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webrtc_m130/webrtc
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Henrik Kjellander 5ddee021dd Landmine: clobber to remove out/{Debug,Release}/args.gn 
Landmine support was added back in
https://codereview.webrtc.org/1402923003/

BUG=webrtc:5070,webrtc:5123
TBR=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1415453006 .

Cr-Commit-Position: refs/heads/master@{#10426}
2015-10-27 11:00:00 +00:00
..
audio
Implement AudioSendStream::GetStats().
2015-10-27 10:35:30 +00:00
base
Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
2015-10-26 21:11:25 +00:00
build
Landmine: clobber to remove out/{Debug,Release}/args.gn
2015-10-27 11:00:00 +00:00
call
Implement AudioSendStream::GetStats().
2015-10-27 10:35:30 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Android: Replace EGL14 with EGL10
2015-10-22 23:52:45 +00:00
libjingle
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
modules
Use webrtc/base/checks.h in desktop_capture.
2015-10-27 10:43:11 +00:00
p2p
Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
2015-10-27 02:21:45 +00:00
sound
Revert "Add GN Build file for rtc_sound target."
2015-10-26 16:39:21 +00:00
system_wrappers
clean up field_trial_default target, to be used by remoting_perftests.
2015-10-23 02:33:22 +00:00
test
Implement AudioSendStream::GetStats().
2015-10-27 10:35:30 +00:00
tools
Fix minor GYP error in webrtc/tools/internal_tools.gyp
2015-10-07 12:51:40 +00:00
video
Add stats for used video codec type for a sent video stream:
2015-10-27 08:32:06 +00:00
video_engine
Use webrtc/base/logging.h for video.
2015-10-26 10:35:26 +00:00
voice_engine
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Implement AudioSendStream::GetStats().
2015-10-27 10:35:30 +00:00
BUILD.gn
Revert "Add GN Build file for rtc_sound target."
2015-10-26 16:39:21 +00:00
call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
codereview.settings
Update lower-level codereview.settings files.
2015-10-06 21:19:34 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +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
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow encoders to fall back dynamically to software.
2015-10-22 06:54:57 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
webrtc.gyp
Remove references to libpeerconnection.
2015-10-08 12:41:05 +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.