API for V3 (#2154)

GPT_SoVITS/module/models.py

@@ -1162,6 +1162,7 @@ class SynthesizerTrnV3(nn.Module):
                  use_sdp=True,
                  semantic_frame_rate=None,
                  freeze_quantizer=None,
+                 version="v3",
                  **kwargs):
 
         super().__init__()
@@ -1182,6 +1183,7 @@ class SynthesizerTrnV3(nn.Module):
         self.segment_size = segment_size
         self.n_speakers = n_speakers
         self.gin_channels = gin_channels
+        self.version = version
 
         self.model_dim=512
         self.use_sdp = use_sdp

GPT_SoVITS/text/chinese.py

@@ -17,6 +17,8 @@ pinyin_to_symbol_map = {
     for line in open(os.path.join(current_file_path, "opencpop-strict.txt")).readlines()
 }
 
+import jieba_fast, logging
+jieba_fast.setLogLevel(logging.CRITICAL)
 import jieba_fast.posseg as psg
 
 

GPT_SoVITS/text/chinese2.py

@@ -18,13 +18,15 @@ pinyin_to_symbol_map = {
     for line in open(os.path.join(current_file_path, "opencpop-strict.txt")).readlines()
 }
 
+import jieba_fast, logging
+jieba_fast.setLogLevel(logging.CRITICAL)
 import jieba_fast.posseg as psg
 
 # is_g2pw_str = os.environ.get("is_g2pw", "True")##默认开启
 # is_g2pw = False#True if is_g2pw_str.lower() == 'true' else False
 is_g2pw = True#True if is_g2pw_str.lower() == 'true' else False
 if is_g2pw:
-    print("当前使用g2pw进行拼音推理")
+    # print("当前使用g2pw进行拼音推理")
     from text.g2pw import G2PWPinyin, correct_pronunciation
     parent_directory = os.path.dirname(current_file_path)
     g2pw = G2PWPinyin(model_dir="GPT_SoVITS/text/G2PWModel",model_source=os.environ.get("bert_path","GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large"),v_to_u=False, neutral_tone_with_five=True)

api.py

@@ -150,9 +150,9 @@ sys.path.append(now_dir)
 sys.path.append("%s/GPT_SoVITS" % (now_dir))
 
 import signal
-import LangSegment
+from text.LangSegmenter import LangSegmenter
 from time import time as ttime
-import torch
+import torch, torchaudio
 import librosa
 import soundfile as sf
 from fastapi import FastAPI, Request, Query, HTTPException
@@ -162,7 +162,8 @@ from transformers import AutoModelForMaskedLM, AutoTokenizer
 import numpy as np
 from feature_extractor import cnhubert
 from io import BytesIO
-from module.models import SynthesizerTrn
+from module.models import SynthesizerTrn, SynthesizerTrnV3
+from peft import LoraConfig, PeftModel, get_peft_model
 from AR.models.t2s_lightning_module import Text2SemanticLightningModule
 from text import cleaned_text_to_sequence
 from text.cleaner import clean_text
@@ -197,6 +198,61 @@ def is_full(*items):  # 任意一项为空返回False
     return True
 
 
+def init_bigvgan():
+    global bigvgan_model
+    from BigVGAN import bigvgan
+    bigvgan_model = bigvgan.BigVGAN.from_pretrained("%s/GPT_SoVITS/pretrained_models/models--nvidia--bigvgan_v2_24khz_100band_256x" % (now_dir,), use_cuda_kernel=False)  # if True, RuntimeError: Ninja is required to load C++ extensions
+    # remove weight norm in the model and set to eval mode
+    bigvgan_model.remove_weight_norm()
+    bigvgan_model = bigvgan_model.eval()
+    if is_half == True:
+        bigvgan_model = bigvgan_model.half().to(device)
+    else:
+        bigvgan_model = bigvgan_model.to(device)
+
+
+resample_transform_dict={}
+def resample(audio_tensor, sr0):
+    global resample_transform_dict
+    if sr0 not in resample_transform_dict:
+        resample_transform_dict[sr0] = torchaudio.transforms.Resample(
+            sr0, 24000
+        ).to(device)
+    return resample_transform_dict[sr0](audio_tensor)
+
+
+from module.mel_processing import spectrogram_torch,mel_spectrogram_torch
+spec_min = -12
+spec_max = 2
+def norm_spec(x):
+    return (x - spec_min) / (spec_max - spec_min) * 2 - 1
+def denorm_spec(x):
+    return (x + 1) / 2 * (spec_max - spec_min) + spec_min
+mel_fn=lambda x: mel_spectrogram_torch(x, **{
+    "n_fft": 1024,
+    "win_size": 1024,
+    "hop_size": 256,
+    "num_mels": 100,
+    "sampling_rate": 24000,
+    "fmin": 0,
+    "fmax": None,
+    "center": False
+})
+
+
+sr_model=None
+def audio_sr(audio,sr):
+    global sr_model
+    if sr_model==None:
+        from tools.audio_sr import AP_BWE
+        try:
+            sr_model=AP_BWE(device,DictToAttrRecursive)
+        except FileNotFoundError:
+            logger.info("你没有下载超分模型的参数，因此不进行超分。如想超分请先参照教程把文件下载")
+            return audio.cpu().detach().numpy(),sr
+    return sr_model(audio,sr)
+
+
 class Speaker:
     def __init__(self, name, gpt, sovits, phones = None, bert = None, prompt = None):
         self.name = name
@@ -214,31 +270,72 @@ class Sovits:
         self.vq_model = vq_model
         self.hps = hps
 
+from process_ckpt import get_sovits_version_from_path_fast,load_sovits_new
 def get_sovits_weights(sovits_path):
-    dict_s2 = torch.load(sovits_path, map_location="cpu")
+    path_sovits_v3="GPT_SoVITS/pretrained_models/s2Gv3.pth"
+    is_exist_s2gv3=os.path.exists(path_sovits_v3)
+
+    version, model_version, if_lora_v3=get_sovits_version_from_path_fast(sovits_path)
+    if if_lora_v3==True and is_exist_s2gv3==False:
+        logger.info("SoVITS V3 底模缺失，无法加载相应 LoRA 权重")
+
+    dict_s2 = load_sovits_new(sovits_path)
     hps = dict_s2["config"]
     hps = DictToAttrRecursive(hps)
     hps.model.semantic_frame_rate = "25hz"
-    if dict_s2['weight']['enc_p.text_embedding.weight'].shape[0] == 322:
+    if 'enc_p.text_embedding.weight' not in dict_s2['weight']:
+        hps.model.version = "v2"#v3model,v2sybomls
+    elif dict_s2['weight']['enc_p.text_embedding.weight'].shape[0] == 322:
         hps.model.version = "v1"
     else:
         hps.model.version = "v2"
-    logger.info(f"模型版本: {hps.model.version}")
+
+    if model_version == "v3":
+        hps.model.version = "v3"
+
     model_params_dict = vars(hps.model)
+    if model_version!="v3":
         vq_model = SynthesizerTrn(
             hps.data.filter_length // 2 + 1,
             hps.train.segment_size // hps.data.hop_length,
             n_speakers=hps.data.n_speakers,
             **model_params_dict
         )
+    else:
+        vq_model = SynthesizerTrnV3(
+            hps.data.filter_length // 2 + 1,
+            hps.train.segment_size // hps.data.hop_length,
+            n_speakers=hps.data.n_speakers,
+            **model_params_dict
+        )
+        init_bigvgan()
+    model_version=hps.model.version
+    logger.info(f"模型版本: {model_version}")
     if ("pretrained" not in sovits_path):
+        try:
             del vq_model.enc_q
+        except:pass
     if is_half == True:
         vq_model = vq_model.half().to(device)
     else:
         vq_model = vq_model.to(device)
     vq_model.eval()
+    if if_lora_v3 == False:
         vq_model.load_state_dict(dict_s2["weight"], strict=False)
+    else:
+        vq_model.load_state_dict(load_sovits_new(path_sovits_v3)["weight"], strict=False)
+        lora_rank=dict_s2["lora_rank"]
+        lora_config = LoraConfig(
+            target_modules=["to_k", "to_q", "to_v", "to_out.0"],
+            r=lora_rank,
+            lora_alpha=lora_rank,
+            init_lora_weights=True,
+        )
+        vq_model.cfm = get_peft_model(vq_model.cfm, lora_config)
+        vq_model.load_state_dict(dict_s2["weight"], strict=False)
+        vq_model.cfm = vq_model.cfm.merge_and_unload()
+        # torch.save(vq_model.state_dict(),"merge_win.pth")
+        vq_model.eval()
 
     sovits = Sovits(vq_model, hps)
     return sovits
@@ -260,8 +357,8 @@ def get_gpt_weights(gpt_path):
         t2s_model = t2s_model.half()
     t2s_model = t2s_model.to(device)
     t2s_model.eval()
-    total = sum([param.nelement() for param in t2s_model.parameters()])
-    logger.info("Number of parameter: %.2fM" % (total / 1e6))
+    # total = sum([param.nelement() for param in t2s_model.parameters()])
+    # logger.info("Number of parameter: %.2fM" % (total / 1e6))
 
     gpt = Gpt(max_sec, t2s_model)
     return gpt
@@ -295,6 +392,7 @@ def get_bert_feature(text, word2ph):
 
 
 def clean_text_inf(text, language, version):
+    language = language.replace("all_","")
     phones, word2ph, norm_text = clean_text(text, language, version)
     phones = cleaned_text_to_sequence(phones, version)
     return phones, word2ph, norm_text
@@ -315,16 +413,10 @@ def get_bert_inf(phones, word2ph, norm_text, language):
 from text import chinese
 def get_phones_and_bert(text,language,version,final=False):
     if language in {"en", "all_zh", "all_ja", "all_ko", "all_yue"}:
-        language = language.replace("all_","")
-        if language == "en":
-            LangSegment.setfilters(["en"])
-            formattext = " ".join(tmp["text"] for tmp in LangSegment.getTexts(text))
-        else:
-            # 因无法区别中日韩文汉字,以用户输入为准
         formattext = text
         while "  " in formattext:
             formattext = formattext.replace("  ", " ")
-        if language == "zh":
+        if language == "all_zh":
             if re.search(r'[A-Za-z]', formattext):
                 formattext = re.sub(r'[a-z]', lambda x: x.group(0).upper(), formattext)
                 formattext = chinese.mix_text_normalize(formattext)
@@ -332,7 +424,7 @@ def get_phones_and_bert(text,language,version,final=False):
             else:
                 phones, word2ph, norm_text = clean_text_inf(formattext, language, version)
                 bert = get_bert_feature(norm_text, word2ph).to(device)
-        elif language == "yue" and re.search(r'[A-Za-z]', formattext):
+        elif language == "all_yue" and re.search(r'[A-Za-z]', formattext):
                 formattext = re.sub(r'[a-z]', lambda x: x.group(0).upper(), formattext)
                 formattext = chinese.mix_text_normalize(formattext)
                 return get_phones_and_bert(formattext,"yue",version)
@@ -345,19 +437,18 @@ def get_phones_and_bert(text,language,version,final=False):
     elif language in {"zh", "ja", "ko", "yue", "auto", "auto_yue"}:
         textlist=[]
         langlist=[]
-        LangSegment.setfilters(["zh","ja","en","ko"])
         if language == "auto":
-            for tmp in LangSegment.getTexts(text):
+            for tmp in LangSegmenter.getTexts(text):
                 langlist.append(tmp["lang"])
                 textlist.append(tmp["text"])
         elif language == "auto_yue":
-            for tmp in LangSegment.getTexts(text):
+            for tmp in LangSegmenter.getTexts(text):
                 if tmp["lang"] == "zh":
                     tmp["lang"] = "yue"
                 langlist.append(tmp["lang"])
                 textlist.append(tmp["text"])
         else:
-            for tmp in LangSegment.getTexts(text):
+            for tmp in LangSegmenter.getTexts(text):
                 if tmp["lang"] == "en":
                     langlist.append(tmp["lang"])
                 else:
@@ -556,10 +647,11 @@ def only_punc(text):
 
 
 splits = {"，", "。", "？", "！", ",", ".", "?", "!", "~", ":", "：", "—", "…", }
-def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language, top_k= 15, top_p = 0.6, temperature = 0.6, speed = 1, inp_refs = None, spk = "default"):
+def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language, top_k= 15, top_p = 0.6, temperature = 0.6, speed = 1, inp_refs = None, sample_steps = 32, if_sr = False, spk = "default"):
     infer_sovits = speaker_list[spk].sovits
     vq_model = infer_sovits.vq_model
     hps = infer_sovits.hps
+    version = vq_model.version
 
     infer_gpt = speaker_list[spk].gpt
     t2s_model = infer_gpt.t2s_model
@@ -587,6 +679,7 @@ def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language,
         prompt_semantic = codes[0, 0]
         prompt = prompt_semantic.unsqueeze(0).to(device)
 
+        if version != "v3":
             refers=[]
             if(inp_refs):
                 for path in inp_refs:
@@ -597,10 +690,11 @@ def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language,
                         logger.error(e)
             if(len(refers)==0):
                 refers = [get_spepc(hps, ref_wav_path).to(dtype).to(device)]
+        else:
+            refer = get_spepc(hps, ref_wav_path).to(device).to(dtype)
 
     t1 = ttime()
-    version = vq_model.version
-    os.environ['version'] = version
+    # os.environ['version'] = version
     prompt_language = dict_language[prompt_language.lower()]
     text_language = dict_language[text_language.lower()]
     phones1, bert1, norm_text1 = get_phones_and_bert(prompt_text, prompt_language, version)
@@ -634,20 +728,82 @@ def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language,
                 early_stop_num=hz * max_sec)
             pred_semantic = pred_semantic[:, -idx:].unsqueeze(0)
         t3 = ttime()
+
+        if version != "v3":
             audio = \
                 vq_model.decode(pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0),
                                 refers,speed=speed).detach().cpu().numpy()[
                     0, 0]  ###试试重建不带上prompt部分
+        else:
+            phoneme_ids0=torch.LongTensor(phones1).to(device).unsqueeze(0)
+            phoneme_ids1=torch.LongTensor(phones2).to(device).unsqueeze(0)
+            # print(11111111, phoneme_ids0, phoneme_ids1)
+            fea_ref,ge = vq_model.decode_encp(prompt.unsqueeze(0), phoneme_ids0, refer)
+            ref_audio, sr = torchaudio.load(ref_wav_path)
+            ref_audio=ref_audio.to(device).float()
+            if (ref_audio.shape[0] == 2):
+                ref_audio = ref_audio.mean(0).unsqueeze(0)
+            if sr!=24000:
+                ref_audio=resample(ref_audio,sr)
+            # print("ref_audio",ref_audio.abs().mean())
+            mel2 = mel_fn(ref_audio)
+            mel2 = norm_spec(mel2)
+            T_min = min(mel2.shape[2], fea_ref.shape[2])
+            mel2 = mel2[:, :, :T_min]
+            fea_ref = fea_ref[:, :, :T_min]
+            if (T_min > 468):
+                mel2 = mel2[:, :, -468:]
+                fea_ref = fea_ref[:, :, -468:]
+                T_min = 468
+            chunk_len = 934 - T_min
+            # print("fea_ref",fea_ref,fea_ref.shape)
+            # print("mel2",mel2)
+            mel2=mel2.to(dtype)
+            fea_todo, ge = vq_model.decode_encp(pred_semantic, phoneme_ids1, refer, ge,speed)
+            # print("fea_todo",fea_todo)
+            # print("ge",ge.abs().mean())
+            cfm_resss = []
+            idx = 0
+            while (1):
+                fea_todo_chunk = fea_todo[:, :, idx:idx + chunk_len]
+                if (fea_todo_chunk.shape[-1] == 0): break
+                idx += chunk_len
+                fea = torch.cat([fea_ref, fea_todo_chunk], 2).transpose(2, 1)
+                # set_seed(123)
+                cfm_res = vq_model.cfm.inference(fea, torch.LongTensor([fea.size(1)]).to(fea.device), mel2, sample_steps, inference_cfg_rate=0)
+                cfm_res = cfm_res[:, :, mel2.shape[2]:]
+                mel2 = cfm_res[:, :, -T_min:]
+                # print("fea", fea)
+                # print("mel2in", mel2)
+                fea_ref = fea_todo_chunk[:, :, -T_min:]
+                cfm_resss.append(cfm_res)
+            cmf_res = torch.cat(cfm_resss, 2)
+            cmf_res = denorm_spec(cmf_res)
+            if bigvgan_model==None:init_bigvgan()
+            with torch.inference_mode():
+                wav_gen = bigvgan_model(cmf_res)
+                audio=wav_gen[0][0].cpu().detach().numpy()
+
         max_audio=np.abs(audio).max()
         if max_audio>1:
             audio/=max_audio
         audio_opt.append(audio)
         audio_opt.append(zero_wav)
+        audio_opt = np.concatenate(audio_opt, 0)
         t4 = ttime()
+
+        sr = hps.data.sampling_rate if version != "v3" else 24000
+        if if_sr and sr == 24000:
+            audio_opt = torch.from_numpy(audio_opt).float().to(device)
+            audio_opt,sr=audio_sr(audio_opt.unsqueeze(0),sr)
+            max_audio=np.abs(audio_opt).max()
+            if max_audio > 1: audio_opt /= max_audio
+            sr = 48000
+
         if is_int32:
-            audio_bytes = pack_audio(audio_bytes,(np.concatenate(audio_opt, 0) * 2147483647).astype(np.int32),hps.data.sampling_rate)
+            audio_bytes = pack_audio(audio_bytes,(audio_opt * 2147483647).astype(np.int32),sr)
         else:
-            audio_bytes = pack_audio(audio_bytes,(np.concatenate(audio_opt, 0) * 32768).astype(np.int16),hps.data.sampling_rate)
+            audio_bytes = pack_audio(audio_bytes,(audio_opt * 32768).astype(np.int16),sr)
     # logger.info("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3))
         if stream_mode == "normal":
             audio_bytes, audio_chunk = read_clean_buffer(audio_bytes)
@@ -655,7 +811,9 @@ def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language,
     
     if not stream_mode == "normal": 
         if media_type == "wav":
-            audio_bytes = pack_wav(audio_bytes,hps.data.sampling_rate)
+            sr = 48000 if if_sr else 24000
+            sr = hps.data.sampling_rate if version != "v3" else sr
+            audio_bytes = pack_wav(audio_bytes,sr)
         yield audio_bytes.getvalue()
 
 
@@ -688,7 +846,7 @@ def handle_change(path, text, language):
     return JSONResponse({"code": 0, "message": "Success"}, status_code=200)
 
 
-def handle(refer_wav_path, prompt_text, prompt_language, text, text_language, cut_punc, top_k, top_p, temperature, speed, inp_refs):
+def handle(refer_wav_path, prompt_text, prompt_language, text, text_language, cut_punc, top_k, top_p, temperature, speed, inp_refs, sample_steps, if_sr):
     if (
             refer_wav_path == "" or refer_wav_path is None
             or prompt_text == "" or prompt_text is None
@@ -702,12 +860,15 @@ def handle(refer_wav_path, prompt_text, prompt_language, text, text_language, cu
         if not default_refer.is_ready():
             return JSONResponse({"code": 400, "message": "未指定参考音频且接口无预设"}, status_code=400)
 
+    if not sample_steps in [4,8,16,32]:
+        sample_steps = 32
+
     if cut_punc == None:
         text = cut_text(text,default_cut_punc)
     else:
         text = cut_text(text,cut_punc)
 
-    return StreamingResponse(get_tts_wav(refer_wav_path, prompt_text, prompt_language, text, text_language, top_k, top_p, temperature, speed, inp_refs), media_type="audio/"+media_type)
+    return StreamingResponse(get_tts_wav(refer_wav_path, prompt_text, prompt_language, text, text_language, top_k, top_p, temperature, speed, inp_refs, sample_steps, if_sr), media_type="audio/"+media_type)
 
 
 
@@ -915,7 +1076,9 @@ async def tts_endpoint(request: Request):
         json_post_raw.get("top_p", 1.0),
         json_post_raw.get("temperature", 1.0),
         json_post_raw.get("speed", 1.0),
-        json_post_raw.get("inp_refs", [])
+        json_post_raw.get("inp_refs", []),
+        json_post_raw.get("sample_steps", 32),
+        json_post_raw.get("if_sr", False) 
     )
 
 
@@ -931,9 +1094,11 @@ async def tts_endpoint(
         top_p: float = 1.0,
         temperature: float = 1.0,
         speed: float = 1.0,
-        inp_refs: list = Query(default=[])
+        inp_refs: list = Query(default=[]),
+        sample_steps: int = 32,
+        if_sr: bool = False
 ):
-    return handle(refer_wav_path, prompt_text, prompt_language, text, text_language, cut_punc, top_k, top_p, temperature, speed, inp_refs)
+    return handle(refer_wav_path, prompt_text, prompt_language, text, text_language, cut_punc, top_k, top_p, temperature, speed, inp_refs, sample_steps, if_sr)
 
 
 if __name__ == "__main__":