Initial commit of ZonePlayer

venv/lib/python3.11/site-packages/pydub/__init__.py

@@ -0,0 +1 @@
+from .audio_segment import AudioSegment

venv/lib/python3.11/site-packages/pydub/effects.py

@@ -0,0 +1,341 @@
+import sys
+import math
+import array
+from .utils import (
+    db_to_float,
+    ratio_to_db,
+    register_pydub_effect,
+    make_chunks,
+    audioop,
+    get_min_max_value
+)
+from .silence import split_on_silence
+from .exceptions import TooManyMissingFrames, InvalidDuration
+
+if sys.version_info >= (3, 0):
+    xrange = range
+
+
+@register_pydub_effect
+def apply_mono_filter_to_each_channel(seg, filter_fn):
+    n_channels = seg.channels
+
+    channel_segs = seg.split_to_mono()
+    channel_segs = [filter_fn(channel_seg) for channel_seg in channel_segs]
+
+    out_data = seg.get_array_of_samples()
+    for channel_i, channel_seg in enumerate(channel_segs):
+        for sample_i, sample in enumerate(channel_seg.get_array_of_samples()):
+            index = (sample_i * n_channels) + channel_i
+            out_data[index] = sample
+
+    return seg._spawn(out_data)
+
+
+@register_pydub_effect
+def normalize(seg, headroom=0.1):
+    """
+    headroom is how close to the maximum volume to boost the signal up to (specified in dB)
+    """
+    peak_sample_val = seg.max
+    
+    # if the max is 0, this audio segment is silent, and can't be normalized
+    if peak_sample_val == 0:
+        return seg
+    
+    target_peak = seg.max_possible_amplitude * db_to_float(-headroom)
+
+    needed_boost = ratio_to_db(target_peak / peak_sample_val)
+    return seg.apply_gain(needed_boost)
+
+
+@register_pydub_effect
+def speedup(seg, playback_speed=1.5, chunk_size=150, crossfade=25):
+    # we will keep audio in 150ms chunks since one waveform at 20Hz is 50ms long
+    # (20 Hz is the lowest frequency audible to humans)
+
+    # portion of AUDIO TO KEEP. if playback speed is 1.25 we keep 80% (0.8) and
+    # discard 20% (0.2)
+    atk = 1.0 / playback_speed
+
+    if playback_speed < 2.0:
+        # throwing out more than half the audio - keep 50ms chunks
+        ms_to_remove_per_chunk = int(chunk_size * (1 - atk) / atk)
+    else:
+        # throwing out less than half the audio - throw out 50ms chunks
+        ms_to_remove_per_chunk = int(chunk_size)
+        chunk_size = int(atk * chunk_size / (1 - atk))
+
+    # the crossfade cannot be longer than the amount of audio we're removing
+    crossfade = min(crossfade, ms_to_remove_per_chunk - 1)
+
+    # DEBUG
+    #print("chunk: {0}, rm: {1}".format(chunk_size, ms_to_remove_per_chunk))
+
+    chunks = make_chunks(seg, chunk_size + ms_to_remove_per_chunk)
+    if len(chunks) < 2:
+        raise Exception("Could not speed up AudioSegment, it was too short {2:0.2f}s for the current settings:\n{0}ms chunks at {1:0.1f}x speedup".format(
+            chunk_size, playback_speed, seg.duration_seconds))
+
+    # we'll actually truncate a bit less than we calculated to make up for the
+    # crossfade between chunks
+    ms_to_remove_per_chunk -= crossfade
+
+    # we don't want to truncate the last chunk since it is not guaranteed to be
+    # the full chunk length
+    last_chunk = chunks[-1]
+    chunks = [chunk[:-ms_to_remove_per_chunk] for chunk in chunks[:-1]]
+
+    out = chunks[0]
+    for chunk in chunks[1:]:
+        out = out.append(chunk, crossfade=crossfade)
+
+    out += last_chunk
+    return out
+    
+
+@register_pydub_effect
+def strip_silence(seg, silence_len=1000, silence_thresh=-16, padding=100):
+    if padding > silence_len:
+        raise InvalidDuration("padding cannot be longer than silence_len")
+
+    chunks = split_on_silence(seg, silence_len, silence_thresh, padding)
+    crossfade = padding / 2
+
+    if not len(chunks):
+        return seg[0:0]
+
+    seg = chunks[0]
+    for chunk in chunks[1:]:
+        seg = seg.append(chunk, crossfade=crossfade)
+
+    return seg
+
+
+@register_pydub_effect
+def compress_dynamic_range(seg, threshold=-20.0, ratio=4.0, attack=5.0, release=50.0):
+    """
+    Keyword Arguments:
+        
+        threshold - default: -20.0
+            Threshold in dBFS. default of -20.0 means -20dB relative to the
+            maximum possible volume. 0dBFS is the maximum possible value so
+            all values for this argument sould be negative.
+
+        ratio - default: 4.0
+            Compression ratio. Audio louder than the threshold will be 
+            reduced to 1/ratio the volume. A ratio of 4.0 is equivalent to
+            a setting of 4:1 in a pro-audio compressor like the Waves C1.
+        
+        attack - default: 5.0
+            Attack in milliseconds. How long it should take for the compressor
+            to kick in once the audio has exceeded the threshold.
+
+        release - default: 50.0
+            Release in milliseconds. How long it should take for the compressor
+            to stop compressing after the audio has falled below the threshold.
+
+    
+    For an overview of Dynamic Range Compression, and more detailed explanation
+    of the related terminology, see: 
+
+        http://en.wikipedia.org/wiki/Dynamic_range_compression
+    """
+
+    thresh_rms = seg.max_possible_amplitude * db_to_float(threshold)
+    
+    look_frames = int(seg.frame_count(ms=attack))
+    def rms_at(frame_i):
+        return seg.get_sample_slice(frame_i - look_frames, frame_i).rms
+    def db_over_threshold(rms):
+        if rms == 0: return 0.0
+        db = ratio_to_db(rms / thresh_rms)
+        return max(db, 0)
+
+    output = []
+
+    # amount to reduce the volume of the audio by (in dB)
+    attenuation = 0.0
+    
+    attack_frames = seg.frame_count(ms=attack)
+    release_frames = seg.frame_count(ms=release)
+    for i in xrange(int(seg.frame_count())):
+        rms_now = rms_at(i)
+        
+        # with a ratio of 4.0 this means the volume will exceed the threshold by
+        # 1/4 the amount (of dB) that it would otherwise
+        max_attenuation = (1 - (1.0 / ratio)) * db_over_threshold(rms_now)
+        
+        attenuation_inc = max_attenuation / attack_frames
+        attenuation_dec = max_attenuation / release_frames
+        
+        if rms_now > thresh_rms and attenuation <= max_attenuation:
+            attenuation += attenuation_inc
+            attenuation = min(attenuation, max_attenuation)
+        else:
+            attenuation -= attenuation_dec
+            attenuation = max(attenuation, 0)
+        
+        frame = seg.get_frame(i)
+        if attenuation != 0.0:
+            frame = audioop.mul(frame,
+                                seg.sample_width,
+                                db_to_float(-attenuation))
+        
+        output.append(frame)
+    
+    return seg._spawn(data=b''.join(output))
+
+
+# Invert the phase of the signal.
+
+@register_pydub_effect
+
+def invert_phase(seg, channels=(1, 1)):
+    """
+    channels- specifies which channel (left or right) to reverse the phase of.
+    Note that mono AudioSegments will become stereo.
+    """
+    if channels == (1, 1):
+        inverted = audioop.mul(seg._data, seg.sample_width, -1.0)  
+        return seg._spawn(data=inverted)
+    
+    else:
+        if seg.channels == 2:
+            left, right = seg.split_to_mono()
+        else:
+            raise Exception("Can't implicitly convert an AudioSegment with " + str(seg.channels) + " channels to stereo.")
+            
+        if channels == (1, 0):    
+            left = left.invert_phase()
+        else:
+            right = right.invert_phase()
+        
+        return seg.from_mono_audiosegments(left, right)
+        
+
+
+# High and low pass filters based on implementation found on Stack Overflow:
+#   http://stackoverflow.com/questions/13882038/implementing-simple-high-and-low-pass-filters-in-c
+
+@register_pydub_effect
+def low_pass_filter(seg, cutoff):
+    """
+        cutoff - Frequency (in Hz) where higher frequency signal will begin to
+            be reduced by 6dB per octave (doubling in frequency) above this point
+    """
+    RC = 1.0 / (cutoff * 2 * math.pi)
+    dt = 1.0 / seg.frame_rate
+
+    alpha = dt / (RC + dt)
+    
+    original = seg.get_array_of_samples()
+    filteredArray = array.array(seg.array_type, original)
+    
+    frame_count = int(seg.frame_count())
+
+    last_val = [0] * seg.channels
+    for i in range(seg.channels):
+        last_val[i] = filteredArray[i] = original[i]
+
+    for i in range(1, frame_count):
+        for j in range(seg.channels):
+            offset = (i * seg.channels) + j
+            last_val[j] = last_val[j] + (alpha * (original[offset] - last_val[j]))
+            filteredArray[offset] = int(last_val[j])
+
+    return seg._spawn(data=filteredArray)
+
+
+@register_pydub_effect
+def high_pass_filter(seg, cutoff):
+    """
+        cutoff - Frequency (in Hz) where lower frequency signal will begin to
+            be reduced by 6dB per octave (doubling in frequency) below this point
+    """
+    RC = 1.0 / (cutoff * 2 * math.pi)
+    dt = 1.0 / seg.frame_rate
+
+    alpha = RC / (RC + dt)
+
+    minval, maxval = get_min_max_value(seg.sample_width * 8)
+    
+    original = seg.get_array_of_samples()
+    filteredArray = array.array(seg.array_type, original)
+    
+    frame_count = int(seg.frame_count())
+
+    last_val = [0] * seg.channels
+    for i in range(seg.channels):
+        last_val[i] = filteredArray[i] = original[i]
+
+    for i in range(1, frame_count):
+        for j in range(seg.channels):
+            offset = (i * seg.channels) + j
+            offset_minus_1 = ((i-1) * seg.channels) + j
+
+            last_val[j] = alpha * (last_val[j] + original[offset] - original[offset_minus_1])
+            filteredArray[offset] = int(min(max(last_val[j], minval), maxval))
+
+    return seg._spawn(data=filteredArray)
+    
+    
+@register_pydub_effect
+def pan(seg, pan_amount):
+    """
+    pan_amount should be between -1.0 (100% left) and +1.0 (100% right)
+    
+    When pan_amount == 0.0 the left/right balance is not changed.
+    
+    Panning does not alter the *perceived* loundness, but since loudness
+    is decreasing on one side, the other side needs to get louder to
+    compensate. When panned hard left, the left channel will be 3dB louder.
+    """
+    if not -1.0 <= pan_amount <= 1.0:
+        raise ValueError("pan_amount should be between -1.0 (100% left) and +1.0 (100% right)")
+    
+    max_boost_db = ratio_to_db(2.0)
+    boost_db = abs(pan_amount) * max_boost_db
+    
+    boost_factor = db_to_float(boost_db)
+    reduce_factor = db_to_float(max_boost_db) - boost_factor
+    
+    reduce_db = ratio_to_db(reduce_factor)
+    
+    # Cut boost in half (max boost== 3dB) - in reality 2 speakers
+    #   do not sum to a full 6 dB.
+    boost_db = boost_db / 2.0
+    
+    if pan_amount < 0:
+        return seg.apply_gain_stereo(boost_db, reduce_db)
+    else:
+        return seg.apply_gain_stereo(reduce_db, boost_db)
+        
+    
+@register_pydub_effect
+def apply_gain_stereo(seg, left_gain=0.0, right_gain=0.0):
+    """
+    left_gain - amount of gain to apply to the left channel (in dB)
+    right_gain - amount of gain to apply to the right channel (in dB)
+    
+    note: mono audio segments will be converted to stereo
+    """
+    if seg.channels == 1:
+        left = right = seg
+    elif seg.channels == 2:
+        left, right = seg.split_to_mono()
+    
+    l_mult_factor = db_to_float(left_gain)
+    r_mult_factor = db_to_float(right_gain)
+    
+    left_data = audioop.mul(left._data, left.sample_width, l_mult_factor)
+    left_data = audioop.tostereo(left_data, left.sample_width, 1, 0)
+    
+    right_data = audioop.mul(right._data, right.sample_width, r_mult_factor)
+    right_data = audioop.tostereo(right_data, right.sample_width, 0, 1)
+    
+    output = audioop.add(left_data, right_data, seg.sample_width)
+    
+    return seg._spawn(data=output,
+                overrides={'channels': 2,
+                           'frame_width': 2 * seg.sample_width})

venv/lib/python3.11/site-packages/pydub/exceptions.py

@@ -0,0 +1,32 @@
+class PydubException(Exception):
+    """
+    Base class for any Pydub exception
+    """
+
+
+class TooManyMissingFrames(PydubException):
+    pass
+
+
+class InvalidDuration(PydubException):
+    pass
+
+
+class InvalidTag(PydubException):
+    pass
+
+
+class InvalidID3TagVersion(PydubException):
+    pass
+
+
+class CouldntDecodeError(PydubException):
+    pass
+
+
+class CouldntEncodeError(PydubException):
+    pass
+
+
+class MissingAudioParameter(PydubException):
+    pass

venv/lib/python3.11/site-packages/pydub/generators.py

@@ -0,0 +1,142 @@
+"""
+Each generator will return float samples from -1.0 to 1.0, which can be 
+converted to actual audio with 8, 16, 24, or 32 bit depth using the
+SiganlGenerator.to_audio_segment() method (on any of it's subclasses).
+
+See Wikipedia's "waveform" page for info on some of the generators included 
+here: http://en.wikipedia.org/wiki/Waveform
+"""
+
+import math
+import array
+import itertools
+import random
+from .audio_segment import AudioSegment
+from .utils import (
+    db_to_float,
+    get_frame_width,
+    get_array_type,
+    get_min_max_value
+)
+
+
+
+class SignalGenerator(object):
+    def __init__(self, sample_rate=44100, bit_depth=16):
+        self.sample_rate = sample_rate
+        self.bit_depth = bit_depth
+
+    def to_audio_segment(self, duration=1000.0, volume=0.0):
+        """
+        Duration in milliseconds
+            (default: 1 second)
+        Volume in DB relative to maximum amplitude
+            (default 0.0 dBFS, which is the maximum value)
+        """
+        minval, maxval = get_min_max_value(self.bit_depth)
+        sample_width = get_frame_width(self.bit_depth)
+        array_type = get_array_type(self.bit_depth)
+
+        gain = db_to_float(volume)
+        sample_count = int(self.sample_rate * (duration / 1000.0))
+
+        sample_data = (int(val * maxval * gain) for val in self.generate())
+        sample_data = itertools.islice(sample_data, 0, sample_count)
+
+        data = array.array(array_type, sample_data)
+        
+        try:
+            data = data.tobytes()
+        except:
+            data = data.tostring()
+
+        return AudioSegment(data=data, metadata={
+            "channels": 1,
+            "sample_width": sample_width,
+            "frame_rate": self.sample_rate,
+            "frame_width": sample_width,
+        })
+
+    def generate(self):
+        raise NotImplementedError("SignalGenerator subclasses must implement the generate() method, and *should not* call the superclass implementation.")
+
+
+
+class Sine(SignalGenerator):
+    def __init__(self, freq, **kwargs):
+        super(Sine, self).__init__(**kwargs)
+        self.freq = freq
+
+    def generate(self):
+        sine_of = (self.freq * 2 * math.pi) / self.sample_rate
+        sample_n = 0
+        while True:
+            yield math.sin(sine_of * sample_n)
+            sample_n += 1
+
+
+
+class Pulse(SignalGenerator):
+    def __init__(self, freq, duty_cycle=0.5, **kwargs):
+        super(Pulse, self).__init__(**kwargs)
+        self.freq = freq
+        self.duty_cycle = duty_cycle
+
+    def generate(self):
+        sample_n = 0
+
+        # in samples
+        cycle_length = self.sample_rate / float(self.freq)
+        pulse_length = cycle_length * self.duty_cycle
+
+        while True:
+            if (sample_n % cycle_length) < pulse_length:
+                yield 1.0
+            else:
+                yield -1.0
+            sample_n += 1
+
+
+
+class Square(Pulse):
+    def __init__(self, freq, **kwargs):
+        kwargs['duty_cycle'] = 0.5
+        super(Square, self).__init__(freq, **kwargs)
+
+
+
+class Sawtooth(SignalGenerator):
+    def __init__(self, freq, duty_cycle=1.0, **kwargs):
+        super(Sawtooth, self).__init__(**kwargs)
+        self.freq = freq
+        self.duty_cycle = duty_cycle
+
+    def generate(self):
+        sample_n = 0
+
+        # in samples
+        cycle_length = self.sample_rate / float(self.freq)
+        midpoint = cycle_length * self.duty_cycle
+        ascend_length = midpoint
+        descend_length = cycle_length - ascend_length
+
+        while True:
+            cycle_position = sample_n % cycle_length
+            if cycle_position < midpoint:
+                yield (2 * cycle_position / ascend_length) - 1.0
+            else:
+                yield 1.0 - (2 * (cycle_position - midpoint) / descend_length)
+            sample_n += 1
+
+
+
+class Triangle(Sawtooth):
+    def __init__(self, freq, **kwargs):
+        kwargs['duty_cycle'] = 0.5
+        super(Triangle, self).__init__(freq, **kwargs)
+
+
+class WhiteNoise(SignalGenerator):
+    def generate(self):
+        while True:
+            yield (random.random() * 2) - 1.0

venv/lib/python3.11/site-packages/pydub/logging_utils.py

@@ -0,0 +1,14 @@
+"""
+
+"""
+import logging
+
+converter_logger = logging.getLogger("pydub.converter")
+
+def log_conversion(conversion_command):
+    converter_logger.debug("subprocess.call(%s)", repr(conversion_command))
+
+def log_subprocess_output(output):
+    if output:
+        for line in output.rstrip().splitlines():
+            converter_logger.debug('subprocess output: %s', line.rstrip())

venv/lib/python3.11/site-packages/pydub/playback.py

@@ -0,0 +1,71 @@
+"""
+Support for playing AudioSegments. Pyaudio will be used if it's installed,
+otherwise will fallback to ffplay. Pyaudio is a *much* nicer solution, but
+is tricky to install. See my notes on installing pyaudio in a virtualenv (on
+OSX 10.10): https://gist.github.com/jiaaro/9767512210a1d80a8a0d
+"""
+
+import subprocess
+from tempfile import NamedTemporaryFile
+from .utils import get_player_name, make_chunks
+
+def _play_with_ffplay(seg):
+    PLAYER = get_player_name()
+    with NamedTemporaryFile("w+b", suffix=".wav") as f:
+        seg.export(f.name, "wav")
+        subprocess.call([PLAYER, "-nodisp", "-autoexit", "-hide_banner", f.name])
+
+
+def _play_with_pyaudio(seg):
+    import pyaudio
+
+    p = pyaudio.PyAudio()
+    stream = p.open(format=p.get_format_from_width(seg.sample_width),
+                    channels=seg.channels,
+                    rate=seg.frame_rate,
+                    output=True)
+
+    # Just in case there were any exceptions/interrupts, we release the resource
+    # So as not to raise OSError: Device Unavailable should play() be used again
+    try:
+        # break audio into half-second chunks (to allows keyboard interrupts)
+        for chunk in make_chunks(seg, 500):
+            stream.write(chunk._data)
+    finally:
+        stream.stop_stream()
+        stream.close()
+
+        p.terminate()
+
+
+def _play_with_simpleaudio(seg):
+    import simpleaudio
+    return simpleaudio.play_buffer(
+        seg.raw_data,
+        num_channels=seg.channels,
+        bytes_per_sample=seg.sample_width,
+        sample_rate=seg.frame_rate
+    )
+
+
+def play(audio_segment):
+    try:
+        playback = _play_with_simpleaudio(audio_segment)
+        try:
+            playback.wait_done()
+        except KeyboardInterrupt:
+            playback.stop()
+    except ImportError:
+        pass
+    else:
+        return
+
+    try:
+        _play_with_pyaudio(audio_segment)
+        return
+    except ImportError:
+        pass
+    else:
+        return
+
+    _play_with_ffplay(audio_segment)

venv/lib/python3.11/site-packages/pydub/pyaudioop.py

@@ -0,0 +1,553 @@
+try:
+    from __builtin__ import max as builtin_max
+    from __builtin__ import min as builtin_min
+except ImportError:
+    from builtins import max as builtin_max
+    from builtins import min as builtin_min
+import math
+import struct
+try:
+    from fractions import gcd
+except ImportError:  # Python 3.9+
+    from math import gcd
+from ctypes import create_string_buffer
+
+
+class error(Exception):
+    pass
+
+
+def _check_size(size):
+    if size != 1 and size != 2 and size != 4:
+        raise error("Size should be 1, 2 or 4")
+
+
+def _check_params(length, size):
+    _check_size(size)
+    if length % size != 0:
+        raise error("not a whole number of frames")
+
+
+def _sample_count(cp, size):
+    return len(cp) / size
+
+
+def _get_samples(cp, size, signed=True):
+    for i in range(_sample_count(cp, size)):
+        yield _get_sample(cp, size, i, signed)
+
+
+def _struct_format(size, signed):
+    if size == 1:
+        return "b" if signed else "B"
+    elif size == 2:
+        return "h" if signed else "H"
+    elif size == 4:
+        return "i" if signed else "I"
+
+
+def _get_sample(cp, size, i, signed=True):
+    fmt = _struct_format(size, signed)
+    start = i * size
+    end = start + size
+    return struct.unpack_from(fmt, buffer(cp)[start:end])[0]
+
+
+def _put_sample(cp, size, i, val, signed=True):
+    fmt = _struct_format(size, signed)
+    struct.pack_into(fmt, cp, i * size, val)
+
+
+def _get_maxval(size, signed=True):
+    if signed and size == 1:
+        return 0x7f
+    elif size == 1:
+        return 0xff
+    elif signed and size == 2:
+        return 0x7fff
+    elif size == 2:
+        return 0xffff
+    elif signed and size == 4:
+        return 0x7fffffff
+    elif size == 4:
+        return 0xffffffff
+
+
+def _get_minval(size, signed=True):
+    if not signed:
+        return 0
+    elif size == 1:
+        return -0x80
+    elif size == 2:
+        return -0x8000
+    elif size == 4:
+        return -0x80000000
+
+
+def _get_clipfn(size, signed=True):
+    maxval = _get_maxval(size, signed)
+    minval = _get_minval(size, signed)
+    return lambda val: builtin_max(min(val, maxval), minval)
+
+
+def _overflow(val, size, signed=True):
+    minval = _get_minval(size, signed)
+    maxval = _get_maxval(size, signed)
+    if minval <= val <= maxval:
+        return val
+
+    bits = size * 8
+    if signed:
+        offset = 2**(bits-1)
+        return ((val + offset) % (2**bits)) - offset
+    else:
+        return val % (2**bits)
+
+
+def getsample(cp, size, i):
+    _check_params(len(cp), size)
+    if not (0 <= i < len(cp) / size):
+        raise error("Index out of range")
+    return _get_sample(cp, size, i)
+
+
+def max(cp, size):
+    _check_params(len(cp), size)
+
+    if len(cp) == 0:
+        return 0
+
+    return builtin_max(abs(sample) for sample in _get_samples(cp, size))
+
+
+def minmax(cp, size):
+    _check_params(len(cp), size)
+
+    max_sample, min_sample = 0, 0
+    for sample in _get_samples(cp, size):
+        max_sample = builtin_max(sample, max_sample)
+        min_sample = builtin_min(sample, min_sample)
+
+    return min_sample, max_sample
+
+
+def avg(cp, size):
+    _check_params(len(cp), size)
+    sample_count = _sample_count(cp, size)
+    if sample_count == 0:
+        return 0
+    return sum(_get_samples(cp, size)) / sample_count
+
+
+def rms(cp, size):
+    _check_params(len(cp), size)
+
+    sample_count = _sample_count(cp, size)
+    if sample_count == 0:
+        return 0
+
+    sum_squares = sum(sample**2 for sample in _get_samples(cp, size))
+    return int(math.sqrt(sum_squares / sample_count))
+
+
+def _sum2(cp1, cp2, length):
+    size = 2
+    total = 0
+    for i in range(length):
+        total += getsample(cp1, size, i) * getsample(cp2, size, i)
+    return total
+
+
+def findfit(cp1, cp2):
+    size = 2
+
+    if len(cp1) % 2 != 0 or len(cp2) % 2 != 0:
+        raise error("Strings should be even-sized")
+
+    if len(cp1) < len(cp2):
+        raise error("First sample should be longer")
+
+    len1 = _sample_count(cp1, size)
+    len2 = _sample_count(cp2, size)
+
+    sum_ri_2 = _sum2(cp2, cp2, len2)
+    sum_aij_2 = _sum2(cp1, cp1, len2)
+    sum_aij_ri = _sum2(cp1, cp2, len2)
+
+    result = (sum_ri_2 * sum_aij_2 - sum_aij_ri * sum_aij_ri) / sum_aij_2
+
+    best_result = result
+    best_i = 0
+
+    for i in range(1, len1 - len2 + 1):
+        aj_m1 = _get_sample(cp1, size, i - 1)
+        aj_lm1 = _get_sample(cp1, size, i + len2 - 1)
+
+        sum_aij_2 += aj_lm1**2 - aj_m1**2
+        sum_aij_ri = _sum2(buffer(cp1)[i*size:], cp2, len2)
+
+        result = (sum_ri_2 * sum_aij_2 - sum_aij_ri * sum_aij_ri) / sum_aij_2
+
+        if result < best_result:
+            best_result = result
+            best_i = i
+
+    factor = _sum2(buffer(cp1)[best_i*size:], cp2, len2) / sum_ri_2
+
+    return best_i, factor
+
+
+def findfactor(cp1, cp2):
+    size = 2
+
+    if len(cp1) % 2 != 0:
+        raise error("Strings should be even-sized")
+
+    if len(cp1) != len(cp2):
+        raise error("Samples should be same size")
+
+    sample_count = _sample_count(cp1, size)
+
+    sum_ri_2 = _sum2(cp2, cp2, sample_count)
+    sum_aij_ri = _sum2(cp1, cp2, sample_count)
+
+    return sum_aij_ri / sum_ri_2
+
+
+def findmax(cp, len2):
+    size = 2
+    sample_count = _sample_count(cp, size)
+
+    if len(cp) % 2 != 0:
+        raise error("Strings should be even-sized")
+
+    if len2 < 0 or sample_count < len2:
+        raise error("Input sample should be longer")
+
+    if sample_count == 0:
+        return 0
+
+    result = _sum2(cp, cp, len2)
+    best_result = result
+    best_i = 0
+
+    for i in range(1, sample_count - len2 + 1):
+        sample_leaving_window = getsample(cp, size, i - 1)
+        sample_entering_window = getsample(cp, size, i + len2 - 1)
+
+        result -= sample_leaving_window**2
+        result += sample_entering_window**2
+
+        if result > best_result:
+            best_result = result
+            best_i = i
+
+    return best_i
+
+
+def avgpp(cp, size):
+    _check_params(len(cp), size)
+    sample_count = _sample_count(cp, size)
+
+    prevextremevalid = False
+    prevextreme = None
+    avg = 0
+    nextreme = 0
+
+    prevval = getsample(cp, size, 0)
+    val = getsample(cp, size, 1)
+
+    prevdiff = val - prevval
+
+    for i in range(1, sample_count):
+        val = getsample(cp, size, i)
+        diff = val - prevval
+
+        if diff * prevdiff < 0:
+            if prevextremevalid:
+                avg += abs(prevval - prevextreme)
+                nextreme += 1
+
+            prevextremevalid = True
+            prevextreme = prevval
+
+        prevval = val
+        if diff != 0:
+            prevdiff = diff
+
+    if nextreme == 0:
+        return 0
+
+    return avg / nextreme
+
+
+def maxpp(cp, size):
+    _check_params(len(cp), size)
+    sample_count = _sample_count(cp, size)
+
+    prevextremevalid = False
+    prevextreme = None
+    max = 0
+
+    prevval = getsample(cp, size, 0)
+    val = getsample(cp, size, 1)
+
+    prevdiff = val - prevval
+
+    for i in range(1, sample_count):
+        val = getsample(cp, size, i)
+        diff = val - prevval
+
+        if diff * prevdiff < 0:
+            if prevextremevalid:
+                extremediff = abs(prevval - prevextreme)
+                if extremediff > max:
+                    max = extremediff
+            prevextremevalid = True
+            prevextreme = prevval
+
+        prevval = val
+        if diff != 0:
+            prevdiff = diff
+
+    return max
+
+
+def cross(cp, size):
+    _check_params(len(cp), size)
+
+    crossings = 0
+    last_sample = 0
+    for sample in _get_samples(cp, size):
+        if sample <= 0 < last_sample or sample >= 0 > last_sample:
+            crossings += 1
+        last_sample = sample
+
+    return crossings
+
+
+def mul(cp, size, factor):
+    _check_params(len(cp), size)
+    clip = _get_clipfn(size)
+
+    result = create_string_buffer(len(cp))
+
+    for i, sample in enumerate(_get_samples(cp, size)):
+        sample = clip(int(sample * factor))
+        _put_sample(result, size, i, sample)
+
+    return result.raw
+
+
+def tomono(cp, size, fac1, fac2):
+    _check_params(len(cp), size)
+    clip = _get_clipfn(size)
+
+    sample_count = _sample_count(cp, size)
+
+    result = create_string_buffer(len(cp) / 2)
+
+    for i in range(0, sample_count, 2):
+        l_sample = getsample(cp, size, i)
+        r_sample = getsample(cp, size, i + 1)
+
+        sample = (l_sample * fac1) + (r_sample * fac2)
+        sample = clip(sample)
+
+        _put_sample(result, size, i / 2, sample)
+
+    return result.raw
+
+
+def tostereo(cp, size, fac1, fac2):
+    _check_params(len(cp), size)
+
+    sample_count = _sample_count(cp, size)
+
+    result = create_string_buffer(len(cp) * 2)
+    clip = _get_clipfn(size)
+
+    for i in range(sample_count):
+        sample = _get_sample(cp, size, i)
+
+        l_sample = clip(sample * fac1)
+        r_sample = clip(sample * fac2)
+
+        _put_sample(result, size, i * 2, l_sample)
+        _put_sample(result, size, i * 2 + 1, r_sample)
+
+    return result.raw
+
+
+def add(cp1, cp2, size):
+    _check_params(len(cp1), size)
+
+    if len(cp1) != len(cp2):
+        raise error("Lengths should be the same")
+
+    clip = _get_clipfn(size)
+    sample_count = _sample_count(cp1, size)
+    result = create_string_buffer(len(cp1))
+
+    for i in range(sample_count):
+        sample1 = getsample(cp1, size, i)
+        sample2 = getsample(cp2, size, i)
+
+        sample = clip(sample1 + sample2)
+
+        _put_sample(result, size, i, sample)
+
+    return result.raw
+
+
+def bias(cp, size, bias):
+    _check_params(len(cp), size)
+
+    result = create_string_buffer(len(cp))
+
+    for i, sample in enumerate(_get_samples(cp, size)):
+        sample = _overflow(sample + bias, size)
+        _put_sample(result, size, i, sample)
+
+    return result.raw
+
+
+def reverse(cp, size):
+    _check_params(len(cp), size)
+    sample_count = _sample_count(cp, size)
+
+    result = create_string_buffer(len(cp))
+    for i, sample in enumerate(_get_samples(cp, size)):
+        _put_sample(result, size, sample_count - i - 1, sample)
+
+    return result.raw
+
+
+def lin2lin(cp, size, size2):
+    _check_params(len(cp), size)
+    _check_size(size2)
+
+    if size == size2:
+        return cp
+
+    new_len = (len(cp) / size) * size2
+
+    result = create_string_buffer(new_len)
+
+    for i in range(_sample_count(cp, size)):
+        sample = _get_sample(cp, size, i)
+        if size < size2:
+            sample = sample << (4 * size2 / size)
+        elif size > size2:
+            sample = sample >> (4 * size / size2)
+
+        sample = _overflow(sample, size2)
+
+        _put_sample(result, size2, i, sample)
+
+    return result.raw
+
+
+def ratecv(cp, size, nchannels, inrate, outrate, state, weightA=1, weightB=0):
+    _check_params(len(cp), size)
+    if nchannels < 1:
+        raise error("# of channels should be >= 1")
+
+    bytes_per_frame = size * nchannels
+    frame_count = len(cp) / bytes_per_frame
+
+    if bytes_per_frame / nchannels != size:
+        raise OverflowError("width * nchannels too big for a C int")
+
+    if weightA < 1 or weightB < 0:
+        raise error("weightA should be >= 1, weightB should be >= 0")
+
+    if len(cp) % bytes_per_frame != 0:
+        raise error("not a whole number of frames")
+
+    if inrate <= 0 or outrate <= 0:
+        raise error("sampling rate not > 0")
+
+    d = gcd(inrate, outrate)
+    inrate /= d
+    outrate /= d
+
+    prev_i = [0] * nchannels
+    cur_i = [0] * nchannels
+
+    if state is None:
+        d = -outrate
+    else:
+        d, samps = state
+
+        if len(samps) != nchannels:
+            raise error("illegal state argument")
+
+        prev_i, cur_i = zip(*samps)
+        prev_i, cur_i = list(prev_i), list(cur_i)
+
+    q = frame_count / inrate
+    ceiling = (q + 1) * outrate
+    nbytes = ceiling * bytes_per_frame
+
+    result = create_string_buffer(nbytes)
+
+    samples = _get_samples(cp, size)
+    out_i = 0
+    while True:
+        while d < 0:
+            if frame_count == 0:
+                samps = zip(prev_i, cur_i)
+                retval = result.raw
+
+                # slice off extra bytes
+                trim_index = (out_i * bytes_per_frame) - len(retval)
+                retval = buffer(retval)[:trim_index]
+
+                return (retval, (d, tuple(samps)))
+
+            for chan in range(nchannels):
+                prev_i[chan] = cur_i[chan]
+                cur_i[chan] = samples.next()
+
+                cur_i[chan] = (
+                    (weightA * cur_i[chan] + weightB * prev_i[chan])
+                    / (weightA + weightB)
+                )
+
+            frame_count -= 1
+            d += outrate
+
+        while d >= 0:
+            for chan in range(nchannels):
+                cur_o = (
+                    (prev_i[chan] * d + cur_i[chan] * (outrate - d))
+                    / outrate
+                )
+                _put_sample(result, size, out_i, _overflow(cur_o, size))
+                out_i += 1
+            d -= inrate
+
+
+def lin2ulaw(cp, size):
+    raise NotImplementedError()
+
+
+def ulaw2lin(cp, size):
+    raise NotImplementedError()
+
+
+def lin2alaw(cp, size):
+    raise NotImplementedError()
+
+
+def alaw2lin(cp, size):
+    raise NotImplementedError()
+
+
+def lin2adpcm(cp, size, state):
+    raise NotImplementedError()
+
+
+def adpcm2lin(cp, size, state):
+    raise NotImplementedError()

venv/lib/python3.11/site-packages/pydub/scipy_effects.py

@@ -0,0 +1,175 @@
+"""
+This module provides scipy versions of high_pass_filter, and low_pass_filter
+as well as an additional band_pass_filter.
+
+Of course, you will need to install scipy for these to work.
+
+When this module is imported the high and low pass filters from this module
+will be used when calling audio_segment.high_pass_filter() and
+audio_segment.high_pass_filter() instead of the slower, less powerful versions
+provided by pydub.effects.
+"""
+from scipy.signal import butter, sosfilt
+from .utils import (register_pydub_effect,stereo_to_ms,ms_to_stereo)
+
+
+def _mk_butter_filter(freq, type, order):
+    """
+    Args:
+        freq: The cutoff frequency for highpass and lowpass filters. For
+            band filters, a list of [low_cutoff, high_cutoff]
+        type: "lowpass", "highpass", or "band"
+        order: nth order butterworth filter (default: 5th order). The
+            attenuation is -6dB/octave beyond the cutoff frequency (for 1st
+            order). A Higher order filter will have more attenuation, each level
+            adding an additional -6dB (so a 3rd order butterworth filter would
+            be -18dB/octave).
+
+    Returns:
+        function which can filter a mono audio segment
+
+    """
+    def filter_fn(seg):
+        assert seg.channels == 1
+
+        nyq = 0.5 * seg.frame_rate
+        try:
+            freqs = [f / nyq for f in freq]
+        except TypeError:
+            freqs = freq / nyq
+
+        sos = butter(order, freqs, btype=type, output='sos')
+        y = sosfilt(sos, seg.get_array_of_samples())
+
+        return seg._spawn(y.astype(seg.array_type))
+
+    return filter_fn
+
+
+@register_pydub_effect
+def band_pass_filter(seg, low_cutoff_freq, high_cutoff_freq, order=5):
+    filter_fn = _mk_butter_filter([low_cutoff_freq, high_cutoff_freq], 'band', order=order)
+    return seg.apply_mono_filter_to_each_channel(filter_fn)
+
+
+@register_pydub_effect
+def high_pass_filter(seg, cutoff_freq, order=5):
+    filter_fn = _mk_butter_filter(cutoff_freq, 'highpass', order=order)
+    return seg.apply_mono_filter_to_each_channel(filter_fn)
+
+
+@register_pydub_effect
+def low_pass_filter(seg, cutoff_freq, order=5):
+    filter_fn = _mk_butter_filter(cutoff_freq, 'lowpass', order=order)
+    return seg.apply_mono_filter_to_each_channel(filter_fn)
+
+
+@register_pydub_effect
+def _eq(seg, focus_freq, bandwidth=100, mode="peak", gain_dB=0, order=2):
+    """
+    Args:
+        focus_freq - middle frequency or known frequency of band (in Hz)
+        bandwidth - range of the equalizer band
+        mode - Mode of Equalization(Peak/Notch(Bell Curve),High Shelf, Low Shelf)
+        order - Rolloff factor(1 - 6dB/Octave 2 - 12dB/Octave)
+    
+    Returns:
+        Equalized/Filtered AudioSegment
+    """
+    filt_mode = ["peak", "low_shelf", "high_shelf"]
+    if mode not in filt_mode:
+        raise ValueError("Incorrect Mode Selection")
+        
+    if gain_dB >= 0:
+        if mode == "peak":
+            sec = band_pass_filter(seg, focus_freq - bandwidth/2, focus_freq + bandwidth/2, order = order)
+            seg = seg.overlay(sec - (3 - gain_dB))
+            return seg
+        
+        if mode == "low_shelf":
+            sec = low_pass_filter(seg, focus_freq, order=order)
+            seg = seg.overlay(sec - (3 - gain_dB))
+            return seg
+        
+        if mode == "high_shelf":
+            sec = high_pass_filter(seg, focus_freq, order=order)
+            seg = seg.overlay(sec - (3 - gain_dB))
+            return seg
+        
+    if gain_dB < 0:
+        if mode == "peak":
+            sec = high_pass_filter(seg, focus_freq - bandwidth/2, order=order)
+            seg = seg.overlay(sec - (3 + gain_dB)) + gain_dB
+            sec = low_pass_filter(seg, focus_freq + bandwidth/2, order=order)
+            seg = seg.overlay(sec - (3 + gain_dB)) + gain_dB
+            return seg
+        
+        if mode == "low_shelf":
+            sec = high_pass_filter(seg, focus_freq, order=order)
+            seg = seg.overlay(sec - (3 + gain_dB)) + gain_dB
+            return seg
+        
+        if mode=="high_shelf":
+            sec=low_pass_filter(seg, focus_freq, order=order)
+            seg=seg.overlay(sec - (3 + gain_dB)) +gain_dB
+            return seg
+        
+
+@register_pydub_effect
+def eq(seg, focus_freq, bandwidth=100, channel_mode="L+R", filter_mode="peak", gain_dB=0, order=2):
+    """
+    Args:
+        focus_freq - middle frequency or known frequency of band (in Hz)
+        bandwidth - range of the equalizer band
+        channel_mode - Select Channels to be affected by the filter.
+            L+R - Standard Stereo Filter
+            L - Only Left Channel is Filtered
+            R - Only Right Channel is Filtered
+            M+S - Blumlien Stereo Filter(Mid-Side)
+            M - Only Mid Channel is Filtered
+            S - Only Side Channel is Filtered
+            Mono Audio Segments are completely filtered.
+        filter_mode - Mode of Equalization(Peak/Notch(Bell Curve),High Shelf, Low Shelf)
+        order - Rolloff factor(1 - 6dB/Octave 2 - 12dB/Octave)
+    
+    Returns:
+        Equalized/Filtered AudioSegment
+    """
+    channel_modes = ["L+R", "M+S", "L", "R", "M", "S"]
+    if channel_mode not in channel_modes:
+        raise ValueError("Incorrect Channel Mode Selection")
+        
+    if seg.channels == 1:
+        return _eq(seg, focus_freq, bandwidth, filter_mode, gain_dB, order)
+        
+    if channel_mode == "L+R":
+        return _eq(seg, focus_freq, bandwidth, filter_mode, gain_dB, order)
+        
+    if channel_mode == "L":
+        seg = seg.split_to_mono()
+        seg = [_eq(seg[0], focus_freq, bandwidth, filter_mode, gain_dB, order), seg[1]]
+        return AudioSegment.from_mono_audio_segements(seg[0], seg[1])
+        
+    if channel_mode == "R":
+        seg = seg.split_to_mono()
+        seg = [seg[0], _eq(seg[1], focus_freq, bandwidth, filter_mode, gain_dB, order)]
+        return AudioSegment.from_mono_audio_segements(seg[0], seg[1])
+        
+    if channel_mode == "M+S":
+        seg = stereo_to_ms(seg)
+        seg = _eq(seg, focus_freq, bandwidth, filter_mode, gain_dB, order)
+        return ms_to_stereo(seg)
+        
+    if channel_mode == "M":
+        seg = stereo_to_ms(seg).split_to_mono()
+        seg = [_eq(seg[0], focus_freq, bandwidth, filter_mode, gain_dB, order), seg[1]]
+        seg = AudioSegment.from_mono_audio_segements(seg[0], seg[1])
+        return ms_to_stereo(seg)
+        
+    if channel_mode == "S":
+        seg = stereo_to_ms(seg).split_to_mono()
+        seg = [seg[0], _eq(seg[1], focus_freq, bandwidth, filter_mode, gain_dB, order)]
+        seg = AudioSegment.from_mono_audio_segements(seg[0], seg[1])
+        return ms_to_stereo(seg)
+
+

venv/lib/python3.11/site-packages/pydub/silence.py

@@ -0,0 +1,182 @@
+"""
+Various functions for finding/manipulating silence in AudioSegments
+"""
+import itertools
+
+from .utils import db_to_float
+
+
+def detect_silence(audio_segment, min_silence_len=1000, silence_thresh=-16, seek_step=1):
+    """
+    Returns a list of all silent sections [start, end] in milliseconds of audio_segment.
+    Inverse of detect_nonsilent()
+
+    audio_segment - the segment to find silence in
+    min_silence_len - the minimum length for any silent section
+    silence_thresh - the upper bound for how quiet is silent in dFBS
+    seek_step - step size for interating over the segment in ms
+    """
+    seg_len = len(audio_segment)
+
+    # you can't have a silent portion of a sound that is longer than the sound
+    if seg_len < min_silence_len:
+        return []
+
+    # convert silence threshold to a float value (so we can compare it to rms)
+    silence_thresh = db_to_float(silence_thresh) * audio_segment.max_possible_amplitude
+
+    # find silence and add start and end indicies to the to_cut list
+    silence_starts = []
+
+    # check successive (1 sec by default) chunk of sound for silence
+    # try a chunk at every "seek step" (or every chunk for a seek step == 1)
+    last_slice_start = seg_len - min_silence_len
+    slice_starts = range(0, last_slice_start + 1, seek_step)
+
+    # guarantee last_slice_start is included in the range
+    # to make sure the last portion of the audio is searched
+    if last_slice_start % seek_step:
+        slice_starts = itertools.chain(slice_starts, [last_slice_start])
+
+    for i in slice_starts:
+        audio_slice = audio_segment[i:i + min_silence_len]
+        if audio_slice.rms <= silence_thresh:
+            silence_starts.append(i)
+
+    # short circuit when there is no silence
+    if not silence_starts:
+        return []
+
+    # combine the silence we detected into ranges (start ms - end ms)
+    silent_ranges = []
+
+    prev_i = silence_starts.pop(0)
+    current_range_start = prev_i
+
+    for silence_start_i in silence_starts:
+        continuous = (silence_start_i == prev_i + seek_step)
+
+        # sometimes two small blips are enough for one particular slice to be
+        # non-silent, despite the silence all running together. Just combine
+        # the two overlapping silent ranges.
+        silence_has_gap = silence_start_i > (prev_i + min_silence_len)
+
+        if not continuous and silence_has_gap:
+            silent_ranges.append([current_range_start,
+                                  prev_i + min_silence_len])
+            current_range_start = silence_start_i
+        prev_i = silence_start_i
+
+    silent_ranges.append([current_range_start,
+                          prev_i + min_silence_len])
+
+    return silent_ranges
+
+
+def detect_nonsilent(audio_segment, min_silence_len=1000, silence_thresh=-16, seek_step=1):
+    """
+    Returns a list of all nonsilent sections [start, end] in milliseconds of audio_segment.
+    Inverse of detect_silent()
+
+    audio_segment - the segment to find silence in
+    min_silence_len - the minimum length for any silent section
+    silence_thresh - the upper bound for how quiet is silent in dFBS
+    seek_step - step size for interating over the segment in ms
+    """
+    silent_ranges = detect_silence(audio_segment, min_silence_len, silence_thresh, seek_step)
+    len_seg = len(audio_segment)
+
+    # if there is no silence, the whole thing is nonsilent
+    if not silent_ranges:
+        return [[0, len_seg]]
+
+    # short circuit when the whole audio segment is silent
+    if silent_ranges[0][0] == 0 and silent_ranges[0][1] == len_seg:
+        return []
+
+    prev_end_i = 0
+    nonsilent_ranges = []
+    for start_i, end_i in silent_ranges:
+        nonsilent_ranges.append([prev_end_i, start_i])
+        prev_end_i = end_i
+
+    if end_i != len_seg:
+        nonsilent_ranges.append([prev_end_i, len_seg])
+
+    if nonsilent_ranges[0] == [0, 0]:
+        nonsilent_ranges.pop(0)
+
+    return nonsilent_ranges
+
+
+def split_on_silence(audio_segment, min_silence_len=1000, silence_thresh=-16, keep_silence=100,
+                     seek_step=1):
+    """
+    Returns list of audio segments from splitting audio_segment on silent sections
+
+    audio_segment - original pydub.AudioSegment() object
+
+    min_silence_len - (in ms) minimum length of a silence to be used for
+        a split. default: 1000ms
+
+    silence_thresh - (in dBFS) anything quieter than this will be
+        considered silence. default: -16dBFS
+
+    keep_silence - (in ms or True/False) leave some silence at the beginning
+        and end of the chunks. Keeps the sound from sounding like it
+        is abruptly cut off.
+        When the length of the silence is less than the keep_silence duration
+        it is split evenly between the preceding and following non-silent
+        segments.
+        If True is specified, all the silence is kept, if False none is kept.
+        default: 100ms
+
+    seek_step - step size for interating over the segment in ms
+    """
+
+    # from the itertools documentation
+    def pairwise(iterable):
+        "s -> (s0,s1), (s1,s2), (s2, s3), ..."
+        a, b = itertools.tee(iterable)
+        next(b, None)
+        return zip(a, b)
+
+    if isinstance(keep_silence, bool):
+        keep_silence = len(audio_segment) if keep_silence else 0
+
+    output_ranges = [
+        [ start - keep_silence, end + keep_silence ]
+        for (start,end)
+            in detect_nonsilent(audio_segment, min_silence_len, silence_thresh, seek_step)
+    ]
+
+    for range_i, range_ii in pairwise(output_ranges):
+        last_end = range_i[1]
+        next_start = range_ii[0]
+        if next_start < last_end:
+            range_i[1] = (last_end+next_start)//2
+            range_ii[0] = range_i[1]
+
+    return [
+        audio_segment[ max(start,0) : min(end,len(audio_segment)) ]
+        for start,end in output_ranges
+    ]
+
+
+def detect_leading_silence(sound, silence_threshold=-50.0, chunk_size=10):
+    """
+    Returns the millisecond/index that the leading silence ends.
+
+    audio_segment - the segment to find silence in
+    silence_threshold - the upper bound for how quiet is silent in dFBS
+    chunk_size - chunk size for interating over the segment in ms
+    """
+    trim_ms = 0 # ms
+    assert chunk_size > 0 # to avoid infinite loop
+    while sound[trim_ms:trim_ms+chunk_size].dBFS < silence_threshold and trim_ms < len(sound):
+        trim_ms += chunk_size
+
+    # if there is no end it should return the length of the segment
+    return min(trim_ms, len(sound))
+
+

venv/lib/python3.11/site-packages/pydub/utils.py

@@ -0,0 +1,434 @@
+from __future__ import division
+
+import json
+import os
+import re
+import sys
+from subprocess import Popen, PIPE
+from math import log, ceil
+from tempfile import TemporaryFile
+from warnings import warn
+from functools import wraps
+
+try:
+    import audioop
+except ImportError:
+    import pyaudioop as audioop
+
+if sys.version_info >= (3, 0):
+    basestring = str
+
+FRAME_WIDTHS = {
+    8: 1,
+    16: 2,
+    32: 4,
+}
+ARRAY_TYPES = {
+    8: "b",
+    16: "h",
+    32: "i",
+}
+ARRAY_RANGES = {
+    8: (-0x80, 0x7f),
+    16: (-0x8000, 0x7fff),
+    32: (-0x80000000, 0x7fffffff),
+}
+
+
+def get_frame_width(bit_depth):
+    return FRAME_WIDTHS[bit_depth]
+
+
+def get_array_type(bit_depth, signed=True):
+    t = ARRAY_TYPES[bit_depth]
+    if not signed:
+        t = t.upper()
+    return t
+
+
+def get_min_max_value(bit_depth):
+    return ARRAY_RANGES[bit_depth]
+
+
+def _fd_or_path_or_tempfile(fd, mode='w+b', tempfile=True):
+    close_fd = False
+    if fd is None and tempfile:
+        fd = TemporaryFile(mode=mode)
+        close_fd = True
+
+    if isinstance(fd, basestring):
+        fd = open(fd, mode=mode)
+        close_fd = True
+
+    try:
+        if isinstance(fd, os.PathLike):
+            fd = open(fd, mode=mode)
+            close_fd = True
+    except AttributeError:
+        # module os has no attribute PathLike, so we're on python < 3.6.
+        # The protocol we're trying to support doesn't exist, so just pass.
+        pass
+
+    return fd, close_fd
+
+
+def db_to_float(db, using_amplitude=True):
+    """
+    Converts the input db to a float, which represents the equivalent
+    ratio in power.
+    """
+    db = float(db)
+    if using_amplitude:
+        return 10 ** (db / 20)
+    else:  # using power
+        return 10 ** (db / 10)
+
+
+def ratio_to_db(ratio, val2=None, using_amplitude=True):
+    """
+    Converts the input float to db, which represents the equivalent
+    to the ratio in power represented by the multiplier passed in.
+    """
+    ratio = float(ratio)
+
+    # accept 2 values and use the ratio of val1 to val2
+    if val2 is not None:
+        ratio = ratio / val2
+
+    # special case for multiply-by-zero (convert to silence)
+    if ratio == 0:
+        return -float('inf')
+
+    if using_amplitude:
+        return 20 * log(ratio, 10)
+    else:  # using power
+        return 10 * log(ratio, 10)
+
+
+def register_pydub_effect(fn, name=None):
+    """
+    decorator for adding pydub effects to the AudioSegment objects.
+    example use:
+        @register_pydub_effect
+        def normalize(audio_segment):
+            ...
+    or you can specify a name:
+        @register_pydub_effect("normalize")
+        def normalize_audio_segment(audio_segment):
+            ...
+    """
+    if isinstance(fn, basestring):
+        name = fn
+        return lambda fn: register_pydub_effect(fn, name)
+
+    if name is None:
+        name = fn.__name__
+
+    from .audio_segment import AudioSegment
+    setattr(AudioSegment, name, fn)
+    return fn
+
+
+def make_chunks(audio_segment, chunk_length):
+    """
+    Breaks an AudioSegment into chunks that are <chunk_length> milliseconds
+    long.
+    if chunk_length is 50 then you'll get a list of 50 millisecond long audio
+    segments back (except the last one, which can be shorter)
+    """
+    number_of_chunks = ceil(len(audio_segment) / float(chunk_length))
+    return [audio_segment[i * chunk_length:(i + 1) * chunk_length]
+            for i in range(int(number_of_chunks))]
+
+
+def which(program):
+    """
+    Mimics behavior of UNIX which command.
+    """
+    # Add .exe program extension for windows support
+    if os.name == "nt" and not program.endswith(".exe"):
+        program += ".exe"
+
+    envdir_list = [os.curdir] + os.environ["PATH"].split(os.pathsep)
+
+    for envdir in envdir_list:
+        program_path = os.path.join(envdir, program)
+        if os.path.isfile(program_path) and os.access(program_path, os.X_OK):
+            return program_path
+
+
+def get_encoder_name():
+    """
+    Return enconder default application for system, either avconv or ffmpeg
+    """
+    if which("avconv"):
+        return "avconv"
+    elif which("ffmpeg"):
+        return "ffmpeg"
+    else:
+        # should raise exception
+        warn("Couldn't find ffmpeg or avconv - defaulting to ffmpeg, but may not work", RuntimeWarning)
+        return "ffmpeg"
+
+
+def get_player_name():
+    """
+    Return enconder default application for system, either avconv or ffmpeg
+    """
+    if which("avplay"):
+        return "avplay"
+    elif which("ffplay"):
+        return "ffplay"
+    else:
+        # should raise exception
+        warn("Couldn't find ffplay or avplay - defaulting to ffplay, but may not work", RuntimeWarning)
+        return "ffplay"
+
+
+def get_prober_name():
+    """
+    Return probe application, either avconv or ffmpeg
+    """
+    if which("avprobe"):
+        return "avprobe"
+    elif which("ffprobe"):
+        return "ffprobe"
+    else:
+        # should raise exception
+        warn("Couldn't find ffprobe or avprobe - defaulting to ffprobe, but may not work", RuntimeWarning)
+        return "ffprobe"
+
+
+def fsdecode(filename):
+    """Wrapper for os.fsdecode which was introduced in python 3.2 ."""
+
+    if sys.version_info >= (3, 2):
+        PathLikeTypes = (basestring, bytes)
+        if sys.version_info >= (3, 6):
+            PathLikeTypes += (os.PathLike,)
+        if isinstance(filename, PathLikeTypes):
+            return os.fsdecode(filename)
+    else:
+        if isinstance(filename, bytes):
+            return filename.decode(sys.getfilesystemencoding())
+        if isinstance(filename, basestring):
+            return filename
+
+    raise TypeError("type {0} not accepted by fsdecode".format(type(filename)))
+
+
+def get_extra_info(stderr):
+    """
+    avprobe sometimes gives more information on stderr than
+    on the json output. The information has to be extracted
+    from stderr of the format of:
+    '    Stream #0:0: Audio: flac, 88200 Hz, stereo, s32 (24 bit)'
+    or (macOS version):
+    '    Stream #0:0: Audio: vorbis'
+    '      44100 Hz, stereo, fltp, 320 kb/s'
+
+    :type stderr: str
+    :rtype: list of dict
+    """
+    extra_info = {}
+
+    re_stream = r'(?P<space_start> +)Stream #0[:\.](?P<stream_id>([0-9]+))(?P<content_0>.+)\n?(?! *Stream)((?P<space_end> +)(?P<content_1>.+))?'
+    for i in re.finditer(re_stream, stderr):
+        if i.group('space_end') is not None and len(i.group('space_start')) <= len(
+                i.group('space_end')):
+            content_line = ','.join([i.group('content_0'), i.group('content_1')])
+        else:
+            content_line = i.group('content_0')
+        tokens = [x.strip() for x in re.split('[:,]', content_line) if x]
+        extra_info[int(i.group('stream_id'))] = tokens
+    return extra_info
+
+
+def mediainfo_json(filepath, read_ahead_limit=-1):
+    """Return json dictionary with media info(codec, duration, size, bitrate...) from filepath
+    """
+    prober = get_prober_name()
+    command_args = [
+        "-v", "info",
+        "-show_format",
+        "-show_streams",
+    ]
+    try:
+        command_args += [fsdecode(filepath)]
+        stdin_parameter = None
+        stdin_data = None
+    except TypeError:
+        if prober == 'ffprobe':
+            command_args += ["-read_ahead_limit", str(read_ahead_limit),
+                             "cache:pipe:0"]
+        else:
+            command_args += ["-"]
+        stdin_parameter = PIPE
+        file, close_file = _fd_or_path_or_tempfile(filepath, 'rb', tempfile=False)
+        file.seek(0)
+        stdin_data = file.read()
+        if close_file:
+            file.close()
+
+    command = [prober, '-of', 'json'] + command_args
+    res = Popen(command, stdin=stdin_parameter, stdout=PIPE, stderr=PIPE)
+    output, stderr = res.communicate(input=stdin_data)
+    output = output.decode("utf-8", 'ignore')
+    stderr = stderr.decode("utf-8", 'ignore')
+
+    info = json.loads(output)
+
+    if not info:
+        # If ffprobe didn't give any information, just return it
+        # (for example, because the file doesn't exist)
+        return info
+
+    extra_info = get_extra_info(stderr)
+
+    audio_streams = [x for x in info['streams'] if x['codec_type'] == 'audio']
+    if len(audio_streams) == 0:
+        return info
+
+    # We just operate on the first audio stream in case there are more
+    stream = audio_streams[0]
+
+    def set_property(stream, prop, value):
+        if prop not in stream or stream[prop] == 0:
+            stream[prop] = value
+
+    for token in extra_info[stream['index']]:
+        m = re.match('([su]([0-9]{1,2})p?) \(([0-9]{1,2}) bit\)$', token)
+        m2 = re.match('([su]([0-9]{1,2})p?)( \(default\))?$', token)
+        if m:
+            set_property(stream, 'sample_fmt', m.group(1))
+            set_property(stream, 'bits_per_sample', int(m.group(2)))
+            set_property(stream, 'bits_per_raw_sample', int(m.group(3)))
+        elif m2:
+            set_property(stream, 'sample_fmt', m2.group(1))
+            set_property(stream, 'bits_per_sample', int(m2.group(2)))
+            set_property(stream, 'bits_per_raw_sample', int(m2.group(2)))
+        elif re.match('(flt)p?( \(default\))?$', token):
+            set_property(stream, 'sample_fmt', token)
+            set_property(stream, 'bits_per_sample', 32)
+            set_property(stream, 'bits_per_raw_sample', 32)
+        elif re.match('(dbl)p?( \(default\))?$', token):
+            set_property(stream, 'sample_fmt', token)
+            set_property(stream, 'bits_per_sample', 64)
+            set_property(stream, 'bits_per_raw_sample', 64)
+    return info
+
+
+def mediainfo(filepath):
+    """Return dictionary with media info(codec, duration, size, bitrate...) from filepath
+    """
+
+    prober = get_prober_name()
+    command_args = [
+        "-v", "quiet",
+        "-show_format",
+        "-show_streams",
+        filepath
+    ]
+
+    command = [prober, '-of', 'old'] + command_args
+    res = Popen(command, stdout=PIPE)
+    output = res.communicate()[0].decode("utf-8")
+
+    if res.returncode != 0:
+        command = [prober] + command_args
+        output = Popen(command, stdout=PIPE).communicate()[0].decode("utf-8")
+
+    rgx = re.compile(r"(?:(?P<inner_dict>.*?):)?(?P<key>.*?)\=(?P<value>.*?)$")
+    info = {}
+
+    if sys.platform == 'win32':
+        output = output.replace("\r", "")
+
+    for line in output.split("\n"):
+        # print(line)
+        mobj = rgx.match(line)
+
+        if mobj:
+            # print(mobj.groups())
+            inner_dict, key, value = mobj.groups()
+
+            if inner_dict:
+                try:
+                    info[inner_dict]
+                except KeyError:
+                    info[inner_dict] = {}
+                info[inner_dict][key] = value
+            else:
+                info[key] = value
+
+    return info
+
+
+def cache_codecs(function):
+    cache = {}
+
+    @wraps(function)
+    def wrapper():
+        try:
+            return cache[0]
+        except:
+            cache[0] = function()
+            return cache[0]
+
+    return wrapper
+
+
+@cache_codecs
+def get_supported_codecs():
+    encoder = get_encoder_name()
+    command = [encoder, "-codecs"]
+    res = Popen(command, stdout=PIPE, stderr=PIPE)
+    output = res.communicate()[0].decode("utf-8")
+    if res.returncode != 0:
+        return []
+
+    if sys.platform == 'win32':
+        output = output.replace("\r", "")
+
+
+    rgx = re.compile(r"^([D.][E.][AVS.][I.][L.][S.]) (\w*) +(.*)")
+    decoders = set()
+    encoders = set()
+    for line in output.split('\n'):
+        match = rgx.match(line.strip())
+        if not match:
+            continue
+        flags, codec, name = match.groups()
+
+        if flags[0] == 'D':
+            decoders.add(codec)
+
+        if flags[1] == 'E':
+            encoders.add(codec)
+
+    return (decoders, encoders)
+
+
+def get_supported_decoders():
+    return get_supported_codecs()[0]
+
+
+def get_supported_encoders():
+    return get_supported_codecs()[1]
+
+def stereo_to_ms(audio_segment):
+	'''
+	Left-Right -> Mid-Side
+	'''
+	channel = audio_segment.split_to_mono()
+	channel = [channel[0].overlay(channel[1]), channel[0].overlay(channel[1].invert_phase())]
+	return AudioSegment.from_mono_audiosegments(channel[0], channel[1])
+
+def ms_to_stereo(audio_segment):
+	'''
+	Mid-Side -> Left-Right
+	'''
+	channel = audio_segment.split_to_mono()
+	channel = [channel[0].overlay(channel[1]) - 3, channel[0].overlay(channel[1].invert_phase()) - 3]
+	return AudioSegment.from_mono_audiosegments(channel[0], channel[1])
+