mirror of
https://github.com/FunAudioLLM/CosyVoice.git
synced 2026-02-04 09:29:25 +08:00
update
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@@ -19,7 +19,7 @@ from hyperpyyaml import load_hyperpyyaml
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from modelscope import snapshot_download
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import torch
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from cosyvoice.cli.frontend import CosyVoiceFrontEnd
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from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model
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from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model, CosyVoice3Model
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from cosyvoice.utils.file_utils import logging
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from cosyvoice.utils.class_utils import get_model_type
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@@ -192,3 +192,43 @@ class CosyVoice2(CosyVoice):
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logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
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yield model_output
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start_time = time.time()
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class CosyVoice3(CosyVoice):
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def __init__(self, model_dir, load_jit=False, load_trt=False, load_vllm=False, fp16=False, trt_concurrent=1):
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self.instruct = True if '-Instruct' in model_dir else False
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self.model_dir = model_dir
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self.fp16 = fp16
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if not os.path.exists(model_dir):
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model_dir = snapshot_download(model_dir)
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hyper_yaml_path = '{}/cosyvoice3.yaml'.format(model_dir)
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if not os.path.exists(hyper_yaml_path):
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raise ValueError('{} not found!'.format(hyper_yaml_path))
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with open(hyper_yaml_path, 'r') as f:
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configs = load_hyperpyyaml(f, overrides={'qwen_pretrain_path': os.path.join(model_dir, 'CosyVoice-BlankEN')})
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assert get_model_type(configs) == CosyVoice2Model, 'do not use {} for CosyVoice2 initialization!'.format(model_dir)
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self.frontend = CosyVoiceFrontEnd(configs['get_tokenizer'],
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configs['feat_extractor'],
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'{}/campplus.onnx'.format(model_dir),
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'{}/speech_tokenizer_v3.onnx'.format(model_dir),
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'{}/spk2info.pt'.format(model_dir),
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configs['allowed_special'])
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self.sample_rate = configs['sample_rate']
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if torch.cuda.is_available() is False and (load_jit is True or load_trt is True or fp16 is True):
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load_jit, load_trt, fp16 = False, False, False
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logging.warning('no cuda device, set load_jit/load_trt/fp16 to False')
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self.model = CosyVoice3Model(configs['llm'], configs['flow'], configs['hift'], fp16)
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self.model.load('{}/llm.pt'.format(model_dir),
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'{}/flow.pt'.format(model_dir),
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'{}/bigvgan.pt'.format(model_dir))
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if load_vllm:
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self.model.load_vllm('{}/vllm'.format(model_dir))
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if load_jit:
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self.model.load_jit('{}/flow.encoder.{}.zip'.format(model_dir, 'fp16' if self.fp16 is True else 'fp32'))
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if load_trt:
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self.model.load_trt('{}/flow.decoder.estimator.{}.mygpu.plan'.format(model_dir, 'fp16' if self.fp16 is True else 'fp32'),
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'{}/flow.decoder.estimator.fp32.onnx'.format(model_dir),
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trt_concurrent,
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self.fp16)
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del configs
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@@ -384,3 +384,70 @@ class CosyVoice2Model(CosyVoiceModel):
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if torch.cuda.is_available():
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torch.cuda.empty_cache()
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torch.cuda.current_stream().synchronize()
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class CosyVoice3Model(CosyVoice2Model):
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def __init__(self,
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llm: torch.nn.Module,
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flow: torch.nn.Module,
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hift: torch.nn.Module,
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fp16: bool = False):
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self.device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
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self.llm = llm
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self.flow = flow
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self.hift = hift
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self.fp16 = fp16
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if self.fp16 is True:
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self.llm.half()
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self.flow.half()
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# NOTE must matching training static_chunk_size
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self.token_hop_len = 25
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# hift cache
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self.mel_cache_len = 8
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self.source_cache_len = int(self.mel_cache_len * 480)
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# speech fade in out
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self.speech_window = np.hamming(2 * self.source_cache_len)
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# rtf and decoding related
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self.llm_context = torch.cuda.stream(torch.cuda.Stream(self.device)) if torch.cuda.is_available() else nullcontext()
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self.lock = threading.Lock()
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# dict used to store session related variable
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self.tts_speech_token_dict = {}
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self.llm_end_dict = {}
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self.hift_cache_dict = {}
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def token2wav(self, token, prompt_token, prompt_feat, embedding, token_offset, uuid, stream=False, finalize=False, speed=1.0):
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with torch.cuda.amp.autocast(self.fp16):
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tts_mel, _ = self.flow.inference(token=token.to(self.device),
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token_len=torch.tensor([token.shape[1]], dtype=torch.int32).to(self.device),
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prompt_token=prompt_token.to(self.device),
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prompt_token_len=torch.tensor([prompt_token.shape[1]], dtype=torch.int32).to(self.device),
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prompt_feat=prompt_feat.to(self.device),
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prompt_feat_len=torch.tensor([prompt_feat.shape[1]], dtype=torch.int32).to(self.device),
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embedding=embedding.to(self.device),
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streaming=stream,
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finalize=finalize)
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tts_mel = tts_mel[:, :, token_offset * self.flow.token_mel_ratio:]
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# append hift cache
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if self.hift_cache_dict[uuid] is not None:
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hift_cache_mel, hift_cache_source = self.hift_cache_dict[uuid]['mel'], self.hift_cache_dict[uuid]['source']
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tts_mel = torch.concat([hift_cache_mel, tts_mel], dim=2)
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else:
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hift_cache_source = torch.zeros(1, 1, 0)
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# keep overlap mel and hift cache
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if finalize is False:
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tts_speech, tts_source = self.hift.inference(speech_feat=tts_mel, cache_source=hift_cache_source)
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if self.hift_cache_dict[uuid] is not None:
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tts_speech = fade_in_out(tts_speech, self.hift_cache_dict[uuid]['speech'], self.speech_window)
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self.hift_cache_dict[uuid] = {'mel': tts_mel[:, :, -self.mel_cache_len:],
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'source': tts_source[:, :, -self.source_cache_len:],
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'speech': tts_speech[:, -self.source_cache_len:]}
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tts_speech = tts_speech[:, :-self.source_cache_len]
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else:
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if speed != 1.0:
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assert self.hift_cache_dict[uuid] is None, 'speed change only support non-stream inference mode'
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tts_mel = F.interpolate(tts_mel, size=int(tts_mel.shape[2] / speed), mode='linear')
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tts_speech, tts_source = self.hift.inference(speech_feat=tts_mel, cache_source=hift_cache_source)
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if self.hift_cache_dict[uuid] is not None:
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tts_speech = fade_in_out(tts_speech, self.hift_cache_dict[uuid]['speech'], self.speech_window)
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return tts_speech
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