leozwang@webrtc.org f3adba499e Add Android include path so that header files can follow google style ...

BUG=1011
TEST=bot
Review URL: https://webrtc-codereview.appspot.com/930018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3107 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-11-15 18:17:40 +00:00

..

audio_coding

Pure Neon assembly coding for WebRtcIsacfix_AutocorrNeon() in iSAC-Fix.

2012-11-14 18:44:24 +00:00

audio_conference_mixer

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

audio_device

Add Android OWNER files

2012-11-09 17:51:55 +00:00

audio_processing

Add Android include path so that header files can follow google style

2012-11-15 18:17:40 +00:00

bitrate_controller

Clarifies the bandwidth estimation interfaces.

2012-11-13 15:02:13 +00:00

interface

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

media_file

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

pacing

Adding pacing module, will replace the transmission_bucket in the RTP module.

2012-11-09 20:56:23 +00:00

remote_bitrate_estimator

Clarifies the bandwidth estimation interfaces.

2012-11-13 15:02:13 +00:00

rtp_rtcp

Enable paced sender.

2012-11-13 21:12:39 +00:00

udp_transport

Clean up TraceCallback::Print.

2012-11-15 05:33:25 +00:00

utility

Add Android include path so that header files can follow google style

2012-11-15 18:17:40 +00:00

video_capture

Fix sorting issues in video_capture.gypi.

2012-11-15 10:30:57 +00:00

video_coding

Relanding r3071 - updates for i420: Making sure that decoded frame is complete and buffer size is sufficient. Re-landing is possible following r3094 - which disabled a problematic test.

2012-11-14 17:56:46 +00:00

video_processing/main

Verify output frame timestamp in VideoProcessingModuleTest.Resampler.

2012-11-08 01:07:21 +00:00

video_render

Fix OpenGL rendering of WebRTCDemo by accounting for stride != width.

2012-11-12 16:39:25 +00:00

modules.gyp

Adding pacing module, will replace the transmission_bucket in the RTP module.

2012-11-09 20:56:23 +00:00