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200 lines
4.3 KiB
Python
200 lines
4.3 KiB
Python
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from base import rObject
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import audioop
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import struct
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import re
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import sys
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import os
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from math import log2
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__all__ = ["rSoundSample"]
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# See: IIgs TechNote #76 Miscellaneous Resource Formats
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# See: HCGS TechNote #3 Pitching Sampled Sounds
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# HyperCard assumes a sample rate of 26.32 KHz (DOC rate w/ 32 oscillators)
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# and a pitch of 261.63 Hz (Middle C, C4)
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# See HyperCard IIgs Tech Note #3: Pitching Sampled Sounds
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def relative_pitch(fS, fW = None):
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# fW = frequency of sample
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# fS = sampling rate
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r = 0
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if fW: r = (261.63 * fS) / (26320 * fW)
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else: r = fS / 26320
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offset = round(3072 * log2(r))
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if (offset < -32767) or (offset > 32767):
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raise Exception("Audio error: offset too big")
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if offset < 0: offset = 0x8000 | abs(offset)
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return offset
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def pitch_to_hz(p):
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if p == None: return 261.63
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if type(p) in (int, float): return float(p)
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if type(p) == str:
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m = re.match("^([A-Ga-g])([#b])?([0-8])$", p)
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if not m: return None
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note = m1[1].upper(); accidental = m[2]; octave = int(m[3])
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a = "CxDxEFxGxAxB".index(note)-9
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if accidental == "#": a += 1
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if accidental == "b": a -= 1
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f = 440.0 * (2 ** (a/12))
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f *= 2 ** (octave-4)
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return f
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return None
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def open_audio(file):
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_, ext = os.path.splitext(os.path.basename(file))
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ext = ext.lower()
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# if ext in (".wav", ".wave"):
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# import wave
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# return wave.open(file, "rb"), 'little', 128
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if ext in (".aiff", ".aifc", ".aif"):
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import aifc
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return aifc.open(file, "rb"), 'big', 'AIFF'
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if ext in (".au", ".snd"):
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import sunau
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return sunau.open(file, "rb"), 'big', 'SUN'
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# default
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import wave
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return wave.open(file, "rb"), 'little', 'WAVE'
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class rSoundSample(rObject):
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"""
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filename: input file to read. format is .wav, .au, .aiff, or .aifc
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pitch: audio pitch, if this is a note. specify hz (eg 261.63) or name (eg c4)
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rate: down/upsample audio to this rate (eg 26320)
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channel: stereo channel
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Native samples are 26320 khz, c4 (261.63 hz)
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"""
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rName = "rSoundSample"
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rType = 0x8024
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def __init__(self, filename, pitch=None, rate=None, channel=0, **kwargs):
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super().__init__(**kwargs)
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new_rate = rate
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freq = pitch_to_hz(pitch)
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if not freq: raise ValueError("Invalid pitch: {}".format(pitch))
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# audio conversion
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verbose = False
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# if verbose: print("Input File: {}".format(filename))
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rv = bytearray()
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tr = b"\x01" + bytes(range(1,256)) # remap 0 -> 1
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rv += struct.pack("<10x") # header filled in later
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src, byteorder, fmt = open_audio(filename)
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width = src.getsampwidth()
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channels = src.getnchannels()
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rate = src.getframerate()
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bias = 128
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swap = width > 1 and sys.byteorder != byteorder
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if width == 1 and fmt == 'wave': bias = 0
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if verbose:
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print("Input: {} ch, {} Hz, {}-bit, {} ({} frames)".format(
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channels,
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rate,
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width*8,
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fmt,
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src.getnframes()
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))
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if channels > 2:
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raise Exception("{}: Too many channels ({})".format(filename, channels))
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cookie = None
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while True:
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frames = src.readframes(32)
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if not frames: break
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if swap:
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frames = audioop.byteswap(frames, width)
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if channels > 1:
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frames = audioop.tomono(frames, width, 0.5, 0.5)
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if new_rate:
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frames, cookie = audioop.ratecv(frames, width, 1, rate, new_rate, cookie)
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if width != 1:
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frames = audioop.lin2lin(frames, width, 1)
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if bias:
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frames = audioop.bias(frames, 1, bias)
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frames = frames.translate(tr)
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rv += frames
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src.close()
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# based on system 6 samples, pages rounds down....
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# probably a bug.
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pages = (len(rv)-10+255) >> 8
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hz = new_rate or rate
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rp = relative_pitch(hz, freq)
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struct.pack_into("<HHHHH", rv, 0,
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0, # format
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pages, # wave size in pages
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rp,
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channel, # stereo
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hz # hz
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)
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self.data = bytes(rv)
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self.pages = pages
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self.channel = channel
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self.relative_pitch = rp
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self.sample_rate = hz
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if verbose:
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print("Output: 1 ch, {} Hz, 8-bit, rSoundSample ({} frames, {:.02f} Hz)".format(
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hz, len(rv)-10, freq or 261.63))
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print()
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def __bytes__(self):
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return self.data
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def _rez_string(self):
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return (
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"\t0, /* format */\n"
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"\t{}, /* size (pages) */\n"
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"\t0x{:04x}, /* relative pitch */\n"
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"\t0x{:04x}, /* stereo channel */\n"
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"\t{:d}, /* sample rate */\n"
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"\t..."
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).format(
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self.pages,
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self.relative_pitch,
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self.channel,
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self.sample_rate
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)
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