webrtc/src/modules
2012-07-13 16:27:51 +00:00
..
audio_coding Corrected one error for Android build. 2012-07-10 21:37:49 +00:00
audio_conference_mixer Downmix before resampling in capture and render paths. 2012-06-27 03:25:31 +00:00
audio_device Remove the useless dummy audio device impl which creates threads and high res timers on windows. 2012-07-06 08:33:13 +00:00
audio_processing Added API to port internal speech probability in NS. 2012-07-12 21:00:43 +00:00
bitrate_controller Move test to src/test. 2012-06-27 01:41:54 +00:00
interface Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. 2012-07-13 16:27:51 +00:00
media_file Move test to src/test. 2012-06-27 01:41:54 +00:00
remote_bitrate_estimator Landing: https://webrtc-codereview.appspot.com/680005/ 2012-07-03 08:19:12 +00:00
rtp_rtcp Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. 2012-07-13 16:27:51 +00:00
udp_transport Move test to src/test. 2012-06-27 01:41:54 +00:00
utility Pass capture time (wallclock) to the RTP sender to compute transmission offset 2012-07-03 13:21:22 +00:00
video_capture Remove files that are not needed from direct_show_base_classes.gyp 2012-07-11 16:52:19 +00:00
video_coding Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. 2012-07-13 16:27:51 +00:00
video_processing/main vpm: Updating module to use CalcBufferSize 2012-07-12 23:52:55 +00:00
video_render Use one OS-independent sleep function in a video test 2012-06-25 11:30:33 +00:00
modules.gyp Refactoring the receive-side bandwidth estimation into its own module. 2012-06-07 08:10:14 +00:00