Add an utility function to TaskQueueForTest to execute all already
posted tasks on the queue.
Bug: webrtc:11380
Change-Id: I6cf75bc543cfd2dd1c363935134d3f7bd55eec58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174140
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31152}
https://webrtc-review.googlesource.com/c/src/+/172847
------------ original description --------------
Preparation for ReceiveStatisticsProxy lock reduction.
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).
Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.
One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.
Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.
Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
Bug: webrtc:11489
Change-Id: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
This change fixes declarations that have initial values but are
technically not definitions by marking them constexpr (which counts as a
definition).
Bug: None
Change-Id: Icbecf8d83faffa83b9f7e1ffe4d6ef3a3f0b0c2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173587
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31073}
These static functions were marked as deprecated and since they
are not used this CL just removes them.
Bug: webrtc:10198
Change-Id: I4872e31701543c988fe71ab4e0b32bd73ff07753
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173467
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31057}
When epoll is enabled in the PhysicalSocketServer, a socket may
not get registered for its epoll events. If an AsyncSocket is
closed and re-created during one of its signal callbacks, its
old epoll events and new epolls events bitmasks may be the same,
even though the fd has changed. This causes the epoll implementation
to not register the new fd for any events.
Fix this by resetting the saved events bitmask when the socket is
closed. This ensures the new fd, if any, is registered if needed.
Bug: webrtc:11497
Change-Id: Idea499e09aefdf292430d1a774a046f963603b95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173103
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31039}
This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.
Reason for revert: Speculative revert: breaks downstream project
Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
>
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
>
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
>
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
>
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
>
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
>
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}
TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).
Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.
One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.
Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.
Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
This patch is a follow up to https://webrtc-review.googlesource.com/c/src/+/172582
and change so that a switch from CELLULAR_X to CELLULAR_Y does not
trigger OnNetworkChange.
This is needed as the OnNetworkChange signals triggers
BasicPortAllocator to rescan all networks and generate new candidates.
The actual adapter type change is still possible to react on using
SignalTypeChanged.
BUG: webrtc:11473
Change-Id: Icc1a945b8a4df1714c6ec4b02ec759ecada92d7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172802
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30992}
This patch adds new enum values for different types of cellular
connections.
The new costs are currently blocked when sending to remote,
(so that arbitrary network switches does not starts occurring).
The end-game for this series to be able to distinguish between
different type of cellular connections in the ice-layer (e.g when
selecting/switching connections).
BUG: webrtc:11473
Change-Id: I587ac8fdff4f6cdd0f8905f327232f58818db4f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172582
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30970}
This reverts commit 7e91482fcc496103f36333a569992c81b6dc9e9c.
Reason for revert: Speculative revert, as Android FYI bots are red
starting https://webrtc.googlesource.com/src/+/7e91482fcc496103f36333a569992c81b6dc9e9c
where this CL landed.
See also https://bugs.chromium.org/p/chromium/issues/detail?id=1065805.
Original change's description:
> Add interface_id to rtc::Network
>
> This patch adds an interface_id property
> to rtc::Network. It is an enumeration of the
> interface names that are present.
>
> This enables a local ICE agent to keep track
> of which connections are using which interfaces,
> something that is useful for predicting how
> connections behave.
>
> This is part 1 of https://webrtc-review.googlesource.com/c/src/+/85520
>
> Bug: webrtc:9446
> Change-Id: Ia6ec1f14ac240799fb1be49d67d82e2733e87acf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171061
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30882}
No-Presubmit: True
Bug: webrtc:9446
TBR=hta@webrtc.org, jonaso@webrtc.org
Change-Id: If86e2e0653b53a8eae26a97ce9fa68748b440607
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172092
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30937}
This patch adds an interface_id property
to rtc::Network. It is an enumeration of the
interface names that are present.
This enables a local ICE agent to keep track
of which connections are using which interfaces,
something that is useful for predicting how
connections behave.
This is part 1 of https://webrtc-review.googlesource.com/c/src/+/85520
BUG: webrtc:9446
Change-Id: Ia6ec1f14ac240799fb1be49d67d82e2733e87acf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171061
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30882}
this patch is a followup to https://webrtc-review.googlesource.com/c/src/+/170628
and removed the now deprecated fields {local/remote}_network_id that
is now no longer used by downstream.
BUG: webrtc:11434
Change-Id: Ia322609c0b4f07b05b8592cbca7f001a115da109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171515
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30874}
This patch extends the NetworkRoute struct with more information
about local/remote endpoints. It adds
- adapter type
- adapter id
- relay
(previously it was "only" network_id)
The patch leaves the {local/remote}_network_id fields
around and populated since downstream projects depend
on them. They will be removed once they have migrated.
OWNER: srte@ call/ test/
OWNER: asapersson@ video/
OWNER: hta@ p2p/ pc/ rtc_base/
BUG: webrtc:11434
Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30848}
This is a reland of de86381161651816c078adeb354902b15d03a35b
Original change's description:
> Leverage dispatch_queue_create_with_target when possible.
>
> Replacing dispatch_queue_create followed by
> dispatch_set_target_queue with dispatch_queue_create_with_target
> is claimed to be source of GCD performance improvement:
> https://developer.apple.com/videos/play/wwdc2017/706/
> Video since 40 min. Slides since 199.
>
> Bug: webrtc:9055
> Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#30781}
Bug: webrtc:9055
Change-Id: I36b0b6423c81c0497f66f7c993741c33ff6ec5ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30821}
Replacing dispatch_queue_create followed by
dispatch_set_target_queue with dispatch_queue_create_with_target
is claimed to be source of GCD performance improvement:
https://developer.apple.com/videos/play/wwdc2017/706/
Video since 40 min. Slides since 199.
Bug: webrtc:9055
Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/master@{#30781}
This reverts commit af1f8655b2cb69af382396ea642eb0a2bf04bb4d
Landing the change with default set to
"enabled" (DTLS 1.0 will continue to work by default),
so that flipping the default can be a separate CL.
Original change's description:
> Revert "Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC."
>
> This reverts commit 7276b974b78ea4f409d8738b1b6f1515f7a8968e.
>
> Reason for revert: Changing to a later Chrome release.
>
> Original change's description:
> > Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
> >
> > This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> > is part of a larger effort at Google to remove old TLS protocols:
> > https://security.googleblog.com/2018/10/modernizing-transport-security.html
> >
> > For the M74 timeline I have added a disabled by default field trial
> > WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> > as consumers move away from these legacy cipher protocols but it will be off
> > in Chrome.
> >
> > This is compliant with the webrtc-security-arch specification which states:
> >
> > All Implementations MUST implement DTLS 1.2 with the
> > TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
> > curve [FIPS186]. Earlier drafts of this specification required DTLS
> > 1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
> > at the time of this writing some implementations do not support DTLS
> > 1.2; endpoints which support only DTLS 1.2 might encounter
> > interoperability issues. The DTLS-SRTP protection profile
> > SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
> > Implementations MUST favor cipher suites which support (Perfect
> > Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
> > over non-AEAD cipher suites.
> >
> > Bug: webrtc:10261
> > Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> > Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: David Benjamin <davidben@webrtc.org>
> > Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27006}
>
> TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10261
> Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27403}
Bug: webrtc:10261
Change-Id: I28c6819d37665976e396df280b4abf48fb91d533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169851
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30733}
This only changes the comments and rename variables.
Bug: chromium:1054403
Change-Id: Ie7419ca23e482361e9f90405587b8c8f839b26d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169101
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30710}
Adding :: before rtc allow us to use the macro in nested rtc namespace for external components like
namespace xxxxxxx {
namespace rtc {
RTC_CHECK(true);
}
}
Bug: webrtc:11400
Change-Id: I79349b847c3fce8197c82aec31b672a1a16e5388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169683
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30684}
Selling point is that it never touches the heap. Intended use case is
cheaply returning a variable, bounded, and small number of things from
a function.
Specifically, there are situations where we'd like to return things like
ArrayView<ArrayView<float>>
where we currently have to allocate an array of ArrayView<float> for
the outer ArrayView to point to, which is a bother; however, although
the outer ArrayView is of variable size, that size is statically
guaranteed to not exceed some small constant. After this CL, we'll be
able to instead return
BoundedInlineVector<ArrayView<float>, kSmallConstant>
which is much more convenient. We already had the option of returning e.g.
std::vector<ArrayView<float>>
but that would bloat our binary with code to handle heap allocations
in places we'd rather be lean and mean.
https://godbolt.org/z/r-vcPj demonstrates that the overhead compared to
a raw C array + a size is ~zero.
Bug: webrtc:11391
Change-Id: Ifb6d937193052588be641aa62cc67ba0ec64ded6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168944
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30663}
This avoid duplication. As part of this moving the overhead calculation
to the IP address class so it's easier to find and more natural to use.
Bug: webrtc:9883
Change-Id: If4d865f445bc1a302572896932966ce30294e339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169445
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30657}
Move definition of AlignedArray to the only code using it, the
test-only LappedTransform class, and delete unused methods.
Bug: webrtc:6424, webrtc:9577
Change-Id: I1bb5f57400f7217345b7ec7376235ad4c4bae858
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168701
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30576}
This is a follow-up of
https://webrtc-review.googlesource.com/c/src/+/168403, removing code
that was only used from the now-deleted code.
The if.h include is still needed for some IFF_xxx flags, but can be
simplified to use the POSIX version as route.h isn't needed anymore.
Bug: None
Change-Id: Ic2def8b54a9d3aa1a0e3eabf6f1a837a0cf8a5a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168483
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@google.com>
Cr-Commit-Position: refs/heads/master@{#30523}
With current congestion window pushback, when congestion window is filling up, it will reduce bitrate directly and encoder may reduce encode quality, resolution, or framerate to adapt to the allocated bitrate, the behavior is depending on the degradation preference.
This change enable congestion window to only drop frames to reduce bitrate (when needed) instead of reduce general bitrate allocation.
Bug: webrtc:11334
Change-Id: I9cf5c20a0858c4d07d006942abe72aa5e1f7cb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168059
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30483}
This CL only includes the necessary changes in PhysicalSocketServer,
and doesn't include the Java or Objective C API.
Note that this is doing exactly the same thing as UDPSocketPosix
in chromium.
BUG=webrtc:5658
Change-Id: I295455eaccba2a83cdd1bc55848f325c310f8d32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168260
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30478}
