This CL was written in preparation for the next CL in the chain and
it contains the following changes:
- SignalWithLevels -> AdaptiveDigitalGainApplier::FrameInfo
- Frame view removed from AdaptiveDigitalGainApplier::FrameInfo
- AdaptiveDigitalGainApplier::Process now gets side info as const& to
avoid unnecessary copies
- AdaptiveAgc::Process: `last_audio_level` renamed to `limiter_envelope`
to better reflect what that actually is
- Missing class/method docstrings added
Tested: bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: Ie25dcd389d6eed74ea9a65f0720eeb8f20f0096b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186040
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32251}
This is the last CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.
This CL adds a second state property to hold temporary updates and a
counter for consecutive speech frames. When enough speech frames are
observed, the reliable state is updated; otherwise, the temporary state
is discarded.
The default for `AdaptiveModeLevelEstimator::min_consecutive_speech_frames_`
is 1, which means that the new feature is disabled.
Tested:
- Bit-exactness verified with audioproc_f
- Not bit-exact if `min_consecutive_speech_frames_` set to 10
Bug: webrtc:7494
No-Try: True
Change-Id: I0daa00e90c27c418c00baec39fb8eacd26eed858
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185125
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32250}
This reverts commit e820cef5340610b9beebbcb63868743b95b97fcd.
Reason for revert: Breaks downstream client. I will investigate and
get back with a suggestion to fix.
Original change's description:
> Begin implementing WGC CaptureFrame
>
> This change introduces the design that will allow us to deliver frames
> synchronously to callers despite the Windows.Graphics.Capture APIs being
> inherently asynchronous.
>
> We achieve this by having WindowCapturerWinWgc create and maintain a
> WgcCaptureSession object for each window that it is asked to capture a
> frame for. The capture session object will be the class that actually
> uses the WGC APIs, and it will store the frames it receives in a frame
> pool and deliver them via GetMostRecentFrame.
>
> The next CL will add the necessary functionality to the
> WgcCaptureSession class.
>
> Bug: webrtc:9273
> Change-Id: I44e164f4874503d8ccc8e6a210e74f9c8458f6c4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184220
> Commit-Queue: Austin Orion <auorion@microsoft.com>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32240}
TBR=mbonadei@webrtc.org,jamiewalch@chromium.org,tommi@webrtc.org,auorion@microsoft.com
Change-Id: I114944357ce5be7d1e2da817703dc95d544aa99a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9273
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186045
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32248}
- State -> LevelEstimatorState
- Mark two methods as const
- Call DumpDebugData() in one place
- DumpDebugData: don't check if data dumper is provided
- Add LevelEstimatorState::operator==
The changes will reduce clutter in follow up CL.
Note: this CL breaks the chain of 3 CLs titled
"AGC2 AdaptiveModeLevelEstimator min consecutive speech frames".
Bug: webrtc:7494
Change-Id: If39ce4b787069bef4af910d718cdfae3af1784a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185811
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32247}
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.
The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.
Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
This change lets the fuzzer modify the first few bytes of the RTP
payload. One of the benefits is that it can cover the RED header
splitter functionality.
The CL also fixes an issue found while running the fuzzer locally.
Bug: webrtc:11640
Change-Id: I7ca73676440897a14a0aaca796f70d381e016575
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185819
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32242}
This change introduces the design that will allow us to deliver frames
synchronously to callers despite the Windows.Graphics.Capture APIs being
inherently asynchronous.
We achieve this by having WindowCapturerWinWgc create and maintain a
WgcCaptureSession object for each window that it is asked to capture a
frame for. The capture session object will be the class that actually
uses the WGC APIs, and it will store the frames it receives in a frame
pool and deliver them via GetMostRecentFrame.
The next CL will add the necessary functionality to the
WgcCaptureSession class.
Bug: webrtc:9273
Change-Id: I44e164f4874503d8ccc8e6a210e74f9c8458f6c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184220
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32240}
`AdaptiveModeLevelEstimator::last_level_dbfs_` doesn't need to be optional.
Note: this CL breaks the chain of 3 CLs titled
"AGC2 AdaptiveModeLevelEstimator min consecutive speech frames".
Bug: webrtc:7494
Change-Id: Id5b409ca5cb5f11ed132c861b7995b9721e167bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185809
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32237}
This CL fixes a bug where the RtpPackeToSend::fec_protect_packet flag
was not cleared when a packet copy was fetched from the packet history
in order to be retransmitted. This caused the packet to be added to the
FEC generator a second time when the retransmission passed through
RtpSenderEgress.
The bug did not affect RTX retransmission and only manifests when using
deferred FEC generation.
Bug: webrtc:11340
Change-Id: Ic7ce2800cce9a99e74bd3dd697bc0779d2a02fda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185817
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32227}
This is the second CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.
In this CL, the `SaturationProtector` class has been replaced by a
struct that define the state and two functions to change it.
This is done in order to use the saturation protector state in
`AdaptiveModeLevelEstimator::State` and will allow to add a
temporary state in `AdaptiveModeLevelEstimator` (see the child CL).
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: Ic5ecd1e174010656ed20664ef7b7e5798ebb7978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185041
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32226}
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.
Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
This change completely disables the use of suppression
when operating in transparent mode.
It also removes the following field trials:
* WebRTC-Aec3UseLowEarlyReflectionsTransparentModeGain
* WebRTC-Aec3UseLowLateReflectionsTransparentModeGain
Bug: webrtc:11985
Change-Id: I1c75efdad2d9c9d0a1aced86bf0278fc96616ea1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185402
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32223}
NV12 frames can be encoded by libvpx now, and this change allows for
encoding of them with VP9.
VP9 encode/decode tests now run with NV12 as well as I420.
Manually tested using video loopback with VP9 and NV12 generated frames.
out/Default/video_loopback.app/Contents/MacOS/video_loopback --clip=GeneratorNV12 --codec="VP9"
Bug: webrtc:11635, webrtc:11974
Change-Id: Ifc5cbf77d2a27821cd5560c253d5d447c7a7cf53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185123
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32220}
I ran into this when using repeating_task, which depends on clock (in
system_wrappers) which in turn added a dependency on rtc_base on Windows
due to win32 files. That's a problem since rtc_base depends on
repeating_task:
//rtc_base:rtc_base ->
//rtc_base/task_utils:repeating_task ->
//system_wrappers:system_wrappers ->
//rtc_base:rtc_base
We could additionally consider moving Clock out of system_wrappers.
Bug: webrtc:9987
Change-Id: I54ed715ad5eb9e3f5dd6c322233c18c05d895dff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185506
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32203}
This is a reland of ad148272b89394978915cb00e1c1be552d908a42
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
Bug: webrtc:11663
Change-Id: I669435c2f4e451ee0766d809443484f2dde09d8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32200}
This is the first CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.
In this CL, the state of the estimator is defined in a separate struct
so that in a follow-up CL a new member of that type can be added to
hold a temporary state (that can be either confirmed or discarded).
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: Ic2ea5ed63c493b9f3a79f19e7f5eaecaa6808ace
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184931
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32199}
Feature added to gain robustness to occasional VAD speech probability
spikes. In such a case, the attack process reduces the chance that the
smoothed values are greater than the speech threshold.
Bug: webrtc:7494
Change-Id: I6babe5afe30ea3dea021181a19d86bb74b33a98c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185046
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32198}
This reverts commit ad148272b89394978915cb00e1c1be552d908a42.
Reason for revert: Causing test failures downstream.
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
TBR=mbonadei@webrtc.org,saza@webrtc.org,peah@webrtc.org,kwiberg@webrtc.org
Change-Id: If2287a0a4b37931ce5f85baae093a66b19d0a78b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11663
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185481
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32196}
Refactoring CL to improve names and allow to inject a VAD into
`VadLevelAnalyzer` (new name for `VadWithLevel`).
The injectable VAD is needed to inject a mock VAD and write better
unit tests as new features are going to be added to the class.
Bug: webrtc:7494
Change-Id: Ic0cea1e86a19a82533bd40fa04c061be3c44f068
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185180
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32195}
This CL activates the newly added AVX2 support by default.
The activation is done beneath a kill-switch.
Beyond the above, the CL also changes an incorrect DCHECK_GT
to a DCHECK_GE.
Bug: webrtc:11663
Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32193}
This change introduces a new Hidden Markov Model based classifier for
AEC3's 'transparent mode'. Transparent mode is used with
headsets/headphones where the speaker signal does not leak into the
microphone signal.
The current classifier suffers from two problems:
1. It sometimes takes a long time to enter transparent mode.
2. Sometimes transparent mode is left (and it once again takes a long
time to re-enter).
Both problems have a severe effect on AEC transparency.
The new classifier enters transparent mode quicker and is less likely
to exit transparent mode when there is no echo. This improves the
audio experience when using headset/headphones.
Another (minor) benefit of this change is that when transparent mode
is disabled no classifier is run (or even created) saving some memory
and CPU cycles.
Bug: webrtc:10232
Change-Id: I509af0e22b59463aeaead53c78c35be1e97fe8c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184500
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32182}
In preparation for a coming refactoring CL, the (fixed) extra saturation
margin is now applied into `AdaptiveModeLevelEstimator`.
This CL also improves the unit tests by hard-coding its saturation
params instead of reading them from a field trial.
This reduces the chances of making the test flaky if a default value
changes.
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: I6765def9887a2f4e55b04d929af754cfecbb1626
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184927
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32172}
Video playout delay is used to give a hint to the receiver
how the video should be played out.
Add the field trial WebRTC-ForceSendPlayoutDelay to set the video playout
delay of outgoing RTP packets to enable experimentation with this feature.
Bug: webrtc:11896
Change-Id: Ie6123b5967763bde6a830f4c5e5a963e73fb0acb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185042
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32169}
- Passing the speech peak power instead of VAD data
- The private class SaturationProtector::PeakEnveloper has been removed
- Added `initial_saturation_margin_db_` parameter to correctly
initialize `last_margin_` (renamed to `margin_db_`)
- Member names have been fixed and/or shortened for better readability
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: I6cad2974397319737c8ac201d44311bf16275f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184925
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32168}
Even if small, the peak delay buffer copies N-1 elements for each frame
whereas a ring buffer is copy-free and scales better if the buffer size
increases.
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: If8c33877b7ab1d881a0606e222b26857a82fff69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184920
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32165}
Unfortunate typo and weak tests made it so if only a middle spatial layer
is active, vp9 encoder would be configured to send two top layers.
Bug: webrtc:11319
Change-Id: I460c245044f60ea7e0127c0e4134d0edab85f4f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185043
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32164}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
CpuSpeedExperiment: Add option to have a separate config for cores below a configurable threshold.
Bug: none
Change-Id: I51562979f3a89a949d014a1ee6fc0802f3c1dae5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184926
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32154}
It should already be enabled by default in libaom, but explicitly enable
it here in case that changes.
Bug: None
Change-Id: I93a1dfc92f9c02bc5ec823c326d8cf6ff163bceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184262
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32114}
