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aluebs@webrtc.org 79b9eba3ab Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands 
Implemented the 3 bands splitting filter bank by:
1. Upsample by 4/3.
2. Split twice into 2 bands.
3. Discard upper most band, because it is empty anyway.

A unittest was also implemented:
1. Generate a signal from presence or absence of sine waves of different frequencies.
2. Split into 3 bands and check their presence or absence.
3. Recombine the bands.
4. Calculate delay (as it is an IIR it depends on frequency).
5. Check that the cross correlation of input and output is high enough at that delay.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7754 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 20:21:38 +00:00
..
base
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
build
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
common_audio
Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands
2014-11-26 20:21:38 +00:00
common_video
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
examples
Update Android projects to API level 21.
2014-10-31 23:26:10 +00:00
libjingle
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
modules
Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands
2014-11-26 20:21:38 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
p2p
Fix an ASSERT that fires in a browser test for renegotiation.
2014-11-26 19:58:50 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
test
Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
2014-11-26 15:50:30 +00:00
tools
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video
Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
2014-11-26 15:50:30 +00:00
video_engine
OWNERS: Remove tomasl@ and mallinath@
2014-11-24 10:05:05 +00:00
voice_engine
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
.gitignore
…
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
call.h
Implement settable min/start/max bitrates in Call.
2014-11-25 14:03:34 +00:00
common_types.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Log formatting fix for VideoEncoderConfig.
2014-11-06 09:35:08 +00:00
config.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
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
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Put send-side bwe probing under finch experiment.
2014-11-10 13:55:16 +00:00
webrtc.gyp
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +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.