This is a first step towards it. I plan to refactor modules_unittests before continuing with this.
BUG=webrtc:6626
NOTRY=True
Review-Url: https://codereview.webrtc.org/2626163004
Cr-Commit-Position: refs/heads/master@{#16109}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
Reason for reland:
Added disabling of the BlockProcessor DEATH tests due to false alarms on the trybots.
These errors cannot be reproduced locally and they occur only in the DEATH test code,
which means that the memory leaks do not occur under normal conditions.
Original issue's description:
> Reason for revert:
> Memcheck buildbot detected memory leaks in the death tests. The code looks fine
> to me, but please investigate what causes the error and reland.
> > Original issue's description:
> > Added first layer of the echo canceller 3 functionality.
> >
> > This CL adds the first layer of the echo canceller 3.
> > All of the code is as it should, apart from
> > block_processor.* which only contains placeholder
> > functionality. (Upcoming CLs will add proper
> > functionality into those files.)
> >
> >
> >
> > BUG=webrtc:6018
> >
> > Review-Url: https://codereview.webrtc.org/2584493002
> > Cr-Commit-Position: refs/heads/master@{#15861}
> > Committed:
> 38fd1758e9
>
> TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6018
BUG=webrtc:6018
TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org
Review-Url: https://codereview.webrtc.org/2608233002
Cr-Commit-Position: refs/heads/master@{#15884}
Reason for revert:
Memcheck buildbot detected memory leaks in the death tests. The code looks fine to me, but please investigate what causes the error and reland.
Original issue's description:
> Added first layer of the echo canceller 3 functionality.
>
> This CL adds the first layer of the echo canceller 3.
> All of the code is as it should, apart from
> block_processor.* which only contains placeholder
> functionality. (Upcoming CLs will add proper
> functionality into those files.)
>
>
>
> BUG=webrtc:6018
>
> Review-Url: https://codereview.webrtc.org/2584493002
> Cr-Commit-Position: refs/heads/master@{#15861}
> Committed: 38fd1758e9TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2603293002
Cr-Commit-Position: refs/heads/master@{#15877}
This CL adds the first layer of the echo canceller 3.
All of the code is as it should, apart from
block_processor.* which only contains placeholder
functionality. (Upcoming CLs will add proper
functionality into those files.)
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2584493002
Cr-Commit-Position: refs/heads/master@{#15861}
that the level of the output in the audio processing
module is monitored. This CL adds that.
BUG=webrtc:6181, webrtc:6183, webrtc:6220
Review-Url: https://codereview.webrtc.org/2549143004
Cr-Commit-Position: refs/heads/master@{#15718}
The previous CL that added the ability to add
and artificial nearend signal had an issue with
null pointer access.
This is addressed in this CL.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2573033003
Cr-Commit-Position: refs/heads/master@{#15600}
It will be followed by a number of other CLs that extends this framework.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2567513003
Cr-Commit-Position: refs/heads/master@{#15593}
orders of the ProcessStream and ProcessReverseStream API calls.
This CL adds the ability to specify that call order in a file.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2561843003
Cr-Commit-Position: refs/heads/master@{#15503}
is the one stored in the aecdump.
During AEC development, it is handy to be able to simulate different
doubletalk scenarios. This CL adds the ability to add an
artificial nearend on top of that present in the aecdump, which
allows for the developer to artificially customize the scenario
being tested
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2562593003
Cr-Commit-Position: refs/heads/master@{#15502}
This moves some GN check configurations out of .gn to individual targets.
The now checked target is:
"//webrtc/modules/audio_processing/*",
BUG=webrtc:6828
NOTRY=True
R=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2558813003
Cr-Commit-Position: refs/heads/master@{#15475}
Reason for revert:
The downstream problem is now fixed, and this should be good to land again.
Original issue's description:
> Revert of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #2 id:20001 of https://codereview.webrtc.org/2547593002/ )
>
> Reason for revert:
> Breaks down-stream dependencies.
>
> Original issue's description:
> > APM: Change 3 UMA metrics to fewer but linearly distributed buckets
> >
> > In this change WebRTC.Audio.ApmCaptureInputLevel{Average,Peak} are
> > changed to 64 buckets, while WebRTC.Audio.AgcLevel is changed to 50
> > buckets. All three are changed to have linear spacing between buckets.
> >
> > Also, the metrics are renamed to avoid stats conflicts because of different bucket schemes:
> > - WebRTC.Audio.AgcLevel -> WebRTC.Audio.AgcSetLevel
> > - WebRTC.Audio.ApmCaptureInputLevelAverage -> WebRTC.Audio.ApmCaptureInputLevelAverageRms
> > - WebRTC.Audio.ApmCaptureInputLevelPeakRms -> WebRTC.Audio.ApmCaptureInputLevelPeakRms
> >
> > BUG=webrtc:6622
> > CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
> >
> > Committed: https://crrev.com/49715fe3be17d8579586d5bc954d626126d53415
> > Cr-Commit-Position: refs/heads/master@{#15418}
>
> TBR=peah@webrtc.org,rkaplow@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6622
>
> Committed: https://crrev.com/63407a9b6ae6f3fc096e01d64e46c6d21d86b517
> Cr-Commit-Position: refs/heads/master@{#15420}
TBR=peah@webrtc.org,rkaplow@chromium.org
BUG=webrtc:6622
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2551863003
Cr-Commit-Position: refs/heads/master@{#15442}
flaky, with many false results and with a huge
variance.
This CL addresses that by changing the way the
API call durations are measured, using a warmup
period and a longer interval for computing the
timing estimates.
Furthermore, this CL reduces the number of tests
to compensate for the fact that the tests now are
more expensive, as well as to reduce the number
of regressions further.
BUG=webrtc:6614,webrtc:6685,666725
Review-Url: https://codereview.webrtc.org/2549403002
Cr-Commit-Position: refs/heads/master@{#15440}
The Bazel build format doesn't support having separate
lists of compilation flags for C and C++; it just has a single
copts list for cc_library:
https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts
This makes it hard to convert our GN targets to Bazel when there are
compiler warnings that aren't supported for C (like -Woverloaded-virtual
being added in bugs.webrtc.org/6653).
The solution for this is to move all .c files to their own targets
and remove C++-only compiler flags during conversion.
New targets:
//webrtc/common_audio:common_audio_c
//webrtc/common_audio:common_audio_neon_c
//webrtc/modules/audio_coding:g711_c
//webrtc/modules/audio_coding:g722_c
//webrtc/modules/audio_coding:ilbc_c
//webrtc/modules/audio_coding:isac_c
//webrtc/modules/audio_coding:isac_fix_c
//webrtc/modules/audio_coding:isac_test_util
//webrtc/modules/audio_coding:pcm16b_c
//webrtc/modules/audio_coding:webrtc_opusj_c
//webrtc/modules/audio_device:mac_portaudio
//webrtc/modules/audio_procssing:audio_processing_c
//webrtc/modules/audio_procssing:audio_processing_neon_c
This CL also adds a PRESUBMIT.py check that will throw an error
if targets are mixing .c and .cc files, to preven this from regressing.
BUG=webrtc:6653
NOTRY=True
Review-Url: https://codereview.webrtc.org/2550563003
Cr-Commit-Position: refs/heads/master@{#15433}
This change enables experimentation with the clipping minimum level
parameter in the gain control.
BUG=webrtc:6622
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2543753006
Cr-Commit-Position: refs/heads/master@{#15426}
Reason for revert:
Breaks down-stream dependencies.
Original issue's description:
> APM: Change 3 UMA metrics to fewer but linearly distributed buckets
>
> In this change WebRTC.Audio.ApmCaptureInputLevel{Average,Peak} are
> changed to 64 buckets, while WebRTC.Audio.AgcLevel is changed to 50
> buckets. All three are changed to have linear spacing between buckets.
>
> Also, the metrics are renamed to avoid stats conflicts because of different bucket schemes:
> - WebRTC.Audio.AgcLevel -> WebRTC.Audio.AgcSetLevel
> - WebRTC.Audio.ApmCaptureInputLevelAverage -> WebRTC.Audio.ApmCaptureInputLevelAverageRms
> - WebRTC.Audio.ApmCaptureInputLevelPeakRms -> WebRTC.Audio.ApmCaptureInputLevelPeakRms
>
> BUG=webrtc:6622
> CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
>
> Committed: https://crrev.com/49715fe3be17d8579586d5bc954d626126d53415
> Cr-Commit-Position: refs/heads/master@{#15418}
TBR=peah@webrtc.org,rkaplow@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2548333002
Cr-Commit-Position: refs/heads/master@{#15420}
In this change WebRTC.Audio.ApmCaptureInputLevel{Average,Peak} are
changed to 64 buckets, while WebRTC.Audio.AgcLevel is changed to 50
buckets. All three are changed to have linear spacing between buckets.
Also, the metrics are renamed to avoid stats conflicts because of different bucket schemes:
- WebRTC.Audio.AgcLevel -> WebRTC.Audio.AgcSetLevel
- WebRTC.Audio.ApmCaptureInputLevelAverage -> WebRTC.Audio.ApmCaptureInputLevelAverageRms
- WebRTC.Audio.ApmCaptureInputLevelPeakRms -> WebRTC.Audio.ApmCaptureInputLevelPeakRms
BUG=webrtc:6622
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2547593002
Cr-Commit-Position: refs/heads/master@{#15418}
This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.
Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.
The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.
TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
These tests have been causing a large number of false alerts, which is
tons of work for the perf sheriff. It's infeasible to test every single
permutation, so we must choose carefully. This CL reduces the number
of RESULT lines from the test from ~450 to ~50. I attempted to choose
interesting permutations, but you probably know better what's
interesting...
https://chromeperf.appspot.com/report?
sid=a7193c96f708018848ca07ad9c78ac657cadab3c70b3939c42bd7d70a092d61a
also suggests most of the metrics have enormous standard deviations,
so maybe you could look into how stable the metrics really are
and remove/stabilize the ones that aren't?
BUG=chromium:666725
Review-Url: https://codereview.webrtc.org/2529393006
Cr-Commit-Position: refs/heads/master@{#15329}
This CL adds WebRTC.Audio.ApmCaptureInputLevelAverage and
WebRTC.Audio.ApmCaptureInputLevelPeak. The metrics are updated once
every 10 seconds.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2534473004
Cr-Commit-Position: refs/heads/master@{#15300}
This change rewrites RMSLevel, making it accept an ArrayView as input,
and modify the implementation somewhat. It also makes the class keep
track of the peak RMS in addition to the average RMS over the
measurement period.
New tests are added to cover the new functionality.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2535523002
Cr-Commit-Position: refs/heads/master@{#15294}
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}
This perf tests the echo detector in 3 scenarios: standalone, as part of APM with only the echo detector enabled and as part of a normally configured APM.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2517523003
Cr-Commit-Position: refs/heads/master@{#15224}
The histogram will log a new value every time the AGC changes level_.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2525963002
Cr-Commit-Position: refs/heads/master@{#15222}
the APM parameters to the high-pass filter.
The introduction will be done in three steps:
1) This CL which introduces the new scheme and
changes the code in webrtcvoiceengine.cc to use it.
2) Introduce the scheme into upstream code.
3) Remove the HighPassFilter interface in APM.
BUG=webrtc::6220, webrtc::6296, webrtc::6297, webrtc::6181, webrtc::5298
Review-Url: https://codereview.webrtc.org/2415403002
Cr-Commit-Position: refs/heads/master@{#15197}
This new histogram will log a value every time input clipping is
detected. The value is a boolean, with "true" meaning that the gain
was in fact adjusted in response to the detected clipping, and "false"
meaning that adjustment was not allowed due to kClippedLevelMin.
BUG=webrtc:6622
Review-Url: https://codereview.webrtc.org/2522543006
Cr-Commit-Position: refs/heads/master@{#15196}
Move the resources to //resources and upload them to Google Storage.
BUG=webrtc:6727
Review-Url: https://codereview.webrtc.org/2508943004
Cr-Commit-Position: refs/heads/master@{#15152}
This will ensure that the estimated likelihood starts at a low value and prevents initial spikes.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2503843004
Cr-Commit-Position: refs/heads/master@{#15131}
The red acronym is already in use in the context of audio coding, so it is better to avoid reusing it here because it could be confusing.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2505993002
Cr-Commit-Position: refs/heads/master@{#15121}
This adds two command-line flags to audioproc_f: -red and -red_graph, which can be used to enable/disable the RED, and to set the output path for the graph. The graph is generated as a python script that depends on matplotlib and numpy to display the graph.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2486763002
Cr-Commit-Position: refs/heads/master@{#15069}
Updates GetStats() function in AudioSendStream to use the new GetStatistics function in APM instead of the corresponding VoE functions.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2463813002
Cr-Commit-Position: refs/heads/master@{#15065}
The likelihood is logged as a percentage, with 100 bins in the histogram.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2487243002
Cr-Commit-Position: refs/heads/master@{#15025}
In the new APM statistics interface, the default values did not match those previously used in AudioSendStream::Stats.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2469783002
Cr-Commit-Position: refs/heads/master@{#14896}
This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2419563003
Cr-Commit-Position: refs/heads/master@{#14824}
