Name | Modified | Size | Downloads / Week |
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Parent folder | |||
WSOLA.v.4.0.tar.bz2 | 2004-11-24 | 75.8 kB | |
README | 2004-11-23 | 3.7 kB | |
WSOLA.tar.bz2 | 2004-04-05 | 48.9 kB | |
WSOLA.3.4.tar.bz2 | 2003-08-08 | 67.1 kB | |
MFFM_FFTw_Wrapper-v1.4.tar.bz2 | 2003-08-08 | 73.8 kB | |
MFFM_FFTw_Wrapper-v1.3.tar.bz2 | 2003-08-08 | 163.5 kB | |
WSOLA.2.7.zip | 2003-05-13 | 844.8 kB | |
WSOLA.2.7.tar.bz2 | 2003-05-13 | 61.9 kB | |
WSOLA.3.3.zip | 2003-05-10 | 1.9 MB | |
WSOLA.3.3.tar.bz2 | 2003-05-10 | 76.4 kB | |
WSOLA.zip | 2003-03-03 | 1.9 MB | |
MFFM_TimeScale.zip | 2002-04-03 | 131.5 kB | |
MFFM_TimeScale.tar.gz | 2002-01-02 | 11.7 kB | |
MFFM_TimeScale.tar.bz2 | 2002-01-02 | 11.1 kB | |
Totals: 14 Items | 5.4 MB | 0 |
Happy user support page : http://sourceforge.net/donate/index.php?group_id=40316 Copyright 2001 Matt Flax <flatmax at ieee d0t org> This application stretches and compresses audio without altering the frequency character of the audio. For reasonable factors, this application will scale audio without altering signal levels or introducing artifacts (in the ideal implementation). The v4.x stream of this project is now targeting embedded operation. The memory footprint is approaching about 2.5 k Bytes to 7.5 k bytes for a mono stream of audio and about 5 k bytes to 15 k Bytes for stereo streams. At this stage the DFT based approach adopted in version 3.x is stable and has excellent sound quiality even for very fast and very slow time scalings. The code has been completely re-worked and this has sped up operation to a large degree. As of Version 3.0, this implementation of WSOLA is now approximatly six times faster then real time (800MHz CPU with coprocessor). Operational time on a 24 second stereo sample with a factor of 1.0 : Linux takes 6 seconds. It is completely stable. Microsoft takes approximatly the same ammout of time. (Using Cygwin GNU*NIX translation) Requirements : * This program can read alot of file types because of the wrapper to libsndfile : http://sourceforge.net/projects/mffmlibsndfilew/ * This program requires an installed version of MFFM multimedia time code handling classes. Try : http://mffmtimecode.sourceforge.net/ For fast operation (> v 3.* only), you will also require MFFM FFTw C++ wrapper. Try: http://mffmfftwrapper.sourceforge.net/ Audio files are read and written using LibSndFile v1 : http://www.zip.com.au/~erikd/libsndfile/ Finally you require a C++ compiler, try : http://gcc.gnu.org/install/binaries.html http://www.cygwin.com (Microsoft users) MS Windows BINARY users wiil require the file 'cygwin1.dll'. If it is not shipped with this zip package then please try to find it at Cygwin: http://www.cygwin.com My other projects : http://sourceforge.net/search/?type_of_search=soft&words=mffm This project's Home Page : http://mffmtimescale.sourceforge.net MFFM Time Scale Modification for Audio is 2 things : a] A compilable program WSOLATest.C which allow you to time stretch and compress mono audio files. Audio files are restricted to be mono 16 bit frame sized. b] A set of 2 header files which are the implementation of [1]. For simple use .... Type 'make' and compile the program WSOLATest Run WSOLA like so : WSOLA inputFile outputFile factor factor = 0.5 for halving the duration of an audio file factor = 2.0 for doubling the duration of an audio file factor = 1.0 for an identical file. [1]"An overlap-add technique based on waveform similarity (WSOLA) for high quality time-scale modification of speech", Verhelst, W.; Roelands, M. Acoustics, Speech, and Signal Processing, 1993. ICASSP-93., 1993 IEEE International Conference on On page(s): 554 - 557 vol.2 27-30 April 1993 Minneapolis, MN, USA 1993 Volume: 2 ISBN: 0-7803-0946-4 Number of Pages: 5 vol. (652+735+606+559+681) References Cited: 4 INSPEC Accession Number: 4771035 Abstract: A concept of waveform similarity for tackling the problem of time-scale modification of speech is proposed. It is worked out in the context of short-time Fourier transform representations. The resulting WSOLA (waveform-similarity-based synchronized overlap-add) algorithm produces high-quality speech output, is algorithmically and computationally efficient and robust, and allows for online processing with arbitrary time-scaling factors that may be specified in a time-varying fashion and can be chosen over a wide continuous range of values.