andrew@webrtc.org 54744918ef Update AudioProcessing::Create docs. ...

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/8039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5488 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-02-05 06:30:29 +00:00

..

audio_coding

Trivial rename of non-compile time consts.

2014-02-04 01:31:28 +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

Update AudioProcessing::Create docs.

2014-02-05 06:30:29 +00:00

bitrate_controller

Change the low-bitrate handling in BitrateControllerImpl

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

desktop_capture

Fix a cursor capturing issue on Windows.

2014-02-04 17:49:12 +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

Fix for libtalkmobile build error

2014-01-21 16:16:58 +00:00

remote_bitrate_estimator

Wire up feedback to VideoSender.

2014-02-03 16:33:50 +00:00

rtp_rtcp

Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered.

2014-02-04 14:38:25 +00:00

utility

Removes usage of ListWrapper from several files.

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

video_capture

Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."

2014-01-29 00:56:02 +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

Remove the requirement to call set_sample_rate_hz and friends.

2014-01-07 17:45:09 +00:00