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webrtc_m130/webrtc
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maxmorin 1aee0b5bd9 Remove old methods in AudioTransport, make it pass a gn build 
when building with default warnings.

This is in preparation for making a gn target for audio_device_tests.

BUG=webrtc:6170, webrtc:163
NOTRY=True

Review-Url: https://codereview.webrtc.org/2219653004
Cr-Commit-Position: refs/heads/master@{#13759}
2016-08-15 18:46:28 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Remove old methods in AudioTransport, make it pass a gn build
2016-08-15 18:46:28 +00:00
audio
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
2016-08-04 12:28:28 +00:00
base
FakeTiming added, an implementation of Timing that can be used for
2016-08-15 14:00:06 +00:00
build
Add Mac bots to MB.
2016-08-11 08:04:39 +00:00
call
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
common_audio
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +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
Remove old methods in AudioTransport, make it pass a gn build
2016-08-15 18:46:28 +00:00
modules
Remove old methods in AudioTransport, make it pass a gn build
2016-08-15 18:46:28 +00:00
p2p
Fixing problems with ICE candidate pair prioritization.
2016-08-11 23:05:15 +00:00
pc
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
sdk
remove unnecessary double allocation
2016-08-15 10:42:08 +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
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
tools
Clarify some function names in visualization tool.
2016-08-15 15:47:21 +00:00
video
Make variable for selecting if intervals without samples should be included in stats configurable (for rate counters).
2016-08-15 08:20:37 +00:00
voice_engine
Remove old methods in AudioTransport, make it pass a gn build
2016-08-15 18:46:28 +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
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
…
BUILD.gn
Add video_loopback to gn.
2016-08-15 13:29:19 +00:00
call.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
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
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
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
Add periodic logging of video stats.
2016-08-11 15:41:26 +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
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Add video_loopback to gn.
2016-08-15 13:29:19 +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.