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}
The custom callback is intended to override errors, so there's no
point in calling it if the status is ok.
Calling it during an otherwise successful verification was an
unintentional change from:
https://webrtc-review.googlesource.com/c/src/+/196941
This is misleading as the return value isn't even used.
Bug: chromium:1247577
Change-Id: Id74411f7364537a3225021e7631bc9ab962889ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231881
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34994}
Call SetMaxBitrate when encoder is configured instead of in OnMaybeEncodeFrame (which is called after the initial frame dropping ->
max bitrate is not set for dropped frames).
Added support for single active stream configuration.
Bug: none
Change-Id: I33ff96e7feed70b9ea3c9b3da89f117859108347
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231681
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34973}
With intent to replace BitBuffer.
This version is optimised for binary size and readability of the parsers
at the cost of slower parsing of invalid bitstreams
Bug: None
Change-Id: Ib054e2a7758b9a69cbf2559e739465b104a7dcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230244
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34886}
The Windows.Graphics.Capture API CreateForMonitor has a bug that was
fixed in 20H1 that causes an exception to be thrown when an HMONITOR
with a value of 0 is provided. This is a valid input used to request
capture of all monitors. To avoid this issue, we can restrict screen
capture using WGC to versions of Windows >=20H1.
Bug: webrtc:13078
Change-Id: Ia66bf2b2738c29813d41e214fdfc1eb96e0a1312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229140
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#34878}
This patch adds support for manually setting subnets that
should be handled as VPN, i.e be subject to VpnPreference,
in case webrtc fails to auto-detect VPNs.
Bug: webrtc:13097
Change-Id: I42514f0677a35cfe30ad053570fa9c2a5b4a856b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230122
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34852}
Also change return type of FinalRefCountedObject::Release() to
RefCountReleaseStatus, for consistency with other refcount classes.
Bug: webrtc:12701
Change-Id: I37c325e78ba7ae3e220b618da02cb243604ca4cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229590
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34849}
This patch adds a vp preference field to RTCConfig.
DEFAULT, // No VPN preference.
ONLY_USE_VPN, // only use VPN connections.
NEVER_USE_VPN, // never use VPN connections
PREFER_VPN, // use a VPN connection if possible, i.e VPN connections sorts higher than all other connections.
AVOID_VPN, // only use VPN if there is no other connections, i.e VPN connections sorts last.
Bug: webrtc:13097
Change-Id: I3f95bdfa9134e082c7d389f803bd08facfb70262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229591
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34842}
This change does not affect downstream dependencies as androidx.annotation
is fully compatible with android.support.annotation.
Bug: webrtc:11962
Change-Id: I714b473df8d0fee8000ddf3a9beca7c5613db5ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226881
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34839}
This patch adds small cost to Vpn connections
so that a "raw" connection identical to a vpn connection
will be chosen first.
The feature is gated by a field trial WebRTC-AddNetworkCostToVpn
for safe roll out.
Bug: webrtc:13097
Change-Id: I4ad40fa00780a6d7f89cacf6f85f3db4ecd0988c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229585
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34822}
Don't DCHECK that network send is successful, it may fail, e.g., EPIPE
if remote end has disconnected.
Bug: None
Change-Id: I7ccff072420498b60fe16598110da91b01bfe7cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229384
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34821}
I meant to do this with the Chromium change but forgot. UMA registers
zero uses of 3DES, so this should be safe. (Not too surprising, since
3DES had already been obsolete for just under a decade by the time
WebRTC existed.)
Bug: chromium:1203442
Change-Id: I5bddd2bd3f24beb486c8246fa5dab5836883b8c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: David Benjamin <davidben@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34806}
::testing::GTEST_FLAG is deprecated and it will be removed in future
versions of gtest.
Bug: None
Change-Id: Icb2ad2a7607073cf9ad63753a4de34f37bda411c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227083
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34600}
This CL removes an unused ScopedMessageData ctor and introduces
ScopedMessageData::Release which is the first step in order to remove
the data() methods that return a reference to a std::unique_ptr (which
is an anti-pattern).
Bug: None
Change-Id: I8f3c3fcfebd127c07fe0b667ca3442a20f458f0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226870
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34563}
Acts as a compile time annotation, with corresponding run-time check
only when DCHECKs are enabled, and built using absl or pthreads mutexes.
Bug: None
Change-Id: Ie044c1ea1e576df71d634301f7df9d75cdf10b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226328
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34555}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).
Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
This is a reland of 9f2a20f4342a3e86e1f9fdfe6f3d76fb539d41c2
See https://webrtc-review.googlesource.com/c/src/+/226563/1..2
for the fix. RTC_DCHECK_ALWAYS_ON needs to be in public_configs
in order to be propagated together with header #includes and
avoid ODR violations.
Original change's description:
> Add WebRTC specific dcheck_always_on.
>
> Inspired by V8 CL: crrev.com/c/3038528.
>
> This makes the WebRTC's dcheck control independent of Chromium's and
> prepares switching Chromium's default behavior without affecting
> WebRTC developers or builders.
>
> Preparation for: https://crrev.com/c/2893204
>
> Bug: chromium:1225701, webrtc:12988
> Change-Id: Ia0d21f9fb8e9d7704fd1beca16504c301a263b3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226465
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Dirk Pranke <dpranke@google.com>
> Cr-Commit-Position: refs/heads/master@{#34512}
Bug: chromium:1225701, webrtc:12988
Change-Id: I1f78587487ee7b1a4a07b8c91b737a9e797b2323
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226563
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34519}
