braveyao@webrtc.org 83a062cc5f AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() ...

BUG=1891
Test=ManualTest

R=fischman@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1622004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4200 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-06-10 08:09:05 +00:00

..

audio_coding

Nack for audio.

2013-06-06 19:00:09 +00:00

audio_conference_mixer

Include files from webrtc/.. paths in audio_conference_mixer/

2013-05-28 08:13:20 +00:00

audio_device

AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout()

2013-06-10 08:09:05 +00:00

audio_processing

Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD

2013-05-28 09:24:03 +00:00

bitrate_controller

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

desktop_capture

Remove fake screen capturer because it's not used anywhere.

2013-06-06 23:11:33 +00:00

interface

Breaking out RTP header parsing from the RTP module.

2013-05-29 12:12:51 +00:00

media_file

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

pacing

Fix build error introduced with r4168.

2013-06-04 09:52:46 +00:00

remote_bitrate_estimator

Update the remote bitrate estimator before passing the packet to the RTP module.

2013-06-04 12:15:40 +00:00

rtp_rtcp

Refactor padding and rtp header functionality.

2013-06-04 13:47:36 +00:00

utility

Merge webrtc_utility_unittests into modules_unittests.

2013-06-05 08:58:46 +00:00

video_capture

Removing functionality for inserting pre-encoded frames instead of raw

2013-06-07 13:57:57 +00:00

video_coding

Add support for padding in pacer.

2013-06-04 09:36:56 +00:00

video_processing/main

Include files from webrtc/.. paths in video_processing/

2013-05-27 14:12:16 +00:00

video_render

Removed unused class members to enable clang=1 android build.

2013-06-04 00:37:21 +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.gyp

Merge webrtc_utility_unittests into modules_unittests.

2013-06-05 08:58:46 +00:00