webrtc/modules
hlundin@google.com 6b04739e04 Route CodecSpecificInfo from encoder to packetizer
Making a long chain of interface changes to route a CodecSpecificInfo
struct from the video encoder function to the RTPSenderVideo. This
will be used to convey information needed by the RTP packetizer when
building the RTP headers.
Review URL: http://webrtc-codereview.appspot.com/56001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@140 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-01 08:32:57 +00:00
..
audio_coding Critical section in constructor, audio coding module 2011-06-22 09:21:51 +00:00
audio_conference_mixer add android makefile, some modification in vpx makefile to build encoder from c source for now 2011-06-07 17:24:39 +00:00
audio_device Adds sanity checks related to IAudioCaptureClient::GetBuffer. 2011-06-23 09:44:59 +00:00
audio_processing Optimization of 'rftbsub': 2011-06-30 19:19:37 +00:00
interface git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:22:19 +00:00
media_file add android makefile, some modification in vpx makefile to build encoder from c source for now 2011-06-07 17:24:39 +00:00
rtp_rtcp Route CodecSpecificInfo from encoder to packetizer 2011-07-01 08:32:57 +00:00
udp_transport Fixed valgrind warning in the udp_module. 2011-06-20 23:06:04 +00:00
utility Route CodecSpecificInfo from encoder to packetizer 2011-07-01 08:32:57 +00:00
video_capture Remove the full header file path to: 2011-06-22 21:17:43 +00:00
video_coding Route CodecSpecificInfo from encoder to packetizer 2011-07-01 08:32:57 +00:00
video_processing/main VPLIB/Interpolation - Delete decode buffer only if too small, this required an API change. In addition, done some clean up and updated test and related code in VCM. 2011-06-29 17:00:03 +00:00
video_render Disable ChangeWindow function for chromium build. 2011-06-17 22:18:43 +00:00