Export NeMo FastConformer Hybrid Transducer-CTC Large Streaming to ONNX. (#843)
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197
scripts/nemo/fast-conformer-hybrid-transducer-ctc/test-onnx-ctc.py
Executable file
197
scripts/nemo/fast-conformer-hybrid-transducer-ctc/test-onnx-ctc.py
Executable file
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#!/usr/bin/env python3
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import argparse
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from pathlib import Path
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import kaldi_native_fbank as knf
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import numpy as np
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import onnxruntime as ort
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import torch
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import soundfile as sf
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import librosa
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def get_args():
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parser = argparse.ArgumentParser()
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parser.add_argument("--model", type=str, required=True, help="Path to model.onnx")
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parser.add_argument("--tokens", type=str, required=True, help="Path to tokens.txt")
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parser.add_argument("--wav", type=str, required=True, help="Path to test.wav")
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return parser.parse_args()
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def create_fbank():
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opts = knf.FbankOptions()
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opts.frame_opts.dither = 0
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opts.frame_opts.remove_dc_offset = False
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opts.frame_opts.window_type = "hann"
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opts.mel_opts.low_freq = 0
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opts.mel_opts.num_bins = 80
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opts.mel_opts.is_librosa = True
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fbank = knf.OnlineFbank(opts)
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return fbank
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def compute_features(audio, fbank):
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assert len(audio.shape) == 1, audio.shape
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fbank.accept_waveform(16000, audio)
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ans = []
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processed = 0
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while processed < fbank.num_frames_ready:
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ans.append(np.array(fbank.get_frame(processed)))
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processed += 1
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ans = np.stack(ans)
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return ans
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class OnnxModel:
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def __init__(
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self,
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filename: str,
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):
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session_opts = ort.SessionOptions()
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session_opts.inter_op_num_threads = 1
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session_opts.intra_op_num_threads = 1
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self.session_opts = session_opts
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self.model = ort.InferenceSession(
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filename,
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sess_options=self.session_opts,
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providers=["CPUExecutionProvider"],
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)
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meta = self.model.get_modelmeta().custom_metadata_map
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print(meta)
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self.window_size = int(meta["window_size"])
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self.chunk_shift = int(meta["chunk_shift"])
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self.cache_last_channel_dim1 = int(meta["cache_last_channel_dim1"])
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self.cache_last_channel_dim2 = int(meta["cache_last_channel_dim2"])
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self.cache_last_channel_dim3 = int(meta["cache_last_channel_dim3"])
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self.cache_last_time_dim1 = int(meta["cache_last_time_dim1"])
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self.cache_last_time_dim2 = int(meta["cache_last_time_dim2"])
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self.cache_last_time_dim3 = int(meta["cache_last_time_dim3"])
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self.init_cache_state()
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def init_cache_state(self):
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self.cache_last_channel = torch.zeros(
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1,
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self.cache_last_channel_dim1,
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self.cache_last_channel_dim2,
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self.cache_last_channel_dim3,
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dtype=torch.float32,
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).numpy()
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self.cache_last_time = torch.zeros(
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1,
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self.cache_last_time_dim1,
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self.cache_last_time_dim2,
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self.cache_last_time_dim3,
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dtype=torch.float32,
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).numpy()
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self.cache_last_channel_len = torch.ones([1], dtype=torch.int64).numpy()
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def __call__(self, x: np.ndarray):
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# x: (T, C)
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x = torch.from_numpy(x)
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x = x.t().unsqueeze(0)
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# x: [1, C, T]
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x_lens = torch.tensor([x.shape[-1]], dtype=torch.int64)
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(
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log_probs,
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log_probs_len,
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cache_last_channel_next,
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cache_last_time_next,
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cache_last_channel_len_next,
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) = self.model.run(
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[
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self.model.get_outputs()[0].name,
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self.model.get_outputs()[1].name,
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self.model.get_outputs()[2].name,
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self.model.get_outputs()[3].name,
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self.model.get_outputs()[4].name,
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],
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{
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self.model.get_inputs()[0].name: x.numpy(),
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self.model.get_inputs()[1].name: x_lens.numpy(),
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self.model.get_inputs()[2].name: self.cache_last_channel,
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self.model.get_inputs()[3].name: self.cache_last_time,
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self.model.get_inputs()[4].name: self.cache_last_channel_len,
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},
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)
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self.cache_last_channel = cache_last_channel_next
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self.cache_last_time = cache_last_time_next
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self.cache_last_channel_len = cache_last_channel_len_next
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# [T, vocab_size]
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return torch.from_numpy(log_probs).squeeze(0)
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def main():
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args = get_args()
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assert Path(args.model).is_file(), args.model
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assert Path(args.tokens).is_file(), args.tokens
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assert Path(args.wav).is_file(), args.wav
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print(vars(args))
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model = OnnxModel(args.model)
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id2token = dict()
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with open(args.tokens, encoding="utf-8") as f:
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for line in f:
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t, idx = line.split()
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id2token[int(idx)] = t
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fbank = create_fbank()
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audio, sample_rate = sf.read(args.wav, dtype="float32", always_2d=True)
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audio = audio[:, 0] # only use the first channel
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if sample_rate != 16000:
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audio = librosa.resample(
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audio,
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orig_sr=sample_rate,
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target_sr=16000,
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)
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sample_rate = 16000
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window_size = model.window_size
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chunk_shift = model.chunk_shift
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blank = len(id2token) - 1
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prev = -1
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ans = []
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features = compute_features(audio, fbank)
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num_chunks = (features.shape[0] - window_size) // chunk_shift + 1
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for i in range(num_chunks):
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start = i * chunk_shift
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end = start + window_size
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chunk = features[start:end, :]
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log_probs = model(chunk)
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ids = torch.argmax(log_probs, dim=1).tolist()
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for i in ids:
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if i != blank and i != prev:
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ans.append(i)
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prev = i
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tokens = [id2token[i] for i in ans]
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underline = "▁"
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# underline = b"\xe2\x96\x81".decode()
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text = "".join(tokens).replace(underline, " ").strip()
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print(args.wav)
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print(text)
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main()
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