6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t. IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
3e1cfa7edba8081faada275683b3d1fc71f37ac7 |
|
12-Jan-2016 |
nisse <nisse@webrtc.org> |
Delete unused method webrtc::VideoRendererInterface::SetSize. BUG=webrtc:5426 Review URL: https://codereview.webrtc.org/1582493002 Cr-Commit-Position: refs/heads/master@{#11223}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
6eca7e3c371383020095ba346e1ac70f38a8c0fd |
|
15-Dec-2015 |
tommi <tommi@webrtc.org> |
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( Additionally: * Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack. * AddSink/RemoveSink are now on all audio sources (like they are for video sources). While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state. BUG=chromium:569526 Review URL: https://codereview.webrtc.org/1522903002 Cr-Commit-Position: refs/heads/master@{#11026}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
f888bb58da04c5095759b5ec7ce2e1fa2cd414fd |
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12-Dec-2015 |
Tommi <tommi@webrtc.org> |
Support for unmixed remote audio into tracks. BUG=chromium:121673 R=solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1505253004 . Cr-Commit-Position: refs/heads/master@{#10995}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
fac0655fd7fe0b40ef50dc5b7f11ea44d72cec6c |
|
25-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
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}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
5def7b9fdea0d027bca3df734d86fb877a83bdbf |
|
20-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
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}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
6834fa10f142bf5e2275142acb834898911d09ae |
|
20-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
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}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
8f46c63f6f764254892f4111b54aa1cc8f32eeeb |
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26-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
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}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
ac9d92ccbe2b29590c53f702e11dc625820480d5 |
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26-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
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}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
c2db810b8958588771282634d00b7e3954c9f5ab |
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09-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Remove VideoRendererInterface::CanApplyRotation() All implementations handle rotation now, both internally in WebRTC and externally in Chromium. R=glaznev@webrtc.org, guoweis@webrtc.org Review URL: https://codereview.webrtc.org/1313753003 . Cr-Commit-Position: refs/heads/master@{#9911}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|
00c509ad1c94805b3332f2ce876c04705abf8ef5 |
|
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add concept of whether video renderer supports rotation. Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation. Tested with peerconnection_client on windows, AppRTCDemo on Mac. BUG=4145 R=glaznev@webrtc.org, pthatcher@webrtc.org Committed: https://code.google.com/p/webrtc/source/detail?r=8660 Committed: https://code.google.com/p/webrtc/source/detail?r=8661 Review URL: https://webrtc-codereview.appspot.com/43569004 Cr-Commit-Position: refs/heads/master@{#8705} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8705 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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f9a75d99b92402c56744121b7bc991a9c71cf324 |
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10-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Add concept of whether video renderer supports rotation." This reverts commit 0ad48935fc5b92be6e10924a9ee3b0dc39c79104. TBR=guoweis@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/41199004 Cr-Commit-Position: refs/heads/master@{#8663} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8663 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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0ad48935fc5b92be6e10924a9ee3b0dc39c79104 |
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10-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add concept of whether video renderer supports rotation. Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation. BUG=4145 R=glaznev@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43569004 Cr-Commit-Position: refs/heads/master@{#8660} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8660 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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5f93d0a140515e3b8cdd1b9a4c6f5871144e5dee |
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20-Jan-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
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
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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d4e598d57aed714a599444a7eab5e8fdde52a950 |
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29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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b90991dade9139e5c14c3b616a9eff07b9d6fdda |
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04-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 62472237->62550414 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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40b3b68cdf47d7c9c3b57fca5d0a372292025f9e |
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03-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 62364298->62472237 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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b9a088b920d1ba16e0593698d4a613bb7bb5481f |
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14-Feb-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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0de29504ab7ac923401c8e4e154f3b72038dbcc2 |
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13-Feb-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5545 "Update libjingle to 61514460" > Update libjingle to 61514460 > > TBR=tommi@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/8649004 TBR=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5547 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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e749c9ebdb2eb2a519c72c827e70107cbc56d270 |
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13-Feb-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 61514460 TBR=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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67ee6b9a6260fa80b83326c4b4fec8857c0e578c |
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03-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 60923971 Review URL: https://webrtc-codereview.appspot.com/7909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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967bfff54d00f176a554bf9f955f14dde99f7bb9 |
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19-Sep-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 52534915. R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/2251004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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32001ef124f5082651c661965dc5d75d7f06a57b |
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13-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection shutdown-time fixes - 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
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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1e09a711263dd105e6f7a03812250084c64e5fd8 |
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26-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49952949 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
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28e20752806a492f5a6a5d343c02f9556f39b1cd |
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10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/talk/app/webrtc/mediastreaminterface.h
|