For x86, x86_64, arm and arm64
Bug: 261600888
Test: build and run cuttlefish x86, x86_64 and arm64
Change-Id: I3ac4dad1ac9ec83b0e626e64715df450e8809b82
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This places the bit exactness testing tools in audioproc_test_utils,
and removes it from audio_processing_unittests.
Bug: webrtc:8240
Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba
Reviewed-on: https://chromium-review.googlesource.com/663545
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19815}
Added script 'apm_quality_assessment_optimize' for finding parameters
that minimize a custom function of the scores generated by APM-QA. The
script reuses the existing functionality for filtering the data on
configs/scores/outputs.
To archieve that, some modularization has been done: the part from
apm_quality_assessment_export that reads in data into a
pandas.DataFrame has been moved into quality_assessment.collect_data.
TESTED = though extensive manual tests. Unit tests for the user
scripts and 'collect_data' are missing, because we don't have a test
framework for loading/exporting fake data.
BUG=webrtc:7218
Change-Id: I5521b952970243da05fc4db1b9feef87a2e5ccad
Reviewed-on: https://chromium-review.googlesource.com/643292
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19780}
This CL reduces the allowed jitter in the api calls to a reasonable
level in order to ensure a quicker revery from audio path glitches.
BUG=webrtc:8224, chromium:763775
Review-Url: https://codereview.webrtc.org/3009273002
Cr-Commit-Position: refs/heads/master@{#19772}
We use Optional in our public API, so its header should be in
webrtc/api/.
BUG=webrtc:8205
Review-Url: https://codereview.webrtc.org/3011943002
Cr-Commit-Position: refs/heads/master@{#19693}
This is done to make UBSan testing more convenient in integration with projects using WebRTC
Some blacklist entries were obsolete so don't need a replacement
Also fix one of the warnings (thanks, kwiberg@)
BUG=webrtc:8189, webrtc:5486, webrtc:5491
Review-Url: https://codereview.webrtc.org/3009123002
Cr-Commit-Position: refs/heads/master@{#19682}
The warnings were (all signed integer overflow):
webrtc/common_audio/signal_processing/levinson_durbin.c:46:25
12 * 268435456 cannot be represented in type 'int'
webrtc/modules/audio_processing/aecm/aecm_core.cc:930:69
522240 * 6115 cannot be represented in type 'int'
webrtc/modules/audio_processing/aecm/aecm_core_c.cc:455:36
72293096 * 50 cannot be represented in type 'int'
webrtc/modules/pacing/alr_detector.cc:70:48
1000000000 * 65 cannot be represented in type 'int'
webrtc/modules/rtp_rtcp/source/rtp_sender.cc:947:20
1929277286 + 321546521 cannot be represented in type 'int'
BUG=webrtc:8195
Review-Url: https://codereview.webrtc.org/3005003002
Cr-Commit-Position: refs/heads/master@{#19670}
We use ArrayView in our public API, so its header should be in
webrtc/api/.
BUG=none
Review-Url: https://codereview.webrtc.org/3007763002
Cr-Commit-Position: refs/heads/master@{#19658}
This CL contains automatically applied fixes suggested by the
ClangTidy analyzer (http://clang.llvm.org/extra/clang-tidy/). The
following kinds of fixes is present:
* renaming variables when the names in the method signature don't
match the names in the method definition
(ClangTidy:readability-inconsistent-declaration-parameter-name)
* ClangTidy:readability-container-size-empty,
ClangTidy:misc-unused-using-decls,
ClangTidy:performance-unnecessary-value-param,
ClangTidy:readability-redundant-control-flow
This is a 'pilot' CL to check if automatic code analyzers can
feasibly be integrated into the WebRTC infrastructuve.
The renamings have been manually expected for consistency with
surrounding code. In echo_cancellation.cc, I changed several names in
the function implementation to match the function declaration. The
tool suggested changing everything to match the function definitions
instead.
Bug: None
Change-Id: Id3b7ba18c51f15b025f26090c7bdcc642e48d8fd
Reviewed-on: https://chromium-review.googlesource.com/635766
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19630}
This CL increases the amount of API call jitter that is allowed in AEC3
without causing resets of AEC3. This increase is now possible, as non-
causal alignments will be detected by the newly imposed delay bound.
BUG=8175
Review-Url: https://codereview.webrtc.org/3012553002
Cr-Commit-Position: refs/heads/master@{#19603}
directives in our DEPS files are not needed anymore.
Includes from webrtc/rtc_base are also whitelisted in webrtc/DEPS
so we don't have to whitelist it in all the others DEPS files.
BUG=webrtc:7634
NOTRY=True
Review-Url: https://codereview.webrtc.org/3006583002
Cr-Commit-Position: refs/heads/master@{#19601}
This CL ensures that the adaptive filter delay is not used for fine
tune echo removal unless the render and capture signals have been
properly aligned.
BUG=webrtc:8189
Review-Url: https://codereview.webrtc.org/3003303002
Cr-Commit-Position: refs/heads/master@{#19492}
This CL robustifies the inaudible echo detection in AEC3 such that a
requirement is that either the render and capture signals are aligned
or that a headset has been detected. This ensures that the inaudible
detection has been able to base the desicion on reliable signals.
BUG=webrtc:8150
Review-Url: https://codereview.webrtc.org/3005503002
Cr-Commit-Position: refs/heads/master@{#19491}
no-unused-lambda-capture was suppressed, but it's been decided as desireable to stop suppressing it. This CL fixes places in the code that trigger it.
1. Some unnecessary captures removed.
2. s/constexpr/const when capturing a float by value - this is good enough to stop the error.
3. Complete removal of the constexpr/const-modifier for int-types as a workaround.
BUG=webrtc:7133
Review-Url: https://codereview.webrtc.org/3005433002
Cr-Commit-Position: refs/heads/master@{#19462}
This CL ensures that AEC3 recovers more quickly when capture data is
lost in such a manner that the echo path, as seen by AEC3, becomes
noncausal due to the AEC3 buffer misalignment caused by the data loss.
The CL adds the assumption of a minimum echo path delay of 5 blocks
and makes the hysteresis in the delay selection one-sided.
BUG=chromium:757796, webrtc:8131
Review-Url: https://codereview.webrtc.org/2998223002
Cr-Commit-Position: refs/heads/master@{#19454}
This CL completely removes the methods
AudioProcessing::{Start,Stop}DebugDumpRecording. These methods have
been replaced with AudioProcessing::{Attach,Detach}AecDump. Their
implementation was removed in the parent CL
https://chromium-review.googlesource.com/c/589147
Bug: webrtc:7404
Change-Id: Ia3d5314985af9c74f79c94c514ded1f8afc78fb5
Reviewed-on: https://chromium-review.googlesource.com/589152
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19334}
AudioProcessingModule has a feature to make a recording of its
configuration, inputs and outputs over a period of time. In the past
CLs, this feature has been rewritten to move file IO away from
real-time audio threads. The interface has changed from
{Start,Stop}DebugDumpRecording to {Attach,Detach}AecDump.
This CL removes the previous implementation of the old interface
StartDebugRecording. The public interface is left to not cause
problems to downstream projects. It will be removed in the dependent
CL https://chromium-review.googlesource.com/c/589152/
With this CL, usage of WEBRTC_AUDIOPROC_DEBUG_DUMP and ~300 LOC of
logging code is removed from AudioProcessingImpl.
Bug: webrtc:7404
Change-Id: I16e7b557774e4bc997e1f5de4f97ed2c31d63879
Reviewed-on: https://chromium-review.googlesource.com/589147
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19332}
That way, the debug printout will tell us which of x and y that was false.
BUG=none
Review-Url: https://codereview.webrtc.org/2988153003
Cr-Commit-Position: refs/heads/master@{#19297}
Found via supersize query:
size_info.symbols.WhereFullNameMatches(r'\bk[A-Z]').WhereInSection('d')
This moves 90 symbols from .data -> .data.rel.ro (5.50kb)
BUG=chromium:747064
Review-Url: https://codereview.webrtc.org/2986163002
Cr-Commit-Position: refs/heads/master@{#19274}
If the input file name matches the "<name>-<params>.wav" pattern and <name> is a valid signal creator name, then <params> is parsed and used to create a new signal which is written in place of the missing file.
This CL only adds a pure tone creator. For instance, 'pure_tone-440_1000.wav' creates a pure tone at 440 Hz, 1000 ms long, mono, sampled at 48kHz.
This feature can be used to simplify the creation of common probe signals - no need to add external .wav files. Also, it will be exploited by a coming CL that adds a new evaluation score requiring the input signal to be a pure tone.
Additional minor fixes:
- apm_quality_assessment_unittest.py: command line arguments replaced to avoid that those for the unit test framework are passed
- simulation_unittest.py: invalid evaluation score name replaced
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2989823002
Cr-Commit-Position: refs/heads/master@{#19200}
- render stream support, required to assess AEC;
- echo path simulation and input mixer, to generate echo and add it to the
speech signal;
- export engine: improved UI, switch to Pandas DataFrames;
- minor design improvements and needed adaptions.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2813883002
Cr-Commit-Position: refs/heads/master@{#19198}
A multiplication result doesn't fit in an int32_t type. This change
rewrites the code to avoid the overflowing multiplication.
Here y[0], y[1] are int16 numbers containing the (truncated) topmost
18 and (scaled Q2 to use the full int16) the least significant 13
bits of a 32-bit value. The change makes y[1] to be calculated
directly instead of using y[0] as an intermediate value.
TESTED=this change passes the bit exactness tests, and has also been
running on the audio_processing fuzzer with a CHECK comparing the
old and new value.
Bug: chromium:747202
Change-Id: Iafc69eb7391d494afdadf65f5b7f399a57bbe9a8
Reviewed-on: https://chromium-review.googlesource.com/580907
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19120}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
The variable 'tmp_int32' in LowCutFilter::BiqueadFilter::Process can
be negative. This replaces a left shift with multiplication.
Bug: chromium:735593, chromium:743330
Change-Id: Idec7fbcc17495f7241eb4bea44920585740e3695
Reviewed-on: https://chromium-review.googlesource.com/575136
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19074}
This avoids a memcopy call which may corrupt audio handling and, in rare cases, crash WebRTC with a buffer over-read.
BUG=webrtc:7845
Review-Url: https://codereview.webrtc.org/2980723002
Cr-Commit-Position: refs/heads/master@{#18984}
The check triggered in 30 / 1000 cases of running PeerConnectionIntegrationTest.CallTransferredForCaller locally, far more often than expected.
It will soon be replaced by more graceful handling.
BUG=webrtc:7845
Review-Url: https://codereview.webrtc.org/2975043002
Cr-Commit-Position: refs/heads/master@{#18983}
This CL adds detection of components in the render signal that are of
strong narrowband nature and therefore may cause problems for the AEC.
This CL also adds functionality in the echo suppressor to suppress
these signals
BUG=webrtc:7967
Review-Url: https://codereview.webrtc.org/2980493002
Cr-Commit-Position: refs/heads/master@{#18968}
This CL robustifies the echo removal in AEC3 during the initial parts
of a call in two ways:
-By extending the period until which a headset is deemed to be used.
-By increasing the assumed echo path gain for unknown echo paths at
higher frequencies.
BUG=webrtc:7971
Review-Url: https://codereview.webrtc.org/2974883002
Cr-Commit-Position: refs/heads/master@{#18967}
This CL adds two changes:
-Adaptive adjustment of the echo suppression to both cover the cases
when the echo path well covers the room, and when when it does not.
-Identification of the case when the echo is too low to be audible
and adaptive handling of this case in the echo suppression.
BUG=webrtc:7519, webrtc:7956,webrtc:7957
Review-Url: https://codereview.webrtc.org/2974583004
Cr-Commit-Position: refs/heads/master@{#18962}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
Ensure that the ring buffer does not return a pointer into the buffer if
no data is available to read.
The ring buffer fix is not directly applicable to issue webrtc:7845, but may cause related memory errors.
BUG=webrtc:7845
Review-Url: https://codereview.webrtc.org/2971313002
Cr-Commit-Position: refs/heads/master@{#18940}
