webrtc_m130

History

asapersson@webrtc.org 9aed002090 Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame.

Controlled by setting enable_extended_processing_usage. Enabled by default.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7460 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-10-16 06:57:12 +00:00

base

Fix data races in ThreadTest.ThreeThreadsInvoke.

2014-10-15 14:54:56 +00:00

build

Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.

2014-10-15 17:30:28 +00:00

common_audio

common_audio: Removed version API from signal_processing

2014-10-15 04:38:42 +00:00

common_video

GN: Add common configs to all targets.

2014-09-28 17:37:22 +00:00

examples

Split video engine android initialization into each internal module initialization.

2014-09-17 11:44:51 +00:00

libjingle

Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.

2014-10-15 17:30:28 +00:00

modules

audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>

2014-10-15 12:51:23 +00:00

overrides

webrtc/overrides: add OWNERS-file.

2014-09-17 08:04:28 +00:00

p2p

Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.

2014-10-15 17:30:28 +00:00

sound

rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.

2014-10-01 16:33:03 +00:00

system_wrappers

Import LappedTransform and friends.

2014-10-01 17:42:18 +00:00

test

GN: Add common configs to tools and test.

2014-09-30 19:07:58 +00:00

tools

Selecting bot_type changed to be specified in the test file

2014-10-15 15:01:11 +00:00

video

Remove -1 from Call::Config::start_bitrate_bps.

2014-10-14 11:52:10 +00:00

video_engine

Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame.

2014-10-16 06:57:12 +00:00

voice_engine

Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.

2014-10-10 09:42:53 +00:00

.gitignore

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

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

BUILD.gn

GN: Enable libvpx, add link target and convert some test targets

2014-09-30 18:05:02 +00:00

call.h

Remove -1 from Call::Config::start_bitrate_bps.

2014-10-14 11:52:10 +00:00

common_types.h

Add VP9 codec to VCM and vie_auto_test.

2014-10-10 16:44:47 +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

Config struct for VideoEncoder.

2014-09-19 12:30:25 +00:00

engine_configurations.h

Add VP9 codec to VCM and vie_auto_test.

2014-10-10 16:44:47 +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

rtc_unittests.isolate

Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.

2014-09-30 14:21:10 +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_encoder.h

Add VP9 codec to VCM and vie_auto_test.

2014-10-10 16:44:47 +00:00

video_engine_tests.isolate

Pass GYP DEPTH variable to isolate.

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

video_frame.h

Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""

2014-09-17 09:02:25 +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

Config struct for VideoEncoder.

2014-09-19 12:30:25 +00:00

webrtc_examples.gyp

Roll chromium_revision 6455c69..deaf2f7 (293954:295079)

2014-09-27 18:10:30 +00:00

webrtc_perf_tests.isolate

Pass GYP DEPTH variable to isolate.

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

webrtc_tests.gypi

Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.

2014-10-15 17:30:28 +00:00

webrtc.gyp

Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.

2014-10-15 17:30:28 +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.