If screen share is set, then we need to tell video source, that it
is screen share source. Also video track should be aware, that it is
screen share track. It is required to choose proper video encoding
settings.
Bug: webrtc:10138
Change-Id: I5c82584ae0325a303a495554d87962a98b676694
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138278
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28098}
This involves catching another callback from usrsctp.
It also moves the definition of "connected" a little later
in the sequence: From "ready to send data" to the reception
of the SCTP_COMM_UP event.
Bug: chromium:943976
Change-Id: Ib9e1b17d0cc356f19cdfa675159b29bf1efdcb55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137435
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28004}
This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes
to avoid existing chromium tests to fail.
Instead of replacing the existing RtpSender::set_stream_ids() to
also fire OnRenegotiationNeeded(), this CL keeps the old
set_stream_ids() and adds the new RtpSender::SetStreams() which sets
the stream IDs and fires the callback.
This allows existing callsites to maintain behavior, and reserve
SetStreams() for the cases when we want OnRenegotiationNeeded() to fire.
Using the SetStreams() name instead of SetStreamIDs() to match the W3C
spec and to make it more different that the existing set_stream_ids().
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}
Bug: webrtc:10129
Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27992}
This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
event if needed and exposes the method on RtpSenderInterface.
This is a spec-compliance change.
Bug: webrtc:10129
Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27974}
This also changes the default when no max-message-size is set
to the protocol defined value of 64K, and prevents messages
from being sent when they are too large to send.
Bug: webrtc:10358
Change-Id: Iacc1dd774d1554d9f27315378fbea6351300b5cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135948
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27945}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
Previously, legacy GetStats would look up the track ID by querying the
local/remote SDP by SSRC. This doesn't work with Unified Plan since the
RtpSender/RtpReceiver track IDs may not correspond to the track ID
stored in the SDP.
This CL changes legacy GetStats to pull the track ID directly from the
RtpSenders and RtpReceivers as it generates the stats. This has a few
additional benefits:
1) Unsignaled receive SSRC stats should now get correctly matched to
the unsigneled RtpReceiver track ID for both Plan B and Unified
Plan.
2) Removes a couple methods on PeerConnection that were only used by
the legacy StatsCollector.
3) Keeps the SSRC -> track ID mapping more localized which should make
the code easier to understand.
Bug: chromium:943493
Change-Id: I43ecde8c3a3d1c5f9c749ba6c8dfb11e8c4950fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129782
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27324}
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
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}
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}
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}
This CL adds a single class to manage the use case of having a task
that repeats itself by a fixed or variable interval. It replaces the
repeating task previously locally defined for rtp transport controller
send as well as the cancelable periodic task. Furthermore, it is
introduced where one off repeating tasks were created before.
It provides the currently used functionality of the cancelable periodic
task, but not some of the unused features, such as allowing cancellation
of tasks before they are started and cancellation of a task after the
owning task queue has been destroyed.
Bug: webrtc:9883
Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca
Reviewed-on: https://webrtc-review.googlesource.com/c/116182
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26313}
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}
Preparation for replacing use of TestVideoCapturer as an interface,
instead using VideoSourceInterface.
Methods kept as non-virtual on the subclass FrameGeneratorCapturer,
but it's changed to be started on creation.
Bug: webrtc:6353
Change-Id: Iae1c9a0ee55d730d4992204f62227ef2f057d58e
Reviewed-on: https://webrtc-review.googlesource.com/c/114425
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26037}
The method for looking up track ID by SSRC was never updated for
Unified Plan so it only looked at the first audio section and the
first video section.
This CL changes the method to look through all audio and video
media sections rather than just the first.
Bug: chromium:906988
Change-Id: Ie79e6162b2bd24b8ac9e983b5fa7360c96f030da
Reviewed-on: https://webrtc-review.googlesource.com/c/112223
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25833}
Auxiliary threads (worker, network) are still active
while PeerConnection is destructed, leading to race condition
in tests such as:
* RTCStatsIntegrationTest.GetStatsFromCaller
* RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection
This CL prevents the conflict to happen by explicitly
closing the PeerConnection.
Bug: webrtc:9847
Change-Id: I40880bb9b193201711031b8c4563c6bbd4983c71
Reviewed-on: https://webrtc-review.googlesource.com/c/104820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25801}
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.
In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory
Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
Fixing bug that allows a channel to be set on a stopped transceiver.
This CL contains the following refactoring:
1. Extracted ChannelInterface from BaseChannel
2. Unified SetXxxMediaChannel (Voice, Video) into SetMediaChannel
Bug: webrtc:9932
Change-Id: I2fbf00c823b7848ad4f2acb6e80b1b58ac45ee38
Reviewed-on: https://webrtc-review.googlesource.com/c/110564
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@{#25641}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Reland Fix:
- cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional
root level configuration.
- peerconnectionfactory - If this optional is set will now overwrite the
underyling value.
This along with the other field will be deprecated once dependent projects
are updated.
TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org
Bug: webrtc:9681
Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d
Reviewed-on: https://webrtc-review.googlesource.com/c/105560
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25135}
This reverts commit ac2f3d14e45398930bc35ff05ed7a3b9b617d328.
Reason for revert: Breaks downstream project
Original change's description:
> Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h
>
> Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
> that only handles SRTP configuration to a more generic structure that can be
> used and extended for all per peer connection CryptoOptions that can be on a
> given PeerConnection.
>
> Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
> accessed as crypto_options.srtp.whatever_option_name. This is more inline with
> other structures we have in WebRTC such as VideoConfig. As additional features
> are added over time this will allow the structure to remain compartmentalized
> and concerned components can only request a subset of the overall configuration
> structure e.g:
>
> void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
>
> In addition to this it made little sense for sslstreamadapter.h to hold all
> Srtp related configuration options. The header has become loo large and takes on
> too many responsibilities and spilting this up will lead to more maintainable
> code going forward.
>
> This will be used in a future CL to enable configuration options for the newly
> supported Frame Crypto.
>
> Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
> Bug: webrtc:9681
> Reviewed-on: https://webrtc-review.googlesource.com/c/105180
> Reviewed-by: Emad Omara <emadomara@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25130}
TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
Bug: webrtc:9681
Change-Id: Ib0075c477c951b540d4deecb3b0cf8cf86ba0fff
Reviewed-on: https://webrtc-review.googlesource.com/c/105541
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25133}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
Bug: webrtc:9681
Reviewed-on: https://webrtc-review.googlesource.com/c/105180
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25130}
This will later allow simulcast to be set up without any SDP
manipulation. Currently limited to only one layer as the SDP
generated is not spec compliant and more work is required
to support simulcast.
Initial encoding parameters are deferred and applied when the ssrc
is set on the sender. This allows parameters to be changed before
negotiation is completed.
Bug: webrtc:7600
Change-Id: I0a31cd1c2bfc72ebb61ce0fa4fa69d87e3d8b74d
Reviewed-on: https://webrtc-review.googlesource.com/95488
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@{#24917}
RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.
Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
