sjlee@webrtc.org fc10c5c97f This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 ...

BUG=1341
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2225004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4761 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-09-17 05:15:37 +00:00

..

audio_coding

Prepare to compile ACM1 and ACM2.

2013-09-13 23:06:59 +00:00

audio_conference_mixer

Replace MapWrapper with std::map<>.

2013-08-12 19:51:57 +00:00

audio_device

Fix win trybot errors due to r4729.

2013-09-13 00:02:13 +00:00

audio_processing

Recognize armv7 target_arch for ios support in webrtc common.gyp

2013-09-05 16:46:36 +00:00

bitrate_controller

Fix some chromium-style warnings in webrtc/modules/bitrate_controller/

2013-07-31 15:16:20 +00:00

desktop_capture

Implement DesktopRegion subtraction.

2013-09-13 19:53:16 +00:00

interface

Adds support for combining RTX and FEC/RED.

2013-09-06 13:40:11 +00:00

media_file

Remove redundant STR_CASE_CMP macro definitions.

2013-09-09 17:50:10 +00:00

pacing

Improving padding rules and breaking out bw allocation to ViEEncoder.

2013-09-06 13:58:01 +00:00

remote_bitrate_estimator

Re-enable tests for Remote Bitrate Estimator

2013-09-09 13:06:52 +00:00

rtp_rtcp

Lock RTPSender statistics.

2013-09-10 11:29:17 +00:00

utility

Remove use of vcm->ResetDecoder from modules/utility.

2013-09-16 11:26:35 +00:00

video_capture

This is related to https://code.google.com/p/webrtc/issues/detail?id=846

2013-09-17 05:15:19 +00:00

video_coding

Split video coding module unit tests into sender and receiver unit tests.

2013-09-16 20:29:13 +00:00

video_processing/main

Switch C++-style C headers with their C equivalents.

2013-08-05 16:22:53 +00:00

video_render

This is related to https://code.google.com/p/webrtc/issues/detail?id=1341

2013-09-17 05:15:37 +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

Add isolate configuration for Android for all tests.

2013-08-27 12:10:09 +00:00

modules_unittests.isolate

Add isolate configuration for Android for all tests.

2013-08-27 12:10:09 +00:00

modules.gyp

Split video coding module unit tests into sender and receiver unit tests.

2013-09-16 20:29:13 +00:00