Merge remote-tracking branch 'refs/remotes/origin/main'

update contextual forward

egs/alimeeting/modular_sa_asr/local/make_textgrid_rttm.py

@@ -1,7 +1,10 @@
 import argparse
 import tqdm
 import codecs
-import textgrid
+try:
+    import textgrid
+except:
+    raise "Please install textgrid firstly: pip install textgrid"
 import pdb
 
 class Segment(object):

egs/alimeeting/modular_sa_asr/local/meeting_speaker_number_process.py

@@ -6,7 +6,10 @@ import argparse
 import codecs
 from distutils.util import strtobool
 from pathlib import Path
-import textgrid
+try:
+    import textgrid
+except:
+    raise "Please install textgrid firstly: pip install textgrid"
 import pdb
 
 class Segment(object):

egs/alimeeting/sa_asr/local/alimeeting_process_overlap_force.py

@@ -6,7 +6,10 @@ import argparse
 import codecs
 from distutils.util import strtobool
 from pathlib import Path
-import textgrid
+try:
+    import textgrid
+except:
+    raise "Please install textgrid firstly: pip install textgrid"
 import pdb
 
 class Segment(object):

egs/alimeeting/sa_asr/local/alimeeting_process_textgrid.py

@@ -6,7 +6,10 @@ import argparse
 import codecs
 from distutils.util import strtobool
 from pathlib import Path
-import textgrid
+try:
+    import textgrid
+except:
+    raise "Please install textgrid firstly: pip install textgrid"
 import pdb
 
 class Segment(object):

egs/alimeeting/sa_asr/local/process_sot_fifo_textchar2spk.py

@@ -6,7 +6,10 @@ import argparse
 import codecs
 from distutils.util import strtobool
 from pathlib import Path
-import textgrid
+try:
+    import textgrid
+except:
+    raise "Please install textgrid firstly: pip install textgrid"
 import pdb
 
 def get_args():

egs/alimeeting/sa_asr/local/process_textgrid_to_single_speaker_wav.py

@@ -6,7 +6,12 @@ import argparse
 import codecs
 from distutils.util import strtobool
 from pathlib import Path
-import textgrid
+
+try:
+    import textgrid
+except:
+    raise "Please install textgrid firstly: pip install textgrid"
+
 import pdb
 import numpy as np
 import sys

funasr/bin/asr_infer.py

@@ -34,8 +34,8 @@ from funasr.modules.beam_search.beam_search_transducer import Hypothesis as Hypo
 from funasr.modules.scorers.ctc import CTCPrefixScorer
 from funasr.modules.scorers.length_bonus import LengthBonus
 from funasr.build_utils.build_asr_model import frontend_choices
-from funasr.text.build_tokenizer import build_tokenizer
-from funasr.text.token_id_converter import TokenIDConverter
+from funasr.tokenizer.build_tokenizer import build_tokenizer
+from funasr.tokenizer.token_id_converter import TokenIDConverter
 from funasr.torch_utils.device_funcs import to_device
 from funasr.utils.timestamp_tools import ts_prediction_lfr6_standard
 
@@ -44,9 +44,9 @@ class Speech2Text:
     """Speech2Text class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech2text = Speech2Text("asr_config.yml", "asr.pb")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2text(audio)
         [(text, token, token_int, hypothesis object), ...]
 
@@ -251,9 +251,9 @@ class Speech2TextParaformer:
     """Speech2Text class
 
     Examples:
-            >>> import soundfile
+            >>> import librosa
             >>> speech2text = Speech2TextParaformer("asr_config.yml", "asr.pb")
-            >>> audio, rate = soundfile.read("speech.wav")
+            >>> audio, rate = librosa.load("speech.wav")
             >>> speech2text(audio)
             [(text, token, token_int, hypothesis object), ...]
 
@@ -625,9 +625,9 @@ class Speech2TextParaformerOnline:
     """Speech2Text class
 
     Examples:
-            >>> import soundfile
+            >>> import librosa
             >>> speech2text = Speech2TextParaformerOnline("asr_config.yml", "asr.pth")
-            >>> audio, rate = soundfile.read("speech.wav")
+            >>> audio, rate = librosa.load("speech.wav")
             >>> speech2text(audio)
             [(text, token, token_int, hypothesis object), ...]
 
@@ -876,9 +876,9 @@ class Speech2TextUniASR:
     """Speech2Text class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech2text = Speech2TextUniASR("asr_config.yml", "asr.pb")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2text(audio)
         [(text, token, token_int, hypothesis object), ...]
 
@@ -1106,9 +1106,9 @@ class Speech2TextMFCCA:
     """Speech2Text class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech2text = Speech2TextMFCCA("asr_config.yml", "asr.pb")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2text(audio)
         [(text, token, token_int, hypothesis object), ...]
 
@@ -1637,9 +1637,9 @@ class Speech2TextSAASR:
     """Speech2Text class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech2text = Speech2TextSAASR("asr_config.yml", "asr.pb")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2text(audio)
         [(text, token, token_int, hypothesis object), ...]
 
@@ -1885,9 +1885,9 @@ class Speech2TextWhisper:
     """Speech2Text class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech2text = Speech2Text("asr_config.yml", "asr.pb")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2text(audio)
         [(text, token, token_int, hypothesis object), ...]
 

funasr/bin/asr_inference_launch.py

@@ -20,7 +20,8 @@ from typing import Union
 import numpy as np
 import torch
 import torchaudio
-import soundfile
+# import librosa
+import librosa
 import yaml
 
 from funasr.bin.asr_infer import Speech2Text
@@ -1281,7 +1282,8 @@ def inference_paraformer_online(
             try:
                 raw_inputs = torchaudio.load(data_path_and_name_and_type[0])[0][0]
             except:
-                raw_inputs = soundfile.read(data_path_and_name_and_type[0], dtype='float32')[0]
+                # raw_inputs = librosa.load(data_path_and_name_and_type[0], dtype='float32')[0]
+                raw_inputs, sr = librosa.load(data_path_and_name_and_type[0], dtype='float32')
                 if raw_inputs.ndim == 2:
                     raw_inputs = raw_inputs[:, 0]
                 raw_inputs = torch.tensor(raw_inputs)

funasr/bin/build_trainer.py

@@ -18,7 +18,7 @@ from funasr.build_utils.build_optimizer import build_optimizer
 from funasr.build_utils.build_scheduler import build_scheduler
 from funasr.build_utils.build_trainer import build_trainer as build_trainer_modelscope
 from funasr.modules.lora.utils import mark_only_lora_as_trainable
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.torch_utils.load_pretrained_model import load_pretrained_model
 from funasr.torch_utils.model_summary import model_summary
 from funasr.torch_utils.pytorch_version import pytorch_cudnn_version

funasr/bin/diar_infer.py

@@ -27,11 +27,11 @@ class Speech2DiarizationEEND:
     """Speech2Diarlization class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> import numpy as np
         >>> speech2diar = Speech2DiarizationEEND("diar_sond_config.yml", "diar_sond.pb")
         >>> profile = np.load("profiles.npy")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2diar(audio, profile)
         {"spk1": [(int, int), ...], ...}
 
@@ -109,11 +109,11 @@ class Speech2DiarizationSOND:
     """Speech2Xvector class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> import numpy as np
         >>> speech2diar = Speech2DiarizationSOND("diar_sond_config.yml", "diar_sond.pb")
         >>> profile = np.load("profiles.npy")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2diar(audio, profile)
         {"spk1": [(int, int), ...], ...}
 

funasr/bin/diar_inference_launch.py

@@ -15,7 +15,8 @@ from typing import Tuple
 from typing import Union
 
 import numpy as np
-import soundfile
+# import librosa
+import librosa
 import torch
 from scipy.signal import medfilt
 
@@ -144,7 +145,9 @@ def inference_sond(
                         # read waveform file
                         example = [load_bytes(x) if isinstance(x, bytes) else x
                                    for x in example]
-                        example = [soundfile.read(x)[0] if isinstance(x, str) else x
+                        # example = [librosa.load(x)[0] if isinstance(x, str) else x
+                        #            for x in example]
+                        example = [librosa.load(x, dtype='float32')[0] if isinstance(x, str) else x
                                    for x in example]
                         # convert torch tensor to numpy array
                         example = [x.numpy() if isinstance(example[0], torch.Tensor) else x

funasr/bin/ss_infer.py

@@ -20,9 +20,9 @@ class SpeechSeparator:
     """SpeechSeparator class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech_separator = MossFormer("ss_config.yml", "ss.pt")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> separated_wavs = speech_separator(audio)        
 
     """

funasr/bin/ss_inference_launch.py

@@ -13,7 +13,7 @@ from typing import Union
 
 import numpy as np
 import torch
-import soundfile as sf
+import librosa
 from funasr.build_utils.build_streaming_iterator import build_streaming_iterator
 from funasr.torch_utils.set_all_random_seed import set_all_random_seed
 from funasr.utils import config_argparse
@@ -104,7 +104,12 @@ def inference_ss(
             ss_results = speech_separator(**batch)
             
             for spk in range(num_spks):
-                sf.write(os.path.join(output_path, keys[0] + '_s' + str(spk+1)+'.wav'), ss_results[spk], sample_rate)
+                # sf.write(os.path.join(output_path, keys[0] + '_s' + str(spk+1)+'.wav'), ss_results[spk], sample_rate)
+                try:
+                    librosa.output.write_wav(os.path.join(output_path, keys[0] + '_s' + str(spk+1)+'.wav'), ss_results[spk], sample_rate)
+                except:
+                    print("To write wav by librosa, you should install librosa<=0.8.0")
+                    raise
         torch.cuda.empty_cache()
         return ss_results
 

funasr/bin/sv_infer.py

@@ -22,9 +22,9 @@ class Speech2Xvector:
     """Speech2Xvector class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech2xvector = Speech2Xvector("sv_config.yml", "sv.pb")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2xvector(audio)
         [(text, token, token_int, hypothesis object), ...]
 

funasr/bin/tokenize_text.py

@@ -9,9 +9,9 @@ from typing import Optional
 
 
 from funasr.utils.cli_utils import get_commandline_args
-from funasr.text.build_tokenizer import build_tokenizer
-from funasr.text.cleaner import TextCleaner
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.build_tokenizer import build_tokenizer
+from funasr.tokenizer.cleaner import TextCleaner
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.utils.types import str2bool
 from funasr.utils.types import str_or_none
 

funasr/bin/tp_infer.py

@@ -11,7 +11,7 @@ import numpy as np
 import torch
 from funasr.build_utils.build_model_from_file import build_model_from_file
 from funasr.models.frontend.wav_frontend import WavFrontend
-from funasr.text.token_id_converter import TokenIDConverter
+from funasr.tokenizer.token_id_converter import TokenIDConverter
 from funasr.torch_utils.device_funcs import to_device
 
 

funasr/bin/train.py

@@ -17,7 +17,7 @@ from funasr.build_utils.build_model import build_model
 from funasr.build_utils.build_optimizer import build_optimizer
 from funasr.build_utils.build_scheduler import build_scheduler
 from funasr.build_utils.build_trainer import build_trainer
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.torch_utils.load_pretrained_model import load_pretrained_model
 from funasr.torch_utils.model_summary import model_summary
 from funasr.torch_utils.pytorch_version import pytorch_cudnn_version

funasr/bin/vad_infer.py

@@ -23,9 +23,9 @@ class Speech2VadSegment:
     """Speech2VadSegment class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech2segment = Speech2VadSegment("vad_config.yml", "vad.pt")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2segment(audio)
         [[10, 230], [245, 450], ...]
 
@@ -118,9 +118,9 @@ class Speech2VadSegmentOnline(Speech2VadSegment):
     """Speech2VadSegmentOnline class
 
     Examples:
-        >>> import soundfile
+        >>> import librosa
         >>> speech2segment = Speech2VadSegmentOnline("vad_config.yml", "vad.pt")
-        >>> audio, rate = soundfile.read("speech.wav")
+        >>> audio, rate = librosa.load("speech.wav")
         >>> speech2segment(audio)
         [[10, 230], [245, 450], ...]
 

funasr/build_utils/build_trainer.py

@@ -246,14 +246,11 @@ class Trainer:
         for iepoch in range(start_epoch, trainer_options.max_epoch + 1):
             if iepoch != start_epoch:
                 logging.info(
-                    "{}/{}epoch started. Estimated time to finish: {}".format(
+                    "{}/{}epoch started. Estimated time to finish: {} hours".format(
                         iepoch,
                         trainer_options.max_epoch,
-                        humanfriendly.format_timespan(
-                            (time.perf_counter() - start_time)
-                            / (iepoch - start_epoch)
-                            * (trainer_options.max_epoch - iepoch + 1)
-                        ),
+                        (time.perf_counter() - start_time) / 3600.0 / (iepoch - start_epoch) * (
+                                trainer_options.max_epoch - iepoch + 1),
                     )
                 )
             else:

funasr/datasets/data_sampler.py

@@ -0,0 +1,74 @@
+import torch
+
+import numpy as np
+
+class BatchSampler(torch.utils.data.BatchSampler):
+	
+	def __init__(self, dataset, batch_size_type: str="example", batch_size: int=14, sort_size: int=30, drop_last: bool=False, shuffle: bool=True, **kwargs):
+		
+		self.drop_last = drop_last
+		self.pre_idx = -1
+		self.dataset = dataset
+		self.total_samples = len(dataset)
+		# self.batch_size_type = args.batch_size_type
+		# self.batch_size = args.batch_size
+		# self.sort_size = args.sort_size
+		# self.max_length_token = args.max_length_token
+		self.batch_size_type = batch_size_type
+		self.batch_size = batch_size
+		self.sort_size = sort_size
+		self.max_length_token = kwargs.get("max_length_token", 5000)
+		self.shuffle_idx = np.arange(self.total_samples)
+		self.shuffle = shuffle
+
+	
+	def __len__(self):
+		return self.total_samples
+
+	def __iter__(self):
+		print("in sampler")
+		
+		if self.shuffle:
+			np.random.shuffle(self.shuffle_idx)
+			
+		batch = []
+		max_token = 0
+		num_sample = 0
+
+		iter_num = (self.total_samples-1) // self.sort_size + 1
+		print("iter_num: ", iter_num)
+		for iter in range(self.pre_idx + 1, iter_num):
+			datalen_with_index = []
+			for i in range(self.sort_size):
+				idx = iter * self.sort_size + i
+				if idx >= self.total_samples:
+					continue
+
+				idx_map = self.shuffle_idx[idx]
+				# prompt = self.dataset.indexed_dataset[idx_map]["prompt"]
+				sample_len_cur = self.dataset.indexed_dataset[idx_map]["source_len"] + \
+				                 self.dataset.indexed_dataset[idx_map]["target_len"]
+
+				datalen_with_index.append([idx, sample_len_cur])
+			
+			datalen_with_index_sort = sorted(datalen_with_index, key=lambda x: x[1])
+			for item in datalen_with_index_sort:
+				idx, sample_len_cur_raw = item
+				if sample_len_cur_raw > self.max_length_token:
+					continue
+
+				max_token_cur = max(max_token, sample_len_cur_raw)
+				max_token_padding = 1 + num_sample
+				if self.batch_size_type == 'token':
+					max_token_padding *= max_token_cur
+				if max_token_padding <= self.batch_size:
+					batch.append(idx)
+					max_token = max_token_cur
+					num_sample += 1
+				else:
+					yield batch
+					batch = [idx]
+					max_token = sample_len_cur_raw
+					num_sample = 1
+					
+		

funasr/datasets/dataloader_fn.py

@@ -0,0 +1,53 @@
+
+import torch
+from funasr.datasets.dataset_jsonl import AudioDataset
+from funasr.datasets.data_sampler import BatchSampler
+from funasr.models.frontend.wav_frontend import WavFrontend
+from funasr.tokenizer.build_tokenizer import build_tokenizer
+from funasr.tokenizer.token_id_converter import TokenIDConverter
+collate_fn = None
+# collate_fn = collate_fn,
+
+jsonl = "/Users/zhifu/funasr_github/test_local/all_task_debug_len.jsonl"
+
+frontend = WavFrontend()
+token_type = 'char'
+bpemodel = None
+delimiter = None
+space_symbol = "<space>"
+non_linguistic_symbols = None
+g2p_type = None
+
+tokenizer = build_tokenizer(
+    token_type=token_type,
+    bpemodel=bpemodel,
+    delimiter=delimiter,
+    space_symbol=space_symbol,
+    non_linguistic_symbols=non_linguistic_symbols,
+    g2p_type=g2p_type,
+)
+token_list = ""
+unk_symbol = "<unk>"
+
+token_id_converter = TokenIDConverter(
+    token_list=token_list,
+    unk_symbol=unk_symbol,
+)
+
+dataset = AudioDataset(jsonl, frontend=frontend, tokenizer=tokenizer)
+batch_sampler = BatchSampler(dataset)
+dataloader_tr = torch.utils.data.DataLoader(dataset,
+                           collate_fn=dataset.collator,
+                           batch_sampler=batch_sampler,
+                           shuffle=False,
+                           num_workers=0,
+                           pin_memory=True)
+
+print(len(dataset))
+for i in range(3):
+    print(i)
+    for data in dataloader_tr:
+        print(len(data), data)
+# data_iter = iter(dataloader_tr)
+# data = next(data_iter)
+pass

funasr/datasets/dataset.py

@@ -16,8 +16,10 @@ from typing import Dict
 from typing import Mapping
 from typing import Tuple
 from typing import Union
-
-import h5py
+try:
+    import h5py
+except:
+    print("If you want use h5py dataset, please pip install h5py, and try it again")
 import humanfriendly
 import kaldiio
 import numpy as np

funasr/datasets/dataset_jsonl.py

@@ -0,0 +1,124 @@
+import torch
+import json
+import torch.distributed as dist
+import numpy as np
+import kaldiio
+import librosa
+
+
+
+def load_audio(audio_path: str, fs: int=16000):
+	audio = None
+	if audio_path.startswith("oss:"):
+		pass
+	elif audio_path.startswith("odps:"):
+		pass
+	else:
+		if ".ark:" in audio_path:
+			audio = kaldiio.load_mat(audio_path)
+		else:
+			audio, fs = librosa.load(audio_path, sr=fs)
+	return audio
+
+def extract_features(data, date_type: str="sound", frontend=None):
+	if date_type == "sound":
+		feat, feats_lens = frontend(data, len(data))
+		feat = feat[0, :, :]
+	else:
+		feat, feats_lens = torch.from_numpy(data).to(torch.float32), torch.tensor([data.shape[0]]).to(torch.int32)
+	return feat, feats_lens
+	
+	
+
+class IndexedDatasetJsonl(torch.utils.data.Dataset):
+	
+	def __init__(self, path):
+		super().__init__()
+		# data_parallel_size = dist.get_world_size()
+		data_parallel_size = 1
+		contents = []
+		with open(path, encoding='utf-8') as fin:
+			for line in fin:
+				data = json.loads(line.strip())
+				if "text" in data:  # for sft
+					self.contents.append(data['text'])
+				if "source" in data:  # for speech lab pretrain
+					prompt = data["prompt"]
+					source = data["source"]
+					target = data["target"]
+					source_len = data["source_len"]
+					target_len = data["target_len"]
+
+					contents.append({"source": source,
+					                 "prompt": prompt,
+					                 "target": target,
+					                 "source_len": source_len,
+					                 "target_len": target_len,
+					                 }
+					                )
+		
+		self.contents = []
+		total_num = len(contents)
+		num_per_rank = total_num // data_parallel_size
+		# rank = dist.get_rank()
+		rank = 0
+		# import ipdb; ipdb.set_trace()
+		self.contents = contents[rank * num_per_rank:(rank + 1) * num_per_rank]
+
+
+	def __len__(self):
+		return len(self.contents)
+	
+	def __getitem__(self, index):
+		return self.contents[index]
+
+
+class AudioDataset(torch.utils.data.Dataset):
+	def __init__(self, path, frontend=None, tokenizer=None):
+		super().__init__()
+		self.indexed_dataset = IndexedDatasetJsonl(path)
+		self.frontend = frontend.forward
+		self.fs = 16000 if frontend is None else frontend.fs
+		self.data_type = "sound"
+		self.tokenizer = tokenizer
+		self.int_pad_value = -1
+		self.float_pad_value = 0.0
+
+	
+
+	
+	def __len__(self):
+		return len(self.indexed_dataset)
+	
+	def __getitem__(self, index):
+		item = self.indexed_dataset[index]
+		source = item["source"]
+		data_src = load_audio(source, fs=self.fs)
+		speech, speech_lengths = extract_features(data_src, self.data_type, self.frontend)
+		target = item["target"]
+		text = self.tokenizer.encode(target)
+		text_lengths = len(text)
+		text, text_lengths = torch.tensor(text, dtype=torch.int64), torch.tensor([text_lengths], dtype=torch.int32)
+		return {"speech": speech,
+		        "speech_lengths": speech_lengths,
+		        "text": text,
+		        "text_lengths": text_lengths,
+		        }
+	
+	
+	def collator(self, samples: list=None):
+		
+		outputs = {}
+		for sample in samples:
+			for key in sample.keys():
+				if key not in outputs:
+					outputs[key] = []
+				outputs[key].append(sample[key])
+		
+		for key, data_list in outputs.items():
+			if data_list[0].dtype.kind == "i":
+				pad_value = self.int_pad_value
+			else:
+				pad_value = self.float_pad_value
+			outputs[key] = torch.nn.utils.rnn.pad_sequence(data_list, batch_first=True, padding_value=pad_value)
+		return samples

funasr/datasets/iterable_dataset.py

@@ -14,7 +14,8 @@ import kaldiio
 import numpy as np
 import torch
 import torchaudio
-import soundfile
+# import librosa
+import librosa
 from torch.utils.data.dataset import IterableDataset
 import os.path
 
@@ -70,7 +71,8 @@ def load_wav(input):
     try:
         return torchaudio.load(input)[0].numpy()
     except:
-        waveform, _ = soundfile.read(input, dtype='float32')
+        # waveform, _ = librosa.load(input, dtype='float32')
+        waveform, _ = librosa.load(input, dtype='float32')
         if waveform.ndim == 2:
             waveform = waveform[:, 0]
         return np.expand_dims(waveform, axis=0)

funasr/datasets/large_datasets/build_dataloader.py

@@ -9,7 +9,7 @@ from torch.utils.data import DataLoader
 
 from funasr.datasets.large_datasets.dataset import Dataset
 from funasr.iterators.abs_iter_factory import AbsIterFactory
-from funasr.text.abs_tokenizer import AbsTokenizer
+from funasr.tokenizer.abs_tokenizer import AbsTokenizer
 
 
 def read_symbol_table(symbol_table_file):

funasr/datasets/large_datasets/dataset.py

@@ -7,7 +7,8 @@ import torch
 import torch.distributed as dist
 import torchaudio
 import numpy as np
-import soundfile
+# import librosa
+import librosa
 from kaldiio import ReadHelper
 from torch.utils.data import IterableDataset
 
@@ -128,7 +129,8 @@ class AudioDataset(IterableDataset):
                         try:
                             waveform, sampling_rate = torchaudio.load(path)
                         except:
-                            waveform, sampling_rate = soundfile.read(path, dtype='float32')
+                            # waveform, sampling_rate = librosa.load(path, dtype='float32')
+                            waveform, sampling_rate = librosa.load(path, dtype='float32')
                             if waveform.ndim == 2:
                                 waveform = waveform[:, 0]
                             waveform = np.expand_dims(waveform, axis=0)

funasr/datasets/preprocessor.py

@@ -10,12 +10,12 @@ from typing import Union
 
 import numpy as np
 import scipy.signal
-import soundfile
+import librosa
 import jieba
 
-from funasr.text.build_tokenizer import build_tokenizer
-from funasr.text.cleaner import TextCleaner
-from funasr.text.token_id_converter import TokenIDConverter
+from funasr.tokenizer.build_tokenizer import build_tokenizer
+from funasr.tokenizer.cleaner import TextCleaner
+from funasr.tokenizer.token_id_converter import TokenIDConverter
 
 
 class AbsPreprocessor(ABC):
@@ -284,7 +284,7 @@ class CommonPreprocessor(AbsPreprocessor):
                 if self.rirs is not None and self.rir_apply_prob >= np.random.random():
                     rir_path = np.random.choice(self.rirs)
                     if rir_path is not None:
-                        rir, _ = soundfile.read(
+                        rir, _ = librosa.load(
                             rir_path, dtype=np.float64, always_2d=True
                         )
 
@@ -310,28 +310,31 @@ class CommonPreprocessor(AbsPreprocessor):
                         noise_db = np.random.uniform(
                             self.noise_db_low, self.noise_db_high
                         )
-                        with soundfile.SoundFile(noise_path) as f:
-                            if f.frames == nsamples:
-                                noise = f.read(dtype=np.float64, always_2d=True)
-                            elif f.frames < nsamples:
-                                offset = np.random.randint(0, nsamples - f.frames)
-                                # noise: (Time, Nmic)
-                                noise = f.read(dtype=np.float64, always_2d=True)
-                                # Repeat noise
-                                noise = np.pad(
-                                    noise,
-                                    [(offset, nsamples - f.frames - offset), (0, 0)],
-                                    mode="wrap",
-                                )
-                            else:
-                                offset = np.random.randint(0, f.frames - nsamples)
-                                f.seek(offset)
-                                # noise: (Time, Nmic)
-                                noise = f.read(
-                                    nsamples, dtype=np.float64, always_2d=True
-                                )
-                                if len(noise) != nsamples:
-                                    raise RuntimeError(f"Something wrong: {noise_path}")
+
+                        audio_data = librosa.load(noise_path, dtype='float32')[0][None, :]
+                        frames = len(audio_data[0])
+                        if frames == nsamples:
+                            noise = audio_data
+                        elif frames < nsamples:
+                            offset = np.random.randint(0, nsamples - frames)
+                            # noise: (Time, Nmic)
+                            noise = audio_data
+                            # Repeat noise
+                            noise = np.pad(
+                                noise,
+                                [(offset, nsamples - frames - offset), (0, 0)],
+                                mode="wrap",
+                            )
+                        else:
+                            noise = audio_data[:, nsamples]
+                            # offset = np.random.randint(0, frames - nsamples)
+                            # f.seek(offset)
+                            # noise: (Time, Nmic)
+                            # noise = f.read(
+                            #     nsamples, dtype=np.float64, always_2d=True
+                            # )
+                            # if len(noise) != nsamples:
+                            #     raise RuntimeError(f"Something wrong: {noise_path}")
                         # noise: (Nmic, Time)
                         noise = noise.T
 

funasr/datasets/small_datasets/preprocessor.py

@@ -9,11 +9,11 @@ from typing import Union
 
 import numpy as np
 import scipy.signal
-import soundfile
+import librosa
 
-from funasr.text.build_tokenizer import build_tokenizer
-from funasr.text.cleaner import TextCleaner
-from funasr.text.token_id_converter import TokenIDConverter
+from funasr.tokenizer.build_tokenizer import build_tokenizer
+from funasr.tokenizer.cleaner import TextCleaner
+from funasr.tokenizer.token_id_converter import TokenIDConverter
 
 
 class AbsPreprocessor(ABC):
@@ -275,7 +275,7 @@ class CommonPreprocessor(AbsPreprocessor):
                 if self.rirs is not None and self.rir_apply_prob >= np.random.random():
                     rir_path = np.random.choice(self.rirs)
                     if rir_path is not None:
-                        rir, _ = soundfile.read(
+                        rir, _ = librosa.load(
                             rir_path, dtype=np.float64, always_2d=True
                         )
 
@@ -301,28 +301,30 @@ class CommonPreprocessor(AbsPreprocessor):
                         noise_db = np.random.uniform(
                             self.noise_db_low, self.noise_db_high
                         )
-                        with soundfile.SoundFile(noise_path) as f:
-                            if f.frames == nsamples:
-                                noise = f.read(dtype=np.float64, always_2d=True)
-                            elif f.frames < nsamples:
-                                offset = np.random.randint(0, nsamples - f.frames)
-                                # noise: (Time, Nmic)
-                                noise = f.read(dtype=np.float64, always_2d=True)
-                                # Repeat noise
-                                noise = np.pad(
-                                    noise,
-                                    [(offset, nsamples - f.frames - offset), (0, 0)],
-                                    mode="wrap",
-                                )
-                            else:
-                                offset = np.random.randint(0, f.frames - nsamples)
-                                f.seek(offset)
-                                # noise: (Time, Nmic)
-                                noise = f.read(
-                                    nsamples, dtype=np.float64, always_2d=True
-                                )
-                                if len(noise) != nsamples:
-                                    raise RuntimeError(f"Something wrong: {noise_path}")
+                        audio_data = librosa.load(noise_path, dtype='float32')[0][None, :]
+                        frames = len(audio_data[0])
+                        if frames == nsamples:
+                            noise = audio_data
+                        elif frames < nsamples:
+                            offset = np.random.randint(0, nsamples - frames)
+                            # noise: (Time, Nmic)
+                            noise = audio_data
+                            # Repeat noise
+                            noise = np.pad(
+                                noise,
+                                [(offset, nsamples - frames - offset), (0, 0)],
+                                mode="wrap",
+                            )
+                        else:
+                            noise = audio_data[:, nsamples]
+                            # offset = np.random.randint(0, frames - nsamples)
+                            # f.seek(offset)
+                            # noise: (Time, Nmic)
+                            # noise = f.read(
+                            #     nsamples, dtype=np.float64, always_2d=True
+                            # )
+                            # if len(noise) != nsamples:
+                            #     raise RuntimeError(f"Something wrong: {noise_path}")
                         # noise: (Nmic, Time)
                         noise = noise.T
 

funasr/export/export_conformer.py

@@ -1,151 +0,0 @@
-import json
-from typing import Union, Dict
-from pathlib import Path
-
-import os
-import logging
-import torch
-
-from funasr.export.models import get_model
-import numpy as np
-import random
-from funasr.utils.types import str2bool, str2triple_str
-# torch_version = float(".".join(torch.__version__.split(".")[:2]))
-# assert torch_version > 1.9
-
-class ModelExport:
-    def __init__(
-        self,
-        cache_dir: Union[Path, str] = None,
-        onnx: bool = True,
-        device: str = "cpu",
-        quant: bool = True,
-        fallback_num: int = 0,
-        audio_in: str = None,
-        calib_num: int = 200,
-        model_revision: str = None,
-    ):
-        self.set_all_random_seed(0)
-
-        self.cache_dir = cache_dir
-        self.export_config = dict(
-            feats_dim=560,
-            onnx=False,
-        )
-        
-        self.onnx = onnx
-        self.device = device
-        self.quant = quant
-        self.fallback_num = fallback_num
-        self.frontend = None
-        self.audio_in = audio_in
-        self.calib_num = calib_num
-        self.model_revision = model_revision
-
-    def _export(
-        self,
-        model,
-        model_dir: str = None,
-        verbose: bool = False,
-    ):
-
-        export_dir = model_dir
-        os.makedirs(export_dir, exist_ok=True)
-
-        self.export_config["model_name"] = "model"
-        model = get_model(
-            model,
-            self.export_config,
-        )
-        model.eval()
-
-        if self.onnx:
-            self._export_onnx(model, verbose, export_dir)
-
-        print("output dir: {}".format(export_dir))
-
-    def _export_onnx(self, model, verbose, path):
-        model._export_onnx(verbose, path)
-
-    def set_all_random_seed(self, seed: int):
-        random.seed(seed)
-        np.random.seed(seed)
-        torch.random.manual_seed(seed)
-
-    def parse_audio_in(self, audio_in):
-        
-        wav_list, name_list = [], []
-        if audio_in.endswith(".scp"):
-            f = open(audio_in, 'r')
-            lines = f.readlines()[:self.calib_num]
-            for line in lines:
-                name, path = line.strip().split()
-                name_list.append(name)
-                wav_list.append(path)
-        else:
-            wav_list = [audio_in,]
-            name_list = ["test",]
-        return wav_list, name_list
-    
-    def load_feats(self, audio_in: str = None):
-        import torchaudio
-
-        wav_list, name_list = self.parse_audio_in(audio_in)
-        feats = []
-        feats_len = []
-        for line in wav_list:
-            path = line.strip()
-            waveform, sampling_rate = torchaudio.load(path)
-            if sampling_rate != self.frontend.fs:
-                waveform = torchaudio.transforms.Resample(orig_freq=sampling_rate,
-                                                          new_freq=self.frontend.fs)(waveform)
-            fbank, fbank_len = self.frontend(waveform, [waveform.size(1)])
-            feats.append(fbank)
-            feats_len.append(fbank_len)
-        return feats, feats_len
-
-    def export(self,
-               mode: str = None,
-               ):
-
-        if mode.startswith('conformer'):
-            from funasr.tasks.asr import ASRTask
-            config = os.path.join(model_dir, 'config.yaml')
-            model_file = os.path.join(model_dir, 'model.pb')
-            cmvn_file = os.path.join(model_dir, 'am.mvn')
-            model, asr_train_args = ASRTask.build_model_from_file(
-                config, model_file, cmvn_file, 'cpu'
-            )
-            self.frontend = model.frontend
-            self.export_config["feats_dim"] = 560
-
-        self._export(model, self.cache_dir)
-
-if __name__ == '__main__':
-    import argparse
-    parser = argparse.ArgumentParser()
-    # parser.add_argument('--model-name', type=str, required=True)
-    parser.add_argument('--model-name', type=str, action="append", required=True, default=[])
-    parser.add_argument('--export-dir', type=str, required=True)
-    parser.add_argument('--type', type=str, default='onnx', help='["onnx", "torch"]')
-    parser.add_argument('--device', type=str, default='cpu', help='["cpu", "cuda"]')
-    parser.add_argument('--quantize', type=str2bool, default=False, help='export quantized model')
-    parser.add_argument('--fallback-num', type=int, default=0, help='amp fallback number')
-    parser.add_argument('--audio_in', type=str, default=None, help='["wav", "wav.scp"]')
-    parser.add_argument('--calib_num', type=int, default=200, help='calib max num')
-    parser.add_argument('--model_revision', type=str, default=None, help='model_revision')
-    args = parser.parse_args()
-
-    export_model = ModelExport(
-        cache_dir=args.export_dir,
-        onnx=args.type == 'onnx',
-        device=args.device,
-        quant=args.quantize,
-        fallback_num=args.fallback_num,
-        audio_in=args.audio_in,
-        calib_num=args.calib_num,
-        model_revision=args.model_revision,
-    )
-    for model_name in args.model_name:
-        print("export model: {}".format(model_name))
-        export_model.export(model_name)

funasr/export/models/__init__.py

@@ -1,7 +1,7 @@
 from funasr.models.e2e_asr_paraformer import Paraformer, BiCifParaformer, ParaformerOnline
 from funasr.export.models.e2e_asr_paraformer import Paraformer as Paraformer_export
 from funasr.export.models.e2e_asr_paraformer import BiCifParaformer as BiCifParaformer_export
-from funasr.export.models.e2e_asr_conformer import Conformer as Conformer_export
+# from funasr.export.models.e2e_asr_conformer import Conformer as Conformer_export
 
 from funasr.models.e2e_vad import E2EVadModel
 from funasr.export.models.e2e_vad import E2EVadModel as E2EVadModel_export
@@ -30,8 +30,8 @@ def get_model(model, export_config=None):
         return [encoder, decoder]
     elif isinstance(model, Paraformer):
         return Paraformer_export(model, **export_config)
-    elif isinstance(model, Conformer_export):
-        return Conformer_export(model, **export_config)
+    # elif isinstance(model, Conformer_export):
+    #     return Conformer_export(model, **export_config)
     elif isinstance(model, E2EVadModel):
         return E2EVadModel_export(model, **export_config)
     elif isinstance(model, PunctuationModel):

funasr/export/models/e2e_asr_conformer.py

@@ -1,69 +0,0 @@
-import os
-import logging
-import torch
-import torch.nn as nn
-
-from funasr.export.utils.torch_function import MakePadMask
-from funasr.export.utils.torch_function import sequence_mask
-from funasr.models.encoder.conformer_encoder import ConformerEncoder
-from funasr.models.decoder.transformer_decoder import TransformerDecoder
-from funasr.export.models.encoder.conformer_encoder import ConformerEncoder as ConformerEncoder_export
-from funasr.export.models.decoder.xformer_decoder import XformerDecoder as TransformerDecoder_export
-
-class Conformer(nn.Module):
-    """
-    export conformer into onnx format
-    """
-
-    def __init__(
-            self,
-            model,
-            max_seq_len=512,
-            feats_dim=560,
-            model_name='model',
-            **kwargs,
-    ):
-        super().__init__()
-        onnx = False
-        if "onnx" in kwargs:
-            onnx = kwargs["onnx"]
-        if isinstance(model.encoder, ConformerEncoder):
-            self.encoder = ConformerEncoder_export(model.encoder, onnx=onnx)
-        elif isinstance(model.decoder, TransformerDecoder):
-            self.decoder = TransformerDecoder_export(model.decoder, onnx=onnx)
-        
-        self.feats_dim = feats_dim
-        self.model_name = model_name
-
-        if onnx:
-            self.make_pad_mask = MakePadMask(max_seq_len, flip=False)
-        else:
-            self.make_pad_mask = sequence_mask(max_seq_len, flip=False)
-
-    def _export_model(self, model, verbose, path):
-        dummy_input = model.get_dummy_inputs()
-        model_script = model
-        model_path = os.path.join(path, f'{model.model_name}.onnx')
-        if not os.path.exists(model_path):
-            torch.onnx.export(
-                model_script,
-                dummy_input,
-                model_path,
-                verbose=verbose,
-                opset_version=14,
-                input_names=model.get_input_names(),
-                output_names=model.get_output_names(),
-                dynamic_axes=model.get_dynamic_axes()
-            )
-
-    def _export_encoder_onnx(self, verbose, path):
-        model_encoder = self.encoder
-        self._export_model(model_encoder, verbose, path)
-
-    def _export_decoder_onnx(self, verbose, path):
-        model_decoder = self.decoder
-        self._export_model(model_decoder, verbose, path)
-
-    def _export_onnx(self, verbose, path):
-        self._export_encoder_onnx(verbose, path)
-        self._export_decoder_onnx(verbose, path)

funasr/export/models/language_models/embed.py

@@ -1,403 +0,0 @@
-"""Positional Encoding Module."""
-
-import math
-
-import torch
-import torch.nn as nn
-from funasr.modules.embedding import (
-    LegacyRelPositionalEncoding, PositionalEncoding, RelPositionalEncoding,
-    ScaledPositionalEncoding, StreamPositionalEncoding)
-from funasr.modules.subsampling import (
-    Conv2dSubsampling, Conv2dSubsampling2, Conv2dSubsampling6,
-    Conv2dSubsampling8)
-from funasr.modules.subsampling_without_posenc import \
-    Conv2dSubsamplingWOPosEnc
-
-from funasr.export.models.language_models.subsampling import (
-    OnnxConv2dSubsampling, OnnxConv2dSubsampling2, OnnxConv2dSubsampling6,
-    OnnxConv2dSubsampling8)
-
-
-def get_pos_emb(pos_emb, max_seq_len=512, use_cache=True):
-    if isinstance(pos_emb, LegacyRelPositionalEncoding):
-        return OnnxLegacyRelPositionalEncoding(pos_emb, max_seq_len, use_cache)
-    elif isinstance(pos_emb, ScaledPositionalEncoding):
-        return OnnxScaledPositionalEncoding(pos_emb, max_seq_len, use_cache)
-    elif isinstance(pos_emb, RelPositionalEncoding):
-        return OnnxRelPositionalEncoding(pos_emb, max_seq_len, use_cache)
-    elif isinstance(pos_emb, PositionalEncoding):
-        return OnnxPositionalEncoding(pos_emb, max_seq_len, use_cache)
-    elif isinstance(pos_emb, StreamPositionalEncoding):
-        return OnnxStreamPositionalEncoding(pos_emb, max_seq_len, use_cache)
-    elif (isinstance(pos_emb, nn.Sequential) and len(pos_emb) == 0) or (
-        isinstance(pos_emb, Conv2dSubsamplingWOPosEnc)
-    ):
-        return pos_emb
-    else:
-        raise ValueError("Embedding model is not supported.")
-
-
-class Embedding(nn.Module):
-    def __init__(self, model, max_seq_len=512, use_cache=True):
-        super().__init__()
-        self.model = model
-        if not isinstance(model, nn.Embedding):
-            if isinstance(model, Conv2dSubsampling):
-                self.model = OnnxConv2dSubsampling(model)
-                self.model.out[-1] = get_pos_emb(model.out[-1], max_seq_len)
-            elif isinstance(model, Conv2dSubsampling2):
-                self.model = OnnxConv2dSubsampling2(model)
-                self.model.out[-1] = get_pos_emb(model.out[-1], max_seq_len)
-            elif isinstance(model, Conv2dSubsampling6):
-                self.model = OnnxConv2dSubsampling6(model)
-                self.model.out[-1] = get_pos_emb(model.out[-1], max_seq_len)
-            elif isinstance(model, Conv2dSubsampling8):
-                self.model = OnnxConv2dSubsampling8(model)
-                self.model.out[-1] = get_pos_emb(model.out[-1], max_seq_len)
-            else:
-                self.model[-1] = get_pos_emb(model[-1], max_seq_len)
-
-    def forward(self, x, mask=None):
-        if mask is None:
-            return self.model(x)
-        else:
-            return self.model(x, mask)
-
-
-def _pre_hook(
-    state_dict,
-    prefix,
-    local_metadata,
-    strict,
-    missing_keys,
-    unexpected_keys,
-    error_msgs,
-):
-    """Perform pre-hook in load_state_dict for backward compatibility.
-
-    Note:
-        We saved self.pe until v.0.5.2 but we have omitted it later.
-        Therefore, we remove the item "pe" from `state_dict` for backward compatibility.
-
-    """
-    k = prefix + "pe"
-    if k in state_dict:
-        state_dict.pop(k)
-
-
-class OnnxPositionalEncoding(torch.nn.Module):
-    """Positional encoding.
-
-    Args:
-        d_model (int): Embedding dimension.
-        dropout_rate (float): Dropout rate.
-        max_seq_len (int): Maximum input length.
-        reverse (bool): Whether to reverse the input position. Only for
-        the class LegacyRelPositionalEncoding. We remove it in the current
-        class RelPositionalEncoding.
-    """
-
-    def __init__(self, model, max_seq_len=512, reverse=False, use_cache=True):
-        """Construct an PositionalEncoding object."""
-        super(OnnxPositionalEncoding, self).__init__()
-        self.d_model = model.d_model
-        self.reverse = reverse
-        self.max_seq_len = max_seq_len
-        self.xscale = math.sqrt(self.d_model)
-        self._register_load_state_dict_pre_hook(_pre_hook)
-        self.pe = model.pe
-        self.use_cache = use_cache
-        self.model = model
-        if self.use_cache:
-            self.extend_pe()
-        else:
-            self.div_term = torch.exp(
-                torch.arange(0, self.d_model, 2, dtype=torch.float32)
-                * -(math.log(10000.0) / self.d_model)
-            )
-
-    def extend_pe(self):
-        """Reset the positional encodings."""
-        pe_length = len(self.pe[0])
-        if self.max_seq_len < pe_length:
-            self.pe = self.pe[:, : self.max_seq_len]
-        else:
-            self.model.extend_pe(torch.tensor(0.0).expand(1, self.max_seq_len))
-            self.pe = self.model.pe
-
-    def _add_pe(self, x):
-        """Computes positional encoding"""
-        if self.reverse:
-            position = torch.arange(
-                x.size(1) - 1, -1, -1.0, dtype=torch.float32
-            ).unsqueeze(1)
-        else:
-            position = torch.arange(0, x.size(1), dtype=torch.float32).unsqueeze(1)
-
-        x = x * self.xscale
-        x[:, :, 0::2] += torch.sin(position * self.div_term)
-        x[:, :, 1::2] += torch.cos(position * self.div_term)
-        return x
-
-    def forward(self, x: torch.Tensor):
-        """Add positional encoding.
-
-        Args:
-            x (torch.Tensor): Input tensor (batch, time, `*`).
-
-        Returns:
-            torch.Tensor: Encoded tensor (batch, time, `*`).
-        """
-        if self.use_cache:
-            x = x * self.xscale + self.pe[:, : x.size(1)]
-        else:
-            x = self._add_pe(x)
-        return x
-
-
-class OnnxScaledPositionalEncoding(OnnxPositionalEncoding):
-    """Scaled positional encoding module.
-
-    See Sec. 3.2  https://arxiv.org/abs/1809.08895
-
-    Args:
-        d_model (int): Embedding dimension.
-        dropout_rate (float): Dropout rate.
-        max_seq_len (int): Maximum input length.
-
-    """
-
-    def __init__(self, model, max_seq_len=512, use_cache=True):
-        """Initialize class."""
-        super().__init__(model, max_seq_len, use_cache=use_cache)
-        self.alpha = torch.nn.Parameter(torch.tensor(1.0))
-
-    def reset_parameters(self):
-        """Reset parameters."""
-        self.alpha.data = torch.tensor(1.0)
-
-    def _add_pe(self, x):
-        """Computes positional encoding"""
-        if self.reverse:
-            position = torch.arange(
-                x.size(1) - 1, -1, -1.0, dtype=torch.float32
-            ).unsqueeze(1)
-        else:
-            position = torch.arange(0, x.size(1), dtype=torch.float32).unsqueeze(1)
-
-        x = x * self.alpha
-        x[:, :, 0::2] += torch.sin(position * self.div_term)
-        x[:, :, 1::2] += torch.cos(position * self.div_term)
-        return x
-
-    def forward(self, x):
-        """Add positional encoding.
-
-        Args:
-            x (torch.Tensor): Input tensor (batch, time, `*`).
-
-        Returns:
-            torch.Tensor: Encoded tensor (batch, time, `*`).
-
-        """
-        if self.use_cache:
-            x = x + self.alpha * self.pe[:, : x.size(1)]
-        else:
-            x = self._add_pe(x)
-        return x
-
-
-class OnnxLegacyRelPositionalEncoding(OnnxPositionalEncoding):
-    """Relative positional encoding module (old version).
-
-    Details can be found in https://github.com/espnet/espnet/pull/2816.
-
-    See : Appendix B in https://arxiv.org/abs/1901.02860
-
-    Args:
-        d_model (int): Embedding dimension.
-        dropout_rate (float): Dropout rate.
-        max_seq_len (int): Maximum input length.
-
-    """
-
-    def __init__(self, model, max_seq_len=512, use_cache=True):
-        """Initialize class."""
-        super().__init__(model, max_seq_len, reverse=True, use_cache=use_cache)
-
-    def _get_pe(self, x):
-        """Computes positional encoding"""
-        if self.reverse:
-            position = torch.arange(
-                x.size(1) - 1, -1, -1.0, dtype=torch.float32
-            ).unsqueeze(1)
-        else:
-            position = torch.arange(0, x.size(1), dtype=torch.float32).unsqueeze(1)
-
-        pe = torch.zeros(x.shape)
-        pe[:, :, 0::2] += torch.sin(position * self.div_term)
-        pe[:, :, 1::2] += torch.cos(position * self.div_term)
-        return pe
-
-    def forward(self, x):
-        """Compute positional encoding.
-
-        Args:
-            x (torch.Tensor): Input tensor (batch, time, `*`).
-
-        Returns:
-            torch.Tensor: Encoded tensor (batch, time, `*`).
-            torch.Tensor: Positional embedding tensor (1, time, `*`).
-
-        """
-        x = x * self.xscale
-        if self.use_cache:
-            pos_emb = self.pe[:, : x.size(1)]
-        else:
-            pos_emb = self._get_pe(x)
-        return x, pos_emb
-
-
-class OnnxRelPositionalEncoding(torch.nn.Module):
-    """Relative positional encoding module (new implementation).
-    Details can be found in https://github.com/espnet/espnet/pull/2816.
-    See : Appendix B in https://arxiv.org/abs/1901.02860
-    Args:
-        d_model (int): Embedding dimension.
-        dropout_rate (float): Dropout rate.
-        max_seq_len (int): Maximum input length.
-    """
-
-    def __init__(self, model, max_seq_len=512, use_cache=True):
-        """Construct an PositionalEncoding object."""
-        super(OnnxRelPositionalEncoding, self).__init__()
-        self.d_model = model.d_model
-        self.xscale = math.sqrt(self.d_model)
-        self.pe = None
-        self.use_cache = use_cache
-        if self.use_cache:
-            self.extend_pe(torch.tensor(0.0).expand(1, max_seq_len))
-        else:
-            self.div_term = torch.exp(
-                torch.arange(0, self.d_model, 2, dtype=torch.float32)
-                * -(math.log(10000.0) / self.d_model)
-            )
-
-    def extend_pe(self, x):
-        """Reset the positional encodings."""
-        if self.pe is not None and self.pe.size(1) >= x.size(1) * 2 - 1:
-            # self.pe contains both positive and negative parts
-            # the length of self.pe is 2 * input_len - 1
-            if self.pe.dtype != x.dtype or self.pe.device != x.device:
-                self.pe = self.pe.to(dtype=x.dtype, device=x.device)
-            return
-        # Suppose `i` means to the position of query vecotr and `j` means the
-        # position of key vector. We use position relative positions when keys
-        # are to the left (i>j) and negative relative positions otherwise (i<j).
-        pe_positive = torch.zeros(x.size(1), self.d_model)
-        pe_negative = torch.zeros(x.size(1), self.d_model)
-        position = torch.arange(0, x.size(1), dtype=torch.float32).unsqueeze(1)
-        div_term = torch.exp(
-            torch.arange(0, self.d_model, 2, dtype=torch.float32)
-            * -(math.log(10000.0) / self.d_model)
-        )
-        pe_positive[:, 0::2] = torch.sin(position * div_term)
-        pe_positive[:, 1::2] = torch.cos(position * div_term)
-        pe_negative[:, 0::2] = torch.sin(-1 * position * div_term)
-        pe_negative[:, 1::2] = torch.cos(-1 * position * div_term)
-
-        # Reserve the order of positive indices and concat both positive and
-        # negative indices. This is used to support the shifting trick
-        # as in https://arxiv.org/abs/1901.02860
-        pe_positive = torch.flip(pe_positive, [0]).unsqueeze(0)
-        pe_negative = pe_negative[1:].unsqueeze(0)
-        pe = torch.cat([pe_positive, pe_negative], dim=1)
-        self.pe = pe.to(device=x.device, dtype=x.dtype)
-
-    def _get_pe(self, x):
-        pe_positive = torch.zeros(x.size(1), self.d_model)
-        pe_negative = torch.zeros(x.size(1), self.d_model)
-        theta = (
-            torch.arange(0, x.size(1), dtype=torch.float32).unsqueeze(1) * self.div_term
-        )
-        pe_positive[:, 0::2] = torch.sin(theta)
-        pe_positive[:, 1::2] = torch.cos(theta)
-        pe_negative[:, 0::2] = -1 * torch.sin(theta)
-        pe_negative[:, 1::2] = torch.cos(theta)
-
-        # Reserve the order of positive indices and concat both positive and
-        # negative indices. This is used to support the shifting trick
-        # as in https://arxiv.org/abs/1901.02860
-        pe_positive = torch.flip(pe_positive, [0]).unsqueeze(0)
-        pe_negative = pe_negative[1:].unsqueeze(0)
-        return torch.cat([pe_positive, pe_negative], dim=1)
-
-    def forward(self, x: torch.Tensor, use_cache=True):
-        """Add positional encoding.
-        Args:
-            x (torch.Tensor): Input tensor (batch, time, `*`).
-        Returns:
-            torch.Tensor: Encoded tensor (batch, time, `*`).
-        """
-        x = x * self.xscale
-        if self.use_cache:
-            pos_emb = self.pe[
-                :,
-                self.pe.size(1) // 2 - x.size(1) + 1 : self.pe.size(1) // 2 + x.size(1),
-            ]
-        else:
-            pos_emb = self._get_pe(x)
-        return x, pos_emb
-
-
-class OnnxStreamPositionalEncoding(torch.nn.Module):
-    """Streaming Positional encoding."""
-
-    def __init__(self, model, max_seq_len=5000, use_cache=True):
-        """Construct an PositionalEncoding object."""
-        super(StreamPositionalEncoding, self).__init__()
-        self.use_cache = use_cache
-        self.d_model = model.d_model
-        self.xscale = model.xscale
-        self.pe = model.pe
-        self.use_cache = use_cache
-        self.max_seq_len = max_seq_len
-        if self.use_cache:
-            self.extend_pe()
-        else:
-            self.div_term = torch.exp(
-                torch.arange(0, self.d_model, 2, dtype=torch.float32)
-                * -(math.log(10000.0) / self.d_model)
-            )
-        self._register_load_state_dict_pre_hook(_pre_hook)
-
-    def extend_pe(self):
-        """Reset the positional encodings."""
-        pe_length = len(self.pe[0])
-        if self.max_seq_len < pe_length:
-            self.pe = self.pe[:, : self.max_seq_len]
-        else:
-            self.model.extend_pe(self.max_seq_len)
-            self.pe = self.model.pe
-
-    def _add_pe(self, x, start_idx):
-        position = torch.arange(start_idx, x.size(1), dtype=torch.float32).unsqueeze(1)
-        x = x * self.xscale
-        x[:, :, 0::2] += torch.sin(position * self.div_term)
-        x[:, :, 1::2] += torch.cos(position * self.div_term)
-        return x
-
-    def forward(self, x: torch.Tensor, start_idx: int = 0):
-        """Add positional encoding.
-
-        Args:
-            x (torch.Tensor): Input tensor (batch, time, `*`).
-
-        Returns:
-            torch.Tensor: Encoded tensor (batch, time, `*`).
-
-        """
-        if self.use_cache:
-            return x * self.xscale + self.pe[:, start_idx : start_idx + x.size(1)]
-        else:
-            return self._add_pe(x, start_idx)

funasr/export/models/language_models/seq_rnn.py

@@ -1,84 +0,0 @@
-import os
-
-import torch
-import torch.nn as nn
-
-class SequentialRNNLM(nn.Module):
-    def __init__(self, model, **kwargs):
-        super().__init__()
-        self.encoder = model.encoder
-        self.rnn = model.rnn
-        self.rnn_type = model.rnn_type
-        self.decoder = model.decoder
-        self.nlayers = model.nlayers
-        self.nhid = model.nhid
-        self.model_name = "seq_rnnlm"
-
-    def forward(self, y, hidden1, hidden2=None):
-        # batch_score function.
-        emb = self.encoder(y)
-        if self.rnn_type == "LSTM":
-            output, (hidden1, hidden2) = self.rnn(emb, (hidden1, hidden2))
-        else:
-            output, hidden1 = self.rnn(emb, hidden1)
-
-        decoded = self.decoder(
-            output.contiguous().view(output.size(0) * output.size(1), output.size(2))
-        )
-        if self.rnn_type == "LSTM":
-            return (
-                decoded.view(output.size(0), output.size(1), decoded.size(1)),
-                hidden1,
-                hidden2,
-            )
-        else:
-            return (
-                decoded.view(output.size(0), output.size(1), decoded.size(1)),
-                hidden1,
-            )
-
-    def get_dummy_inputs(self):
-        tgt = torch.LongTensor([0, 1]).unsqueeze(0)
-        hidden = torch.randn(self.nlayers, 1, self.nhid)
-        if self.rnn_type == "LSTM":
-            return (tgt, hidden, hidden)
-        else:
-            return (tgt, hidden)
-
-    def get_input_names(self):
-        if self.rnn_type == "LSTM":
-            return ["x", "in_hidden1", "in_hidden2"]
-        else:
-            return ["x", "in_hidden1"]
-
-    def get_output_names(self):
-        if self.rnn_type == "LSTM":
-            return ["y", "out_hidden1", "out_hidden2"]
-        else:
-            return ["y", "out_hidden1"]
-
-    def get_dynamic_axes(self):
-        ret = {
-            "x": {0: "x_batch", 1: "x_length"},
-            "y": {0: "y_batch"},
-            "in_hidden1": {1: "hidden1_batch"},
-            "out_hidden1": {1: "out_hidden1_batch"},
-        }
-        if self.rnn_type == "LSTM":
-            ret.update(
-                {
-                    "in_hidden2": {1: "hidden2_batch"},
-                    "out_hidden2": {1: "out_hidden2_batch"},
-                }
-            )
-        return ret
-
-    def get_model_config(self, path):
-        return {
-            "use_lm": True,
-            "model_path": os.path.join(path, f"{self.model_name}.onnx"),
-            "lm_type": "SequentialRNNLM",
-            "rnn_type": self.rnn_type,
-            "nhid": self.nhid,
-            "nlayers": self.nlayers,
-        }

funasr/export/models/language_models/subsampling.py

@@ -1,185 +0,0 @@
-"""Subsampling layer definition."""
-
-import torch
-
-
-class OnnxConv2dSubsampling(torch.nn.Module):
-    """Convolutional 2D subsampling (to 1/4 length).
-
-    Args:
-        idim (int): Input dimension.
-        odim (int): Output dimension.
-        dropout_rate (float): Dropout rate.
-        pos_enc (torch.nn.Module): Custom position encoding layer.
-
-    """
-
-    def __init__(self, model):
-        """Construct an Conv2dSubsampling object."""
-        super().__init__()
-        self.conv = model.conv
-        self.out = model.out
-
-    def forward(self, x, x_mask):
-        """Subsample x.
-
-        Args:
-            x (torch.Tensor): Input tensor (#batch, time, idim).
-            x_mask (torch.Tensor): Input mask (#batch, 1, time).
-
-        Returns:
-            torch.Tensor: Subsampled tensor (#batch, time', odim),
-                where time' = time // 4.
-            torch.Tensor: Subsampled mask (#batch, 1, time'),
-                where time' = time // 4.
-
-        """
-        x = x.unsqueeze(1)  # (b, c, t, f)
-        x = self.conv(x)
-        b, c, t, f = x.size()
-        x = self.out(x.transpose(1, 2).contiguous().view(b, t, c * f))
-        if x_mask is None:
-            return x, None
-        return x, x_mask[:, :-2:2][:, :-2:2]
-
-    def __getitem__(self, key):
-        """Get item.
-
-        When reset_parameters() is called, if use_scaled_pos_enc is used,
-            return the positioning encoding.
-
-        """
-        if key != -1:
-            raise NotImplementedError("Support only `-1` (for `reset_parameters`).")
-        return self.out[key]
-
-
-class OnnxConv2dSubsampling2(torch.nn.Module):
-    """Convolutional 2D subsampling (to 1/2 length).
-
-    Args:
-        idim (int): Input dimension.
-        odim (int): Output dimension.
-        dropout_rate (float): Dropout rate.
-        pos_enc (torch.nn.Module): Custom position encoding layer.
-
-    """
-
-    def __init__(self, model):
-        """Construct an Conv2dSubsampling object."""
-        super().__init__()
-        self.conv = model.conv
-        self.out = model.out
-
-    def forward(self, x, x_mask):
-        """Subsample x.
-
-        Args:
-            x (torch.Tensor): Input tensor (#batch, time, idim).
-            x_mask (torch.Tensor): Input mask (#batch, 1, time).
-
-        Returns:
-            torch.Tensor: Subsampled tensor (#batch, time', odim),
-                where time' = time // 2.
-            torch.Tensor: Subsampled mask (#batch, 1, time'),
-                where time' = time // 2.
-
-        """
-        x = x.unsqueeze(1)  # (b, c, t, f)
-        x = self.conv(x)
-        b, c, t, f = x.size()
-        x = self.out(x.transpose(1, 2).contiguous().view(b, t, c * f))
-        if x_mask is None:
-            return x, None
-        return x, x_mask[:, :-2:2][:, :-2:1]
-
-    def __getitem__(self, key):
-        """Get item.
-
-        When reset_parameters() is called, if use_scaled_pos_enc is used,
-            return the positioning encoding.
-
-        """
-        if key != -1:
-            raise NotImplementedError("Support only `-1` (for `reset_parameters`).")
-        return self.out[key]
-
-
-class OnnxConv2dSubsampling6(torch.nn.Module):
-    """Convolutional 2D subsampling (to 1/6 length).
-
-    Args:
-        idim (int): Input dimension.
-        odim (int): Output dimension.
-        dropout_rate (float): Dropout rate.
-        pos_enc (torch.nn.Module): Custom position encoding layer.
-
-    """
-
-    def __init__(self, model):
-        """Construct an Conv2dSubsampling object."""
-        super().__init__()
-        self.conv = model.conv
-        self.out = model.out
-
-    def forward(self, x, x_mask):
-        """Subsample x.
-
-        Args:
-            x (torch.Tensor): Input tensor (#batch, time, idim).
-            x_mask (torch.Tensor): Input mask (#batch, 1, time).
-
-        Returns:
-            torch.Tensor: Subsampled tensor (#batch, time', odim),
-                where time' = time // 6.
-            torch.Tensor: Subsampled mask (#batch, 1, time'),
-                where time' = time // 6.
-
-        """
-        x = x.unsqueeze(1)  # (b, c, t, f)
-        x = self.conv(x)
-        b, c, t, f = x.size()
-        x = self.out(x.transpose(1, 2).contiguous().view(b, t, c * f))
-        if x_mask is None:
-            return x, None
-        return x, x_mask[:, :-2:2][:, :-4:3]
-
-
-class OnnxConv2dSubsampling8(torch.nn.Module):
-    """Convolutional 2D subsampling (to 1/8 length).
-
-    Args:
-        idim (int): Input dimension.
-        odim (int): Output dimension.
-        dropout_rate (float): Dropout rate.
-        pos_enc (torch.nn.Module): Custom position encoding layer.
-
-    """
-
-    def __init__(self, model):
-        """Construct an Conv2dSubsampling object."""
-        super().__init__()
-        self.conv = model.conv
-        self.out = model.out
-
-    def forward(self, x, x_mask):
-        """Subsample x.
-
-        Args:
-            x (torch.Tensor): Input tensor (#batch, time, idim).
-            x_mask (torch.Tensor): Input mask (#batch, 1, time).
-
-        Returns:
-            torch.Tensor: Subsampled tensor (#batch, time', odim),
-                where time' = time // 8.
-            torch.Tensor: Subsampled mask (#batch, 1, time'),
-                where time' = time // 8.
-
-        """
-        x = x.unsqueeze(1)  # (b, c, t, f)
-        x = self.conv(x)
-        b, c, t, f = x.size()
-        x = self.out(x.transpose(1, 2).contiguous().view(b, t, c * f))
-        if x_mask is None:
-            return x, None
-        return x, x_mask[:, :-2:2][:, :-2:2][:, :-2:2]

funasr/export/models/language_models/transformer.py

@@ -1,110 +0,0 @@
-import os
-
-import torch
-import torch.nn as nn
-from funasr.modules.vgg2l import import VGG2L
-from funasr.modules.attention import MultiHeadedAttention
-from funasr.modules.subsampling import (
-    Conv2dSubsampling, Conv2dSubsampling6, Conv2dSubsampling8)
-
-from funasr.export.models.modules.encoder_layer import EncoderLayerConformer as OnnxEncoderLayer
-from funasr.export.models.language_models.embed import Embedding
-from funasr.export.models.modules.multihead_att import OnnxMultiHeadedAttention
-
-from funasr.export.utils.torch_function import MakePadMask
-
-class TransformerLM(nn.Module, AbsExportModel):
-    def __init__(self, model, max_seq_len=512, **kwargs):
-        super().__init__()
-        self.embed = Embedding(model.embed, max_seq_len)
-        self.encoder = model.encoder
-        self.decoder = model.decoder
-        self.make_pad_mask = MakePadMask(max_seq_len, flip=False)
-        # replace multihead attention module into customized module.
-        for i, d in enumerate(self.encoder.encoders):
-            # d is EncoderLayer
-            if isinstance(d.self_attn, MultiHeadedAttention):
-                d.self_attn = OnnxMultiHeadedAttention(d.self_attn)
-            self.encoder.encoders[i] = OnnxEncoderLayer(d)
-
-        self.model_name = "transformer_lm"
-        self.num_heads = self.encoder.encoders[0].self_attn.h
-        self.hidden_size = self.encoder.encoders[0].self_attn.linear_out.out_features
-
-    def prepare_mask(self, mask):
-        if len(mask.shape) == 2:
-            mask = mask[:, None, None, :]
-        elif len(mask.shape) == 3:
-            mask = mask[:, None, :]
-        mask = 1 - mask
-        return mask * -10000.0
-
-    def forward(self, y, cache):
-        feats_length = torch.ones(y.shape).sum(dim=-1).type(torch.long)
-        mask = self.make_pad_mask(feats_length)  # (B, T)
-        mask = (y != 0) * mask
-
-        xs = self.embed(y)
-        # forward_one_step of Encoder
-        if isinstance(
-            self.encoder.embed,
-            (Conv2dSubsampling, Conv2dSubsampling6, Conv2dSubsampling8, VGG2L),
-        ):
-            xs, mask = self.encoder.embed(xs, mask)
-        else:
-            xs = self.encoder.embed(xs)
-
-        new_cache = []
-        mask = self.prepare_mask(mask)
-        for c, e in zip(cache, self.encoder.encoders):
-            xs, mask = e(xs, mask, c)
-            new_cache.append(xs)
-
-        if self.encoder.normalize_before:
-            xs = self.encoder.after_norm(xs)
-
-        h = self.decoder(xs[:, -1])
-        return h, new_cache
-
-    def get_dummy_inputs(self):
-        tgt = torch.LongTensor([1]).unsqueeze(0)
-        cache = [
-            torch.zeros((1, 1, self.encoder.encoders[0].size))
-            for _ in range(len(self.encoder.encoders))
-        ]
-        return (tgt, cache)
-
-    def is_optimizable(self):
-        return True
-
-    def get_input_names(self):
-        return ["tgt"] + ["cache_%d" % i for i in range(len(self.encoder.encoders))]
-
-    def get_output_names(self):
-        return ["y"] + ["out_cache_%d" % i for i in range(len(self.encoder.encoders))]
-
-    def get_dynamic_axes(self):
-        ret = {"tgt": {0: "tgt_batch", 1: "tgt_length"}}
-        ret.update(
-            {
-                "cache_%d" % d: {0: "cache_%d_batch" % d, 1: "cache_%d_length" % d}
-                for d in range(len(self.encoder.encoders))
-            }
-        )
-        ret.update(
-            {
-                "out_cache_%d"
-                % d: {0: "out_cache_%d_batch" % d, 1: "out_cache_%d_length" % d}
-                for d in range(len(self.encoder.encoders))
-            }
-        )
-        return ret
-
-    def get_model_config(self, path):
-        return {
-            "use_lm": True,
-            "model_path": os.path.join(path, f"{self.model_name}.onnx"),
-            "lm_type": "TransformerLM",
-            "odim": self.encoder.encoders[0].size,
-            "nlayers": len(self.encoder.encoders),
-        }

funasr/fileio/sound_scp.py

@@ -4,7 +4,7 @@ from typing import List, Tuple, Union
 
 import random
 import numpy as np
-import soundfile
+import librosa
 import librosa
 
 import torch
@@ -116,7 +116,7 @@ class SoundScpReader(collections.abc.Mapping):
     def __getitem__(self, key):
         wav = self.data[key]
         if self.normalize:
-            # soundfile.read normalizes data to [-1,1] if dtype is not given
+            # librosa.load normalizes data to [-1,1] if dtype is not given
             array, rate = librosa.load(
                 wav, sr=self.dest_sample_rate, mono=self.always_2d
             )

funasr/layers/complex_utils.py

@@ -5,8 +5,12 @@ from typing import Tuple
 from typing import Union
 
 import torch
-from torch_complex import functional as FC
-from torch_complex.tensor import ComplexTensor
+try:
+    from torch_complex import functional as FC
+    from torch_complex.tensor import ComplexTensor
+except:
+    print("Please install torch_complex firstly")
+
 
 
 EPS = torch.finfo(torch.double).eps

funasr/layers/stft.py

@@ -4,8 +4,11 @@ from typing import Tuple
 from typing import Union
 
 import torch
-from torch_complex.tensor import ComplexTensor
 
+try:
+    from torch_complex.tensor import ComplexTensor
+except:
+    print("Please install torch_complex firstly")
 from funasr.modules.nets_utils import make_pad_mask
 from funasr.layers.complex_utils import is_complex
 from funasr.layers.inversible_interface import InversibleInterface

funasr/models/encoder/mossformer_encoder.py

@@ -1,8 +1,10 @@
 import torch
 import torch.nn as nn
 import torch.nn.functional as F
-
-from rotary_embedding_torch import RotaryEmbedding
+try:
+    from rotary_embedding_torch import RotaryEmbedding
+except:
+    print("Please install rotary_embedding_torch by: \n pip install -U rotary_embedding_torch")
 from funasr.modules.layer_norm import GlobalLayerNorm, CumulativeLayerNorm, ScaleNorm
 from funasr.modules.embedding import ScaledSinuEmbedding
 from funasr.modules.mossformer import FLASH_ShareA_FFConvM

funasr/models/frontend/default.py

@@ -6,12 +6,15 @@ import logging
 import humanfriendly
 import numpy as np
 import torch
-from torch_complex.tensor import ComplexTensor
+try:
+    from torch_complex.tensor import ComplexTensor
+except:
+    print("Please install torch_complex firstly")
 
 from funasr.layers.log_mel import LogMel
 from funasr.layers.stft import Stft
 from funasr.models.frontend.abs_frontend import AbsFrontend
-from funasr.modules.frontends.frontend import Frontend
+from funasr.models.frontend.frontends_utils.frontend import Frontend
 from funasr.utils.get_default_kwargs import get_default_kwargs
 from funasr.modules.nets_utils import make_pad_mask
 

funasr/models/frontend/frontends_utils/dnn_beamformer.py

@@ -4,12 +4,12 @@ from typing import Tuple
 import torch
 from torch.nn import functional as F
 
-from funasr.modules.frontends.beamformer import apply_beamforming_vector
-from funasr.modules.frontends.beamformer import get_mvdr_vector
-from funasr.modules.frontends.beamformer import (
+from funasr.models.frontend.frontends_utils.beamformer import apply_beamforming_vector
+from funasr.models.frontend.frontends_utils.beamformer import get_mvdr_vector
+from funasr.models.frontend.frontends_utils.beamformer import (
     get_power_spectral_density_matrix,  # noqa: H301
 )
-from funasr.modules.frontends.mask_estimator import MaskEstimator
+from funasr.models.frontend.frontends_utils.mask_estimator import MaskEstimator
 from torch_complex.tensor import ComplexTensor
 
 

funasr/models/frontend/frontends_utils/dnn_wpe.py

@@ -4,7 +4,7 @@ from pytorch_wpe import wpe_one_iteration
 import torch
 from torch_complex.tensor import ComplexTensor
 
-from funasr.modules.frontends.mask_estimator import MaskEstimator
+from funasr.models.frontend.frontends_utils.mask_estimator import MaskEstimator
 from funasr.modules.nets_utils import make_pad_mask
 
 

funasr/models/frontend/frontends_utils/frontend.py

@@ -8,8 +8,8 @@ import torch
 import torch.nn as nn
 from torch_complex.tensor import ComplexTensor
 
-from funasr.modules.frontends.dnn_beamformer import DNN_Beamformer
-from funasr.modules.frontends.dnn_wpe import DNN_WPE
+from funasr.models.frontend.frontends_utils.dnn_beamformer import DNN_Beamformer
+from funasr.models.frontend.frontends_utils.dnn_wpe import DNN_WPE
 
 
 class Frontend(nn.Module):

funasr/models/frontend/s3prl.py

@@ -10,7 +10,7 @@ import humanfriendly
 import torch
 
 from funasr.models.frontend.abs_frontend import AbsFrontend
-from funasr.modules.frontends.frontend import Frontend
+from funasr.models.frontend.frontends_utils.frontend import Frontend
 from funasr.modules.nets_utils import pad_list
 from funasr.utils.get_default_kwargs import get_default_kwargs
 

funasr/models/frontend/wav_frontend.py

@@ -145,9 +145,12 @@ class WavFrontend(AbsFrontend):
             feats_lens.append(feat_length)
 
         feats_lens = torch.as_tensor(feats_lens)
-        feats_pad = pad_sequence(feats,
-                                 batch_first=True,
-                                 padding_value=0.0)
+        if batch_size == 1:
+            feats_pad = feats[0][None, :, :]
+        else:
+            feats_pad = pad_sequence(feats,
+                                     batch_first=True,
+                                     padding_value=0.0)
         return feats_pad, feats_lens
 
     def forward_fbank(

funasr/modules/eend_ola/utils/kaldi_data.py

@@ -9,7 +9,7 @@ import os
 import sys
 import numpy as np
 import subprocess
-import soundfile as sf
+import librosa as sf
 import io
 from functools import lru_cache
 
@@ -67,18 +67,18 @@ def load_wav(wav_rxfilename, start=0, end=None):
         # input piped command
         p = subprocess.Popen(wav_rxfilename[:-1], shell=True,
                              stdout=subprocess.PIPE)
-        data, samplerate = sf.read(io.BytesIO(p.stdout.read()),
+        data, samplerate = sf.load(io.BytesIO(p.stdout.read()),
                                    dtype='float32')
         # cannot seek
         data = data[start:end]
     elif wav_rxfilename == '-':
         # stdin
-        data, samplerate = sf.read(sys.stdin, dtype='float32')
+        data, samplerate = sf.load(sys.stdin, dtype='float32')
         # cannot seek
         data = data[start:end]
     else:
         # normal wav file
-        data, samplerate = sf.read(wav_rxfilename, start=start, stop=end)
+        data, samplerate = sf.load(wav_rxfilename, start=start, stop=end)
     return data, samplerate
 
 

funasr/quick_start.md

@@ -16,7 +16,7 @@ Supports real-time streaming speech recognition, uses non-streaming models for e
 #### Server Deployment
 
 ```shell
-cd funasr/runtime/python/websocket
+cd runtime/python/websocket
 python funasr_wss_server.py --port 10095
 ```
 
@@ -161,4 +161,4 @@ If you want to train from scratch, usually for academic models, you can start tr
 cd egs/aishell/paraformer
 . ./run.sh --CUDA_VISIBLE_DEVICES="0,1" --gpu_num=2
 ```
-More examples could be found in [docs](https://alibaba-damo-academy.github.io/FunASR/en/modelscope_pipeline/quick_start.html)
+More examples could be found in [docs](https://alibaba-damo-academy.github.io/FunASR/en/modelscope_pipeline/quick_start.html)

funasr/quick_start_zh.md

@@ -17,7 +17,7 @@
 
 ##### 服务端部署
 ```shell
-cd funasr/runtime/python/websocket
+cd runtime/python/websocket
 python funasr_wss_server.py --port 10095
 ```
 
@@ -161,4 +161,4 @@ cd egs/aishell/paraformer
 . ./run.sh --CUDA_VISIBLE_DEVICES="0,1" --gpu_num=2
 ```
 
-更多例子可以参考（[点击此处](https://alibaba-damo-academy.github.io/FunASR/en/academic_recipe/asr_recipe.html)）
+更多例子可以参考（[点击此处](https://alibaba-damo-academy.github.io/FunASR/en/academic_recipe/asr_recipe.html)）

funasr/tasks/asr.py

@@ -76,7 +76,7 @@ from funasr.models.specaug.specaug import SpecAug
 from funasr.models.specaug.specaug import SpecAugLFR
 from funasr.modules.subsampling import Conv1dSubsampling
 from funasr.tasks.abs_task import AbsTask
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.torch_utils.initialize import initialize
 from funasr.models.base_model import FunASRModel
 from funasr.train.class_choices import ClassChoices

funasr/tasks/data2vec.py

@@ -25,7 +25,7 @@ from funasr.models.preencoder.sinc import LightweightSincConvs
 from funasr.models.specaug.abs_specaug import AbsSpecAug
 from funasr.models.specaug.specaug import SpecAug
 from funasr.tasks.abs_task import AbsTask
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.torch_utils.initialize import initialize
 from funasr.train.class_choices import ClassChoices
 from funasr.train.trainer import Trainer

funasr/tasks/lm.py

@@ -17,7 +17,7 @@ from funasr.train.abs_model import LanguageModel
 from funasr.models.seq_rnn_lm import SequentialRNNLM
 from funasr.models.transformer_lm import TransformerLM
 from funasr.tasks.abs_task import AbsTask
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.torch_utils.initialize import initialize
 from funasr.train.class_choices import ClassChoices
 from funasr.train.trainer import Trainer

funasr/tasks/punctuation.py

@@ -16,7 +16,7 @@ from funasr.train.abs_model import PunctuationModel
 from funasr.models.target_delay_transformer import TargetDelayTransformer
 from funasr.models.vad_realtime_transformer import VadRealtimeTransformer
 from funasr.tasks.abs_task import AbsTask
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.torch_utils.initialize import initialize
 from funasr.train.class_choices import ClassChoices
 from funasr.train.trainer import Trainer

funasr/tasks/sa_asr.py

@@ -71,7 +71,7 @@ from funasr.models.specaug.specaug import SpecAugLFR
 from funasr.models.base_model import FunASRModel
 from funasr.modules.subsampling import Conv1dSubsampling
 from funasr.tasks.abs_task import AbsTask
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.torch_utils.initialize import initialize
 from funasr.train.class_choices import ClassChoices
 from funasr.train.trainer import Trainer

funasr/tasks/whisper.py

@@ -76,7 +76,7 @@ from funasr.models.specaug.specaug import SpecAug
 from funasr.models.specaug.specaug import SpecAugLFR
 from funasr.modules.subsampling import Conv1dSubsampling
 from funasr.tasks.abs_task import AbsTask
-from funasr.text.phoneme_tokenizer import g2p_choices
+from funasr.tokenizer.phoneme_tokenizer import g2p_choices
 from funasr.torch_utils.initialize import initialize
 from funasr.models.base_model import FunASRModel
 from funasr.train.class_choices import ClassChoices

funasr/tokenizer/build_tokenizer.py

@@ -3,11 +3,11 @@ from typing import Iterable
 from typing import Union
 
 
-from funasr.text.abs_tokenizer import AbsTokenizer
-from funasr.text.char_tokenizer import CharTokenizer
-from funasr.text.phoneme_tokenizer import PhonemeTokenizer
-from funasr.text.sentencepiece_tokenizer import SentencepiecesTokenizer
-from funasr.text.word_tokenizer import WordTokenizer
+from funasr.tokenizer.abs_tokenizer import AbsTokenizer
+from funasr.tokenizer.char_tokenizer import CharTokenizer
+from funasr.tokenizer.phoneme_tokenizer import PhonemeTokenizer
+from funasr.tokenizer.sentencepiece_tokenizer import SentencepiecesTokenizer
+from funasr.tokenizer.word_tokenizer import WordTokenizer
 
 
 def build_tokenizer(

funasr/tokenizer/char_tokenizer.py

@@ -5,7 +5,7 @@ from typing import Union
 import warnings
 
 
-from funasr.text.abs_tokenizer import AbsTokenizer
+from funasr.tokenizer.abs_tokenizer import AbsTokenizer
 
 
 class CharTokenizer(AbsTokenizer):

funasr/tokenizer/phoneme_tokenizer.py

@@ -10,7 +10,7 @@ import warnings
 # import g2p_en
 import jamo
 
-from funasr.text.abs_tokenizer import AbsTokenizer
+from funasr.tokenizer.abs_tokenizer import AbsTokenizer
 
 
 g2p_choices = [
@@ -107,7 +107,7 @@ def pyopenjtalk_g2p_prosody(text: str, drop_unvoiced_vowels: bool = True) -> Lis
         List[str]: List of phoneme + prosody symbols.
 
     Examples:
-        >>> from funasr.text.phoneme_tokenizer import pyopenjtalk_g2p_prosody
+        >>> from funasr.tokenizer.phoneme_tokenizer import pyopenjtalk_g2p_prosody
         >>> pyopenjtalk_g2p_prosody("こんにちは。")
         ['^', 'k', 'o', '[', 'N', 'n', 'i', 'ch', 'i', 'w', 'a', '$']
 

funasr/tokenizer/sentencepiece_tokenizer.py

@@ -5,7 +5,7 @@ from typing import Union
 
 import sentencepiece as spm
 
-from funasr.text.abs_tokenizer import AbsTokenizer
+from funasr.tokenizer.abs_tokenizer import AbsTokenizer
 
 
 class SentencepiecesTokenizer(AbsTokenizer):

funasr/tokenizer/word_tokenizer.py

@@ -5,7 +5,7 @@ from typing import Union
 import warnings
 
 
-from funasr.text.abs_tokenizer import AbsTokenizer
+from funasr.tokenizer.abs_tokenizer import AbsTokenizer
 
 
 class WordTokenizer(AbsTokenizer):

funasr/train/trainer.py

@@ -278,14 +278,11 @@ class Trainer:
         for iepoch in range(start_epoch, trainer_options.max_epoch + 1):
             if iepoch != start_epoch:
                 logging.info(
-                    "{}/{}epoch started. Estimated time to finish: {}".format(
+                    "{}/{}epoch started. Estimated time to finish: {} hours".format(
                         iepoch,
                         trainer_options.max_epoch,
-                        humanfriendly.format_timespan(
-                            (time.perf_counter() - start_time)
-                            / (iepoch - start_epoch)
-                            * (trainer_options.max_epoch - iepoch + 1)
-                        ),
+                        (time.perf_counter() - start_time) / 3600.0 / (iepoch - start_epoch) * (
+                                trainer_options.max_epoch - iepoch + 1),
                     )
                 )
             else:

funasr/utils/asr_utils.py

@@ -5,7 +5,7 @@ import struct
 from typing import Any, Dict, List, Union
 
 import torchaudio
-import soundfile
+import librosa
 import numpy as np
 import pkg_resources
 from modelscope.utils.logger import get_logger
@@ -139,7 +139,7 @@ def get_sr_from_wav(fname: str):
                     try:
                         audio, fs = torchaudio.load(fname)
                     except:
-                        audio, fs = soundfile.read(fname)
+                        audio, fs = librosa.load(fname)
                 break
         if audio_type.rfind(".scp") >= 0:
             with open(fname, encoding="utf-8") as f:

funasr/utils/prepare_data.py

@@ -5,7 +5,7 @@ from multiprocessing import Pool
 
 import kaldiio
 import numpy as np
-import soundfile
+import librosa
 import torch.distributed as dist
 import torchaudio
 
@@ -46,7 +46,7 @@ def wav2num_frame(wav_path, frontend_conf):
     try:
         waveform, sampling_rate = torchaudio.load(wav_path)
     except:
-        waveform, sampling_rate = soundfile.read(wav_path)
+        waveform, sampling_rate = librosa.load(wav_path)
         waveform = np.expand_dims(waveform, axis=0)
     n_frames = (waveform.shape[1] * 1000.0) / (sampling_rate * frontend_conf["frame_shift"] * frontend_conf["lfr_n"])
     feature_dim = frontend_conf["n_mels"] * frontend_conf["lfr_m"]

funasr/utils/speaker_utils.py

@@ -12,7 +12,7 @@ import os
 from typing import Any, Dict, List, Union
 
 import numpy as np
-import soundfile as sf
+import librosa as sf
 import torch
 import torchaudio
 import logging
@@ -43,7 +43,7 @@ def sv_preprocess(inputs: Union[np.ndarray, list]):
         for i in range(len(inputs)):
             if isinstance(inputs[i], str):
                 file_bytes = File.read(inputs[i])
-                data, fs = sf.read(io.BytesIO(file_bytes), dtype='float32')
+                data, fs = sf.load(io.BytesIO(file_bytes), dtype='float32')
                 if len(data.shape) == 2:
                     data = data[:, 0]
                 data = torch.from_numpy(data).unsqueeze(0)

funasr/utils/timestamp_tools.py

@@ -3,7 +3,7 @@ import codecs
 import logging
 import argparse
 import numpy as np
-import edit_distance
+# import edit_distance
 from itertools import zip_longest
 
 
@@ -160,112 +160,112 @@ def time_stamp_sentence(punc_id_list, time_stamp_postprocessed, text_postprocess
     return res
 
 
-class AverageShiftCalculator():
-    def __init__(self):
-        logging.warning("Calculating average shift.")
-    def __call__(self, file1, file2):
-        uttid_list1, ts_dict1 = self.read_timestamps(file1)
-        uttid_list2, ts_dict2 = self.read_timestamps(file2)
-        uttid_intersection = self._intersection(uttid_list1, uttid_list2)
-        res = self.as_cal(uttid_intersection, ts_dict1, ts_dict2)
-        logging.warning("Average shift of {} and {}: {}.".format(file1, file2, str(res)[:8]))
-        logging.warning("Following timestamp pair differs most: {}, detail:{}".format(self.max_shift, self.max_shift_uttid))
-
-    def _intersection(self, list1, list2):
-        set1 = set(list1)
-        set2 = set(list2)
-        if set1 == set2:
-            logging.warning("Uttid same checked.")
-            return set1
-        itsc = list(set1 & set2)
-        logging.warning("Uttid differs: file1 {}, file2 {}, lines same {}.".format(len(list1), len(list2), len(itsc)))
-        return itsc
-
-    def read_timestamps(self, file):
-        # read timestamps file in standard format
-        uttid_list = []
-        ts_dict = {}
-        with codecs.open(file, 'r') as fin:
-            for line in fin.readlines():
-                text = ''
-                ts_list = []
-                line = line.rstrip()
-                uttid = line.split()[0]
-                uttid_list.append(uttid)
-                body = " ".join(line.split()[1:])
-                for pd in body.split(';'):
-                    if not len(pd): continue
-                    # pdb.set_trace() 
-                    char, start, end = pd.lstrip(" ").split(' ')
-                    text += char + ','
-                    ts_list.append((float(start), float(end)))
-                # ts_lists.append(ts_list)
-                ts_dict[uttid] = (text[:-1], ts_list)
-        logging.warning("File {} read done.".format(file))
-        return uttid_list, ts_dict
-
-    def _shift(self, filtered_timestamp_list1, filtered_timestamp_list2):
-        shift_time = 0
-        for fts1, fts2 in zip(filtered_timestamp_list1, filtered_timestamp_list2):
-            shift_time += abs(fts1[0] - fts2[0]) + abs(fts1[1] - fts2[1])
-        num_tokens = len(filtered_timestamp_list1)
-        return shift_time, num_tokens
-
-    def as_cal(self, uttid_list, ts_dict1, ts_dict2):
-        # calculate average shift between timestamp1 and timestamp2
-        # when characters differ, use edit distance alignment
-        # and calculate the error between the same characters
-        self._accumlated_shift = 0
-        self._accumlated_tokens = 0
-        self.max_shift = 0
-        self.max_shift_uttid = None
-        for uttid in uttid_list:
-            (t1, ts1) = ts_dict1[uttid]
-            (t2, ts2) = ts_dict2[uttid]
-            _align, _align2, _align3 = [], [], []
-            fts1, fts2 = [], []
-            _t1, _t2 = [], []
-            sm = edit_distance.SequenceMatcher(t1.split(','), t2.split(','))
-            s = sm.get_opcodes()
-            for j in range(len(s)):
-                if s[j][0] == "replace" or s[j][0] == "insert":
-                    _align.append(0)
-                if s[j][0] == "replace" or s[j][0] == "delete":
-                    _align3.append(0)
-                elif s[j][0] == "equal":
-                    _align.append(1)
-                    _align3.append(1)
-                else:
-                    continue
-            # use s to index t2
-            for a, ts , t in zip(_align, ts2, t2.split(',')):
-                if a: 
-                    fts2.append(ts)
-                    _t2.append(t)
-            sm2 = edit_distance.SequenceMatcher(t2.split(','), t1.split(','))
-            s = sm2.get_opcodes()
-            for j in range(len(s)):
-                if s[j][0] == "replace" or s[j][0] == "insert":
-                    _align2.append(0)
-                elif s[j][0] == "equal":
-                    _align2.append(1)
-                else:
-                    continue
-            # use s2 tp index t1
-            for a, ts, t in zip(_align3, ts1, t1.split(',')):
-                if a: 
-                    fts1.append(ts)
-                    _t1.append(t)
-            if len(fts1) == len(fts2):
-                shift_time, num_tokens = self._shift(fts1, fts2)
-                self._accumlated_shift += shift_time
-                self._accumlated_tokens += num_tokens
-                if shift_time/num_tokens > self.max_shift:
-                    self.max_shift = shift_time/num_tokens
-                    self.max_shift_uttid = uttid
-            else:
-                logging.warning("length mismatch")
-        return self._accumlated_shift / self._accumlated_tokens
+# class AverageShiftCalculator():
+#     def __init__(self):
+#         logging.warning("Calculating average shift.")
+#     def __call__(self, file1, file2):
+#         uttid_list1, ts_dict1 = self.read_timestamps(file1)
+#         uttid_list2, ts_dict2 = self.read_timestamps(file2)
+#         uttid_intersection = self._intersection(uttid_list1, uttid_list2)
+#         res = self.as_cal(uttid_intersection, ts_dict1, ts_dict2)
+#         logging.warning("Average shift of {} and {}: {}.".format(file1, file2, str(res)[:8]))
+#         logging.warning("Following timestamp pair differs most: {}, detail:{}".format(self.max_shift, self.max_shift_uttid))
+#
+#     def _intersection(self, list1, list2):
+#         set1 = set(list1)
+#         set2 = set(list2)
+#         if set1 == set2:
+#             logging.warning("Uttid same checked.")
+#             return set1
+#         itsc = list(set1 & set2)
+#         logging.warning("Uttid differs: file1 {}, file2 {}, lines same {}.".format(len(list1), len(list2), len(itsc)))
+#         return itsc
+#
+#     def read_timestamps(self, file):
+#         # read timestamps file in standard format
+#         uttid_list = []
+#         ts_dict = {}
+#         with codecs.open(file, 'r') as fin:
+#             for line in fin.readlines():
+#                 text = ''
+#                 ts_list = []
+#                 line = line.rstrip()
+#                 uttid = line.split()[0]
+#                 uttid_list.append(uttid)
+#                 body = " ".join(line.split()[1:])
+#                 for pd in body.split(';'):
+#                     if not len(pd): continue
+#                     # pdb.set_trace()
+#                     char, start, end = pd.lstrip(" ").split(' ')
+#                     text += char + ','
+#                     ts_list.append((float(start), float(end)))
+#                 # ts_lists.append(ts_list)
+#                 ts_dict[uttid] = (text[:-1], ts_list)
+#         logging.warning("File {} read done.".format(file))
+#         return uttid_list, ts_dict
+#
+#     def _shift(self, filtered_timestamp_list1, filtered_timestamp_list2):
+#         shift_time = 0
+#         for fts1, fts2 in zip(filtered_timestamp_list1, filtered_timestamp_list2):
+#             shift_time += abs(fts1[0] - fts2[0]) + abs(fts1[1] - fts2[1])
+#         num_tokens = len(filtered_timestamp_list1)
+#         return shift_time, num_tokens
+#
+#     # def as_cal(self, uttid_list, ts_dict1, ts_dict2):
+#     #     # calculate average shift between timestamp1 and timestamp2
+#     #     # when characters differ, use edit distance alignment
+#     #     # and calculate the error between the same characters
+#     #     self._accumlated_shift = 0
+#     #     self._accumlated_tokens = 0
+#     #     self.max_shift = 0
+#     #     self.max_shift_uttid = None
+#     #     for uttid in uttid_list:
+#     #         (t1, ts1) = ts_dict1[uttid]
+#     #         (t2, ts2) = ts_dict2[uttid]
+#     #         _align, _align2, _align3 = [], [], []
+#     #         fts1, fts2 = [], []
+#     #         _t1, _t2 = [], []
+#     #         sm = edit_distance.SequenceMatcher(t1.split(','), t2.split(','))
+#     #         s = sm.get_opcodes()
+#     #         for j in range(len(s)):
+#     #             if s[j][0] == "replace" or s[j][0] == "insert":
+#     #                 _align.append(0)
+#     #             if s[j][0] == "replace" or s[j][0] == "delete":
+#     #                 _align3.append(0)
+#     #             elif s[j][0] == "equal":
+#     #                 _align.append(1)
+#     #                 _align3.append(1)
+#     #             else:
+#     #                 continue
+#     #         # use s to index t2
+#     #         for a, ts , t in zip(_align, ts2, t2.split(',')):
+#     #             if a:
+#     #                 fts2.append(ts)
+#     #                 _t2.append(t)
+#     #         sm2 = edit_distance.SequenceMatcher(t2.split(','), t1.split(','))
+#     #         s = sm2.get_opcodes()
+#     #         for j in range(len(s)):
+#     #             if s[j][0] == "replace" or s[j][0] == "insert":
+#     #                 _align2.append(0)
+#     #             elif s[j][0] == "equal":
+#     #                 _align2.append(1)
+#     #             else:
+#     #                 continue
+#     #         # use s2 tp index t1
+#     #         for a, ts, t in zip(_align3, ts1, t1.split(',')):
+#     #             if a:
+#     #                 fts1.append(ts)
+#     #                 _t1.append(t)
+#     #         if len(fts1) == len(fts2):
+#     #             shift_time, num_tokens = self._shift(fts1, fts2)
+#     #             self._accumlated_shift += shift_time
+#     #             self._accumlated_tokens += num_tokens
+#     #             if shift_time/num_tokens > self.max_shift:
+#     #                 self.max_shift = shift_time/num_tokens
+#     #                 self.max_shift_uttid = uttid
+#     #         else:
+#     #             logging.warning("length mismatch")
+#     #     return self._accumlated_shift / self._accumlated_tokens
 
 
 def convert_external_alphas(alphas_file, text_file, output_file):
@@ -311,10 +311,10 @@ SUPPORTED_MODES = ['cal_aas', 'read_ext_alphas']
 
 
 def main(args):
-    if args.mode == 'cal_aas':
-        asc = AverageShiftCalculator()
-        asc(args.input, args.input2)
-    elif args.mode == 'read_ext_alphas':
+    # if args.mode == 'cal_aas':
+    #     asc = AverageShiftCalculator()
+    #     asc(args.input, args.input2)
+    if args.mode == 'read_ext_alphas':
         convert_external_alphas(args.input, args.input2, args.output)
     else:
         logging.error("Mode {} not in SUPPORTED_MODES: {}.".format(args.mode, SUPPORTED_MODES))

funasr/utils/wav_utils.py

@@ -11,7 +11,7 @@ import librosa
 import numpy as np
 import torch
 import torchaudio
-import soundfile
+import librosa
 import torchaudio.compliance.kaldi as kaldi
 
 
@@ -166,7 +166,7 @@ def compute_fbank(wav_file,
         try:
             waveform, audio_sr = torchaudio.load(wav_file)
         except:
-            waveform, audio_sr = soundfile.read(wav_file, dtype='float32')
+            waveform, audio_sr = librosa.load(wav_file, dtype='float32')
             if waveform.ndim == 2:
                 waveform = waveform[:, 0]
             waveform = torch.tensor(np.expand_dims(waveform, axis=0))
@@ -191,7 +191,7 @@ def wav2num_frame(wav_path, frontend_conf):
     try:
         waveform, sampling_rate = torchaudio.load(wav_path)
     except:
-        waveform, sampling_rate = soundfile.read(wav_path)
+        waveform, sampling_rate = librosa.load(wav_path)
         waveform = torch.tensor(np.expand_dims(waveform, axis=0))
     speech_length = (waveform.shape[1] / sampling_rate) * 1000.
     n_frames = (waveform.shape[1] * 1000.0) / (sampling_rate * frontend_conf["frame_shift"] * frontend_conf["lfr_n"])

funasr/utils/whisper_utils/audio.py

@@ -1,8 +1,11 @@
 import os
 from functools import lru_cache
 from typing import Union
+try:
+    import ffmpeg
+except:
+    print("Please Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.")
 
-import ffmpeg
 import numpy as np
 import torch
 import torch.nn.functional as F

funasr/version.txt

@@ -1 +1 @@
-0.8.5
+0.8.6

runtime/docs/SDK_advanced_guide_offline.md

@@ -94,7 +94,9 @@ Introduction to run_server.sh parameters:
 --punc-quant: True for quantized PUNC model, False for non-quantized PUNC model. Default is True.
 --itn-dir modelscope model ID or local model path.
 --port: Port number that the server listens on. Default is 10095.
---decoder-thread-num: Number of inference threads that the server starts. Default is 8.
+--decoder-thread-num: The number of thread pools on the server side that can handle concurrent requests. The default value is 8.
+--model-thread-num: The number of internal threads for each recognition route to control the parallelism of the ONNX model. 
+        The default value is 1. It is recommended that decoder-thread-num * model-thread-num equals the total number of threads.
 --io-thread-num: Number of IO threads that the server starts. Default is 1.
 --certfile <string>: SSL certificate file. Default is ../../../ssl_key/server.crt. If you want to close ssl，set 0
 --keyfile <string>: SSL key file. Default is ../../../ssl_key/server.key. 

runtime/docs/SDK_advanced_guide_offline_en.md

@@ -73,7 +73,9 @@ Introduction to run_server.sh parameters:
 --punc-quant: True for quantized PUNC model, False for non-quantized PUNC model. Default is True.
 --itn-dir modelscope model ID or local model path.
 --port: Port number that the server listens on. Default is 10095.
---decoder-thread-num: Number of inference threads that the server starts. Default is 8.
+--decoder-thread-num: The number of thread pools on the server side that can handle concurrent requests. The default value is 8.
+--model-thread-num: The number of internal threads for each recognition route to control the parallelism of the ONNX model. 
+        The default value is 1. It is recommended that decoder-thread-num * model-thread-num equals the total number of threads.
 --io-thread-num: Number of IO threads that the server starts. Default is 1.
 --certfile <string>: SSL certificate file. Default is ../../../ssl_key/server.crt. If you want to close ssl，set 0
 --keyfile <string>: SSL key file. Default is ../../../ssl_key/server.key. 

runtime/docs/SDK_advanced_guide_offline_en_zh.md

@@ -158,7 +158,9 @@ nohup bash run_server.sh \
 --punc-quant   True为量化PUNC模型，False为非量化PUNC模型，默认是True
 --itn-dir modelscope model ID 或者 本地模型路径
 --port  服务端监听的端口号，默认为 10095
---decoder-thread-num  服务端启动的推理线程数，默认为 8
+--decoder-thread-num  服务端线程池个数(支持的最大并发路数)，默认为 8
+--model-thread-num  每路识别的内部线程数(控制ONNX模型的并行)，默认为 1，
+                    其中建议 decoder-thread-num*model-thread-num 等于总线程数
 --io-thread-num  服务端启动的IO线程数，默认为 1
 --certfile  ssl的证书文件，默认为：../../../ssl_key/server.crt，如果需要关闭ssl，参数设置为0
 --keyfile   ssl的密钥文件，默认为：../../../ssl_key/server.key

runtime/docs/SDK_advanced_guide_offline_zh.md

@@ -175,11 +175,14 @@ nohup bash run_server.sh \
 --lm-dir modelscope model ID 或者 本地模型路径
 --itn-dir modelscope model ID 或者 本地模型路径
 --port  服务端监听的端口号，默认为 10095
---decoder-thread-num  服务端启动的推理线程数，默认为 8
+--decoder-thread-num  服务端线程池个数(支持的最大并发路数)，默认为 8
+--model-thread-num  每路识别的内部线程数(控制ONNX模型的并行)，默认为 1，
+                    其中建议 decoder-thread-num*model-thread-num 等于总线程数
 --io-thread-num  服务端启动的IO线程数，默认为 1
 --certfile  ssl的证书文件，默认为：../../../ssl_key/server.crt，如果需要关闭ssl，参数设置为0
 --keyfile   ssl的密钥文件，默认为：../../../ssl_key/server.key
---hotword   热词文件路径，每行一个热词，格式：热词 权重(例如:阿里巴巴 20)，如果客户端提供热词，则与客户端提供的热词合并一起使用。
+--hotword   热词文件路径，每行一个热词，格式：热词 权重(例如:阿里巴巴 20)，
+            如果客户端提供热词，则与客户端提供的热词合并一起使用，服务端热词全局生效，客户端热词只针对对应客户端生效。
 ```
 
 ### 关闭FunASR服务

runtime/docs/SDK_advanced_guide_online.md

@@ -111,7 +111,9 @@ nohup bash run_server_2pass.sh \
 --punc-quant: True for quantized PUNC model, False for non-quantized PUNC model. Default is True.
 --itn-dir modelscope model ID or local model path.
 --port: Port number that the server listens on. Default is 10095.
---decoder-thread-num: Number of inference threads that the server starts. Default is 8.
+--decoder-thread-num: The number of thread pools on the server side that can handle concurrent requests. The default value is 8.
+--model-thread-num: The number of internal threads for each recognition route to control the parallelism of the ONNX model. 
+        The default value is 1. It is recommended that decoder-thread-num * model-thread-num equals the total number of threads.
 --io-thread-num: Number of IO threads that the server starts. Default is 1.
 --certfile <string>: SSL certificate file. Default is ../../../ssl_key/server.crt. If you want to close ssl，set 0
 --keyfile <string>: SSL key file. Default is ../../../ssl_key/server.key. 

runtime/docs/SDK_advanced_guide_online_zh.md

@@ -120,11 +120,14 @@ nohup bash run_server_2pass.sh \
 --punc-quant   True为量化PUNC模型，False为非量化PUNC模型，默认是True
 --itn-dir modelscope model ID 或者 本地模型路径
 --port  服务端监听的端口号，默认为 10095
---decoder-thread-num  服务端启动的推理线程数，默认为 8
+--decoder-thread-num  服务端线程池个数(支持的最大并发路数)，默认为 8
+--model-thread-num  每路识别的内部线程数(控制ONNX模型的并行)，默认为 1，
+                    其中建议 decoder-thread-num*model-thread-num 等于总线程数
 --io-thread-num  服务端启动的IO线程数，默认为 1
 --certfile  ssl的证书文件，默认为：../../../ssl_key/server.crt，如果需要关闭ssl，参数设置为0
 --keyfile   ssl的密钥文件，默认为：../../../ssl_key/server.key
---hotword   热词文件路径，每行一个热词，格式：热词 权重(例如:阿里巴巴 20)，如果客户端提供热词，则与客户端提供的热词合并一起使用。
+--hotword   热词文件路径，每行一个热词，格式：热词 权重(例如:阿里巴巴 20)，
+            如果客户端提供热词，则与客户端提供的热词合并一起使用，服务端热词全局生效，客户端热词只针对对应客户端生效。
 ```
 
 ### 关闭FunASR服务

runtime/python/websocket/README.md

@@ -16,7 +16,7 @@ git clone https://github.com/alibaba/FunASR.git && cd FunASR
 ### Install the requirements for server
 
 ```shell
-cd funasr/runtime/python/websocket
+cd runtime/python/websocket
 pip install -r requirements_server.txt
 ```
 

runtime/python/websocket/funasr_wss_server.py

@@ -53,13 +53,13 @@ parser.add_argument("--ncpu",
                     help="cpu cores")
 parser.add_argument("--certfile",
                     type=str,
-                    default="../ssl_key/server.crt",
+                    default="../../ssl_key/server.crt",
                     required=False,
                     help="certfile for ssl")
 
 parser.add_argument("--keyfile",
                     type=str,
-                    default="../ssl_key/server.key",
+                    default="../../ssl_key/server.key",
                     required=False,
                     help="keyfile for ssl")
 args = parser.parse_args()

setup.py

@@ -10,36 +10,36 @@ from setuptools import setup
 
 requirements = {
     "install": [
-        "setuptools>=38.5.1",
+        # "setuptools>=38.5.1",
         "humanfriendly",
         "scipy>=1.4.1",
         "librosa",
-        "jamo",  # For kss
+        # "jamo",  # For kss
         "PyYAML>=5.1.2",
-        "soundfile>=0.12.1",
-        "h5py>=3.1.0",
+        # "soundfile>=0.12.1",
+        # "h5py>=3.1.0",
         "kaldiio>=2.17.0",
-        "torch_complex",
-        "nltk>=3.4.5",
+        # "torch_complex",
+        # "nltk>=3.4.5",
         # ASR
-        "sentencepiece",
+        "sentencepiece", # train
         "jieba",
-        "rotary_embedding_torch",
-        "ffmpeg",
+        # "rotary_embedding_torch",
+        # "ffmpeg-python",
         # TTS
-        "pypinyin>=0.44.0",
-        "espnet_tts_frontend",
+        # "pypinyin>=0.44.0",
+        # "espnet_tts_frontend",
         # ENH
-        "pytorch_wpe",
+        # "pytorch_wpe",
         "editdistance>=0.5.2",
         "tensorboard",
-        "g2p",
-        "nara_wpe",
+        # "g2p",
+        # "nara_wpe",
         # PAI
         "oss2",
-        "edit-distance",
-        "textgrid",
-        "protobuf",
+        # "edit-distance",
+        # "textgrid",
+        # "protobuf",
         "tqdm",
         "hdbscan",
         "umap",
@@ -104,7 +104,7 @@ setup(
     name="funasr",
     version=version,
     url="https://github.com/alibaba-damo-academy/FunASR.git",
-    author="Speech Lab of DAMO Academy, Alibaba Group",
+    author="Speech Lab of Alibaba Group",
     author_email="funasr@list.alibaba-inc.com",
     description="FunASR: A Fundamental End-to-End Speech Recognition Toolkit",
     long_description=open(os.path.join(dirname, "README.md"), encoding="utf-8").read(),

web-pages/.browserslistrc

@@ -0,0 +1,3 @@
+> 1%
+last 2 versions
+not ie < 11

web-pages/.editorconfig

@@ -0,0 +1,7 @@
+[*.{js,jsx,ts,tsx,vue}]
+charset = utf-8
+indent_style = space
+indent_size = 4
+end_of_line = lf
+trim_trailing_whitespace = true
+insert_final_newline = true

web-pages/.eslintignore

@@ -0,0 +1,7 @@
+node_modules
+dist
+fonts
+*.md
+*.woff
+*.ttf
+public

web-pages/.eslintrc.js

@@ -0,0 +1,24 @@
+module.exports = {
+    root: true,
+    env: {
+        node: true
+    },
+    extends: ['plugin:vue/essential', '@vue/standard'],
+    parserOptions: {
+        parser: '@babel/eslint-parser'
+    },
+    rules: {
+        'no-console': process.env.NODE_ENV === 'production' ? 'warn' : 'off',
+        'no-debugger': process.env.NODE_ENV === 'production' ? 'warn' : 'off',
+        'quote-props': 'off',
+        indent: 'off',
+        // "vue/script-indent": [
+        //     "error",
+        //     4,
+        //     {
+        //         baseIndent: 1,
+        //     },
+        // ],
+        'vue/order-in-components': 'error'
+    }
+}

web-pages/babel.config.js

@@ -0,0 +1,13 @@
+module.exports = {
+    presets: ['@vue/cli-plugin-babel/preset'],
+    plugins: [
+        [
+            'import',
+            {
+                libraryName: 'ant-design-vue',
+                libraryDirectory: 'es',
+                style: 'css'
+            }
+        ]
+    ]
+}

web-pages/jsconfig.json

@@ -0,0 +1,12 @@
+{
+	"compilerOptions": {
+		"target": "es5",
+		"module": "esnext",
+		"baseUrl": "./",
+		"moduleResolution": "node",
+		"paths": {
+			"@/*": ["src/*"]
+		},
+		"lib": ["esnext", "dom", "dom.iterable", "scripthost"]
+	}
+}

web-pages/mock/index.js

@@ -0,0 +1,11 @@
+import Mock from 'mockjs'
+
+import getUserInfo from './user/getUserInfo.js'
+import getMenuList from './user/getMenuList.js'
+Mock.setup({
+    timeout: 500
+})
+
+console.log('启动mock请求数据')
+getUserInfo(Mock)
+getMenuList(Mock)

web-pages/mock/user/getMenuList.js

@@ -0,0 +1 @@
+export default Mock => {}

web-pages/mock/user/getUserInfo.js

@@ -0,0 +1 @@
+export default Mock => {}

web-pages/mock/util.js

@@ -0,0 +1,3 @@
+export function randomNum (n, m) {
+    return Math.floor(Math.random() * (m - n + 1) + n)
+}

web-pages/package.json

@@ -0,0 +1,44 @@
+{
+    "name": "template-vue",
+    "version": "0.1.0",
+    "private": true,
+    "scripts": {
+        "dev": "vue-cli-service serve --env VUE_APP_frontendConfigUrl=/template-vue/frontend-config-dev.json",
+        "devmock": "vue-cli-service serve --env VUE_APP_frontendConfigUrl=/template-vue/frontend-config-devmock.json --env VUE_APP_env=devmock",
+        "example": "vue-cli-service build --env VUE_APP_frontendConfigUrl=/template-vue/frontend-config-dev.json --env NODE_ENV=production"
+    },
+    "dependencies": {
+        "ant-design-vue": "1.7.5",
+        "@liveqing/liveplayer": "2.7.10",
+        "axios": "0.19.2",
+        "core-js": "3.22.5",
+        "mockjs": "1.1.0",
+        "vue": "2.6.14",
+        "vue-router": "3.5.3",
+        "vuex": "3.6.2",
+        "swiper": "5.4.5",
+        "vuex-persistedstate": "3.0.1"
+    },
+    "devDependencies": {
+        "@babel/core": "7.17.10",
+        "@babel/eslint-parser": "7.17.0",
+        "@vue/cli-plugin-babel": "5.0.4",
+        "@vue/cli-plugin-eslint": "5.0.4",
+        "@vue/cli-plugin-router": "5.0.4",
+        "@vue/cli-plugin-vuex": "5.0.4",
+        "@vue/cli-service": "5.0.4",
+        "@vue/eslint-config-standard": "6.1.0",
+        "babel-plugin-component": "1.1.1",
+        "babel-plugin-import": "1.12.2",
+        "compression-webpack-plugin": "3.1.0",
+        "css-unicode-loader": "1.0.3",
+        "eslint": "7.32.0",
+        "eslint-plugin-import": "2.26.0",
+        "eslint-plugin-node": "11.1.0",
+        "eslint-plugin-promise": "5.2.0",
+        "eslint-plugin-vue": "8.7.1",
+        "sass": "1.51.0",
+        "sass-loader": "12.6.0",
+        "vue-template-compiler": "2.6.14"
+    }
+}