kma@webrtc.org adf8ddf4aa Assembly coding for pitch filter in iSAC for ARMv6. ...

Review URL: https://webrtc-codereview.appspot.com/631004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2501 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-07-10 19:30:57 +00:00

..

audio_coding

Assembly coding for pitch filter in iSAC for ARMv6.

2012-07-10 19:30:57 +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

Fix Android build after test/ -> src/test/

2012-06-27 03:22:37 +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

Move test to src/test.

2012-06-27 01:41:54 +00:00

video_coding

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

2012-07-03 13:21:22 +00:00

video_processing/main

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

2012-07-03 13:21:22 +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