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michaelt 584c35a334 Make a the decisions of ANA optional for the opus encoder. 
BUG=webrtc:6936, webrtc:6303

Review-Url: https://codereview.webrtc.org/2592253004
Cr-Commit-Position: refs/heads/master@{#15807}
2016-12-27 16:21:29 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
2016-12-24 09:43:32 +00:00
audio
Add ossu@ to OWNERS of audio/ and modules/audio_coding/
2016-12-15 15:52:14 +00:00
base
Remove WebRTC-specific test runner script for Android
2016-12-27 08:21:19 +00:00
build
Remove WebRTC-specific test runner script for Android
2016-12-27 08:21:19 +00:00
call
Make class of static functions in rtp_to_ntp.h:
2016-12-22 15:53:51 +00:00
common_audio
Fix for left shift of negative value in NetEq.
2016-12-14 10:00:02 +00:00
common_video
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
examples
Created a java wrapper for the callback OnAddTrack to PeerConnection.Observer
2016-12-21 22:08:03 +00:00
libjingle
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
logging
Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.
2016-12-20 13:03:58 +00:00
media
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
2016-12-24 09:43:32 +00:00
modules
Make a the decisions of ANA optional for the opus encoder.
2016-12-27 16:21:29 +00:00
p2p
If network enumeration fails, try binding to the "ANY" address.
2016-12-23 09:43:45 +00:00
pc
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.
2016-12-22 18:53:38 +00:00
sdk
Created a java wrapper for the callback OnAddTrack to PeerConnection.Observer
2016-12-21 22:08:03 +00:00
stats
RTCStatsReport::AddStats DCHECKs that the ID is unique.
2016-12-21 09:29:05 +00:00
system_wrappers
Make class of static functions in rtp_to_ntp.h:
2016-12-22 15:53:51 +00:00
test
Remove WebRTC-specific test runner script for Android
2016-12-27 08:21:19 +00:00
tools
Add method needed to extract frame capture and arrival timestamps from rtc event logs.
2016-12-20 16:51:52 +00:00
video
Make class of static functions in rtp_to_ntp.h:
2016-12-22 15:53:51 +00:00
voice_engine
Drop unneeded include of media_file.h.
2016-12-21 11:05:44 +00:00
.gitignore
…
BUILD.gn
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.
2016-12-22 18:53:38 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
CodecInst operator<<
2016-12-19 14:04:04 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
2016-12-19 14:32:16 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +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.