This is a reland of 2a11b2451a4068746fa0c55fa210efd4a15e4423
There are no changes compared to the first attempt.
Original change's description:
> Enable using a custom NetEqFactory in simulations
>
> Bug: webrtc:11005
> Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30286}
TBR=kwiberg
Bug: webrtc:11005
Change-Id: I4aa377e05916bd23f8f63aece9d0e27731c80d3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166465
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30319}
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.
Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
This is a reland of b70c5c5ce97e7dcf2e1d8453f5ea0639d4b60453
Original change's description:
> Using simulated rtc::Thread for peer connection scenario tests.
>
> Bug: webrtc:11255
> Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30258}
Bug: webrtc:11255
Change-Id: If65cd56b59158cebec5609407a721fbdb47cfd1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30294}
This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
Reason for revert: breaks negotiation with send-only clients
(webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
(peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
(peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
TBR=steveanton@webrtc.org,kron@webrtc.org
Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30292}
Even though send and receive codecs may be the same, they might have
different support in HW. Distinguish between send and receive codecs
to be able to keep track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30284}
This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
Reason for revert: Downstream tests have been updated.
Original change's description:
> Revert "Distinguish between send and receive video codecs"
>
> This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
>
> Reason for revert: Breaks downstream test.
>
> Original change's description:
> > Distinguish between send and receive video codecs
> >
> > Even though send and receive codecs are the same,
> > they might have different support in HW.
> > Distinguish between send and receive codecs to be able to keep
> > track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30041}
>
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
>
> Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30042}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1029737
Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30078}
This is used to allow using a pre-configured builders as arguments to
fixture code.
Bug: webrtc:9510
Change-Id: I7837d284580fdbc926535ce5b2d8f582056534ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161948
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30070}
This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
Reason for revert: Breaks downstream test.
Original change's description:
> Distinguish between send and receive video codecs
>
> Even though send and receive codecs are the same,
> they might have different support in HW.
> Distinguish between send and receive codecs to be able to keep
> track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30041}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30042}
Even though send and receive codecs are the same,
they might have different support in HW.
Distinguish between send and receive codecs to be able to keep
track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30041}
Introduce FrameGeneratorInterface to make FrameGenerator API available
for downstream projects.
Bug: webrtc:10138
Change-Id: I4216775e4b8b54c3f1c72d67ffbda31eb082fd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161234
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30009}
This makes the class easier to use at a minor cost of making it slightly
more magic.
Bug: webrtc:9883
Change-Id: If807cfbf046615333c3bcd3b58a001813102a9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161231
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30008}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
This change renames TimeController's Sleep method to AdvanceTime, unifying
the same name with the same semantic as for downstream projects.
Bug: webrtc:11154
Change-Id: Id79bcf0eafcd0b47a76407ba220479d84df5a736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161092
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29989}
This CL decouples NetEqFactory and AudioDecoderFactory.
AudioDecoderFactory is used in more places than just inside of NetEq, so
decoupling these makes sense.
Bug: webrtc:11005
Change-Id: I78dd856e4248e398e69a65816b062ef30555b055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161005
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29961}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
The NetEqFactory is currently expected to wrap the AudioDecoderFactory,
but this turns out not to be a good idea. Instead, it makes more sense
to pass the AudioDecoderFactory through the CreateNetEq method.
Bug: webrtc:11005
Change-Id: I8027ff6593f40c92072e7e88157631dcf329a984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160644
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29918}
Creates an abstraction for an "alarm clock" which can schedule
time-controller callbacks and exposes a time controller driven by
an external alarm.
Bug: webrtc:9719
Change-Id: I08c2aa9dba25603043bfba48f55c925716a55bae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158969
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29879}
Add ability to provide custom implementation of rtc::VideoSourceInterface
as source for video track in PC-framework based media quality tests.
Bug: webrtc:10138
Change-Id: I8ffd3015230c733a0a9a2e97fd4bb93a0c02b283
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159680
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29776}
This CL also introduces NetEqFactory and NetEqControllerFactory
interfaces, as well as several convenience classes for working with
them: DefaultNetEqFactory, DefaultNetEqControllerFactory and
CustomNetEqFactory.
Bug: webrtc:11005
Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29671}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport. When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section. This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section. Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.
This change adds an a=x-alt-protocol: line to SDP. The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line. This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.
Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data. It is
still not possible to use it for audio but not video, or vice versa.
PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media. It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels. PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.
JsepTransport now negotiates use of the datagram transport independently for
media and data channels. It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.
Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b
Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
>
> Also clears SctpTransport before deleting JsepTransport.
>
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport. This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
>
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
>
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
>
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP. Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left. For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports. Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}
Bug: webrtc:9719
Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29290}
