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henrika 8eadead3f4 Adds support for USB audio devices in AppRTCMobile on Android. 
This change extends the definition of wired headset to also include USB
devices. The effect is that audio will now be routed to USB audio devices
when used in combination with AppRTCMobile.

BUG=webrtc:7931

Review-Url: https://codereview.webrtc.org/2971613003
Cr-Commit-Position: refs/heads/master@{#18889}
2017-07-04 12:10:48 +00:00
..
api
Reland of "VideoFrameBuffer: Remove deprecated functions"
2017-06-30 16:39:27 +00:00
audio
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
base
Delete SignalThread class.
2017-06-29 13:21:20 +00:00
call
Reland of Add received audio/video call duration metrics based on packets.
2017-07-04 11:05:06 +00:00
common_audio
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
common_video
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
examples
Adds support for USB audio devices in AppRTCMobile on Android.
2017-07-04 12:10:48 +00:00
logging
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
media
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
modules
Reduce send rate to 50% if overusing before we have an acknowledged bitrate.
2017-07-04 11:52:58 +00:00
ortc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
p2p
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
pc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
rtc_base
Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)"
2017-06-30 17:45:21 +00:00
rtc_tools
fix comment length
2017-07-03 07:47:44 +00:00
sdk
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
stats
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
system_wrappers
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
test
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
tools
Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ )
2017-07-04 09:02:49 +00:00
video
Reland of Add received audio/video call duration metrics based on packets.
2017-07-04 11:05:06 +00:00
voice_engine
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
.gitignore
…
BUILD.gn
Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )
2017-06-30 15:11:44 +00:00
codereview.settings
…
common_types.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
common_types.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
config_unittest.cc
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
config.cc
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
config.h
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
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_receive_stream.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_send_stream.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +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.