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View Code? Open in Web Editor NEWTDHS (time domain harmonic scaling) library with command-line demo
License: BSD 3-Clause "New" or "Revised" License
TDHS (time domain harmonic scaling) library with command-line demo
License: BSD 3-Clause "New" or "Revised" License
//////////////////////////////////////////////////////////////////////////// // **** AUDIO-STRETCH **** // // Time Domain Harmonic Scaler // // Copyright (c) 2022 David Bryant // // All Rights Reserved. // // Distributed under the BSD Software License (see license.txt) // //////////////////////////////////////////////////////////////////////////// New Python wrapper for audio-stretch: https://github.com/twardoch/audiostretchy From Wikipedia, the free encyclopedia: Time-domain harmonic scaling (TDHS) is a method for time-scale modification of speech (or other audio signals), allowing the apparent rate of speech articulation to be changed without affecting the pitch-contour and the time-evolution of the formant structure. TDHS differs from other time-scale modification algorithms in that time-scaling operations are performed in the time domain (not the frequency domain). This project is an implementation of a TDHS library and a command-line demo program to utilize it with standard WAV files. The command-line program also incorporates silence detection so that can be handled differently. There are two effects possible with TDHS and the audio-stretch demo. The first is the more obvious mentioned above of changing the duration (or speed) of a speech (or other audio) sample without modifying its pitch. The other effect is similar, but after applying the duration change we change the sampling rate in a complimentary manner to restore the original duration and timing, which then results in the pitch being altered. So when a ratio is supplied to the audio-stretch program, the default operation is for the total duration of the audio file to be scaled by exactly that ratio (0.5X to 2.0X), with the pitches remaining constant. If the option to scale the sample-rate proportionally is specified (-s) then the total duration and timing of the audio file will be preserved, but the pitches will be scaled by the specified ratio instead. This is useful for creating a "helium voice" effect and lots of other fun stuff. Note that unless ratios of exactly 0.5 or 2.0 are used with the -s option, non-standard sampling rates will probably result. Many programs will still properly play these files, and audio editing programs will likely import them correctly (by resampling), but it is possible that some applications will barf on them. They can also be resampled to a standard rate using an audio resampling tool I wrote that's also available here on GitHub: https://github.com/dbry/audio-resampler There's an option to cycle through the full possible ratio range in a sinusoidal pattern, starting at 1.0, and either going up (-c) or down (-cc) first. In this case any specified ratio is ignored (except if the -s option is also specified to scale the sampling rate). The total period is fixed at 2ฯ seconds, at which point the output will again be exactly aligned with the input. *** Version 0.4 Enhancements *** For version 0.4 two useful features were added. First, the ability to cascade two instances of the stretcher was added. This is enabled by including the flag STRETCH_DUAL_FLAG when initializing the stretcher and allows double the stretch ratio of the regular code (i.e., now 0.25X to 4.00X). Note that the audio quality degrades some when slowed beyond 2X, and generally voice becomes unintelligible when sped faster than 2X, however these values may still be useful for some applications, and specifically the very high speed values are useful for silence gaps (see the next feature). The other feature added is the ability to detect silence gaps in the audio and apply a different (likely lower) stretch ratio to these areas. This is currently not performed in the library itself, but in the demo command-line program where it is highly configurable, but it should be relatively easy to copy the functionality into another application. If I get requests for it, I will consider moving it into the library. There is a script to build the demo app on Linux (build.sh), and this also allows building the app to test for UB (undefined behavior) and ASAN (bad addressing). Also, some artificial test signals (both mono and stereo) and a script (test.sh) for running them at various ratios has been added. The current "help" display from the demo app: AUDIO-STRETCH Time Domain Harmonic Scaling Demo Version 0.4 Copyright (c) 2022 David Bryant. All Rights Reserved. Usage: AUDIO-STRETCH [-options] infile.wav outfile.wav Options: -r<n.n> = stretch ratio (0.25 to 4.0, default = 1.0) -g<n.n> = gap/silence stretch ratio (if different) -u<n> = upper freq period limit (default = 333 Hz) -l<n> = lower freq period limit (default = 55 Hz) -b<n> = audio buffer/window length (ms, default = 25) -t<n> = gap/silence threshold (dB re FS, default = -40) -c = cycle through all ratios, starting higher -cc = cycle through all ratios, starting lower -d = force dual instance even for shallow ratios -s = scale rate to preserve duration (not pitch) -f = fast pitch detection (default >= 32 kHz) -n = normal pitch detection (default < 32 kHz) -q = quiet mode (display errors only) -v = verbose (display lots of info) -y = overwrite outfile if it exists Web: Visit www.github.com/dbry/audio-stretch for latest version Notes: 1. The program will handle only mono or stereo files in the WAV format. In case of stereo, the two channels shouldn't be independent. The audio must be 16-bit PCM and the acceptable sampling rates are from 8,000 to 48,000 Hz. Any additional RIFF info in the WAV file will be discarded. The command-line program is only for little-endian architectures. 2. For stereo files, the pitch detection is done on a mono conversion of the audio, but the scaling transformation is done on the independent channels. If it is desired to have completely independent processing this can only be done with two mono files. Note that this is not a limitation of the library but of the demo utility (the library has no problem with multiple contexts). 3. This technique (TDHS) is ideal for speech signals, but can also be used for homophonic musical instruments. As the sound becomes increasingly polyphonic, however, the quality and effectiveness will decrease. Also, the period frequency limits provided by default are optimized for speech; adjusting these may be required for best quality with non-speech audio. 4. The vast majority of the time required for TDHS is in the pitch detection, and so this library implements two versions. The first is the standard one that includes every sample and pitch period, and the second is an optimized one that uses pairs of samples and only even pitch periods. This second version is about 4X faster than the standard version, but provides virtually the same quality. It is used by default for files with sample rates of 32 kHz or higher, but its use can be forced on or off from the command-line (see options above).
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