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f9e7c9d8653495ec466d50665fb6299deacea98c
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stefan@webrtc.org f9e7c9d865 Add option to bwe_rtp_to_text to output arrival times only in nanoseconds. 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5665 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10 09:11:21 +00:00
..
audio_coding
Remove std:: prefixes from C functions in webrtc/.
2014-03-07 15:23:34 +00:00
audio_conference_mixer
Add Config struct for experimental AGC.
2014-01-29 04:57:25 +00:00
audio_device
ifdef the alsa code based on macro USE_X11
2014-02-20 03:05:05 +00:00
audio_processing
Switch to correct interpretation of int and float input data in audio_processing_unittest
2014-03-05 00:18:53 +00:00
bitrate_controller
Adding a new ramp-up-down-up test
2014-03-06 07:19:28 +00:00
desktop_capture
Remove std:: prefixes from C functions in webrtc/.
2014-03-07 15:23:34 +00:00
interface
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
2014-03-06 23:49:08 +00:00
media_file
Removes usage of ListWrapper from several files.
2014-01-13 15:21:30 +00:00
pacing
Adds APIs for reporting pacer queuing delay.
2014-02-27 22:32:40 +00:00
remote_bitrate_estimator
Add option to bwe_rtp_to_text to output arrival times only in nanoseconds.
2014-03-10 09:11:21 +00:00
rtp_rtcp
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
2014-03-06 23:49:08 +00:00
utility
AviRecorder is missing a critical section.
2014-02-24 09:19:36 +00:00
video_capture
VideoCaptureAndroid: don't deliver frames after stopCapture().
2014-03-04 18:17:55 +00:00
video_coding
Remove std:: prefixes from C functions in webrtc/.
2014-03-07 15:23:34 +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
Removes usage of ListWrapper from several files.
2014-01-13 15:21:30 +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
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
modules_unittests.isolate
Add trace-based delivery filter to BWE test framework.
2014-01-24 10:00:27 +00:00
modules.gyp
This CL is to add Opus complexity knob and to test it.
2014-03-07 08:55:48 +00:00