The UMA histograms WebRTC.Audio.AecSystemDelayJump and WebRTC.Audio.PlatformReportedStreamDelayJump triggers if the jump is larger than kMinDiffDelayMs.
Especially WebRTC.Audio.AecSystemDelayJump is sensitive around 50 ms differences, since the granularity is 4 ms and we can get a significant amount of hits at 52 ms.
Therefore, a change to 60 ms can make the logging more robust. The effect of not logging jumps in the interval 50-60 ms is of minor importance since they are not likely to affect the AEC performance. It's when we get values from ~100 ms and above that we should be worried.
Tested with a local ToT Chromium build where 52, 64 and 200 ms jumps were forced.
BUG=488124
TBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1208313003.
Cr-Commit-Position: refs/heads/master@{#9540}
We use this Config struct for enabling/disabling the delay agnostic
AEC. This change renames it to DelayAgnostic for readability reasons.
NOTE: The logic is reversed in this CL. The old ReportedDelay config
turned DA-AEC off, while the new DelayAgnostic turns it on.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, ReportedDelay is disabled or DelayAgnostic is enabled, DA-AEC
is engaged in APM.
BUG=webrtc:4651
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1211053006
Cr-Commit-Position: refs/heads/master@{#9531}
Sudden platform system delay jumps can hurt AEC and we have no stats that monitor these jumps. How often do they occur, and when they are reported are they accurate?
This CL logs all jumps in both the reported and actual delay.
The histogram has been tested with a chromium build where a fake jump of 200 ms was applied after 5 seconds and it was registered correctly in chrome://histograms
BUG=488124
R=henrik.lundin@webrtc.org, peah@webrtc.org
Review URL: https://codereview.webrtc.org/1213733004.
Cr-Commit-Position: refs/heads/master@{#9513}
Revert of original: https://codereview.webrtc.org/1187033005/
Changes in original:
- Added files to gyp and BUILD
- Made minor fixes to get everything to compile
and intelligibility_proc to run
- Added comments
- Auto-reformatting
New Changes:
- Added <numeric> header to intelligibility_enhancer.cc to address buildbot errors
- Switched to use WAV for i/o in intelligibility_proc.cc to address windows errors
- clean up
Note: Patch 1 duplicates Patch 7 of https://codereview.webrtc.org/1182323005/R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1190733004.
Cr-Commit-Position: refs/heads/master@{#9486}
Original review at https://codereview.webrtc.org/1180423006
SystemDelayTests was not updated w.r.t. extended_filter mode and some tests were disabled on Android since DA-AEC is automatically set.
All tests have now been updated for both extended_filter mode as well as DA-AEC, hence are now enabled on Android.
Also
* Moves default settings of extended_filter and DA-AEC form Init() to Create() to avoid unintentional loss of state during a reset.
* Fixes a potential bug of starting from scratch in extended_filter mode + DA-AEC.
This reverts commit 01c9b012e9171c813ace9e405c32fc75f4262bf6.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1187943005.
Cr-Commit-Position: refs/heads/master@{#9458}
The code only affects DA-AEC, but since DA-AEC is the default AEC if run on Android tests failed. Reverting to fix that test.
This reverts commit 9002cc426dab7a576f5247f45ba888cd081a39f0.
BUG=
TBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1183243003.
Cr-Commit-Position: refs/heads/master@{#9453}
We've seen that if we get a buffer underrun followed by a sudden buffer build up the DA-AEC can't really catch up even though it should be possible to estimate the upcoming difference. We have a feature for this already, but that is only used in the regular AEC. This CL turns that feature on also for DA-AEC.
- Adds a helper function MoveFarReadPtrWithoutSystemDelayUpdate()
- Only apply conservative correction for positive delays, where we can put the AEC into a non-causal state
- Stuff the farend buffer if we don't have enough data to process w.r.t. to current nearend buffer.
- Always run delay estimation based on reported delays to catch buffer starvation.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1180423006.
Cr-Commit-Position: refs/heads/master@{#9452}
This is a follow-up to r9401, where the configuration DelayCorrection
was replaced by ExtendedFilter.
This change also removes the media constraint
kExperimentalEchoCancellation which was replaced by
kExtendedFilterEchoCancellation in the same CL.
Both settings that are now being removed were kept in the code to avoid
API breakages. In https://codereview.chromium.org/1167343004,
depending code has been updated to avoid breakages.
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1181413004.
Cr-Commit-Position: refs/heads/master@{#9444}
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question. This is preparation for a future change
that will convert a variety of types to size_t.
There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.
BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm
Review URL: https://codereview.webrtc.org/1174813003
Cr-Commit-Position: refs/heads/master@{#9413}
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)
The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.
Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec."
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1151573021.
Cr-Commit-Position: refs/heads/master@{#9401}
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec.
BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54659004
Cr-Commit-Position: refs/heads/master@{#9378}
The delay agnostic AEC uses a signal based delay correction method to adjust buffer synchronization between loudspeaker and microphone. On Mac in particular we have seen deviations in UMA stats that point towards an echo already at startup. This is likely due to an early and incorrect correction based on poor audio data.
By waiting 15 seconds before we turn on the ability to correct we can avoid a majority of these.
The reported delay values are in general accurate enough and relying on them in the beginning is fine. The value 15 seconds is chosen because we have seen from UMA data that a significant amount of calls tend to end before 15 seconds when being in the UseDelayAgnosticAEC Finch experiment.
We turn this "feature" on for all platforms but Android, where the reported system delays are inaccurate and we want to take action as soon as possible.
In addition, the set of "good" delay values has been increased from 25% to 75% of the filter length.
BUG=webrtc:3504
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50199004
Cr-Commit-Position: refs/heads/master@{#9376}
Reduces musical noise with minimal impact on interferer suppression.
This also unifies the treatment of "mean bins". The "end" bin is now
inclusive in the mean range as with the "start" bin.
Corrects interpretation of quantile.
BUG=chromium:490477
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50939004
Cr-Commit-Position: refs/heads/master@{#9317}
Changed Create() to return a pointer to the object rather than an error message, which is in line with how objects should be created.
BUG=441
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51939004
Cr-Commit-Position: refs/heads/master@{#9315}
Changed the WebRtcVad_Create() function to the more conventional format of returning the handle directly instead of an error code to take care of.
In addition NULL was changed to nullptr in the files where it applied.
Affected components:
* AGC
* VAD
* NetEQ
BUG=441, 3347
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51919004
Cr-Commit-Position: refs/heads/master@{#9291}
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}
Because of the Finch experiment, this will not affect Chrome's behaviour at all.
The SNRs in AudioProcessingTest.Formats were only increased to the next multiple of 5.
BUG=webrtc:3146
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43359004
Cr-Commit-Position: refs/heads/master@{#9263}
BUG=3056, 1320
TEST=AutoTest
Mainly add threadchecker and remove unnecessary lock.
And some more styling working.
- audio_device_pulse_linux.cc: wrap lines longer than 80 chars. And add '.' to some comments around. Not do it to all places.
- audio_mixer_manager_pulse_linux.cc: Here I adopt some chromium practice. We use to do many things to the failure of pulse operation, which causes most of the data race issue. In chromium, if we failed to call any pulse function, we just fail it w/o use the previous results. Here I did same. Please check if it's good.
R=bjornv@webrtc.org, henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52479004
Cr-Commit-Position: refs/heads/master@{#9243}
It is necessary for adding 48kHz support to the AudioProcessing::AnalyzeReverseStream int interface (It was not necessary for 32kHz since in that case the splitting filter is more efficient).
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56389004
Cr-Commit-Position: refs/heads/master@{#9241}
compile time.
The condition of static_assert() is evaluated at compile time which is safer and
more efficient.
Note that static_assert() requires C++11.
The changes were generated by the misc-static-assert ClangTidy check by alexfh@google.comR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51019004
Cr-Commit-Position: refs/heads/master@{#9231}
Merge WEBRTC_ARCH_ARM64_NEON and WEBRTC_ARCH_ARM_NEON into one
WEBRTC_HAS_NEON.
Replace WEBRTC_DETECT_ARM_NEON by WEBRTC_DETECT_NEON.
Replace WEBRTC_ARCH_ARM by WEBRTC_ARCH_ARM64 for arm64 cpu.
BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org
Change-Id: I870a4d0682b80633b671c9aab733153f6d95a980
Review URL: https://webrtc-codereview.appspot.com/49309004
Cr-Commit-Position: refs/heads/master@{#9228}
From 6kHz-6.5kHz to 3kHz-5kHz. Previous range had unreliable mask values, letting high frequencies from all directions through. The new range is wider and lower, which results in better estimates.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47089004
Cr-Commit-Position: refs/heads/master@{#9213}
This adds functionality from audioproc_float. The geometry parsing code
is now shared from test_utils.h. I removed the "mic_spacing" flag from
audioproc_float because it's a redundancy that I suspect isn't very
useful.
Includes a cleanup of the audio_processing test utils. They're now
packaged in targets, with the protobuf-using ones split out to avoid
requiring users to depend on protobufs.
pcm_utils is no longer needed and removed.
The primary motivation for this CL is that AudioProcessing currently
doesn't support more than two channels and we'd like a way to pass
more channels to the beamformer.
R=aluebs@webrtc.org, mgraczyk@chromium.org
Review URL: https://webrtc-codereview.appspot.com/50899004
Cr-Commit-Position: refs/heads/master@{#9157}
When running AEC in extended_filter mode there is no startup phase to evaluate the reported system delay values.
Instead we simply use the first value and scale by two to avoid over compensating when synchronizing render and capture.
We don't need to be too accurate since we have extended the filter length.
On Android we use fixed (measured) reported delay values.
There is no need to be extra conservative here, because that is already built-in in the measured value.
In fact, the difference between devices is large and with such an extra conservative approach the true delay can not be caught by the filter length.
With this change we can improve performance on some devices.
BUG=4472
TESTED=offline on recordings from various devices
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49909004
Cr-Commit-Position: refs/heads/master@{#9144}
