Add support of negotiating simulcast offer/answer. Also fix some minor
issues around to make it finally work.
Bug: webrtc:10138
Change-Id: I382f5df04ca6ac04d8ed1e030e7b2ae5706dd10c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137425
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28274}
Also removes SessionDescription::Copy.
Bug: webrtc:10612
Change-Id: Ib652d717531738c3ed5d1054e32a03961e16dba9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139903
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28239}
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
recent changed decreased timeout from forever to 1s which is not enough on some platforms
Increase timeout to forever for posting 65k tasks.
Also increase timeout for eventual destruction of the tasks to reduce change it would flake.
Bug: chromium:972917
Change-Id: I4948d49c1514833ab190856fdd25a47a5bad91eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141410
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28235}
This reverts commit 11dfff0878c949f2e19d95a0ddc209cdad94b3b4.
Reason for revert: Downstream import failure.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org
Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().
Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
This is a reland of 890bc3069cbababa19b40ec02684253d60e051b2
Zero bitrate caused division by zero in DCHECK for max bitrate.
Added unit tests to ensure that setting zero bitrate does not crash.
> Original change's description:
> > Cleanup of video packet overhead calculation.
> >
> > This CL updates the video packet overhead calculation to make it more
> > clear. This prepares for future work on improving the accuracy of the
> > calculation.
> >
> > Bug: webrtc:9883
> > Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28040}
Bug: webrtc:10674
Change-Id: I156d1ee5546ede7e43ae1d9a298dcaba6071230f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140890
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28212}
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.
Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
avoid waiting while executing a Task to discourage blocking.
fix accessing tasks_clean_up counter since after TaskQueue is destroyed,
it doesn't guarantee sequential execution of the destructors, nor
that all pending tasks are destroyed at that moment.
Instead verify that all posted tasks will be destroyed eventually.
Bug: None
Change-Id: I4cfc97ac0787fe2d0b9d2f0d712a37ae0ca9e1aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140288
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28208}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28198}
This reverts commit 71c6482baf0ff17141c635e6a7639493db68a65c.
Reason for revert: Lands too much at once and breaks downstream tests that need to implement new interfaces first.
Original change's description:
> Implement true negotiation for DatagramTransport with fallback to RTP.
>
> In short, the caller places a x-opaque line in SDP for each m= section that
> uses datagram transport. If the answerer supports datagram transport, it will
> parse this line and create a datagram transport. It will then echo the x-opaque
> line into the answer (to indicate that it accepted use of datagram transport).
>
> If the offer and answer contain exactly the same x-opaque line, both peers will
> use datagram transport. If the x-opaque line is omitted from the answer (or is
> different in the answer) they will fall back to RTP.
>
> Note that a different x-opaque line in the answer means the answerer did not
> understand something in the negotiation proto. Since WebRTC cannot know what
> was misunderstood, or whether it's still possible to use the datagram transport,
> it must fall back to RTP. This may change in the future, possibly by passing
> the answer to the datagram transport, but it's good enough for now.
>
> Negotiation consists of four parts:
> 1. DatagramTransport exposes transport parameters for both client and server
> perspectives. The client just echoes what it received from the server (modulo
> any fields it might not have understood).
>
> 2. SDP adds a x-opaque line for opaque transport parameters. Identical to
> x-mt, but this is specific to datagram transport and goes in each m= section,
> and appears in the answer as well as the offer.
> - This is propagated to Jsep as part of the TransportDescription.
> - SDP files: transport_description.h,cc, transport_description_factory.h,cc,
> media_session.cc, webrtc_sdp.cc
>
> 3. JsepTransport/Controller:
> - Exposes opaque parameters for each mid (m= section). On offerer, this means
> pre-allocating a datagram transport and getting its parameters. On the
> answerer, this means echoing the offerer's parameters.
> - Uses a composite RTP transport to receive from either default RTP or
> datagram transport until both offer and answer arrive.
> - If a provisional answer arrives, sets the composite to send on the
> provisionally selected transport.
> - Once both offer and answer are set, deletes the unneeded transports and
> keeps whichever transport is selected.
>
> 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
>
> Bug: webrtc:9719
> Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28182}
TBR=steveanton@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org
Change-Id: I0d502c4a6d27516c35ed85154f3fa5869f88b3b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140822
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28188}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28182}
This is a reland of 9469c784dbf732472e3b2a60a5fcca0a2f432313
Original change's description:
> Added OnIceCandidateError to API and implementation
>
> Bug: webrtc:3098
> Change-Id: I27ffd015ebf9e8130c1288f7331b0e2fdafb01ef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135953
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28173}
TBR=steveanton@webrtc.org
Bug: webrtc:3098
Change-Id: I77af2065fc1479273f399e2b3d919f98fe8ac23d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140641
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28179}
Bug: webrtc:3098
Change-Id: I27ffd015ebf9e8130c1288f7331b0e2fdafb01ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135953
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28173}
In several places VideoFrame::Builder is used to create a new VideoFrame
when intent is to change only one or two fields of a const VideoFrame&.
This approach is bad because each and every metadata field have to be
added to all the places.
Instead, this CL adds missing setters and refactors the code to use
full copy of a VideoFrame and update required fields only.
Along the way few actual bugs are fixed, e.g. when ColorSpace isn't copied
when frame rotation or buffer is cropped or converted.
Bug: webrtc:10460
Change-Id: I2895a473ca938b150eed2916c689060bdf58cb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140102
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28170}
Using this flag, an encoder may inform the RTP sender module that
the packet is not elligible for retransmission. Specifically, it
may not be retransmitted in response to a NACK message,
nor because of early loss detection (see CL #135881).
Bug: webrtc:10702
Change-Id: Ib6a9cc361cf10ea7214cf672e05940c27899a6be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140105
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28169}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
from a field trial to RTCConfiguration.
The test coverage is also expanded for the underlying feature.
Bug: None
Change-Id: Ic9c1362867e4a956c5453be7a9355083b6a442f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28143}
This change adds classes so that we later can plumb information about received packets to each audio and video frame. It's not wired up to do anything yet.
Bug: webrtc:10668
Change-Id: I962df493a76692f668314f78d6792d7636c5a31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138203
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28138}
Bug: None
Change-Id: I11a7ae5d1b0157b1d7b537fa7c071f0f48efe307
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/113147
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28130}
The datagram sink needs to know when datagrams are lost in addition to
when they are acked.
DatagramAck::receive_timestamp needs a default value so that
DatagramAck's default ctor is not implicitly deleted. Without a default
ctor, it's not possible to make this struct without specifying all its
fields, so users will still be broken when the interface adds a new
field.
Bug: webrtc:9719
Change-Id: I6688a938d68eea133f12b13a1228d4df4771d1b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139480
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28117}
Whenever a datagram is acked, the datagram transport will provide the
remote peer's receive timestamp in this field.
Bug: webrtc:9719
Change-Id: I516b9d602e62179a3deda001e0ee9b484aa20d37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139440
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28114}
Bug: webrtc:9719
Change-Id: Ib394e02a4f0dcb36da64b9f005f068a53a50854c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139280
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28111}
In addition to the 48 kHz that we've always used.
Bug: webrtc:10631
Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28104}
In addition to the 48 kHz that we've always used.
Bug: webrtc:10631
Change-Id: I5e4f6600e39a463d20d3988db098c7e38281f4a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138264
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28074}
This implements the essentials of RTCRemoteInboundRtpStreamStats. This
includes:
- ssrc
- transportId
- codecId
- packetsLost
- jitter
- localId
- roundTripTime
https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
The following members are not implemented because they require more
work...
- From RTCReceivedRtpStreamStats: packetsReceived, packetsDiscarded,
packetsRepaired, burstPacketsLost, burstPacketsDiscarded,
burstLossCount, burstDiscardCount, burstLossRate, burstDiscardRate,
gapLossRate and gapDiscardRate.
- From RTCRemoteInboundRtpStreamStats: fractionLost.
Bug: webrtc:10455, webrtc:10456
Change-Id: If2ab0da7105d8c93bba58e14aa93bd22ffe57f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138067
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28073}
These are unused except in tests, and just add clutter.
Bug: webrtc:9824
Change-Id: Ica209d09850f5ff9b122ce21306aaf1bbfc7bda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28064}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
- Implement datagram transport adaptor, which wraps datagram transport in DtlsTransportInternal. Datagram adaptor owns both ICE and Datagram Transports.
- Implement setup of datagram transport based on RTCConfiguration flag use_datagram_transport. This is very similar to MediaTransport setup with the exception that we create DTLS datagram adaptor.
- Propagate maximum datagram size to video encoder via MediaTransportConfig.
TODO: Currently this CL can only be tested in downstream projects. Once we add fake datagram transport, we will be able to implement unit tests similar to loopback media transport.
Bug: webrtc:9719
Change-Id: I4fa4a5725598dfee5da4f0f374269a7e289d48ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138100
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28047}
