mirror of
https://github.com/correl/dejavu.git
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commit
1fe7d4fd45
2 changed files with 149 additions and 10 deletions
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@ -2,7 +2,8 @@ import os
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import fnmatch
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import fnmatch
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import numpy as np
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import numpy as np
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from pydub import AudioSegment
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from pydub import AudioSegment
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from pydub.utils import audioop
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import wavio
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def find_files(path, extensions):
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def find_files(path, extensions):
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# Allow both with ".mp3" and without "mp3" to be used for extensions
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# Allow both with ".mp3" and without "mp3" to be used for extensions
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@ -18,7 +19,8 @@ def find_files(path, extensions):
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def read(filename, limit=None):
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def read(filename, limit=None):
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"""
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"""
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Reads any file supported by pydub (ffmpeg) and returns the data contained
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Reads any file supported by pydub (ffmpeg) and returns the data contained
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within.
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within. If file reading fails due to input being a 24-bit wav file,
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wavio is used as a backup.
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Can be optionally limited to a certain amount of seconds from the start
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Can be optionally limited to a certain amount of seconds from the start
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of the file by specifying the `limit` parameter. This is the amount of
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of the file by specifying the `limit` parameter. This is the amount of
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@ -26,18 +28,34 @@ def read(filename, limit=None):
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returns: (channels, samplerate)
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returns: (channels, samplerate)
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"""
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"""
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audiofile = AudioSegment.from_file(filename)
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# pydub does not support 24-bit wav files, use wavio when this occurs
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try:
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audiofile = AudioSegment.from_file(filename)
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if limit:
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if limit:
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audiofile = audiofile[:limit * 1000]
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audiofile = audiofile[:limit * 1000]
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data = np.fromstring(audiofile._data, np.int16)
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data = np.fromstring(audiofile._data, np.int16)
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channels = []
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channels = []
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for chn in xrange(audiofile.channels):
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for chn in xrange(audiofile.channels):
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channels.append(data[chn::audiofile.channels])
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channels.append(data[chn::audiofile.channels])
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return channels, audiofile.frame_rate
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fs = audiofile.frame_rate
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except audioop.error:
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fs, _, audiofile = wavio.readwav(filename)
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if limit:
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audiofile = audiofile[:limit * 1000]
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audiofile = audiofile.T
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audiofile = audiofile.astype(np.int16)
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channels = []
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for chn in audiofile:
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channels.append(chn)
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return channels, fs
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def path_to_songname(path):
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def path_to_songname(path):
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121
dejavu/wavio.py
Normal file
121
dejavu/wavio.py
Normal file
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@ -0,0 +1,121 @@
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# wavio.py
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# Author: Warren Weckesser
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# License: BSD 3-Clause (http://opensource.org/licenses/BSD-3-Clause)
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# Synopsis: A Python module for reading and writing 24 bit WAV files.
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# Github: github.com/WarrenWeckesser/wavio
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import wave as _wave
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import numpy as _np
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def _wav2array(nchannels, sampwidth, data):
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"""data must be the string containing the bytes from the wav file."""
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num_samples, remainder = divmod(len(data), sampwidth * nchannels)
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if remainder > 0:
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raise ValueError('The length of data is not a multiple of '
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'sampwidth * num_channels.')
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if sampwidth > 4:
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raise ValueError("sampwidth must not be greater than 4.")
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if sampwidth == 3:
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a = _np.empty((num_samples, nchannels, 4), dtype=_np.uint8)
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raw_bytes = _np.fromstring(data, dtype=_np.uint8)
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a[:, :, :sampwidth] = raw_bytes.reshape(-1, nchannels, sampwidth)
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a[:, :, sampwidth:] = (a[:, :, sampwidth - 1:sampwidth] >> 7) * 255
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result = a.view('<i4').reshape(a.shape[:-1])
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else:
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# 8 bit samples are stored as unsigned ints; others as signed ints.
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dt_char = 'u' if sampwidth == 1 else 'i'
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a = _np.fromstring(data, dtype='<%s%d' % (dt_char, sampwidth))
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result = a.reshape(-1, nchannels)
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return result
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def readwav(file):
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"""
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Read a WAV file.
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Parameters
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----------
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file : string or file object
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Either the name of a file or an open file pointer.
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Return Values
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-------------
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rate : float
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The sampling frequency (i.e. frame rate)
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sampwidth : float
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The sample width, in bytes. E.g. for a 24 bit WAV file,
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sampwidth is 3.
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data : numpy array
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The array containing the data. The shape of the array is
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(num_samples, num_channels). num_channels is the number of
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audio channels (1 for mono, 2 for stereo).
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Notes
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-----
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This function uses the `wave` module of the Python standard libary
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to read the WAV file, so it has the same limitations as that library.
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In particular, the function does not read compressed WAV files.
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"""
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wav = _wave.open(file)
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rate = wav.getframerate()
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nchannels = wav.getnchannels()
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sampwidth = wav.getsampwidth()
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nframes = wav.getnframes()
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data = wav.readframes(nframes)
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wav.close()
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array = _wav2array(nchannels, sampwidth, data)
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return rate, sampwidth, array
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def writewav24(filename, rate, data):
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"""
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Create a 24 bit wav file.
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Parameters
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----------
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filename : string
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Name of the file to create.
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rate : float
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The sampling frequency (i.e. frame rate) of the data.
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data : array-like collection of integer or floating point values
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data must be "array-like", either 1- or 2-dimensional. If it
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is 2-d, the rows are the frames (i.e. samples) and the columns
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are the channels.
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Notes
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-----
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The data is assumed to be signed, and the values are assumed to be
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within the range of a 24 bit integer. Floating point values are
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converted to integers. The data is not rescaled or normalized before
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writing it to the file.
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Example
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-------
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Create a 3 second 440 Hz sine wave.
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>>> rate = 22050 # samples per second
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>>> T = 3 # sample duration (seconds)
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>>> f = 440.0 # sound frequency (Hz)
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>>> t = np.linspace(0, T, T*rate, endpoint=False)
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>>> x = (2**23 - 1) * np.sin(2 * np.pi * f * t)
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>>> writewav24("sine24.wav", rate, x)
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"""
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a32 = _np.asarray(data, dtype=_np.int32)
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if a32.ndim == 1:
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# Convert to a 2D array with a single column.
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a32.shape = a32.shape + (1,)
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# By shifting first 0 bits, then 8, then 16, the resulting output
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# is 24 bit little-endian.
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a8 = (a32.reshape(a32.shape + (1,)) >> _np.array([0, 8, 16])) & 255
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wavdata = a8.astype(_np.uint8).tostring()
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w = _wave.open(filename, 'wb')
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w.setnchannels(a32.shape[1])
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w.setsampwidth(3)
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w.setframerate(rate)
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w.writeframes(wavdata)
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w.close()
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