master@{#14878}
The requirements have been relaxed a little bit which should allow the test to pass on our VMs that run the tests.
BUG=6610
Review-Url: https://codereview.webrtc.org/2458713005
Cr-Commit-Position: refs/heads/master@{#14836}
Introduce a convention on categorizing GN targets:
1. Production code
2. Tests
3. Examples
4. Tools
The first two have targets spread out all over the tree,
while the latter are isolated to examples/ and tools/ directories.
Another new convention: Each directory's BUILD.gn file shall contain
a target named similar to the directory name. This target shall
contain the 'most common' production code, i.e. so that most
consumers of the directory can depend on only the directory
(which implicitly means that target in GN).
//webrtc:webrtc_tests is changed to depend on all WebRTC tests.
From now on, it's necessary to add new test targets to this dependency
tree in order to get them compiled.
Two new group targets are created:
//webrtc/modules/audio_coding:audio_coding_tests
//webrtc/modules/audio_processing:audio_processing_tests
to reduce the long list of tests in //webrtc:webrtc_tests.
Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted
to the root target, to avoid circular dependencies due to the monolithic
property of these targets (a problem we've had in the past).
The 'root' target at the top level is renamed to 'default', which means GN will
build this target instead of _all_ generated targets
(see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default).
This target now depends on everything we want to build, meaning all targets now
explicitly needs to be wired up from the root target in order to get build.
Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build.
BUG=webrtc:6440
TESTED=git cl try --clobber
NOTRY=True
Review-Url: https://codereview.webrtc.org/2441383002
Cr-Commit-Position: refs/heads/master@{#14821}
The loopback range is 127.0.0.0/8, which is everything from 127.0.0.0 to
127.255.255.255.
BUG=chromium:649118
Review-Url: https://codereview.webrtc.org/2445933003
Cr-Commit-Position: refs/heads/master@{#14807}
Before this change, with DCHECKs switched off, this sort of check
size_t index = ...;
RTC_DCHECK_GE(index, 0u);
would cause GCC (but no other compiler that we use) to complain
that unsigned values are always greater than or equal to 0. With
this change, it no longer complains.
(It was and remains the case that there was no complaints if
DCHECKs were switched on, or if you used RTC_CHECK_op.)
The reason for doing this change is that it isn't useful for the
compiler to force us to remove DCHECKs just because their
condition can be verified statically. That causes us to remove
the checks, and once that's happened, future code changes are free
to violate the removed checks and no one will know...
BUG=webrtc:6620
Review-Url: https://codereview.webrtc.org/2455943002
Cr-Commit-Position: refs/heads/master@{#14805}
The LOGGING define is only used in a single location in our whole codebase:
$ git gs "f LOGGING"
webrtc/base/physicalsocketserver.cc:1584:#if LOGGING
$ git gs "defined(LOGGING"
(no hits)
The above commands also give no hits in Chromium's code base.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2442743002
Cr-Commit-Position: refs/heads/master@{#14799}
It turns out that that audio network adaptor can always be created with knowledge of receiver frame length range. There is no need to keep some infrastructure that is used for runtime setting of receiver frame length ranges.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2429503002
Cr-Commit-Position: refs/heads/master@{#14748}
