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Fix typo mismatch in field trial string for delay bwe weight #16
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@taste1981, please review. |
taste1981
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Aug 2, 2019
grgustaf
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taste1981
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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 <[email protected]>
Reviewed-by: Mirko Bonadei <[email protected]>
Cr-Commit-Position: refs/heads/master@{#32244}
taste1981
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Dec 30, 2020
This reverts commit 76d3e7a. Reason for revert: Causes multiple Chromium WPT tests to crash, preventing rolls. Sample failed run: https://ci.chromium.org/p/chromium/builders/try/win10_chromium_x64_rel_ng/685757? Sample stack trace: #0 0x7ff8623fbde9 base::debug::CollectStackTrace() STDERR: #1 0x7ff862311ca3 [2665012:17:1009/162250.249660:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652370324, interval (us) = 834 STDERR: base::debug::StackTrace::StackTrace() STDERR: #2 0x7ff8623fb93b base::debug::(anonymous namespace)::StackDumpSignalHandler() STDERR: #3 0x7ff857a70140 [2665012:17:1009/162250.249947:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652370634, interval (us) = 742 STDERR: (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f) STDERR: #4 0x7ff85778edb1 gsignal STDERR: #5 0x7ff857778537 abort STDERR: #6 0x7ff855d5eee2 [2665012:17:1009/162250.250342:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652371030, interval (us) = 706 STDERR: [2665012:17:1009/162250.250514:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652371204, interval (us) = 963 STDERR: rtc::webrtc_checks_impl::FatalLog() STDERR: #7 0x7ff855f14e62 webrtc::LibvpxVp8Encoder::PrepareRawImagesForEncoding() STDERR: #8 0x7ff855f14412 webrtc::LibvpxVp8Encoder::Encode() STDERR: #9 0x7ff855bae765 webrtc::SimulcastEncoderAdapter::Encode() STDERR: #10 0x7ff85607d598 webrtc::VideoStreamEncoder::EncodeVideoFrame() STDERR: #11 0x7ff85607c60d webrtc::VideoStreamEncoder::MaybeEncodeVideoFrame() STDERR: #12 0x7ff8560816f5 webrtc::webrtc_new_closure_impl::ClosureTask<>::Run() STDERR: #13 0x7ff855b352b5 (anonymous namespace)::WebrtcTaskQueue::RunTask() STDERR: #14 0x7ff855b3531e base::internal::Invoker<>::RunOnce() STDERR: #15 0x7ff86239785b base::TaskAnnotator::RunTask() STDERR: #16 0x7ff8623c8557 base::internal::TaskTracker::RunSkipOnShutdown() STDERR: #17 0x7ff8623c7d92 base::internal::TaskTracker::RunTask() STDERR: #18 0x7ff862415a06 base::internal::TaskTrackerPosix::RunTask() STDERR: #19 0x7ff8623c75e6 base::internal::TaskTracker::RunAndPopNextTask() STDERR: #20 0x7ff8623d3a8d base::internal::WorkerThread::RunWorker() STDERR: #21 0x7ff8623d368d base::internal::WorkerThread::RunPooledWorker() STDERR: #22 0x7ff862416509 base::(anonymous namespace)::ThreadFunc() STDERR: #23 0x7ff857a64ea7 start_thread Original change's description: > NV12 support for VP8 simulcast > > Tested using video_loopback with generated NV12 frames. > > Bug: webrtc:11635, webrtc:11975 > Change-Id: I14b2d663c55a83d80e48e226fcf706cb18903193 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186722 > Commit-Queue: Evan Shrubsole <[email protected]> > Reviewed-by: Ilya Nikolaevskiy <[email protected]> > Cr-Commit-Position: refs/heads/master@{#32325} [email protected],[email protected] # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:11635 Bug: webrtc:11975 Change-Id: I61c8aed1270bc9c2f7f0577fa5ca717a325f548a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187484 Reviewed-by: Guido Urdaneta <[email protected]> Reviewed-by: Ilya Nikolaevskiy <[email protected]> Commit-Queue: Guido Urdaneta <[email protected]> Cr-Commit-Position: refs/heads/master@{#32369}
taste1981
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Dec 30, 2020
This reverts commit c5f7108. Reason for revert: Causes Chromium WPT Tests to fail, preventing rolls. Sample failed run: https://ci.chromium.org/p/chromium/builders/try/linux-rel/511995? Sample logs: STDERR: # Fatal error in: ../../third_party/webrtc/pc/peer_connection.cc, line 575 STDERR: # last system error: 0 STDERR: # Check failed: (signaling_thread())->IsCurrent() STDERR: # Received signal 6 STDERR: #0 0x7f81d39e3de9 base::debug::CollectStackTrace() STDERR: #1 0x7f81d38f9ca3 base::debug::StackTrace::StackTrace() STDERR: #2 0x7f81d39e393b base::debug::(anonymous namespace)::StackDumpSignalHandler() STDERR: #3 0x7f81c9054140 (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f) STDERR: #4 0x7f81c8d72db1 gsignal STDERR: #5 0x7f81c8d5c537 abort STDERR: #6 0x7f81c7344032 rtc::webrtc_checks_impl::FatalLog() STDERR: #7 0x7f81c722e5c0 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #8 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #9 0x7f81c72d6e8e webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #10 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #11 0x7f81c71c6df3 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #12 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #13 0x7f81c73135bc rtc::OpenSSLStreamAdapter::ContinueSSL() STDERR: #14 0x7f81c7312fd4 rtc::OpenSSLStreamAdapter::OnEvent() STDERR: #15 0x7f81c71c30d9 cricket::StreamInterfaceChannel::OnPacketReceived() STDERR: #16 0x7f81c71c640a cricket::DtlsTransport::OnReadPacket() STDERR: #17 0x7f81c71cad61 cricket::P2PTransportChannel::OnReadPacket() STDERR: #18 0x7f81c71bc90f cricket::Connection::OnReadPacket() STDERR: #19 0x7f81c71e6255 cricket::UDPPort::HandleIncomingPacket() STDERR: #20 0x7f81cd1f1bff blink::(anonymous namespace)::IpcPacketSocket::OnDataReceived() STDERR: #21 0x7f81cd1f645d blink::P2PSocketClientImpl::DataReceived() STDERR: #22 0x7f81cd50a16b network::mojom::blink::P2PSocketClientStubDispatch::Accept() STDERR: #23 0x7f81d2b4f642 mojo::InterfaceEndpointClient::HandleValidatedMessage() STDERR: #24 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #25 0x7f81d2b50bb1 mojo::InterfaceEndpointClient::HandleIncomingMessage() STDERR: #26 0x7f81d2b58a3a mojo::internal::MultiplexRouter::ProcessIncomingMessage() STDERR: #27 0x7f81d2b57f7f mojo::internal::MultiplexRouter::Accept() STDERR: #28 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #29 0x7f81d2b48851 mojo::Connector::DispatchMessage() STDERR: #30 0x7f81d2b492e7 mojo::Connector::ReadAllAvailableMessages() STDERR: #31 0x7f81d2b490a3 mojo::Connector::OnHandleReadyInternal() STDERR: #32 0x7f81d2b498f0 mojo::SimpleWatcher::DiscardReadyState() STDERR: #33 0x7f81d2d0e67a mojo::SimpleWatcher::OnHandleReady() STDERR: #34 0x7f81d2d0eb2b base::internal::Invoker<>::RunOnce() STDERR: #35 0x7f81d397f85b base::TaskAnnotator::RunTask() STDERR: #36 0x7f81d399a04c base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl() STDERR: #37 0x7f81d3999c78 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork() STDERR: #38 0x7f81d391fe64 base::MessagePumpDefault::Run() STDERR: #39 0x7f81d399a8dc base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run() STDERR: #40 0x7f81d395ae55 base::RunLoop::Run() STDERR: #41 0x7f81d39c87f2 base::Thread::Run() Original change's description: > Reland "Replace sigslot usages with robocaller library." > > This is a reland of 40261c3 > > Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError > added a new member with a different name and used it in webrtc code. > After this change do two more follow up CLs to completely remove the old code > from google3. > > Original change's description: > > Replace sigslot usages with robocaller library. > > > > - Replace all the top level signals from jsep_transport_controller. > > - There are still sigslot usages in this file so keep the inheritance > > and that is the reason for not having a binary size gain in this CL. > > > > Bug: webrtc:11943 > > Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540 > > Commit-Queue: Lahiru Ginnaliya Gamathige <[email protected]> > > Reviewed-by: Mirko Bonadei <[email protected]> > > Reviewed-by: Karl Wiberg <[email protected]> > > Cr-Commit-Position: refs/heads/master@{#32321} > > Bug: webrtc:11943 > Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946 > Commit-Queue: Lahiru Ginnaliya Gamathige <[email protected]> > Reviewed-by: Karl Wiberg <[email protected]> > Cr-Commit-Position: refs/heads/master@{#32359} [email protected],[email protected],[email protected] Change-Id: I6bce1775d10758ac4a9d05b855f473dd70bd9815 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11943 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187487 Reviewed-by: Guido Urdaneta <[email protected]> Commit-Queue: Guido Urdaneta <[email protected]> Cr-Commit-Position: refs/heads/master@{#32372}
JamesTerm
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Apr 15, 2021
The AdaptiveAgc often boosts the signal outside of Float S16 range. It is expected, which is why we have a limiter after it in the process chain. But it turns out that this happens regularly even for simple input examples. The output signal peaks can be as high as +4 dBFs for a single speaker example (which should be easy). It leads to excessive gain modulation by the limiter. This CL is a new tuning designed to produce a safer gain. After this, we shouldn't hit the saturation region of the limiter as often. But we will still maintain a high gain. We have a 'configurable kill-switch': the settings can be changed via field trials WebRTC-Audio-Agc2Force(Initial|Extra)SaturationMargin. Difference from the CL reviewed in https://webrtc-review.googlesource.com/c/102922: * Rename GN dependency '//system_wrappers:field_trial' to '//system_wrappers:field_trial_api'. * Add GN dependencies '//system_wrappers:field_trial_default' to binaries apm_vad, transient_suppression_test, click_annotate. TBR: [email protected] (cherry picked from commit 4bb1e4a) Bug: webrtc:7494, chromium:892043 Change-Id: I5014377050c74c32ae8998282991141eae31cf58 Reviewed-on: https://webrtc-review.googlesource.com/c/102922 Commit-Queue: Alex Loiko <[email protected]> Reviewed-by: Sam Zackrisson <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#25006} Reviewed-on: https://webrtc-review.googlesource.com/c/104762 Cr-Commit-Position: refs/branch-heads/70@{open-webrtc-toolkit#16} Cr-Branched-From: f18b352-refs/heads/master@{#24472}
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Discovered this problem while debugging today.