andresp@webrtc.org 36947bb635 Fix logging calls in bitrate_controller module. ...

BUG=3153
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11069005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5851 4adac7df-926f-26a2-2b94-8c16560cd09d

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

..

audio_coding

Consolidate audio conversion from Channel and TransmitMixer.

2014-04-03 21:56:01 +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 a crash in WindowCapturereMac when capture() fails.

2014-04-04 20:26:41 +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

Added simulations of capacity variations and wifi recordings.

2014-04-01 14:00:05 +00:00

rtp_rtcp

Protect write of send_target_bitrate.

2014-03-26 14:27:34 +00:00

utility

AviRecorder is missing a critical section.

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

video_capture

sink_filter_ds.cc: add lock to Receive procedure to Pause().

2014-04-02 02:14:55 +00:00

video_coding

Fix "unreachable code" warnings (MSVC warning 4702) in webrtc.

2014-03-21 10:26:42 +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

iOS VideoEngine: move video_{capture,render} to ARC.

2014-03-25 05:15:44 +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