webrtc

History

wu@webrtc.org 54231f0662 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log.

BUG=crbug/371714
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6166 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-05-14 23:06:23 +00:00

base

Added namespace rtc to some base classes and functions. It was causing linker error in the FYI bots: http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Android%20Builder%20%28dbg%29/builds/1808/steps/compile/logs/stdio but also, not doing it pollutes the global namespace.

2014-05-14 18:24:13 +00:00

build

Adds a modified copy of talk/base to webrtc/base. It is the first step in

2014-05-13 18:00:26 +00:00

common_audio

Fix constness of AudioBuffer accessors.

2014-04-30 16:44:13 +00:00

common_video

Added include of assert.h for files calling assert but missing the include.

2014-04-29 17:54:17 +00:00

examples

WebRTCDemo: correct set trace filter operation.

2014-05-05 04:24:47 +00:00

modules

Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.

2014-05-14 19:00:59 +00:00

overrides/webrtc/base

Adds a modified copy of talk/base to webrtc/base. It is the first step in

2014-05-13 18:00:26 +00:00

system_wrappers

Add webrtc field trials API.

2014-05-14 12:24:04 +00:00

test

Add DeliveryStatus enum to DeliverPacket().

2014-05-14 13:57:12 +00:00

tools

Add webrtc field trials API.

2014-05-14 12:24:04 +00:00

video

Add DeliveryStatus enum to DeliverPacket().

2014-05-14 13:57:12 +00:00

video_engine

Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log.

2014-05-14 23:06:23 +00:00

voice_engine

Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.

2014-05-14 19:00:59 +00:00

.gitignore

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

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

call.h

Add DeliveryStatus enum to DeliverPacket().

2014-05-14 13:57:12 +00:00

common_types.h

Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65

2014-04-29 17:50:47 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

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

config.h

Wire up statistics in video receive stream of new API

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

engine_configurations.h

Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-05-12 12:19:19 +00:00

experiments.h

Adding API for setting bandwidth estimation configurations.

2014-03-25 10:37:31 +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

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

PATENTS

Move src/ -> webrtc/

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

PRESUBMIT.py

Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.

2013-07-22 22:32:50 +00:00

README.chromium

Move src/ -> webrtc/

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

supplement.gypi

Roll chromium_revision 260462:266514

2014-04-29 09:36:40 +00:00

transport.h

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

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

typedefs.h

Remove ALLOW_UNUSED.

2014-05-05 18:18:02 +00:00

video_engine_tests.isolate

Merge metrics_unittests into video_engine_tests.

2013-12-13 14:31:47 +00:00

video_receive_stream.h

Rename Start/Stop in Video{Send,Receive}Streams.

2014-04-24 11:13:21 +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

Rename Start/Stop in Video{Send,Receive}Streams.

2014-04-24 11:13:21 +00:00

webrtc_examples.gyp

Add webrtc field trials API.

2014-05-14 12:24:04 +00:00

webrtc_perf_tests.isolate

Move realtime tests to webrtc_perf_tests.

2013-12-13 12:48:05 +00:00

webrtc_tests.gypi

Add webrtc field trials API.

2014-05-14 12:24:04 +00:00

webrtc.gyp

Adds a modified copy of talk/base to webrtc/base. It is the first step in

2014-05-13 18:00:26 +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.