The intended signalling from StartTimer() to Process() is that
created_at_.tv_sec is set to 0, and timer_event_->Set() is then called
in order to wake the process thread from timer_event_->Wait(). When this
happens the process thread will return early and the run Process()
again. This time it will pick up created_at_.tv_sec = 0 and run a new
Wait() call with the desired end time.
However if the process thread was NOT blocking on timer_event_->Wait()
when timer_event_->Set() was called from StartTimer() it will mean that
the first call to timer_event_->Wait() from Process(), AFTER the new
time has been configured (count_ = 1), will return early.
If the timer is not periodic it means that Set() will never be called,
and any calls will Wait() will block until the time out.
The solution is to always reset the event in timer_event_ on the first
call to timerEvent_->Wait(), after a timer has started.
Also some general cleanup.
BUG=
Review URL: https://codereview.webrtc.org/1812533002
Cr-Commit-Position: refs/heads/master@{#12082}
Removes code duplication and use of the dangerous public destructor in
RefCountImpl.
Also making wider use of scoped_refptr and fixing various leaks in the
process.
BUG=webrtc:5229
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1477013005 .
Cr-Commit-Position: refs/heads/master@{#12075}
Reason for revert:
The openmax_dl include change breaks downstream projects.
Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623
Review URL: https://codereview.webrtc.org/1808573002
Cr-Commit-Position: refs/heads/master@{#12009}
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1796413002 .
Cr-Commit-Position: refs/heads/master@{#12008}
"WebRTC.Video.AVSyncOffsetInMs"
The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed.
Updated sync tests in call_perf_tests.cc to use this implementation.
BUG=webrtc:5493
Review URL: https://codereview.webrtc.org/1756193005
Cr-Commit-Position: refs/heads/master@{#11993}
Sparse macro is replaced and new implementation in metrics.h is used.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1564923008
Cr-Commit-Position: refs/heads/master@{#11483}
Real time clock may cause problems as they can move (even backwards) if
the clock is changed, eg updated by NTP.
Non-monotonic clocks still in use on some platform (I'm looking at you,
Apple) for timed waits, but that should be less of an issue than actual
timestamps.
BUG=webrtc:5452
Review URL: https://codereview.webrtc.org/1613013002
Cr-Commit-Position: refs/heads/master@{#11375}
While transitioning over to rtc::CriticalSection completely, this gives perf benefits that rtc::CriticalSection has on Mac to current users of CriticalSectionWrapper.
BUG=
Review URL: https://codereview.webrtc.org/1614373002
Cr-Commit-Position: refs/heads/master@{#11356}
There's no need for this class to have a vtable since there exists only a single implementation (per platform). It's also not good for performance.
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1601743004 .
Cr-Commit-Position: refs/heads/master@{#11306}
Since this is on Windows only, I'm also using the CriticalSectionWrapper and ConditionVariableWrapper Windows types directly which allows us to skip 3 extra heap allocations. It also helps with the removal of the 'friend' relationship between ConditionVariableWrapper and CriticalSectionWrapper, which is causing headaches on Mac.
BUG=
Review URL: https://codereview.webrtc.org/1595983002
Cr-Commit-Position: refs/heads/master@{#11300}
This removes dependency on ConditionVariableWrapper and CriticalSectionWrapper which currently have a 'friend' relationship that I'd like to get rid of.
BUG=
Review URL: https://codereview.webrtc.org/1590983005
Cr-Commit-Position: refs/heads/master@{#11292}
There's no reason not to use std::move instead now that we can use the
C++11 standard library.
BUG=webrtc:5373
Review URL: https://codereview.webrtc.org/1531013003
Cr-Commit-Position: refs/heads/master@{#11225}
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
Update test implementation (test/histograms.h) to be more similar a real implementation (where histogram get functions return a Histogram pointer). Add check that the name of a histogram does not change.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1528403003
Cr-Commit-Position: refs/heads/master@{#11161}
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}
Removes the global simulated time that affects (or breaks) following
tests in the same binary and replaces it with SimulatedClock.
BUG=webrtc:5318
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512853002 .
Cr-Commit-Position: refs/heads/master@{#10947}
Reason for revert:
Broke downstream compile step, possibly relandable when using a MSVC version that has constexpr, other than that I'm out of ideas.
.../webrtc/base/atomicops.h:71:8: note: no known conversion for argument 1 from '<brace-enclosed initializer list>' to 'const rtc::AtomicInt&'
Original issue's description:
> Reland of "Create rtc::AtomicInt POD struct."
>
> Relands https://codereview.webrtc.org/1420043008/ with brace initializers
> instead of constructors hoping that they won't introduce static
> initializers.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/84f0970d100e67a1dc4fe9a1b16b7d293302044e
> Cr-Commit-Position: refs/heads/master@{#10920}
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1505053002
Cr-Commit-Position: refs/heads/master@{#10922}
Seconds and fractions parts of the ntp time presented with two values, but used as one.
This helper structure can make that use more clear.
(initially introduced into rtp_rtcp as https://codereview.webrtc.org/1435833003)
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1482593002
Cr-Commit-Position: refs/heads/master@{#10814}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
Reason for revert:
Potentially breaks a threading test under DrMemory. Rolling back while I investigate.
Original issue's description:
> Several Tick counter improvements.
>
> Move logic into cc file
> Simplify interval calculation
> Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
> Remove double divide on each ::Now() invocation on mac
>
> Move TickTime and TickInterval funcitons to cc file in prep for refactoring.
>
> BUG=
> R=mflodman@webrtc.org, pbos@webrtc.org
>
> Committed: https://crrev.com/4c27e4b62da2047063d88eedfeec3e939fea7843
> Cr-Commit-Position: refs/heads/master@{#10661}
TBR=pbos@webrtc.org,mflodman@webrtc.org,noahric@chromium.org,thaloun@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1450203002
Cr-Commit-Position: refs/heads/master@{#10663}
Move logic into cc file
Simplify interval calculation
Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
Remove double divide on each ::Now() invocation on mac
Move TickTime and TickInterval funcitons to cc file in prep for refactoring.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1415923010 .
Cr-Commit-Position: refs/heads/master@{#10661}
This is the second revert. The first attempt in https://codereview.webrtc.org/1423693008/
was missing a subtle curly brace caused by a merge conflict.
I'm going to let this one go through the CQ.
Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/1412963007
Cr-Commit-Position: refs/heads/master@{#10532}
Reason for revert:
Oh dear, this broke compilation.
I guess more was built on top of this CL before I reverted it.
Reverting now for futher investigation (and re-land using CQ)
Original issue's description:
> Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
>
> Reason for revert:
> This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
> I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
>
> See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
>
> Original issue's description:
> > Add aecdump support to audioproc_f.
> >
> > Add a new interface to abstract away file operations. This CL temporarily
> > removes support for dumping the output of reverse streams. It will be easy to
> > restore in the new framework, although we may decide to only allow it with
> > the aecdump format.
> >
> > We also now require the user to specify the output format, rather than
> > defaulting to the input format.
> >
> > TEST=Bit-exact output to the previous audioproc_f version using an input wav
> > file, and to the legacy audioproc using an aecdump file.
> >
> > Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> > Cr-Commit-Position: refs/heads/master@{#10460}
>
> TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/d279941bb54bfdc6e7324bf36cac76581474b96d
> Cr-Commit-Position: refs/heads/master@{#10523}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1419953010
Cr-Commit-Position: refs/heads/master@{#10524}
Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}
TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1423693008
Cr-Commit-Position: refs/heads/master@{#10523}
field_trial_default and metrics_default don't use system_wrappers and
don't need to depend on it. The dependency on field_trial_default was
added in libjingle in crrev.com/356135 and that broke compilation of
libjingle for NaCl with GN because system_wrappers currently doesn't
compile for NaCl.
TBR=niklas.enbom@webrtc.org
Review URL: https://codereview.webrtc.org/1412003007
Cr-Commit-Position: refs/heads/master@{#10495}
