import numpy as np from . import wavfile import warnings import torch import librosa def time_to_x_coords(time_in_file, sampling_rate, fft_win_length, fft_overlap): nfft = np.floor(fft_win_length*sampling_rate) # int() uses floor noverlap = np.floor(fft_overlap*nfft) return (time_in_file*sampling_rate-noverlap) / (nfft - noverlap) # NOTE this is also defined in post_process def x_coords_to_time(x_pos, sampling_rate, fft_win_length, fft_overlap): nfft = np.floor(fft_win_length*sampling_rate) noverlap = np.floor(fft_overlap*nfft) return ((x_pos*(nfft - noverlap)) + noverlap) / sampling_rate #return (1.0 - fft_overlap) * fft_win_length * (x_pos + 0.5) # 0.5 is for center of temporal window def generate_spectrogram(audio, sampling_rate, params, return_spec_for_viz=False, check_spec_size=True): # generate spectrogram spec = gen_mag_spectrogram(audio, sampling_rate, params['fft_win_length'], params['fft_overlap']) # crop to min/max freq max_freq = round(params['max_freq']*params['fft_win_length']) min_freq = round(params['min_freq']*params['fft_win_length']) if spec.shape[0] < max_freq: freq_pad = max_freq - spec.shape[0] spec = np.vstack((np.zeros((freq_pad, spec.shape[1]), dtype=spec.dtype), spec)) spec_cropped = spec[-max_freq:spec.shape[0]-min_freq, :] if params['spec_scale'] == 'log': log_scaling = 2.0 * (1.0 / sampling_rate) * (1.0/(np.abs(np.hanning(int(params['fft_win_length']*sampling_rate)))**2).sum()) #log_scaling = (1.0 / sampling_rate)*0.1 #log_scaling = (1.0 / sampling_rate)*10e4 spec = np.log1p(log_scaling*spec_cropped) elif params['spec_scale'] == 'pcen': spec = pcen(spec_cropped, sampling_rate) elif params['spec_scale'] == 'none': pass if params['denoise_spec_avg']: spec = spec - np.mean(spec, 1)[:, np.newaxis] spec.clip(min=0, out=spec) if params['max_scale_spec']: spec = spec / (spec.max() + 10e-6) # needs to be divisible by specific factor - if not it should have been padded #if check_spec_size: #assert((int(spec.shape[0]*params['resize_factor']) % params['spec_divide_factor']) == 0) #assert((int(spec.shape[1]*params['resize_factor']) % params['spec_divide_factor']) == 0) # for visualization purposes - use log scaled spectrogram if return_spec_for_viz: log_scaling = 2.0 * (1.0 / sampling_rate) * (1.0/(np.abs(np.hanning(int(params['fft_win_length']*sampling_rate)))**2).sum()) spec_for_viz = np.log1p(log_scaling*spec_cropped).astype(np.float32) else: spec_for_viz = None return spec, spec_for_viz def load_audio_file(audio_file, time_exp_fact, target_samp_rate, scale=False): with warnings.catch_warnings(): warnings.filterwarnings('ignore', category=wavfile.WavFileWarning) #sampling_rate, audio_raw = wavfile.read(audio_file) audio_raw, sampling_rate = librosa.load(audio_file, sr=None) if len(audio_raw.shape) > 1: raise Exception('Currently does not handle stereo files') sampling_rate = sampling_rate * time_exp_fact # resample - need to do this after correcting for time expansion sampling_rate_old = sampling_rate sampling_rate = target_samp_rate audio_raw = librosa.resample(audio_raw, orig_sr=sampling_rate_old, target_sr=sampling_rate, res_type='polyphase') # convert to float32 and scale audio_raw = audio_raw.astype(np.float32) if scale: audio_raw = audio_raw - audio_raw.mean() audio_raw = audio_raw / (np.abs(audio_raw).max() + 10e-6) return sampling_rate, audio_raw def pad_audio(audio_raw, fs, ms, overlap_perc, resize_factor, divide_factor, fixed_width=None): # Adds zeros to the end of the raw data so that the generated sepctrogram # will be evenly divisible by `divide_factor` # Also deals with very short audio clips and fixed_width during training # This code could be clearer, clean up nfft = int(ms*fs) noverlap = int(overlap_perc*nfft) step = nfft - noverlap min_size = int(divide_factor*(1.0/resize_factor)) spec_width = ((audio_raw.shape[0]-noverlap)//step) spec_width_rs = spec_width * resize_factor if fixed_width is not None and spec_width < fixed_width: # too small # used during training to ensure all the batches are the same size diff = fixed_width*step + noverlap - audio_raw.shape[0] audio_raw = np.hstack((audio_raw, np.zeros(diff, dtype=audio_raw.dtype))) elif fixed_width is not None and spec_width > fixed_width: # too big # used during training to ensure all the batches are the same size diff = fixed_width*step + noverlap - audio_raw.shape[0] audio_raw = audio_raw[:diff] elif spec_width_rs < min_size or (np.floor(spec_width_rs) % divide_factor) != 0: # need to be at least min_size div_amt = np.ceil(spec_width_rs / float(divide_factor)) div_amt = np.maximum(1, div_amt) target_size = int(div_amt*divide_factor*(1.0/resize_factor)) diff = target_size*step + noverlap - audio_raw.shape[0] audio_raw = np.hstack((audio_raw, np.zeros(diff, dtype=audio_raw.dtype))) return audio_raw def gen_mag_spectrogram(x, fs, ms, overlap_perc): # Computes magnitude spectrogram by specifying time. x = x.astype(np.float32) nfft = int(ms*fs) noverlap = int(overlap_perc*nfft) # window data step = nfft - noverlap # compute spec spec, _ = librosa.core.spectrum._spectrogram(y=x, power=1, n_fft=nfft, hop_length=step, center=False) # remove DC component and flip vertical orientation spec = np.flipud(spec[1:, :]) return spec.astype(np.float32) def gen_mag_spectrogram_pt(x, fs, ms, overlap_perc): nfft = int(ms*fs) nstep = round((1.0-overlap_perc)*nfft) han_win = torch.hann_window(nfft, periodic=False).to(x.device) complex_spec = torch.stft(x, nfft, nstep, window=han_win, center=False) spec = complex_spec.pow(2.0).sum(-1) # remove DC component and flip vertically spec = torch.flipud(spec[0, 1:,:]) return spec def pcen(spec_cropped, sampling_rate): # TODO should be passing hop_length too i.e. step spec = librosa.pcen(spec_cropped * (2**31), sr=sampling_rate/10).astype(np.float32) return spec