webrtc/src/modules
ajm@google.com f8dc8dc5f6 Generate protobuf classes at build-time.
This method is well-established in Chromium. The new code is largely boilerplate copied from there. The advantage is that we don't have to maintain various versions of the classes; we just generate against whatever compiler version happens to exist at build-time.
Review URL: http://webrtc-codereview.appspot.com/93008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@271 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-29 17:29:08 +00:00
..
audio_coding Clean up ANDROID macro definitions [audio_coding] 2011-07-28 20:39:08 +00:00
audio_conference_mixer Add license statement 2011-07-26 18:31:26 +00:00
audio_device exclude pulse audio when building with Chromium. 2011-07-21 20:45:31 +00:00
audio_processing Generate protobuf classes at build-time. 2011-07-29 17:29:08 +00:00
interface git-svn-id: http://webrtc.googlecode.com/svn/trunk@164 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:43:35 +00:00
media_file git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
rtp_rtcp Changing the default VP8 packetization mode setting to kAggregate and balanced, from the previous settig of kStrict and balanced. 2011-07-21 16:49:54 +00:00
udp_transport git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
utility git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
video_capture git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
video_coding Fix an ambiguous call to pow() error. 2011-07-28 18:43:18 +00:00
video_processing/main git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
video_render Clean up ANDROID macro definitions [video_render] 2011-07-26 05:03:10 +00:00