dce40cf804
Update a ton of audio code to use size_t more correctly and in general reduce
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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}
2015-08-24 21:52:45 +00:00
00c509ad1c
Add concept of whether video renderer supports rotation.
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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
2015-03-12 21:38:19 +00:00
f9a75d99b9
Revert "Add concept of whether video renderer supports rotation."
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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
2015-03-10 06:37:41 +00:00
0ad48935fc
Add concept of whether video renderer supports rotation.
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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
2015-03-10 02:43:40 +00:00
5f93d0a140
Update libjingle license statements at top of talk files for consistency
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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
d4e598d57a
(Auto)update libjingle 72097588-> 72159069
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
b90991dade
Update libjingle 62472237->62550414
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-04 19:54:57 +00:00
40b3b68cdf
Update libjingle 62364298->62472237
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03 18:30:11 +00:00
b9a088b920
Update talk to 61538839.
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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
0de29504ab
Revert 5545 "Update libjingle to 61514460"
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> 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
2014-02-13 19:54:28 +00:00
e749c9ebdb
Update libjingle to 61514460
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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
2014-02-13 15:09:40 +00:00
67ee6b9a62
Update talk to 60923971
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Review URL: https://webrtc-codereview.appspot.com/7909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 16:57:16 +00:00
967bfff54d
Update talk to 52534915.
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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
2013-09-19 05:49:50 +00:00
32001ef124
PeerConnection shutdown-time fixes
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- 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
2013-08-12 23:26:21 +00:00
1e09a71126
Update talk folder to revision=49952949
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-26 19:17:59 +00:00
28e2075280
Adds trunk/talk folder of revision 359 from libjingles google code to
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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
