mikhal@webrtc.org 0aeb22e32c Adding tl0idx consideration for continuity ...

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5046 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-10-28 22:26:14 +00:00

..

audio_coding

Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.

2013-10-28 09:21:07 +00:00

audio_conference_mixer

Upgrade scoped_ptr to Chromium's latest version.

2013-10-22 12:50:00 +00:00

audio_device

WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.

2013-10-22 10:27:23 +00:00

audio_processing

Add an extended filter option to audioproc.

2013-10-23 14:07:17 +00:00

bitrate_controller

Change the low-bitrate handling in BitrateControllerImpl

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

desktop_capture

Upgrade scoped_ptr to Chromium's latest version.

2013-10-22 12:50:00 +00:00

interface

Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h

2013-10-25 18:15:09 +00:00

media_file

Remove redundant STR_CASE_CMP macro definitions.

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

pacing

Remove include_dirs from pacing.

2013-09-26 09:23:31 +00:00

remote_bitrate_estimator

Framework for testing bandwidth estimation.

2013-10-21 14:23:26 +00:00

rtp_rtcp

Fix for incorrect RTT estimation. A too low RTT value could be estimated.

2013-10-23 13:58:31 +00:00

utility

WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.

2013-10-22 10:27:23 +00:00

video_capture

Make sure the first frame isn't dropped.

2013-10-21 09:02:30 +00:00

video_coding

Adding tl0idx consideration for continuity

2013-10-28 22:26:14 +00:00

video_processing/main

VPM: Fixing namespace

2013-10-04 20:21:30 +00:00

video_render

Add APK and isolate target for video_engine_tests

2013-10-04 18:20:38 +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 APK and isolate target for video_engine_tests

2013-10-04 18:20:38 +00:00

modules_unittests.isolate

Add APK and isolate target for video_engine_tests

2013-10-04 18:20:38 +00:00

modules.gyp

Fix for incorrect RTT estimation. A too low RTT value could be estimated.

2013-10-23 13:58:31 +00:00