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82 lines
2.8 KiB
82 lines
2.8 KiB
import json
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import pathlib
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import numpy as np
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import librosa
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import soundfile
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"""
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The dataset has to be downloaded manually from https://www.openslr.org/12/ and put in `extra/datasets/librispeech`.
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For mlperf validation the dev-clean dataset is used.
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Then all the flacs have to be converted to wav using something like:
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```fish
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for file in $(find * | grep flac); do ffmpeg -i $file -ar 16k "$(dirname $file)/$(basename $file .flac).wav"; done
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```
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Then this [file](https://github.com/mlcommons/inference/blob/master/speech_recognition/rnnt/dev-clean-wav.json) has to also be put in `extra/datasets/librispeech`.
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"""
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BASEDIR = pathlib.Path(__file__).parent / "librispeech"
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with open(BASEDIR / "dev-clean-wav.json") as f:
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ci = json.load(f)
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FILTER_BANK = np.expand_dims(librosa.filters.mel(sr=16000, n_fft=512, n_mels=80, fmin=0, fmax=8000), 0)
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WINDOW = librosa.filters.get_window("hann", 320)
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def feature_extract(x, x_lens):
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x_lens = np.ceil((x_lens / 160) / 3).astype(np.int32)
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# pre-emphasis
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x = np.concatenate((np.expand_dims(x[:, 0], 1), x[:, 1:] - 0.97 * x[:, :-1]), axis=1)
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# stft
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x = librosa.stft(x, n_fft=512, window=WINDOW, hop_length=160, win_length=320, center=True, pad_mode="reflect")
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x = np.stack((x.real, x.imag), axis=-1)
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# power spectrum
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x = (x**2).sum(-1)
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# mel filter bank
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x = np.matmul(FILTER_BANK, x)
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# log
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x = np.log(x + 1e-20)
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# feature splice
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seq = [x]
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for i in range(1, 3):
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tmp = np.zeros_like(x)
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tmp[:, :, :-i] = x[:, :, i:]
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seq.append(tmp)
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features = np.concatenate(seq, axis=1)[:, :, ::3]
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# normalize
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features_mean = np.zeros((features.shape[0], features.shape[1]), dtype=np.float32)
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features_std = np.zeros((features.shape[0], features.shape[1]), dtype=np.float32)
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for i in range(features.shape[0]):
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features_mean[i, :] = features[i, :, :x_lens[i]].mean(axis=1)
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features_std[i, :] = features[i, :, :x_lens[i]].std(axis=1, ddof=1)
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features_std += 1e-5
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features = (features - np.expand_dims(features_mean, 2)) / np.expand_dims(features_std, 2)
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return features.transpose(2, 0, 1), x_lens.astype(np.float32)
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def load_wav(file):
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sample = soundfile.read(file)[0].astype(np.float32)
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return sample, sample.shape[0]
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def iterate(bs=1, start=0):
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print(f"there are {len(ci)} samples in the dataset")
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for i in range(start, len(ci), bs):
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samples, sample_lens = zip(*[load_wav(BASEDIR / v["files"][0]["fname"]) for v in ci[i : i + bs]])
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samples = list(samples)
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# pad to same length
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max_len = max(sample_lens)
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for j in range(len(samples)):
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samples[j] = np.pad(samples[j], (0, max_len - sample_lens[j]), "constant")
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samples, sample_lens = np.array(samples), np.array(sample_lens)
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yield feature_extract(samples, sample_lens), np.array([v["transcript"] for v in ci[i : i + bs]])
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if __name__ == "__main__":
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X, Y = next(iterate())
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print(X[0].shape, Y.shape)
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