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webrtc_m130/webrtc
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maxmorin 3a11933a63 Remove audio_device_test_func. 
This code does not work and hasn't been used in a long time. It also
lacks a GN target. There's no reason to save it.

BUG=none

Review-Url: https://codereview.webrtc.org/2255173002
Cr-Commit-Position: refs/heads/master@{#13820}
2016-08-18 14:20:48 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Roll chromium_revision 915e47250f..e3860bd297 (412201:412289)
2016-08-18 08:45:53 +00:00
audio
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
base
Do not switch a connection if the new connection is not ready to send packets.
2016-08-17 01:19:21 +00:00
build
Fix get_landmines.py script.
2016-08-17 11:00:47 +00:00
call
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
common_audio
WebRtcSpl_SynthesisQMF: Fix UBSan fuzzer bug (left shift of negative value)
2016-08-18 10:17:10 +00:00
common_video
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
examples
Remove LooperExecutor and replace it with built-in HandlerThread.
2016-08-09 09:03:41 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Updated AudioDecoderFactory to list AudioCodecSpecs instead of SdpAudioFormats.
2016-08-18 09:02:15 +00:00
modules
Remove audio_device_test_func.
2016-08-18 14:20:48 +00:00
p2p
Do not switch to a high-cost connection that is not receiving.
2016-08-17 23:12:58 +00:00
pc
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
sdk
Nil out EAGLContext explicitly on RTCEAGLVideoView dealloc.
2016-08-17 23:03:09 +00:00
system_wrappers
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
test
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
tools
Don't simulate probing based on rtc event logs since we don't have that info logged.
2016-08-17 09:14:38 +00:00
video
Reland of Adding audio to video_quality_test.
2016-08-18 13:28:59 +00:00
voice_engine
voice_engine: Removed old variants of Channel constructor and CreateChannel
2016-08-18 11:23:04 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
audio_sink.h
…
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
GN: Add Android support to video_engine_tests.
2016-08-18 07:42:05 +00:00
call.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
codereview.settings
…
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.