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mflodman@webrtc.org b4e5d1b34e Remove RTX SSRC when deleting the default receive stream. 
BUG=crbug 448632
TEST=New unittest hitting assert without this change.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8059 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 15:07:07 +00:00
..
constants.h
Enable VP9 video codec support on webrtcvideoengine behind a field trial.
2014-11-07 13:21:04 +00:00
dummyinstantiation.cc
Update talk to 58037405.
2013-12-11 18:25:07 +00:00
fakewebrtccommon.h
(Auto)update libjingle 72097588-> 72159069
2014-07-29 17:36:52 +00:00
fakewebrtcdeviceinfo.h
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
fakewebrtcvcmfactory.h
Adds trunk/talk folder of revision 359 from libjingles google code to
2013-07-10 00:45:36 +00:00
fakewebrtcvideocapturemodule.h
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
fakewebrtcvideoengine.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
fakewebrtcvoiceengine.h
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
2014-12-09 16:22:09 +00:00
OWNERS
Add pbos@webrtc.org (myself) to talk/media/webrtc/.
2014-09-16 16:14:51 +00:00
simulcast_unittest.cc
Add field trial for screenshare bitrates when using temporal layers.
2014-12-23 15:19:35 +00:00
simulcast.cc
Remove min bitrate from simulcast streams.
2015-01-02 15:40:56 +00:00
simulcast.h
Remove min bitrate from simulcast streams.
2015-01-02 15:40:56 +00:00
webrtccommon.h
Adds trunk/talk folder of revision 359 from libjingles google code to
2013-07-10 00:45:36 +00:00
webrtcexport.h
(Auto)update libjingle 72205295-> 72320533
2014-07-31 15:08:53 +00:00
webrtcmediaengine.cc
Skeleton for registering external encoders/decoders.
2014-09-23 09:40:22 +00:00
webrtcmediaengine.h
Removing unused method GetDefaultVideoEncoderConfig.
2014-11-06 11:16:32 +00:00
webrtcpassthroughrender_unittest.cc
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
webrtcpassthroughrender.cc
(Auto)update libjingle 72097588-> 72159069
2014-07-29 17:36:52 +00:00
webrtcpassthroughrender.h
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
webrtctexturevideoframe_unittest.cc
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
webrtctexturevideoframe.cc
cricket::VideoFrame int64 to int64_t.
2014-12-05 09:42:57 +00:00
webrtctexturevideoframe.h
cricket::VideoFrame int64 to int64_t.
2014-12-05 09:42:57 +00:00
webrtcvideocapturer_unittest.cc
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
webrtcvideocapturer.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
webrtcvideocapturer.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
webrtcvideocapturerfactory.cc
(Auto)update libjingle 72839629-> 72847605
2014-08-07 22:09:08 +00:00
webrtcvideocapturerfactory.h
(Auto)update libjingle 72839629-> 72847605
2014-08-07 22:09:08 +00:00
webrtcvideochannelfactory.h
Revert r6110 and r6109.
2014-05-13 11:07:01 +00:00
webrtcvideodecoderfactory.h
(Auto)update libjingle 72097588-> 72159069
2014-07-29 17:36:52 +00:00
webrtcvideoencoderfactory.h
(Auto)update libjingle 77414393-> 77554188
2014-10-13 06:35:10 +00:00
webrtcvideoengine2_unittest.cc
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
webrtcvideoengine2_unittest.h
Make setting identical RTP extensions a no-op.
2015-01-05 18:51:13 +00:00
webrtcvideoengine2.cc
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
webrtcvideoengine2.h
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
webrtcvideoengine_unittest.cc
Remove RTX SSRC when deleting the default receive stream.
2015-01-14 15:07:07 +00:00
webrtcvideoengine.cc
Remove RTX SSRC when deleting the default receive stream.
2015-01-14 15:07:07 +00:00
webrtcvideoengine.h
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
webrtcvideoframe_unittest.cc
Fix webrtcvideoframe tests.
2014-09-05 16:34:13 +00:00
webrtcvideoframe.cc
Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
2014-12-14 11:09:23 +00:00
webrtcvideoframe.h
Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
2014-12-14 11:09:23 +00:00
webrtcvideoframefactory.cc
(Auto)update libjingle 72682155-> 72785180
2014-08-07 04:47:36 +00:00
webrtcvideoframefactory.h
(Auto)update libjingle 72682155-> 72785180
2014-08-07 04:47:36 +00:00
webrtcvie.h
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
webrtcvoe.h
(Auto)update libjingle 73222930-> 73226398
2014-08-13 17:26:08 +00:00
webrtcvoiceengine_unittest.cc
Setting Opus FEC as default
2014-11-17 09:26:39 +00:00
webrtcvoiceengine.cc
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
2014-12-09 16:22:09 +00:00
webrtcvoiceengine.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00