webrtc

History

henrik.lundin@webrtc.org 3154a1cf9d Reland r8210 "Add a new parameter to ACMGenericCodec constructor""

This effectively reverts r8211.

The problem with r8210 was that the change in constructor signature was not done for other codec selections that then default one. That is, some code that was hidden under #ifdef did not get updated. This is now fixed.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37879004

Cr-Commit-Position: refs/heads/master@{#8215}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8215 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-01-30 12:30:22 +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

Reland r8210 "Add a new parameter to ACMGenericCodec constructor""

2015-01-30 12:30:22 +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

Remove unused method, SetAffinity, from the ThreadWrapper class.

2015-01-30 09:43:20 +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.