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webrtc_m130/webrtc/modules
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stefan@webrtc.org c4726d06fa Make RTPSender::SendPadData public. 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5219 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 09:16:33 +00:00
..
audio_coding
Small fixes to plot_neteq_delay.m
2013-12-04 12:28:47 +00:00
audio_conference_mixer
Upgrade scoped_ptr to Chromium's latest version.
2013-10-22 12:50:00 +00:00
audio_device
Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close).
2013-12-03 19:10:20 +00:00
audio_processing
Remove unused ThreadData struct.
2013-12-04 17:13:47 +00:00
bitrate_controller
Change the low-bitrate handling in BitrateControllerImpl
2013-10-24 09:24:06 +00:00
desktop_capture
Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin.
2013-11-22 20:39:16 +00:00
interface
Remove ".." from include_dirs in build/common.
2013-11-11 10:20:27 +00:00
media_file
Hook up audio/video sync to Call.
2013-11-18 11:45:11 +00:00
pacing
Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large.
2013-11-28 14:00:09 +00:00
remote_bitrate_estimator
Add baseline generation/verification to BWE test framework.
2013-12-03 15:11:14 +00:00
rtp_rtcp
Make RTPSender::SendPadData public.
2013-12-05 09:16:33 +00:00
utility
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
2013-10-22 10:27:23 +00:00
video_capture
Replace VideoFrameI420 with I420VideoFrame.
2013-11-22 13:10:13 +00:00
video_coding
Add send frame rate statistics callback
2013-12-04 15:09:27 +00:00
video_processing/main
Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error.
2013-11-25 09:06:33 +00:00
video_render
Remove ".." from include_dirs in build/common.
2013-11-11 10:20:27 +00:00
module_common_types_unittest.cc
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
2013-04-11 17:48:02 +00:00
modules_java_chromium.gyp
Fixes build breakage when building WebRTC in Chromium and having include_tests=1.
2013-07-08 18:14:58 +00:00
modules_java.gyp
Build all java files into jar for each module on Android
2013-07-01 17:52:39 +00:00
modules_tests.isolate
Add APK and isolate target for video_engine_tests
2013-10-04 18:20:38 +00:00
modules_unittests.isolate
Add baseline generation/verification to BWE test framework.
2013-12-03 15:11:14 +00:00
modules.gyp
Add baseline generation/verification to BWE test framework.
2013-12-03 15:11:14 +00:00