Bug: webrtc:7600
Change-Id: I2a48426a29ac67b6bdbd7817fe07273cdd5fd980
Reviewed-on: https://webrtc-review.googlesource.com/31647
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21305}
This reverts commit 8b13f96e2d.
Original change's description:
> Revert "Add AddTransceiver and GetTransceivers to PeerConnection"
>
> This reverts commit f93d2800d9.
>
> Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
>
> Original change's description:
> > Add AddTransceiver and GetTransceivers to PeerConnection
> >
> > WebRTC 1.0 has added the transceiver API to PeerConnection. This
> > is the first step towards exposing this to WebRTC consumers. For
> > now, transceivers can be added and fetched but there is not yet
> > support for creating offers/answers or setting local/remote
> > descriptions. That support ("Unified Plan") will be added in
> > follow-up CLs.
> >
> > The transceiver API is currently only available if the application
> > opts in by specifying the kUnifiedPlan SDP semantics when creating
> > the PeerConnection.
> >
> > Bug: webrtc:7600
> > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
> > Reviewed-on: https://webrtc-review.googlesource.com/23880
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20896}
>
> TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
>
> Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7600
> Reviewed-on: https://webrtc-review.googlesource.com/26400
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20897}
TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7600
Reviewed-on: https://webrtc-review.googlesource.com/26401
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20898}
This reverts commit f93d2800d9.
Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
Original change's description:
> Add AddTransceiver and GetTransceivers to PeerConnection
>
> WebRTC 1.0 has added the transceiver API to PeerConnection. This
> is the first step towards exposing this to WebRTC consumers. For
> now, transceivers can be added and fetched but there is not yet
> support for creating offers/answers or setting local/remote
> descriptions. That support ("Unified Plan") will be added in
> follow-up CLs.
>
> The transceiver API is currently only available if the application
> opts in by specifying the kUnifiedPlan SDP semantics when creating
> the PeerConnection.
>
> Bug: webrtc:7600
> Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
> Reviewed-on: https://webrtc-review.googlesource.com/23880
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20896}
TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7600
Reviewed-on: https://webrtc-review.googlesource.com/26400
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20897}
WebRTC 1.0 has added the transceiver API to PeerConnection. This
is the first step towards exposing this to WebRTC consumers. For
now, transceivers can be added and fetched but there is not yet
support for creating offers/answers or setting local/remote
descriptions. That support ("Unified Plan") will be added in
follow-up CLs.
The transceiver API is currently only available if the application
opts in by specifying the kUnifiedPlan SDP semantics when creating
the PeerConnection.
Bug: webrtc:7600
Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
Reviewed-on: https://webrtc-review.googlesource.com/23880
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20896}
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 6c7ec32bd6
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 6c7ec32bd6.
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}
This reverts commit 33c5c7f5e4.
Reason for revert: Fix broken API change.
TBR=sprang@webrtc.org,solenberg@webrtc.org
TBRing because only pc/ and api/ have changed since last LGTMed version.
Original change's description:
> Revert "Encode log events periodically instead of for every event."
>
> This reverts commit b154c27e72.
>
> Reason for revert: Broke the internal project.
>
> Original change's description:
> > Encode log events periodically instead of for every event.
> >
> > Updated unit test to take output_period and random seed as parameters.
> > Updated the peerconnection interface to allow passing in an output_period.
> >
> > This is in preparation of some upcoming CLs that will change the format
> > to store batches of delta-encoded values.
> >
> >
> > Bug: webrtc:8111
> > Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
> > Reviewed-on: https://webrtc-review.googlesource.com/22600
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20736}
>
> Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229
> Bug: webrtc:8111
> Reviewed-on: https://webrtc-review.googlesource.com/24160
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20738}
Bug: webrtc:8111
Change-Id: Ie69862cd52d11c1e15adeb6e2caacafe16863c80
Reviewed-on: https://webrtc-review.googlesource.com/24620
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20811}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}
This reverts commit 90bace0958.
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}
This reverts commit 54d1da13a5.
Reason for revert: Breaking tests
Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
>
> This CL implements the main logic and IOS appRTC integration.
>
> Unit tests and Android appRTC will be in separate CL.
>
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}
TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
This CL implements the main logic and IOS appRTC integration.
Unit tests and Android appRTC will be in separate CL.
Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
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}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
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}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/api/peerconnectionproxy.h (Browse further)
