| Age | Commit message (Collapse) | Author |
|---|
|
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1486004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4042 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
Rolling to the latest Chromium picks up a new clang, which catches a fresh error:
error: 'reinterpret_cast' to class 'webrtc::VideoEngineImpl *' from its base at non-zero offset 'webrtc::VideoEngine *' behaves differently from 'static_cast' [-Werror,-Wreinterpret-base-class]
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../webrtc/video_engine/vie_codec_impl.cc:36:31: note: use 'static_cast' to adjust the pointer correctly while downcasting
VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
^~~~~~~~~~~~~~~~
static_cast
This was triggered by André's change here:
https://code.google.com/p/webrtc/source/detail?r=3986
which made VideoEngineImpl a derived class of VideoEngine (good).
Picked up one other error as well:
error: implicit conversion from 'long' to 'int' changes value from 9223372036854775807 to -1 [-Werror,-Wconstant-conversion]
AutoTestSleep(std::numeric_limits<long>::max());
~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This fixes the errors and is required before stable can be rolled in Chromium.
TBR=mflodman,andresp
Review URL: https://webrtc-codereview.appspot.com/1450004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3989 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1302005
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3801 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
Review URL: https://webrtc-codereview.appspot.com/1203009
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
Review URL: https://webrtc-codereview.appspot.com/1232004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3754 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
buffered at the sender. When the buffer grows above the target delay
encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
the pacer to faster get rid of the queue after a network down event.
(Work based on issue 1237004)
BUG=1524
TESTS=trybots,vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/1258004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
Review URL: https://webrtc-codereview.appspot.com/1199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
the functionality is now provided via the test frame work.
Review URL: https://webrtc-codereview.appspot.com/1029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
|
|
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
|
