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webrtc_m130/webrtc
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Bjorn Volcker 3cfa756f37 audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz 
In AEC a fixed fft size is used, but processing can in the lower band be in either 8 or 16 kHz.
Therefore we need a multiplier/rate factor to, for example, map frequency bands in Hz to frequency bins.

The multiplier/rate factor can only be either 1 or 2, but when 48 kHz support was added it was assigned 3.

BUG=crbug.com/482424
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43329004

Cr-Commit-Position: refs/heads/master@{#9117}
2015-04-29 18:22:50 +00:00
..
base
Added a BitBufferWriter subclass that contains methods for writing bit and byte-sized data, along with exponential golomb encoded data.
2015-04-28 22:13:50 +00:00
build
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
common_audio
Add 3-band filter-bank implementation
2015-04-27 18:34:16 +00:00
common_video
Remove i420_video_frame.h from common_video.gyp
2015-04-10 12:40:21 +00:00
examples/android
Use short include paths for icu headers.
2015-04-23 06:58:02 +00:00
libjingle
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
modules
audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz
2015-04-29 18:22:50 +00:00
overrides
Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.
2015-04-04 23:56:56 +00:00
p2p
Add safeguards against signalling peer-reflexive candidates.
2015-04-28 16:06:34 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Set correct .type directive for asm functions.
2015-04-28 20:51:57 +00:00
test
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
tools
Disables mic bump-up level if not built with chromium
2015-04-22 04:39:47 +00:00
video
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
video_engine
Remove ViEImageProcess usage in VideoSendStream.
2015-04-29 12:08:49 +00:00
voice_engine
Added SetBitRate function to VoE API to allow changing the audio bitrate.
2015-04-29 14:03:45 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +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
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_encoder.h
Dynamic resolution change for VP8 HW encode.
2015-04-21 22:29:53 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
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
Use IncomingVideoStream in VideoReceiveStream.
2015-04-24 13:15:40 +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
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
Add a BWE fairness test.
2015-04-16 18:20:26 +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.