4ab4fc0044
Add test for automatically disabling padding when no video is being captured.
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BUG=2648
TEST=trybots
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5169 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-25 11:54:24 +00:00
331d4402fc
Connect pacer/padding to SuspendBelowMinBitrate
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The suspend function must not be engaged unless padding is also enabled.
This CL makes the connection so that the pacer and padding is enabled
when SuspendBelowMinBitrate is.
Had to change the unit test to make it aware of the padding packets.
BUG=2606
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5153 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 14:05:40 +00:00
2c46f8d854
Rename DestroyStream methods to include Video.
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Matches r5135 which renames CreateSendStream->CreateVideoSendStream for
instance.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5151 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 13:49:43 +00:00
ce90eff345
Rename RTP-extension constants.
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BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 11:48:56 +00:00
53c8573525
Rename video streams' start/stop methods.
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{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 11:36:47 +00:00
5a63655ab0
Rename Call::Create{Receive,Send}Stream().
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Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 10:40:25 +00:00
ce8e0936d9
Rename AutoMute to SuspendBelowMinBitrate
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Changes all instances throughout the WebRTC stack.
BUG=2436
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 12:18:43 +00:00
4cfa6050f6
Fix breakage after introducing new test.
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TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3899005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5127 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 13:15:56 +00:00
69969e2e2f
Improve Call tests for RTX.
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Also does some refactoring to reuse RtpRtcpObserver.
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 12:32:15 +00:00
6488761f2e
Implement VideoSendStream::SetCodec().
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Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.
This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-14 08:58:14 +00:00
def22b455b
Stop DirectTransports in VideoSendStreamTests.
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Prevents racy packet delivery during or after Call destruction.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3099005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5049 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29 10:12:10 +00:00
16e03b7bd8
Separate Call API/build files from video_engine/.
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BUG=2535
R=andrew@webrtc.org , mflodman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 16:32:01 +00:00
