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# Copyright © 2021 Arm Ltd and Contributors. All rights reserved.
# SPDX-License-Identifier: MIT
"""Automatic speech recognition with PyArmNN demo for processing audio clips to text."""
import sys
import os
import numpy as np
script_dir = os.path.dirname(__file__)
sys.path.insert(1, os.path.join(script_dir, '..', 'common'))
from argparse import ArgumentParser
from network_executor import ArmnnNetworkExecutor
from utils import prepare_input_tensors
from audio_capture import AudioCaptureParams, capture_audio
from audio_utils import decode_text, display_text
from wav2letter_mfcc import Wav2LetterMFCC, W2LAudioPreprocessor
from mfcc import MFCCParams
# Model Specific Labels
labels = {0: 'a', 1: 'b', 2: 'c', 3: 'd', 4: 'e', 5: 'f', 6: 'g', 7: 'h', 8: 'i', 9: 'j', 10: 'k', 11: 'l', 12: 'm',
13: 'n',
14: 'o', 15: 'p', 16: 'q', 17: 'r', 18: 's', 19: 't', 20: 'u', 21: 'v', 22: 'w', 23: 'x', 24: 'y',
25: 'z',
26: "'", 27: ' ', 28: '$'}
def parse_args():
parser = ArgumentParser(description="ASR with PyArmNN")
parser.add_argument(
"--audio_file_path",
required=True,
type=str,
help="Path to the audio file to perform ASR",
)
parser.add_argument(
"--model_file_path",
required=True,
type=str,
help="Path to ASR model to use",
)
parser.add_argument(
"--preferred_backends",
type=str,
nargs="+",
default=["CpuAcc", "CpuRef"],
help="""List of backends in order of preference for optimizing
subgraphs, falling back to the next backend in the list on unsupported
layers. Defaults to [CpuAcc, CpuRef]""",
)
return parser.parse_args()
def main(args):
# Read command line args
audio_file = args.audio_file_path
# Create the ArmNN inference runner
network = ArmnnNetworkExecutor(args.model_file_path, args.preferred_backends)
# Specify model specific audio data requirements
audio_capture_params = AudioCaptureParams(dtype=np.float32, overlap=31712, min_samples=47712, sampling_freq=16000,
mono=True)
buffer = capture_audio(audio_file, audio_capture_params)
# Extract features and create the preprocessor
mfcc_params = MFCCParams(sampling_freq=16000, num_fbank_bins=128, mel_lo_freq=0, mel_hi_freq=8000,
num_mfcc_feats=13, frame_len=512, use_htk_method=False, n_fft=512)
wmfcc = Wav2LetterMFCC(mfcc_params)
preprocessor = W2LAudioPreprocessor(wmfcc, model_input_size=296, stride=160)
current_r_context = ""
is_first_window = True
print("Processing Audio Frames...")
for audio_data in buffer:
# Prepare the input Tensors
input_tensors = prepare_input_tensors(audio_data, network.input_binding_info, preprocessor)
# Run inference
output_result = network.run(input_tensors)
# Slice and Decode the text, and store the right context
current_r_context, text = decode_text(is_first_window, labels, output_result)
is_first_window = False
display_text(text)
print(current_r_context, flush=True)
if __name__ == "__main__":
args = parse_args()
main(args)
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