Also updated the OperationsChain and CallbackHandle classes to not use
any virtual methods.
Bug: webrtc:13464
Change-Id: I3437d1b7b043339e66411f5a46c226624b7ff9a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246102
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35682}
Update RtpPacketInfos internals to use rtc::make_ref_counted, and a
Data class with no virtual methods.
Bug: webrtc:13464, webrtc:12701
Change-Id: I03f6bee69a9f060dcf287284fc779268d5eb433e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244505
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35660}
Replaced with a constructor with a SocketFactory argument.
Bug: webrtc:13145
Change-Id: I30db4ad089009284e1be8a6bbdadd5a671e93713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35508}
Current rate statistic tracker has assumption, the tracking window will
always be full after first filled up. This assumption looks not always
true. One example is the input_framerate_ tracker inside
video_stream_sender.cc which is used for setup frame droper and encoder.
Whenever there is a gap in video stream, like mute/unmute,
pacer pause/unpause etc. The fps detected from the rate_statistics
becomes samples_filled_partial_window / full_window_size, which could
be extremely low for a while. This creates a misalignment between the
fps we told encoder/frame dropper, and the real fps we fed into them,
which causes short-term serious overshot and very bad experience on
delay, avsync, congestion etc. This may also depends on how fast
encoder could react to the gap between set fps and real fps, but
libvpx and openh264 at least cannot handle this well.
So propose a fix to update first timestamp after tracker window
drained. This will give more accurate fps estimate similar based on
active window after sample gets drained
Bug: webrtc:13403
Change-Id: I96792c11091fe8bfa63e669f4360a3b3e95593e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35447}
In logging.cc, use the pointer of the static variable so that
it doesn't need a global constructor/exit time destructor.
In RTCFieldTrials.mm, store the field trial string as a char pointer
instead of a std::unique_ptr to ensure that it is never freed.
LSAN will be unhappy with this fix, but WebRTC itself hasn't been
tested with LSAN enabled, and any code changed in this CL does not
build with build_with_chromium=true, so it shouldn't be a problem.
Bug: webrtc:9693, webrtc:11665
Change-Id: Ia28e3534170e0817b815717f6efe862f7b51ef62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237320
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35391}
The method is not used so can be safely deleted before the full
(and eventual) removal of the implementation.
Bug: webrtc:12339
Change-Id: I7726313c46562041f670c3baec2db955de0b4298
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238141
Auto-Submit: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35375}
This emphasizes the "hint" to potential external users that the
class has been deprecated.
Bug: webrtc:12339
Change-Id: Iab83481af69a505059297cce959f02b5ab649f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237805
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35368}
The VSS is used in tests, usually from the signaling thread, but all the
network emulation happens in the network thread. TSAN will then complain
about variable access from different threads without any synchronization.
Bug: b/204654931
Change-Id: I164f5d73e559f00e6bf390ef5e5f112bcc58ce11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237784
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35357}
Use a fake task queue to test the RepeatingTask rather than a real
task queue, which removes the need for Sleep(). This fixes the flakiness
issues as the class is no deterministic.
BUG=webrtc:12808
Change-Id: I8c6a8535165b076f5fe6ec3e65ebcf7f07008737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237803
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35349}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This makes it not an error to bind a SequenceChecker before the
global TaskQueueBase::Current() is set.
Unbreaks the SDP integration fuzzer.
Bug: webrtc:13374
Change-Id: Ic4c23fa29f4598290cf9196550e5133ba753f44f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237620
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35341}
Change search for next separator to be linear in length of the string
(instead of potentially quadratic)
Reduce copying of std::string by switch to string_view
Throttle logging about unknown key.
Bug: b/204541739
Change-Id: I81d5cd4432966a0a5808077f9001bc62960e5e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237500
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35330}
The AsyncListenSocket::SetOption method then gets unused, and can be
deleted.
Bug: webrtc:13065
Change-Id: Idcf70a75b96036290fdceff6e0f96a8d5617f87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35302}
The main change is to remove sockets from epoll if there are no
requested events, which happens when a socket is considered closed
(due to an error or otherwise). This prevents a busy loop when a socket
is an error condition where it will constantly be signaled, but not
deleted by higher level code.
Other related changes:
* Set DE_CLOSE on errors regardless of whether the socket is readable or
writable.
* Don't set DE_ACCEPT on errors.
* Handle getsockopt(SO_ERROR) errors.
* In IsDescriptorClosed:
* Retry recv on EINTR.
* Treat ECONNABORTED and EPIPE as errors.
Original patch contributed by andrey.semashev@gmail.com.
Bug: webrtc:11124
Change-Id: I67f33213efc1418b1ffc8f4867f606b7f8dc4ece
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235863
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35300}
This is a reland of b141c162ee2ef88a7498ba8cb8bc852287f93ad2
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
Bug: webrtc:13065
Change-Id: I88bebdd80ebe6bcf6ac635023924d79fbfb76813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35260}
Only affects turn server. Refactored to wrap sockets with SSLAdapter
after Accept, using the SSLAdapterFactory to hold needed configuration.
Bug: webrtc:13065
Change-Id: I5df65aad5728d8d40d95b22db6398a573ec7a36f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235823
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35258}
This reverts commit b141c162ee2ef88a7498ba8cb8bc852287f93ad2.
Reason for revert: Breaking WebRTC rolls. See https://ci.chromium.org/ui/b/8832847811929676465 for an example failed build.
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13065
Change-Id: Id5d5b35cb21704ca4e3006caf1636906df062609
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235824
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35249}
Since `PostTaskToGlobalQueue` is somewhat different from
other TaskQueue APIs, there are concerns that it should not be
a public API.
Remove this from task_queue_gcd.h and make it a private static function
of AsyncResolver.
Bug: webrtc:13237
Change-Id: Ib4aff296f894a4f3b051969d176369e13a10088f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234900
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35236}
The advantage is that GCD maintains the internal thread pool and
spawns threads when needed. I would expect the behavior to be
almost identical to creating a thread using PlatformThread.
Bug: webrtc:13237
Change-Id: Ie4406b5d128c244f66a73830d5a27f2d8fd88549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35165}
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.
Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
The |slice_qp_detla| reported by the hardware is not credible, which
causing the quality scaler cannot work properly,the resolution cannot
be adjusted correctly.
To fix this issue, this CL implements a bandwidth scaler which is used
for adjust resolution, this scaler will be used when QP based quality
scaler is not working due to untrusted QP reported by HW AVC encoder.
Bug: webrtc:12942
Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35120}
Used by QueryDefaultLocalAddress, instead of relying on the update
thread's associated socket server.
This is not the only use of rtc::Thread::socketserver() in the
BasicNetworkManager class. It also interacts with the thread's
socket server to call set_network_binder. That is unchanged by this cl,
perhaps those calls can be moved to the caller of StartNetworkMonitor and
StopNetworkMonitor.
Bug: webrtc:13145
Change-Id: If109c2dcb0e74b183e10bb3db7a5aefcc95d1a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232613
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35118}
It's useful for other parts of WebRTC and there is no real reason why
it should be located in net/dcsctp.
Bug: None
Change-Id: Iccaed4e943e21ddaea8603182d693114b2da9f6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232606
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35055}
Unlike ReadBits, ConsumeBits doesn't limit number of bits it may advance,
and thus should work when that number is close to the integer limit
Bug: chromium:1250730
Change-Id: Ia7847869ef9d3fc16450d572c9e2be6e1aa36741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232332
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35042}
GCC complains that explicit specialization in non-namespace scope
is happening for webrtc::BitstreamReader::Read(). However,
specializationvfor bool isn't used because std::is_unsigned<bool>::value
returns true. Add std::is_same for bool check and enable second
specialization only for bool types.
Bug: chromium:819294
Change-Id: I1873cd59e2737516bd4012fb952da65d6bf3172b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231561
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35007}
This patch adds VPN detection for windows
based on known MAC addresses.
- Cisco AnyConnect
- GlobalProtect Virtual Ethernet
Bug: webrtc:13097
Change-Id: Ia90ee50be0dc2dcd2e6e9de1493fdd2c5e7d9d3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230245
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34997}
