1 files changed, 341 insertions, 0 deletions
diff --git a/venv/lib/python3.9/site-packages/pydub/effects.py b/venv/lib/python3.9/site-packages/pydub/effects.py
new file mode 100644
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--- /dev/null
+++ b/venv/lib/python3.9/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})