webrtc

History

henrik.lundin@webrtc.org 6752b85ff7 Revert r8210 "Add a new parameter to ACMGenericCodec constructor"

The change failed to compile on some bots.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34949004

Cr-Commit-Position: refs/heads/master@{#8211}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8211 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-01-30 06:36:41 +00:00

base

Control the max IPv6 Networks used by WebRTC.

2015-01-30 00:09:42 +00:00

build

Enable Clang warning implicit-fallthrough and annotate the code.

2015-01-28 18:38:13 +00:00

common_audio

Move channel_buffer.{h,cc} to common_audio.

2015-01-28 19:57:44 +00:00

common_video

Make it easier to use external libyuv + cleanup GYP files.

2015-01-26 19:17:26 +00:00

examples

Add arraysize() macro from Chromium, and make use of it in a few places.

2015-01-28 21:37:13 +00:00

libjingle

Enable Clang warning implicit-fallthrough and annotate the code.

2015-01-28 18:38:13 +00:00

modules

Revert r8210 "Add a new parameter to ACMGenericCodec constructor"

2015-01-30 06:36:41 +00:00

overrides

Remove COMPILE_ASSERT and use static_assert everywhere

2015-01-14 10:51:54 +00:00

p2p

Adding constness.

2015-01-28 17:33:45 +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

Change a GYP reference to cpufeatures.gypi

2015-01-23 16:35:17 +00:00

test

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

tools

Re-land "Remove <(webrtc_root) from source file entries."

2015-01-29 14:30:41 +00:00

video

Add tracing for slow paths in new video API.

2015-01-29 12:33:42 +00:00

video_engine

Revert 8203 "Reducing locking in OveruseFrameDetector and increa..."

2015-01-29 16:09:07 +00:00

voice_engine

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

.gitignore

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

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

BUILD.gn

Make it easier to use external libyuv + cleanup GYP files.

2015-01-26 19:17:26 +00:00

call.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

2014-12-05 13:43:35 +00:00

common_types.h

Update StreamDataCounter with FEC bytes.

2015-01-27 12:17:29 +00:00

common.gyp

Make it easier to use external libyuv + cleanup GYP files.

2015-01-26 19:17:26 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.cc

Revert 8028 "Support associated payload type when registering Rt..."

2015-01-09 20:22:46 +00:00

config.h

Revert 8028 "Support associated payload type when registering Rt..."

2015-01-09 20:22:46 +00:00

engine_configurations.h

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +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

Update all .isolate files for the new format.

2014-10-31 18:08:09 +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

Enable Clang warning implicit-fallthrough and annotate the code.

2015-01-28 18:38:13 +00:00

video_decoder.h

Add support for VP9 in webrtc::Call and video_loopback.

2014-11-04 19:41:15 +00:00

video_encoder.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

video_frame.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_receive_stream.h

Log configs when creating video streams in Call.

2015-01-15 10:09:39 +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

Log configs when creating video streams in Call.

2015-01-15 10:09:39 +00:00

webrtc_examples.gyp

Move internal capture+render to build_with_chromium==0 condition

2015-01-20 11:40:45 +00:00

webrtc_perf_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

webrtc_tests.gypi

Re-land "Remove <(webrtc_root) from source file entries."

2015-01-29 14:30:41 +00:00

webrtc.gyp

Make it easier to use external libyuv + cleanup GYP files.

2015-01-26 19:17:26 +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.