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Andrew MacDonald 645299d4e0 Add frequency smoothing to postfilter. 
Reduces musical noise with minimal impact on interferer suppression.

This also unifies the treatment of "mean bins". The "end" bin is now
inclusive in the mean range as with the "start" bin.

Corrects interpretation of quantile.

BUG=chromium:490477
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50939004

Cr-Commit-Position: refs/heads/master@{#9317}
2015-05-28 20:10:32 +00:00
..
base
Prevent potential double-free if srtp_create fails.
2015-05-27 21:41:52 +00:00
build
Remove GetSendSideDelay from RtpRtcp.
2015-05-28 12:45:45 +00:00
common_audio
common_audio/vad: Removes head allocation failure check
2015-05-28 12:30:29 +00:00
common_video
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
examples/android/media_demo
VoE: Remove unused interfaces
2015-05-26 08:25:00 +00:00
libjingle
Some cleanup for base/logging and base/stream.h
2015-05-25 09:26:08 +00:00
modules
Add frequency smoothing to postfilter.
2015-05-28 20:10:32 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Stop video candidates getting down to audio.
2015-05-28 16:48:30 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
test
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
tools
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
video
Remove ViESender.
2015-05-28 12:10:48 +00:00
video_engine
Remove ViESender.
2015-05-28 12:10:48 +00:00
voice_engine
Remove remaining .mk files.
2015-05-28 09:05:11 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
PRESUBMIT: Improve PyLint check and add GN format check.
2015-05-25 10:55:50 +00:00
call.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
2015-05-11 10:44:20 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
…
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
Remove unused VideoDecoder methods.
2015-05-21 07:42:14 +00:00
video_encoder.h
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
video_engine_tests.isolate
…
video_frame.h
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
video_receive_stream.h
Configure default render delay as 10 ms.
2015-05-27 15:59:21 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.