pbos@webrtc.org 4ca7d3f9fe Replace MapWrapper with std::map<>. ...

MapWrapper was needed on some platforms where STL wasn't supported, we
now use std::map<> directly.

BUG=2164
TEST=trybots
R=henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4530 4adac7df-926f-26a2-2b94-8c16560cd09d

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

..

audio_coding

Added Opus stereo support

2013-08-12 06:48:09 +00:00

audio_conference_mixer

Replace MapWrapper with std::map<>.

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

audio_device

Remove unused unreferenced code in webrtc/

2013-08-09 08:47:51 +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

Fix crash in screen capturer on Mac

2013-08-10 01:30:23 +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

Don't pace out packets or generate padding when the pacer is disabled.

2013-08-09 11:31:23 +00:00

remote_bitrate_estimator

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

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

rtp_rtcp

Remove unused unreferenced code in webrtc/

2013-08-09 08:47:51 +00:00

utility

* Update libjingle to 50389769.

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

video_capture

Replace MapWrapper with std::map<>.

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

video_coding

Avoid acquiring VCM::_receiveCritSect during decode callback.

2013-08-12 14:20:49 +00:00

video_processing/main

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

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

video_render

Replace MapWrapper with std::map<>.

2013-08-12 19:51:57 +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