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#!/usr/bin/env python | |
# -*- coding: utf-8 -*- | |
from __future__ import absolute_import, division, print_function | |
import argparse | |
import numpy as np | |
import shlex | |
import subprocess | |
import sys | |
import wave | |
import json | |
from deepspeech import Model, version | |
from timeit import default_timer as timer | |
try: | |
from shhlex import quote | |
except ImportError: | |
from pipes import quote | |
def convert_samplerate(audio_path, desired_sample_rate): | |
sox_cmd = 'sox {} --type raw --bits 16 --channels 1 --rate {} --encoding signed-integer --endian little --compression 0.0 --no-dither - '.format( | |
quote(audio_path), desired_sample_rate) | |
try: | |
output = subprocess.check_output( | |
shlex.split(sox_cmd), stderr=subprocess.PIPE) | |
except subprocess.CalledProcessError as e: | |
raise RuntimeError('SoX returned non-zero status: {}'.format(e.stderr)) | |
except OSError as e: | |
raise OSError(e.errno, 'SoX not found, use {}hz files or install it: {}'.format( | |
desired_sample_rate, e.strerror)) | |
return desired_sample_rate, np.frombuffer(output, np.int16) | |
def metadata_to_string(metadata): | |
return ''.join(token.text for token in metadata.tokens) | |
def words_from_candidate_transcript(metadata): | |
word = "" | |
word_list = [] | |
word_start_time = 0 | |
# Loop through each character | |
for i, token in enumerate(metadata.tokens): | |
# Append character to word if it's not a space | |
if token.text != " ": | |
if len(word) == 0: | |
# Log the start time of the new word | |
word_start_time = token.start_time | |
word = word + token.text | |
# Word boundary is either a space or the last character in the array | |
if token.text == " " or i == len(metadata.tokens) - 1: | |
word_duration = token.start_time - word_start_time | |
if word_duration < 0: | |
word_duration = 0 | |
each_word = dict() | |
each_word["word"] = word | |
each_word["start_time "] = round(word_start_time, 4) | |
each_word["duration"] = round(word_duration, 4) | |
word_list.append(each_word) | |
# Reset | |
word = "" | |
word_start_time = 0 | |
return word_list | |
def metadata_json_output(metadata): | |
json_result = dict() | |
json_result["transcripts"] = [{ | |
"confidence": transcript.confidence, | |
"words": words_from_candidate_transcript(transcript), | |
} for transcript in metadata.transcripts] | |
return json.dumps(json_result, indent=2) | |
class VersionAction(argparse.Action): | |
def __init__(self, *args, **kwargs): | |
super(VersionAction, self).__init__(nargs=0, *args, **kwargs) | |
def __call__(self, *args, **kwargs): | |
print('DeepSpeech ', version()) | |
exit(0) | |
def client(audio_file, lang="uk"): | |
model_load_start = timer() | |
# sphinx-doc: python_ref_model_start | |
model_path = "uk.tflite" | |
if lang not in ["en", "uk"]: | |
lang = "uk" | |
if lang == "uk": | |
model_path = "./uk.tflite" | |
if lang == "en": | |
model_path = "./deepspeech-0.7.3-models.tflite" | |
print(lang) | |
ds = Model(model_path) | |
# sphinx-doc: python_ref_model_stop | |
model_load_end = timer() - model_load_start | |
print('Loaded model in {:.3}s.'.format(model_load_end), file=sys.stderr) | |
desired_sample_rate = ds.sampleRate() | |
fin = wave.open(audio_file, 'rb') | |
fs_orig = fin.getframerate() | |
audio = np.frombuffer(fin.readframes(fin.getnframes()), np.int16) | |
audio_length = fin.getnframes() * (1/fs_orig) | |
fin.close() | |
print('Running inference.', file=sys.stderr) | |
inference_start = timer() | |
# sphinx-doc: python_ref_inference_start | |
result = ds.stt(audio) | |
print(result) | |
# sphinx-doc: python_ref_inference_stop | |
inference_end = timer() - inference_start | |
print('Inference took %0.3fs for %0.3fs audio file.' % | |
(inference_end, audio_length), file=sys.stderr) | |
return result | |