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enginex-mr_series-sherpa-onnx/scripts/nemo/fast-conformer-hybrid-transducer-ctc/test-onnx-ctc.py
2024-05-10 16:26:43 +08:00

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Python
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#!/usr/bin/env python3
# Copyright 2024 Xiaomi Corp. (authors: Fangjun Kuang)
import argparse
from pathlib import Path
import kaldi_native_fbank as knf
import numpy as np
import onnxruntime as ort
import torch
import soundfile as sf
import librosa
def get_args():
parser = argparse.ArgumentParser()
parser.add_argument("--model", type=str, required=True, help="Path to model.onnx")
parser.add_argument("--tokens", type=str, required=True, help="Path to tokens.txt")
parser.add_argument("--wav", type=str, required=True, help="Path to test.wav")
return parser.parse_args()
def create_fbank():
opts = knf.FbankOptions()
opts.frame_opts.dither = 0
opts.frame_opts.remove_dc_offset = False
opts.frame_opts.window_type = "hann"
opts.mel_opts.low_freq = 0
opts.mel_opts.num_bins = 80
opts.mel_opts.is_librosa = True
fbank = knf.OnlineFbank(opts)
return fbank
def compute_features(audio, fbank):
assert len(audio.shape) == 1, audio.shape
fbank.accept_waveform(16000, audio)
ans = []
processed = 0
while processed < fbank.num_frames_ready:
ans.append(np.array(fbank.get_frame(processed)))
processed += 1
ans = np.stack(ans)
return ans
class OnnxModel:
def __init__(
self,
filename: str,
):
session_opts = ort.SessionOptions()
session_opts.inter_op_num_threads = 1
session_opts.intra_op_num_threads = 1
self.session_opts = session_opts
self.model = ort.InferenceSession(
filename,
sess_options=self.session_opts,
providers=["CPUExecutionProvider"],
)
meta = self.model.get_modelmeta().custom_metadata_map
print(meta)
self.window_size = int(meta["window_size"])
self.chunk_shift = int(meta["chunk_shift"])
self.cache_last_channel_dim1 = int(meta["cache_last_channel_dim1"])
self.cache_last_channel_dim2 = int(meta["cache_last_channel_dim2"])
self.cache_last_channel_dim3 = int(meta["cache_last_channel_dim3"])
self.cache_last_time_dim1 = int(meta["cache_last_time_dim1"])
self.cache_last_time_dim2 = int(meta["cache_last_time_dim2"])
self.cache_last_time_dim3 = int(meta["cache_last_time_dim3"])
self.init_cache_state()
def init_cache_state(self):
self.cache_last_channel = torch.zeros(
1,
self.cache_last_channel_dim1,
self.cache_last_channel_dim2,
self.cache_last_channel_dim3,
dtype=torch.float32,
).numpy()
self.cache_last_time = torch.zeros(
1,
self.cache_last_time_dim1,
self.cache_last_time_dim2,
self.cache_last_time_dim3,
dtype=torch.float32,
).numpy()
self.cache_last_channel_len = torch.zeros([1], dtype=torch.int64).numpy()
def __call__(self, x: np.ndarray):
# x: (T, C)
x = torch.from_numpy(x)
x = x.t().unsqueeze(0)
# x: [1, C, T]
x_lens = torch.tensor([x.shape[-1]], dtype=torch.int64)
(
log_probs,
log_probs_len,
cache_last_channel_next,
cache_last_time_next,
cache_last_channel_len_next,
) = self.model.run(
[
self.model.get_outputs()[0].name,
self.model.get_outputs()[1].name,
self.model.get_outputs()[2].name,
self.model.get_outputs()[3].name,
self.model.get_outputs()[4].name,
],
{
self.model.get_inputs()[0].name: x.numpy(),
self.model.get_inputs()[1].name: x_lens.numpy(),
self.model.get_inputs()[2].name: self.cache_last_channel,
self.model.get_inputs()[3].name: self.cache_last_time,
self.model.get_inputs()[4].name: self.cache_last_channel_len,
},
)
self.cache_last_channel = cache_last_channel_next
self.cache_last_time = cache_last_time_next
self.cache_last_channel_len = cache_last_channel_len_next
# [T, vocab_size]
return torch.from_numpy(log_probs).squeeze(0)
def main():
args = get_args()
assert Path(args.model).is_file(), args.model
assert Path(args.tokens).is_file(), args.tokens
assert Path(args.wav).is_file(), args.wav
print(vars(args))
model = OnnxModel(args.model)
id2token = dict()
with open(args.tokens, encoding="utf-8") as f:
for line in f:
t, idx = line.split()
id2token[int(idx)] = t
fbank = create_fbank()
audio, sample_rate = sf.read(args.wav, dtype="float32", always_2d=True)
audio = audio[:, 0] # only use the first channel
if sample_rate != 16000:
audio = librosa.resample(
audio,
orig_sr=sample_rate,
target_sr=16000,
)
sample_rate = 16000
window_size = model.window_size
chunk_shift = model.chunk_shift
blank = len(id2token) - 1
prev = -1
ans = []
features = compute_features(audio, fbank)
num_chunks = (features.shape[0] - window_size) // chunk_shift + 1
for i in range(num_chunks):
start = i * chunk_shift
end = start + window_size
chunk = features[start:end, :]
log_probs = model(chunk)
ids = torch.argmax(log_probs, dim=1).tolist()
for i in ids:
if i != blank and i != prev:
ans.append(i)
prev = i
tokens = [id2token[i] for i in ans]
underline = ""
# underline = b"\xe2\x96\x81".decode()
text = "".join(tokens).replace(underline, " ").strip()
print(args.wav)
print(text)
main()