|is_posix| will be switched to false for Fuchsia, this is a preliminary change.
Bug: chromium:812974
Change-Id: I3bfda3e056ad1e5229834286ce5d095d9204a428
Reviewed-on: https://webrtc-review.googlesource.com/65782
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Fabrice de Gans-Riberi <fdegans@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22753}
We put back the old noise estimator from LevelController. We add a few
new unit tests. We also re-arrange the code so that it fits with how
it is used in AGC2. The differences are:
1. The NoiseLevelEstimator is now fully self-contained.
2. The NoiseLevelEstimator is responsible for calling SignalClassifier
and computing the signal energy. Previously the signal type and
energy were used in several places. It made sense to compute the
values independently of the noise calculation.
3. Re-initialization doesn't have to be done by the caller.
4. The interface is AudioFrameView instead of AudioBuffer.
# Bots are green, nothing should break internal stuff
NOTRY=True
Bug: webrtc:7494
Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d
Reviewed-on: https://webrtc-review.googlesource.com/66380
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22738}
The level estimator (AdaptiveModeLevelEstimator) produces a biased
estimate of the speech level. In our model, we use another module
(the SaturationProtector) to compute the bias. This CL contains the
estimator and a stub of the saturation protector.
Bug: webrtc:7494
Change-Id: I0df736d0346063f544fa680b4cc84177ea548545
Reviewed-on: https://webrtc-review.googlesource.com/64820
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22641}
This CL defines the control flow of the adaptive AGC. It also defines
method and class stubs.
Contents:
1. Divide the 'agc2' build target into 'fixed_digital' and
'adaptive_digital'.
1. Update the dependencies of everything that depended on 'agc2'.
2. Define the sub-modules of the adaptive digital AGC 2. They are:
1. Level Estimator - it gets the energy and a speech probability
and updates a speech level estimate.
2. Noise Estimator - it gets an immutable view of the speech frame
and updates the noise level estimate
3. Gain applier - it gets the speech frame, the current speech and
noise estimates, and the speech probability. It finds a gain to
apply and applies it.
4. AdaptiveAgc - sets up and controls the sub-modules described
above.
Bug: webrtc:7494
Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1
Reviewed-on: https://webrtc-review.googlesource.com/64440
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22628}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I440974da4d347b09ff042478720d7983056b62b9
Reviewed-on: https://webrtc-review.googlesource.com/21226
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22579}
The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world.
Bug: webrtc:8732
Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8
Reviewed-on: https://webrtc-review.googlesource.com/61801
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22449}
This refactoring makes it easier to experiment with injectable components.
Bug: webrtc:8732
Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217
Reviewed-on: https://webrtc-review.googlesource.com/60863
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22372}
This is a reland of 6f37ed78d99daa36e964ff0a65b205f0916d9949
CQ dry run OK except for missing iOS swarming bots.
NOTRY=True
Original change's description:
> Deprecate the adaptive level controller
>
> Level control handled by default-on AGC.
>
> Bug: none
> Change-Id: I405daeceece12c896d41156b649fcfd556726f77
> Reviewed-on: https://webrtc-review.googlesource.com/59682
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22305}
Bug: none
Change-Id: I0b9b8e2f3457d5efd4603efbfbbc6b84651df315
Reviewed-on: https://webrtc-review.googlesource.com/60720
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22352}
This CL extends the options for the audioproc_f tool to match the options
for AEC3.
Bug: webrtc:8671
Change-Id: I39972eae33dba461b94118ec47a8560eb9cfe5a6
Reviewed-on: https://webrtc-review.googlesource.com/43120
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22344}
This provides the empty shell of an AudioGenerator class.
It is intended to be used for debugging purposes and can be inserted
into the APM much like an AecDump. It allows for playing out diagnostic
audio unaffected by codecs and network jitter, while still capturing
API interaction like in a normal call.
NOTRY=True
Bug: webrtc:8882
Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7
Reviewed-on: https://webrtc-review.googlesource.com/53300
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22282}
This CL moves the AEC3 API call skew estimator into a separate file.
This has the advantage that it can more easily be tested.
The CL also simplifies the code and adds unittests.
Bug: webrtc:8671
Change-Id: I19bc31ca5666cdc87a1ed14770ef20ead1b5b80d
Reviewed-on: https://webrtc-review.googlesource.com/55860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22144}
Targets containing files in api/audio are moved from api/BUILD.gn to
api/audio/BUILD.gn.
Bug: webrtc:8844
Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981
Reviewed-on: https://webrtc-review.googlesource.com/54360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22074}
This CL adds the Level Estimator of the new gain controller. The Level
Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub
frames. We take the maximal sample values in every sub frame. We then
apply attack/decay smoothing. This is the final level estimate.
The results will be used with InterpolatedGainCurve (see this CL
https://webrtc-review.googlesource.com/c/src/+/51920). For every level
estimate value, we look up a gain with
InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to
the signal.
Bug: webrtc:7949
Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1
Reviewed-on: https://webrtc-review.googlesource.com/52381
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22054}
The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.
The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().
This CL contains
* build changes to make modules/audio_processing/agc2 an independent
target
* a new MutableFloatAudioFrame which is the audio interface between
AGC2 and APM
* move of the fixed gain application from GainController2 to
FixedGainController.
If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#
Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
Avoid including audio_processing.h from within AEC3.
Bug: webrtc:8844
Change-Id: I02c475c2fb84e2c24eac86baac3c7edaa08bebc0
Reviewed-on: https://webrtc-review.googlesource.com/53065
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22029}
WHAT: made a BUILD.gn with library and tests in the Audio Processing
Module Voice Activity Detector directory. Updated depending
code. Fixed a Clang warning.
WHY: to make it possible for a target to depend on just the VAD and
not the whole APM. There are other benefits:
* Sometimes faster compilation.
* The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making
a peerconnection shared object file without the VAD has to be done in
steps. The first step is a custom target for the VAD. Hence this Cl.
Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af
BUG: webrtc:5716, webrtc:7494
Reviewed-on: https://webrtc-review.googlesource.com/47521
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21893}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
The faster AEC3 alignment introduced recently may in
cases cause the alignment (and the AEC3) to repeatedly
reset. This CL avoids these resets by handling buffer
issues (which are triggering the resets) separately
during the initial coarse alignment phase.
Change-Id: Idf5e2ffda2591906da8060d03ec8ca73cdaedf53
Bug: webrtc:8798,chromium:805815
Reviewed-on: https://webrtc-review.googlesource.com/43480
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21758}
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
This CL forces the AEC2 to assume a stream delay of 0, thereby
avoiding that the incorrect stream delays reported on Chrome OS
causes echo issues.
Bug: chromium:797274, chromium:797272
Change-Id: I10f295c9f1d735622c55fc56be99a14c6cdd88a2
Reviewed-on: https://webrtc-review.googlesource.com/36081
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21432}
I realized I could use configs to fix some duplication that I
partially introduced.
Verified APM_DEBUG_DUMP is set appropriately by looking at the
compiler command line.
Bug: webrtc:6828
Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da
Reviewed-on: https://webrtc-review.googlesource.com/34642
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21349}
This solves a circular dep and eliminates a target.
This means we will apply neon copts to some files that weren't before,
but I don't think that is a problem.
Bug: webrtc:6828,webrtc:7042
Change-Id: I3bb656ba5b13d6104b519c2dbf6a4b2814575b87
Reviewed-on: https://webrtc-review.googlesource.com/34183
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21330}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
This tool is used downstream, so we want to christen rtc_tools as
a kind of api dir for tools. Tools in other locations should be
considered off limits.
I chose rtc_tools because video_quality_toolchain is already there,
which is also used downstream.
Bug: None
Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836
Reviewed-on: https://webrtc-review.googlesource.com/32340
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21236}
Arguably we should add a few more targets, for instance a utility
target, but I tried to create as few targets as possible here based on
the current usage.
Bug: webrtc:6828
Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8
Reviewed-on: https://webrtc-review.googlesource.com/28301
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21199}
