webrtc/src/modules
2012-06-12 20:12:13 +00:00
..
audio_coding Use new fileutil functions for trace in ACM 2012-06-12 07:16:24 +00:00
audio_conference_mixer Move audio_frame_operations to the utility module. 2012-05-29 22:13:14 +00:00
audio_device Added gyp variable to include/exclude all tests. 2012-05-24 13:23:35 +00:00
audio_processing Refactored Neon code for AECM module, by using pure assembly code. 2012-06-07 16:17:17 +00:00
bitrate_controller Attempt to fix broken encoding. 2012-06-04 11:04:05 +00:00
interface Re-added ChangeUniqueId temporary for chrome. 2012-05-24 09:52:19 +00:00
media_file Added gyp variable to include/exclude all tests. 2012-05-24 13:23:35 +00:00
remote_bitrate_estimator abs() was used instead of fabsf(), which returns int and not float and therefore truncated the return value. 2012-06-07 13:48:04 +00:00
rtp_rtcp FEC: Fix to coverity issue 14448: unintended sign extension. 2012-06-12 20:12:13 +00:00
udp_transport Fix compilation errors on ChromeOS 2012-05-30 16:46:09 +00:00
utility Fixing gyp bug in https://webrtc-codereview.appspot.com/599006 2012-05-30 14:32:42 +00:00
video_capture Fix for the alignment problems/mismatch in ViECapture and VP8Encoder. 2012-06-05 23:52:59 +00:00
video_coding Fix for the alignment problems/mismatch in ViECapture and VP8Encoder. 2012-06-05 23:52:59 +00:00
video_processing/main Check return value of fwrite. [Video Module] 2012-05-29 17:33:13 +00:00
video_render Refactored IncomingVideoStream and VideoRenderFrame, to get code in better shape when hunting BUG=481. 2012-05-30 10:45:18 +00:00
modules.gyp Refactoring the receive-side bandwidth estimation into its own module. 2012-06-07 08:10:14 +00:00