jiayl@webrtc.org 9fd8d87ff5 Adds APIs for reporting pacer queuing delay. ...

BUG=2775
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8959005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5621 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-02-27 22:32:40 +00:00

..

audio_coding

Removing a variable that was never read

2014-02-20 15:27:00 +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

Remove ProcessingComponent's dependence on AudioProcessingImpl.

2014-02-27 22:23:17 +00:00

bitrate_controller

Change the low-bitrate handling in BitrateControllerImpl

2013-10-24 09:24:06 +00:00

desktop_capture

Call PrintWindow for the first time of capturing to capture the window frames correctly.

2014-02-27 16:43:12 +00:00

interface

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +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

Change the type of propagation delta from int64 to int.

2014-02-12 19:19:23 +00:00

rtp_rtcp

Add RTCP packet class.

2014-02-21 08:14:45 +00:00

utility

AviRecorder is missing a critical section.

2014-02-24 09:19:36 +00:00

video_capture

Remove "Too long processing time of Incoming frame" logspam.

2014-02-11 17:48:11 +00:00

video_coding

Split the implementation of VP8Encoder|Decoder::Create into a seperated file

2014-02-24 23:56:39 +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

Add RTCP packet class.

2014-02-21 08:14:45 +00:00