webrtc_m130

History

stefan@webrtc.org fe16167507 Fix RTT calculations for send-only channels.

As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations.

BUG=3781
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7097 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-08 08:45:25 +00:00

base

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

build

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

common_audio

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

common_video

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

examples

Remove former team members from OWNERS and WATCHLISTS

2014-08-26 06:12:08 +00:00

libjingle/xmllite

Fix strange owners files with comments that crashs "git cl presubmit"

2014-09-04 14:25:07 +00:00

modules

Ignore FEC packet in stats, if it is first packet on ssrc.

2014-09-08 08:20:18 +00:00

overrides/webrtc/base

Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.

2014-05-21 16:52:14 +00:00

sound

Add talk owners to migrated talk folders

2014-08-28 16:03:58 +00:00

system_wrappers

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

test

Drop buildbot_tests.py script

2014-09-05 08:25:38 +00:00

tools

Moving the api.js and bot.js to /rtcbot/bot/ to be shared between

2014-09-04 13:44:47 +00:00

video

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +00:00

video_engine

Fix RTT calculations for send-only channels.

2014-09-08 08:45:25 +00:00

voice_engine

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

BUILD.gn

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

call.h

Network up/down signaling in Call.

2014-09-03 16:17:12 +00:00

common_types.h

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +00:00

common.gyp

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.cc

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

config.h

Check before send/receive rtp header extensions.

2014-07-20 15:27:35 +00:00

engine_configurations.h

Add boilerplate code for H.264.

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

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

2014-07-22 07:17:17 +00:00

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

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

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

supplement.gypi

Roll chromium_revision 289723:291647

2014-08-25 14:16:32 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Add CHECK and friends from Chromium.

2014-08-28 16:28:26 +00:00

video_engine_tests.isolate

Pass GYP DEPTH variable to isolate.

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

video_receive_stream.h

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +00:00

video_renderer.h

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +00:00

video_send_stream.h

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +00:00

webrtc_examples.gyp

Fix rm command for class cleanup in r7091

2014-09-06 22:11:28 +00:00

webrtc_perf_tests.isolate

Pass GYP DEPTH variable to isolate.

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

webrtc_tests.gypi

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

webrtc.gyp

Create a copy of talk/xmllite under webrtc/xmllite.

2014-09-02 15:41:12 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.