Update voice_processing.py
Browse files- voice_processing.py +30 -45
voice_processing.py
CHANGED
@@ -23,11 +23,7 @@ from lib.infer_pack.models import (
|
|
23 |
from rmvpe import RMVPE
|
24 |
from vc_infer_pipeline import VC
|
25 |
|
26 |
-
# Set
|
27 |
-
logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(name)s - %(levelname)s - %(message)s')
|
28 |
-
logger = logging.getLogger(__name__)
|
29 |
-
|
30 |
-
# Set logging levels for other libraries
|
31 |
logging.getLogger("fairseq").setLevel(logging.WARNING)
|
32 |
logging.getLogger("numba").setLevel(logging.WARNING)
|
33 |
logging.getLogger("markdown_it").setLevel(logging.WARNING)
|
@@ -56,7 +52,7 @@ def model_data(model_name):
|
|
56 |
for f in os.listdir(f"{model_root}/{model_name}")
|
57 |
if f.endswith(".pth")
|
58 |
][0]
|
59 |
-
|
60 |
cpt = torch.load(pth_path, map_location="cpu")
|
61 |
tgt_sr = cpt["config"][-1]
|
62 |
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
|
@@ -76,7 +72,7 @@ def model_data(model_name):
|
|
76 |
raise ValueError("Unknown version")
|
77 |
del net_g.enc_q
|
78 |
net_g.load_state_dict(cpt["weight"], strict=False)
|
79 |
-
|
80 |
net_g.eval().to(config.device)
|
81 |
if config.is_half:
|
82 |
net_g = net_g.half()
|
@@ -90,11 +86,11 @@ def model_data(model_name):
|
|
90 |
if f.endswith(".index")
|
91 |
]
|
92 |
if len(index_files) == 0:
|
93 |
-
|
94 |
index_file = ""
|
95 |
else:
|
96 |
index_file = index_files[0]
|
97 |
-
|
98 |
|
99 |
return tgt_sr, net_g, vc, version, index_file, if_f0
|
100 |
|
@@ -123,8 +119,6 @@ def run_async_in_thread(fn, *args):
|
|
123 |
loop.close()
|
124 |
return result
|
125 |
|
126 |
-
executor = ThreadPoolExecutor(max_workers=config.n_cpu)
|
127 |
-
|
128 |
def parallel_tts(tasks):
|
129 |
with ThreadPoolExecutor() as executor:
|
130 |
futures = [executor.submit(run_async_in_thread, tts, *task) for task in tasks]
|
@@ -139,24 +133,21 @@ async def tts(
|
|
139 |
use_uploaded_voice,
|
140 |
uploaded_voice,
|
141 |
):
|
142 |
-
|
143 |
-
|
144 |
-
|
145 |
-
|
146 |
-
|
147 |
-
|
148 |
-
|
149 |
-
|
150 |
-
|
151 |
-
|
152 |
-
|
153 |
-
edge_output_filename = get_unique_filename("mp3")
|
154 |
-
|
155 |
-
logger.info(f"Starting TTS process for text: {tts_text[:50]}...")
|
156 |
|
|
|
157 |
if use_uploaded_voice:
|
158 |
if uploaded_voice is None:
|
159 |
-
logger.error("No voice file uploaded.")
|
160 |
return "No voice file uploaded.", None, None
|
161 |
|
162 |
# Process the uploaded voice file
|
@@ -165,11 +156,9 @@ async def tts(
|
|
165 |
uploaded_file_path = tmp_file.name
|
166 |
|
167 |
audio, sr = librosa.load(uploaded_file_path, sr=16000, mono=True)
|
168 |
-
logger.info(f"Uploaded voice file loaded. Shape: {audio.shape}, SR: {sr}")
|
169 |
else:
|
170 |
# EdgeTTS processing
|
171 |
if limitation and len(tts_text) > 12000:
|
172 |
-
logger.error(f"Text characters exceed limit: {len(tts_text)} characters.")
|
173 |
return (
|
174 |
f"Text characters should be at most 12000 in this huggingface space, but got {len(tts_text)} characters.",
|
175 |
None,
|
@@ -186,13 +175,11 @@ async def tts(
|
|
186 |
edge_time = t1 - t0
|
187 |
|
188 |
audio, sr = librosa.load(edge_output_filename, sr=16000, mono=True)
|
189 |
-
logger.info(f"Edge TTS audio generated. Shape: {audio.shape}, SR: {sr}")
|
190 |
|
191 |
# Common processing after loading the audio
|
192 |
duration = len(audio) / sr
|
193 |
-
|
194 |
if limitation and duration >= 20000:
|
195 |
-
logger.error(f"Audio duration exceeds limit: {duration}s")
|
196 |
return (
|
197 |
f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.",
|
198 |
None,
|
@@ -208,7 +195,6 @@ async def tts(
|
|
208 |
|
209 |
# Perform voice conversion pipeline
|
210 |
times = [0, 0, 0]
|
211 |
-
logger.info(f"Starting voice conversion with audio shape: {audio.shape}")
|
212 |
audio_opt = vc.pipeline(
|
213 |
hubert_model,
|
214 |
net_g,
|
@@ -229,22 +215,28 @@ async def tts(
|
|
229 |
protect,
|
230 |
None,
|
231 |
)
|
232 |
-
logger.info(f"Voice conversion completed. Output shape: {audio_opt.shape}")
|
233 |
|
234 |
if tgt_sr != resample_sr and resample_sr >= 16000:
|
235 |
tgt_sr = resample_sr
|
236 |
|
237 |
-
info = f"Success. Time: tts: {edge_time
|
238 |
-
|
239 |
return (
|
240 |
info,
|
241 |
edge_output_filename if not use_uploaded_voice else None,
|
242 |
(tgt_sr, audio_opt),
|
243 |
)
|
244 |
|
|
|
|
|
|
|
|
|
|
|
|
|
245 |
except Exception as e:
|
246 |
-
|
247 |
-
|
|
|
248 |
|
249 |
voice_mapping = {
|
250 |
"Mongolian Male": "mn-MN-BataaNeural",
|
@@ -294,11 +286,4 @@ async def parallel_tts_processor(tasks):
|
|
294 |
|
295 |
def parallel_tts_wrapper(tasks):
|
296 |
loop = asyncio.get_event_loop()
|
297 |
-
return loop.run_until_complete(parallel_tts_processor(tasks))
|
298 |
-
|
299 |
-
# Keep the original parallel_tts function
|
300 |
-
# def parallel_tts(tasks):
|
301 |
-
# with ThreadPoolExecutor() as executor:
|
302 |
-
# futures = [executor.submit(run_async_in_thread, tts, *task) for task in tasks]
|
303 |
-
# results = [future.result() for future in futures]
|
304 |
-
# return results
|
|
|
23 |
from rmvpe import RMVPE
|
24 |
from vc_infer_pipeline import VC
|
25 |
|
26 |
+
# Set logging levels
|
|
|
|
|
|
|
|
|
27 |
logging.getLogger("fairseq").setLevel(logging.WARNING)
|
28 |
logging.getLogger("numba").setLevel(logging.WARNING)
|
29 |
logging.getLogger("markdown_it").setLevel(logging.WARNING)
|
|
|
52 |
for f in os.listdir(f"{model_root}/{model_name}")
|
53 |
if f.endswith(".pth")
|
54 |
][0]
|
55 |
+
print(f"Loading {pth_path}")
|
56 |
cpt = torch.load(pth_path, map_location="cpu")
|
57 |
tgt_sr = cpt["config"][-1]
|
58 |
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
|
|
|
72 |
raise ValueError("Unknown version")
|
73 |
del net_g.enc_q
|
74 |
net_g.load_state_dict(cpt["weight"], strict=False)
|
75 |
+
print("Model loaded")
|
76 |
net_g.eval().to(config.device)
|
77 |
if config.is_half:
|
78 |
net_g = net_g.half()
|
|
|
86 |
if f.endswith(".index")
|
87 |
]
|
88 |
if len(index_files) == 0:
|
89 |
+
print("No index file found")
|
90 |
index_file = ""
|
91 |
else:
|
92 |
index_file = index_files[0]
|
93 |
+
print(f"Index file found: {index_file}")
|
94 |
|
95 |
return tgt_sr, net_g, vc, version, index_file, if_f0
|
96 |
|
|
|
119 |
loop.close()
|
120 |
return result
|
121 |
|
|
|
|
|
122 |
def parallel_tts(tasks):
|
123 |
with ThreadPoolExecutor() as executor:
|
124 |
futures = [executor.submit(run_async_in_thread, tts, *task) for task in tasks]
|
|
|
133 |
use_uploaded_voice,
|
134 |
uploaded_voice,
|
135 |
):
|
136 |
+
# Default values for parameters used in EdgeTTS
|
137 |
+
speed = 0 # Default speech speed
|
138 |
+
f0_up_key = 0 # Default pitch adjustment
|
139 |
+
f0_method = "rmvpe" # Default pitch extraction method
|
140 |
+
protect = 0.33 # Default protect value
|
141 |
+
filter_radius = 3
|
142 |
+
resample_sr = 0
|
143 |
+
rms_mix_rate = 0.25
|
144 |
+
edge_time = 0 # Initialize edge_time
|
145 |
+
|
146 |
+
edge_output_filename = get_unique_filename("mp3")
|
|
|
|
|
|
|
147 |
|
148 |
+
try:
|
149 |
if use_uploaded_voice:
|
150 |
if uploaded_voice is None:
|
|
|
151 |
return "No voice file uploaded.", None, None
|
152 |
|
153 |
# Process the uploaded voice file
|
|
|
156 |
uploaded_file_path = tmp_file.name
|
157 |
|
158 |
audio, sr = librosa.load(uploaded_file_path, sr=16000, mono=True)
|
|
|
159 |
else:
|
160 |
# EdgeTTS processing
|
161 |
if limitation and len(tts_text) > 12000:
|
|
|
162 |
return (
|
163 |
f"Text characters should be at most 12000 in this huggingface space, but got {len(tts_text)} characters.",
|
164 |
None,
|
|
|
175 |
edge_time = t1 - t0
|
176 |
|
177 |
audio, sr = librosa.load(edge_output_filename, sr=16000, mono=True)
|
|
|
178 |
|
179 |
# Common processing after loading the audio
|
180 |
duration = len(audio) / sr
|
181 |
+
print(f"Audio duration: {duration}s")
|
182 |
if limitation and duration >= 20000:
|
|
|
183 |
return (
|
184 |
f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.",
|
185 |
None,
|
|
|
195 |
|
196 |
# Perform voice conversion pipeline
|
197 |
times = [0, 0, 0]
|
|
|
198 |
audio_opt = vc.pipeline(
|
199 |
hubert_model,
|
200 |
net_g,
|
|
|
215 |
protect,
|
216 |
None,
|
217 |
)
|
|
|
218 |
|
219 |
if tgt_sr != resample_sr and resample_sr >= 16000:
|
220 |
tgt_sr = resample_sr
|
221 |
|
222 |
+
info = f"Success. Time: tts: {edge_time}s, npy: {times[0]}s, f0: {times[1]}s, infer: {times[2]}s"
|
223 |
+
print(info)
|
224 |
return (
|
225 |
info,
|
226 |
edge_output_filename if not use_uploaded_voice else None,
|
227 |
(tgt_sr, audio_opt),
|
228 |
)
|
229 |
|
230 |
+
except EOFError:
|
231 |
+
info = (
|
232 |
+
"output not valid. This may occur when input text and speaker do not match."
|
233 |
+
)
|
234 |
+
print(info)
|
235 |
+
return info, None, None
|
236 |
except Exception as e:
|
237 |
+
traceback_info = traceback.format_exc()
|
238 |
+
print(traceback_info)
|
239 |
+
return str(e), None, None
|
240 |
|
241 |
voice_mapping = {
|
242 |
"Mongolian Male": "mn-MN-BataaNeural",
|
|
|
286 |
|
287 |
def parallel_tts_wrapper(tasks):
|
288 |
loop = asyncio.get_event_loop()
|
289 |
+
return loop.run_until_complete(parallel_tts_processor(tasks))
|
|
|
|
|
|
|
|
|
|
|
|
|
|