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