(c) Timothy James Becker 2016, Andy Selby 2024
- python 3.10.12+
# without virtual environment: pip install -r ./requirements.txt # with virtual environment in bash/zsh: python -m venv . . bin/activate pip install -r ./requirements.txt # with virtual environment in Powershell: python -m venv . .\bin\Activate.ps1 pip install -r .\requirements.txt
- 24 bit file handling is based on the python multiple bit depth converter tool wavio written by Warren Weckesser (c) 2015
- phase_vocoder written by Victor Lazzarini (c) 2010
./mungo_utils.py -h usage: mungo_utils.py [-h] [-I AUDIO_INPUT_DIR] [-E AUDIO_EXT] [-T TARGET_MUNGO_MODULE] [-O MUNGO_OUTPUT_DIR] [-m] [-n] [-f FADE] [-l] [-p] [-t]
Automated Mungo Enterprises Eurorack Focused WAV Bit Depth and Sample Rate Conversion Utilities Tested with WAV and AIFF audio files with 1|2 chnannels and 16/24 bit depths provide a folder glob pattern to search and this utility will match those full paths and one by one convert to the target Mungo Module based on the published buffer sizes there. For example the C0 has a buffer of 12000, so any audio file will be auto resampled to that size converted to a 16 bit WAV file format
-h, --help show this help message and exit -I AUDIO_INPUT_DIR, --audio_input_dir AUDIO_INPUT_DIR audio directory to search [required] -E AUDIO_EXT, --audio_ext AUDIO_EXT either or aif and wav audio extensions to convert [aif,wav] -T TARGET_MUNGO_MODULE, --target_mungo_module TARGET_MUNGO_MODULE the mungo target module to write to [G0=500K] -O MUNGO_OUTPUT_DIR, --mungo_output_dir MUNGO_OUTPUT_DIR mungo output directory [required] -m, --mix mix multiple channels [False] -n, --norm normalize audio and remove DC offset [False] -f FADE, --fade FADE target buffer fade out in samples default is exponential fade [256] -r, --reverse reverse the buffer [False] -p, --phase apply phase vocoder timestretch [False] -t, --trim trim begining and end of file based on amplitude[False]
./mungo_utils.py -I "./test/*" -O ./test/converted/ -T C0 -n using audio input directory: ./test/* processing ./test/aifs/Arp_2600.aif 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 0 audio array of <type 'numpy.int32'> amplitude center at 1 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/aifs/GP CHH 3.aif 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 13 audio array of <type 'numpy.int32'> amplitude center at 0 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/aifs/GP CR 3.aif 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 28 audio array of <type 'numpy.int32'> amplitude center at 0 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/aifs/GP KD 2.aif 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 60 audio array of <type 'numpy.int32'> amplitude center at 0 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/aifs/GP OHH 2.aif 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 24 audio array of <type 'numpy.int32'> amplitude center at 0 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/aifs/GP RCB 3.aif 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 0 audio array of <type 'numpy.int32'> amplitude center at 0 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/aifs/GP RS 1.aif 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 15 audio array of <type 'numpy.int32'> amplitude center at 0 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/aifs/GP SN 2.aif 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 9 audio array of <type 'numpy.int32'> amplitude center at 0 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/wavs/DC_Offset_Test.wav 1 audio channels detected 16 bit sample depth detected 44100Hz sample rate detected resampled to 12000Hz trimed to sample start 27 audio array of <type 'numpy.int32'> amplitude center at 0 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out --------------------------------------------------- processing ./test/wavs/High_Resolution_24b_Test.wav 2 audio channels detected 24 bit sample depth detected 96000Hz sample rate detected resampled to 12000Hz trimed to sample start 0 audio array of <type 'numpy.int32'> amplitude center at 1 correcting for DC offset correction of amplitude center now at 0 correcting for DC offset processing fade out ---------------------------------------------------
Additionally the mungo_utils.py python script can be imported as a module, exposing three high-level functions: 1. Reading audio files and converting to a 1D array structure for signal processing, 2. Writing 1D array strutures to 16 bit mono wav (aka what all Mungo modules want), 3. Experimental Generative IR creator.
This function does not take in audio files, but starts generation with an empty 1D array buffer (absolute silence). Next a series of random passes pick among parameters and add impulses into the buffer. The user can specify the total number of IRs to generate and this function will make as many folders as required (with the Mungo W0.wav-W9.wav file names inside each). Suggested use is to play around with the parameters (like the H list H=[16,32,64]) and start a run that generates many IRs, auditioning the ones that sound interesting.
impulse type
HARM = integer related impulses
EXP = exponential decay impulses (reverb/spatial like)
RAND = ranomly distributed impulse
reverse probability
provides a chance to reverse the buffer
H
harmonic list of integers to generate harmonics