platform-external-webrtc

Files

henrik.lundin@webrtc.org 4d363ae305 The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.

This part of the the aftermath of issue 3255.

BUG=3255
R=andrew@webrtc.org, henrike@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6117 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-05-13 08:50:02 +00:00

build

Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."

2014-05-13 08:15:48 +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

The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.

2014-05-13 08:50:02 +00:00

system_wrappers

Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."

2014-05-13 08:15:48 +00:00

test

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

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

tools

Replace scoped_array<T> with scoped_ptr<T[]>.

2014-04-25 23:10:28 +00:00

video

Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.

2014-05-07 17:09:44 +00:00

video_engine

Fix failing test introduced with r6111.

2014-05-12 23:17:29 +00:00

voice_engine

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

2014-05-12 12:24:10 +00:00

.gitignore

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

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

call.h

Remove TraceCallback use from Call.

2014-04-24 11:35:33 +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

Make WebRTC Android examples build without sourcing envsetup.sh

2014-04-15 08:35:00 +00:00

webrtc_perf_tests.isolate

Move realtime tests to webrtc_perf_tests.

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

webrtc_tests.gypi

Fix the Android compilation (better structure for NetEq test libs)

2014-04-24 13:19:04 +00:00

webrtc.gyp

Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."

2014-05-13 08:15:48 +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.