21 Commits

Author SHA1 Message Date
deadbeef
bd7d8f7e2b Adding a MediaStream parameter to createSender.
This will allow an app to create senders with the same stream id,
without SDP munging.

Review URL: https://codereview.webrtc.org/1538673002

Cr-Commit-Position: refs/heads/master@{#11092}
2015-12-19 00:58:51 +00:00
deadbeef
fac0655fd7 Reland of Adding the ability to create an RtpSender without a track.
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )

Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.

Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}

Review URL: https://codereview.webrtc.org/1468113002

Cr-Commit-Position: refs/heads/master@{#10790}
2015-11-25 19:26:08 +00:00
deadbeef
5def7b9fde Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
Reason for revert:
Still breaking CallAndModifyStream. Chromium CL intended to fix it (https://codereview.chromium.org/1435713002/) wasn't sufficient, because I forgot to call addStream/removeStream on the second connection.

Original issue's description:
> Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
>
> Reason for revert:
> Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.
>
> Original issue's description:
> > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
> >
> > Reason for revert:
> > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
> >
> > Original issue's description:
> > > Adding the ability to create an RtpSender without a track.
> > >
> > > This CL also changes AddStream to immediately create a sender, rather
> > > than waiting until the track is seen in SDP. And the PeerConnection now
> > > builds the list of "send streams" from the list of senders, rather than
> > > the collection of local media streams.
> > >
> > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > > Cr-Commit-Position: refs/heads/master@{#10414}
> >
> > TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> >
> > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> > Cr-Commit-Position: refs/heads/master@{#10417}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/6834fa10f142bf5e2275142acb834898911d09ae
> Cr-Commit-Position: refs/heads/master@{#10730}

TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1460323002

Cr-Commit-Position: refs/heads/master@{#10732}
2015-11-20 19:43:27 +00:00
deadbeef
6834fa10f1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
Reason for revert:
Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.

Original issue's description:
> Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
>
> Reason for revert:
> Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
>
> Original issue's description:
> > Adding the ability to create an RtpSender without a track.
> >
> > This CL also changes AddStream to immediately create a sender, rather
> > than waiting until the track is seen in SDP. And the PeerConnection now
> > builds the list of "send streams" from the list of senders, rather than
> > the collection of local media streams.
> >
> > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > Cr-Commit-Position: refs/heads/master@{#10414}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> Cr-Commit-Position: refs/heads/master@{#10417}

TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1413983004

Cr-Commit-Position: refs/heads/master@{#10730}
2015-11-20 17:50:02 +00:00
deadbeef
8f46c63f6f Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Reason for revert:
Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.

Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}

TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1426443007

Cr-Commit-Position: refs/heads/master@{#10417}
2015-10-26 21:11:25 +00:00
deadbeef
ac9d92ccbe Adding the ability to create an RtpSender without a track.
This CL also changes AddStream to immediately create a sender, rather
than waiting until the track is seen in SDP. And the PeerConnection now
builds the list of "send streams" from the list of senders, rather than
the collection of local media streams.

Review URL: https://codereview.webrtc.org/1413713003

Cr-Commit-Position: refs/heads/master@{#10414}
2015-10-26 18:48:26 +00:00
deadbeef
a67696b3cd Reland of Adding PeerConnectionInterface::SetConfiguration method. (patchset #1 id:1 of https://codereview.webrtc.org/1361263002/ )
Reason for revert:
Relanding with SetConfiguration not pure virtual.

Original issue's description:
> Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ )
>
> Reason for revert:
> Broke FYI bots because SetConfiguration is pure virtual and MockPeerConnectionImpl doesn't implement it. Need to reland with SetConfiguration not pure virtual.
>
> Original issue's description:
> > Adding PeerConnectionInterface::SetConfiguration method.
> >
> > Also updated the JNI and Objective-C bindings. Later, will have a CL to
> > remove UpdateIce, which this method effectively replaces.
> >
> > BUG=webrtc:4945
> >
> > Committed: https://crrev.com/70702afbcb8418fe93747e7ed63bcbf5e56b90e9
> > Cr-Commit-Position: refs/heads/master@{#10040}
>
> TBR=guoweis@webrtc.org,pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4945
>
> Committed: https://crrev.com/7603c76ab077b1e2033bb179595129bd96797345
> Cr-Commit-Position: refs/heads/master@{#10041}

TBR=guoweis@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4945

Review URL: https://codereview.webrtc.org/1361273002

Cr-Commit-Position: refs/heads/master@{#10112}
2015-09-29 18:56:30 +00:00
deadbeef
70ab1a1ca8 Exposing RtpSenders and RtpReceivers from PeerConnection.
This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.

It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.

Review URL: https://codereview.webrtc.org/1351803002

Cr-Commit-Position: refs/heads/master@{#10100}
2015-09-28 23:54:02 +00:00
deadbeef
7603c76ab0 Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ )
Reason for revert:
Broke FYI bots because SetConfiguration is pure virtual and MockPeerConnectionImpl doesn't implement it. Need to reland with SetConfiguration not pure virtual.

Original issue's description:
> Adding PeerConnectionInterface::SetConfiguration method.
>
> Also updated the JNI and Objective-C bindings. Later, will have a CL to
> remove UpdateIce, which this method effectively replaces.
>
> BUG=webrtc:4945
>
> Committed: https://crrev.com/70702afbcb8418fe93747e7ed63bcbf5e56b90e9
> Cr-Commit-Position: refs/heads/master@{#10040}

TBR=guoweis@webrtc.org,pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4945

Review URL: https://codereview.webrtc.org/1361263002

Cr-Commit-Position: refs/heads/master@{#10041}
2015-09-24 00:37:20 +00:00
deadbeef
70702afbcb Adding PeerConnectionInterface::SetConfiguration method.
Also updated the JNI and Objective-C bindings. Later, will have a CL to
remove UpdateIce, which this method effectively replaces.

BUG=webrtc:4945

Review URL: https://codereview.webrtc.org/1317353005

Cr-Commit-Position: refs/heads/master@{#10040}
2015-09-23 23:32:38 +00:00
honghaiz
4edc39c569 Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter.
BUG= 4901

Review URL: https://codereview.webrtc.org/1315503003

Cr-Commit-Position: refs/heads/master@{#9832}
2015-09-01 16:54:02 +00:00
honghaiz
900996290c Add methods to set the ICE connection receiving_timeout values.
BUG=

Review URL: https://codereview.webrtc.org/1231913003

Cr-Commit-Position: refs/heads/master@{#9572}
2015-07-13 19:19:42 +00:00
jlmiller@webrtc.org
5f93d0a140 Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
perkj@webrtc.org
fd0efb694a Remove deprecated PeerConnection APIs.
Removes PeerConnectionObserver::OnError.
Removes MediaConstraints argument to PeerConnection::AddStream.
None of these have ever been implemented and have been removed from the spec.

R=tommi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/24189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7650 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 12:16:36 +00:00
perkj@webrtc.org
b5d045e94d ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:01:33 +00:00
perkj@webrtc.org
c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:31:29 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
buildbot@webrtc.org
1567b8cf8c (Auto)update libjingle 66540208-> 66541346
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6085 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-08 19:54:16 +00:00
jiayl@webrtc.org
db41b4dbcd Remove the deprecated GetStats method from PeerConnectionInterface.
R=fischman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03 21:30:06 +00:00
wu@webrtc.org
b9a088b920 Update talk to 61538839.
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/8669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 23:18:49 +00:00
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00