andresp@webrtc.org 4e69f782b0 Small refactor on send_side_bandwidth_estimation. ...

R=stefan@webrtc.org
BUG=3065

Review URL: https://webrtc-codereview.appspot.com/10029005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5710 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-03-17 17:07:48 +00:00

..

audio_coding

References to includes in third_party should be relative, not absolute.

2014-03-14 16:22:18 +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

Disable all protobuf dependent targets when enable_protobuf=0.

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

bitrate_controller

Small refactor on send_side_bandwidth_estimation.

2014-03-17 17:07:48 +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 max delay to trace based filters and enhances drop tail queues with delay statistics.

2014-03-13 15:01:32 +00:00

rtp_rtcp

Race condition in RTPSender

2014-03-13 15:12:37 +00:00

utility

AviRecorder is missing a critical section.

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

video_capture

AppRTCDemo(iOS): video support; part 1 of 2: webrtc/.

2014-03-10 17:12:08 +00:00

video_coding

Replace labs with std::abs.

2014-03-13 08:53:39 +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

iOS video_render: omit no-op setNeedsDisplay

2014-03-10 18:48:19 +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