webrtc

History

andresp@webrtc.org d2cf48de1a Fix mac video_render implementation on cocoa.

Hit this while playing around with all compile possibilities for issue 3770.

BUG=3770
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7166 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-12 13:57:47 +00:00

base

Fix stack limit exceeded in http client.

2014-09-12 13:35:05 +00:00

build

Split suppressons of thread.cc and messagequeue.cc.

2014-09-11 14:59:06 +00:00

common_audio

Add ctors to ChannelBuffer to enable copying on construction.

2014-09-08 20:27:04 +00:00

common_video

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

2014-09-11 09:48:30 +00:00

examples

Fix compile error on JDK 1.7.

2014-09-10 12:35:59 +00:00

libjingle/xmllite

Put base tests in webrtc_tests.gyp

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

modules

Fix mac video_render implementation on cocoa.

2014-09-12 13:57:47 +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

Put base tests in webrtc_tests.gyp

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

system_wrappers

Put base tests in webrtc_tests.gyp

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

test

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

2014-09-11 09:48:30 +00:00

tools

Fix printing of error stack in rtcbot when a test fails via test.fail().

2014-09-10 14:35:35 +00:00

video

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

2014-09-11 09:48:30 +00:00

video_engine

Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.

2014-09-12 07:42:33 +00:00

voice_engine

Calculating round-trip-time in send-only channel in VoE.

2014-09-11 07:51:53 +00:00

.gitignore

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

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

BUILD.gn

Fix the RTC+Chromium GN build.

2014-09-09 19:15:33 +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

Put base tests in webrtc_tests.gyp

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

webrtc.gyp

Put base tests in webrtc_tests.gyp

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