Spaces:
Running
on
Zero
Running
on
Zero
File size: 8,234 Bytes
6852edb |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 |
import io
import threading
import time
import os
import numpy as np
import torch
import torchaudio
import onnxruntime
import whisper
from funasr_detach import AutoModel
from utils import resample_audio, energy_norm_fn, trim_silence
from model_loader import model_loader, ModelSource
class StepAudioTokenizer:
def __init__(
self,
encoder_path,
model_source=ModelSource.AUTO,
funasr_model_id="dengcunqin/speech_paraformer-large_asr_nat-zh-cantonese-en-16k-vocab8501-online"
):
"""
Initialize StepAudioTokenizer
Args:
encoder_path: Encoder path
model_source: Model source (auto/local/modelscope/huggingface)
funasr_model_id: FunASR model ID or path
"""
funasr_model_path = os.path.join(encoder_path, funasr_model_id)
# Load FunASR model - use unified loader to handle all modes
try:
self.funasr_model = model_loader.load_funasr_model(
encoder_path,
funasr_model_path,
source=model_source,
model_revision="main"
)
except Exception as e:
print(f"Failed to load FunASR model from {model_source}: {e}")
# Fallback to default method
self.funasr_model = AutoModel(model=funasr_model_path, model_revision="main")
# Load other resource files (these are usually local files)
kms_path = os.path.join(self.funasr_model.repo_path, "linguistic_tokenizer.npy")
cosy_tokenizer_path = os.path.join(self.funasr_model.repo_path, "speech_tokenizer_v1.onnx")
if not os.path.exists(kms_path):
raise FileNotFoundError(f"KMS file not found: {kms_path}")
if not os.path.exists(cosy_tokenizer_path):
raise FileNotFoundError(f"Cosy tokenizer file not found: {cosy_tokenizer_path}")
self.kms = torch.tensor(np.load(kms_path))
providers = ["CUDAExecutionProvider"]
session_option = onnxruntime.SessionOptions()
session_option.graph_optimization_level = (
onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
)
session_option.intra_op_num_threads = 1
self.ort_session = onnxruntime.InferenceSession(
cosy_tokenizer_path, sess_options=session_option, providers=providers
)
self.chunk_size = [0, 4, 5]
self.encoder_chunk_look_back = 4
self.decoder_chunk_look_back = 1
self.vq02_sessions = {}
self.vq02_lock = threading.Lock()
self.vq06_lock = threading.Lock()
def __call__(self, audio, sr):
_, vq02, vq06 = self.wav2token(audio, sr, False)
text = self.merge_vq0206_to_token_str(vq02, vq06)
return text
def preprocess_wav(self, audio, sample_rate, enable_trim=True, energy_norm=True):
audio = resample_audio(audio, sample_rate, 16000)
if energy_norm:
audio = energy_norm_fn(audio)
if enable_trim:
audio = audio.cpu().numpy().squeeze(0)
audio = trim_silence(audio, 16000)
audio = torch.from_numpy(audio)
audio = audio.unsqueeze(0)
return audio
def wav2token(self, audio, sample_rate, enable_trim=True, energy_norm=True):
audio = self.preprocess_wav(
audio, sample_rate, enable_trim=enable_trim, energy_norm=energy_norm
)
vq02_ori = self.get_vq02_code(audio)
vq02 = [int(x) + 65536 for x in vq02_ori]
vq06_ori = self.get_vq06_code(audio)
vq06 = [int(x) + 65536 + 1024 for x in vq06_ori]
chunk = 1
chunk_nums = min(len(vq06) // (3 * chunk), len(vq02) // (2 * chunk))
speech_tokens = []
for idx in range(chunk_nums):
speech_tokens += vq02[idx * chunk * 2 : (idx + 1) * chunk * 2]
speech_tokens += vq06[idx * chunk * 3 : (idx + 1) * chunk * 3]
return speech_tokens, vq02_ori, vq06_ori
def get_vq02_code(self, audio, session_id=None, is_final=True):
_tmp_wav = io.BytesIO()
torchaudio.save(_tmp_wav, audio, 16000, format="wav")
_tmp_wav.seek(0)
with self.vq02_lock:
cache = {}
if session_id in self.vq02_sessions:
cache = self.vq02_sessions[session_id].get("cache", {})
res, new_cache = self.funasr_model.infer_encoder(
input=[_tmp_wav],
chunk_size=self.chunk_size,
encoder_chunk_look_back=self.encoder_chunk_look_back,
decoder_chunk_look_back=self.decoder_chunk_look_back,
device=0,
is_final=is_final,
cache=cache,
)
c_list = []
for j, res_ in enumerate(res):
feat = res_["enc_out"]
if len(feat) > 0:
c_list = self.dump_label([feat], self.kms)[0]
if is_final:
if session_id in self.vq02_sessions:
self.vq02_sessions.pop(session_id)
else:
if isinstance(session_id, str) and len(session_id) > 0:
self.vq02_sessions[session_id] = {
"cache": new_cache,
"update_time": time.time(),
}
return c_list
def get_vq06_code(self, audio):
def split_audio(audio, chunk_duration=480000):
start = 0
chunks = []
while start < len(audio):
end = min(start + chunk_duration, len(audio))
chunk = audio[start:end]
if len(chunk) < 480:
pass
else:
chunks.append(chunk)
start = end
return chunks
with self.vq06_lock:
audio = audio.squeeze(0)
chunk_audios = split_audio(audio, chunk_duration=30 * 16000) # Maximum support 30s
speech_tokens = []
for chunk in chunk_audios:
duration = round(chunk.shape[0] / 16000, 2)
feat = whisper.log_mel_spectrogram(chunk, n_mels=128)
feat = feat.unsqueeze(0)
feat_len = np.array([feat.shape[2]], dtype=np.int32)
chunk_token = (
self.ort_session.run(
None,
{
self.ort_session.get_inputs()[0]
.name: feat.detach()
.cpu()
.numpy(),
self.ort_session.get_inputs()[1].name: feat_len,
},
)[0]
.flatten()
.tolist()
)
assert abs(len(chunk_token) - duration * 25) <= 2
speech_tokens += chunk_token
return speech_tokens
def kmean_cluster(self, samples, means):
dists = torch.cdist(samples, means)
indices = dists.argmin(dim=1).cpu().numpy()
return indices.tolist()
def dump_label(self, samples, mean):
dims = samples[0].shape[-1]
x_lens = [x.shape[1] for x in samples]
total_len = sum(x_lens)
x_sel = torch.FloatTensor(1, total_len, dims)
start_len = 0
for sample in samples:
sample_len = sample.shape[1]
end_len = start_len + sample_len
x_sel[:, start_len:end_len] = sample
start_len = end_len
dense_x = x_sel.squeeze(0)
indices = self.kmean_cluster(dense_x, mean)
indices_list = []
start_len = 0
for x_len in x_lens:
end_len = start_len + end_len
indices_list.append(indices[start_len:end_len])
return indices_list
def merge_vq0206_to_token_str(self, vq02, vq06):
_vq06 = [1024 + x for x in vq06]
result = []
i = 0
j = 0
while i < len(vq02) - 1 and j < len(_vq06) - 2:
sublist = vq02[i : i + 2] + _vq06[j : j + 3]
result.extend(sublist)
i += 2
j += 3
return "".join([f"<audio_{x}>" for x in result])
|