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mellem 90bdb3d138 Make WebRtcAudioEffects and its create method public. 
This allows it to be reused with Android AudioRecords created outside
WebRtcAudioRecord.  WebRtcAudioEffects provides useful methods for enabling
hardware effects (such as echo cancellation) only on appropriate devices.  It
also allows some control of these effects through WebRtcAudioUtils.

BUG=webrtc:7448

Review-Url: https://codereview.webrtc.org/2786603004
Cr-Commit-Position: refs/heads/master@{#18053}
2017-05-08 17:21:30 +00:00
..
api
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
audio
Add --quick flag to low bandwidth audio test
2017-05-04 10:06:18 +00:00
base
Delete unused class SharedExclusiveLock.
2017-05-08 13:24:29 +00:00
call
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
2017-05-02 13:46:30 +00:00
common_audio
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
common_video
Actually move CoreVideoFrameBuffer from webrtc/common_video/ to webrtc/sdk/objc/
2017-05-08 12:32:05 +00:00
examples
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
logging
Create tool to print statistics about the file size usage of an RTC event log.
2017-04-28 14:48:17 +00:00
media
Update comments for removal of MediaController.
2017-05-08 10:21:43 +00:00
modules
Make WebRtcAudioEffects and its create method public.
2017-05-08 17:21:30 +00:00
ortc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
p2p
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
pc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
sdk
Actually move CoreVideoFrameBuffer from webrtc/common_video/ to webrtc/sdk/objc/
2017-05-08 12:32:05 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
test
Fixing pseudotcp_parser_fuzzer crash with NO_MAIN_THREAD_WRAPPING.
2017-05-04 19:16:28 +00:00
tools
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
video
Remove deprecated API
2017-05-08 09:59:38 +00:00
voice_engine
Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()
2017-05-03 15:30:15 +00:00
.gitignore
…
BUILD.gn
Remove config.h from //webrtc:webrtc target.
2017-05-03 12:28:46 +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
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_frame.h
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ )
2017-05-04 09:11:12 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Remove the expat and jsoncpp dependencies to reduce binary size.
2017-04-24 19:05:06 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.