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webrtc/talk/media/webrtc
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pkasting@chromium.org 0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +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 cricket::VideoFrame int64 to int64_t. 2014-12-05 09:42:57 +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.cc (Auto)update libjingle 81702493-> 81755413 2014-12-10 09:01:18 +00:00
simulcast.h (Auto)update libjingle 81702493-> 81755413 2014-12-10 09:01:18 +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 Wire up RTT statistics to webrtc::Call. 2014-12-11 13:26:09 +00:00
webrtcvideoengine2_unittest.h Wire up RTT statistics to webrtc::Call. 2014-12-11 13:26:09 +00:00
webrtcvideoengine2.cc Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h 2014-12-14 11:09:23 +00:00
webrtcvideoengine2.h Wire up RTT statistics to webrtc::Call. 2014-12-11 13:26:09 +00:00
webrtcvideoengine_unittest.cc (Auto)update libjingle 81702493-> 81755413 2014-12-10 09:01:18 +00:00
webrtcvideoengine.cc Cleanup little things found when refactoring. 2014-12-12 02:44:30 +00:00
webrtcvideoengine.h (Auto)update libjingle 81702493-> 81755413 2014-12-10 09:01:18 +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