webrtc

History

xians@webrtc.org 7aad5e5cce Revert 7338 "Fixed the android build by making the interface pur..."

> Fixed the android build by making the interface pure virtual.
> 
> TBR=asapersson@webrtc.org, bjornv@webrtc.org,
> 
> Review URL: https://webrtc-codereview.appspot.com/24789004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-30 15:26:15 +00:00

base

Reland disallowing blocking calls on the worker thread.

2014-09-29 22:45:55 +00:00

build

Revert 7327 "Update isolate.gypi files + link to isolate_driver.py"

2014-09-30 08:44:00 +00:00

common_audio

audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with <<

2014-09-29 10:56:27 +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/xmllite

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +00:00

modules

Revert 7338 "Fixed the android build by making the interface pur..."

2014-09-30 15:26:15 +00:00

overrides

webrtc/overrides: add OWNERS-file.

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

sound

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

2014-09-30 14:21:10 +00:00

system_wrappers

Set thread scheduling parameters inside the new thread.

2014-09-29 18:25:27 +00:00

test

Minor modifications to test::RtpFileReader

2014-09-30 11:08:44 +00:00

tools

Collecting stats every fixed time in webrtc_video_streaming.js test

2014-09-30 15:20:15 +00:00

video

GN: Add common configs to all targets.

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

video_engine

GN: Add common configs to all targets.

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

voice_engine

GN: Add common configs to all targets.

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

.gitignore

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

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

BUILD.gn

GN: Add common configs to all targets.

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

call.h

Config struct for VideoEncoder.

2014-09-19 12:30:25 +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

Config struct for VideoEncoder.

2014-09-19 12:30:25 +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

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

Expose VP8/H264 defaults through video_encoder.h.

2014-09-18 12:42:28 +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

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

2014-09-30 14:21:10 +00:00

webrtc.gyp

Split video_render_module implementation into default and internal implementation.

2014-09-18 08:58:15 +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.