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webrtc_m130/webrtc
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henrika fe55c38eff Removes automatic setting of COMM mode in WebRTC. 
It is now up to the application to ensure that it is in COMM mode before any audio streaming is started.

BUG=b/21571563
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1165923002

Cr-Commit-Position: refs/heads/master@{#9383}
2015-06-05 09:46:02 +00:00
..
base
Remove win32toolhelp.h.
2015-06-04 08:25:12 +00:00
build
Roll chromium_revision ccef3cb..7779e7d (331232:332119)
2015-06-01 09:49:28 +00:00
common_audio
Add LappedTransform accessors.
2015-05-29 01:01:43 +00:00
common_video
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
examples/android/media_demo
VoE: Remove unused interfaces
2015-05-26 08:25:00 +00:00
libjingle
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
2015-05-29 13:05:52 +00:00
modules
Removes automatic setting of COMM mode in WebRTC.
2015-06-05 09:46:02 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Disable some PortAllocatorTest on asan due to flakiness
2015-06-05 07:41:36 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Use one scoped_refptr.
2015-06-04 13:18:28 +00:00
test
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
tools
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
video
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_engine
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
voice_engine
Add options for NetEq fast accelerate mode through libjingle
2015-06-01 08:29:55 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
call.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add options for NetEq fast accelerate mode through libjingle
2015-06-01 08:29:55 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +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
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_encoder.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_receive_stream.h
Configure default render delay as 10 ms.
2015-05-27 15:59:21 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +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 HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.