asapersson@webrtc.org 8b2ec15d1e Convert WEBRTC_TRACE to LOG in utility. ...

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5886 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-04-11 07:59:43 +00:00

..

audio_coding

Move a chatty creation log in neteq to LS_VERBOSE.

2014-04-09 17:48:48 +00:00

audio_conference_mixer

Add Config struct for experimental AGC.

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

audio_device

Remove AudioDevice::{Microphone,Speaker}IsAvailable.

2014-04-02 16:49:26 +00:00

audio_processing

Fix the library path for android 64-bit build

2014-04-04 04:44:19 +00:00

bitrate_controller

Fix logging calls in bitrate_controller module.

2014-04-07 08:45:16 +00:00

desktop_capture

Fix DesktopSize::is_empty() for the case when only width or only height is 0.

2014-04-09 01:04:22 +00:00

interface

VoE changes to allow forwarding of packets from VoE to ViE BWE.

2014-03-24 10:38:25 +00:00

media_file

Removes usage of ListWrapper from several files.

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

pacing

Simplify pacer interface.

2014-03-19 10:59:52 +00:00

remote_bitrate_estimator

Clean up traces and logs in RemoteBitrateEstimator.

2014-04-07 12:53:28 +00:00

rtp_rtcp

Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.

2014-04-08 11:06:12 +00:00

utility

Convert WEBRTC_TRACE to LOG in utility.

2014-04-11 07:59:43 +00:00

video_capture

Remove self-assignment hacks that were added to avoid unused variable warnings.

2014-04-09 21:19:55 +00:00

video_coding

Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.

2014-04-08 11:06:12 +00:00

video_processing/main

Remove usage of webrtc trace in video processing modules.

2014-04-10 11:30:49 +00:00

video_render

VideoCaptureAndroid: stop referencing ViERenderer

2014-04-08 22:55:07 +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

Classes and tests for audio an classifier. The class can be used to classify whether a frame of audio contains speech or music. The classifier uses the music/speech classifier in Opus.

2014-03-10 22:50:19 +00:00

modules.gyp

Disable all protobuf dependent targets when enable_protobuf=0.

2014-03-13 00:57:52 +00:00