mirror of
https://github.com/storytold/storyteller-ml.git
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174 lines
6.5 KiB
Python
174 lines
6.5 KiB
Python
# Copyright (c) 2020, NVIDIA CORPORATION. All rights reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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import logging
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import logging.handlers
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import multiprocessing
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import os
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from pathlib import PosixPath
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from typing import List, Tuple, Union
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import ctc_segmentation as cs
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import numpy as np
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def get_segments(
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log_probs: np.ndarray,
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path_wav: Union[PosixPath, str],
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transcript_file: Union[PosixPath, str],
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output_file: str,
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vocabulary: List[str],
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window_size: int = 8000,
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frame_duration_ms: int = 20,
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) -> None:
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"""
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Segments the audio into segments and saves segments timings to a file
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Args:
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log_probs: Log probabilities for the original audio from an ASR model, shape T * |vocabulary|.
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values for blank should be at position 0
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path_wav: path to the audio .wav file
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transcript_file: path to
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output_file: path to the file to save timings for segments
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vocabulary: vocabulary used to train the ASR model, note blank is at position 0
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window_size: the length of each utterance (in terms of frames of the CTC outputs) fits into that window.
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frame_duration_ms: frame duration in ms
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"""
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config = cs.CtcSegmentationParameters()
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config.char_list = vocabulary
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config.min_window_size = window_size
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config.frame_duration_ms = frame_duration_ms
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config.blank = config.space
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config.subsampling_factor = 2
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with open(transcript_file, "r") as f:
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text = f.readlines()
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text = [t.strip() for t in text if t.strip()]
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# add corresponding original text without pre-processing
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transcript_file_no_preprocessing = transcript_file.replace('.txt', '_with_punct.txt')
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if not os.path.exists(transcript_file_no_preprocessing):
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raise ValueError(f'{transcript_file_no_preprocessing} not found.')
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with open(transcript_file_no_preprocessing, "r") as f:
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text_no_preprocessing = f.readlines()
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text_no_preprocessing = [t.strip() for t in text_no_preprocessing if t.strip()]
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# add corresponding normalized original text
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transcript_file_normalized = transcript_file.replace('.txt', '_with_punct_normalized.txt')
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if not os.path.exists(transcript_file_normalized):
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raise ValueError(f'{transcript_file_normalized} not found.')
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with open(transcript_file_normalized, "r") as f:
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text_normalized = f.readlines()
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text_normalized = [t.strip() for t in text_normalized if t.strip()]
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if len(text_no_preprocessing) != len(text):
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raise ValueError(f'{transcript_file} and {transcript_file_no_preprocessing} do not match')
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if len(text_normalized) != len(text):
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raise ValueError(f'{transcript_file} and {transcript_file_normalized} do not match')
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ground_truth_mat, utt_begin_indices = cs.prepare_text(config, text)
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logging.debug(f"Syncing {transcript_file}")
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logging.debug(
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f"Audio length {os.path.basename(path_wav)}: {log_probs.shape[0]}. "
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f"Text length {os.path.basename(transcript_file)}: {len(ground_truth_mat)}"
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)
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timings, char_probs, char_list = cs.ctc_segmentation(config, log_probs, ground_truth_mat)
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segments = cs.determine_utterance_segments(config, utt_begin_indices, char_probs, timings, text)
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write_output(output_file, path_wav, segments, text, text_no_preprocessing, text_normalized)
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def write_output(
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out_path: str,
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path_wav: str,
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segments: List[Tuple[float]],
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text: str,
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text_no_preprocessing: str,
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text_normalized: str,
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stride: int = 2,
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):
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"""
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Write the segmentation output to a file
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out_path: Path to output file
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path_wav: Path to the original audio file
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segments: Segments include start, end and alignment score
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text: Text used for alignment
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text_no_preprocessing: Reference txt without any pre-processing
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text_normalized: Reference text normalized
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stride: Stride applied to an ASR input
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"""
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# Uses char-wise alignments to get utterance-wise alignments and writes them into the given file
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with open(str(out_path), "w") as outfile:
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outfile.write(str(path_wav) + "\n")
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for i, (start, end, score) in enumerate(segments):
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outfile.write(
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f'{start/stride} {end/stride} {score} | {text[i]} | {text_no_preprocessing[i]} | {text_normalized[i]}\n'
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)
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#####################
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# logging utils
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#####################
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def listener_configurer(log_file, level):
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root = logging.getLogger()
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h = logging.handlers.RotatingFileHandler(log_file, 'w')
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f = logging.Formatter('%(asctime)s %(processName)-10s %(name)s %(levelname)-8s %(message)s')
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h.setFormatter(f)
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ch = logging.StreamHandler()
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root.addHandler(h)
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root.setLevel(level)
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root.addHandler(ch)
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def listener_process(queue, configurer, log_file, level):
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configurer(log_file, level)
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while True:
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try:
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record = queue.get()
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if record is None: # We send this as a sentinel to tell the listener to quit.
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break
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logger = logging.getLogger(record.name)
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logger.setLevel(logging.INFO)
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logger.handle(record) # No level or filter logic applied - just do it!
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except Exception:
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import sys
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import traceback
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print('Problem:', file=sys.stderr)
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traceback.print_exc(file=sys.stderr)
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def worker_configurer(queue, level):
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h = logging.handlers.QueueHandler(queue) # Just the one handler needed
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root = logging.getLogger()
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root.addHandler(h)
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root.setLevel(level)
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def worker_process(
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queue, configurer, level, log_probs, path_wav, transcript_file, output_file, vocabulary, window_len
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):
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configurer(queue, level)
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name = multiprocessing.current_process().name
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innerlogger = logging.getLogger('worker')
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innerlogger.info(f'{name} is processing {path_wav}, window_len={window_len}')
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get_segments(log_probs, path_wav, transcript_file, output_file, vocabulary, window_len)
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innerlogger.info(f'{name} completed segmentation of {path_wav}, segments saved to {output_file}')
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