pbos@webrtc.org 72491b9a90 Count total bytes sent in RTPSender::Bytes(). ...

Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-07-10 16:24:54 +00:00

..

audio_coding

Receiver bit-exactness test for AudioCoding Module

2014-06-26 19:07:04 +00:00

audio_conference_mixer

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

audio_device

Fixing compile error.

2014-07-08 11:48:37 +00:00

audio_processing

audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h

2014-07-10 08:09:50 +00:00

bitrate_controller

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

desktop_capture

Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault.

2014-07-09 19:41:32 +00:00

interface

Add boilerplate code for H.264.

2014-07-04 12:42:07 +00:00

media_file

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

pacing

Fix pacer to accept duplicate sequence numbers on different SSRCs.

2014-07-07 10:20:35 +00:00

remote_bitrate_estimator

Fix test issues and a win compile error introduced with r6605.

2014-07-04 17:00:06 +00:00

rtp_rtcp

Count total bytes sent in RTPSender::Bytes().

2014-07-10 16:24:54 +00:00

utility

Some refactoring inside rtp_rtcp/.

2014-07-08 12:10:51 +00:00

video_capture

Add tkchin@ to OWNERS.

2014-07-01 18:38:28 +00:00

video_coding

Fix data race in VCMTiming::ResetDecodeTime.

2014-07-10 15:25:37 +00:00

video_processing

Add initial gn build files for video_coding and video_processing.

2014-07-07 11:46:43 +00:00

video_render

Add tkchin@ to OWNERS.

2014-07-01 18:38:28 +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

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules_unittests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules.gyp

Receiver bit-exactness test for AudioCoding Module

2014-06-26 19:07:04 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00