Skip to content

Instantly share code, notes, and snippets.

@knandersen
Forked from ferrihydrite/morphagene_audacity.py
Last active August 23, 2024 01:55
Show Gist options
  • Save knandersen/a1da6859e3ef84f3c0ce1979536d85c8 to your computer and use it in GitHub Desktop.
Save knandersen/a1da6859e3ef84f3c0ce1979536d85c8 to your computer and use it in GitHub Desktop.
Allows you to use Ableton projects and exports as reels for the Make Noise Morphagene eurorack module. Since a few people have found the script not working or difficulty getting python to work, I have created a web-based tool: https://knandersen.github.io/morphaweb/
#!/usr/bin/env python2
# -*- coding: utf-8 -*-
"""
USAGE:
morphagene_ableton.py -w <inputwavfile> -l <inputlabels> -o <outputfile>'
Instructions in Ableton:
Insert locators as splice markers in your project (Create > Add Locator)
Export Audio/Video with
Sample Rate: 48000 Hz
Encode PCM: enabled
File Type: WAV
Bit Depth: 16
Save your Ableton project.
The associated Ableton Live Set .als-file will serve as the inputlabels argument
Used to convert Ableton Locators from an Ableton Live Set file on .WAV files into
single 32-bit float .WAV with CUE markers within the file, directly
compatible with the Make Noise Morphagene.
Does not require input file to be 48000Hz, only that the Ableton label matches
the .WAV file that generated it, and that the input .WAV is stereo.
See the Morphagene manual for naming conventions of output files:
http://www.makenoisemusic.com/content/manuals/morphagene-manual.pdf
# see http://stackoverflow.com/questions/15576798/create-32bit-float-wav-file-in-python
# see... http://blog.theroyweb.com/extracting-wav-file-header-information-using-a-python-script
# marker code from Joseph Basquin [https://gist.github.com/josephernest/3f22c5ed5dabf1815f16efa8fa53d476]
"""
import sys, getopt
import struct
import numpy as np
from scipy import interpolate
import gzip
import xml.etree.ElementTree as ET
def float32_wav_file(file_name, sample_array, sample_rate,
markers=None, verbose=False):
(M,N)=sample_array.shape
#print "len sample_array=(%d,%d)" % (M,N)
byte_count = M * N * 4 # (len(sample_array)) * 4 # 32-bit floats
wav_file = ""
# write the header
wav_file += struct.pack('<ccccIccccccccIHHIIHH',
'R', 'I', 'F', 'F',
byte_count + 0x2c - 8, # header size
'W', 'A', 'V', 'E', 'f', 'm', 't', ' ',
0x10, # size of 'fmt ' header
3, # format 3 = floating-point PCM
M, # channels
sample_rate, # samples / second
sample_rate * 4, # bytes / second
4, # block alignment
32) # bits / sample
wav_file += struct.pack('<ccccI',
'd', 'a', 't', 'a', byte_count)
if verbose:
print("packing...")
# flatten data in an alternating fashion
# see: http://soundfile.sapp.org/doc/WaveFormat/
reordered_wav = [sample_array[k,j] for j in range(N) for k in range(M)]
wav_file += struct.pack('<%df' % len(reordered_wav), *reordered_wav)
if verbose:
print("saving audio...")
fid=open(file_name,'wb')
for value in wav_file:
fid.write(value)
if markers: # != None and != []
if verbose:
print("saving cue markers...")
if isinstance(markers[0], dict):# then we have [{'position': 100, 'label': 'marker1'}, ...]
labels = [m['label'] for m in markers]
markers = [m['position'] for m in markers]
else:
labels = ['' for m in markers]
fid.write(b'cue ')
size = 4 + len(markers) * 24
fid.write(struct.pack('<ii', size, len(markers)))
for i, c in enumerate(markers):
s = struct.pack('<iiiiii', i + 1, c, 1635017060, 0, 0, c)# 1635017060 is struct.unpack('<i',b'data')
fid.write(s)
lbls = ''
for i, lbl in enumerate(labels):
lbls += b'labl'
label = lbl + ('\x00' if len(lbl) % 2 == 1 else '\x00\x00')
size = len(lbl) + 1 + 4 # because \x00
lbls += struct.pack('<ii', size, i + 1)
lbls += label
fid.write(b'LIST')
size = len(lbls) + 4
fid.write(struct.pack('<i', size))
fid.write(b'adtl')# https://web.archive.org/web/20141226210234/http://www.sonicspot.com/guide/wavefiles.html#list
fid.write(lbls)
fid.close()
def wav_file_read(filename,verbose=False):
# read file and close
fi=open(filename,'rb')
data=fi.read()
fi.close()
# take raw data and read subsections for important format data
A,B,C,D=struct.unpack('4c', data[0:4]) # 'RIFF'
ChunkSize=struct.unpack('<l', data[4:8])[0] #4+(8+SubChunk1Size)+8+SubChunk2Size)
A,B,C,D=struct.unpack('4c', data[8:12]) # 'WAVE'
A,B,C,D=struct.unpack('4c', data[12:16]) # 'fmt '
Subchunk1Size=struct.unpack('<l', data[16:20])[0] # LITTLE ENDIAN, long, 16
AudioFormat=struct.unpack('<h', data[20:22])[0] # LITTLE ENDIAN, short, 1
NumChannels=struct.unpack('<h', data[22:24])[0] # LITTLE ENDIAN, short, Mono = 1, Stereo = 2
SampleRate =struct.unpack('<l', data[24:28])[0] # LITTLE ENDIAN, long, sample rate in samples per second
ByteRate=struct.unpack('<l', data[28:32])[0] # self.SampleRate * self.NumChannels * self.BitsPerSample/8)) # (ByteRate) LITTLE ENDIAN, long
BlockAlign=struct.unpack('<h', data[32:34])[0] # self.NumChannels * self.BitsPerSample/8)) # (BlockAlign) LITTLE ENDIAN, short
BitsPerSample=struct.unpack('<h', data[34:36])[0] # LITTLE ENDIAN, short
A,B,C,D=struct.unpack('4c', data[36:40]) # BIG ENDIAN, char*4
SubChunk2Size=struct.unpack('<l', data[40:44])[0] # LITTLE ENDIAN, long
waveData=data[44:]
(M,N)=(len(waveData),len(waveData[0]))
if verbose:
print("ChunkSize =%d\nSubchunk1Size =%d\nAudioFormat =%d\nNumChannels =%d\nSampleRate =%d\nByteRate =%d\nBlockAlign =%d\nBitsPerSample =%d\nA:%c, B:%c, C:%c, D:%c\nSubChunk2Size =%d" %
(ChunkSize ,
Subchunk1Size,
AudioFormat ,
NumChannels ,
SampleRate ,
ByteRate ,
BlockAlign ,
BitsPerSample ,
A, B, C, D ,
SubChunk2Size ))
# convert audio data to float based on bitdepth
if BitsPerSample==8:
if verbose:
print("Unpacking 8 bits on len(waveData)=%d" % len(waveData))
d=np.fromstring(waveData,np.uint8)
floatdata=d.astype(np.float64)/np.float(127)
elif BitsPerSample==16:
if verbose:
print("Unpacking 16 bits on len(waveData)=%d" % len(waveData))
d=np.zeros(SubChunk2Size/2, dtype=np.int16)
j=0
for k in range(0, SubChunk2Size, 2):
d[j]=struct.unpack('<h',waveData[k:k+2])[0]
j=j+1
floatdata=d.astype(np.float64)/np.float(32767)
elif BitsPerSample==24:
if verbose:
print("Unpacking 24 bits on len(waveData)=%d" % len(waveData))
d=np.zeros(SubChunk2Size/3, dtype=np.int32)
j=0
for k in range(0, SubChunk2Size, 3):
d[j]=struct.unpack('<l',struct.pack('c',waveData[k])+waveData[k:k+3])[0]
j=j+1
floatdata=d.astype(np.float64)/np.float(2147483647)
else: # anything else will be considered 32 bits
if verbose:
print("Unpacking 32 bits on len(waveData)=%d" % len(waveData))
d=np.fromstring(waveData,np.int32)
floatdata=d.astype(np.float64)/np.float(2147483647)
v=floatdata[0::NumChannels]
for i in range(1,NumChannels):
v=np.vstack((v,floatdata[i::NumChannels]))
#return (np.vstack((floatdata[0::2],floatdata[1::2])), SampleRate, NumChannels, BitsPerSample)
return (v, SampleRate, NumChannels, BitsPerSample)
def load_ableton_labels(label_file):
'''
Loads Ableton Live locators and calculates the timecode based on tempo and locator measure
'''
# Open Ableton ALS file as gzip and read tempo and locator data as XML
with gzip.open(label_file, mode='r') as f:
data = f.read()
root = ET.fromstring(data)
bpm = None
markers = []
for tempo in root.iter('Tempo'):
for manual in tempo.findall('Manual'):
bpm = float(manual.get('Value'))
bps = bpm / 60
print("BPM: {0}, BPS: {1}".format(bpm,bps))
for locator in root.iter('Locator'):
v = float(locator.find('Time').get('Value', 'nan'))
print("Locator {0} found at: {1}".format(locator.get('Id'),v/bps))
markers.append(v/bps)
return np.array(markers).astype('float')
def change_samplerate_interp(old_audio,old_rate,new_rate):
'''
Change sample rate to new sample rate by simple interpolation.
If old_rate > new_rate, there may be aliasing / data loss.
Input should be in column format, as the interpolation will be completed
on each channel this way.
Modified from:
https://stackoverflow.com/questions/33682490/how-to-read-a-wav-file-using-scipy-at-a-different-sampling-rate
'''
if old_rate != new_rate:
# duration of audio
duration = old_audio.shape[0] / old_rate
# length of old and new audio
time_old = np.linspace(0, duration, old_audio.shape[0])
time_new = np.linspace(0, duration, int(old_audio.shape[0] * new_rate / old_rate))
# fit old_audio into new_audio length by interpolation
interpolator = interpolate.interp1d(time_old, old_audio.T)
new_audio = interpolator(time_new).T
return new_audio
else:
print('Conversion not needed, old and new rates match')
return old_audio # conversion not needed
def main(argv):
inputwavefile = ''
inputlabelfile = ''
outputfile = ''
try:
opts, args = getopt.getopt(argv,"hw:l:o:",["wavfile=","labelfile=","outputfile="])
except getopt.GetoptError:
print('Error in usage, correct format:\n'+\
'morphagene_ableton.py -w <inputwavfile> -l <inputlabels> -o <outputfile>')
sys.exit(2)
for opt, arg in opts:
if opt == '-h':
print('morphagene_ableton.py -w <inputwavfile> -l <inputlabels> -o <outputfile>')
sys.exit()
elif opt in ("-w", "--wavfile"):
inputwavefile = arg
elif opt in ("-l", "--labelfile"):
inputlabelfile = arg
elif opt in ("-o", "--outputfile"):
outputfile = arg
print('Input wave file: %s'%inputwavefile)
print('Input label file: %s'%inputlabelfile)
print('Output Morphagene reel: %s'%outputfile)
###########################################################################
'''
Write single file, edited in Ableton with labels, to Morphagene 32bit
WAV file at 48000hz sample rate.
'''
###########################################################################
morph_srate = 48000 # required samplerate for Morphagene
# read labels from stereo Audacity label file, ignore text, and use one channel
audac_labs = load_ableton_labels(inputlabelfile)
# read pertinent info from audio file, exit if input wave file is broken
try:
(array,sample_rate,num_channels,bits_per_sample)=wav_file_read(inputwavefile)
except:
print('Input .wav file %s is poorly formatted, exiting'%inputwavefile)
sys.exit()
# check if input wav has a different rate than desired Morphagene rate,
# and correct by interpolation
if sample_rate != morph_srate:
print("Correcting input sample rate %iHz to Morphagene rate %iHz"%(sample_rate,morph_srate))
# perform interpolation on each channel, then transpose back
array = change_samplerate_interp(array.T,float(sample_rate),float(morph_srate)).T
# convert labels in seconds to labels in frames, adjusting for change
# in rate
sc = float(morph_srate) / float(sample_rate)
frame_labs = (audac_labs * sample_rate * sc).astype(np.int)
else:
frame_labs = (audac_labs * sample_rate).astype(np.int)
frame_dict = [{'position': l, 'label': 'marker%i'%(i+1)} for i,l in enumerate(frame_labs)]
# write wav file with additional cue markers from labels
float32_wav_file(outputfile,array,morph_srate,markers=frame_dict)
print('Saved Morphagene reel with %i splices: %s'%(len(frame_labs),outputfile))
if __name__ == "__main__":
main(sys.argv[1:])
@KiLm0
Copy link

KiLm0 commented Feb 23, 2021

Hi. Thank you, this is just what I am looking to be able to do. Can you advise me on the upload process? Does this get uploaded to the morphogene as firmware right? Its not just a file on the sd card. TIA

@knandersen
Copy link
Author

Hi, glad to hear! No it doesn't handle the upload process, but should be possible to write a separate script that uploads all the reel to the SD card. I don't mind just doing that manually and a script would have be concerned with handling existing reels on the card, name conflicts etc.

@KiLm0
Copy link

KiLm0 commented Feb 26, 2021

Hi. Sorry I meant can you give me instructions on how and where to install your program to make it work. Noob over here!

@knandersen
Copy link
Author

oh, sure, I can try! If you don't know how to run python scripts, here's a tutorial for most operating systems: https://www.pythonforbeginners.com/development/how-run-your-python-scripts

Did you read the USAGE-section of the script? There are instructions for what to do inside Ableton to prep your Ableton file and wav-file. With those, you can run the script by following the instructions in the tutorial I just mentioned.

@alevillejo
Copy link

Hey! first of all thank you for sharing your work :) I did everything as described above but I'm getting this message "Input .wav file is poorly formatted, exiting" I've tried everything, any ideas?

@knandersen
Copy link
Author

knandersen commented Apr 1, 2022

Hey! first of all thank you for sharing your work :) I did everything as described above but I'm getting this message "Input .wav file is poorly formatted, exiting" I've tried everything, any ideas?

@alevillejo sorry you're experiencing issues! Can you give me a bit of troubleshooting info, like what kind of input file are you using? bit rate, sample rate, number of channels? could also help to know where you got the file from, i.e. which software generated the file in the first place

@bullpencatcher
Copy link

Hello, I am having the same problem that @alevillejo was having. I have exported using the specs in the instructions, WAV input file, ALS input labels. 16bit, 48000 Hz, stereo/2 channels, PCM enabled. Still getting the "Input .wav file is poorly formatted, exiting". If anyone has advice, I'd appreciate it.

@knandersen
Copy link
Author

@bullpencatcher sorry about that! Would it be possible to share some kind of sample I can use to test with? Might take a look at it over the weekend.

@knandersen
Copy link
Author

I just ran test and works fine here.

When asking for help in the future, would be really helpful that you tell me:

  • Error message
  • Command you are executing
  • System info: Which OS are you running, which version of Python, etc.

Alternatively you can also try a web-based version I made for taking a single audio file, adding markers and exporting as a morphagene-compatible reel: https://knandersen.github.io/morphaweb/

@jak2030
Copy link

jak2030 commented Nov 10, 2022

Hey @knandersen,

The issue I'm getting specifically if I raise an exception for the same error that others have noted occurs at line 121:

https://gist.github.com/knandersen/a1da6859e3ef84f3c0ce1979536d85c8#file-morphagene_ableton-py-L121

Traceback (most recent call last):
  File "/Users/jak/Desktop/music/samples/morphagene/morphagene_ableton.py", line 257, in main
    (array,sample_rate,num_channels,bits_per_sample)=wav_file_read(inputwavefile)
  File "/Users/jak/Desktop/music/samples/morphagene/morphagene_ableton.py", line 120, in wav_file_read
    (M,N)=(len(waveData),len(waveData[0]))
TypeError: object of type 'int' has no len()

This is occurring because the first index of the byte information in waveData returns an int (0 for me, though I suppose 1 would be possible too 😄).

When I comment out that line (since the offending variables aren't used), I get a new error that has to do with the wav_file byte object that is getting packed at line 47:

https://gist.github.com/knandersen/a1da6859e3ef84f3c0ce1979536d85c8#file-morphagene_ableton-py-L47

Traceback (most recent call last):
  File "/Users/jak/Desktop/music/samples/morphagene/morphagene_ableton.py", line 292, in <module>
    main(sys.argv[1:])
  File "/Users/jak/Desktop/music/samples/morphagene/morphagene_ableton.py", line 288, in main
    float32_wav_file(outputfile,array,morph_srate,markers=frame_dict)
  File "/Users/jak/Desktop/music/samples/morphagene/morphagene_ableton.py", line 48, in float32_wav_file
    wav_file += struct.pack('<ccccIccccccccIHHIIHH',
struct.error: char format requires a bytes object of length 1

This one is definitely beyond me but would love to understand!

Some specs:

  • 2.7.18 virtualenv (since your script specifies python 2)
  • OSX 12.3.1
  • I've also exported all of the audio parameters as you've specified

@jak2030
Copy link

jak2030 commented Nov 10, 2022

*Also, the webapp you made is super cool, thank you so much for creating it!!

@knandersen
Copy link
Author

knandersen commented Nov 18, 2022

*Also, the webapp you made is super cool, thank you so much for creating it!!

Sorry you're having trouble with the python-script. Works on my side, also using 2.7.18 on OSX 13.0 and is hard to debug because of so many things that could be going wrong. That's one of the reasons I created the webapp, which I'm super happy that you like!

@premini
Copy link

premini commented Aug 23, 2024

Hi, I'm very late to the party. I am also getting the "File is poorly formatted" error. Has anyone had any luck?

Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment