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ferrihydrite/morphagene_audacity3.py

Created March 28, 2021 18:19
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morphagene_audacity3.py
# -*- coding: utf-8 -*-
"""
USAGE:
morphagene_audacity3.py -w <inputwavfile> -l <inputlabels> -o <outputfile>'
Used to convert Audacity labels in .txt form 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 Audacity 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]
updated to Python3 by Phiala (March 2021)
"""
import sys, getopt
import struct
import numpy as np
from scipy import interpolate
import warnings
import io
from enum import IntEnum
__all__ = [
'WavFileWarning',
'read',
'write'
]
class WavFileWarning(UserWarning):
pass
class WAVE_FORMAT(IntEnum):
"""
WAVE form wFormatTag IDs
Complete list is in mmreg.h in Windows 10 SDK. ALAC and OPUS are the
newest additions, in v10.0.14393 2016-07
"""
UNKNOWN = 0x0000
PCM = 0x0001
ADPCM = 0x0002
IEEE_FLOAT = 0x0003
VSELP = 0x0004
IBM_CVSD = 0x0005
ALAW = 0x0006
MULAW = 0x0007
DTS = 0x0008
DRM = 0x0009
WMAVOICE9 = 0x000A
WMAVOICE10 = 0x000B
OKI_ADPCM = 0x0010
DVI_ADPCM = 0x0011
IMA_ADPCM = 0x0011 # Duplicate
MEDIASPACE_ADPCM = 0x0012
SIERRA_ADPCM = 0x0013
G723_ADPCM = 0x0014
DIGISTD = 0x0015
DIGIFIX = 0x0016
DIALOGIC_OKI_ADPCM = 0x0017
MEDIAVISION_ADPCM = 0x0018
CU_CODEC = 0x0019
HP_DYN_VOICE = 0x001A
YAMAHA_ADPCM = 0x0020
SONARC = 0x0021
DSPGROUP_TRUESPEECH = 0x0022
ECHOSC1 = 0x0023
AUDIOFILE_AF36 = 0x0024
APTX = 0x0025
AUDIOFILE_AF10 = 0x0026
PROSODY_1612 = 0x0027
LRC = 0x0028
DOLBY_AC2 = 0x0030
GSM610 = 0x0031
MSNAUDIO = 0x0032
ANTEX_ADPCME = 0x0033
CONTROL_RES_VQLPC = 0x0034
DIGIREAL = 0x0035
DIGIADPCM = 0x0036
CONTROL_RES_CR10 = 0x0037
NMS_VBXADPCM = 0x0038
CS_IMAADPCM = 0x0039
ECHOSC3 = 0x003A
ROCKWELL_ADPCM = 0x003B
ROCKWELL_DIGITALK = 0x003C
XEBEC = 0x003D
G721_ADPCM = 0x0040
G728_CELP = 0x0041
MSG723 = 0x0042
INTEL_G723_1 = 0x0043
INTEL_G729 = 0x0044
SHARP_G726 = 0x0045
MPEG = 0x0050
RT24 = 0x0052
PAC = 0x0053
MPEGLAYER3 = 0x0055
LUCENT_G723 = 0x0059
CIRRUS = 0x0060
ESPCM = 0x0061
VOXWARE = 0x0062
CANOPUS_ATRAC = 0x0063
G726_ADPCM = 0x0064
G722_ADPCM = 0x0065
DSAT = 0x0066
DSAT_DISPLAY = 0x0067
VOXWARE_BYTE_ALIGNED = 0x0069
VOXWARE_AC8 = 0x0070
VOXWARE_AC10 = 0x0071
VOXWARE_AC16 = 0x0072
VOXWARE_AC20 = 0x0073
VOXWARE_RT24 = 0x0074
VOXWARE_RT29 = 0x0075
VOXWARE_RT29HW = 0x0076
VOXWARE_VR12 = 0x0077
VOXWARE_VR18 = 0x0078
VOXWARE_TQ40 = 0x0079
VOXWARE_SC3 = 0x007A
VOXWARE_SC3_1 = 0x007B
SOFTSOUND = 0x0080
VOXWARE_TQ60 = 0x0081
MSRT24 = 0x0082
G729A = 0x0083
MVI_MVI2 = 0x0084
DF_G726 = 0x0085
DF_GSM610 = 0x0086
ISIAUDIO = 0x0088
ONLIVE = 0x0089
MULTITUDE_FT_SX20 = 0x008A
INFOCOM_ITS_G721_ADPCM = 0x008B
CONVEDIA_G729 = 0x008C
CONGRUENCY = 0x008D
SBC24 = 0x0091
DOLBY_AC3_SPDIF = 0x0092
MEDIASONIC_G723 = 0x0093
PROSODY_8KBPS = 0x0094
ZYXEL_ADPCM = 0x0097
PHILIPS_LPCBB = 0x0098
PACKED = 0x0099
MALDEN_PHONYTALK = 0x00A0
RACAL_RECORDER_GSM = 0x00A1
RACAL_RECORDER_G720_A = 0x00A2
RACAL_RECORDER_G723_1 = 0x00A3
RACAL_RECORDER_TETRA_ACELP = 0x00A4
NEC_AAC = 0x00B0
RAW_AAC1 = 0x00FF
RHETOREX_ADPCM = 0x0100
IRAT = 0x0101
VIVO_G723 = 0x0111
VIVO_SIREN = 0x0112
PHILIPS_CELP = 0x0120
PHILIPS_GRUNDIG = 0x0121
DIGITAL_G723 = 0x0123
SANYO_LD_ADPCM = 0x0125
SIPROLAB_ACEPLNET = 0x0130
SIPROLAB_ACELP4800 = 0x0131
SIPROLAB_ACELP8V3 = 0x0132
SIPROLAB_G729 = 0x0133
SIPROLAB_G729A = 0x0134
SIPROLAB_KELVIN = 0x0135
VOICEAGE_AMR = 0x0136
G726ADPCM = 0x0140
DICTAPHONE_CELP68 = 0x0141
DICTAPHONE_CELP54 = 0x0142
QUALCOMM_PUREVOICE = 0x0150
QUALCOMM_HALFRATE = 0x0151
TUBGSM = 0x0155
MSAUDIO1 = 0x0160
WMAUDIO2 = 0x0161
WMAUDIO3 = 0x0162
WMAUDIO_LOSSLESS = 0x0163
WMASPDIF = 0x0164
UNISYS_NAP_ADPCM = 0x0170
UNISYS_NAP_ULAW = 0x0171
UNISYS_NAP_ALAW = 0x0172
UNISYS_NAP_16K = 0x0173
SYCOM_ACM_SYC008 = 0x0174
SYCOM_ACM_SYC701_G726L = 0x0175
SYCOM_ACM_SYC701_CELP54 = 0x0176
SYCOM_ACM_SYC701_CELP68 = 0x0177
KNOWLEDGE_ADVENTURE_ADPCM = 0x0178
FRAUNHOFER_IIS_MPEG2_AAC = 0x0180
DTS_DS = 0x0190
CREATIVE_ADPCM = 0x0200
CREATIVE_FASTSPEECH8 = 0x0202
CREATIVE_FASTSPEECH10 = 0x0203
UHER_ADPCM = 0x0210
ULEAD_DV_AUDIO = 0x0215
ULEAD_DV_AUDIO_1 = 0x0216
QUARTERDECK = 0x0220
ILINK_VC = 0x0230
RAW_SPORT = 0x0240
ESST_AC3 = 0x0241
GENERIC_PASSTHRU = 0x0249
IPI_HSX = 0x0250
IPI_RPELP = 0x0251
CS2 = 0x0260
SONY_SCX = 0x0270
SONY_SCY = 0x0271
SONY_ATRAC3 = 0x0272
SONY_SPC = 0x0273
TELUM_AUDIO = 0x0280
TELUM_IA_AUDIO = 0x0281
NORCOM_VOICE_SYSTEMS_ADPCM = 0x0285
FM_TOWNS_SND = 0x0300
MICRONAS = 0x0350
MICRONAS_CELP833 = 0x0351
BTV_DIGITAL = 0x0400
INTEL_MUSIC_CODER = 0x0401
INDEO_AUDIO = 0x0402
QDESIGN_MUSIC = 0x0450
ON2_VP7_AUDIO = 0x0500
ON2_VP6_AUDIO = 0x0501
VME_VMPCM = 0x0680
TPC = 0x0681
LIGHTWAVE_LOSSLESS = 0x08AE
OLIGSM = 0x1000
OLIADPCM = 0x1001
OLICELP = 0x1002
OLISBC = 0x1003
OLIOPR = 0x1004
LH_CODEC = 0x1100
LH_CODEC_CELP = 0x1101
LH_CODEC_SBC8 = 0x1102
LH_CODEC_SBC12 = 0x1103
LH_CODEC_SBC16 = 0x1104
NORRIS = 0x1400
ISIAUDIO_2 = 0x1401
SOUNDSPACE_MUSICOMPRESS = 0x1500
MPEG_ADTS_AAC = 0x1600
MPEG_RAW_AAC = 0x1601
MPEG_LOAS = 0x1602
NOKIA_MPEG_ADTS_AAC = 0x1608
NOKIA_MPEG_RAW_AAC = 0x1609
VODAFONE_MPEG_ADTS_AAC = 0x160A
VODAFONE_MPEG_RAW_AAC = 0x160B
MPEG_HEAAC = 0x1610
VOXWARE_RT24_SPEECH = 0x181C
SONICFOUNDRY_LOSSLESS = 0x1971
INNINGS_TELECOM_ADPCM = 0x1979
LUCENT_SX8300P = 0x1C07
LUCENT_SX5363S = 0x1C0C
CUSEEME = 0x1F03
NTCSOFT_ALF2CM_ACM = 0x1FC4
DVM = 0x2000
DTS2 = 0x2001
MAKEAVIS = 0x3313
DIVIO_MPEG4_AAC = 0x4143
NOKIA_ADAPTIVE_MULTIRATE = 0x4201
DIVIO_G726 = 0x4243
LEAD_SPEECH = 0x434C
LEAD_VORBIS = 0x564C
WAVPACK_AUDIO = 0x5756
OGG_VORBIS_MODE_1 = 0x674F
OGG_VORBIS_MODE_2 = 0x6750
OGG_VORBIS_MODE_3 = 0x6751
OGG_VORBIS_MODE_1_PLUS = 0x676F
OGG_VORBIS_MODE_2_PLUS = 0x6770
OGG_VORBIS_MODE_3_PLUS = 0x6771
ALAC = 0x6C61
_3COM_NBX = 0x7000 # Can't have leading digit
OPUS = 0x704F
FAAD_AAC = 0x706D
AMR_NB = 0x7361
AMR_WB = 0x7362
AMR_WP = 0x7363
GSM_AMR_CBR = 0x7A21
GSM_AMR_VBR_SID = 0x7A22
COMVERSE_INFOSYS_G723_1 = 0xA100
COMVERSE_INFOSYS_AVQSBC = 0xA101
COMVERSE_INFOSYS_SBC = 0xA102
SYMBOL_G729_A = 0xA103
VOICEAGE_AMR_WB = 0xA104
INGENIENT_G726 = 0xA105
MPEG4_AAC = 0xA106
ENCORE_G726 = 0xA107
ZOLL_ASAO = 0xA108
SPEEX_VOICE = 0xA109
VIANIX_MASC = 0xA10A
WM9_SPECTRUM_ANALYZER = 0xA10B
WMF_SPECTRUM_ANAYZER = 0xA10C
GSM_610 = 0xA10D
GSM_620 = 0xA10E
GSM_660 = 0xA10F
GSM_690 = 0xA110
GSM_ADAPTIVE_MULTIRATE_WB = 0xA111
POLYCOM_G722 = 0xA112
POLYCOM_G728 = 0xA113
POLYCOM_G729_A = 0xA114
POLYCOM_SIREN = 0xA115
GLOBAL_IP_ILBC = 0xA116
RADIOTIME_TIME_SHIFT_RADIO = 0xA117
NICE_ACA = 0xA118
NICE_ADPCM = 0xA119
VOCORD_G721 = 0xA11A
VOCORD_G726 = 0xA11B
VOCORD_G722_1 = 0xA11C
VOCORD_G728 = 0xA11D
VOCORD_G729 = 0xA11E
VOCORD_G729_A = 0xA11F
VOCORD_G723_1 = 0xA120
VOCORD_LBC = 0xA121
NICE_G728 = 0xA122
FRACE_TELECOM_G729 = 0xA123
CODIAN = 0xA124
FLAC = 0xF1AC
EXTENSIBLE = 0xFFFE
DEVELOPMENT = 0xFFFF
KNOWN_WAVE_FORMATS = {WAVE_FORMAT.PCM, WAVE_FORMAT.IEEE_FLOAT}
def _raise_bad_format(format_tag):
try:
format_name = WAVE_FORMAT(format_tag).name
except ValueError:
format_name = f'{format_tag:#06x}'
raise ValueError(f"Unknown wave file format: {format_name}. Supported "
"formats: " +
', '.join(x.name for x in KNOWN_WAVE_FORMATS))
def _read_fmt_chunk(fid, is_big_endian):
"""
Returns
-------
size : int
size of format subchunk in bytes (minus 8 for "fmt " and itself)
format_tag : int
PCM, float, or compressed format
channels : int
number of channels
fs : int
sampling frequency in samples per second
bytes_per_second : int
overall byte rate for the file
block_align : int
bytes per sample, including all channels
bit_depth : int
bits per sample
Notes
-----
Assumes file pointer is immediately after the 'fmt ' id
"""
if is_big_endian:
fmt = '>'
else:
fmt = '<'
size = struct.unpack(fmt+'I', fid.read(4))[0]
if size < 16:
raise ValueError("Binary structure of wave file is not compliant")
res = struct.unpack(fmt+'HHIIHH', fid.read(16))
bytes_read = 16
format_tag, channels, fs, bytes_per_second, block_align, bit_depth = res
if format_tag == WAVE_FORMAT.EXTENSIBLE and size >= (16+2):
ext_chunk_size = struct.unpack(fmt+'H', fid.read(2))[0]
bytes_read += 2
if ext_chunk_size >= 22:
extensible_chunk_data = fid.read(22)
bytes_read += 22
raw_guid = extensible_chunk_data[2+4:2+4+16]
# GUID template {XXXXXXXX-0000-0010-8000-00AA00389B71} (RFC-2361)
# MS GUID byte order: first three groups are native byte order,
# rest is Big Endian
if is_big_endian:
tail = b'\x00\x00\x00\x10\x80\x00\x00\xAA\x00\x38\x9B\x71'
else:
tail = b'\x00\x00\x10\x00\x80\x00\x00\xAA\x00\x38\x9B\x71'
if raw_guid.endswith(tail):
format_tag = struct.unpack(fmt+'I', raw_guid[:4])[0]
else:
raise ValueError("Binary structure of wave file is not compliant")
if format_tag not in KNOWN_WAVE_FORMATS:
_raise_bad_format(format_tag)
# move file pointer to next chunk
if size > bytes_read:
fid.read(size - bytes_read)
# fmt should always be 16, 18 or 40, but handle it just in case
_handle_pad_byte(fid, size)
return (size, format_tag, channels, fs, bytes_per_second, block_align,
bit_depth)
def _read_data_chunk(fid, format_tag, channels, bit_depth, is_big_endian,
block_align, mmap=False):
"""
Notes
-----
Assumes file pointer is immediately after the 'data' id
It's possible to not use all available bits in a container, or to store
samples in a container bigger than necessary, so bytes_per_sample uses
the actual reported container size (nBlockAlign / nChannels). Real-world
examples:
Adobe Audition's "24-bit packed int (type 1, 20-bit)"
nChannels = 2, nBlockAlign = 6, wBitsPerSample = 20
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples/AFsp/M1F1-int12-AFsp.wav
is:
nChannels = 2, nBlockAlign = 4, wBitsPerSample = 12
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Docs/multichaudP.pdf
gives an example of:
nChannels = 2, nBlockAlign = 8, wBitsPerSample = 20
"""
if is_big_endian:
fmt = '>'
else:
fmt = '<'
# Size of the data subchunk in bytes
size = struct.unpack(fmt+'I', fid.read(4))[0]
# Number of bytes per sample (sample container size)
bytes_per_sample = block_align // channels
n_samples = size // bytes_per_sample
if format_tag == WAVE_FORMAT.PCM:
if 1 <= bit_depth <= 8:
dtype = 'u1' # WAV of 8-bit integer or less are unsigned
elif bytes_per_sample in {3, 5, 6, 7}:
# No compatible dtype. Load as raw bytes for reshaping later.
dtype = 'V1'
elif bit_depth <= 64:
# Remaining bit depths can map directly to signed numpy dtypes
dtype = f'{fmt}i{bytes_per_sample}'
else:
raise ValueError("Unsupported bit depth: the WAV file "
f"has {bit_depth}-bit integer data.")
elif format_tag == WAVE_FORMAT.IEEE_FLOAT:
if bit_depth in {32, 64}:
dtype = f'{fmt}f{bytes_per_sample}'
else:
raise ValueError("Unsupported bit depth: the WAV file "
f"has {bit_depth}-bit floating-point data.")
else:
_raise_bad_format(format_tag)
start = fid.tell()
if not mmap:
try:
count = size if dtype == 'V1' else n_samples
data = np.fromfile(fid, dtype=dtype, count=count)
except io.UnsupportedOperation: # not a C-like file
fid.seek(start, 0) # just in case it seeked, though it shouldn't
data = np.frombuffer(fid.read(size), dtype=dtype)
if dtype == 'V1':
# Rearrange raw bytes into smallest compatible numpy dtype
dt = f'{fmt}i4' if bytes_per_sample == 3 else f'{fmt}i8'
a = np.zeros((len(data) // bytes_per_sample, np.dtype(dt).itemsize),
dtype='V1')
if is_big_endian:
a[:, :bytes_per_sample] = data.reshape((-1, bytes_per_sample))
else:
a[:, -bytes_per_sample:] = data.reshape((-1, bytes_per_sample))
data = a.view(dt).reshape(a.shape[:-1])
else:
if bytes_per_sample in {1, 2, 4, 8}:
start = fid.tell()
data = np.memmap(fid, dtype=dtype, mode='c', offset=start,
shape=(n_samples,))
fid.seek(start + size)
else:
raise ValueError("mmap=True not compatible with "
f"{bytes_per_sample}-byte container size.")
_handle_pad_byte(fid, size)
if channels > 1:
data = data.reshape(-1, channels)
return data
def _skip_unknown_chunk(fid, is_big_endian):
if is_big_endian:
fmt = '>I'
else:
fmt = '<I'
data = fid.read(4)
# call unpack() and seek() only if we have really read data from file
# otherwise empty read at the end of the file would trigger
# unnecessary exception at unpack() call
# in case data equals somehow to 0, there is no need for seek() anyway
if data:
size = struct.unpack(fmt, data)[0]
fid.seek(size, 1)
_handle_pad_byte(fid, size)
def _read_riff_chunk(fid):
str1 = fid.read(4) # File signature
if str1 == b'RIFF':
is_big_endian = False
fmt = '<I'
elif str1 == b'RIFX':
is_big_endian = True
fmt = '>I'
else:
# There are also .wav files with "FFIR" or "XFIR" signatures?
raise ValueError(f"File format {repr(str1)} not understood. Only "
"'RIFF' and 'RIFX' supported.")
# Size of entire file
file_size = struct.unpack(fmt, fid.read(4))[0] + 8
str2 = fid.read(4)
if str2 != b'WAVE':
raise ValueError(f"Not a WAV file. RIFF form type is {repr(str2)}.")
return file_size, is_big_endian
def _handle_pad_byte(fid, size):
# "If the chunk size is an odd number of bytes, a pad byte with value zero
# is written after ckData." So we need to seek past this after each chunk.
if size % 2:
fid.seek(1, 1)
def read(filename, mmap=False):
"""
Open a WAV file.
Return the sample rate (in samples/sec) and data from an LPCM WAV file.
Parameters
----------
filename : string or open file handle
Input WAV file.
mmap : bool, optional
Whether to read data as memory-mapped (default: False). Not compatible
with some bit depths; see Notes. Only to be used on real files.
.. versionadded:: 0.12.0
Returns
-------
rate : int
Sample rate of WAV file.
data : numpy array
Data read from WAV file. Data-type is determined from the file;
see Notes. Data is 1-D for 1-channel WAV, or 2-D of shape
(Nsamples, Nchannels) otherwise. If a file-like input without a
C-like file descriptor (e.g., :class:`python:io.BytesIO`) is
passed, this will not be writeable.
Notes
-----
Common data types: [1]_
===================== =========== =========== =============
WAV format Min Max NumPy dtype
===================== =========== =========== =============
32-bit floating-point -1.0 +1.0 float32
32-bit integer PCM -2147483648 +2147483647 int32
24-bit integer PCM -2147483648 +2147483392 int32
16-bit integer PCM -32768 +32767 int16
8-bit integer PCM 0 255 uint8
===================== =========== =========== =============
WAV files can specify arbitrary bit depth, and this function supports
reading any integer PCM depth from 1 to 64 bits. Data is returned in the
smallest compatible numpy int type, in left-justified format. 8-bit and
lower is unsigned, while 9-bit and higher is signed.
For example, 24-bit data will be stored as int32, with the MSB of the
24-bit data stored at the MSB of the int32, and typically the least
significant byte is 0x00. (However, if a file actually contains data past
its specified bit depth, those bits will be read and output, too. [2]_)
This bit justification and sign matches WAV's native internal format, which
allows memory mapping of WAV files that use 1, 2, 4, or 8 bytes per sample
(so 24-bit files cannot be memory-mapped, but 32-bit can).
IEEE float PCM in 32- or 64-bit format is supported, with or without mmap.
Values exceeding [-1, +1] are not clipped.
Non-linear PCM (mu-law, A-law) is not supported.
References
----------
.. [1] IBM Corporation and Microsoft Corporation, "Multimedia Programming
Interface and Data Specifications 1.0", section "Data Format of the
Samples", August 1991
http://www.tactilemedia.com/info/MCI_Control_Info.html
.. [2] Adobe Systems Incorporated, "Adobe Audition 3 User Guide", section
"Audio file formats: 24-bit Packed Int (type 1, 20-bit)", 2007
Examples
--------
>>> from os.path import dirname, join as pjoin
>>> from scipy.io import wavfile
>>> import scipy.io
Get the filename for an example .wav file from the tests/data directory.
>>> data_dir = pjoin(dirname(scipy.io.__file__), 'tests', 'data')
>>> wav_fname = pjoin(data_dir, 'test-44100Hz-2ch-32bit-float-be.wav')
Load the .wav file contents.
>>> samplerate, data = wavfile.read(wav_fname)
>>> print(f"number of channels = {data.shape[1]}")
number of channels = 2
>>> length = data.shape[0] / samplerate
>>> print(f"length = {length}s")
length = 0.01s
Plot the waveform.
>>> import matplotlib.pyplot as plt
>>> import numpy as np
>>> time = np.linspace(0., length, data.shape[0])
>>> plt.plot(time, data[:, 0], label="Left channel")
>>> plt.plot(time, data[:, 1], label="Right channel")
>>> plt.legend()
>>> plt.xlabel("Time [s]")
>>> plt.ylabel("Amplitude")
>>> plt.show()
"""
if hasattr(filename, 'read'):
fid = filename
mmap = False
else:
fid = open(filename, 'rb')
try:
file_size, is_big_endian = _read_riff_chunk(fid)
fmt_chunk_received = False
data_chunk_received = False
while fid.tell() < file_size:
# read the next chunk
chunk_id = fid.read(4)
if not chunk_id:
if data_chunk_received:
# End of file but data successfully read
warnings.warn(
"Reached EOF prematurely; finished at {:d} bytes, "
"expected {:d} bytes from header."
.format(fid.tell(), file_size),
WavFileWarning, stacklevel=2)
break
else:
raise ValueError("Unexpected end of file.")
elif len(chunk_id) < 4:
msg = f"Incomplete chunk ID: {repr(chunk_id)}"
# If we have the data, ignore the broken chunk
if fmt_chunk_received and data_chunk_received:
warnings.warn(msg + ", ignoring it.", WavFileWarning,
stacklevel=2)
else:
raise ValueError(msg)
if chunk_id == b'fmt ':
fmt_chunk_received = True
fmt_chunk = _read_fmt_chunk(fid, is_big_endian)
format_tag, channels, fs = fmt_chunk[1:4]
bit_depth = fmt_chunk[6]
block_align = fmt_chunk[5]
elif chunk_id == b'fact':
_skip_unknown_chunk(fid, is_big_endian)
elif chunk_id == b'data':
data_chunk_received = True
if not fmt_chunk_received:
raise ValueError("No fmt chunk before data")
data = _read_data_chunk(fid, format_tag, channels, bit_depth,
is_big_endian, block_align, mmap)
elif chunk_id == b'LIST':
# Someday this could be handled properly but for now skip it
_skip_unknown_chunk(fid, is_big_endian)
elif chunk_id in {b'JUNK', b'Fake'}:
# Skip alignment chunks without warning
_skip_unknown_chunk(fid, is_big_endian)
else:
warnings.warn("Chunk (non-data) not understood, skipping it.",
WavFileWarning, stacklevel=2)
_skip_unknown_chunk(fid, is_big_endian)
finally:
if not hasattr(filename, 'read'):
fid.close()
else:
fid.seek(0)
return fs, data
def write(filename, rate, data, markers=False, verbose=True):
"""
Write a NumPy array as a WAV file.
Parameters
----------
filename : string or open file handle
Output wav file.
rate : int
The sample rate (in samples/sec).
data : ndarray
A 1-D or 2-D NumPy array of either integer or float data-type.
Notes
-----
* Writes a simple uncompressed WAV file.
* To write multiple-channels, use a 2-D array of shape
(Nsamples, Nchannels).
* The bits-per-sample and PCM/float will be determined by the data-type.
Common data types: [1]_
===================== =========== =========== =============
WAV format Min Max NumPy dtype
===================== =========== =========== =============
32-bit floating-point -1.0 +1.0 float32
32-bit PCM -2147483648 +2147483647 int32
16-bit PCM -32768 +32767 int16
8-bit PCM 0 255 uint8
===================== =========== =========== =============
Note that 8-bit PCM is unsigned.
References
----------
.. [1] IBM Corporation and Microsoft Corporation, "Multimedia Programming
Interface and Data Specifications 1.0", section "Data Format of the
Samples", August 1991
http://www.tactilemedia.com/info/MCI_Control_Info.html
Examples
--------
Create a 100Hz sine wave, sampled at 44100Hz.
Write to 16-bit PCM, Mono.
>>> from scipy.io.wavfile import write
>>> samplerate = 44100; fs = 100
>>> t = np.linspace(0., 1., samplerate)
>>> amplitude = np.iinfo(np.int16).max
>>> data = amplitude * np.sin(2. * np.pi * fs * t)
>>> write("example.wav", samplerate, data.astype(np.int16))
"""
if hasattr(filename, 'write'):
fid = filename
else:
fid = open(filename, 'wb')
fs = rate
try:
dkind = data.dtype.kind
if not (dkind == 'i' or dkind == 'f' or (dkind == 'u' and
data.dtype.itemsize == 1)):
raise ValueError("Unsupported data type '%s'" % data.dtype)
header_data = b''
header_data += b'RIFF'
header_data += b'\x00\x00\x00\x00'
header_data += b'WAVE'
# fmt chunk
header_data += b'fmt '
if dkind == 'f':
format_tag = WAVE_FORMAT.IEEE_FLOAT
else:
format_tag = WAVE_FORMAT.PCM
if data.ndim == 1:
channels = 1
else:
channels = data.shape[1]
bit_depth = data.dtype.itemsize * 8
bytes_per_second = fs*(bit_depth // 8)*channels
block_align = channels * (bit_depth // 8)
fmt_chunk_data = struct.pack('<HHIIHH', format_tag, channels, fs,
bytes_per_second, block_align, bit_depth)
if not (dkind == 'i' or dkind == 'u'):
# add cbSize field for non-PCM files
fmt_chunk_data += b'\x00\x00'
header_data += struct.pack('<I', len(fmt_chunk_data))
header_data += fmt_chunk_data
# fact chunk (non-PCM files)
if not (dkind == 'i' or dkind == 'u'):
header_data += b'fact'
header_data += struct.pack('<II', 4, data.shape[0])
# check data size (needs to be immediately before the data chunk)
if ((len(header_data)-4-4) + (4+4+data.nbytes)) > 0xFFFFFFFF:
raise ValueError("Data exceeds wave file size limit")
fid.write(header_data)
# data chunk
fid.write(b'data')
fid.write(struct.pack('<I', data.nbytes))
if data.dtype.byteorder == '>' or (data.dtype.byteorder == '=' and
sys.byteorder == 'big'):
data = data.byteswap()
_array_tofile(fid, data)
# cue/marker chunk
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 = [b'' 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 = b''
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 += bytes(label, encoding='ascii')
fid.write(b'LIST')
size = len(lbls) + 4
fid.write(struct.pack('<i', size))
fid.write(b'adtl')
fid.write(lbls)
# Determine file size and place it in correct
# position at start of the file.
size = fid.tell()
fid.seek(4)
fid.write(struct.pack('<I', size-8))
finally:
if not hasattr(filename, 'write'):
fid.close()
else:
fid.seek(0)
def _array_tofile(fid, data):
# ravel gives a c-contiguous buffer
fid.write(data.ravel().view('b').data)
def test_normalized(array):
'''
Determine if an array is entirely -1 < array[i,j] < 1, to see if array is
normalized
'''
return (array > -1).all() and (array < 1).all()
def norm_to_32float(array):
'''
Convert a variety of audio types to float32 while normalizing if needed
'''
if array.dtype == 'int16':
bits=16
normfactor = 2 ** (bits-1)
data = np.float32(array) * 1.0 / normfactor
if array.dtype == 'int32':
bits=32
normfactor = 2 ** (bits-1)
data = np.float32(array) * 1.0 / normfactor
if array.dtype == 'float32':
if test_normalized(array):
data = np.float32(array) # nothing needed
else:
bits=32
normfactor = 2 ** (bits-1)
data = np.float32(array) * 1.0 / normfactor
if array.dtype == 'float64':
bits=64
normfactor = 2 ** (bits-1)
data = np.float32(array) * 1.0 / normfactor
elif array.dtype == 'uint8':
if isinstance(data[0], (int, np.uint8)):
bits=8
# handle uint8 data by shifting to center at 0
normfactor = 2 ** (bits-1)
data = (np.float32(array) * 1.0 / normfactor) -\
((normfactor)/(normfactor-1))
return data
def load_audacity_labels(label_file):
'''
Load Audacity labels, ignoring the additional frequency range info lines,
if labels were exported from a spectrogram.
'''
fi = open(label_file, 'r')
labs = [line.strip().split()[0] for line in fi if not line.startswith('\\')]
fi.close()
return np.array(labs).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_audacity3.py -w <inputwavfile> -l <inputlabels> -o <outputfile>')
sys.exit(2)
for opt, arg in opts:
if opt == '-h':
print('morphagene_audacity3.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 Audacity 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_audacity_labels(inputlabelfile)
# read pertinent info from audio file, convert, exit if input wave file is broken
try:
sample_rate, array = read(inputwavefile)
array = norm_to_32float(array)
except:
print('Input file %s.wav 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
# no need to transpose again for data from Audacity
write(outputfile,morph_srate,array.astype('float32'),markers=frame_dict)
print('Saved Morphagene reel with %i splices: %s'%(len(frame_labs),outputfile))
if __name__ == "__main__":
main(sys.argv[1:])
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