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@ -5,7 +5,14 @@ import math
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import random
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import pyrubberband
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import math
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#import noisereduce as nr
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import platform
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OPERATING_SYSTEM = platform.system()
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if OPERATING_SYSTEM == 'Windows':
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wav_resolution = "sinc_fastest"
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else:
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wav_resolution = "polyphase"
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MAX_SPEC = 'Max Spec'
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MIN_SPEC = 'Min Spec'
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@ -189,7 +196,7 @@ def reduce_vocal_aggressively(X, y, softmask):
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return y_mag * np.exp(1.j * np.angle(y))
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def merge_artifacts(y_mask, thres=0.05, min_range=64, fade_size=32):
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def merge_artifacts(y_mask, thres=0.01, min_range=64, fade_size=32):
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if min_range < fade_size * 2:
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raise ValueError('min_range must be >= fade_size * 2')
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@ -224,7 +231,7 @@ def merge_artifacts(y_mask, thres=0.05, min_range=64, fade_size=32):
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return y_mask
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def mask_silence(mag, ref, thres=0.2, min_range=64, fade_size=32):
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def mask_silence(mag, ref, thres=0.1, min_range=64, fade_size=32):
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if min_range < fade_size * 2:
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raise ValueError('min_range must be >= fade_area * 2')
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@ -329,12 +336,12 @@ def cmb_spectrogram_to_wave(spec_m, mp, extra_bins_h=None, extra_bins=None):
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sr = mp.param['band'][d+1]['sr']
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if d == 1: # lower
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spec_s = fft_lp_filter(spec_s, bp['lpf_start'], bp['lpf_stop'])
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wave = librosa.resample(spectrogram_to_wave(spec_s, bp['hl'], mp.param['mid_side'], mp.param['mid_side_b2'], mp.param['reverse']), bp['sr'], sr, res_type="sinc_fastest")
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wave = librosa.resample(spectrogram_to_wave(spec_s, bp['hl'], mp.param['mid_side'], mp.param['mid_side_b2'], mp.param['reverse']), bp['sr'], sr, res_type=wav_resolution)
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else: # mid
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spec_s = fft_hp_filter(spec_s, bp['hpf_start'], bp['hpf_stop'] - 1)
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spec_s = fft_lp_filter(spec_s, bp['lpf_start'], bp['lpf_stop'])
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wave2 = np.add(wave, spectrogram_to_wave(spec_s, bp['hl'], mp.param['mid_side'], mp.param['mid_side_b2'], mp.param['reverse']))
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wave = librosa.resample(wave2, bp['sr'], sr, res_type="sinc_fastest")
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wave = librosa.resample(wave2, bp['sr'], sr, res_type=wav_resolution)
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return wave
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@ -460,30 +467,6 @@ def wave_to_spectrogram_no_mp(wave):
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return spec
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# def noise_reduction(audio_file):
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# noise_pro = 'noise_pro.wav'
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# wav, sr = librosa.load(audio_file, sr=44100, mono=False)
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# wav_noise, noise_rate = librosa.load(noise_pro, sr=44100, mono=False)
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# if wav.ndim == 1:
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# wav = np.asfortranarray([wav,wav])
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# wav_1 = nr.reduce_noise(audio_clip=wav[0], noise_clip=wav_noise, verbose=True)
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# wav_2 = nr.reduce_noise(audio_clip=wav[1], noise_clip=wav_noise, verbose=True)
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# if wav_1.shape > wav_2.shape:
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# wav_2 = to_shape(wav_2, wav_1.shape)
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# if wav_1.shape < wav_2.shape:
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# wav_1 = to_shape(wav_1, wav_2.shape)
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# #print('wav_1.shape: ', wav_1.shape)
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# wav_mix = np.asfortranarray([wav_1, wav_2])
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# return wav_mix, sr
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def invert_audio(specs, invert_p=True):
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ln = min([specs[0].shape[2], specs[1].shape[2]])
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@ -518,8 +501,6 @@ def ensembling(a, specs):
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spec = spec[:,:,:ln]
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specs[i] = specs[i][:,:,:ln]
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#print('spec: ', a)
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if MIN_SPEC == a:
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spec = np.where(np.abs(specs[i]) <= np.abs(spec), specs[i], spec)
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if MAX_SPEC == a:
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@ -531,8 +512,6 @@ def ensembling(a, specs):
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def ensemble_inputs(audio_input, algorithm, is_normalization, wav_type_set, save_path):
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#print(algorithm)
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if algorithm == AVERAGE:
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output = average_audio(audio_input)
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samplerate = 44100
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@ -543,9 +522,6 @@ def ensemble_inputs(audio_input, algorithm, is_normalization, wav_type_set, save
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wave, samplerate = librosa.load(audio_input[i], mono=False, sr=44100)
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spec = wave_to_spectrogram_no_mp(wave)
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specs.append(spec)
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#print('output size: ', sys.getsizeof(spec))
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#print('output size: ', sys.getsizeof(specs))
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output = spectrogram_to_wave_no_mp(ensembling(algorithm, specs))
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@ -572,8 +548,6 @@ def to_shape_minimize(x: np.ndarray, target_shape):
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def augment_audio(export_path, audio_file, rate, is_normalization, wav_type_set, save_format=None, is_pitch=False):
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#print(rate)
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wav, sr = librosa.load(audio_file, sr=44100, mono=False)
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if wav.ndim == 1:
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