webrtc_m130

History

henrike@webrtc.org cc774a69cb Mark all virtual overrides in the hierarchies of RtpDump and

VCMPacketizationCallback as such.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also marks all other such overrides in the affected files.

BUG=none
TEST=none
R=henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7161 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-11 22:45:54 +00:00

base

Restore webrtc_base target until r7140 is rolled into Chromium.

2014-09-11 09:22:13 +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

Mark all virtual overrides in the hierarchies of RtpDump and

2014-09-11 22:45:54 +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 hierarchies of RtpDump and

2014-09-11 22:45:54 +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.