#!/usr/bin/env python3 # # Copyright (c) 2023 by manyeyes # Copyright (c) 2023 Xiaomi Corporation """ This file demonstrates how to use sherpa-onnx Python API to transcribe file(s) with a non-streaming model. (1) For paraformer ./python-api-examples/offline-decode-files.py \ --tokens=/path/to/tokens.txt \ --paraformer=/path/to/paraformer.onnx \ --num-threads=2 \ --decoding-method=greedy_search \ --debug=false \ --sample-rate=16000 \ --feature-dim=80 \ /path/to/0.wav \ /path/to/1.wav (2) For transducer models from icefall ./python-api-examples/offline-decode-files.py \ --tokens=/path/to/tokens.txt \ --encoder=/path/to/encoder.onnx \ --decoder=/path/to/decoder.onnx \ --joiner=/path/to/joiner.onnx \ --num-threads=2 \ --decoding-method=greedy_search \ --debug=false \ --sample-rate=16000 \ --feature-dim=80 \ /path/to/0.wav \ /path/to/1.wav (3) For CTC models from NeMo python3 ./python-api-examples/offline-decode-files.py \ --tokens=./sherpa-onnx-nemo-ctc-en-citrinet-512/tokens.txt \ --nemo-ctc=./sherpa-onnx-nemo-ctc-en-citrinet-512/model.onnx \ --num-threads=2 \ --decoding-method=greedy_search \ --debug=false \ ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/0.wav \ ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/1.wav \ ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/8k.wav (4) For Whisper models python3 ./python-api-examples/offline-decode-files.py \ --whisper-encoder=./sherpa-onnx-whisper-base.en/base.en-encoder.int8.onnx \ --whisper-decoder=./sherpa-onnx-whisper-base.en/base.en-decoder.int8.onnx \ --tokens=./sherpa-onnx-whisper-base.en/base.en-tokens.txt \ --whisper-task=transcribe \ --num-threads=1 \ ./sherpa-onnx-whisper-base.en/test_wavs/0.wav \ ./sherpa-onnx-whisper-base.en/test_wavs/1.wav \ ./sherpa-onnx-whisper-base.en/test_wavs/8k.wav (5) For CTC models from WeNet python3 ./python-api-examples/offline-decode-files.py \ --wenet-ctc=./sherpa-onnx-zh-wenet-wenetspeech/model.onnx \ --tokens=./sherpa-onnx-zh-wenet-wenetspeech/tokens.txt \ ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/0.wav \ ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/1.wav \ ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/8k.wav (6) For tdnn models of the yesno recipe from icefall python3 ./python-api-examples/offline-decode-files.py \ --sample-rate=8000 \ --feature-dim=23 \ --tdnn-model=./sherpa-onnx-tdnn-yesno/model-epoch-14-avg-2.onnx \ --tokens=./sherpa-onnx-tdnn-yesno/tokens.txt \ ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_0_1_0_0_0_1.wav \ ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_1_0_0_0_1_0.wav \ ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_1_0_0_1_1_1.wav Please refer to https://k2-fsa.github.io/sherpa/onnx/index.html to install sherpa-onnx and to download non-streaming pre-trained models used in this file. """ import argparse import time import wave from pathlib import Path from typing import List, Tuple import numpy as np import sherpa_onnx import soundfile as sf def get_args(): parser = argparse.ArgumentParser( formatter_class=argparse.ArgumentDefaultsHelpFormatter ) parser.add_argument( "--tokens", type=str, help="Path to tokens.txt", ) parser.add_argument( "--hotwords-file", type=str, default="", help=""" The file containing hotwords, one words/phrases per line, like HELLO WORLD 你好世界 """, ) parser.add_argument( "--hotwords-score", type=float, default=1.5, help=""" The hotword score of each token for biasing word/phrase. Used only if --hotwords-file is given. """, ) parser.add_argument( "--modeling-unit", type=str, default="", help=""" The modeling unit of the model, valid values are cjkchar, bpe, cjkchar+bpe. Used only when hotwords-file is given. """, ) parser.add_argument( "--bpe-vocab", type=str, default="", help=""" The path to the bpe vocabulary, the bpe vocabulary is generated by sentencepiece, you can also export the bpe vocabulary through a bpe model by `scripts/export_bpe_vocab.py`. Used only when hotwords-file is given and modeling-unit is bpe or cjkchar+bpe. """, ) parser.add_argument( "--encoder", default="", type=str, help="Path to the encoder model", ) parser.add_argument( "--decoder", default="", type=str, help="Path to the decoder model", ) parser.add_argument( "--joiner", default="", type=str, help="Path to the joiner model", ) parser.add_argument( "--paraformer", default="", type=str, help="Path to the model.onnx from Paraformer", ) parser.add_argument( "--nemo-ctc", default="", type=str, help="Path to the model.onnx from NeMo CTC", ) parser.add_argument( "--wenet-ctc", default="", type=str, help="Path to the model.onnx from WeNet CTC", ) parser.add_argument( "--tdnn-model", default="", type=str, help="Path to the model.onnx for the tdnn model of the yesno recipe", ) parser.add_argument( "--num-threads", type=int, default=1, help="Number of threads for neural network computation", ) parser.add_argument( "--whisper-encoder", default="", type=str, help="Path to whisper encoder model", ) parser.add_argument( "--whisper-decoder", default="", type=str, help="Path to whisper decoder model", ) parser.add_argument( "--whisper-language", default="", type=str, help="""It specifies the spoken language in the input audio file. Example values: en, fr, de, zh, jp. Available languages for multilingual models can be found at https://github.com/openai/whisper/blob/main/whisper/tokenizer.py#L10 If not specified, we infer the language from the input audio file. """, ) parser.add_argument( "--whisper-task", default="transcribe", choices=["transcribe", "translate"], type=str, help="""For multilingual models, if you specify translate, the output will be in English. """, ) parser.add_argument( "--whisper-tail-paddings", default=-1, type=int, help="""Number of tail padding frames. We have removed the 30-second constraint from whisper, so you need to choose the amount of tail padding frames by yourself. Use -1 to use a default value for tail padding. """, ) parser.add_argument( "--blank-penalty", type=float, default=0.0, help=""" The penalty applied on blank symbol during decoding. Note: It is a positive value that would be applied to logits like this `logits[:, 0] -= blank_penalty` (suppose logits.shape is [batch_size, vocab] and blank id is 0). """, ) parser.add_argument( "--decoding-method", type=str, default="greedy_search", help="Valid values are greedy_search and modified_beam_search", ) parser.add_argument( "--debug", type=bool, default=False, help="True to show debug messages", ) parser.add_argument( "--sample-rate", type=int, default=16000, help="""Sample rate of the feature extractor. Must match the one expected by the model. Note: The input sound files can have a different sample rate from this argument.""", ) parser.add_argument( "--feature-dim", type=int, default=80, help="Feature dimension. Must match the one expected by the model", ) parser.add_argument( "sound_files", type=str, nargs="+", help="The input sound file(s) to decode. Each file must be of WAVE" "format with a single channel, and each sample has 16-bit, " "i.e., int16_t. " "The sample rate of the file can be arbitrary and does not need to " "be 16 kHz", ) parser.add_argument( "--name", type=str, default="", help="The directory containing the input sound files to decode", ) parser.add_argument( "--log-dir", type=str, default="", help="The directory containing the input sound files to decode", ) parser.add_argument( "--label", type=str, default=None, help="wav_base_name label", ) return parser.parse_args() def assert_file_exists(filename: str): assert Path(filename).is_file(), ( f"{filename} does not exist!\n" "Please refer to " "https://k2-fsa.github.io/sherpa/onnx/pretrained_models/index.html to download it" ) def read_wave(wave_filename: str) -> Tuple[np.ndarray, int]: """ Args: wave_filename: Path to a wave file. It should be single channel and can be of type 32-bit floating point PCM. Its sample rate does not need to be 24kHz. Returns: Return a tuple containing: - A 1-D array of dtype np.float32 containing the samples, which are normalized to the range [-1, 1]. - Sample rate of the wave file. """ samples, sample_rate = sf.read(wave_filename, dtype="float32") assert ( samples.ndim == 1 ), f"Expected single channel, but got {samples.ndim} channels." samples_float32 = samples.astype(np.float32) return samples_float32, sample_rate def normalize_text_alimeeting(text: str) -> str: """ Text normalization similar to M2MeT challenge baseline. See: https://github.com/yufan-aslp/AliMeeting/blob/main/asr/local/text_normalize.pl """ import re text = text.replace(" ", "") text = text.replace("", "") text = text.replace("<%>", "") text = text.replace("<->", "") text = text.replace("<$>", "") text = text.replace("<#>", "") text = text.replace("<_>", "") text = text.replace("", "") text = text.replace("`", "") text = text.replace("&", "") text = text.replace(",", "") if re.search("[a-zA-Z]", text): text = text.upper() text = text.replace("A", "A") text = text.replace("a", "A") text = text.replace("b", "B") text = text.replace("c", "C") text = text.replace("k", "K") text = text.replace("t", "T") text = text.replace(",", "") text = text.replace("丶", "") text = text.replace("。", "") text = text.replace("、", "") text = text.replace("?", "") return text def main(): args = get_args() assert_file_exists(args.tokens) assert args.num_threads > 0, args.num_threads assert len(args.nemo_ctc) == 0, args.nemo_ctc assert len(args.wenet_ctc) == 0, args.wenet_ctc assert len(args.whisper_encoder) == 0, args.whisper_encoder assert len(args.whisper_decoder) == 0, args.whisper_decoder assert len(args.tdnn_model) == 0, args.tdnn_model assert_file_exists(args.paraformer) recognizer = sherpa_onnx.OfflineRecognizer.from_paraformer( paraformer=args.paraformer, tokens=args.tokens, num_threads=args.num_threads, sample_rate=args.sample_rate, feature_dim=args.feature_dim, decoding_method=args.decoding_method, debug=args.debug, ) print("Started!") start_time = time.time() streams, results = [], [] total_duration = 0 for i, wave_filename in enumerate(args.sound_files): assert_file_exists(wave_filename) samples, sample_rate = read_wave(wave_filename) duration = len(samples) / sample_rate total_duration += duration s = recognizer.create_stream() s.accept_waveform(sample_rate, samples) streams.append(s) if i % 10 == 0: recognizer.decode_streams(streams) results += [s.result.text for s in streams] streams = [] print(f"Processed {i} files") # process the last batch if streams: recognizer.decode_streams(streams) results += [s.result.text for s in streams] end_time = time.time() print("Done!") results_dict = {} for wave_filename, result in zip(args.sound_files, results): print(f"{wave_filename}\n{result}") print("-" * 10) wave_basename = Path(wave_filename).stem results_dict[wave_basename] = result elapsed_seconds = end_time - start_time rtf = elapsed_seconds / total_duration print(f"num_threads: {args.num_threads}") print(f"decoding_method: {args.decoding_method}") print(f"Wave duration: {total_duration:.3f} s") print(f"Elapsed time: {elapsed_seconds:.3f} s") print( f"Real time factor (RTF): {elapsed_seconds:.3f}/{total_duration:.3f} = {rtf:.3f}" ) if args.label: from icefall.utils import store_transcripts, write_error_stats labels_dict = {} with open(args.label, "r") as f: for line in f: # fields = line.strip().split(" ") # fields = [item for item in fields if item] # assert len(fields) == 4 # prompt_text, prompt_audio, text, audio_path = fields fields = line.strip().split("|") fields = [item for item in fields if item] assert len(fields) == 4 audio_path, prompt_text, prompt_audio, text = fields labels_dict[Path(audio_path).stem] = normalize_text_alimeeting(text) final_results = [] for key, value in results_dict.items(): final_results.append((key, labels_dict[key], value)) store_transcripts( filename=f"{args.log_dir}/recogs-{args.name}.txt", texts=final_results ) with open(f"{args.log_dir}/errs-{args.name}.txt", "w") as f: write_error_stats(f, "test-set", final_results, enable_log=True) with open(f"{args.log_dir}/errs-{args.name}.txt", "r") as f: print(f.readline()) # WER print(f.readline()) # Detailed errors if __name__ == "__main__": main()