Earlier CLs assumed that the object pointed to by call_ had to be
accessed on the worker thread. While this is generally the case,
Call::MediaTransportChange is explicitly thread safe, so
PeerConnection::OnTransportChanged doesn't have to run on the worker
thread for that reason.
Which is fortunate, because it actually runs on the network thread.
The RTC_RUN_ON(worker_thread()) annotation on the method declaration
was ineffective because this method is being called via a base class
pointer; replacing it with a call to
RTC_DCHECK_RUN_ON(worker_thread()) in the function body immediately
triggered assertions in the unit tests.
Bug: webrtc:9987
Change-Id: I08cf558a74f4ca2b2eff8ef4810ebbd1287a9726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129442
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27287}
Plus all the annotations that were necessary to make things compile
again.
We needed a special twist for call_. The value it points to is owned
by the worker thread, but the signal thread needs to read the pointer.
We could have made the pointer const, except that we explicitly reset
it in the destructor (in an invoke to the worker thread).
Bug: webrtc:9987
Change-Id: I31f024547f4be0e50967133b0d452c80ae38d7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128863
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27278}
Plus all the annotations that were necessary to make things compile
again.
port_allocator_flags_ was accessed on both the signaling and the
network thread, but I was able to replace it with a return value.
Bug: webrtc:9987
Change-Id: Iab977a49d6588ce2240487475ec3588ae579caa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128772
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27254}
Metrics are added to measure:
1. The number of send encodings in calls to AddTransceiver.
2. The number of times that simulcast is disabled because there is no
support from remote peer.
3. The number of times simulcast is indicated in ApplyLocal and
ApplyRemote and with which API surface (no simulcast, legacy munging,
spec-compliant).
Bug: webrtc:10372
Change-Id: I84717a1911efdf8aaf43cd6c04c7f09fcf2c58f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125482
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26979}
Simulcast is disabled if the RIDs are not negotiated.
This change addresses the scenario in which RIDs are negotiated but
support for the RID extension is not negotiated.
In such cases, the RID extension cannot be used, so support for
simulcast should be turned off, as if RIDs were not negotiated.
A similar case can be made for MIDs, however MIDs are not explicitly
specified in simulcast. RIDs are only guaranteed to be unique within
a media section so it would seem that MIDs should be required.
However, applications supply RID values and can guarantee their
uniqueness, so unlike RIDs, the use of MIDs is not enforced as mandatory.
Bug: webrtc:10075
Change-Id: Ic1b27878ea152eaee43a38bbfda11144307766fe
Reviewed-on: https://webrtc-review.googlesource.com/c/125176
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26934}
We already support decoding of the x-mt line. This change adds the
a=x-mt line to the SDP offer. This is not a backward compatible change
for media transport (because of the changes in pre-shared key handling)
1) if media transport is enabled, and SDES is enabled, generate the
media transport offer.
2) if media transport generated the offer, add that offer to the x-mt
line.
3) in order to create media transport, require an x-mt line (backward incompatible).
The way it works is that
1) PeerConnection, on the offerer, asks jsep transport for the
configuration of the media transport.
2) Tentative media transport is created in JsepTransportController when
that happens.
3) SessionDescription will include configuration from this tentative
media transport.
4) When the LocalDescription is set on the offerer, the tentative media
transport is promoted to the real media transport.
Caveats:
- now we really only support MaxBundle. In the previous implementations,
two media transports were briefly created in some tests, and the second
one was destroyed shortly after instantiation.
- we, for now, enforce SDES. In the future, whether SDES is used will be
refactored out of the peer connection.
In the future (on the callee) we should ignore 'is_media_transport' setting. If
Offer contains x-mt, media transport should be used (if the factory is
present). However, we need to decide how to negotiate media transport
for data channels vs data transport for media (x-mt line at this point
doesn't differentiate the two, so we still need to use app setting).
This change also removes the negotation of pre-shared key from the
a=crypto line. Instead, media transport will have its own, 256bit key.
Such key should be transported in the x-mt line. This makes the code
much simpler, and simplifies the dependency / a=crypto lines parsing.
Also, adds a proper test for the connection re-offer (on both sides: callee and caller).
Before, it was possible that media transport could get recreated, based on the offer.
The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test.
This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even
when there is a re-offer.
Bug: webrtc:9719
Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01
Reviewed-on: https://webrtc-review.googlesource.com/c/125040
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26933}
This involves inserting an extra layer between jsep_transport_controller
and the cricket::SctpTransportInternal layer. The objects at this layer
are reference counted.
Bug: chromium:818643
Change-Id: Ibed57c4a538de981cee63e0f7f1f319f029cab39
Reviewed-on: https://webrtc-review.googlesource.com/c/123884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26889}
RtpSender uses a transactional model when getting and setting RtpParameters.
One must call GetParameters() and then can use the returned object in a
subsequent call to SetParameters().
PeerConnection was calling GetParameters() and SetParameters() during
negotiation in SetLocalDescription and SetRemoteDescription effectively
invalidating any parameters that the client previously held.
This change introduces an internal way for the platform to modify
parameters without invalidating the transactional model, provided that
the modification is not severe.
Ex. removing simulcast layers is a severe modification and will
invalidate any outstanding parameters.
Bug: webrtc:10339
Change-Id: I362e8ca4d9556e04a1aa7a3e74e2c275f8d16fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/124504
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26864}
This informs the media transport that PeerConnection wants to use a data channel
and gives it a chance to set up before the data channel sends the first message.
Bug: webrtc:9719
Change-Id: I6ea905a74b29b8735e77ac68bc8606e7bca77f18
Reviewed-on: https://webrtc-review.googlesource.com/c/124020
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26823}
Most of the implementation in rtp_sender.cc is a copy paste for both
Audio & Video RTP senders. This change moves all the common behavior
into the base RtpSenderInternal class.
Template method pattern is used to accomodate for the very slight differences
between audio and video senders.
Bug: None
Change-Id: I6d4e93cd32fbb0fb361fd0e1883791019bde9a92
Reviewed-on: https://webrtc-review.googlesource.com/c/123411
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26758}
Added a layer in RtpSender that bridges the gap between the layers
that the user sees and the layer that the media engine sees.
Media engine still maintains the invariant that the number of layers
cannot be changed, while RtpSender adds and removes layers between
the user GetParameters and SetParameters calls and the media engine.
Bug: webrtc:10251
Change-Id: I33839c1f9a9052cb6130253e5a582606f2cbe54a
Reviewed-on: https://webrtc-review.googlesource.com/c/122641
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26756}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: If7bbd5a468a8c50ac3cfe03cd2ed4f5b5f461195
Reviewed-on: https://webrtc-review.googlesource.com/c/123047
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26727}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: I68a51bfa3342c6d66d67276d5979144af34692c9
Reviewed-on: https://webrtc-review.googlesource.com/c/123046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26716}
Plus all the annotations that are necessary to make things compile
again.
(The latter could only be made const, since it is accessed from both
the signal thread and the worker thread.)
Bug: webrtc:9987
Change-Id: I49f1568d26a327f48cd94c2c70bc5939e9fbd59a
Reviewed-on: https://webrtc-review.googlesource.com/c/122842
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26715}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: I53caa3c6644c5d0fcb50f7c1e89e853eddd769a5
Reviewed-on: https://webrtc-review.googlesource.com/c/122882
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26705}
This is a reland of ce470aab518f067a67aa03aaab1fc61a45afa0ec
Original change's description:
> Enabling Simulcast use via AddTransceiver.
>
> This change removes the restriction on multiple send encodings when
> calling AddTransceiver. If RIDs are not provided in the
> simulcast scenario, they are auto-generated by the platform.
>
> This effectively enables the use of spec-compliant simulcast.
> Tests are also added to verify simulcast functionality.
>
> Bug: webrtc:10075
> Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2
> Reviewed-on: https://webrtc-review.googlesource.com/c/121389
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26590}
Bug: webrtc:10075
Change-Id: Ib4392e36d5d60b966ecff56d4df69bf60d13a73d
Reviewed-on: https://webrtc-review.googlesource.com/c/122962
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26694}
This change removes the restriction on multiple send encodings when
calling AddTransceiver. If RIDs are not provided in the
simulcast scenario, they are auto-generated by the platform.
This effectively enables the use of spec-compliant simulcast.
Tests are also added to verify simulcast functionality.
Bug: webrtc:10075
Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2
Reviewed-on: https://webrtc-review.googlesource.com/c/121389
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26590}
Support varies by codec, especially in the simulcast case, but using
the EncoderSimulcastProxy codec should fix this.
Bug: webrtc:10069
Change-Id: Idb6a5f400ffda1cdb139004f540961a9cf85d224
Reviewed-on: https://webrtc-review.googlesource.com/c/119400
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26449}
In the January 22nd 2019 WebRTC meeting it was agreed that an offer
for sending (or receiving) simulcast should only contain the RIDs
of the layers that are sent by the client.
This change removes the complexity that was added to support sending
and receiving the single layer (and RID) that are sent from the server.
Bug: webrtc:10076
Change-Id: I8bae1336d5cb8ba2f91c5b62332dc69e67ddfd47
Reviewed-on: https://webrtc-review.googlesource.com/c/120242
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26432}
CreateOffer and CreateAnswer will now examine the layers on the
transceiver to determine if multiple layers are requested (Simulcast).
In this scenario RIDs will be used in the layers (instead of SSRCs).
When the offer is created, only RIDs are signalled in the offer.
When the offer is set locally SetLocalDescription() SSRCs will be
generated for each layer by the Channel and sent downstream to the
MediaChannel.
The MediaChannel receives configuration that looks identical to that of
legacy simulcast, and should be able to integrate the streams correctly
regardless of how they were signalled.
Setting multiple layers on the transciever is still not supported
through the API.
Bug: webrtc:10075
Change-Id: Id4ad3637b87b68ef6ca7eec69166fee2d9dfa36f
Reviewed-on: https://webrtc-review.googlesource.com/c/119780
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26428}
Currently it's possible that no-op DTLS is created if media transport is only used for data channels.
Changing it so that no-op DTLS is only created when both media & data will flow through media transport.
Bug: webrtc:9719
Change-Id: I87f27fc778ea21b12f2904bad1452d893f66b541
Reviewed-on: https://webrtc-review.googlesource.com/c/119909
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26416}
UniqueIdGenerator classes are useful outside the pc directory.
This change moves them to the rtc_base directory to enable code
in all directories to reference them.
Bug: None
Change-Id: I1c77da87ea26d9611f37dc1d4d2c16006a6589c6
Reviewed-on: https://webrtc-review.googlesource.com/c/119460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26378}
The parameter was expanded twice in the macro,
leading to double use and move.
This is both an example of:
* Issue spotted by clandtidy's bug-prone patterns.
* Premature optimization.
Bug: webrtc:9855
Change-Id: I1a0cb2c99f95c6aec79ba1eb198aa39743ccbcd9
Reviewed-on: https://webrtc-review.googlesource.com/c/119042
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26367}
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.
Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26289}
