(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}
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}
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}
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}
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}
- TCPPort::~TCPPort() was leaking incoming_ sockets; now they are deleted.
- PeerConnection::RemoveStream() now removes streams even if the
PeerConnection::IsClosed(). Previously such streams would never get removed.
- Gave MediaStreamTrackInterface a virtual destructor to ensure deletes on base
pointers are dispatched virtually.
- VideoTrack.dispose() delegates to super.dispose() (instead of leaking)
- PeerConnection.dispose() now removes streams before disposing of them.
- MediaStream.dispose() now removes tracks before disposing of them.
- VideoCapturer.dispose() only unowned VideoCapturers (mirroring C++ API)
- AppRTCDemo.disconnectAndExit() now correctly .dispose()s its
VideoSource and PeerConnectionFactory.
- CHECK that Release()d objects are deleted when expected to (i.e. no ref-cycles
or missing .dispose() calls) in the Java API.
- Create & Return webrtc::Traces at factory birth/death to be able to assert
that _all_ threads started during the test are collected by the end.
- Name threads attached to the JVM more informatively for debugging.
- Removed a bunch of unnecessary scoped_refptr instances in
peerconnection_jni.cc whose only job was messing with refcounts.
RISK=P2
TESTED=AppRTCDemo can be ended and restarted just fine instead of crashing on camera unavailability. No more post-app-exit logcat lines. PCTest.java now asserts that all threads are collected before exit.
BUG=2183
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2005004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4534 4adac7df-926f-26a2-2b94-8c16560cd09d
