mikhal@webrtc.org d96dcef422 vpm: Updating module to use CalcBufferSize ...

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/666008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2513 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-07-12 23:52:55 +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

Re-added ChangeUniqueId temporary for chrome.

2012-05-24 09:52:19 +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

Pass capture time (wallclock) to the RTP sender to compute transmission offset

2012-07-03 13:21:22 +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

vp8: Updating wrapper to use CalcBufferSize (includes odd size support).

2012-07-11 18:43:36 +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