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Alejandro Luebs fa639f0bb3 Surface the noise estimate of the NS to be used by other components 
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://codereview.webrtc.org/1654443004 .

Cr-Commit-Position: refs/heads/master@{#11541}
2016-02-09 19:24:51 +00:00
..
api
Fix GYP and GN references that are invalid in Chromium builds.
2016-02-08 07:04:33 +00:00
audio
Clean up of CongestionController.
2016-02-08 13:31:53 +00:00
base
Added thread annotations to FifoBuffer.
2016-02-09 10:30:43 +00:00
build
Revert of Analyze support in gyp_webrtc (patchset #1 id:1 of https://codereview.webrtc.org/1369683004/ )
2016-02-09 07:46:39 +00:00
call
Clean up of CongestionController.
2016-02-08 13:31:53 +00:00
common_audio
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common_video
H264: Improve FFmpeg decoder performance by using I420BufferPool.
2016-02-05 16:08:39 +00:00
examples
Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete
2016-02-09 11:09:50 +00:00
libjingle
…
media
Add perkj+magjed to webrtc/media/OWNERS
2016-02-08 14:46:04 +00:00
modules
Surface the noise estimate of the NS to be used by other components
2016-02-09 19:24:51 +00:00
p2p
Create QuicSession
2016-02-09 01:35:55 +00:00
sound
…
system_wrappers
Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics).
2016-02-04 08:33:29 +00:00
test
Fuzzer tests for AudioDecoder's DecodeRedundant and IncomingPacket
2016-02-09 08:00:32 +00:00
tools
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
video
Clean up of CongestionController.
2016-02-08 13:31:53 +00:00
voice_engine
Remove mutable from rtc::CriticalSections.
2016-02-01 17:00:59 +00:00
.gitignore
…
audio_receive_stream.h
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
2016-02-04 12:12:31 +00:00
audio_send_stream.h
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audio_state.h
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BUILD.gn
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
call.h
…
codereview.settings
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common_types.cc
…
common_types.h
…
common.gyp
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common.h
…
config.cc
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
config.h
…
engine_configurations.h
…
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
libjingle_media_unittest.isolate
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
…
PATENTS
…
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
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video_decoder.h
Implement NullVideoDecoder to avoid crash on unsupported decoders.
2016-02-03 13:51:56 +00:00
video_encoder.h
…
video_engine_tests.isolate
…
video_frame.h
…
video_receive_stream.h
…
video_renderer.h
…
video_send_stream.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
webrtc_examples.gyp
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtc.gyp
Move talk/media to webrtc/media
2016-02-05 07:52:35 +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.