turaj@webrtc.org 78f0db4710 Fix the break caused by r5579. ...

TBR=tlegrand@google.com
BUG=

Review URL: https://webrtc-codereview.appspot.com/8939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5581 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-02-19 23:07:31 +00:00

..

audio_coding

Fix the break caused by r5579.

2014-02-19 23:07:31 +00:00

audio_conference_mixer

Add Config struct for experimental AGC.

2014-01-29 04:57:25 +00:00

audio_device

Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base

2014-02-02 15:30:20 +00:00

audio_processing

Move the volume quantization workaround from VoE to AGC.

2014-02-18 20:24:56 +00:00

bitrate_controller

Change the low-bitrate handling in BitrateControllerImpl

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

desktop_capture

Make WindowCapturerLinux handling window resize events.

2014-02-19 17:28:41 +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

Fix for libtalkmobile build error

2014-01-21 16:16:58 +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 type counter (for getting statistics such as sent/received NACK and FIR).

2014-02-19 11:59:02 +00:00

utility

Removes usage of ListWrapper from several files.

2014-01-13 15:21:30 +00:00

video_capture

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

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

video_coding

Add BWE tools for parsing RTP files.

2014-01-31 09:15:48 +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 delay and send/receive throughput plots to BWE simulation.

2014-02-05 15:57:14 +00:00