This change creates trace events with a single parameter
composed of ClassName::Method.
The change additionally causes the duration of the proxy call to be
traced, not only the occurrence.
Fixed: webrtc:12787
Change-Id: I1689862318d4c6fc1dcef343c3ccf3ae9f7e17df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219788
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34149}
This change prepares for a later change in Chromium that makes it
stop depending on headers exposed by WebRTC that require inclusion of
api/proxy.h.
No-Try because of lack of infra lack of capacity on macs.
No-Try: True
Bug: webrtc:12787
Change-Id: I628424fe49e873027595b80336be2b821c22245e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219688
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34072}
The channel classes have stored the negotiated headers but a more
natural place to store them is in the RtpTransceiver class where
RtpHeaderExtension state is managed as well as the implementation of
the only method that depends on the stored state,
HeaderExtensionsNegotiated().
Also adding a TODO for further improvements where we're unnecessarily
storing state in the channel classes for the purposes of the transports.
Bug: webrtc:12726
Change-Id: If36668e3e49782ddeada23ebed126ee2c4935b8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216691
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33917}
This changes the notification to a single std::function pointer
instead of being a sigslot::signal1<> collection.
Summary:
* Remove SignalFirstPacketReceived_, the last sigslot member variable.
(still inherits from sigslot::has_slots<>)
* BaseChannel doesn't post to the signaling thread anymore. The only
reason that remains for the signaling_thread_ variable, is for
thread checking.
* Remove BaseChannel's reliance on MessageHandlerAutoCleanup
(still inherits from MessageHandler)
RtpTransceiver is the consumer of this event. That class is also the
class that sits between the PC classes and the channel object, holding
a pointer to the channel and managing calls that come in on the
signaling thread, such as SetChannel. The responsibility of delivering
the first packet received on the signaling thread is now with
RtpTransceiver:
* RtpTransceiver always requires a ChannelManager instance. Previously
this variable was sometimes set, but it's now required.
* Updated tests in rtp_transceiver_unittest.cc to include a
ChannelManager as well as fix them to include call expectations for
mock sender and receivers.
Bug: webrtc:11993, webrtc:11988
Change-Id: If49d6be157cd7599fa6fe3a42cd0a363464e3a74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215979
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33853}
This is a reland of a37f2bd9421868e222d591d3371486a6ab939fd6
Original change's description:
> Rename SIGNALING and WORKER to PRIMARY and SECONDARY
>
> This makes the proxy macros less confusing when the secondary thread
> is sometimes the worker thread, sometimes the networking thread.
>
> Bug: none
> Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33346}
Bug: none
Change-Id: If46a6679ac0fc947797dd7be87626ef7702faca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208845
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33349}
This makes the proxy macros less confusing when the secondary thread
is sometimes the worker thread, sometimes the networking thread.
Bug: none
Change-Id: I1a8cebb82d09be44fe40e80c861bcfb47b9928e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208763
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33346}
at the cost of adding a WeakPointerFactory.
Moves the RtpTransceiver "NegotiationNeeded" signal to a callback
function that is passed as a constructor argument.
Bug: webrtc:11943
Change-Id: I37b2027379acce38dbaf0f396daebdb3e579ee54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192540
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32575}
This bypasses the proxy for the following properties:
* MediaStream::id()
* AudioTrack::kind() and AudioTrack::id()
* VideoTrack::kind() and VideoTrack::id()
* RtpReceiver::media_type() and RtpReceiver::id()
* RtpSender::media_type() and RtpSender::id()
* VideoTrackSource::remote() and VideoTrackSource::is_screencast()
* RtpTransceiver::media_type()
Bug: webrtc:11923
Change-Id: If7edea1781f778af3775515fc4af9a9e151c8103
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183767
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32071}
Also add an unit test for RtpTransceiver under Unified Plan, and
refactor so that we no longer use StopInternal() internally.
This will make removing it easier.
Bug: chromium:980879
Change-Id: I46219112e3aba8e7513c08336b10e95b1ea5d68b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182681
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31999}
Also moves implementation of legacy setDirection() without error to the
api/ directory.
This is one step in the plan for changing the API
to return RTCError.
Bug: chromium:980879
Change-Id: Ibce8edf8e3c6d41de7ce49d2ffc33f5b282a0e9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181520
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31943}
This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.
Note: SDP negotiation is not modified by this change.
Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
indicating either kStopped (extension available but not signalled),
or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
default value of the attribute comes from the voice and video
engines as before.
https://chromestatus.com/feature/5680189201711104.
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30800}
