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}
Currently, key frames are scheduled even when the encoder is not reset
during reconfigeration. This means whenever new parameters like max
bitrate or min bitrate are updated through SetRtpParameters(), the
triggered encoder reconfigeration will always schedule key frames even
they are not necessary. Since parameters' changes like bitrate doesn't
require encoder instance reset.
This causes flood of key frames in our app since we do regularly max
bitrate update according to server control message.
Bug: None
Change-Id: I15d953b24c30e6026c0e97b30f44495d845f293f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185380
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32245}
Essentially, instead of having the inlined UntypedFunction::Create(f)
return an UntypedFunction which is then passed as an argument to
non-inlined RoboCallerReceivers::AddReceiverImpl(), we let
UntypedFunction::PrepareArgs(f) return a few different kinds of
trivial structs (depending on what sort of type f has) which are
passed as arguments to non-inlined RoboCallerReceivers::AddReceiver()
(which then converts them to UntypedFunction by calling
UntypedFunction::Create()). These structs are smaller than
UntypedFunction and optimized for argument passing, so many fewer
instructions are needed.
Example code:
struct Foo {
void Receive(int, float, int, float);
void TestAddLambdaReceiver();
webrtc::RoboCaller<int, float, int, float> rc;
};
void Foo::TestAddLambdaReceiver() {
rc.AddReceiver([this](int a, float b, int c, float d){
Receive(a, b, c, d);});
}
On arm32, we get before this CL:
Foo::TestAddLambdaReceiver():
push {r11, lr}
mov r11, sp
sub sp, sp, #24
ldr r1, .LCPI0_0
mov r2, #0
stm sp, {r0, r2}
add r1, pc, r1
str r2, [sp, #20]
str r1, [sp, #16]
mov r1, sp
bl RoboCallerReceivers::AddReceiverImpl
mov sp, r11
pop {r11, pc}
.LCPI0_0:
.long CallInlineStorage<Foo::TestAddLambdaReceiver()::$_0>
CallInlineStorage<Foo::TestAddLambdaReceiver()::$_0>:
ldr r0, [r0]
b Foo::Receive(int, float, int, float)
After this CL:
Foo::TestAddLambdaReceiver():
ldr r3, .LCPI0_0
mov r2, r0
add r3, pc, r3
b RoboCallerReceivers::AddReceiver<1u>
.LCPI0_0:
.long CallInlineStorage<Foo::TestAddLambdaReceiver()::$_0>
CallInlineStorage<Foo::TestAddLambdaReceiver()::$_0>:
ldr r0, [r0]
b Foo::Receive(int, float, int, float)
(Symbol names abbreviated so that they'll fit on one line.)
So a reduction from 64 to 28 bytes. The improvements on arm64 and
x86_64 are similar.
Bug: webrtc:11943
Change-Id: I93fbba083be0235051c3279d3e3f6852a4a9fdad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185960
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32244}
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}
Currently isolated output directory is created in flags_compatibility.py script.
This doesn't work for android swarming tasks because this script isn't called.
Bug: webrtc:11895
Change-Id: I8b8f01850d6e5970292b524d104314eef7ab17be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185883
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32236}
This reduces the degree of interdependency among modules related
to the PeerConnection class, and makes it easier to isolate inappropriate
external dependencies.
Bug: webrtc:11967
Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32235}
Add ability to specify which metrics to plot on the plotter level and
add sorting of plottable data because there is no guarantee on the perf
writer side that output is sorted by time.
Bug: webrtc:11959
Change-Id: I87e6f5720fff2b259f58e3fc5f7ed2462568e0d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32233}
This component is heavily referenced by both PeerConnection and
SdpOfferAnswerHandler; it's likely that it will end up in
SdpOfferAnswerHandler.
Encapsulation makes it easier to move around.
Bug: webrtc:11995
Change-Id: I5329d9a90159d203510bf3698962cd246eea7324
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32229}
Also remove test code that can cause leaks into production.
Add sequence checkers.
Bug: webrtc:11988
Change-Id: I67b4cec6ee77d73ccffbbc88c9081ebb3c3cc423
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185503
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32228}
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}
SignalDtlsSrtpSetupFailure is never fired, so the setup code for it,
is dead code. Also removing declarations for methods that have no
implementation.
For other public signals in BaseChannel I've added an accessor which
has revealed a threading problem due to the member variable being public.
Bug: webrtc:11994
Change-Id: Iec6046c6a598066b92c956002ba4160708ae7dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32211}
The callback that the asyncinvoker was being used for, will now use
a safety flag to check if call_ is valid before issuing calls.
Using the flag is a step towards removing the call_ptr_ variable
but in this CL we're just looking at replacing use of the async invoker.
The safety flag is cleared at the same time as call_ is, which prevents
pending callbacks for that call instance from running.
Also adding TODOs related to this change that will be
followed upon in other CLs.
Bug: webrtc:11988, webrtc:11992, webrtc:11993
Change-Id: If3986758af6d01d39b2db0cce82e57fc48be9d7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185508
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32208}
This can be used in the future to test NV12 video frames with encoders, both
from unittests and from tools like video_loopback.
Tested using video_loopback with generator NV12.
Bug: webrtc:11978
Change-Id: I0d24ae3ebab2267f076703cbda81e99cec465ec8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185045
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32206}
OnSctpInboundPacket is called not only for incoming packets, but for
notifications, which can be delivered on the usrsctp timer thread.
I suspect that these notifications can be delivered after we attempt to
close the socket, because if we attempt to close it while the timer
thread holds a reference, it isn't actually destroyed until the timer
thread finishes its operation.
Bug: chromium:1127774
Change-Id: Id6a883b14796e8f5bf1c2990f3d9d389d72c8a46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32204}
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}
