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webrtc_m130/webrtc
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brandtr 03bc3cfb3e GN target for test_packet_masks_metrics. 
NOTRY=True
BUG=webrtc:6193

Review-Url: https://codereview.webrtc.org/2229473002
Cr-Commit-Position: refs/heads/master@{#13672}
2016-08-08 15:09:00 +00:00
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androidjunit
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api
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
audio
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
2016-08-04 12:28:28 +00:00
base
Add argument support for stand-alone WebRTC tracing.
2016-08-08 11:10:17 +00:00
build
Cleaning up mb_config.pyl style.
2016-08-08 13:59:24 +00:00
call
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common_audio
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common_video
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examples
GN: Add targets for AppRTCDemo on Mac+iOS
2016-08-08 04:37:56 +00:00
libjingle
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media
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
modules
GN target for test_packet_masks_metrics.
2016-08-08 15:09:00 +00:00
p2p
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
pc
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
sdk
iOS: Add support for rendering native CVPixelBuffers directly
2016-08-08 10:10:12 +00:00
system_wrappers
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test
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tools
Add fraction loss in visualization tool.
2016-08-04 17:00:19 +00:00
video
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
voice_engine
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.gitignore
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audio_receive_stream.h
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audio_send_stream.h
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audio_sink.h
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audio_state.h
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BUILD.gn
GN: Add targets for AppRTCDemo on Mac+iOS
2016-08-08 04:37:56 +00:00
call.h
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codereview.settings
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common_types.cc
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common_types.h
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common.gyp
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common.h
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config.cc
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config.h
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
DEPS
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engine_configurations.h
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LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
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PATENTS
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PRESUBMIT.py
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README.chromium
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rtc_unittests_apk.isolate
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rtc_unittests.isolate
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supplement.gypi
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transport.h
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typedefs.h
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video_decoder.h
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video_encoder.h
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video_engine_tests_apk.isolate
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video_engine_tests.isolate
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video_frame.h
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video_receive_stream.h
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
video_send_stream.h
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webrtc_examples.gyp
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webrtc_nonparallel_tests_apk.isolate
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webrtc_nonparallel_tests.isolate
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webrtc_perf_tests_apk.isolate
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webrtc_perf_tests.isolate
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webrtc_tests.gypi
Reland Enable socketservers unittests on windows.
2016-08-05 12:55:49 +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.