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}
First step to only expose the relevant RNN VAD API to AGC2.
Bug: webrtc:7494
Change-Id: I7f11f6eebded124c30cabd64963c8e3ccc35e58f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185124
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32185}
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}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
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}
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.
Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.
Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
Currently test code passes pointer to temporary objects, while
RtcpSender passes raw pointers to objects that are then seen as owned,
and will be manually deleted by a overloaded destructor, which is scary
and fragile.
This CL moves all usage to std::unique_ptr<RtcpPacket> instead, which
may create some heap churn in unit tests but that should be fine.
Bug: webrtc:11925
Change-Id: I981bc7ccd6a74115c5a3de64b8427adbf3f16cc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183920
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32084}
It is meant for Pinpoint to run only the relevant tests when running a bisection.
The Pinpoint side of this change can be found here:
https://crrev.com/c/2404161
Bug: webrtc:11084
Change-Id: I466f39816b83e2f83a3a49845c99605f4d5a857b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183763
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32082}
Currently is_linux is set to true on Chrome OS build,
but it is planned to be set false. This CL is the preparation
to keep the compatibility.
Bug: chromium:1110266
Test: Build locally.
Change-Id: Ic79a202b0b3baeff157955cd03a07556bfb958a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Hidehiko Abe <hidehiko@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32073}
This CL adds explicit initialization of the FilterAnalyzer in AEC3.
While the current code never uses any fields before they are initialized,
it makes sense to be on the safe side and add initialization during
construction.
Bug: webrtc:11918
Change-Id: I467c4c8b8d6dd859a1b216baef28ac1e9d3f76c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183764
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32069}
For Non-DPI aware windows, we need to figure out the current DPI
and scale the content accordingly, the current behavior works ok
for until the clipped region pushes the content outside of the
frame and then the capture will fail. When this happens, the
captured frame may be blank or it could cause the browser to crash.
The issue is that the left and top clipped regions are not being
scaled along with the content (the captured window region is
contained within a larger window frame). When the clipped window
and window frame are scaled, the original offset for left and top
are not adjusted so after a certain DPI, this offset causes the
clipped region to get pushed outside of the frame which is why
the capture fails.
The fix is to scale the left and top clipped regions and translate
the clipped region accordingly. This change will only affect non-DPI
aware windows.
Bug: chromium:1083527
Change-Id: I893c2cb362cbaa01170d1e58465e43c3517139ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183660
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32065}
Removes the need for specifying a fixed number of parameters.
Bug: none
Change-Id: I1324861807cb4929963aedccb6c2755b9c6ea3fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180421
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32055}
This is a reland of 79098821a23f9de49f70cb3794b51e2730bffa01
with changes to disable the tests when not building with X11 support.
TBR=sergeyu@chromium.org
Original change's description:
> reenable mouse_cursor_monitor tests on linux
>
> BUG=webrtc:3245
>
> Change-Id: Ibf9cd929b22a0a519950621da46eb9f5b3febd73
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181367
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Sergey Ulanov <sergeyu@google.com>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31940}
BUG=webrtc:3245
Change-Id: I882e08f6f425df357f16fa4db25dcdf79db1f367
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181882
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32047}
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.
Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
