asapersson@webrtc.org 86bb56a7f5 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-12-13 16:16:45 +00:00

..

audio_coding

ACM 2 compatibility with ACM 1.

2013-12-13 00:10:44 +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

Change uses of the obsolete armv7 setting to arm_version==7.

2013-12-10 09:26:07 +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

Remove ".." from include_dirs in build/common.

2013-11-11 10:20:27 +00:00

media_file

Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.

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

pacing

Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large.

2013-11-28 14:00:09 +00:00

remote_bitrate_estimator

Making RemoteRateControl::min_configured_bit_rate_ configurable

2013-12-13 08:42:42 +00:00

rtp_rtcp

Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."

2013-12-13 16:16:45 +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

Add SwapFrame() to VideoSendStreamInput.

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