This restricts the interface such that only a single onclose handler
can be set and that only one OnClose() notification will be fired.
That behavior is the same as how the previous sigslot was being
used, but the difference is that, in addition to removing sigslot,
this pattern is now more explicitly checked in the design.
Bug: webrtc:11943
Change-Id: I469c8cab3d62544988c8145b326af60b06b76d8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343340
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41920}
The class is now only used by one test class.
Bug: none
Change-Id: Ib7714469254bd507d027385d2825b1c14bd63c94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343123
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41917}
TRACE_EVENT is already scoped!
#rtc_fixit
Tested: Compiled the patch in Chromium and confirmed the Proxy events are still present. I can send the resulting trace to reviewers if desired.
Bug: webrtc:15867
Change-Id: I5717a85c0ee25e8e20123afa08064c9b6666ba96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41916}
This reverts commit 1ee24a650c116509d855e2ed23b8d28a0bb37384.
Reason for revert: Suspected upstream test breakage.
Original change's description:
> Propagate known Encoder SinkWants when configured instead of after first frame.
>
> Propagate requested resolution and max frame rate to the source when
> configured rather than after the first frame.
> This is so that the source can be configured immediately. There is no
> reason why source should be updated until after first frame since it may lead
> to unnecessary reconfigurations and thread jumps. Wants that depend on actual frame size is not moved.
>
> Cl also change default behaviour in VideoStreamEncoderTest to not
> set restriction on max frame rate. This aligns with how its used.
>
> Bug: webrtc:14451
> Change-Id: I96a3675d3ccabb1d2ecb4354b6932bc6563b1760
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342801
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41906}
Bug: webrtc:14451
Change-Id: I3aa669f8cc61a43b0820a06edf1497f3c86e3958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343220
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41911}
And thus require Environment to be propagated to this test helper
Bug: webrtc:15860
Change-Id: Ia4796d7a6a8e6f5dcb947899617df43e991419e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343181
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41910}
Keeping the old setting for the total queue size
limit, which avoids breaking a downstream.
This reverts commit 47ce449afaf9ba38785437fdd338630cad24a77b
and relands commit 4c990e2e56157175324e651f95f3d8c6a0e5c030.
Bug: chromium:40072842
Change-Id: I1e7d14b5d0026232d1fc9277172b6947b8be3490
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343120
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41907}
Propagate requested resolution and max frame rate to the source when
configured rather than after the first frame.
This is so that the source can be configured immediately. There is no
reason why source should be updated until after first frame since it may lead
to unnecessary reconfigurations and thread jumps. Wants that depend on actual frame size is not moved.
Cl also change default behaviour in VideoStreamEncoderTest to not
set restriction on max frame rate. This aligns with how its used.
Bug: webrtc:14451
Change-Id: I96a3675d3ccabb1d2ecb4354b6932bc6563b1760
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342801
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41906}
When an error occurs, the callback needs to be invoked or the
signaling thread may block indefinitely waiting for it.
Bug: webrtc:15871
Change-Id: Ib73382aff07b3632794300985223c70c24f554f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342901
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41904}
This reverts commit 4c990e2e56157175324e651f95f3d8c6a0e5c030.
Reason for revert: Breaks downstream build.
Original change's description:
> dcsctp: Add per-stream-limit, refactor limits.
>
> The limits have been moved out from the Send Queue as they were enforced
> outside the queue anyway (in the socket). That was a preparation for
> adding even more limits; There is now also a per-stream limit, allowing
> individual streams to have one (global) limit, and the entire socket to
> have another limit.
>
> These limits are very small in the default options. In Chrome, the limit
> is 16MB per stream, so expect the defaults to be updated when the
> additional buffering outside dcSCTP is removed.
>
> Bug: chromium:41221056
> Change-Id: I9f835be05d349cbfce3e9235d34b5ea0e2fe87d1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342481
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Commit-Queue: Victor Boivie <boivie@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41895}
Bug: chromium:41221056
Change-Id: Icd57fbfca87d6b512cfc7f7682ae709000c2bcad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343080
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41901}
Webrtc is build with FFmpeg sources on defined in the include path
through the -I flag, so they should be included this way instead. This
would otherwise cause a conflict when the chromium ffmpeg sources move
from third_party/ffmpeg/* to third_party/ffmpeg/src/*
BUG: chromium:329282834
Change-Id: Id8f7e91446bdc536db77e74388a73e51f5111ffc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342820
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ted (Chromium) Meyer <tmathmeyer@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41899}
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.
Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.
This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.
This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.
Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
This seems to confuse perfetto, and the data ends up on its own track
and the end event is just ignored. As it was invalid, I am assuming it
is not used, and can be simply removed.
#rtc_fixit
Bug: webrtc:15867
Change-Id: I31a814f6c2147c3ce534726bf9046a79369b9eb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342761
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41896}
The limits have been moved out from the Send Queue as they were enforced
outside the queue anyway (in the socket). That was a preparation for
adding even more limits; There is now also a per-stream limit, allowing
individual streams to have one (global) limit, and the entire socket to
have another limit.
These limits are very small in the default options. In Chrome, the limit
is 16MB per stream, so expect the defaults to be updated when the
additional buffering outside dcSCTP is removed.
Bug: chromium:41221056
Change-Id: I9f835be05d349cbfce3e9235d34b5ea0e2fe87d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342481
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41895}
This seems to confuse perfetto, and the data ends up on its own track
and the end event is just ignored. As it was invalid, I am assuming it
is not used, and can be simply removed.
#rtc_fixit
Bug: webrtc:15867
Change-Id: I77e59adcd35c51911474446a5f92505bf6b860f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342780
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41892}
Enabling Socket options on keep-alive related function that may enable clients to detect any stale connection early on.
Bug: webrtc:15866
Change-Id: Ib4f15e0c933aeb6cf4fd18ff8cc708d118ea8645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342223
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41888}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
Two new socket options are introduced OPT_SEND_ECN used for setting ECN bits. OPT_RECV_ECN used for reading the ECN bits.
If ECN bits are set on received IP packets, ECT(1) and CE is propagated via rtc::ReceivedPacket.
Bug: webrtc:15368
Change-Id: I3ac335007e2f7d30564569bbc80ce47fa541bef1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332380
Reviewed-by: Jonas Oreland <jonaso@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41885}
These lines cause an error when building a project with libwebrtc as a dependency in Microsoft Visual Studio.
Bug: webrtc:15864
Change-Id: I1abfe257d0ea1c16c4c5b718594e8085036f7763
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342320
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41881}
Before this CL, the StreamId class represented either a valid SCTP
stream ID, or "nothing", which means that it was a wrapped
absl::optional. Since created data channels don't have a SCTP stream ID
until it's known whether this peer will use odd or even numbers, the
"nothing" value was used for that state.
This unfortunately made it a bit hard to work with objects of this type,
as one always had to check if it contained a value. And even if a caller
would check this, and then pass the StreamId to a different function,
that function would have to do the check itself (often as a RTC_DCHECK)
since the passed StreamId always could have that state.
This CL simply extracts the "absl::optional" part of it, forcing holders
to wrap it in an optional type - when it can be "nothing". But allowing
the other code to just pass StreamId that can't be "nothing". That
simplifies the code a bit, potentially removing some bugs.
Bug: chromium:41221056
Change-Id: I93104cdd5d2f5fc1dbeb9d9dfc4cf361f11a9d68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342440
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41880}
This reverts commit ed8390d21a7b15091d01bc8e843193d0a6efd23a.
Reason for revert: Fix has landed in chrome, ready to reland.
Original change's description:
> Revert "Deprecate old constructors and set_type() in Candidate and Port"
>
> This reverts commit aaa6851d53741179a591d79fc82c4dd6651a7ba5.
>
> Reason for revert: breaks chromium webrtc import
>
> Original change's description:
> > Deprecate old constructors and set_type() in Candidate and Port
> >
> > * Deprecates constructors that use string based `type`
> > * Deprecates string based type functions in favor of enum based.
> > * Restrict possible values of Candidate::type. Ensure a valid value
> > is assigned at construction.
> > * Make Port constructors protected to limit their use to subclasses.
> > - The reason for this is to make sure that use of SharedSocket()
> > is controlled (it adds a bit of complexity).
> > * Simplify construction of Port (remove Construct() etc)
> >
> > Bug: webrtc:15846
> > Change-Id: If24ed674e175642efa49da37fd2bc847dd14f613
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339860
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41865}
>
> Bug: webrtc:15846
> Change-Id: Ic8b7cba97f8fb207ef51a88900e704658ade28b7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342140
> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#41867}
Bug: webrtc:15846
Change-Id: I3d52643bbb537d1c072643528828d26eb18fea94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342200
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41875}
