Description for changes from the original CL:
Calling legacy SRD, implemented using
SetRemoteDescriptionObserverAdapter wrapping the old observer, was
meant to have the exact same behavior as the legacy SRD implementation
which invokes the callbacks in a Post.
However, in the CL that landed and got reverted (PS1), the Adapter had
its own message handler, and callbacks would be invoked even if the PC
was destroyed.
In PS2 I've changed the Adapter to use the PeerConnection's message
handler. If the PC is destroyed, the callback will not be invoked.
This gives identical behavior to before this CL, and the legacy
behavior is covered by a new unittest.
I changed the adapter to be an implementation detail of
peerconnection.cc, therefor some stuff was moved, and the only tests
covering this is now in peerconnection_rtp_unittest.cc.
This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72
Original change's description:
> SetRemoteDescriptionObserverInterface added.
>
> The new observer replaced SetSessionDescriptionObserver for
> SetRemoteDescription. Unlike SetSessionDescriptionObserver,
> SetRemoteDescriptionObserverInterface is invoked synchronously so
> that the you can rely on the state of the PeerConnection to represent
> the result of the SetRemoteDescription call in the callback.
>
> The new observer succeeds or fails with an RTCError.
>
> This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
> and SetSessionDescriptionObserver, with the benefit that all media
> object changes can be processed in a single callback by the application
> in a synchronous callback. This will help Chromium keep objects in-sync
> across layers and threads in a non-racy and straight-forward way, see
> design doc (Proposal 2):
> https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
>
> An adapter for SetSessionDescriptionObserver is added to allow calling
> the old SetRemoteDescription signature and get the old behavior
> (OnSuccess/OnFailure callback in a Post) until third parties switch.
>
> Bug: webrtc:8473
> Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
> Reviewed-on: https://webrtc-review.googlesource.com/17523
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20841}
TBR=pthatcher@webrtc.org
Bug: webrtc:8473
Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5
Reviewed-on: https://webrtc-review.googlesource.com/25640
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20854}
This reverts commit 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72.
Reason for revert: Third party project breaks due to use-after-free
in the callback. I suspect this is because the adapter is processing
the async callback instead of the pc, i.e. callback is called from
SetRemoteDescriptionObserverAdapter::OnMessage instead of from
PeerConnection::OnMessage. This makes it possible for the callback to
be invoked after the PC is destroyed.
I argue this is how it should be done, and that if you're using a raw
pointer in an async callback you're doing it wrong, but I will reland
this CL with the callback processed in PeerConnection::OnMessage
instead as to not change the behavior of the old SRD signature.
Original change's description:
> SetRemoteDescriptionObserverInterface added.
>
> The new observer replaced SetSessionDescriptionObserver for
> SetRemoteDescription. Unlike SetSessionDescriptionObserver,
> SetRemoteDescriptionObserverInterface is invoked synchronously so
> that the you can rely on the state of the PeerConnection to represent
> the result of the SetRemoteDescription call in the callback.
>
> The new observer succeeds or fails with an RTCError.
>
> This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
> and SetSessionDescriptionObserver, with the benefit that all media
> object changes can be processed in a single callback by the application
> in a synchronous callback. This will help Chromium keep objects in-sync
> across layers and threads in a non-racy and straight-forward way, see
> design doc (Proposal 2):
> https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
>
> An adapter for SetSessionDescriptionObserver is added to allow calling
> the old SetRemoteDescription signature and get the old behavior
> (OnSuccess/OnFailure callback in a Post) until third parties switch.
>
> Bug: webrtc:8473
> Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
> Reviewed-on: https://webrtc-review.googlesource.com/17523
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20841}
TBR=hbos@webrtc.org,hta@webrtc.org,pthatcher@webrtc.org,guidou@webrtc.org
Change-Id: I715555e084d9aae49ee2a8831c70dc006dbdb74c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8473
Reviewed-on: https://webrtc-review.googlesource.com/25580
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20850}
The new observer replaced SetSessionDescriptionObserver for
SetRemoteDescription. Unlike SetSessionDescriptionObserver,
SetRemoteDescriptionObserverInterface is invoked synchronously so
that the you can rely on the state of the PeerConnection to represent
the result of the SetRemoteDescription call in the callback.
The new observer succeeds or fails with an RTCError.
This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack
and SetSessionDescriptionObserver, with the benefit that all media
object changes can be processed in a single callback by the application
in a synchronous callback. This will help Chromium keep objects in-sync
across layers and threads in a non-racy and straight-forward way, see
design doc (Proposal 2):
https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing
An adapter for SetSessionDescriptionObserver is added to allow calling
the old SetRemoteDescription signature and get the old behavior
(OnSuccess/OnFailure callback in a Post) until third parties switch.
Bug: webrtc:8473
Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99
Reviewed-on: https://webrtc-review.googlesource.com/17523
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20841}
Some targets depend on "api/peerconnectioninterface.h" which is part of
//api:peerconnection_and_implicit_call_api.
Furthermore, peerconnection_and_implicit_call_api depends among other
things of headers in //media:rtc_media_base, so we should add it as a
dependency as well.
Bug: webrtc:7504
Change-Id: Ifab69253d52532876509b3507917b1c93d99a2ac
Reviewed-on: https://webrtc-review.googlesource.com/24660
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20815}
The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel.
DtlsSrtpTransport is responsible for exporting keys from DtlsTransport
and setting up the wrapped SrtpTransport.
The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL.
Bug: webrtc:7013
Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713
Reviewed-on: https://webrtc-review.googlesource.com/6700
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20804}
Introduces the public API interface corresponding to the
standardized RtpTransceiver object in the WebRTC spec.
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
The RtpTransceiver will be the internal representation for both
Plan B and Unified Plan SDP, but the public API interface will
only support Unified Plan (existing users should continue to use
GetSenders/GetReceivers, which will still be supported).
Bug: webrtc:7600
Change-Id: I417ffda683209ba9a9b4cbd274f91ca8295779a7
Reviewed-on: https://webrtc-review.googlesource.com/21460
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20659}
This literally copies & pastes the code from WebRtcSession into
PeerConnection as private methods. The only other changes were to
inline the WebRtcSession construction/initialization/destruction
into PeerConnection and fix issues using rtc::Bind on the
reference-counted PeerConnection.
Bug: webrtc:8323
Change-Id: Ib3f071ac10d18566a21a3b04813b1d4ec691ef3c
Reviewed-on: https://webrtc-review.googlesource.com/15160
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20574}
We need to get rid of the ones that don't take audio codec factory
arguments in order to eliminate the dependency on audio codec
implementations.
BUG=webrtc:8396
Change-Id: Id0c1c3b70c2b3479da81ba1056cc69e857e454bd
Reviewed-on: https://webrtc-review.googlesource.com/12281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20555}
This reverts commit 90bace095806a635411edd40fb8490a144e59e63.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
This SetAudioPlayout method lets applications disable audio playout while
still processing incoming audio data and generating statistics on the
received audio.
This may be useful if the application wants to set up media flows as
soon as possible, but isn't ready to play audio yet. Currently, native
applications don't have any API point to control this, unless they
completely implement their own AudioDeviceModule.
The SetAudioRecording works in a similar fashion but for the recorded
audio. One difference is that calling SetAudioRecording(false) does not
keep any audio processing alive.
TBR=solenberg
Bug: webrtc:7313
Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
Reviewed-on: https://webrtc-review.googlesource.com/16180
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20499}
Originally, the idea was to implement QUIC data channels as a
PeerConnection API. Now, the effort has shifted to implementing it as a
part of ORTC which will live in Chromium. Since this code has not been
maintained and is not currently being used, remove it to reduce
maintenance overhead while a copy will be retained in the Git history.
Bug: webrtc:8385
Change-Id: I2719c007a0de0118b67d41a425f900b66c52f65a
Reviewed-on: https://webrtc-review.googlesource.com/14100
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20412}
This is a reland of b49b66109ea8a0a33a3192ebccf91366af2e49ae.
Original change's description:
> Rewrite WebRtcSession BUNDLE tests as PeerConnection tests
>
> Bug: webrtc:8222
> Change-Id: Id47e4544dc073564ad7e63d02865ca80dd5a85ff
> Reviewed-on: https://webrtc-review.googlesource.com/8280
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20365}
Bug: webrtc:8222
Change-Id: If3dcd8090875c641881e2b9e92fc1db387ba1de5
Reviewed-on: https://webrtc-review.googlesource.com/14400
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20397}
This is a reland of 3df5dcac9b339ba4d3f4969602f094c2c8035b51
Original change's description:
> Rewrite WebRtcSession media tests as PeerConnection tests
>
> Bug: webrtc:8222
> Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad
> Reviewed-on: https://webrtc-review.googlesource.com/6640
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20364}
Bug: webrtc:8222
Change-Id: I0a5398170d469eb9223bc781bfb417a85a72a2d2
Reviewed-on: https://webrtc-review.googlesource.com/14380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20377}
This will allow Chrome to provide a RtcEventLogOutput object that reports the log back to Chrome, allowing Chrome to manage the log by itself - write it to a file, upload it to a server, etc.
Bug: webrtc:8111
Change-Id: I6a2a6945fc8586ef10e0fb9c56eaa8fda00dfc98
Reviewed-on: https://webrtc-review.googlesource.com/8081
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20295}
This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40.
Reason for revert:
Fixed the flake.
Original change's description:
> Revert "Added PeerConnectionObserver::OnRemoveTrack."
>
> This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc.
>
> Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway.
>
> Original change's description:
> > Added PeerConnectionObserver::OnRemoveTrack.
> >
> > This corresponds to processing the removal of a remote track step of
> > the spec, with processing the addition of a remote track already
> > covered by OnAddTrack.
> > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
> >
> > Bug: webrtc:8260, webrtc:8315
> > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
> > Reviewed-on: https://webrtc-review.googlesource.com/4722
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20214}
>
> TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
>
> Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8260, webrtc:8315
> Reviewed-on: https://webrtc-review.googlesource.com/7940
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20218}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org
Change-Id: Iab7500bebf98535754b102874259de43831fff6b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8260, webrtc:8315
Reviewed-on: https://webrtc-review.googlesource.com/8180
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20227}
This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc.
Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway.
Original change's description:
> Added PeerConnectionObserver::OnRemoveTrack.
>
> This corresponds to processing the removal of a remote track step of
> the spec, with processing the addition of a remote track already
> covered by OnAddTrack.
> https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
>
> Bug: webrtc:8260, webrtc:8315
> Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
> Reviewed-on: https://webrtc-review.googlesource.com/4722
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20214}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8260, webrtc:8315
Reviewed-on: https://webrtc-review.googlesource.com/7940
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20218}
This corresponds to processing the removal of a remote track step of
the spec, with processing the addition of a remote track already
covered by OnAddTrack.
https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
Bug: webrtc:8260, webrtc:8315
Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
Reviewed-on: https://webrtc-review.googlesource.com/4722
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20214}
The TransportController was in p2p/base before and it cannot depend on
pc/ or media/ level targets because of the circular dependency. To make the
TransportController be responsible for creating and managing
the RtpTransport related objects which are pc/ level targets, the
TransportController is moved from p2p/base to pc/.
The TransportController makes more sense in pc/ anyway, since its main
responsibility is processing the "transport" parts of SDP which is
PeerConnection-specific.
This is also easier than moving RtpTransport related objects to p2p/base
because those objects also depend on other media/ and pc/ level targets
such as srtpfilter, cryptoparams etc.
Bug: webrtc:7013
Change-Id: Ic48dd5c454046ff3c81331f4b459f96a3255f328
Reviewed-on: https://webrtc-review.googlesource.com/4560
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20049}
This lays the groundwork for splitting up the
PeerConnectionInterface unit tests into multiple files so that
the tests can be organized better. The intent is for each unit
test file to declare a test fixture which subclasses
PeerConnectionUnitTestFixture and creates PeerConnectionWrappers
to write assertions against.
Bug: webrtc:8222
Change-Id: I21175b1e1828a6cd5012305a8a27faaf4eecf81c
Reviewed-on: https://webrtc-review.googlesource.com/1120
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20004}
This CL exposes the new type of video codec factories that represent all
video codecs in the PeerConnectionFactory API, i.e. no extra internal SW
video codecs will be added. Clients of the new functions will be
responsible for adding all SW video codecs themselves, and also handling
SW fallback and simulcast.
BUG=webrtc:7925
R=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/3004353002 .
Cr-Commit-Position: refs/heads/master@{#19866}
There were a number of unused includes and undeclared
dependencies. I removed the includes that were causing
problems and added dependencies for the includes that
turned out to be needed.
Bug: webrtc:7239,webrtc:6828
Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284
Reviewed-on: https://webrtc-review.googlesource.com/1185
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19858}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
