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terelius 2845a02339 Remove unused enum RTPDirections. 
BUG=

Review URL: https://codereview.webrtc.org/1582523002

Cr-Commit-Position: refs/heads/master@{#11221}
2016-01-12 13:01:02 +00:00
..
api
Update API for Objective-C RTCMediaSource.
2016-01-11 22:39:05 +00:00
audio
Wire-up BWE feedback for audio receive streams.
2016-01-12 12:55:11 +00:00
base
Change DTLS default from 1.0 to 1.2 for webrtc.
2016-01-11 23:27:12 +00:00
build
H.264: Default flags and pulling in openh264 and ffmpeg.
2016-01-11 18:19:06 +00:00
call
Wire-up BWE feedback for audio receive streams.
2016-01-12 12:55:11 +00:00
common_audio
Misc. small cleanups.
2016-01-08 21:50:32 +00:00
common_video
Add unit test for stand-alone denoiser and fixed some bugs.
2016-01-12 05:34:14 +00:00
examples
Use getExternalStorageDirectory() for trace file.
2015-12-30 17:23:35 +00:00
libjingle
Remove DISABLED_ON_ macros.
2016-01-04 21:44:16 +00:00
modules
Wire-up BWE feedback for audio receive streams.
2016-01-12 12:55:11 +00:00
p2p
Change DTLS default from 1.0 to 1.2 for webrtc.
2016-01-11 23:27:12 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Misc. small cleanups.
2016-01-08 21:50:32 +00:00
test
Add CreateSend/ReceiveTransport() methods to CallTest.
2016-01-08 14:47:21 +00:00
tools
Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
2015-12-10 11:11:48 +00:00
video
Add CreateSend/ReceiveTransport() methods to CallTest.
2016-01-08 14:47:21 +00:00
voice_engine
Wire-up BWE feedback for audio receive streams.
2016-01-12 12:55:11 +00:00
.gitignore
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audio_receive_stream.h
Wire-up BWE feedback for audio receive streams.
2016-01-12 12:55:11 +00:00
audio_send_stream.h
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audio_state.h
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BUILD.gn
Add VP8 and H264 depacketizer fuzzers.
2015-12-17 13:28:28 +00:00
call.h
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codereview.settings
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common_types.cc
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common_types.h
Remove unused enum RTPDirections.
2016-01-12 13:01:02 +00:00
common.gyp
…
common.h
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config.cc
…
config.h
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engine_configurations.h
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frame_callback.h
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LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
…
PATENTS
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PRESUBMIT.py
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README.chromium
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rtc_unittests.isolate
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stream.h
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supplement.gypi
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transport.h
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typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +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 unit test for stand-alone denoiser and fixed some bugs.
2016-01-12 05:34:14 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
…
video_send_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
Disable warnings failing when using Clang on Windows.
2015-12-16 22:05:36 +00:00
webrtc_nonparallel_tests.isolate
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webrtc_perf_tests.isolate
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webrtc_tests.gypi
Step 1 to prepare call_test.* for combined audio/video tests.
2015-12-21 11:14:05 +00:00
webrtc.gyp
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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.