pwestin@webrtc.org 401ef361ac Added configuration of max delay to ACM and NetEq ...

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1964004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4499 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-08-06 21:01:36 +00:00

..

audio_coding

Added configuration of max delay to ACM and NetEq

2013-08-06 21:01:36 +00:00

audio_conference_mixer

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

audio_device

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

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

audio_processing

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

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

bitrate_controller

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

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

desktop_capture

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

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

interface

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

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

media_file

Invert dependency between webrtc_utility and media_file targets to reflect reality.

2013-08-05 18:45:19 +00:00

pacing

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

2013-07-31 15:18:19 +00:00

remote_bitrate_estimator

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

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

rtp_rtcp

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

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

utility

* Update libjingle to 50389769.

2013-08-05 20:36:57 +00:00

video_capture

PeerConnectionTest.java: make the test work for the bots' v4l2loopback.

2013-07-29 20:43:15 +00:00

video_coding

Added Decoding with errors API to video_coding.h and removed unused DecodeError enum.

2013-08-06 18:27:41 +00:00

video_processing/main

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

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

video_render

* Update libjingle to 50389769.

2013-08-05 20:36:57 +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_unittests.isolate

Fixes resources and data path in modules_unittests.isolate.

2013-07-25 22:44:04 +00:00

modules.gyp

Invert dependency between webrtc_utility and media_file targets to reflect reality.

2013-08-05 18:45:19 +00:00