asapersson@webrtc.org e7b1e11283 Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." ...

> Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
> 
> > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> > 
> > R=holmer@google.com
> > 
> > Review URL: https://webrtc-codereview.appspot.com/5049004
> 
> TBR=asapersson@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5799004

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-12-16 14:40:36 +00:00

..

audio_coding

Update talk to 58174641 together with http://review.webrtc.org/4319005/.

2013-12-13 19:17:43 +00:00

audio_conference_mixer

Upgrade scoped_ptr to Chromium's latest version.

2013-10-22 12:50:00 +00:00

audio_device

Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).

2013-12-11 21:42:44 +00:00

audio_processing

Simplification of histogram normalization in delay estimator.

2013-12-16 13:37:28 +00:00

bitrate_controller

Change the low-bitrate handling in BitrateControllerImpl

2013-10-24 09:24:06 +00:00

desktop_capture

Add shape in DesktopFrame.

2013-12-07 01:03:28 +00:00

interface

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

media_file

Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.

2013-12-09 16:23:00 +00:00

pacing

Measure pacer queue size based on when packets are inserted rather than captured.

2013-12-13 22:03:27 +00:00

remote_bitrate_estimator

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

rtp_rtcp

Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."

2013-12-16 14:40:36 +00:00

utility

Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.

2013-12-09 16:23:00 +00:00

video_capture

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

video_coding

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +00:00

video_processing/main

Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error.

2013-11-25 09:06:33 +00:00

video_render

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

module_common_types_unittest.cc

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

modules_java_chromium.gyp

Fixes build breakage when building WebRTC in Chromium and having include_tests=1.

2013-07-08 18:14:58 +00:00

modules_java.gyp

Build all java files into jar for each module on Android

2013-07-01 17:52:39 +00:00

modules_tests.isolate

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

modules_unittests.isolate

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

modules.gyp

Update talk to 58174641 together with http://review.webrtc.org/4319005/.

2013-12-13 19:17:43 +00:00