Update voice_processing.py
Browse files- voice_processing.py +31 -26
voice_processing.py
CHANGED
@@ -23,7 +23,11 @@ from lib.infer_pack.models import (
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from rmvpe import RMVPE
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from vc_infer_pipeline import VC
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-
# Set logging
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logging.getLogger("fairseq").setLevel(logging.WARNING)
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logging.getLogger("numba").setLevel(logging.WARNING)
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logging.getLogger("markdown_it").setLevel(logging.WARNING)
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@@ -52,7 +56,7 @@ def model_data(model_name):
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for f in os.listdir(f"{model_root}/{model_name}")
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if f.endswith(".pth")
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][0]
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-
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cpt = torch.load(pth_path, map_location="cpu")
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tgt_sr = cpt["config"][-1]
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cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
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@@ -72,7 +76,7 @@ def model_data(model_name):
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raise ValueError("Unknown version")
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del net_g.enc_q
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net_g.load_state_dict(cpt["weight"], strict=False)
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-
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net_g.eval().to(config.device)
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if config.is_half:
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net_g = net_g.half()
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@@ -86,11 +90,11 @@ def model_data(model_name):
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if f.endswith(".index")
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]
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if len(index_files) == 0:
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-
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index_file = ""
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else:
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index_file = index_files[0]
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-
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return tgt_sr, net_g, vc, version, index_file, if_f0
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@@ -119,6 +123,8 @@ def run_async_in_thread(fn, *args):
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loop.close()
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return result
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def parallel_tts(tasks):
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with ThreadPoolExecutor() as executor:
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futures = [executor.submit(run_async_in_thread, tts, *task) for task in tasks]
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@@ -133,21 +139,21 @@ async def tts(
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use_uploaded_voice,
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uploaded_voice,
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):
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# Default values for parameters used in EdgeTTS
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speed = 0 # Default speech speed
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f0_up_key = 0 # Default pitch adjustment
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f0_method = "rmvpe" # Default pitch extraction method
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protect = 0.33 # Default protect value
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filter_radius = 3
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resample_sr = 0
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rms_mix_rate = 0.25
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edge_time = 0 # Initialize edge_time
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-
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edge_output_filename = get_unique_filename("mp3")
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-
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try:
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if use_uploaded_voice:
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if uploaded_voice is None:
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return "No voice file uploaded.", None, None
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# Process the uploaded voice file
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@@ -159,6 +165,7 @@ async def tts(
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else:
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# EdgeTTS processing
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if limitation and len(tts_text) > 12000:
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return (
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f"Text characters should be at most 12000 in this huggingface space, but got {len(tts_text)} characters.",
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None,
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@@ -178,8 +185,9 @@ async def tts(
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# Common processing after loading the audio
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duration = len(audio) / sr
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if limitation and duration >= 20000:
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return (
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f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.",
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None,
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@@ -219,8 +227,8 @@ async def tts(
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if tgt_sr != resample_sr and resample_sr >= 16000:
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tgt_sr = resample_sr
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info = f"Success. Time: tts: {edge_time}s, npy: {times[0]}s, f0: {times[1]}s, infer: {times[2]}s"
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return (
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info,
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edge_output_filename if not use_uploaded_voice else None,
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@@ -228,14 +236,11 @@ async def tts(
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)
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except EOFError:
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info =
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)
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print(info)
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return info, None, None
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except Exception as e:
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print(traceback_info)
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return str(e), None, None
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voice_mapping = {
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from rmvpe import RMVPE
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from vc_infer_pipeline import VC
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+
# Set up logging
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logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(name)s - %(levelname)s - %(message)s')
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logger = logging.getLogger(__name__)
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# Set logging levels for other libraries
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logging.getLogger("fairseq").setLevel(logging.WARNING)
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logging.getLogger("numba").setLevel(logging.WARNING)
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logging.getLogger("markdown_it").setLevel(logging.WARNING)
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for f in os.listdir(f"{model_root}/{model_name}")
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if f.endswith(".pth")
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][0]
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logger.info(f"Loading {pth_path}")
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cpt = torch.load(pth_path, map_location="cpu")
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tgt_sr = cpt["config"][-1]
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cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
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raise ValueError("Unknown version")
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del net_g.enc_q
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net_g.load_state_dict(cpt["weight"], strict=False)
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logger.info("Model loaded")
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net_g.eval().to(config.device)
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if config.is_half:
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net_g = net_g.half()
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if f.endswith(".index")
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]
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if len(index_files) == 0:
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logger.info("No index file found")
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index_file = ""
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else:
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index_file = index_files[0]
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logger.info(f"Index file found: {index_file}")
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return tgt_sr, net_g, vc, version, index_file, if_f0
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loop.close()
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return result
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executor = ThreadPoolExecutor(max_workers=config.n_cpu)
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def parallel_tts(tasks):
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with ThreadPoolExecutor() as executor:
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futures = [executor.submit(run_async_in_thread, tts, *task) for task in tasks]
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use_uploaded_voice,
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uploaded_voice,
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):
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try:
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# Default values for parameters used in EdgeTTS
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speed = 0 # Default speech speed
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f0_up_key = 0 # Default pitch adjustment
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f0_method = "rmvpe" # Default pitch extraction method
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protect = 0.33 # Default protect value
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filter_radius = 3
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resample_sr = 0
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rms_mix_rate = 0.25
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edge_output_filename = get_unique_filename("mp3")
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if use_uploaded_voice:
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if uploaded_voice is None:
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logger.error("No voice file uploaded.")
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return "No voice file uploaded.", None, None
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# Process the uploaded voice file
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else:
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# EdgeTTS processing
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if limitation and len(tts_text) > 12000:
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logger.error(f"Text characters exceed limit: {len(tts_text)} characters.")
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return (
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f"Text characters should be at most 12000 in this huggingface space, but got {len(tts_text)} characters.",
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None,
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# Common processing after loading the audio
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duration = len(audio) / sr
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logger.info(f"Audio duration: {duration}s")
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if limitation and duration >= 20000:
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logger.error(f"Audio duration exceeds limit: {duration}s")
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return (
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f"Audio should be less than 20 seconds in this huggingface space, but got {duration}s.",
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None,
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if tgt_sr != resample_sr and resample_sr >= 16000:
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tgt_sr = resample_sr
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info = f"Success. Time: tts: {edge_time:.2f}s, npy: {times[0]:.2f}s, f0: {times[1]:.2f}s, infer: {times[2]:.2f}s"
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logger.info(info)
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return (
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info,
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edge_output_filename if not use_uploaded_voice else None,
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)
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except EOFError:
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info = "output not valid. This may occur when input text and speaker do not match."
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logger.error(info)
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return info, None, None
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except Exception as e:
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logger.exception("Error in TTS processing")
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return str(e), None, None
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voice_mapping = {
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