Merge pull request #1670 from FunAudioLLM/dev/lyuxiang.lx

Dev/lyuxiang.lx

.github/workflows/lint.yml

@@ -52,5 +52,5 @@ jobs:
           set -eux
           pip install flake8==3.8.2 flake8-bugbear flake8-comprehensions flake8-executable flake8-pyi==20.5.0 mccabe pycodestyle==2.6.0 pyflakes==2.2.0
           flake8 --version
-          flake8 --max-line-length 180 --ignore B006,B008,B905,C408,E402,E731,E741,W503,W504,F401,F403,F405,F841 --exclude ./third_party/,./runtime/python/grpc/cosyvoice_pb2*py
+          flake8 --max-line-length 180 --ignore B006,B008,B905,C408,E402,E731,E741,W503,W504,F401,F403,F405,F722,F841 --exclude ./third_party/,./runtime/python/grpc/cosyvoice_pb2*py
           if [ $? != 0 ]; then exit 1; fi

README.md

@@ -2,7 +2,7 @@
 
 ## 👉🏻 CosyVoice 👈🏻
 
-**CosyVoice 3.0**: [Demos](https://funaudiollm.github.io/cosyvoice3/); [Paper](https://arxiv.org/abs/2505.17589); [CV3-Eval](https://github.com/FunAudioLLM/CV3-Eval)
+**Fun-CosyVoice 3.0**: [Demos](https://funaudiollm.github.io/cosyvoice3/); [Paper](https://arxiv.org/abs/2505.17589); [Modelscope](https://www.modelscope.cn/studios/FunAudioLLM/Fun-CosyVoice3-0.5B); [CV3-Eval](https://github.com/FunAudioLLM/CV3-Eval)
 
 **CosyVoice 2.0**: [Demos](https://funaudiollm.github.io/cosyvoice2/); [Paper](https://arxiv.org/abs/2412.10117); [Modelscope](https://www.modelscope.cn/studios/iic/CosyVoice2-0.5B); [HuggingFace](https://huggingface.co/spaces/FunAudioLLM/CosyVoice2-0.5B)
 
@@ -10,45 +10,43 @@
 
 ## Highlight🔥
 
-**CosyVoice 2.0** has been released! Compared to version 1.0, the new version offers more accurate, more stable, faster, and better speech generation capabilities.
-### Multilingual
-- **Supported Language**: Chinese, English, Japanese, Korean, Chinese dialects (Cantonese, Sichuanese, Shanghainese, Tianjinese, Wuhanese, etc.)
-- **Crosslingual & Mixlingual**：Support zero-shot voice cloning for cross-lingual and code-switching scenarios.
-### Ultra-Low Latency
-- **Bidirectional Streaming Support**: CosyVoice 2.0 integrates offline and streaming modeling technologies.
-- **Rapid First Packet Synthesis**: Achieves latency as low as 150ms while maintaining high-quality audio output.
-### High Accuracy
-- **Improved Pronunciation**: Reduces pronunciation errors by 30% to 50% compared to CosyVoice 1.0.
-- **Benchmark Achievements**: Attains the lowest character error rate on the hard test set of the Seed-TTS evaluation set.
-### Strong Stability
-- **Consistency in Timbre**: Ensures reliable voice consistency for zero-shot and cross-language speech synthesis.
-- **Cross-language Synthesis**: Marked improvements compared to version 1.0.
-### Natural Experience
-- **Enhanced Prosody and Sound Quality**: Improved alignment of synthesized audio, raising MOS evaluation scores from 5.4 to 5.53.
-- **Emotional and Dialectal Flexibility**: Now supports more granular emotional controls and accent adjustments.
+**Fun-CosyVoice 3.0** is an advanced text-to-speech (TTS) system based on large language models (LLM), surpassing its predecessor (CosyVoice 2.0) in content consistency, speaker similarity, and prosody naturalness. It is designed for zero-shot multilingual speech synthesis in the wild.
+### Key Features
+- **Language Coverage**: Covers 9 common languages (Chinese, English, Japanese, Korean, German, Spanish, French, Italian, Russian), 18+ Chinese dialects/accents (Guangdong, Minnan, Sichuan, Dongbei, Shan3xi, Shan1xi, Shanghai, Tianjin, Shan1dong, Ningxia, Gansu, etc.) and meanwhile supports both multi-lingual/cross-lingual zero-shot voice cloning.
+- **Content Consistency & Naturalness**: Achieves state-of-the-art performance in content consistency, speaker similarity, and prosody naturalness.
+- **Pronunciation Inpainting**: Supports pronunciation inpainting of Chinese Pinyin and English CMU phonemes, providing more controllability and thus suitable for production use.
+- **Text Normalization**: Supports reading of numbers, special symbols and various text formats without a traditional frontend module.
+- **Bi-Streaming**: Support both text-in streaming and audio-out streaming, and achieves latency as low as 150ms while maintaining high-quality audio output.
+- **Instruct Support**: Supports various instructions such as languages, dialects, emotions, speed, volume, etc.
+
 
 ## Roadmap
 
+- [x] 2025/12
+
+    - [x] release Fun-CosyVoice3-0.5B-2512 base model, rl model and its training/inference script
+    - [x] release Fun-CosyVoice3-0.5B modelscope gradio space
+
 - [x] 2025/08
 
     - [x] Thanks to the contribution from NVIDIA Yuekai Zhang, add triton trtllm runtime support and cosyvoice2 grpo training support
 
 - [x] 2025/07
 
-    - [x] release cosyvoice 3.0 eval set
+    - [x] release Fun-CosyVoice 3.0 eval set
 
 - [x] 2025/05
 
-    - [x] add cosyvoice 2.0 vllm support
+    - [x] add CosyVoice2-0.5B vllm support
 
 - [x] 2024/12
 
-    - [x] 25hz cosyvoice 2.0 released
+    - [x] 25hz CosyVoice2-0.5B released
 
 - [x] 2024/09
 
-    - [x] 25hz cosyvoice base model
-    - [x] 25hz cosyvoice voice conversion model
+    - [x] 25hz CosyVoice-300M base model
+    - [x] 25hz CosyVoice-300M voice conversion function
 
 - [x] 2024/08
 
@@ -61,6 +59,25 @@
     - [x] WeTextProcessing support when ttsfrd is not available
     - [x] Fastapi server and client
 
+## Evaluation
+| Model | CER (%) ↓ (test-zh) | WER (%) ↓ (test-en) | CER (%) ↓ (test-hard) |
+|-----|------------------|------------------|------------------|
+| Human | 1.26 | 2.14 | - |
+| F5-TTS | 1.53 | 2.00 | 8.67 |
+| SparkTTS | 1.20 | 1.98 | - |
+| Seed-TTS | 1.12 | 2.25 | 7.59 |
+| CosyVoice2 | 1.45 | 2.57 | 6.83 |
+| FireRedTTS-2 | 1.14 | 1.95 | - |
+| IndexTTS2 | 1.01 | 1.52 | 7.12 |
+| VibeVoice | 1.16 | 3.04 | - |
+| HiggsAudio | 1.79 | 2.44 | - |
+| MiniMax-Speech | 0.83 | 1.65 | - |
+| VoxPCM | 0.93 | 1.85 | 8.87 |
+| GLM-TTS | 1.03 | - | - |
+| GLM-TTS_RL | 0.89 | - | - |
+| Fun-CosyVoice3-0.5B-2512 | 1.21 |  2.24 | 6.71 |
+| Fun-CosyVoice3-0.5B-2512_RL | 0.81 | 1.68 | 5.44 |
+
 
 ## Install
 
@@ -91,11 +108,12 @@
 
 ### Model download
 
-We strongly recommend that you download our pretrained `CosyVoice2-0.5B` `CosyVoice-300M` `CosyVoice-300M-SFT` `CosyVoice-300M-Instruct` model and `CosyVoice-ttsfrd` resource.
+We strongly recommend that you download our pretrained `Fun-CosyVoice3-0.5B` `CosyVoice2-0.5B` `CosyVoice-300M` `CosyVoice-300M-SFT` `CosyVoice-300M-Instruct` model and `CosyVoice-ttsfrd` resource.
 
 ``` python
 # SDK模型下载
 from modelscope import snapshot_download
+snapshot_download('FunAudioLLM/Fun-CosyVoice3-0.5B-2512', local_dir='pretrained_models/Fun-CosyVoice3-0.5B')
 snapshot_download('iic/CosyVoice2-0.5B', local_dir='pretrained_models/CosyVoice2-0.5B')
 snapshot_download('iic/CosyVoice-300M', local_dir='pretrained_models/CosyVoice-300M')
 snapshot_download('iic/CosyVoice-300M-SFT', local_dir='pretrained_models/CosyVoice-300M-SFT')
@@ -103,16 +121,6 @@ snapshot_download('iic/CosyVoice-300M-Instruct', local_dir='pretrained_models/Co
 snapshot_download('iic/CosyVoice-ttsfrd', local_dir='pretrained_models/CosyVoice-ttsfrd')
 ```
 
-``` sh
-# git模型下载，请确保已安装git lfs
-mkdir -p pretrained_models
-git clone https://www.modelscope.cn/iic/CosyVoice2-0.5B.git pretrained_models/CosyVoice2-0.5B
-git clone https://www.modelscope.cn/iic/CosyVoice-300M.git pretrained_models/CosyVoice-300M
-git clone https://www.modelscope.cn/iic/CosyVoice-300M-SFT.git pretrained_models/CosyVoice-300M-SFT
-git clone https://www.modelscope.cn/iic/CosyVoice-300M-Instruct.git pretrained_models/CosyVoice-300M-Instruct
-git clone https://www.modelscope.cn/iic/CosyVoice-ttsfrd.git pretrained_models/CosyVoice-ttsfrd
-```
-
 Optionally, you can unzip `ttsfrd` resource and install `ttsfrd` package for better text normalization performance.
 
 Notice that this step is not necessary. If you do not install `ttsfrd` package, we will use wetext by default.
@@ -126,50 +134,10 @@ pip install ttsfrd-0.4.2-cp310-cp310-linux_x86_64.whl
 
 ### Basic Usage
 
-We strongly recommend using `CosyVoice2-0.5B` for better performance.
-Follow the code below for detailed usage of each model.
-
-``` python
-import sys
-sys.path.append('third_party/Matcha-TTS')
-from cosyvoice.cli.cosyvoice import CosyVoice, CosyVoice2
-from cosyvoice.utils.file_utils import load_wav
-import torchaudio
-```
-
-#### CosyVoice2 Usage
-```python
-cosyvoice = CosyVoice2('pretrained_models/CosyVoice2-0.5B', load_jit=False, load_trt=False, load_vllm=False, fp16=False)
-
-# NOTE if you want to reproduce the results on https://funaudiollm.github.io/cosyvoice2, please add text_frontend=False during inference
-# zero_shot usage
-prompt_speech_16k = load_wav('./asset/zero_shot_prompt.wav', 16000)
-for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '希望你以后能够做的比我还好呦。', prompt_speech_16k, stream=False)):
-    torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-
-# save zero_shot spk for future usage
-assert cosyvoice.add_zero_shot_spk('希望你以后能够做的比我还好呦。', prompt_speech_16k, 'my_zero_shot_spk') is True
-for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '', '', zero_shot_spk_id='my_zero_shot_spk', stream=False)):
-    torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-cosyvoice.save_spkinfo()
-
-# fine grained control, for supported control, check cosyvoice/tokenizer/tokenizer.py#L248
-for i, j in enumerate(cosyvoice.inference_cross_lingual('在他讲述那个荒诞故事的过程中，他突然[laughter]停下来，因为他自己也被逗笑了[laughter]。', prompt_speech_16k, stream=False)):
-    torchaudio.save('fine_grained_control_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-
-# instruct usage
-for i, j in enumerate(cosyvoice.inference_instruct2('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '用四川话说这句话', prompt_speech_16k, stream=False)):
-    torchaudio.save('instruct_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-
-# bistream usage, you can use generator as input, this is useful when using text llm model as input
-# NOTE you should still have some basic sentence split logic because llm can not handle arbitrary sentence length
-def text_generator():
-    yield '收到好友从远方寄来的生日礼物，'
-    yield '那份意外的惊喜与深深的祝福'
-    yield '让我心中充满了甜蜜的快乐，'
-    yield '笑容如花儿般绽放。'
-for i, j in enumerate(cosyvoice.inference_zero_shot(text_generator(), '希望你以后能够做的比我还好呦。', prompt_speech_16k, stream=False)):
-    torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+We strongly recommend using `Fun-CosyVoice3-0.5B` for better performance.
+Follow the code in `example.py` for detailed usage of each model.
+```sh
+python example.py
 ```
 
 #### CosyVoice2 vllm Usage
@@ -184,36 +152,6 @@ pip install vllm==v0.9.0 transformers==4.51.3 -i https://mirrors.aliyun.com/pypi
 python vllm_example.py
 ```
 
-#### CosyVoice Usage
-```python
-cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M-SFT', load_jit=False, load_trt=False, fp16=False)
-# sft usage
-print(cosyvoice.list_available_spks())
-# change stream=True for chunk stream inference
-for i, j in enumerate(cosyvoice.inference_sft('你好，我是通义生成式语音大模型，请问有什么可以帮您的吗？', '中文女', stream=False)):
-    torchaudio.save('sft_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-
-cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M')
-# zero_shot usage, <|zh|><|en|><|jp|><|yue|><|ko|> for Chinese/English/Japanese/Cantonese/Korean
-prompt_speech_16k = load_wav('./asset/zero_shot_prompt.wav', 16000)
-for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '希望你以后能够做的比我还好呦。', prompt_speech_16k, stream=False)):
-    torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-# cross_lingual usage
-prompt_speech_16k = load_wav('./asset/cross_lingual_prompt.wav', 16000)
-for i, j in enumerate(cosyvoice.inference_cross_lingual('<|en|>And then later on, fully acquiring that company. So keeping management in line, interest in line with the asset that\'s coming into the family is a reason why sometimes we don\'t buy the whole thing.', prompt_speech_16k, stream=False)):
-    torchaudio.save('cross_lingual_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-# vc usage
-prompt_speech_16k = load_wav('./asset/zero_shot_prompt.wav', 16000)
-source_speech_16k = load_wav('./asset/cross_lingual_prompt.wav', 16000)
-for i, j in enumerate(cosyvoice.inference_vc(source_speech_16k, prompt_speech_16k, stream=False)):
-    torchaudio.save('vc_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-
-cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M-Instruct')
-# instruct usage, support <laughter></laughter><strong></strong>[laughter][breath]
-for i, j in enumerate(cosyvoice.inference_instruct('在面对挑战时，他展现了非凡的<strong>勇气</strong>与<strong>智慧</strong>。', '中文男', 'Theo \'Crimson\', is a fiery, passionate rebel leader. Fights with fervor for justice, but struggles with impulsiveness.', stream=False)):
-    torchaudio.save('instruct_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
-```
-
 #### Start web demo
 
 You can use our web demo page to get familiar with CosyVoice quickly.

cosyvoice/bin/export_jit.py

@@ -23,8 +23,10 @@ import torch
 ROOT_DIR = os.path.dirname(os.path.abspath(__file__))
 sys.path.append('{}/../..'.format(ROOT_DIR))
 sys.path.append('{}/../../third_party/Matcha-TTS'.format(ROOT_DIR))
-from cosyvoice.cli.cosyvoice import CosyVoice, CosyVoice2
+from cosyvoice.cli.cosyvoice import AutoModel
+from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model, CosyVoice3Model
 from cosyvoice.utils.file_utils import logging
+from cosyvoice.utils.class_utils import get_model_type
 
 
 def get_args():
@@ -57,15 +59,17 @@ def main():
     torch._C._jit_set_profiling_mode(False)
     torch._C._jit_set_profiling_executor(False)
 
-    try:
-        model = CosyVoice(args.model_dir)
-    except Exception:
-        try:
-            model = CosyVoice2(args.model_dir)
-        except Exception:
-            raise TypeError('no valid model_type!')
+    model = AutoModel(model_dir=args.model_dir)
 
-    if not isinstance(model, CosyVoice2):
+    if get_model_type(model.model) == CosyVoiceModel:
+        # 1. export flow encoder
+        flow_encoder = model.model.flow.encoder
+        script = get_optimized_script(flow_encoder)
+        script.save('{}/flow.encoder.fp32.zip'.format(args.model_dir))
+        script = get_optimized_script(flow_encoder.half())
+        script.save('{}/flow.encoder.fp16.zip'.format(args.model_dir))
+        logging.info('successfully export flow_encoder')
+    elif get_model_type(model.model) == CosyVoice2Model:
         # 1. export llm text_encoder
         llm_text_encoder = model.model.llm.text_encoder
         script = get_optimized_script(llm_text_encoder)
@@ -90,13 +94,7 @@ def main():
         script.save('{}/flow.encoder.fp16.zip'.format(args.model_dir))
         logging.info('successfully export flow_encoder')
     else:
-        # 3. export flow encoder
-        flow_encoder = model.model.flow.encoder
-        script = get_optimized_script(flow_encoder)
-        script.save('{}/flow.encoder.fp32.zip'.format(args.model_dir))
-        script = get_optimized_script(flow_encoder.half())
-        script.save('{}/flow.encoder.fp16.zip'.format(args.model_dir))
-        logging.info('successfully export flow_encoder')
+        raise ValueError('unsupported model type')
 
 
 if __name__ == '__main__':

cosyvoice/bin/export_onnx.py

@@ -27,7 +27,7 @@ from tqdm import tqdm
 ROOT_DIR = os.path.dirname(os.path.abspath(__file__))
 sys.path.append('{}/../..'.format(ROOT_DIR))
 sys.path.append('{}/../../third_party/Matcha-TTS'.format(ROOT_DIR))
-from cosyvoice.cli.cosyvoice import CosyVoice, CosyVoice2
+from cosyvoice.cli.cosyvoice import AutoModel
 from cosyvoice.utils.file_utils import logging
 
 
@@ -58,13 +58,7 @@ def main():
     logging.basicConfig(level=logging.DEBUG,
                         format='%(asctime)s %(levelname)s %(message)s')
 
-    try:
-        model = CosyVoice(args.model_dir)
-    except Exception:
-        try:
-            model = CosyVoice2(args.model_dir)
-        except Exception:
-            raise TypeError('no valid model_type!')
+    model = AutoModel(model_dir=args.model_dir)
 
     # 1. export flow decoder estimator
     estimator = model.model.flow.decoder.estimator

cosyvoice/bin/inference_deprecated.py

@@ -1,126 +0,0 @@
-# Copyright (c) 2024 Alibaba Inc (authors: Xiang Lyu)
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#   http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-from __future__ import print_function
-
-import argparse
-import logging
-logging.getLogger('matplotlib').setLevel(logging.WARNING)
-import os
-import torch
-from torch.utils.data import DataLoader
-import torchaudio
-from hyperpyyaml import load_hyperpyyaml
-from tqdm import tqdm
-from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model
-from cosyvoice.dataset.dataset import Dataset
-
-
-def get_args():
-    parser = argparse.ArgumentParser(description='inference with your model')
-    parser.add_argument('--config', required=True, help='config file')
-    parser.add_argument('--prompt_data', required=True, help='prompt data file')
-    parser.add_argument('--prompt_utt2data', required=True, help='prompt data file')
-    parser.add_argument('--tts_text', required=True, help='tts input file')
-    parser.add_argument('--qwen_pretrain_path', required=False, help='qwen pretrain path')
-    parser.add_argument('--llm_model', required=True, help='llm model file')
-    parser.add_argument('--flow_model', required=True, help='flow model file')
-    parser.add_argument('--hifigan_model', required=True, help='hifigan model file')
-    parser.add_argument('--gpu',
-                        type=int,
-                        default=-1,
-                        help='gpu id for this rank, -1 for cpu')
-    parser.add_argument('--mode',
-                        default='sft',
-                        choices=['sft', 'zero_shot'],
-                        help='inference mode')
-    parser.add_argument('--result_dir', required=True, help='asr result file')
-    args = parser.parse_args()
-    print(args)
-    return args
-
-
-def main():
-    args = get_args()
-    logging.basicConfig(level=logging.DEBUG,
-                        format='%(asctime)s %(levelname)s %(message)s')
-    os.environ['CUDA_VISIBLE_DEVICES'] = str(args.gpu)
-
-    # Init cosyvoice models from configs
-    use_cuda = args.gpu >= 0 and torch.cuda.is_available()
-    device = torch.device('cuda' if use_cuda else 'cpu')
-    try:
-        with open(args.config, 'r') as f:
-            configs = load_hyperpyyaml(f, overrides={'qwen_pretrain_path': args.qwen_pretrain_path})
-        model = CosyVoice2Model(configs['llm'], configs['flow'], configs['hift'])
-    except Exception:
-        try:
-            with open(args.config, 'r') as f:
-                configs = load_hyperpyyaml(f)
-            model = CosyVoiceModel(configs['llm'], configs['flow'], configs['hift'])
-        except Exception:
-            raise TypeError('no valid model_type!')
-
-    model.load(args.llm_model, args.flow_model, args.hifigan_model)
-
-    test_dataset = Dataset(args.prompt_data, data_pipeline=configs['data_pipeline'], mode='inference', shuffle=False, partition=False,
-                           tts_file=args.tts_text, prompt_utt2data=args.prompt_utt2data)
-    test_data_loader = DataLoader(test_dataset, batch_size=None, num_workers=0)
-
-    sample_rate = configs['sample_rate']
-    del configs
-    os.makedirs(args.result_dir, exist_ok=True)
-    fn = os.path.join(args.result_dir, 'wav.scp')
-    f = open(fn, 'w')
-    with torch.no_grad():
-        for _, batch in tqdm(enumerate(test_data_loader)):
-            utts = batch["utts"]
-            assert len(utts) == 1, "inference mode only support batchsize 1"
-            text_token = batch["text_token"].to(device)
-            text_token_len = batch["text_token_len"].to(device)
-            tts_index = batch["tts_index"]
-            tts_text_token = batch["tts_text_token"].to(device)
-            tts_text_token_len = batch["tts_text_token_len"].to(device)
-            speech_token = batch["speech_token"].to(device)
-            speech_token_len = batch["speech_token_len"].to(device)
-            speech_feat = batch["speech_feat"].to(device)
-            speech_feat_len = batch["speech_feat_len"].to(device)
-            utt_embedding = batch["utt_embedding"].to(device)
-            spk_embedding = batch["spk_embedding"].to(device)
-            if args.mode == 'sft':
-                model_input = {'text': tts_text_token, 'text_len': tts_text_token_len,
-                               'llm_embedding': spk_embedding, 'flow_embedding': spk_embedding}
-            else:
-                model_input = {'text': tts_text_token, 'text_len': tts_text_token_len,
-                               'prompt_text': text_token, 'prompt_text_len': text_token_len,
-                               'llm_prompt_speech_token': speech_token, 'llm_prompt_speech_token_len': speech_token_len,
-                               'flow_prompt_speech_token': speech_token, 'flow_prompt_speech_token_len': speech_token_len,
-                               'prompt_speech_feat': speech_feat, 'prompt_speech_feat_len': speech_feat_len,
-                               'llm_embedding': utt_embedding, 'flow_embedding': utt_embedding}
-            tts_speeches = []
-            for model_output in model.tts(**model_input):
-                tts_speeches.append(model_output['tts_speech'])
-            tts_speeches = torch.concat(tts_speeches, dim=1)
-            tts_key = '{}_{}'.format(utts[0], tts_index[0])
-            tts_fn = os.path.join(args.result_dir, '{}.wav'.format(tts_key))
-            torchaudio.save(tts_fn, tts_speeches, sample_rate=sample_rate, backend='soundfile')
-            f.write('{} {}\n'.format(tts_key, tts_fn))
-            f.flush()
-    f.close()
-    logging.info('Result wav.scp saved in {}'.format(fn))
-
-
-if __name__ == '__main__':
-    logging.warning('this code has been deprecated, please refer to README for CosyVoice inference usage!')
-    main()

cosyvoice/cli/cosyvoice.py

@@ -19,7 +19,7 @@ from hyperpyyaml import load_hyperpyyaml
 from modelscope import snapshot_download
 import torch
 from cosyvoice.cli.frontend import CosyVoiceFrontEnd
-from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model
+from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model, CosyVoice3Model
 from cosyvoice.utils.file_utils import logging
 from cosyvoice.utils.class_utils import get_model_type
 
@@ -27,7 +27,6 @@ from cosyvoice.utils.class_utils import get_model_type
 class CosyVoice:
 
     def __init__(self, model_dir, load_jit=False, load_trt=False, fp16=False, trt_concurrent=1):
-        self.instruct = True if '-Instruct' in model_dir else False
         self.model_dir = model_dir
         self.fp16 = fp16
         if not os.path.exists(model_dir):
@@ -37,7 +36,7 @@ class CosyVoice:
             raise ValueError('{} not found!'.format(hyper_yaml_path))
         with open(hyper_yaml_path, 'r') as f:
             configs = load_hyperpyyaml(f)
-        assert get_model_type(configs) != CosyVoice2Model, 'do not use {} for CosyVoice initialization!'.format(model_dir)
+        assert get_model_type(configs) == CosyVoiceModel, 'do not use {} for CosyVoice initialization!'.format(model_dir)
         self.frontend = CosyVoiceFrontEnd(configs['get_tokenizer'],
                                           configs['feat_extractor'],
                                           '{}/campplus.onnx'.format(model_dir),
@@ -67,9 +66,9 @@ class CosyVoice:
         spks = list(self.frontend.spk2info.keys())
         return spks
 
-    def add_zero_shot_spk(self, prompt_text, prompt_speech_16k, zero_shot_spk_id):
+    def add_zero_shot_spk(self, prompt_text, prompt_wav, zero_shot_spk_id):
         assert zero_shot_spk_id != '', 'do not use empty zero_shot_spk_id'
-        model_input = self.frontend.frontend_zero_shot('', prompt_text, prompt_speech_16k, self.sample_rate, '')
+        model_input = self.frontend.frontend_zero_shot('', prompt_text, prompt_wav, self.sample_rate, '')
         del model_input['text']
         del model_input['text_len']
         self.frontend.spk2info[zero_shot_spk_id] = model_input
@@ -89,12 +88,12 @@ class CosyVoice:
                 yield model_output
                 start_time = time.time()
 
-    def inference_zero_shot(self, tts_text, prompt_text, prompt_speech_16k, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
+    def inference_zero_shot(self, tts_text, prompt_text, prompt_wav, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
         prompt_text = self.frontend.text_normalize(prompt_text, split=False, text_frontend=text_frontend)
         for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
             if (not isinstance(i, Generator)) and len(i) < 0.5 * len(prompt_text):
                 logging.warning('synthesis text {} too short than prompt text {}, this may lead to bad performance'.format(i, prompt_text))
-            model_input = self.frontend.frontend_zero_shot(i, prompt_text, prompt_speech_16k, self.sample_rate, zero_shot_spk_id)
+            model_input = self.frontend.frontend_zero_shot(i, prompt_text, prompt_wav, self.sample_rate, zero_shot_spk_id)
             start_time = time.time()
             logging.info('synthesis text {}'.format(i))
             for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
@@ -103,9 +102,9 @@ class CosyVoice:
                 yield model_output
                 start_time = time.time()
 
-    def inference_cross_lingual(self, tts_text, prompt_speech_16k, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
+    def inference_cross_lingual(self, tts_text, prompt_wav, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
         for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
-            model_input = self.frontend.frontend_cross_lingual(i, prompt_speech_16k, self.sample_rate, zero_shot_spk_id)
+            model_input = self.frontend.frontend_cross_lingual(i, prompt_wav, self.sample_rate, zero_shot_spk_id)
             start_time = time.time()
             logging.info('synthesis text {}'.format(i))
             for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
@@ -116,8 +115,6 @@ class CosyVoice:
 
     def inference_instruct(self, tts_text, spk_id, instruct_text, stream=False, speed=1.0, text_frontend=True):
         assert isinstance(self.model, CosyVoiceModel), 'inference_instruct is only implemented for CosyVoice!'
-        if self.instruct is False:
-            raise ValueError('{} do not support instruct inference'.format(self.model_dir))
         instruct_text = self.frontend.text_normalize(instruct_text, split=False, text_frontend=text_frontend)
         for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
             model_input = self.frontend.frontend_instruct(i, spk_id, instruct_text)
@@ -129,8 +126,8 @@ class CosyVoice:
                 yield model_output
                 start_time = time.time()
 
-    def inference_vc(self, source_speech_16k, prompt_speech_16k, stream=False, speed=1.0):
-        model_input = self.frontend.frontend_vc(source_speech_16k, prompt_speech_16k, self.sample_rate)
+    def inference_vc(self, source_wav, prompt_wav, stream=False, speed=1.0):
+        model_input = self.frontend.frontend_vc(source_wav, prompt_wav, self.sample_rate)
         start_time = time.time()
         for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
             speech_len = model_output['tts_speech'].shape[1] / self.sample_rate
@@ -142,7 +139,6 @@ class CosyVoice:
 class CosyVoice2(CosyVoice):
 
     def __init__(self, model_dir, load_jit=False, load_trt=False, load_vllm=False, fp16=False, trt_concurrent=1):
-        self.instruct = True if '-Instruct' in model_dir else False
         self.model_dir = model_dir
         self.fp16 = fp16
         if not os.path.exists(model_dir):
@@ -160,9 +156,9 @@ class CosyVoice2(CosyVoice):
                                           '{}/spk2info.pt'.format(model_dir),
                                           configs['allowed_special'])
         self.sample_rate = configs['sample_rate']
-        if torch.cuda.is_available() is False and (load_jit is True or load_trt is True or fp16 is True):
-            load_jit, load_trt, fp16 = False, False, False
-            logging.warning('no cuda device, set load_jit/load_trt/fp16 to False')
+        if torch.cuda.is_available() is False and (load_jit is True or load_trt is True or load_vllm is True or fp16 is True):
+            load_jit, load_trt, load_vllm, fp16 = False, False, False, False
+            logging.warning('no cuda device, set load_jit/load_trt/load_vllm/fp16 to False')
         self.model = CosyVoice2Model(configs['llm'], configs['flow'], configs['hift'], fp16)
         self.model.load('{}/llm.pt'.format(model_dir),
                         '{}/flow.pt'.format(model_dir),
@@ -178,13 +174,9 @@ class CosyVoice2(CosyVoice):
                                 self.fp16)
         del configs
 
-    def inference_instruct(self, *args, **kwargs):
-        raise NotImplementedError('inference_instruct is not implemented for CosyVoice2!')
-
-    def inference_instruct2(self, tts_text, instruct_text, prompt_speech_16k, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
-        assert isinstance(self.model, CosyVoice2Model), 'inference_instruct2 is only implemented for CosyVoice2!'
+    def inference_instruct2(self, tts_text, instruct_text, prompt_wav, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
         for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
-            model_input = self.frontend.frontend_instruct2(i, instruct_text, prompt_speech_16k, self.sample_rate, zero_shot_spk_id)
+            model_input = self.frontend.frontend_instruct2(i, instruct_text, prompt_wav, self.sample_rate, zero_shot_spk_id)
             start_time = time.time()
             logging.info('synthesis text {}'.format(i))
             for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
@@ -192,3 +184,55 @@ class CosyVoice2(CosyVoice):
                 logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
                 yield model_output
                 start_time = time.time()
+
+
+class CosyVoice3(CosyVoice2):
+
+    def __init__(self, model_dir, load_trt=False, load_vllm=False, fp16=False, trt_concurrent=1):
+        self.model_dir = model_dir
+        self.fp16 = fp16
+        if not os.path.exists(model_dir):
+            model_dir = snapshot_download(model_dir)
+        hyper_yaml_path = '{}/cosyvoice3.yaml'.format(model_dir)
+        if not os.path.exists(hyper_yaml_path):
+            raise ValueError('{} not found!'.format(hyper_yaml_path))
+        with open(hyper_yaml_path, 'r') as f:
+            configs = load_hyperpyyaml(f, overrides={'qwen_pretrain_path': os.path.join(model_dir, 'CosyVoice-BlankEN')})
+        assert get_model_type(configs) == CosyVoice3Model, 'do not use {} for CosyVoice3 initialization!'.format(model_dir)
+        self.frontend = CosyVoiceFrontEnd(configs['get_tokenizer'],
+                                          configs['feat_extractor'],
+                                          '{}/campplus.onnx'.format(model_dir),
+                                          '{}/speech_tokenizer_v3.onnx'.format(model_dir),
+                                          '{}/spk2info.pt'.format(model_dir),
+                                          configs['allowed_special'])
+        self.sample_rate = configs['sample_rate']
+        if torch.cuda.is_available() is False and (load_trt is True or fp16 is True):
+            load_trt, fp16 = False, False
+            logging.warning('no cuda device, set load_trt/fp16 to False')
+        self.model = CosyVoice3Model(configs['llm'], configs['flow'], configs['hift'], fp16)
+        self.model.load('{}/llm.pt'.format(model_dir),
+                        '{}/flow.pt'.format(model_dir),
+                        '{}/hift.pt'.format(model_dir))
+        if load_vllm:
+            self.model.load_vllm('{}/vllm'.format(model_dir))
+        if load_trt:
+            if self.fp16 is True:
+                logging.warning('DiT tensorRT fp16 engine have some performance issue, use at caution!')
+            self.model.load_trt('{}/flow.decoder.estimator.{}.mygpu.plan'.format(model_dir, 'fp16' if self.fp16 is True else 'fp32'),
+                                '{}/flow.decoder.estimator.fp32.onnx'.format(model_dir),
+                                trt_concurrent,
+                                self.fp16)
+        del configs
+
+
+def AutoModel(**kwargs):
+    if not os.path.exists(kwargs['model_dir']):
+        kwargs['model_dir'] = snapshot_download(kwargs['model_dir'])
+    if os.path.exists('{}/cosyvoice.yaml'.format(kwargs['model_dir'])):
+        return CosyVoice(**kwargs)
+    elif os.path.exists('{}/cosyvoice2.yaml'.format(kwargs['model_dir'])):
+        return CosyVoice2(**kwargs)
+    elif os.path.exists('{}/cosyvoice3.yaml'.format(kwargs['model_dir'])):
+        return CosyVoice3(**kwargs)
+    else:
+        raise TypeError('No valid model type found!')

cosyvoice/cli/frontend.py

@@ -32,7 +32,7 @@ except ImportError:
     from wetext import Normalizer as ZhNormalizer
     from wetext import Normalizer as EnNormalizer
     use_ttsfrd = False
-from cosyvoice.utils.file_utils import logging
+from cosyvoice.utils.file_utils import logging, load_wav
 from cosyvoice.utils.frontend_utils import contains_chinese, replace_blank, replace_corner_mark, remove_bracket, spell_out_number, split_paragraph, is_only_punctuation
 
 
@@ -89,7 +89,8 @@ class CosyVoiceFrontEnd:
             for i in range(text_token.shape[1]):
                 yield text_token[:, i: i + 1]
 
-    def _extract_speech_token(self, speech):
+    def _extract_speech_token(self, prompt_wav):
+        speech = load_wav(prompt_wav, 16000)
         assert speech.shape[1] / 16000 <= 30, 'do not support extract speech token for audio longer than 30s'
         feat = whisper.log_mel_spectrogram(speech, n_mels=128)
         speech_token = self.speech_tokenizer_session.run(None,
@@ -101,7 +102,8 @@ class CosyVoiceFrontEnd:
         speech_token_len = torch.tensor([speech_token.shape[1]], dtype=torch.int32).to(self.device)
         return speech_token, speech_token_len
 
-    def _extract_spk_embedding(self, speech):
+    def _extract_spk_embedding(self, prompt_wav):
+        speech = load_wav(prompt_wav, 16000)
         feat = kaldi.fbank(speech,
                            num_mel_bins=80,
                            dither=0,
@@ -112,7 +114,8 @@ class CosyVoiceFrontEnd:
         embedding = torch.tensor([embedding]).to(self.device)
         return embedding
 
-    def _extract_speech_feat(self, speech):
+    def _extract_speech_feat(self, prompt_wav):
+        speech = load_wav(prompt_wav, 24000)
         speech_feat = self.feat_extractor(speech).squeeze(dim=0).transpose(0, 1).to(self.device)
         speech_feat = speech_feat.unsqueeze(dim=0)
         speech_feat_len = torch.tensor([speech_feat.shape[1]], dtype=torch.int32).to(self.device)
@@ -122,6 +125,9 @@ class CosyVoiceFrontEnd:
         if isinstance(text, Generator):
             logging.info('get tts_text generator, will skip text_normalize!')
             return [text]
+        # NOTE skip text_frontend when ssml symbol in text
+        if '<|' in text and '|>' in text:
+            text_frontend = False
         if text_frontend is False or text == '':
             return [text] if split is True else text
         text = text.strip()
@@ -154,19 +160,18 @@ class CosyVoiceFrontEnd:
         model_input = {'text': tts_text_token, 'text_len': tts_text_token_len, 'llm_embedding': embedding, 'flow_embedding': embedding}
         return model_input
 
-    def frontend_zero_shot(self, tts_text, prompt_text, prompt_speech_16k, resample_rate, zero_shot_spk_id):
+    def frontend_zero_shot(self, tts_text, prompt_text, prompt_wav, resample_rate, zero_shot_spk_id):
         tts_text_token, tts_text_token_len = self._extract_text_token(tts_text)
         if zero_shot_spk_id == '':
             prompt_text_token, prompt_text_token_len = self._extract_text_token(prompt_text)
-            prompt_speech_resample = torchaudio.transforms.Resample(orig_freq=16000, new_freq=resample_rate)(prompt_speech_16k)
-            speech_feat, speech_feat_len = self._extract_speech_feat(prompt_speech_resample)
-            speech_token, speech_token_len = self._extract_speech_token(prompt_speech_16k)
+            speech_feat, speech_feat_len = self._extract_speech_feat(prompt_wav)
+            speech_token, speech_token_len = self._extract_speech_token(prompt_wav)
             if resample_rate == 24000:
                 # cosyvoice2, force speech_feat % speech_token = 2
                 token_len = min(int(speech_feat.shape[1] / 2), speech_token.shape[1])
                 speech_feat, speech_feat_len[:] = speech_feat[:, :2 * token_len], 2 * token_len
                 speech_token, speech_token_len[:] = speech_token[:, :token_len], token_len
-            embedding = self._extract_spk_embedding(prompt_speech_16k)
+            embedding = self._extract_spk_embedding(prompt_wav)
             model_input = {'prompt_text': prompt_text_token, 'prompt_text_len': prompt_text_token_len,
                            'llm_prompt_speech_token': speech_token, 'llm_prompt_speech_token_len': speech_token_len,
                            'flow_prompt_speech_token': speech_token, 'flow_prompt_speech_token_len': speech_token_len,
@@ -178,8 +183,8 @@ class CosyVoiceFrontEnd:
         model_input['text_len'] = tts_text_token_len
         return model_input
 
-    def frontend_cross_lingual(self, tts_text, prompt_speech_16k, resample_rate, zero_shot_spk_id):
-        model_input = self.frontend_zero_shot(tts_text, '', prompt_speech_16k, resample_rate, zero_shot_spk_id)
+    def frontend_cross_lingual(self, tts_text, prompt_wav, resample_rate, zero_shot_spk_id):
+        model_input = self.frontend_zero_shot(tts_text, '', prompt_wav, resample_rate, zero_shot_spk_id)
         # in cross lingual mode, we remove prompt in llm
         del model_input['prompt_text']
         del model_input['prompt_text_len']
@@ -191,22 +196,21 @@ class CosyVoiceFrontEnd:
         model_input = self.frontend_sft(tts_text, spk_id)
         # in instruct mode, we remove spk_embedding in llm due to information leakage
         del model_input['llm_embedding']
-        instruct_text_token, instruct_text_token_len = self._extract_text_token(instruct_text + '<endofprompt>')
+        instruct_text_token, instruct_text_token_len = self._extract_text_token(instruct_text)
         model_input['prompt_text'] = instruct_text_token
         model_input['prompt_text_len'] = instruct_text_token_len
         return model_input
 
-    def frontend_instruct2(self, tts_text, instruct_text, prompt_speech_16k, resample_rate, zero_shot_spk_id):
-        model_input = self.frontend_zero_shot(tts_text, instruct_text + '<|endofprompt|>', prompt_speech_16k, resample_rate, zero_shot_spk_id)
+    def frontend_instruct2(self, tts_text, instruct_text, prompt_wav, resample_rate, zero_shot_spk_id):
+        model_input = self.frontend_zero_shot(tts_text, instruct_text, prompt_wav, resample_rate, zero_shot_spk_id)
         del model_input['llm_prompt_speech_token']
         del model_input['llm_prompt_speech_token_len']
         return model_input
 
-    def frontend_vc(self, source_speech_16k, prompt_speech_16k, resample_rate):
-        prompt_speech_token, prompt_speech_token_len = self._extract_speech_token(prompt_speech_16k)
-        prompt_speech_resample = torchaudio.transforms.Resample(orig_freq=16000, new_freq=resample_rate)(prompt_speech_16k)
-        prompt_speech_feat, prompt_speech_feat_len = self._extract_speech_feat(prompt_speech_resample)
-        embedding = self._extract_spk_embedding(prompt_speech_16k)
+    def frontend_vc(self, source_speech_16k, prompt_wav, resample_rate):
+        prompt_speech_token, prompt_speech_token_len = self._extract_speech_token(prompt_wav)
+        prompt_speech_feat, prompt_speech_feat_len = self._extract_speech_feat(prompt_wav)
+        embedding = self._extract_spk_embedding(prompt_wav)
         source_speech_token, source_speech_token_len = self._extract_speech_token(source_speech_16k)
         model_input = {'source_speech_token': source_speech_token, 'source_speech_token_len': source_speech_token_len,
                        'flow_prompt_speech_token': prompt_speech_token, 'flow_prompt_speech_token_len': prompt_speech_token_len,

cosyvoice/cli/model.py

@@ -38,9 +38,6 @@ class CosyVoiceModel:
         self.flow = flow
         self.hift = hift
         self.fp16 = fp16
-        if self.fp16 is True:
-            self.llm.half()
-            self.flow.half()
         self.token_min_hop_len = 2 * self.flow.input_frame_rate
         self.token_max_hop_len = 4 * self.flow.input_frame_rate
         self.token_overlap_len = 20
@@ -129,7 +126,7 @@ class CosyVoiceModel:
 
     def token2wav(self, token, prompt_token, prompt_feat, embedding, uuid, finalize=False, speed=1.0):
         with torch.cuda.amp.autocast(self.fp16):
-            tts_mel, self.flow_cache_dict[uuid] = self.flow.inference(token=token.to(self.device),
+            tts_mel, self.flow_cache_dict[uuid] = self.flow.inference(token=token.to(self.device, dtype=torch.int32),
                                                                       token_len=torch.tensor([token.shape[1]], dtype=torch.int32).to(self.device),
                                                                       prompt_token=prompt_token.to(self.device),
                                                                       prompt_token_len=torch.tensor([prompt_token.shape[1]], dtype=torch.int32).to(self.device),
@@ -249,9 +246,6 @@ class CosyVoice2Model(CosyVoiceModel):
         self.flow = flow
         self.hift = hift
         self.fp16 = fp16
-        if self.fp16 is True:
-            self.llm.half()
-            self.flow.half()
         # NOTE must matching training static_chunk_size
         self.token_hop_len = 25
         # hift cache
@@ -284,7 +278,7 @@ class CosyVoice2Model(CosyVoiceModel):
 
     def token2wav(self, token, prompt_token, prompt_feat, embedding, token_offset, uuid, stream=False, finalize=False, speed=1.0):
         with torch.cuda.amp.autocast(self.fp16):
-            tts_mel, _ = self.flow.inference(token=token.to(self.device),
+            tts_mel, _ = self.flow.inference(token=token.to(self.device, dtype=torch.int32),
                                              token_len=torch.tensor([token.shape[1]], dtype=torch.int32).to(self.device),
                                              prompt_token=prompt_token.to(self.device),
                                              prompt_token_len=torch.tensor([prompt_token.shape[1]], dtype=torch.int32).to(self.device),
@@ -384,3 +378,53 @@ class CosyVoice2Model(CosyVoiceModel):
         if torch.cuda.is_available():
             torch.cuda.empty_cache()
             torch.cuda.current_stream().synchronize()
+
+
+class CosyVoice3Model(CosyVoice2Model):
+
+    def __init__(self,
+                 llm: torch.nn.Module,
+                 flow: torch.nn.Module,
+                 hift: torch.nn.Module,
+                 fp16: bool = False):
+        self.device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
+        self.llm = llm
+        self.flow = flow
+        self.hift = hift
+        self.fp16 = fp16
+        # NOTE must matching training static_chunk_size
+        self.token_hop_len = 25
+        # rtf and decoding related
+        self.llm_context = torch.cuda.stream(torch.cuda.Stream(self.device)) if torch.cuda.is_available() else nullcontext()
+        self.lock = threading.Lock()
+        # dict used to store session related variable
+        self.tts_speech_token_dict = {}
+        self.llm_end_dict = {}
+        self.hift_cache_dict = {}
+
+    def token2wav(self, token, prompt_token, prompt_feat, embedding, token_offset, uuid, stream=False, finalize=False, speed=1.0):
+        with torch.cuda.amp.autocast(self.fp16):
+            tts_mel, _ = self.flow.inference(token=token.to(self.device, dtype=torch.int32),
+                                             token_len=torch.tensor([token.shape[1]], dtype=torch.int32).to(self.device),
+                                             prompt_token=prompt_token.to(self.device),
+                                             prompt_token_len=torch.tensor([prompt_token.shape[1]], dtype=torch.int32).to(self.device),
+                                             prompt_feat=prompt_feat.to(self.device),
+                                             prompt_feat_len=torch.tensor([prompt_feat.shape[1]], dtype=torch.int32).to(self.device),
+                                             embedding=embedding.to(self.device),
+                                             streaming=stream,
+                                             finalize=finalize)
+            tts_mel = tts_mel[:, :, token_offset * self.flow.token_mel_ratio:]
+            # append mel cache
+            if self.hift_cache_dict[uuid] is not None:
+                hift_cache_mel = self.hift_cache_dict[uuid]['mel']
+                tts_mel = torch.concat([hift_cache_mel, tts_mel], dim=2)
+                self.hift_cache_dict[uuid]['mel'] = tts_mel
+            else:
+                self.hift_cache_dict[uuid] = {'mel': tts_mel, 'speech_offset': 0}
+            if speed != 1.0:
+                assert token_offset == 0 and finalize is True, 'speed change only support non-stream inference mode'
+                tts_mel = F.interpolate(tts_mel, size=int(tts_mel.shape[2] / speed), mode='linear')
+            tts_speech, _ = self.hift.inference(speech_feat=tts_mel, finalize=finalize)
+            tts_speech = tts_speech[:, self.hift_cache_dict[uuid]['speech_offset']:]
+            self.hift_cache_dict[uuid]['speech_offset'] += tts_speech.shape[1]
+        return tts_speech

cosyvoice/dataset/processor.py

@@ -242,6 +242,10 @@ def tokenize(data, get_tokenizer, allowed_special, mode='train'):
     for sample in data:
         assert 'text' in sample
         sample['text_token'] = tokenizer.encode(sample['text'], allowed_special=allowed_special)
+        if 'instruct' in sample:
+            sample['instruct_token'] = tokenizer.encode(sample['instruct'], allowed_special=allowed_special)
+        else:
+            sample['instruct_token'] = tokenizer.encode('', allowed_special=allowed_special)
         yield sample
 
 
@@ -390,6 +394,9 @@ def padding(data, use_spk_embedding, mode='train', gan=False, dpo=False):
         text_token = [torch.tensor(sample[i]['text_token']) for i in order]
         text_token_len = torch.tensor([i.size(0) for i in text_token], dtype=torch.int32)
         text_token = pad_sequence(text_token, batch_first=True, padding_value=0)
+        instruct_token = [torch.tensor(sample[i]['instruct_token']) for i in order]
+        instruct_token_len = torch.tensor([i.size(0) for i in instruct_token], dtype=torch.int32)
+        instruct_token = pad_sequence(instruct_token, batch_first=True, padding_value=0)
         utt_embedding = torch.stack([sample[i]['utt_embedding'] for i in order], dim=0)
         spk_embedding = torch.stack([sample[i]['spk_embedding'] for i in order], dim=0)
         batch = {
@@ -403,6 +410,8 @@ def padding(data, use_spk_embedding, mode='train', gan=False, dpo=False):
             "text": text,
             "text_token": text_token,
             "text_token_len": text_token_len,
+            "instruct_token": instruct_token,
+            "instruct_token_len": instruct_token_len,
             "utt_embedding": utt_embedding,
             "spk_embedding": spk_embedding,
         }

cosyvoice/flow/DiT/dit.py

@@ -0,0 +1,176 @@
+
+"""
+ein notation:
+b - batch
+n - sequence
+nt - text sequence
+nw - raw wave length
+d - dimension
+"""
+
+from __future__ import annotations
+
+import torch
+from torch import nn
+import torch.nn.functional as F
+from einops import repeat
+from x_transformers.x_transformers import RotaryEmbedding
+from cosyvoice.utils.mask import add_optional_chunk_mask
+from cosyvoice.flow.DiT.modules import (
+    TimestepEmbedding,
+    ConvNeXtV2Block,
+    CausalConvPositionEmbedding,
+    DiTBlock,
+    AdaLayerNormZero_Final,
+    precompute_freqs_cis,
+    get_pos_embed_indices,
+)
+
+
+# Text embedding
+
+
+class TextEmbedding(nn.Module):
+    def __init__(self, text_num_embeds, text_dim, conv_layers=0, conv_mult=2):
+        super().__init__()
+        self.text_embed = nn.Embedding(text_num_embeds + 1, text_dim)  # use 0 as filler token
+
+        if conv_layers > 0:
+            self.extra_modeling = True
+            self.precompute_max_pos = 4096  # ~44s of 24khz audio
+            self.register_buffer("freqs_cis", precompute_freqs_cis(text_dim, self.precompute_max_pos), persistent=False)
+            self.text_blocks = nn.Sequential(
+                *[ConvNeXtV2Block(text_dim, text_dim * conv_mult) for _ in range(conv_layers)]
+            )
+        else:
+            self.extra_modeling = False
+
+    def forward(self, text: int["b nt"], seq_len, drop_text=False):  # noqa: F722
+        batch, text_len = text.shape[0], text.shape[1]
+        text = text + 1  # use 0 as filler token. preprocess of batch pad -1, see list_str_to_idx()
+        text = text[:, :seq_len]  # curtail if character tokens are more than the mel spec tokens
+        text = F.pad(text, (0, seq_len - text_len), value=0)
+
+        if drop_text:  # cfg for text
+            text = torch.zeros_like(text)
+
+        text = self.text_embed(text)  # b n -> b n d
+
+        # possible extra modeling
+        if self.extra_modeling:
+            # sinus pos emb
+            batch_start = torch.zeros((batch,), dtype=torch.long)
+            pos_idx = get_pos_embed_indices(batch_start, seq_len, max_pos=self.precompute_max_pos)
+            text_pos_embed = self.freqs_cis[pos_idx]
+            text = text + text_pos_embed
+
+            # convnextv2 blocks
+            text = self.text_blocks(text)
+
+        return text
+
+
+# noised input audio and context mixing embedding
+
+
+class InputEmbedding(nn.Module):
+    def __init__(self, mel_dim, text_dim, out_dim, spk_dim=None):
+        super().__init__()
+        spk_dim = 0 if spk_dim is None else spk_dim
+        self.spk_dim = spk_dim
+        self.proj = nn.Linear(mel_dim * 2 + text_dim + spk_dim, out_dim)
+        self.conv_pos_embed = CausalConvPositionEmbedding(dim=out_dim)
+
+    def forward(
+            self,
+            x: float["b n d"],
+            cond: float["b n d"],
+            text_embed: float["b n d"],
+            spks: float["b d"],
+    ):
+        to_cat = [x, cond, text_embed]
+        if self.spk_dim > 0:
+            spks = repeat(spks, "b c -> b t c", t=x.shape[1])
+            to_cat.append(spks)
+
+        x = self.proj(torch.cat(to_cat, dim=-1))
+        x = self.conv_pos_embed(x) + x
+        return x
+
+
+# Transformer backbone using DiT blocks
+
+
+class DiT(nn.Module):
+    def __init__(
+        self,
+        *,
+        dim,
+        depth=8,
+        heads=8,
+        dim_head=64,
+        dropout=0.1,
+        ff_mult=4,
+        mel_dim=80,
+        mu_dim=None,
+        long_skip_connection=False,
+        spk_dim=None,
+        out_channels=None,
+        static_chunk_size=50,
+        num_decoding_left_chunks=2
+    ):
+        super().__init__()
+
+        self.time_embed = TimestepEmbedding(dim)
+        if mu_dim is None:
+            mu_dim = mel_dim
+        self.input_embed = InputEmbedding(mel_dim, mu_dim, dim, spk_dim)
+
+        self.rotary_embed = RotaryEmbedding(dim_head)
+
+        self.dim = dim
+        self.depth = depth
+
+        self.transformer_blocks = nn.ModuleList(
+            [DiTBlock(dim=dim, heads=heads, dim_head=dim_head, ff_mult=ff_mult, dropout=dropout) for _ in range(depth)]
+        )
+        self.long_skip_connection = nn.Linear(dim * 2, dim, bias=False) if long_skip_connection else None
+
+        self.norm_out = AdaLayerNormZero_Final(dim)  # final modulation
+        self.proj_out = nn.Linear(dim, mel_dim)
+        self.out_channels = out_channels
+        self.static_chunk_size = static_chunk_size
+        self.num_decoding_left_chunks = num_decoding_left_chunks
+
+    def forward(self, x, mask, mu, t, spks=None, cond=None, streaming=False):
+        x = x.transpose(1, 2)
+        mu = mu.transpose(1, 2)
+        cond = cond.transpose(1, 2)
+        spks = spks.unsqueeze(dim=1)
+        batch, seq_len = x.shape[0], x.shape[1]
+        if t.ndim == 0:
+            t = t.repeat(batch)
+
+        # t: conditioning time, c: context (text + masked cond audio), x: noised input audio
+        t = self.time_embed(t)
+        x = self.input_embed(x, cond, mu, spks.squeeze(1))
+
+        rope = self.rotary_embed.forward_from_seq_len(seq_len)
+
+        if self.long_skip_connection is not None:
+            residual = x
+
+        if streaming is True:
+            attn_mask = add_optional_chunk_mask(x, mask.bool(), False, False, 0, self.static_chunk_size, -1).unsqueeze(dim=1)
+        else:
+            attn_mask = add_optional_chunk_mask(x, mask.bool(), False, False, 0, 0, -1).repeat(1, x.size(1), 1).unsqueeze(dim=1)
+
+        for block in self.transformer_blocks:
+            x = block(x, t, mask=attn_mask.bool(), rope=rope)
+
+        if self.long_skip_connection is not None:
+            x = self.long_skip_connection(torch.cat((x, residual), dim=-1))
+
+        x = self.norm_out(x, t)
+        output = self.proj_out(x).transpose(1, 2)
+        return output

cosyvoice/flow/DiT/modules.py

@@ -0,0 +1,616 @@
+
+"""
+ein notation:
+b - batch
+n - sequence
+nt - text sequence
+nw - raw wave length
+d - dimension
+"""
+
+from __future__ import annotations
+from typing import Optional
+import math
+
+import torch
+from torch import nn
+import torch.nn.functional as F
+import torchaudio
+
+from x_transformers.x_transformers import apply_rotary_pos_emb
+
+
+# raw wav to mel spec
+class MelSpec(nn.Module):
+    def __init__(
+        self,
+        filter_length=1024,
+        hop_length=256,
+        win_length=1024,
+        n_mel_channels=100,
+        target_sample_rate=24_000,
+        normalize=False,
+        power=1,
+        norm=None,
+        center=True,
+    ):
+        super().__init__()
+        self.n_mel_channels = n_mel_channels
+
+        self.mel_stft = torchaudio.transforms.MelSpectrogram(
+            sample_rate=target_sample_rate,
+            n_fft=filter_length,
+            win_length=win_length,
+            hop_length=hop_length,
+            n_mels=n_mel_channels,
+            power=power,
+            center=center,
+            normalized=normalize,
+            norm=norm,
+        )
+
+        self.register_buffer("dummy", torch.tensor(0), persistent=False)
+
+    def forward(self, inp):
+        if len(inp.shape) == 3:
+            inp = inp.squeeze(1)  # 'b 1 nw -> b nw'
+
+        assert len(inp.shape) == 2
+
+        if self.dummy.device != inp.device:
+            self.to(inp.device)
+
+        mel = self.mel_stft(inp)
+        mel = mel.clamp(min=1e-5).log()
+        return mel
+
+
+# sinusoidal position embedding
+
+
+class SinusPositionEmbedding(nn.Module):
+    def __init__(self, dim):
+        super().__init__()
+        self.dim = dim
+
+    def forward(self, x, scale=1000):
+        device = x.device
+        half_dim = self.dim // 2
+        emb = math.log(10000) / (half_dim - 1)
+        emb = torch.exp(torch.arange(half_dim, device=device).float() * -emb)
+        emb = scale * x.unsqueeze(1) * emb.unsqueeze(0)
+        emb = torch.cat((emb.sin(), emb.cos()), dim=-1)
+        return emb
+
+
+# convolutional position embedding
+
+
+class ConvPositionEmbedding(nn.Module):
+    def __init__(self, dim, kernel_size=31, groups=16):
+        super().__init__()
+        assert kernel_size % 2 != 0
+        self.conv1d = nn.Sequential(
+            nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=kernel_size // 2),
+            nn.Mish(),
+            nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=kernel_size // 2),
+            nn.Mish(),
+        )
+
+    def forward(self, x: float["b n d"], mask: bool["b n"] | None = None):  # noqa: F722
+        if mask is not None:
+            mask = mask[..., None]
+            x = x.masked_fill(~mask, 0.0)
+
+        x = x.permute(0, 2, 1)
+        x = self.conv1d(x)
+        out = x.permute(0, 2, 1)
+
+        if mask is not None:
+            out = out.masked_fill(~mask, 0.0)
+
+        return out
+
+
+class CausalConvPositionEmbedding(nn.Module):
+    def __init__(self, dim, kernel_size=31, groups=16):
+        super().__init__()
+        assert kernel_size % 2 != 0
+        self.kernel_size = kernel_size
+        self.conv1 = nn.Sequential(
+            nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=0),
+            nn.Mish(),
+        )
+        self.conv2 = nn.Sequential(
+            nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=0),
+            nn.Mish(),
+        )
+
+    def forward(self, x: float["b n d"], mask: bool["b n"] | None = None):  # noqa: F722
+        if mask is not None:
+            mask = mask[..., None]
+            x = x.masked_fill(~mask, 0.0)
+
+        x = x.permute(0, 2, 1)
+        x = F.pad(x, (self.kernel_size - 1, 0, 0, 0))
+        x = self.conv1(x)
+        x = F.pad(x, (self.kernel_size - 1, 0, 0, 0))
+        x = self.conv2(x)
+        out = x.permute(0, 2, 1)
+
+        if mask is not None:
+            out = out.masked_fill(~mask, 0.0)
+
+        return out
+
+
+# rotary positional embedding related
+
+
+def precompute_freqs_cis(dim: int, end: int, theta: float = 10000.0, theta_rescale_factor=1.0):
+    # proposed by reddit user bloc97, to rescale rotary embeddings to longer sequence length without fine-tuning
+    # has some connection to NTK literature
+    # https://www.reddit.com/r/LocalLLaMA/comments/14lz7j5/ntkaware_scaled_rope_allows_llama_models_to_have/
+    # https://github.com/lucidrains/rotary-embedding-torch/blob/main/rotary_embedding_torch/rotary_embedding_torch.py
+    theta *= theta_rescale_factor ** (dim / (dim - 2))
+    freqs = 1.0 / (theta ** (torch.arange(0, dim, 2)[: (dim // 2)].float() / dim))
+    t = torch.arange(end, device=freqs.device)  # type: ignore
+    freqs = torch.outer(t, freqs).float()  # type: ignore
+    freqs_cos = torch.cos(freqs)  # real part
+    freqs_sin = torch.sin(freqs)  # imaginary part
+    return torch.cat([freqs_cos, freqs_sin], dim=-1)
+
+
+def get_pos_embed_indices(start, length, max_pos, scale=1.0):
+    # length = length if isinstance(length, int) else length.max()
+    scale = scale * torch.ones_like(start, dtype=torch.float32)  # in case scale is a scalar
+    pos = (
+        start.unsqueeze(1)
+        + (torch.arange(length, device=start.device, dtype=torch.float32).unsqueeze(0) * scale.unsqueeze(1)).long()
+    )
+    # avoid extra long error.
+    pos = torch.where(pos < max_pos, pos, max_pos - 1)
+    return pos
+
+
+# Global Response Normalization layer (Instance Normalization ?)
+
+
+class GRN(nn.Module):
+    def __init__(self, dim):
+        super().__init__()
+        self.gamma = nn.Parameter(torch.zeros(1, 1, dim))
+        self.beta = nn.Parameter(torch.zeros(1, 1, dim))
+
+    def forward(self, x):
+        Gx = torch.norm(x, p=2, dim=1, keepdim=True)
+        Nx = Gx / (Gx.mean(dim=-1, keepdim=True) + 1e-6)
+        return self.gamma * (x * Nx) + self.beta + x
+
+
+# ConvNeXt-V2 Block https://github.com/facebookresearch/ConvNeXt-V2/blob/main/models/convnextv2.py
+# ref: https://github.com/bfs18/e2_tts/blob/main/rfwave/modules.py#L108
+
+
+class ConvNeXtV2Block(nn.Module):
+    def __init__(
+        self,
+        dim: int,
+        intermediate_dim: int,
+        dilation: int = 1,
+    ):
+        super().__init__()
+        padding = (dilation * (7 - 1)) // 2
+        self.dwconv = nn.Conv1d(
+            dim, dim, kernel_size=7, padding=padding, groups=dim, dilation=dilation
+        )  # depthwise conv
+        self.norm = nn.LayerNorm(dim, eps=1e-6)
+        self.pwconv1 = nn.Linear(dim, intermediate_dim)  # pointwise/1x1 convs, implemented with linear layers
+        self.act = nn.GELU()
+        self.grn = GRN(intermediate_dim)
+        self.pwconv2 = nn.Linear(intermediate_dim, dim)
+
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
+        residual = x
+        x = x.transpose(1, 2)  # b n d -> b d n
+        x = self.dwconv(x)
+        x = x.transpose(1, 2)  # b d n -> b n d
+        x = self.norm(x)
+        x = self.pwconv1(x)
+        x = self.act(x)
+        x = self.grn(x)
+        x = self.pwconv2(x)
+        return residual + x
+
+
+# AdaLayerNormZero
+# return with modulated x for attn input, and params for later mlp modulation
+
+
+class AdaLayerNormZero(nn.Module):
+    def __init__(self, dim):
+        super().__init__()
+
+        self.silu = nn.SiLU()
+        self.linear = nn.Linear(dim, dim * 6)
+
+        self.norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+
+    def forward(self, x, emb=None):
+        emb = self.linear(self.silu(emb))
+        shift_msa, scale_msa, gate_msa, shift_mlp, scale_mlp, gate_mlp = torch.chunk(emb, 6, dim=1)
+
+        x = self.norm(x) * (1 + scale_msa[:, None]) + shift_msa[:, None]
+        return x, gate_msa, shift_mlp, scale_mlp, gate_mlp
+
+
+# AdaLayerNormZero for final layer
+# return only with modulated x for attn input, cuz no more mlp modulation
+
+
+class AdaLayerNormZero_Final(nn.Module):
+    def __init__(self, dim):
+        super().__init__()
+
+        self.silu = nn.SiLU()
+        self.linear = nn.Linear(dim, dim * 2)
+
+        self.norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+
+    def forward(self, x, emb):
+        emb = self.linear(self.silu(emb))
+        scale, shift = torch.chunk(emb, 2, dim=1)
+
+        x = self.norm(x) * (1 + scale)[:, None, :] + shift[:, None, :]
+        return x
+
+
+# FeedForward
+
+
+class FeedForward(nn.Module):
+    def __init__(self, dim, dim_out=None, mult=4, dropout=0.0, approximate: str = "none"):
+        super().__init__()
+        inner_dim = int(dim * mult)
+        dim_out = dim_out if dim_out is not None else dim
+
+        activation = nn.GELU(approximate=approximate)
+        project_in = nn.Sequential(nn.Linear(dim, inner_dim), activation)
+        self.ff = nn.Sequential(project_in, nn.Dropout(dropout), nn.Linear(inner_dim, dim_out))
+
+    def forward(self, x):
+        return self.ff(x)
+
+
+# Attention with possible joint part
+# modified from diffusers/src/diffusers/models/attention_processor.py
+
+
+class Attention(nn.Module):
+    def __init__(
+        self,
+        processor: JointAttnProcessor | AttnProcessor,
+        dim: int,
+        heads: int = 8,
+        dim_head: int = 64,
+        dropout: float = 0.0,
+        context_dim: Optional[int] = None,  # if not None -> joint attention
+        context_pre_only=None,
+    ):
+        super().__init__()
+
+        if not hasattr(F, "scaled_dot_product_attention"):
+            raise ImportError("Attention equires PyTorch 2.0, to use it, please upgrade PyTorch to 2.0.")
+
+        self.processor = processor
+
+        self.dim = dim
+        self.heads = heads
+        self.inner_dim = dim_head * heads
+        self.dropout = dropout
+
+        self.context_dim = context_dim
+        self.context_pre_only = context_pre_only
+
+        self.to_q = nn.Linear(dim, self.inner_dim)
+        self.to_k = nn.Linear(dim, self.inner_dim)
+        self.to_v = nn.Linear(dim, self.inner_dim)
+
+        if self.context_dim is not None:
+            self.to_k_c = nn.Linear(context_dim, self.inner_dim)
+            self.to_v_c = nn.Linear(context_dim, self.inner_dim)
+            if self.context_pre_only is not None:
+                self.to_q_c = nn.Linear(context_dim, self.inner_dim)
+
+        self.to_out = nn.ModuleList([])
+        self.to_out.append(nn.Linear(self.inner_dim, dim))
+        self.to_out.append(nn.Dropout(dropout))
+
+        if self.context_pre_only is not None and not self.context_pre_only:
+            self.to_out_c = nn.Linear(self.inner_dim, dim)
+
+    def forward(
+        self,
+        x: float["b n d"],  # noised input x  # noqa: F722
+        c: float["b n d"] = None,  # context c  # noqa: F722
+        mask: bool["b n"] | None = None,  # noqa: F722
+        rope=None,  # rotary position embedding for x
+        c_rope=None,  # rotary position embedding for c
+    ) -> torch.Tensor:
+        if c is not None:
+            return self.processor(self, x, c=c, mask=mask, rope=rope, c_rope=c_rope)
+        else:
+            return self.processor(self, x, mask=mask, rope=rope)
+
+
+# Attention processor
+
+
+class AttnProcessor:
+    def __init__(self):
+        pass
+
+    def __call__(
+        self,
+        attn: Attention,
+        x: float["b n d"],  # noised input x  # noqa: F722
+        mask: bool["b n"] | None = None,  # noqa: F722
+        rope=None,  # rotary position embedding
+    ) -> torch.FloatTensor:
+        batch_size = x.shape[0]
+
+        # `sample` projections.
+        query = attn.to_q(x)
+        key = attn.to_k(x)
+        value = attn.to_v(x)
+
+        # apply rotary position embedding
+        if rope is not None:
+            freqs, xpos_scale = rope
+            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
+
+            query = apply_rotary_pos_emb(query, freqs, q_xpos_scale)
+            key = apply_rotary_pos_emb(key, freqs, k_xpos_scale)
+
+        # attention
+        inner_dim = key.shape[-1]
+        head_dim = inner_dim // attn.heads
+        query = query.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+        key = key.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+        value = value.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+
+        # mask. e.g. inference got a batch with different target durations, mask out the padding
+        if mask is not None:
+            attn_mask = mask
+            if attn_mask.dim() == 2:
+                attn_mask = attn_mask.unsqueeze(1).unsqueeze(1)  # 'b n -> b 1 1 n'
+                attn_mask = attn_mask.expand(batch_size, attn.heads, query.shape[-2], key.shape[-2])
+        else:
+            attn_mask = None
+
+        x = F.scaled_dot_product_attention(query, key, value, attn_mask=attn_mask, dropout_p=0.0, is_causal=False)
+        x = x.transpose(1, 2).reshape(batch_size, -1, attn.heads * head_dim)
+        x = x.to(query.dtype)
+
+        # linear proj
+        x = attn.to_out[0](x)
+        # dropout
+        x = attn.to_out[1](x)
+
+        if mask is not None:
+            if mask.dim() == 2:
+                mask = mask.unsqueeze(-1)
+            else:
+                mask = mask[:, 0, -1].unsqueeze(-1)
+            x = x.masked_fill(~mask, 0.0)
+
+        return x
+
+
+# Joint Attention processor for MM-DiT
+# modified from diffusers/src/diffusers/models/attention_processor.py
+
+
+class JointAttnProcessor:
+    def __init__(self):
+        pass
+
+    def __call__(
+        self,
+        attn: Attention,
+        x: float["b n d"],  # noised input x  # noqa: F722
+        c: float["b nt d"] = None,  # context c, here text # noqa: F722
+        mask: bool["b n"] | None = None,  # noqa: F722
+        rope=None,  # rotary position embedding for x
+        c_rope=None,  # rotary position embedding for c
+    ) -> torch.FloatTensor:
+        residual = x
+
+        batch_size = c.shape[0]
+
+        # `sample` projections.
+        query = attn.to_q(x)
+        key = attn.to_k(x)
+        value = attn.to_v(x)
+
+        # `context` projections.
+        c_query = attn.to_q_c(c)
+        c_key = attn.to_k_c(c)
+        c_value = attn.to_v_c(c)
+
+        # apply rope for context and noised input independently
+        if rope is not None:
+            freqs, xpos_scale = rope
+            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
+            query = apply_rotary_pos_emb(query, freqs, q_xpos_scale)
+            key = apply_rotary_pos_emb(key, freqs, k_xpos_scale)
+        if c_rope is not None:
+            freqs, xpos_scale = c_rope
+            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
+            c_query = apply_rotary_pos_emb(c_query, freqs, q_xpos_scale)
+            c_key = apply_rotary_pos_emb(c_key, freqs, k_xpos_scale)
+
+        # attention
+        query = torch.cat([query, c_query], dim=1)
+        key = torch.cat([key, c_key], dim=1)
+        value = torch.cat([value, c_value], dim=1)
+
+        inner_dim = key.shape[-1]
+        head_dim = inner_dim // attn.heads
+        query = query.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+        key = key.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+        value = value.view(batch_size, -1, attn.heads, head_dim).transpose(1, 2)
+
+        # mask. e.g. inference got a batch with different target durations, mask out the padding
+        if mask is not None:
+            attn_mask = F.pad(mask, (0, c.shape[1]), value=True)  # no mask for c (text)
+            attn_mask = attn_mask.unsqueeze(1).unsqueeze(1)  # 'b n -> b 1 1 n'
+            attn_mask = attn_mask.expand(batch_size, attn.heads, query.shape[-2], key.shape[-2])
+        else:
+            attn_mask = None
+
+        x = F.scaled_dot_product_attention(query, key, value, attn_mask=attn_mask, dropout_p=0.0, is_causal=False)
+        x = x.transpose(1, 2).reshape(batch_size, -1, attn.heads * head_dim)
+        x = x.to(query.dtype)
+
+        # Split the attention outputs.
+        x, c = (
+            x[:, : residual.shape[1]],
+            x[:, residual.shape[1]:],
+        )
+
+        # linear proj
+        x = attn.to_out[0](x)
+        # dropout
+        x = attn.to_out[1](x)
+        if not attn.context_pre_only:
+            c = attn.to_out_c(c)
+
+        if mask is not None:
+            mask = mask.unsqueeze(-1)
+            x = x.masked_fill(~mask, 0.0)
+            # c = c.masked_fill(~mask, 0.)  # no mask for c (text)
+
+        return x, c
+
+
+# DiT Block
+
+
+class DiTBlock(nn.Module):
+    def __init__(self, dim, heads, dim_head, ff_mult=4, dropout=0.1):
+        super().__init__()
+
+        self.attn_norm = AdaLayerNormZero(dim)
+        self.attn = Attention(
+            processor=AttnProcessor(),
+            dim=dim,
+            heads=heads,
+            dim_head=dim_head,
+            dropout=dropout,
+        )
+
+        self.ff_norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+        self.ff = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
+
+    def forward(self, x, t, mask=None, rope=None):  # x: noised input, t: time embedding
+        # pre-norm & modulation for attention input
+        norm, gate_msa, shift_mlp, scale_mlp, gate_mlp = self.attn_norm(x, emb=t)
+
+        # attention
+        attn_output = self.attn(x=norm, mask=mask, rope=rope)
+
+        # process attention output for input x
+        x = x + gate_msa.unsqueeze(1) * attn_output
+
+        ff_norm = self.ff_norm(x) * (1 + scale_mlp[:, None]) + shift_mlp[:, None]
+        ff_output = self.ff(ff_norm)
+        x = x + gate_mlp.unsqueeze(1) * ff_output
+
+        return x
+
+
+# MMDiT Block https://arxiv.org/abs/2403.03206
+
+
+class MMDiTBlock(nn.Module):
+    r"""
+    modified from diffusers/src/diffusers/models/attention.py
+
+    notes.
+    _c: context related. text, cond, etc. (left part in sd3 fig2.b)
+    _x: noised input related. (right part)
+    context_pre_only: last layer only do prenorm + modulation cuz no more ffn
+    """
+
+    def __init__(self, dim, heads, dim_head, ff_mult=4, dropout=0.1, context_pre_only=False):
+        super().__init__()
+
+        self.context_pre_only = context_pre_only
+
+        self.attn_norm_c = AdaLayerNormZero_Final(dim) if context_pre_only else AdaLayerNormZero(dim)
+        self.attn_norm_x = AdaLayerNormZero(dim)
+        self.attn = Attention(
+            processor=JointAttnProcessor(),
+            dim=dim,
+            heads=heads,
+            dim_head=dim_head,
+            dropout=dropout,
+            context_dim=dim,
+            context_pre_only=context_pre_only,
+        )
+
+        if not context_pre_only:
+            self.ff_norm_c = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+            self.ff_c = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
+        else:
+            self.ff_norm_c = None
+            self.ff_c = None
+        self.ff_norm_x = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
+        self.ff_x = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
+
+    def forward(self, x, c, t, mask=None, rope=None, c_rope=None):  # x: noised input, c: context, t: time embedding
+        # pre-norm & modulation for attention input
+        if self.context_pre_only:
+            norm_c = self.attn_norm_c(c, t)
+        else:
+            norm_c, c_gate_msa, c_shift_mlp, c_scale_mlp, c_gate_mlp = self.attn_norm_c(c, emb=t)
+        norm_x, x_gate_msa, x_shift_mlp, x_scale_mlp, x_gate_mlp = self.attn_norm_x(x, emb=t)
+
+        # attention
+        x_attn_output, c_attn_output = self.attn(x=norm_x, c=norm_c, mask=mask, rope=rope, c_rope=c_rope)
+
+        # process attention output for context c
+        if self.context_pre_only:
+            c = None
+        else:  # if not last layer
+            c = c + c_gate_msa.unsqueeze(1) * c_attn_output
+
+            norm_c = self.ff_norm_c(c) * (1 + c_scale_mlp[:, None]) + c_shift_mlp[:, None]
+            c_ff_output = self.ff_c(norm_c)
+            c = c + c_gate_mlp.unsqueeze(1) * c_ff_output
+
+        # process attention output for input x
+        x = x + x_gate_msa.unsqueeze(1) * x_attn_output
+
+        norm_x = self.ff_norm_x(x) * (1 + x_scale_mlp[:, None]) + x_shift_mlp[:, None]
+        x_ff_output = self.ff_x(norm_x)
+        x = x + x_gate_mlp.unsqueeze(1) * x_ff_output
+
+        return c, x
+
+
+# time step conditioning embedding
+
+
+class TimestepEmbedding(nn.Module):
+    def __init__(self, dim, freq_embed_dim=256):
+        super().__init__()
+        self.time_embed = SinusPositionEmbedding(freq_embed_dim)
+        self.time_mlp = nn.Sequential(nn.Linear(freq_embed_dim, dim), nn.SiLU(), nn.Linear(dim, dim))
+
+    def forward(self, timestep: float["b"]):  # noqa: F821
+        time_hidden = self.time_embed(timestep)
+        time_hidden = time_hidden.to(timestep.dtype)
+        time = self.time_mlp(time_hidden)  # b d
+        return time

cosyvoice/flow/flow.py

@@ -37,14 +37,11 @@ class MaskedDiffWithXvec(torch.nn.Module):
                                        'cfm_params': DictConfig({'sigma_min': 1e-06, 'solver': 'euler', 't_scheduler': 'cosine',
                                                                  'training_cfg_rate': 0.2, 'inference_cfg_rate': 0.7, 'reg_loss_type': 'l1'}),
                                        'decoder_params': {'channels': [256, 256], 'dropout': 0.0, 'attention_head_dim': 64,
-                                                          'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}},
-                 mel_feat_conf: Dict = {'n_fft': 1024, 'num_mels': 80, 'sampling_rate': 22050,
-                                        'hop_size': 256, 'win_size': 1024, 'fmin': 0, 'fmax': 8000}):
+                                                          'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}}):
         super().__init__()
         self.input_size = input_size
         self.output_size = output_size
         self.decoder_conf = decoder_conf
-        self.mel_feat_conf = mel_feat_conf
         self.vocab_size = vocab_size
         self.output_type = output_type
         self.input_frame_rate = input_frame_rate
@@ -165,14 +162,11 @@ class CausalMaskedDiffWithXvec(torch.nn.Module):
                                        'cfm_params': DictConfig({'sigma_min': 1e-06, 'solver': 'euler', 't_scheduler': 'cosine',
                                                                  'training_cfg_rate': 0.2, 'inference_cfg_rate': 0.7, 'reg_loss_type': 'l1'}),
                                        'decoder_params': {'channels': [256, 256], 'dropout': 0.0, 'attention_head_dim': 64,
-                                                          'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}},
-                 mel_feat_conf: Dict = {'n_fft': 1024, 'num_mels': 80, 'sampling_rate': 22050,
-                                        'hop_size': 256, 'win_size': 1024, 'fmin': 0, 'fmax': 8000}):
+                                                          'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}}):
         super().__init__()
         self.input_size = input_size
         self.output_size = output_size
         self.decoder_conf = decoder_conf
-        self.mel_feat_conf = mel_feat_conf
         self.vocab_size = vocab_size
         self.output_type = output_type
         self.input_frame_rate = input_frame_rate
@@ -279,3 +273,160 @@ class CausalMaskedDiffWithXvec(torch.nn.Module):
         feat = feat[:, :, mel_len1:]
         assert feat.shape[2] == mel_len2
         return feat.float(), None
+
+
+class CausalMaskedDiffWithDiT(torch.nn.Module):
+    def __init__(self,
+                 input_size: int = 512,
+                 output_size: int = 80,
+                 spk_embed_dim: int = 192,
+                 output_type: str = "mel",
+                 vocab_size: int = 4096,
+                 input_frame_rate: int = 50,
+                 only_mask_loss: bool = True,
+                 token_mel_ratio: int = 2,
+                 pre_lookahead_len: int = 3,
+                 pre_lookahead_layer: torch.nn.Module = None,
+                 decoder: torch.nn.Module = None,
+                 decoder_conf: Dict = {'in_channels': 240, 'out_channel': 80, 'spk_emb_dim': 80, 'n_spks': 1,
+                                       'cfm_params': DictConfig({'sigma_min': 1e-06, 'solver': 'euler', 't_scheduler': 'cosine',
+                                                                 'training_cfg_rate': 0.2, 'inference_cfg_rate': 0.7, 'reg_loss_type': 'l1'}),
+                                       'decoder_params': {'channels': [256, 256], 'dropout': 0.0, 'attention_head_dim': 64,
+                                                          'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}}):
+        super().__init__()
+        self.input_size = input_size
+        self.output_size = output_size
+        self.decoder_conf = decoder_conf
+        self.vocab_size = vocab_size
+        self.output_type = output_type
+        self.input_frame_rate = input_frame_rate
+        logging.info(f"input frame rate={self.input_frame_rate}")
+        self.input_embedding = nn.Embedding(vocab_size, input_size)
+        self.spk_embed_affine_layer = torch.nn.Linear(spk_embed_dim, output_size)
+        self.pre_lookahead_len = pre_lookahead_len
+        self.pre_lookahead_layer = pre_lookahead_layer
+        self.decoder = decoder
+        self.only_mask_loss = only_mask_loss
+        self.token_mel_ratio = token_mel_ratio
+
+    def forward(
+            self,
+            batch: dict,
+            device: torch.device,
+    ) -> Dict[str, Optional[torch.Tensor]]:
+        token = batch['speech_token'].to(device)
+        token_len = batch['speech_token_len'].to(device)
+        feat = batch['speech_feat'].to(device)
+        feat_len = batch['speech_feat_len'].to(device)
+        embedding = batch['embedding'].to(device)
+
+        # NOTE unified training, static_chunk_size > 0 or = 0
+        streaming = True if random.random() < 0.5 else False
+
+        # xvec projection
+        embedding = F.normalize(embedding, dim=1)
+        embedding = self.spk_embed_affine_layer(embedding)
+
+        # concat text and prompt_text
+        mask = (~make_pad_mask(token_len)).float().unsqueeze(-1).to(device)
+        token = self.input_embedding(torch.clamp(token, min=0)) * mask
+
+        # text encode
+        h, h_lengths = self.encoder(token, token_len, streaming=streaming)
+        h = self.encoder_proj(h)
+
+        # get conditions
+        conds = torch.zeros(feat.shape, device=token.device)
+        for i, j in enumerate(feat_len):
+            if random.random() < 0.5:
+                continue
+            index = random.randint(0, int(0.3 * j))
+            conds[i, :index] = feat[i, :index]
+        conds = conds.transpose(1, 2)
+
+        mask = (~make_pad_mask(h_lengths.sum(dim=-1).squeeze(dim=1))).to(h)
+        loss, _ = self.decoder.compute_loss(
+            feat.transpose(1, 2).contiguous(),
+            mask.unsqueeze(1),
+            h.transpose(1, 2).contiguous(),
+            embedding,
+            cond=conds,
+            streaming=streaming,
+        )
+        return {'loss': loss}
+
+    @torch.inference_mode()
+    def inference(self,
+                  token,
+                  token_len,
+                  prompt_token,
+                  prompt_token_len,
+                  prompt_feat,
+                  prompt_feat_len,
+                  embedding,
+                  streaming,
+                  finalize):
+        assert token.shape[0] == 1
+        # xvec projection
+        embedding = F.normalize(embedding, dim=1)
+        embedding = self.spk_embed_affine_layer(embedding)
+
+        # concat text and prompt_text
+        token, token_len = torch.concat([prompt_token, token], dim=1), prompt_token_len + token_len
+        mask = (~make_pad_mask(token_len)).unsqueeze(-1).to(embedding)
+        token = self.input_embedding(torch.clamp(token, min=0)) * mask
+
+        # text encode
+        if finalize is True:
+            h = self.pre_lookahead_layer(token)
+        else:
+            h = self.pre_lookahead_layer(token[:, :-self.pre_lookahead_len], context=token[:, -self.pre_lookahead_len:])
+        h = h.repeat_interleave(self.token_mel_ratio, dim=1)
+        mel_len1, mel_len2 = prompt_feat.shape[1], h.shape[1] - prompt_feat.shape[1]
+
+        # get conditions
+        conds = torch.zeros([1, mel_len1 + mel_len2, self.output_size], device=token.device).to(h.dtype)
+        conds[:, :mel_len1] = prompt_feat
+        conds = conds.transpose(1, 2)
+
+        mask = (~make_pad_mask(torch.tensor([mel_len1 + mel_len2]))).to(h)
+        feat, _ = self.decoder(
+            mu=h.transpose(1, 2).contiguous(),
+            mask=mask.unsqueeze(1),
+            spks=embedding,
+            cond=conds,
+            n_timesteps=10,
+            streaming=streaming
+        )
+        feat = feat[:, :, mel_len1:]
+        assert feat.shape[2] == mel_len2
+        return feat.float(), None
+
+
+if __name__ == '__main__':
+    torch.backends.cudnn.deterministic = True
+    torch.backends.cudnn.benchmark = False
+    from hyperpyyaml import load_hyperpyyaml
+    with open('./pretrained_models/Fun-CosyVoice3-0.5B/cosyvoice3.yaml', 'r') as f:
+        configs = load_hyperpyyaml(f, overrides={'llm': None, 'hift': None})
+    model = configs['flow']
+    device = 'cuda' if torch.cuda.is_available() else 'cpu'
+    model.to(device)
+    model.eval()
+    max_len = 10 * model.decoder.estimator.static_chunk_size
+    chunk_size = model.decoder.estimator.static_chunk_size
+    context_size = model.pre_lookahead_layer.pre_lookahead_len
+    token = torch.randint(0, 6561, size=(1, max_len)).to(device)
+    token_len = torch.tensor([max_len]).to(device)
+    prompt_token = torch.randint(0, 6561, size=(1, chunk_size)).to(device)
+    prompt_token_len = torch.tensor([chunk_size]).to(device)
+    prompt_feat = torch.rand(1, chunk_size * 2, 80).to(device)
+    prompt_feat_len = torch.tensor([chunk_size * 2]).to(device)
+    prompt_embedding = torch.rand(1, 192).to(device)
+    pred_gt, _ = model.inference(token, token_len, prompt_token, prompt_token_len, prompt_feat, prompt_feat_len, prompt_embedding, streaming=True, finalize=True)
+    for i in range(0, max_len, chunk_size):
+        finalize = True if i + chunk_size + context_size >= max_len else False
+        pred_chunk, _ = model.inference(token[:, :i + chunk_size + context_size], torch.tensor([token[:, :i + chunk_size + context_size].shape[1]]).to(device),
+                                        prompt_token, prompt_token_len, prompt_feat, prompt_feat_len, prompt_embedding, streaming=True, finalize=finalize)
+        pred_chunk = pred_chunk[:, :, i * model.token_mel_ratio:]
+        print((pred_gt[:, :, i * model.token_mel_ratio: i * model.token_mel_ratio + pred_chunk.shape[2]] - pred_chunk).abs().max().item())

cosyvoice/flow/flow_matching.py

@@ -91,12 +91,13 @@ class ConditionalCFM(BASECFM):
         sol = []
 
         # Do not use concat, it may cause memory format changed and trt infer with wrong results!
-        x_in = torch.zeros([2, 80, x.size(2)], device=x.device, dtype=x.dtype)
-        mask_in = torch.zeros([2, 1, x.size(2)], device=x.device, dtype=x.dtype)
-        mu_in = torch.zeros([2, 80, x.size(2)], device=x.device, dtype=x.dtype)
-        t_in = torch.zeros([2], device=x.device, dtype=x.dtype)
-        spks_in = torch.zeros([2, 80], device=x.device, dtype=x.dtype)
-        cond_in = torch.zeros([2, 80, x.size(2)], device=x.device, dtype=x.dtype)
+        # NOTE when flow run in amp mode, x.dtype is float32, which cause nan in trt fp16 inference, so set dtype=spks.dtype
+        x_in = torch.zeros([2, 80, x.size(2)], device=x.device, dtype=spks.dtype)
+        mask_in = torch.zeros([2, 1, x.size(2)], device=x.device, dtype=spks.dtype)
+        mu_in = torch.zeros([2, 80, x.size(2)], device=x.device, dtype=spks.dtype)
+        t_in = torch.zeros([2], device=x.device, dtype=spks.dtype)
+        spks_in = torch.zeros([2, 80], device=x.device, dtype=spks.dtype)
+        cond_in = torch.zeros([2, 80, x.size(2)], device=x.device, dtype=spks.dtype)
         for step in range(1, len(t_span)):
             # Classifier-Free Guidance inference introduced in VoiceBox
             x_in[:] = x

cosyvoice/hifigan/f0_predictor.py

@@ -17,6 +17,7 @@ try:
     from torch.nn.utils.parametrizations import weight_norm
 except ImportError:
     from torch.nn.utils import weight_norm
+from cosyvoice.transformer.convolution import CausalConv1d
 
 
 class ConvRNNF0Predictor(nn.Module):
@@ -56,3 +57,47 @@ class ConvRNNF0Predictor(nn.Module):
         x = self.condnet(x)
         x = x.transpose(1, 2)
         return torch.abs(self.classifier(x).squeeze(-1))
+
+
+class CausalConvRNNF0Predictor(nn.Module):
+    def __init__(self,
+                 num_class: int = 1,
+                 in_channels: int = 80,
+                 cond_channels: int = 512
+                 ):
+        super().__init__()
+
+        self.num_class = num_class
+        self.condnet = nn.Sequential(
+            weight_norm(
+                CausalConv1d(in_channels, cond_channels, kernel_size=4, causal_type='right')
+            ),
+            nn.ELU(),
+            weight_norm(
+                CausalConv1d(cond_channels, cond_channels, kernel_size=3, causal_type='left')
+            ),
+            nn.ELU(),
+            weight_norm(
+                CausalConv1d(cond_channels, cond_channels, kernel_size=3, causal_type='left')
+            ),
+            nn.ELU(),
+            weight_norm(
+                CausalConv1d(cond_channels, cond_channels, kernel_size=3, causal_type='left')
+            ),
+            nn.ELU(),
+            weight_norm(
+                CausalConv1d(cond_channels, cond_channels, kernel_size=3, causal_type='left')
+            ),
+            nn.ELU(),
+        )
+        self.classifier = nn.Linear(in_features=cond_channels, out_features=self.num_class)
+
+    def forward(self, x: torch.Tensor, finalize: bool = True) -> torch.Tensor:
+        if finalize is True:
+            x = self.condnet[0](x)
+        else:
+            x = self.condnet[0](x[:, :, :-self.condnet[0].causal_padding], x[:, :, -self.condnet[0].causal_padding:])
+        for i in range(1, len(self.condnet)):
+            x = self.condnet[i](x)
+        x = x.transpose(1, 2)
+        return torch.abs(self.classifier(x).squeeze(-1))

cosyvoice/hifigan/generator.py

@@ -28,7 +28,7 @@ try:
 except ImportError:
     from torch.nn.utils import weight_norm
 from torch.distributions.uniform import Uniform
-
+from cosyvoice.transformer.convolution import CausalConv1d, CausalConv1dDownSample, CausalConv1dUpsample
 from cosyvoice.transformer.activation import Snake
 from cosyvoice.utils.common import get_padding
 from cosyvoice.utils.common import init_weights
@@ -50,8 +50,10 @@ class ResBlock(torch.nn.Module):
         channels: int = 512,
         kernel_size: int = 3,
         dilations: List[int] = [1, 3, 5],
+        causal: bool = False,
     ):
         super(ResBlock, self).__init__()
+        self.causal = causal
         self.convs1 = nn.ModuleList()
         self.convs2 = nn.ModuleList()
 
@@ -64,7 +66,14 @@ class ResBlock(torch.nn.Module):
                         kernel_size,
                         1,
                         dilation=dilation,
-                        padding=get_padding(kernel_size, dilation)
+                        padding=get_padding(kernel_size, dilation)) if causal is False else
+                    CausalConv1d(
+                        channels,
+                        channels,
+                        kernel_size,
+                        1,
+                        dilation=dilation,
+                        causal_type='left'
                     )
                 )
             )
@@ -76,7 +85,14 @@ class ResBlock(torch.nn.Module):
                         kernel_size,
                         1,
                         dilation=1,
-                        padding=get_padding(kernel_size, 1)
+                        padding=get_padding(kernel_size, 1)) if causal is False else
+                    CausalConv1d(
+                        channels,
+                        channels,
+                        kernel_size,
+                        1,
+                        dilation=1,
+                        causal_type='left'
                     )
                 )
             )
@@ -139,11 +155,13 @@ class SineGen(torch.nn.Module):
 
     @torch.no_grad()
     def forward(self, f0):
+        """ sine_tensor, uv = forward(f0)
+        input F0: tensor(batchsize=1, dim=1, length)
+                  f0 for unvoiced steps should be 0
+        output sine_tensor: tensor(batchsize=1, length, dim)
+        output uv: tensor(batchsize=1, length, 1)
         """
-        :param f0: [B, 1, sample_len], Hz
-        :return: [B, 1, sample_len]
-        """
-
+        f0 = f0.transpose(1, 2)
         F_mat = torch.zeros((f0.size(0), self.harmonic_num + 1, f0.size(-1))).to(f0.device)
         for i in range(self.harmonic_num + 1):
             F_mat[:, i: i + 1, :] = f0 * (i + 1) / self.sampling_rate
@@ -168,59 +186,7 @@ class SineGen(torch.nn.Module):
         # first: set the unvoiced part to 0 by uv
         # then: additive noise
         sine_waves = sine_waves * uv + noise
-        return sine_waves, uv, noise
-
-
-class SourceModuleHnNSF(torch.nn.Module):
-    """ SourceModule for hn-nsf
-    SourceModule(sampling_rate, harmonic_num=0, sine_amp=0.1,
-                 add_noise_std=0.003, voiced_threshod=0)
-    sampling_rate: sampling_rate in Hz
-    harmonic_num: number of harmonic above F0 (default: 0)
-    sine_amp: amplitude of sine source signal (default: 0.1)
-    add_noise_std: std of additive Gaussian noise (default: 0.003)
-        note that amplitude of noise in unvoiced is decided
-        by sine_amp
-    voiced_threshold: threhold to set U/V given F0 (default: 0)
-    Sine_source, noise_source = SourceModuleHnNSF(F0_sampled)
-    F0_sampled (batchsize, length, 1)
-    Sine_source (batchsize, length, 1)
-    noise_source (batchsize, length 1)
-    uv (batchsize, length, 1)
-    """
-
-    def __init__(self, sampling_rate, upsample_scale, harmonic_num=0, sine_amp=0.1,
-                 add_noise_std=0.003, voiced_threshod=0):
-        super(SourceModuleHnNSF, self).__init__()
-
-        self.sine_amp = sine_amp
-        self.noise_std = add_noise_std
-
-        # to produce sine waveforms
-        self.l_sin_gen = SineGen(sampling_rate, harmonic_num,
-                                 sine_amp, add_noise_std, voiced_threshod)
-
-        # to merge source harmonics into a single excitation
-        self.l_linear = torch.nn.Linear(harmonic_num + 1, 1)
-        self.l_tanh = torch.nn.Tanh()
-
-    def forward(self, x):
-        """
-        Sine_source, noise_source = SourceModuleHnNSF(F0_sampled)
-        F0_sampled (batchsize, length, 1)
-        Sine_source (batchsize, length, 1)
-        noise_source (batchsize, length 1)
-        """
-        # source for harmonic branch
-        with torch.no_grad():
-            sine_wavs, uv, _ = self.l_sin_gen(x.transpose(1, 2))
-            sine_wavs = sine_wavs.transpose(1, 2)
-            uv = uv.transpose(1, 2)
-        sine_merge = self.l_tanh(self.l_linear(sine_wavs))
-
-        # source for noise branch, in the same shape as uv
-        noise = torch.randn_like(uv) * self.sine_amp / 3
-        return sine_merge, noise, uv
+        return sine_waves.transpose(1, 2), uv.transpose(1, 2), noise
 
 
 class SineGen2(torch.nn.Module):
@@ -242,7 +208,8 @@ class SineGen2(torch.nn.Module):
     def __init__(self, samp_rate, upsample_scale, harmonic_num=0,
                  sine_amp=0.1, noise_std=0.003,
                  voiced_threshold=0,
-                 flag_for_pulse=False):
+                 flag_for_pulse=False,
+                 causal=False):
         super(SineGen2, self).__init__()
         self.sine_amp = sine_amp
         self.noise_std = noise_std
@@ -252,6 +219,11 @@ class SineGen2(torch.nn.Module):
         self.voiced_threshold = voiced_threshold
         self.flag_for_pulse = flag_for_pulse
         self.upsample_scale = upsample_scale
+        self.causal = causal
+        if causal is True:
+            self.rand_ini = torch.rand(1, 9)
+            self.rand_ini[:, 0] = 0
+            self.sine_waves = torch.rand(1, 300 * 24000, 9)
 
     def _f02uv(self, f0):
         # generate uv signal
@@ -267,9 +239,12 @@ class SineGen2(torch.nn.Module):
         rad_values = (f0_values / self.sampling_rate) % 1
 
         # initial phase noise (no noise for fundamental component)
-        rand_ini = torch.rand(f0_values.shape[0], f0_values.shape[2], device=f0_values.device)
-        rand_ini[:, 0] = 0
-        rad_values[:, 0, :] = rad_values[:, 0, :] + rand_ini
+        if self.training is False and self.causal is True:
+            rad_values[:, 0, :] = rad_values[:, 0, :] + self.rand_ini.to(rad_values.device)
+        else:
+            rand_ini = torch.rand(f0_values.shape[0], f0_values.shape[2], device=f0_values.device)
+            rand_ini[:, 0] = 0
+            rad_values[:, 0, :] = rad_values[:, 0, :] + rand_ini
 
         # instantanouse phase sine[t] = sin(2*pi \sum_i=1 ^{t} rad)
         if not self.flag_for_pulse:
@@ -279,7 +254,7 @@ class SineGen2(torch.nn.Module):
 
             phase = torch.cumsum(rad_values, dim=1) * 2 * np.pi
             phase = torch.nn.functional.interpolate(phase.transpose(1, 2) * self.upsample_scale,
-                                                    scale_factor=self.upsample_scale, mode="linear").transpose(1, 2)
+                                                    scale_factor=self.upsample_scale, mode="nearest" if self.causal is True else 'linear').transpose(1, 2)
             sines = torch.sin(phase)
         else:
             # If necessary, make sure that the first time step of every
@@ -331,7 +306,10 @@ class SineGen2(torch.nn.Module):
         #        std = self.sine_amp/3 -> max value ~ self.sine_amp
         # .       for voiced regions is self.noise_std
         noise_amp = uv * self.noise_std + (1 - uv) * self.sine_amp / 3
-        noise = noise_amp * torch.randn_like(sine_waves)
+        if self.training is False and self.causal is True:
+            noise = noise_amp * self.sine_waves[:, :sine_waves.shape[1]].to(sine_waves.device)
+        else:
+            noise = noise_amp * torch.randn_like(sine_waves)
 
         # first: set the unvoiced part to 0 by uv
         # then: additive noise
@@ -339,7 +317,7 @@ class SineGen2(torch.nn.Module):
         return sine_waves, uv, noise
 
 
-class SourceModuleHnNSF2(torch.nn.Module):
+class SourceModuleHnNSF(torch.nn.Module):
     """ SourceModule for hn-nsf
     SourceModule(sampling_rate, harmonic_num=0, sine_amp=0.1,
                  add_noise_std=0.003, voiced_threshod=0)
@@ -358,19 +336,24 @@ class SourceModuleHnNSF2(torch.nn.Module):
     """
 
     def __init__(self, sampling_rate, upsample_scale, harmonic_num=0, sine_amp=0.1,
-                 add_noise_std=0.003, voiced_threshod=0):
-        super(SourceModuleHnNSF2, self).__init__()
+                 add_noise_std=0.003, voiced_threshod=0, sinegen_type='1', causal=False):
+        super(SourceModuleHnNSF, self).__init__()
 
         self.sine_amp = sine_amp
         self.noise_std = add_noise_std
 
         # to produce sine waveforms
-        self.l_sin_gen = SineGen2(sampling_rate, upsample_scale, harmonic_num,
-                                  sine_amp, add_noise_std, voiced_threshod)
+        if sinegen_type == '1':
+            self.l_sin_gen = SineGen(sampling_rate, harmonic_num, sine_amp, add_noise_std, voiced_threshod)
+        else:
+            self.l_sin_gen = SineGen2(sampling_rate, upsample_scale, harmonic_num, sine_amp, add_noise_std, voiced_threshod, causal=causal)
 
         # to merge source harmonics into a single excitation
         self.l_linear = torch.nn.Linear(harmonic_num + 1, 1)
         self.l_tanh = torch.nn.Tanh()
+        self.causal = causal
+        if causal is True:
+            self.uv = torch.rand(1, 300 * 24000, 1)
 
     def forward(self, x):
         """
@@ -385,7 +368,10 @@ class SourceModuleHnNSF2(torch.nn.Module):
         sine_merge = self.l_tanh(self.l_linear(sine_wavs))
 
         # source for noise branch, in the same shape as uv
-        noise = torch.randn_like(uv) * self.sine_amp / 3
+        if self.training is False and self.causal is True:
+            noise = self.uv[:, :uv.shape[1]] * self.sine_amp / 3
+        else:
+            noise = torch.randn_like(uv) * self.sine_amp / 3
         return sine_merge, noise, uv
 
 
@@ -425,15 +411,16 @@ class HiFTGenerator(nn.Module):
 
         self.num_kernels = len(resblock_kernel_sizes)
         self.num_upsamples = len(upsample_rates)
-        # NOTE in CosyVoice2, we use the original SourceModuleHnNSF implementation
-        this_SourceModuleHnNSF = SourceModuleHnNSF if self.sampling_rate == 22050 else SourceModuleHnNSF2
-        self.m_source = this_SourceModuleHnNSF(
+        # NOTE in CosyVoice2, we use the original SineGen implementation
+        self.m_source = SourceModuleHnNSF(
             sampling_rate=sampling_rate,
             upsample_scale=np.prod(upsample_rates) * istft_params["hop_len"],
             harmonic_num=nb_harmonics,
             sine_amp=nsf_alpha,
             add_noise_std=nsf_sigma,
-            voiced_threshod=nsf_voiced_threshold)
+            voiced_threshod=nsf_voiced_threshold,
+            sinegen_type='1' if self.sampling_rate == 22050 else '2',
+            causal=False)
         self.f0_upsamp = torch.nn.Upsample(scale_factor=np.prod(upsample_rates) * istft_params["hop_len"])
 
         self.conv_pre = weight_norm(
@@ -580,3 +567,180 @@ class HiFTGenerator(nn.Module):
             s[:, :, :cache_source.shape[2]] = cache_source
         generated_speech = self.decode(x=speech_feat, s=s)
         return generated_speech, s
+
+
+class CausalHiFTGenerator(HiFTGenerator):
+    """
+    HiFTNet Generator: Neural Source Filter + ISTFTNet
+    https://arxiv.org/abs/2309.09493
+    """
+    def __init__(
+            self,
+            in_channels: int = 80,
+            base_channels: int = 512,
+            nb_harmonics: int = 8,
+            sampling_rate: int = 22050,
+            nsf_alpha: float = 0.1,
+            nsf_sigma: float = 0.003,
+            nsf_voiced_threshold: float = 10,
+            upsample_rates: List[int] = [8, 8],
+            upsample_kernel_sizes: List[int] = [16, 16],
+            istft_params: Dict[str, int] = {"n_fft": 16, "hop_len": 4},
+            resblock_kernel_sizes: List[int] = [3, 7, 11],
+            resblock_dilation_sizes: List[List[int]] = [[1, 3, 5], [1, 3, 5], [1, 3, 5]],
+            source_resblock_kernel_sizes: List[int] = [7, 11],
+            source_resblock_dilation_sizes: List[List[int]] = [[1, 3, 5], [1, 3, 5]],
+            lrelu_slope: float = 0.1,
+            audio_limit: float = 0.99,
+            conv_pre_look_right: int = 4,
+            f0_predictor: torch.nn.Module = None,
+    ):
+        torch.nn.Module.__init__(self)
+
+        self.out_channels = 1
+        self.nb_harmonics = nb_harmonics
+        self.sampling_rate = sampling_rate
+        self.istft_params = istft_params
+        self.lrelu_slope = lrelu_slope
+        self.audio_limit = audio_limit
+
+        self.num_kernels = len(resblock_kernel_sizes)
+        self.num_upsamples = len(upsample_rates)
+        self.m_source = SourceModuleHnNSF(
+            sampling_rate=sampling_rate,
+            upsample_scale=np.prod(upsample_rates) * istft_params["hop_len"],
+            harmonic_num=nb_harmonics,
+            sine_amp=nsf_alpha,
+            add_noise_std=nsf_sigma,
+            voiced_threshod=nsf_voiced_threshold,
+            sinegen_type='1' if self.sampling_rate == 22050 else '2',
+            causal=True)
+        self.upsample_rates = upsample_rates
+        self.f0_upsamp = torch.nn.Upsample(scale_factor=np.prod(upsample_rates) * istft_params["hop_len"])
+
+        self.conv_pre = weight_norm(
+            CausalConv1d(in_channels, base_channels, conv_pre_look_right + 1, 1, causal_type='right')
+        )
+
+        # Up
+        self.ups = nn.ModuleList()
+        for i, (u, k) in enumerate(zip(upsample_rates, upsample_kernel_sizes)):
+            self.ups.append(
+                weight_norm(
+                    CausalConv1dUpsample(
+                        base_channels // (2**i),
+                        base_channels // (2**(i + 1)),
+                        k,
+                        u,
+                    )
+                )
+            )
+
+        # Down
+        self.source_downs = nn.ModuleList()
+        self.source_resblocks = nn.ModuleList()
+        downsample_rates = [1] + upsample_rates[::-1][:-1]
+        downsample_cum_rates = np.cumprod(downsample_rates)
+        for i, (u, k, d) in enumerate(zip(downsample_cum_rates[::-1], source_resblock_kernel_sizes, source_resblock_dilation_sizes)):
+            if u == 1:
+                self.source_downs.append(
+                    CausalConv1d(istft_params["n_fft"] + 2, base_channels // (2 ** (i + 1)), 1, 1, causal_type='left')
+                )
+            else:
+                self.source_downs.append(
+                    CausalConv1dDownSample(istft_params["n_fft"] + 2, base_channels // (2 ** (i + 1)), u * 2, u)
+                )
+
+            self.source_resblocks.append(
+                ResBlock(base_channels // (2 ** (i + 1)), k, d, causal=True)
+            )
+
+        self.resblocks = nn.ModuleList()
+        for i in range(len(self.ups)):
+            ch = base_channels // (2**(i + 1))
+            for _, (k, d) in enumerate(zip(resblock_kernel_sizes, resblock_dilation_sizes)):
+                self.resblocks.append(ResBlock(ch, k, d, causal=True))
+
+        self.conv_post = weight_norm(CausalConv1d(ch, istft_params["n_fft"] + 2, 7, 1, causal_type='left'))
+        self.ups.apply(init_weights)
+        self.conv_post.apply(init_weights)
+        self.reflection_pad = nn.ReflectionPad1d((1, 0))
+        self.stft_window = torch.from_numpy(get_window("hann", istft_params["n_fft"], fftbins=True).astype(np.float32))
+        self.conv_pre_look_right = conv_pre_look_right
+        self.f0_predictor = f0_predictor
+
+    def decode(self, x: torch.Tensor, s: torch.Tensor = torch.zeros(1, 1, 0), finalize: bool = True) -> torch.Tensor:
+        s_stft_real, s_stft_imag = self._stft(s.squeeze(1))
+        if finalize is True:
+            x = self.conv_pre(x)
+        else:
+            x = self.conv_pre(x[:, :, :-self.conv_pre_look_right], x[:, :, -self.conv_pre_look_right:])
+            s_stft_real = s_stft_real[:, :, :-int(np.prod(self.upsample_rates) * self.conv_pre_look_right)]
+            s_stft_imag = s_stft_imag[:, :, :-int(np.prod(self.upsample_rates) * self.conv_pre_look_right)]
+        s_stft = torch.cat([s_stft_real, s_stft_imag], dim=1)
+
+        for i in range(self.num_upsamples):
+            x = F.leaky_relu(x, self.lrelu_slope)
+            x = self.ups[i](x)
+
+            if i == self.num_upsamples - 1:
+                x = self.reflection_pad(x)
+
+            # fusion
+            si = self.source_downs[i](s_stft)
+            si = self.source_resblocks[i](si)
+            x = x + si
+
+            xs = None
+            for j in range(self.num_kernels):
+                if xs is None:
+                    xs = self.resblocks[i * self.num_kernels + j](x)
+                else:
+                    xs += self.resblocks[i * self.num_kernels + j](x)
+            x = xs / self.num_kernels
+
+        x = F.leaky_relu(x)
+        x = self.conv_post(x)
+        magnitude = torch.exp(x[:, :self.istft_params["n_fft"] // 2 + 1, :])
+        phase = torch.sin(x[:, self.istft_params["n_fft"] // 2 + 1:, :])  # actually, sin is redundancy
+
+        x = self._istft(magnitude, phase)
+        if finalize is False:
+            x = x[:, :-int(np.prod(self.upsample_rates) * self.istft_params['hop_len'])]
+        x = torch.clamp(x, -self.audio_limit, self.audio_limit)
+        return x
+
+    @torch.inference_mode()
+    def inference(self, speech_feat: torch.Tensor, finalize: bool = True) -> torch.Tensor:
+        # mel->f0 NOTE f0_predictor precision is crucial for causal inference, move self.f0_predictor to cpu if necessary
+        self.f0_predictor.to('cpu')
+        f0 = self.f0_predictor(speech_feat.cpu(), finalize=finalize).to(speech_feat)
+        # f0->source
+        s = self.f0_upsamp(f0[:, None]).transpose(1, 2)  # bs,n,t
+        s, _, _ = self.m_source(s)
+        s = s.transpose(1, 2)
+        if finalize is True:
+            generated_speech = self.decode(x=speech_feat, s=s, finalize=finalize)
+        else:
+            generated_speech = self.decode(x=speech_feat[:, :, :-self.f0_predictor.condnet[0].causal_padding], s=s, finalize=finalize)
+        return generated_speech, s
+
+
+if __name__ == '__main__':
+    torch.backends.cudnn.deterministic = True
+    torch.backends.cudnn.benchmark = False
+    from hyperpyyaml import load_hyperpyyaml
+    with open('./pretrained_models/Fun-CosyVoice3-0.5B/cosyvoice3.yaml', 'r') as f:
+        configs = load_hyperpyyaml(f, overrides={'llm': None, 'flow': None})
+    model = configs['hift']
+    device = 'cuda' if torch.cuda.is_available() else 'cpu'
+    model.to(device)
+    model.eval()
+    max_len, chunk_size, context_size = 300, 30, 8
+    mel = torch.rand(1, 80, max_len).to(device)
+    pred_gt, _ = model.inference(mel)
+    for i in range(0, max_len, chunk_size):
+        finalize = True if i + chunk_size + context_size >= max_len else False
+        pred_chunk, _ = model.inference(mel[:, :, : i + chunk_size + context_size], finalize=finalize)
+        pred_chunk = pred_chunk[:, i * 480:]
+        print((pred_gt[:, i * 480:i * 480 + pred_chunk.shape[1]] - pred_chunk).abs().max().item())

cosyvoice/llm/llm.py

@@ -17,6 +17,7 @@ import random
 import time
 import threading
 from typing import Dict, Optional, Callable, List, Generator
+import numpy as np
 import torch
 from torch import nn
 import torch.nn.functional as F
@@ -56,8 +57,9 @@ class TransformerLM(torch.nn.Module):
         )
 
         # 2. build speech token language model related modules
-        self.sos_eos = 0
+        self.sos = 0
         self.task_id = 1
+        self.eos_token = self.speech_token_size
         self.llm_embedding = torch.nn.Embedding(2, llm_input_size)
         self.llm = llm
         self.llm_decoder = nn.Linear(llm_output_size, speech_token_size + 1)
@@ -85,10 +87,10 @@ class TransformerLM(torch.nn.Module):
         encoder_out = self.text_encoder_affine_layer(encoder_out)
         return encoder_out, encoder_out_lens
 
-    def pad_unpad_sequence(self, sos_eos_emb, embedding, text_token, text_token_len, task_id_emb, speech_token, speech_token_len):
+    def pad_unpad_sequence(self, sos_emb, embedding, text_token, text_token_len, task_id_emb, speech_token, speech_token_len):
         text_token = unpad_sequence(text_token, text_token_len.cpu(), batch_first=True)
         speech_token = unpad_sequence(speech_token, speech_token_len.cpu(), batch_first=True)
-        lm_input = [torch.concat([sos_eos_emb.squeeze(dim=0), embedding[i], text_token[i], task_id_emb.squeeze(dim=0), speech_token[i]], dim=0)
+        lm_input = [torch.concat([sos_emb.squeeze(dim=0), embedding[i], text_token[i], task_id_emb.squeeze(dim=0), speech_token[i]], dim=0)
                     for i in range(len(text_token))]
         lm_input_len = torch.tensor([i.size(0) for i in lm_input], dtype=torch.int32)
         lm_input = pad_sequence(lm_input, batch_first=True, padding_value=IGNORE_ID)
@@ -126,15 +128,15 @@ class TransformerLM(torch.nn.Module):
         embedding = self.spk_embed_affine_layer(embedding)
         embedding = embedding.unsqueeze(1)
 
-        # 3. eos and task_id
-        sos_eos_emb = self.llm_embedding.weight[self.sos_eos].reshape(1, 1, -1)
+        # 3. sos and task_id
+        sos_emb = self.llm_embedding.weight[self.sos].reshape(1, 1, -1)
         task_id_emb = self.llm_embedding.weight[self.task_id].reshape(1, 1, -1)
 
         # 4. encode speech_token
         speech_token = self.speech_embedding(speech_token)
 
         # 5. unpad and pad
-        lm_input, lm_input_len = self.pad_unpad_sequence(sos_eos_emb, embedding, text_token, text_token_len,
+        lm_input, lm_input_len = self.pad_unpad_sequence(sos_emb, embedding, text_token, text_token_len,
                                                          task_id_emb, speech_token, speech_token_len)
 
         # 6. run lm forward
@@ -154,7 +156,7 @@ class TransformerLM(torch.nn.Module):
         num_trials, max_trials = 0, 100
         while True:
             top_ids = self.sampling(weighted_scores, decoded_tokens, sampling)
-            if (not ignore_eos) or (self.speech_token_size not in top_ids):
+            if (not ignore_eos) or (top_ids < self.speech_token_size):
                 break
             num_trials += 1
             if num_trials > max_trials:
@@ -193,13 +195,13 @@ class TransformerLM(torch.nn.Module):
             embedding = torch.zeros(1, 0, self.llm_input_size, dtype=text.dtype).to(device).to(text.dtype)
 
         # 3. concat llm_input
-        sos_eos_emb = self.llm_embedding.weight[self.sos_eos].reshape(1, 1, -1)
+        sos_emb = self.llm_embedding.weight[self.sos].reshape(1, 1, -1)
         task_id_emb = self.llm_embedding.weight[self.task_id].reshape(1, 1, -1)
         if prompt_speech_token_len != 0:
             prompt_speech_token_emb = self.speech_embedding(prompt_speech_token)
         else:
             prompt_speech_token_emb = torch.zeros(1, 0, self.llm_input_size, dtype=text.dtype).to(device)
-        lm_input = torch.concat([sos_eos_emb, embedding, text, task_id_emb, prompt_speech_token_emb], dim=1)
+        lm_input = torch.concat([sos_emb, embedding, text, task_id_emb, prompt_speech_token_emb], dim=1)
 
         # 4. cal min/max_length
         min_len = int((text_len - prompt_text_len) * min_token_text_ratio)
@@ -215,11 +217,8 @@ class TransformerLM(torch.nn.Module):
                                                                   att_mask=torch.tril(torch.ones((1, lm_input.shape[1], lm_input.shape[1]),
                                                                                                  device=lm_input.device)).to(torch.bool))
             logp = self.llm_decoder(y_pred[:, -1]).log_softmax(dim=-1)
-            # force continue decode first token
-            if i == 0:
-                logp[:, self.speech_token_size] = -float('inf')
-            top_ids = self.sampling_ids(logp.squeeze(dim=0), out_tokens, sampling, ignore_eos=True if i < min_len else False).item()
-            if top_ids == self.speech_token_size:
+            top_ids = self.sampling_ids(logp.squeeze(dim=0), out_tokens, sampling, ignore_eos=True if i < min_len else False)
+            if top_ids == self.eos_token:
                 break
             # in stream mode, yield token one by one
             yield top_ids
@@ -276,9 +275,10 @@ class Qwen2LM(TransformerLM):
         self.llm_output_size = llm_output_size
         self.speech_token_size = speech_token_size
         # 2. build speech token language model related modules
-        self.sos_eos = 0
+        self.sos = 0
         self.task_id = 1
-        self.fill_token = 2
+        self.eos_token = speech_token_size
+        self.fill_token = speech_token_size + 2
 
         self.llm_embedding = torch.nn.Embedding(2, llm_input_size)
         self.llm = llm
@@ -301,7 +301,7 @@ class Qwen2LM(TransformerLM):
         self.stop_token_ids = [speech_token_size + i for i in range(3)]
         self.vllm_output_queue = {}
 
-    def prepare_lm_input_target(self, text_token, text_token_emb, text_token_len, speech_token, speech_token_emb, speech_token_len):
+    def prepare_lm_input_target(self, sos_emb, text_token, text_token_emb, text_token_len, task_id_emb, speech_token, speech_token_emb, speech_token_len):
         lm_target, lm_input = [], []
         text_token = unpad_sequence(text_token, text_token_len.cpu(), batch_first=True)
         speech_token = unpad_sequence(speech_token, speech_token_len.cpu(), batch_first=True)
@@ -312,7 +312,7 @@ class Qwen2LM(TransformerLM):
             if random.random() < 0.5 and speech_token_len[i] / text_token_len[i] > self.mix_ratio[1] / self.mix_ratio[0]:
                 this_lm_target, this_lm_input = [], []
                 this_lm_target.append(IGNORE_ID)
-                this_lm_input.append(self.llm_embedding.weight[self.sos_eos].reshape(1, -1))
+                this_lm_input.append(sos_emb.squeeze(dim=0))
                 for j in range(((text_token_len[i] + 1) / self.mix_ratio[0]).ceil().int().item()):
                     this_text_token = text_token[i][j * self.mix_ratio[0]: (j + 1) * self.mix_ratio[0]].tolist()
                     this_speech_token = speech_token[i][j * self.mix_ratio[1]: (j + 1) * self.mix_ratio[1]].tolist()
@@ -320,22 +320,21 @@ class Qwen2LM(TransformerLM):
                         assert len(this_speech_token) == self.mix_ratio[1]
                         this_lm_target += [IGNORE_ID] * (self.mix_ratio[0] - 1)
                         this_lm_target += this_speech_token
-                        this_lm_target.append(self.speech_token_size + 2)
+                        this_lm_target.append(self.fill_token)
                         this_lm_input.append(text_token_emb[i][j * self.mix_ratio[0]: (j + 1) * self.mix_ratio[0]])
                         this_lm_input.append(speech_token_emb[i][j * self.mix_ratio[1]: (j + 1) * self.mix_ratio[1]])
                     else:
                         this_lm_target += [-1] * len(this_text_token)
                         this_lm_target += speech_token[i][j * self.mix_ratio[1]:].tolist()
-                        this_lm_target.append(self.speech_token_size)
+                        this_lm_target.append(self.eos_token)
                         this_lm_input.append(text_token_emb[i][j * self.mix_ratio[0]:])
-                        this_lm_input.append(self.llm_embedding.weight[self.task_id].reshape(1, -1))
+                        this_lm_input.append(task_id_emb.squeeze(dim=0))
                         this_lm_input.append(speech_token_emb[i][j * self.mix_ratio[1]:])
                 this_lm_target, this_lm_input = torch.tensor(this_lm_target), torch.concat(this_lm_input, dim=0)
             # unistream sequence
             else:
-                this_lm_target = torch.tensor([IGNORE_ID] * (1 + text_token_len[i]) + speech_token[i].tolist() + [self.speech_token_size])
-                this_lm_input = torch.concat([self.llm_embedding.weight[self.sos_eos].reshape(1, -1), text_token_emb[i],
-                                              self.llm_embedding.weight[self.task_id].reshape(1, -1), speech_token_emb[i]], dim=0)
+                this_lm_target = torch.tensor([IGNORE_ID] * (1 + text_token_len[i]) + speech_token[i].tolist() + [self.eos_token])
+                this_lm_input = torch.concat([sos_emb.squeeze(dim=0), text_token_emb[i], task_id_emb.squeeze(dim=0), speech_token_emb[i]], dim=0)
             lm_target.append(this_lm_target)
             lm_input.append(this_lm_input)
         lm_input_len = torch.tensor([i.size(0) for i in lm_input], dtype=torch.int32)
@@ -363,11 +362,16 @@ class Qwen2LM(TransformerLM):
         # 1. encode text_token
         text_token_emb = self.llm.model.model.embed_tokens(text_token)
 
+        # 3. sos and task_id
+        sos_emb = self.llm_embedding.weight[self.sos].reshape(1, 1, -1)
+        task_id_emb = self.llm_embedding.weight[self.task_id].reshape(1, 1, -1)
+
         # 2. encode speech_token
         speech_token_emb = self.speech_embedding(speech_token)
 
         # 3. prepare llm_input/target
-        lm_target, lm_input, lm_input_len = self.prepare_lm_input_target(text_token, text_token_emb, text_token_len, speech_token, speech_token_emb, speech_token_len)
+        lm_target, lm_input, lm_input_len = self.prepare_lm_input_target(sos_emb, text_token, text_token_emb, text_token_len, task_id_emb,
+                                                                         speech_token, speech_token_emb, speech_token_len)
         lm_target = lm_target.to(device)
 
         # 4. run lm forward
@@ -392,6 +396,10 @@ class Qwen2LM(TransformerLM):
         # 1. encode text_token
         text_token_emb = self.llm.model.model.embed_tokens(text_token)
 
+        # 3. sos and task_id
+        sos_emb = self.llm_embedding.weight[self.sos].reshape(1, 1, -1)
+        task_id_emb = self.llm_embedding.weight[self.task_id].reshape(1, 1, -1)
+
         # 2. encode speech_token
         speech_token = unpad_sequence(speech_token, speech_token_len.cpu(), batch_first=True)
         reject_speech_token = unpad_sequence(reject_speech_token, reject_speech_token_len.cpu(), batch_first=True)
@@ -401,8 +409,8 @@ class Qwen2LM(TransformerLM):
         speech_token_combined_emb = self.speech_embedding(speech_token_combined)
 
         # 3. prepare llm_input/target
-        lm_target, lm_input, lm_input_len = self.prepare_lm_input_target(text_token.repeat(2, 1), text_token_emb.repeat(2, 1, 1), text_token_len.repeat(2),
-                                                                         speech_token_combined, speech_token_combined_emb, speech_token_combined_len)
+        lm_target, lm_input, lm_input_len = self.prepare_lm_input_target(sos_emb, text_token.repeat(2, 1), text_token_emb.repeat(2, 1, 1), text_token_len.repeat(2),
+                                                                         task_id_emb, speech_token_combined, speech_token_combined_emb, speech_token_combined_len)
         lm_target = lm_target.to(device)
 
         # 4. run lm forward
@@ -445,13 +453,13 @@ class Qwen2LM(TransformerLM):
         text = self.llm.model.model.embed_tokens(text)
 
         # 3. concat llm_input
-        sos_eos_emb = self.llm_embedding.weight[self.sos_eos].reshape(1, 1, -1)
+        sos_emb = self.llm_embedding.weight[self.sos].reshape(1, 1, -1)
         task_id_emb = self.llm_embedding.weight[self.task_id].reshape(1, 1, -1)
         if prompt_speech_token_len != 0:
             prompt_speech_token_emb = self.speech_embedding(prompt_speech_token)
         else:
             prompt_speech_token_emb = torch.zeros(1, 0, self.llm_input_size, dtype=text.dtype).to(device)
-        lm_input = torch.concat([sos_eos_emb, text, task_id_emb, prompt_speech_token_emb], dim=1)
+        lm_input = torch.concat([sos_emb, text, task_id_emb, prompt_speech_token_emb], dim=1)
 
         # 4. cal min/max_length
         min_len = int((text_len - prompt_text_len) * min_token_text_ratio)
@@ -500,11 +508,9 @@ class Qwen2LM(TransformerLM):
                                                           masks=torch.tril(torch.ones((1, lm_input.shape[1], lm_input.shape[1]), device=lm_input.device)).to(torch.bool),
                                                           cache=cache)
                 logp = self.llm_decoder(y_pred[:, -1]).log_softmax(dim=-1)
-                top_ids = self.sampling_ids(logp.squeeze(dim=0), out_tokens, sampling, ignore_eos=True if i < min_len else False).item()
-                if top_ids == self.speech_token_size:
+                top_ids = self.sampling_ids(logp.squeeze(dim=0), out_tokens, sampling, ignore_eos=True if i < min_len else False)
+                if top_ids in self.stop_token_ids:
                     break
-                if top_ids > self.speech_token_size:
-                    continue
                 # in stream mode, yield token one by one
                 yield top_ids
                 out_tokens.append(top_ids)
@@ -526,20 +532,20 @@ class Qwen2LM(TransformerLM):
 
         device = prompt_text.device
         # 1. prepare input
-        sos_eos_emb = self.llm_embedding.weight[self.sos_eos].reshape(1, 1, -1)
+        sos_emb = self.llm_embedding.weight[self.sos].reshape(1, 1, -1)
         task_id_emb = self.llm_embedding.weight[self.task_id].reshape(1, 1, -1)
         if prompt_speech_token_len != 0:
             prompt_speech_token_emb = self.speech_embedding(prompt_speech_token)
         else:
             prompt_speech_token_emb = torch.zeros(1, 0, self.llm_input_size, dtype=prompt_text.dtype).to(device)
-        lm_input = torch.concat([sos_eos_emb], dim=1)
+        lm_input = torch.concat([sos_emb], dim=1)
 
         # 2. iterate text
         out_tokens = []
         cache = None
         # NOTE init prompt_text as text_cache as it is basically impossible prompt_speech_token/prompt_text < 15/5
         text_cache = self.llm.model.model.embed_tokens(prompt_text)
-        next_fill_index = -1
+        next_fill_index = (int(prompt_speech_token.shape[1] / self.mix_ratio[1]) + 1) * self.mix_ratio[1] - prompt_speech_token.shape[1]
         for this_text in text:
             text_cache = torch.concat([text_cache, self.llm.model.model.embed_tokens(this_text)], dim=1)
             # prompt_speech_token_emb not empty, try append to lm_input
@@ -554,12 +560,12 @@ class Qwen2LM(TransformerLM):
                     break
             # no prompt_speech_token_emb remain, can decode some speech token
             if prompt_speech_token_emb.size(1) == 0:
-                if (len(out_tokens) != 0 and out_tokens[-1] == self.speech_token_size + 2) or (len(out_tokens) == 0 and lm_input.size(1) == 1):
+                if (len(out_tokens) != 0 and out_tokens[-1] == self.fill_token) or (len(out_tokens) == 0 and lm_input.size(1) == 1):
                     logging.info('get fill token, need to append more text token')
                     if text_cache.size(1) >= self.mix_ratio[0]:
                         lm_input_text = text_cache[:, :self.mix_ratio[0]]
                         logging.info('append {} text token'.format(lm_input_text.size(1)))
-                        if len(out_tokens) != 0 and out_tokens[-1] == self.speech_token_size + 2:
+                        if len(out_tokens) != 0 and out_tokens[-1] == self.fill_token:
                             lm_input = lm_input_text
                         else:
                             lm_input = torch.concat([lm_input, lm_input_text], dim=1)
@@ -574,16 +580,16 @@ class Qwen2LM(TransformerLM):
                                                               cache=cache)
                     logp = self.llm_decoder(y_pred[:, -1]).log_softmax(dim=-1)
                     if next_fill_index != -1 and len(out_tokens) == next_fill_index:
-                        top_ids = self.speech_token_size + 2
+                        top_ids = self.fill_token
                         next_fill_index += (self.mix_ratio[1] + 1)
                     else:
-                        top_ids = self.sampling_ids(logp.squeeze(dim=0), out_tokens, sampling, ignore_eos=True).item()
-                    if top_ids == self.speech_token_size + 2:
+                        top_ids = self.sampling_ids(logp.squeeze(dim=0), out_tokens, sampling, ignore_eos=True)
+                    if top_ids == self.fill_token:
                         next_fill_index = len(out_tokens) + self.mix_ratio[1] + 1
                         logging.info('fill_token index {} next fill_token index {}'.format(len(out_tokens), next_fill_index))
                     out_tokens.append(top_ids)
                     if top_ids >= self.speech_token_size:
-                        if top_ids == self.speech_token_size + 2:
+                        if top_ids == self.fill_token:
                             break
                         else:
                             raise ValueError('should not get token {}'.format(top_ids))
@@ -599,13 +605,135 @@ class Qwen2LM(TransformerLM):
                                                       masks=torch.tril(torch.ones((1, seq_len, seq_len), device=lm_input.device)).to(torch.bool),
                                                       cache=cache)
             logp = self.llm_decoder(y_pred[:, -1]).log_softmax(dim=-1)
-            top_ids = self.sampling_ids(logp.squeeze(dim=0), out_tokens, sampling, ignore_eos=False).item()
+            top_ids = self.sampling_ids(logp.squeeze(dim=0), out_tokens, sampling, ignore_eos=False)
             out_tokens.append(top_ids)
             if top_ids >= self.speech_token_size:
-                if top_ids == self.speech_token_size:
+                if top_ids == self.eos_token:
                     break
                 else:
                     raise ValueError('should not get token {}'.format(top_ids))
             # in stream mode, yield token one by one
             yield top_ids
             lm_input = self.speech_embedding.weight[top_ids].reshape(1, 1, -1)
+
+
+class CosyVoice3LM(Qwen2LM):
+    def __init__(
+            self,
+            llm_input_size: int,
+            llm_output_size: int,
+            speech_token_size: int,
+            llm: torch.nn.Module,
+            sampling: Callable,
+            length_normalized_loss: bool = True,
+            lsm_weight: float = 0.0,
+            mix_ratio: List[int] = [5, 15],
+    ):
+        torch.nn.Module.__init__(self)
+        self.llm_input_size = llm_input_size
+        self.llm_output_size = llm_output_size
+        self.speech_token_size = speech_token_size
+        # 2. build speech token language model related modules
+        self.sos = speech_token_size + 0
+        self.eos_token = speech_token_size + 1
+        self.task_id = speech_token_size + 2
+        self.fill_token = speech_token_size + 3
+
+        self.llm = llm
+        self.llm_decoder = nn.Linear(llm_output_size, speech_token_size + 200, bias=False)
+        self.criterion_ce = LabelSmoothingLoss(
+            size=speech_token_size + 200,
+            padding_idx=IGNORE_ID,
+            smoothing=lsm_weight,
+            normalize_length=length_normalized_loss,
+        )
+
+        # 3. [Optional] build speech token related modules
+        self.speech_embedding = torch.nn.Embedding(speech_token_size + 200, llm_input_size)
+
+        # 4. sampling method
+        self.sampling = sampling
+        self.mix_ratio = mix_ratio
+
+        # 5. vllm related
+        self.stop_token_ids = [speech_token_size + i for i in range(200)]
+        self.vllm_output_queue = {}
+
+    def forward(
+            self,
+            batch: dict,
+            device: torch.device,
+    ) -> Dict[str, Optional[torch.Tensor]]:
+        """
+        Args:
+            text: (B, L, D)
+            text_lengths: (B,)
+            audio: (B, T, N) or (B, T)
+            audio_lengths: (B,)
+        """
+        text_token = batch['text_token'].to(device)
+        text_token_len = batch['text_token_len'].to(device)
+        speech_token = batch['speech_token'].to(device)
+        speech_token_len = batch['speech_token_len'].to(device)
+        # NOTE should append instruct_token to sequence, not implemented yet
+        instruct_token = batch['instruct_token'].to(device)
+        instruct_token_len = batch['instruct_token_len'].to(device)
+
+        # 1. encode text_token
+        text_token_emb = self.llm.model.model.embed_tokens(text_token)
+
+        # 3. sos and task_id
+        sos_emb = self.speech_embedding.weight[self.sos].reshape(1, 1, -1)
+        task_id_emb = self.speech_embedding.weight[self.task_id].reshape(1, 1, -1)
+
+        # 2. encode speech_token
+        speech_token_emb = self.speech_embedding(speech_token)
+
+        # 3. prepare llm_input/target
+        lm_target, lm_input, lm_input_len = self.prepare_lm_input_target(sos_emb, text_token, text_token_emb, text_token_len, task_id_emb,
+                                                                         speech_token, speech_token_emb, speech_token_len)
+        lm_target = lm_target.to(device)
+
+        # 4. run lm forward
+        lm_output, lm_output_mask = self.llm(lm_input, lm_input_len.to(device))
+        logits = self.llm_decoder(lm_output)
+        loss = self.criterion_ce(logits, lm_target.to(device))
+        acc = th_accuracy(logits.view(-1, self.speech_token_size + 3), lm_target, ignore_label=IGNORE_ID)
+        return {'loss': loss, 'acc': acc}
+
+    @torch.inference_mode()
+    def inference(
+            self,
+            text: torch.Tensor,
+            text_len: torch.Tensor,
+            prompt_text: torch.Tensor,
+            prompt_text_len: torch.Tensor,
+            prompt_speech_token: torch.Tensor,
+            prompt_speech_token_len: torch.Tensor,
+            embedding: torch.Tensor,
+            sampling: int = 25,
+            max_token_text_ratio: float = 20,
+            min_token_text_ratio: float = 2,
+            uuid: str = '',
+    ) -> Generator[torch.Tensor, None, None]:
+        device = text.device
+        text = torch.concat([prompt_text, text], dim=1)
+        text_len += prompt_text_len
+        text = self.llm.model.model.embed_tokens(text)
+
+        # 3. concat llm_input
+        sos_emb = self.speech_embedding.weight[self.sos].reshape(1, 1, -1)
+        task_id_emb = self.speech_embedding.weight[self.task_id].reshape(1, 1, -1)
+        if prompt_speech_token_len != 0:
+            prompt_speech_token_emb = self.speech_embedding(prompt_speech_token)
+        else:
+            prompt_speech_token_emb = torch.zeros(1, 0, self.llm_input_size, dtype=text.dtype).to(device)
+        lm_input = torch.concat([sos_emb, text, task_id_emb, prompt_speech_token_emb], dim=1)
+
+        # 4. cal min/max_length
+        min_len = int((text_len - prompt_text_len) * min_token_text_ratio)
+        max_len = int((text_len - prompt_text_len) * max_token_text_ratio)
+
+        # 5. step by step decode
+        for token in self.inference_wrapper(lm_input, sampling, min_len, max_len, uuid):
+            yield token

cosyvoice/tokenizer/tokenizer.py

@@ -238,7 +238,7 @@ def get_tokenizer(
     )
 
 
-class QwenTokenizer():
+class CosyVoice2Tokenizer():
     def __init__(self, token_path, skip_special_tokens=True):
         super().__init__()
         # NOTE: non-chat model, all these special tokens keep randomly initialized.
@@ -271,9 +271,57 @@ class QwenTokenizer():
         return text
 
 
+class CosyVoice3Tokenizer(CosyVoice2Tokenizer):
+    def __init__(self, token_path, skip_special_tokens=True):
+        # NOTE: non-chat model, all these special tokens keep randomly initialized.
+        special_tokens = {
+            'eos_token': '<|endoftext|>',
+            'pad_token': '<|endoftext|>',
+            'additional_special_tokens': [
+                '<|im_start|>', '<|im_end|>', '<|endofprompt|>',
+                '[breath]', '<strong>', '</strong>', '[noise]',
+                '[laughter]', '[cough]', '[clucking]', '[accent]',
+                '[quick_breath]',
+                "<laughter>", "</laughter>",
+                "[hissing]", "[sigh]", "[vocalized-noise]",
+                "[lipsmack]", "[mn]", "<|endofsystem|>",
+                "[AA]", "[AA0]", "[AA1]", "[AA2]", "[AE]", "[AE0]", "[AE1]", "[AE2]", "[AH]", "[AH0]", "[AH1]", "[AH2]",
+                "[AO]", "[AO0]", "[AO1]", "[AO2]", "[AW]", "[AW0]", "[AW1]", "[AW2]", "[AY]", "[AY0]", "[AY1]", "[AY2]",
+                "[B]", "[CH]", "[D]", "[DH]", "[EH]", "[EH0]", "[EH1]", "[EH2]", "[ER]", "[ER0]", "[ER1]", "[ER2]", "[EY]",
+                "[EY0]", "[EY1]", "[EY2]", "[F]", "[G]", "[HH]", "[IH]", "[IH0]", "[IH1]", "[IH2]", "[IY]", "[IY0]", "[IY1]",
+                "[IY2]", "[JH]", "[K]", "[L]", "[M]", "[N]", "[NG]", "[OW]", "[OW0]", "[OW1]", "[OW2]", "[OY]", "[OY0]",
+                "[OY1]", "[OY2]", "[P]", "[R]", "[S]", "[SH]", "[T]", "[TH]", "[UH]", "[UH0]", "[UH1]", "[UH2]", "[UW]",
+                "[UW0]", "[UW1]", "[UW2]", "[V]", "[W]", "[Y]", "[Z]", "[ZH]",
+                "[a]", "[ai]", "[an]", "[ang]", "[ao]", "[b]", "[c]", "[ch]", "[d]", "[e]", "[ei]", "[en]", "[eng]", "[f]",
+                "[g]", "[h]", "[i]", "[ian]", "[in]", "[ing]", "[iu]", "[ià]", "[iàn]", "[iàng]", "[iào]", "[iá]", "[ián]",
+                "[iáng]", "[iáo]", "[iè]", "[ié]", "[iòng]", "[ióng]", "[iù]", "[iú]", "[iā]", "[iān]", "[iāng]", "[iāo]",
+                "[iē]", "[iě]", "[iōng]", "[iū]", "[iǎ]", "[iǎn]", "[iǎng]", "[iǎo]", "[iǒng]", "[iǔ]", "[j]", "[k]", "[l]",
+                "[m]", "[n]", "[o]", "[ong]", "[ou]", "[p]", "[q]", "[r]", "[s]", "[sh]", "[t]", "[u]", "[uang]", "[ue]",
+                "[un]", "[uo]", "[uà]", "[uài]", "[uàn]", "[uàng]", "[uá]", "[uái]", "[uán]", "[uáng]", "[uè]", "[ué]", "[uì]",
+                "[uí]", "[uò]", "[uó]", "[uā]", "[uāi]", "[uān]", "[uāng]", "[uē]", "[uě]", "[uī]", "[uō]", "[uǎ]", "[uǎi]",
+                "[uǎn]", "[uǎng]", "[uǐ]", "[uǒ]", "[vè]", "[w]", "[x]", "[y]", "[z]", "[zh]", "[à]", "[ài]", "[àn]", "[àng]",
+                "[ào]", "[á]", "[ái]", "[án]", "[áng]", "[áo]", "[è]", "[èi]", "[èn]", "[èng]", "[èr]", "[é]", "[éi]", "[én]",
+                "[éng]", "[ér]", "[ì]", "[ìn]", "[ìng]", "[í]", "[ín]", "[íng]", "[ò]", "[òng]", "[òu]", "[ó]", "[óng]", "[óu]",
+                "[ù]", "[ùn]", "[ú]", "[ún]", "[ā]", "[āi]", "[ān]", "[āng]", "[āo]", "[ē]", "[ēi]", "[ēn]", "[ēng]", "[ě]",
+                "[ěi]", "[ěn]", "[ěng]", "[ěr]", "[ī]", "[īn]", "[īng]", "[ō]", "[ōng]", "[ōu]", "[ū]", "[ūn]", "[ǎ]", "[ǎi]",
+                "[ǎn]", "[ǎng]", "[ǎo]", "[ǐ]", "[ǐn]", "[ǐng]", "[ǒ]", "[ǒng]", "[ǒu]", "[ǔ]", "[ǔn]", "[ǘ]", "[ǚ]", "[ǜ]"
+            ]
+        }
+        self.special_tokens = special_tokens
+        self.tokenizer = AutoTokenizer.from_pretrained(token_path)
+        self.tokenizer.add_special_tokens(special_tokens)
+        self.skip_special_tokens = skip_special_tokens
+
+
 @lru_cache(maxsize=None)
 def get_qwen_tokenizer(
     token_path: str,
-    skip_special_tokens: bool
-) -> QwenTokenizer:
-    return QwenTokenizer(token_path=token_path, skip_special_tokens=skip_special_tokens)
+    skip_special_tokens: bool,
+    version: str = 'cosyvoice2'
+):
+    if version == 'cosyvoice2':
+        return CosyVoice2Tokenizer(token_path=token_path, skip_special_tokens=skip_special_tokens)
+    elif version == 'cosyvoice3':
+        return CosyVoice3Tokenizer(token_path=token_path, skip_special_tokens=skip_special_tokens)
+    else:
+        raise ValueError

cosyvoice/transformer/convolution.py

@@ -19,6 +19,7 @@ from typing import Tuple
 
 import torch
 from torch import nn
+import torch.nn.functional as F
 
 
 class ConvolutionModule(nn.Module):
@@ -143,3 +144,115 @@ class ConvolutionModule(nn.Module):
             x.masked_fill_(~mask_pad, 0.0)
 
         return x.transpose(1, 2), new_cache
+
+
+# NOTE(Xiang Lyu) causal conv module used in convolution-based vocoder
+class CausalConv1d(torch.nn.Conv1d):
+    def __init__(
+        self,
+        in_channels: int,
+        out_channels: int,
+        kernel_size: int,
+        stride: int = 1,
+        dilation: int = 1,
+        groups: int = 1,
+        bias: bool = True,
+        padding_mode: str = 'zeros',
+        causal_type: str = 'left',
+        device=None,
+        dtype=None
+    ) -> None:
+        super(CausalConv1d, self).__init__(in_channels, out_channels,
+                                           kernel_size, stride=1,
+                                           padding=0, dilation=dilation,
+                                           groups=groups, bias=bias,
+                                           padding_mode=padding_mode,
+                                           device=device, dtype=dtype)
+        assert stride == 1
+        self.causal_padding = int((kernel_size * dilation - dilation) / 2) * 2 + (kernel_size + 1) % 2
+        assert causal_type in ['left', 'right']
+        self.causal_type = causal_type
+
+    def forward(self, x: torch.Tensor, cache: torch.Tensor = torch.zeros(0, 0, 0)) -> Tuple[torch.Tensor]:
+        input_timestep = x.shape[2]
+        if cache.size(2) == 0:
+            cache = torch.zeros(x.shape[0], x.shape[1], self.causal_padding).to(x)
+        assert cache.size(2) == self.causal_padding
+        if self.causal_type == 'left':
+            x = torch.concat([cache, x], dim=2)
+        else:
+            x = torch.concat([x, cache], dim=2)
+        x = super(CausalConv1d, self).forward(x)
+        assert x.shape[2] == input_timestep
+        return x
+
+
+class CausalConv1dDownSample(torch.nn.Conv1d):
+    def __init__(
+        self,
+        in_channels: int,
+        out_channels: int,
+        kernel_size: int,
+        stride: int = 1,
+        dilation: int = 1,
+        groups: int = 1,
+        bias: bool = True,
+        padding_mode: str = 'zeros',
+        device=None,
+        dtype=None
+    ) -> None:
+        super(CausalConv1dDownSample, self).__init__(in_channels, out_channels,
+                                                     kernel_size, stride,
+                                                     padding=0, dilation=dilation,
+                                                     groups=groups, bias=bias,
+                                                     padding_mode=padding_mode,
+                                                     device=device, dtype=dtype)
+        assert stride != 1 and dilation == 1
+        assert kernel_size % stride == 0
+        self.causal_padding = stride - 1
+
+    def forward(self, x: torch.Tensor, cache: torch.Tensor = torch.zeros(0, 0, 0)) -> Tuple[torch.Tensor, torch.Tensor]:
+        if cache.size(2) == 0:
+            x = F.pad(x, (self.causal_padding, 0), value=0.0)
+        else:
+            assert cache.size(2) == self.causal_padding
+            x = torch.concat([cache, x], dim=2)
+        x = super(CausalConv1dDownSample, self).forward(x)
+        return x
+
+
+class CausalConv1dUpsample(torch.nn.Conv1d):
+    def __init__(
+        self,
+        in_channels: int,
+        out_channels: int,
+        kernel_size: int,
+        stride: int = 1,
+        dilation: int = 1,
+        groups: int = 1,
+        bias: bool = True,
+        padding_mode: str = 'zeros',
+        device=None,
+        dtype=None
+    ) -> None:
+        super(CausalConv1dUpsample, self).__init__(in_channels, out_channels,
+                                                   kernel_size, 1,
+                                                   padding=0, dilation=dilation,
+                                                   groups=groups, bias=bias,
+                                                   padding_mode=padding_mode,
+                                                   device=device, dtype=dtype)
+        assert dilation == 1
+        self.causal_padding = kernel_size - 1
+        self.upsample = torch.nn.Upsample(scale_factor=stride, mode='nearest')
+
+    def forward(self, x: torch.Tensor, cache: torch.Tensor = torch.zeros(0, 0, 0)) -> Tuple[torch.Tensor, torch.Tensor]:
+        x = self.upsample(x)
+        input_timestep = x.shape[2]
+        if cache.size(2) == 0:
+            x = F.pad(x, (self.causal_padding, 0), value=0.0)
+        else:
+            assert cache.size(2) == self.causal_padding
+            x = torch.concat([cache, x], dim=2)
+        x = super(CausalConv1dUpsample, self).forward(x)
+        assert input_timestep == x.shape[2]
+        return x

cosyvoice/transformer/upsample_encoder.py

@@ -64,17 +64,18 @@ class Upsample1D(nn.Module):
 
 
 class PreLookaheadLayer(nn.Module):
-    def __init__(self, channels: int, pre_lookahead_len: int = 1):
+    def __init__(self, in_channels: int, channels: int, pre_lookahead_len: int = 1):
         super().__init__()
+        self.in_channels = in_channels
         self.channels = channels
         self.pre_lookahead_len = pre_lookahead_len
         self.conv1 = nn.Conv1d(
-            channels, channels,
+            in_channels, channels,
             kernel_size=pre_lookahead_len + 1,
             stride=1, padding=0,
         )
         self.conv2 = nn.Conv1d(
-            channels, channels,
+            channels, in_channels,
             kernel_size=3, stride=1, padding=0,
         )
 
@@ -199,7 +200,7 @@ class UpsampleConformerEncoder(torch.nn.Module):
         # convolution module definition
         convolution_layer_args = (output_size, cnn_module_kernel, activation,
                                   cnn_module_norm, causal)
-        self.pre_lookahead_layer = PreLookaheadLayer(channels=512, pre_lookahead_len=3)
+        self.pre_lookahead_layer = PreLookaheadLayer(in_channels=512, channels=512, pre_lookahead_len=3)
         self.encoders = torch.nn.ModuleList([
             ConformerEncoderLayer(
                 output_size,

cosyvoice/utils/class_utils.py

@@ -32,10 +32,10 @@ from cosyvoice.transformer.attention import (MultiHeadedAttention,
                                              RelPositionMultiHeadedAttention)
 from cosyvoice.transformer.embedding import EspnetRelPositionalEncoding
 from cosyvoice.transformer.subsampling import LegacyLinearNoSubsampling
-from cosyvoice.llm.llm import TransformerLM, Qwen2LM
-from cosyvoice.flow.flow import MaskedDiffWithXvec, CausalMaskedDiffWithXvec
-from cosyvoice.hifigan.generator import HiFTGenerator
-from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model
+from cosyvoice.llm.llm import TransformerLM, Qwen2LM, CosyVoice3LM
+from cosyvoice.flow.flow import MaskedDiffWithXvec, CausalMaskedDiffWithXvec, CausalMaskedDiffWithDiT
+from cosyvoice.hifigan.generator import HiFTGenerator, CausalHiFTGenerator
+from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model, CosyVoice3Model
 
 
 COSYVOICE_ACTIVATION_CLASSES = {
@@ -80,4 +80,6 @@ def get_model_type(configs):
         return CosyVoiceModel
     if isinstance(configs['llm'], Qwen2LM) and isinstance(configs['flow'], CausalMaskedDiffWithXvec) and isinstance(configs['hift'], HiFTGenerator):
         return CosyVoice2Model
+    if isinstance(configs['llm'], CosyVoice3LM) and isinstance(configs['flow'], CausalMaskedDiffWithDiT) and isinstance(configs['hift'], CausalHiFTGenerator):
+        return CosyVoice3Model
     raise TypeError('No valid model type found!')

cosyvoice/utils/common.py

@@ -25,6 +25,33 @@ import torch
 
 IGNORE_ID = -1
 
+instruct_list = ["You are a helpful assistant. 请用广东话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用东北话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用甘肃话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用贵州话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用河南话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用湖北话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用湖南话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用江西话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用闽南话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用宁夏话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用山西话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用陕西话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用山东话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用上海话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用四川话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用天津话表达。<|endofprompt|>",
+                 "You are a helpful assistant. 请用云南话表达。<|endofprompt|>",
+                 "You are a helpful assistant. Please say a sentence as loudly as possible.<|endofprompt|>",
+                 "You are a helpful assistant. Please say a sentence in a very soft voice.<|endofprompt|>",
+                 "You are a helpful assistant. 请用尽可能慢地语速说一句话。<|endofprompt|>",
+                 "You are a helpful assistant. 请用尽可能快地语速说一句话。<|endofprompt|>",
+                 "You are a helpful assistant. 请非常开心地说一句话。<|endofprompt|>",
+                 "You are a helpful assistant. 请非常伤心地说一句话。<|endofprompt|>",
+                 "You are a helpful assistant. 请非常生气地说一句话。<|endofprompt|>",
+                 "You are a helpful assistant. 我想体验一下小猪佩奇风格，可以吗？<|endofprompt|>",
+                 "You are a helpful assistant. 你可以尝试用机器人的方式解答吗？<|endofprompt|>"]
+
 
 def pad_list(xs: List[torch.Tensor], pad_value: int):
     """Perform padding for the list of tensors.
@@ -130,12 +157,12 @@ def nucleus_sampling(weighted_scores, top_p=0.8, top_k=25):
             break
     prob = torch.tensor(prob).to(weighted_scores)
     indices = torch.tensor(indices, dtype=torch.long).to(weighted_scores.device)
-    top_ids = indices[prob.multinomial(1, replacement=True)]
+    top_ids = indices[prob.multinomial(1, replacement=True)].item()
     return top_ids
 
 
 def random_sampling(weighted_scores, decoded_tokens, sampling):
-    top_ids = weighted_scores.softmax(dim=0).multinomial(1, replacement=True)
+    top_ids = weighted_scores.softmax(dim=0).multinomial(1, replacement=True).item()
     return top_ids
 
 

cosyvoice/utils/file_utils.py

@@ -41,11 +41,11 @@ def read_json_lists(list_file):
     return results
 
 
-def load_wav(wav, target_sr):
+def load_wav(wav, target_sr, min_sr=16000):
     speech, sample_rate = torchaudio.load(wav, backend='soundfile')
     speech = speech.mean(dim=0, keepdim=True)
     if sample_rate != target_sr:
-        assert sample_rate > target_sr, 'wav sample rate {} must be greater than {}'.format(sample_rate, target_sr)
+        assert sample_rate >= min_sr, 'wav sample rate {} must be greater than {}'.format(sample_rate, target_sr)
         speech = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=target_sr)(speech)
     return speech
 
@@ -88,30 +88,18 @@ def convert_onnx_to_trt(trt_model, trt_kwargs, onnx_model, fp16):
     logging.info("Succesfully convert onnx to trt...")
 
 
+# NOTE do not support bistream inference as only speech token embedding/head is kept
 def export_cosyvoice2_vllm(model, model_path, device):
     if os.path.exists(model_path):
         return
-    pad_to = DEFAULT_VOCAB_PADDING_SIZE = 64
-    vocab_size = model.speech_embedding.num_embeddings
-    feature_size = model.speech_embedding.embedding_dim
-    pad_vocab_size = ((vocab_size + pad_to - 1) // pad_to) * pad_to
 
     dtype = torch.bfloat16
     # lm_head
-    new_lm_head = torch.nn.Linear(in_features=feature_size, out_features=pad_vocab_size, bias=True)
-    with torch.no_grad():
-        new_lm_head.weight[:vocab_size] = model.llm_decoder.weight
-        new_lm_head.bias[:vocab_size] = model.llm_decoder.bias
-        new_lm_head.weight[vocab_size:] = 0
-        new_lm_head.bias[vocab_size:] = 0
-    model.llm.model.lm_head = new_lm_head
-    new_codec_embed = torch.nn.Linear(in_features=feature_size, out_features=pad_vocab_size)
+    use_bias = True if model.llm_decoder.bias is not None else False
+    model.llm.model.lm_head = model.llm_decoder
     # embed_tokens
     embed_tokens = model.llm.model.model.embed_tokens
-    with torch.no_grad():
-        new_codec_embed.weight[:vocab_size] = model.speech_embedding.weight
-        new_codec_embed.weight[vocab_size:] = 0
-    model.llm.model.set_input_embeddings(new_codec_embed)
+    model.llm.model.set_input_embeddings(model.speech_embedding)
     model.llm.model.to(device)
     model.llm.model.to(dtype)
     tmp_vocab_size = model.llm.model.config.vocab_size
@@ -119,11 +107,12 @@ def export_cosyvoice2_vllm(model, model_path, device):
     del model.llm.model.generation_config.eos_token_id
     del model.llm.model.config.bos_token_id
     del model.llm.model.config.eos_token_id
-    model.llm.model.config.vocab_size = pad_vocab_size
+    model.llm.model.config.vocab_size = model.speech_embedding.num_embeddings
     model.llm.model.config.tie_word_embeddings = False
-    model.llm.model.config.use_bias = True
+    model.llm.model.config.use_bias = use_bias
     model.llm.model.save_pretrained(model_path)
-    os.system('sed -i s@Qwen2ForCausalLM@CosyVoice2ForCausalLM@g {}/config.json'.format(os.path.abspath(model_path)))
+    if use_bias is True:
+        os.system('sed -i s@Qwen2ForCausalLM@CosyVoice2ForCausalLM@g {}/config.json'.format(os.path.abspath(model_path)))
     model.llm.model.config.vocab_size = tmp_vocab_size
     model.llm.model.config.tie_word_embeddings = tmp_tie_embedding
     model.llm.model.set_input_embeddings(embed_tokens)

example.py

@@ -0,0 +1,106 @@
+import sys
+sys.path.append('third_party/Matcha-TTS')
+from cosyvoice.cli.cosyvoice import AutoModel
+import torchaudio
+
+
+def cosyvoice_example():
+    """ CosyVoice Usage, check https://fun-audio-llm.github.io/ for more details
+    """
+    cosyvoice = AutoModel(model_dir='pretrained_models/CosyVoice-300M-SFT')
+    # sft usage
+    print(cosyvoice.list_available_spks())
+    # change stream=True for chunk stream inference
+    for i, j in enumerate(cosyvoice.inference_sft('你好，我是通义生成式语音大模型，请问有什么可以帮您的吗？', '中文女', stream=False)):
+        torchaudio.save('sft_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+    cosyvoice = AutoModel(model_dir='pretrained_models/CosyVoice-300M')
+    # zero_shot usage
+    for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '希望你以后能够做的比我还好呦。', './asset/zero_shot_prompt.wav')):
+        torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+    # cross_lingual usage, <|zh|><|en|><|jp|><|yue|><|ko|> for Chinese/English/Japanese/Cantonese/Korean
+    for i, j in enumerate(cosyvoice.inference_cross_lingual('<|en|>And then later on, fully acquiring that company. So keeping management in line, interest in line with the asset that\'s coming into the family is a reason why sometimes we don\'t buy the whole thing.',
+                                                            './asset/cross_lingual_prompt.wav')):
+        torchaudio.save('cross_lingual_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+    # vc usage
+    for i, j in enumerate(cosyvoice.inference_vc('./asset/cross_lingual_prompt.wav', './asset/zero_shot_prompt.wav')):
+        torchaudio.save('vc_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+    cosyvoice = AutoModel(model_dir='pretrained_models/CosyVoice-300M-Instruct')
+    # instruct usage, support <laughter></laughter><strong></strong>[laughter][breath]
+    for i, j in enumerate(cosyvoice.inference_instruct('在面对挑战时，他展现了非凡的<strong>勇气</strong>与<strong>智慧</strong>。', '中文男',
+                                                       'Theo \'Crimson\', is a fiery, passionate rebel leader. Fights with fervor for justice, but struggles with impulsiveness.<|endofprompt|>')):
+        torchaudio.save('instruct_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+
+def cosyvoice2_example():
+    """ CosyVoice2 Usage, check https://funaudiollm.github.io/cosyvoice2/ for more details
+    """
+    cosyvoice = AutoModel(model_dir='pretrained_models/CosyVoice2-0.5B')
+
+    # NOTE if you want to reproduce the results on https://funaudiollm.github.io/cosyvoice2, please add text_frontend=False during inference
+    # zero_shot usage
+    for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '希望你以后能够做的比我还好呦。', './asset/zero_shot_prompt.wav')):
+        torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+    # save zero_shot spk for future usage
+    assert cosyvoice.add_zero_shot_spk('希望你以后能够做的比我还好呦。', './asset/zero_shot_prompt.wav', 'my_zero_shot_spk') is True
+    for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '', '', zero_shot_spk_id='my_zero_shot_spk')):
+        torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+    cosyvoice.save_spkinfo()
+
+    # fine grained control, for supported control, check cosyvoice/tokenizer/tokenizer.py#L248
+    for i, j in enumerate(cosyvoice.inference_cross_lingual('在他讲述那个荒诞故事的过程中，他突然[laughter]停下来，因为他自己也被逗笑了[laughter]。', './asset/zero_shot_prompt.wav')):
+        torchaudio.save('fine_grained_control_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+    # instruct usage
+    for i, j in enumerate(cosyvoice.inference_instruct2('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '用四川话说这句话<|endofprompt|>', './asset/zero_shot_prompt.wav')):
+        torchaudio.save('instruct_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+    # bistream usage, you can use generator as input, this is useful when using text llm model as input
+    # NOTE you should still have some basic sentence split logic because llm can not handle arbitrary sentence length
+    def text_generator():
+        yield '收到好友从远方寄来的生日礼物，'
+        yield '那份意外的惊喜与深深的祝福'
+        yield '让我心中充满了甜蜜的快乐，'
+        yield '笑容如花儿般绽放。'
+    for i, j in enumerate(cosyvoice.inference_zero_shot(text_generator(), '希望你以后能够做的比我还好呦。', './asset/zero_shot_prompt.wav', stream=False)):
+        torchaudio.save('zero_shot_bistream_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+
+def cosyvoice3_example():
+    """ CosyVoice3 Usage, check https://funaudiollm.github.io/cosyvoice3/ for more details
+    """
+    cosyvoice = AutoModel(model_dir='pretrained_models/Fun-CosyVoice3-0.5B')
+    # zero_shot usage
+    for i, j in enumerate(cosyvoice.inference_zero_shot('八百标兵奔北坡，北坡炮兵并排跑，炮兵怕把标兵碰，标兵怕碰炮兵炮。', 'You are a helpful assistant.<|endofprompt|>希望你以后能够做的比我还好呦。',
+                                                        './asset/zero_shot_prompt.wav', stream=False)):
+        torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+    # fine grained control, for supported control, check cosyvoice/tokenizer/tokenizer.py#L280
+    for i, j in enumerate(cosyvoice.inference_cross_lingual('You are a helpful assistant.<|endofprompt|>[breath]因为他们那一辈人[breath]在乡里面住的要习惯一点，[breath]邻居都很活络，[breath]嗯，都很熟悉。[breath]',
+                                                            './asset/zero_shot_prompt.wav', stream=False)):
+        torchaudio.save('fine_grained_control_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+    # instruct usage, for supported control, check cosyvoice/utils/common.py#L28
+    for i, j in enumerate(cosyvoice.inference_instruct2('好少咯，一般系放嗰啲国庆啊，中秋嗰啲可能会咯。', 'You are a helpful assistant. 请用广东话表达。<|endofprompt|>',
+                                                        './asset/zero_shot_prompt.wav', stream=False)):
+        torchaudio.save('instruct_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+    for i, j in enumerate(cosyvoice.inference_instruct2('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', 'You are a helpful assistant. 请用尽可能快地语速说一句话。<|endofprompt|>',
+                                                        './asset/zero_shot_prompt.wav', stream=False)):
+        torchaudio.save('instruct_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+    # hotfix usage
+    for i, j in enumerate(cosyvoice.inference_zero_shot('高管也通过电话、短信、微信等方式对报道[j][ǐ]予好评。', 'You are a helpful assistant.<|endofprompt|>希望你以后能够做的比我还好呦。',
+                                                        './asset/zero_shot_prompt.wav', stream=False)):
+        torchaudio.save('hotfix_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
+
+
+def main():
+    # cosyvoice_example()
+    # cosyvoice2_example()
+    cosyvoice3_example()
+
+
+if __name__ == '__main__':
+    main()

examples/libritts/cosyvoice/local/prepare_data.py

@@ -40,6 +40,11 @@ def main():
     with open('{}/spk2utt'.format(args.des_dir), 'w') as f:
         for k, v in spk2utt.items():
             f.write('{} {}\n'.format(k, ' '.join(v)))
+    if args.instruct is True:
+        with open('{}/instruct'.format(args.des_dir), 'w') as f:
+            for k, v in utt2text.items():
+                # NOTE in CosyVoice3, we add instruct in sequence
+                f.write('{} You are a helpful assistant.<|endofprompt|>\n'.format(k, v))
     return
 
 
@@ -49,7 +54,9 @@ if __name__ == "__main__":
                         type=str)
     parser.add_argument('--des_dir',
                         type=str)
-    parser.add_argument('--ref_model',
-                        type=str)
+    parser.add_argument('--instruct',
+                        action='store_true',
+                        default=False,
+                        help='create instruct file or not')
     args = parser.parse_args()
     main()

examples/libritts/cosyvoice3/conf/cosyvoice3.yaml

@@ -0,0 +1,234 @@
+# set random seed, so that you may reproduce your result.
+__set_seed1: !apply:random.seed [1986]
+__set_seed2: !apply:numpy.random.seed [1986]
+__set_seed3: !apply:torch.manual_seed [1986]
+__set_seed4: !apply:torch.cuda.manual_seed_all [1986]
+
+# fixed params
+sample_rate: 24000
+llm_input_size: 896
+llm_output_size: 896
+spk_embed_dim: 192
+qwen_pretrain_path: ''
+token_frame_rate: 25
+token_mel_ratio: 2
+
+# stream related params
+chunk_size: 25 # streaming inference chunk size, in token
+num_decoding_left_chunks: -1 # streaming inference flow decoder left chunk size, <0 means use all left chunks
+
+# model params
+# for all class/function included in this repo, we use !<name> or !<new> for intialization, so that user may find all corresponding class/function according to one single yaml.
+# for system/third_party class/function, we do not require this.
+llm: !new:cosyvoice.llm.llm.Qwen2LM
+    llm_input_size: !ref <llm_input_size>
+    llm_output_size: !ref <llm_output_size>
+    speech_token_size: 6561
+    length_normalized_loss: True
+    lsm_weight: 0
+    mix_ratio: [5, 15]
+    llm: !new:cosyvoice.llm.llm.Qwen2Encoder
+        pretrain_path: !ref <qwen_pretrain_path>
+    sampling: !name:cosyvoice.utils.common.ras_sampling
+        top_p: 0.8
+        top_k: 25
+        win_size: 10
+        tau_r: 0.1
+
+flow: !new:cosyvoice.flow.flow.CausalMaskedDiffWithXvec
+    input_size: 512
+    output_size: 80
+    spk_embed_dim: !ref <spk_embed_dim>
+    output_type: 'mel'
+    vocab_size: 6561
+    input_frame_rate: !ref <token_frame_rate>
+    only_mask_loss: True
+    token_mel_ratio: !ref <token_mel_ratio>
+    pre_lookahead_len: 3
+    encoder: !new:cosyvoice.transformer.upsample_encoder.UpsampleConformerEncoder
+        output_size: 512
+        attention_heads: 8
+        linear_units: 2048
+        num_blocks: 6
+        dropout_rate: 0.1
+        positional_dropout_rate: 0.1
+        attention_dropout_rate: 0.1
+        normalize_before: True
+        input_layer: 'linear'
+        pos_enc_layer_type: 'rel_pos_espnet'
+        selfattention_layer_type: 'rel_selfattn'
+        input_size: 512
+        use_cnn_module: False
+        macaron_style: False
+        static_chunk_size: !ref <chunk_size>
+    decoder: !new:cosyvoice.flow.flow_matching.CausalConditionalCFM
+        in_channels: 240
+        n_spks: 1
+        spk_emb_dim: 80
+        cfm_params: !new:omegaconf.DictConfig
+            content:
+                sigma_min: 1e-06
+                solver: 'euler'
+                t_scheduler: 'cosine'
+                training_cfg_rate: 0.2
+                inference_cfg_rate: 0.7
+                reg_loss_type: 'l1'
+        estimator: !new:cosyvoice.flow.decoder.CausalConditionalDecoder
+            in_channels: 320
+            out_channels: 80
+            channels: [256]
+            dropout: 0.0
+            attention_head_dim: 64
+            n_blocks: 4
+            num_mid_blocks: 12
+            num_heads: 8
+            act_fn: 'gelu'
+            static_chunk_size: !ref <chunk_size> * <token_mel_ratio>
+            num_decoding_left_chunks: !ref <num_decoding_left_chunks>
+
+hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
+    in_channels: 80
+    base_channels: 512
+    nb_harmonics: 8
+    sampling_rate: !ref <sample_rate>
+    nsf_alpha: 0.1
+    nsf_sigma: 0.003
+    nsf_voiced_threshold: 10
+    upsample_rates: [8, 5, 3]
+    upsample_kernel_sizes: [16, 11, 7]
+    istft_params:
+        n_fft: 16
+        hop_len: 4
+    resblock_kernel_sizes: [3, 7, 11]
+    resblock_dilation_sizes: [[1, 3, 5], [1, 3, 5], [1, 3, 5]]
+    source_resblock_kernel_sizes: [7, 7, 11]
+    source_resblock_dilation_sizes: [[1, 3, 5], [1, 3, 5], [1, 3, 5]]
+    lrelu_slope: 0.1
+    audio_limit: 0.99
+    f0_predictor: !new:cosyvoice.hifigan.f0_predictor.ConvRNNF0Predictor
+        num_class: 1
+        in_channels: 80
+        cond_channels: 512
+
+# gan related module
+mel_spec_transform1: !name:matcha.utils.audio.mel_spectrogram
+    n_fft: 1920
+    num_mels: 80
+    sampling_rate: !ref <sample_rate>
+    hop_size: 480
+    win_size: 1920
+    fmin: 0
+    fmax: null
+    center: False
+hifigan: !new:cosyvoice.hifigan.hifigan.HiFiGan
+    generator: !ref <hift>
+    discriminator: !new:cosyvoice.hifigan.discriminator.MultipleDiscriminator
+        mpd: !new:matcha.hifigan.models.MultiPeriodDiscriminator
+        mrd: !new:cosyvoice.hifigan.discriminator.MultiResSpecDiscriminator
+    mel_spec_transform: [
+        !ref <mel_spec_transform1>
+    ]
+
+# processor functions
+parquet_opener: !name:cosyvoice.dataset.processor.parquet_opener
+get_tokenizer: !name:cosyvoice.tokenizer.tokenizer.get_qwen_tokenizer
+    token_path: !ref <qwen_pretrain_path>
+    skip_special_tokens: True
+allowed_special: 'all'
+tokenize: !name:cosyvoice.dataset.processor.tokenize
+    get_tokenizer: !ref <get_tokenizer>
+    allowed_special: !ref <allowed_special>
+filter: !name:cosyvoice.dataset.processor.filter
+    max_length: 40960
+    min_length: 100
+    token_max_length: 200
+    token_min_length: 1
+resample: !name:cosyvoice.dataset.processor.resample
+    resample_rate: !ref <sample_rate>
+truncate: !name:cosyvoice.dataset.processor.truncate
+    truncate_length: 24480 # must be a multiplier of hop_size
+feat_extractor: !name:matcha.utils.audio.mel_spectrogram
+    n_fft: 1920
+    num_mels: 80
+    sampling_rate: !ref <sample_rate>
+    hop_size: 480
+    win_size: 1920
+    fmin: 0
+    fmax: 8000
+    center: False
+compute_fbank: !name:cosyvoice.dataset.processor.compute_fbank
+    feat_extractor: !ref <feat_extractor>
+    token_mel_ratio: 2
+compute_f0: !name:cosyvoice.dataset.processor.compute_f0
+    sample_rate: !ref <sample_rate>
+    hop_size: 480
+parse_embedding: !name:cosyvoice.dataset.processor.parse_embedding
+    normalize: True
+shuffle: !name:cosyvoice.dataset.processor.shuffle
+    shuffle_size: 1000
+sort: !name:cosyvoice.dataset.processor.sort
+    sort_size: 500  # sort_size should be less than shuffle_size
+batch: !name:cosyvoice.dataset.processor.batch
+    batch_type: 'dynamic'
+    max_frames_in_batch: 2000
+padding: !name:cosyvoice.dataset.processor.padding
+    use_spk_embedding: False # change to True during sft
+
+
+# dataset processor pipeline
+data_pipeline: [
+    !ref <parquet_opener>,
+    !ref <tokenize>,
+    !ref <filter>,
+    !ref <resample>,
+    !ref <compute_fbank>,
+    !ref <parse_embedding>,
+    !ref <shuffle>,
+    !ref <sort>,
+    !ref <batch>,
+    !ref <padding>,
+]
+data_pipeline_gan: [
+    !ref <parquet_opener>,
+    !ref <tokenize>,
+    !ref <filter>,
+    !ref <resample>,
+    !ref <truncate>,
+    !ref <compute_fbank>,
+    !ref <compute_f0>,
+    !ref <parse_embedding>,
+    !ref <shuffle>,
+    !ref <sort>,
+    !ref <batch>,
+    !ref <padding>,
+]
+
+# llm flow train conf
+train_conf:
+    optim: adam
+    optim_conf:
+        lr: 1e-5 # change to 1e-5 during sft
+    scheduler: constantlr # change to constantlr during sft
+    scheduler_conf:
+        warmup_steps: 2500
+    max_epoch: 200
+    grad_clip: 5
+    accum_grad: 2
+    log_interval: 100
+    save_per_step: -1
+
+# gan train conf
+train_conf_gan:
+    optim: adam
+    optim_conf:
+        lr: 0.0002 # use small lr for gan training
+    scheduler: constantlr
+    optim_d: adam
+    optim_conf_d:
+        lr: 0.0002 # use small lr for gan training
+    scheduler_d: constantlr
+    max_epoch: 200
+    grad_clip: 5
+    accum_grad: 1 # in gan training, accum_grad must be 1
+    log_interval: 100
+    save_per_step: -1

examples/libritts/cosyvoice3/conf/ds_stage2.json

@@ -0,0 +1,42 @@
+{
+  "train_micro_batch_size_per_gpu": 1,
+  "gradient_accumulation_steps": 1,
+  "steps_per_print": 100,
+  "gradient_clipping": 5,
+  "fp16": {
+    "enabled": false,
+    "auto_cast": false,
+    "loss_scale": 0,
+    "initial_scale_power": 16,
+    "loss_scale_window": 256,
+    "hysteresis": 2,
+    "consecutive_hysteresis": false,
+    "min_loss_scale": 1
+  },
+  "bf16": {
+    "enabled": false
+  },
+  "zero_force_ds_cpu_optimizer": false,
+  "zero_optimization": {
+    "stage": 2,
+    "offload_optimizer": {
+      "device": "none",
+      "pin_memory": true
+    },
+    "allgather_partitions": true,
+    "allgather_bucket_size": 5e8,
+    "overlap_comm": false,
+    "reduce_scatter": true,
+    "reduce_bucket_size": 5e8,
+    "contiguous_gradients" : true
+  },
+  "optimizer": {
+    "type": "AdamW",
+    "params": {
+        "lr": 0.001,
+        "weight_decay": 0.0001,
+        "torch_adam": true,
+        "adam_w_mode": true
+    }
+  }
+}

examples/libritts/cosyvoice3/cosyvoice

@@ -0,0 +1 @@
+../../../cosyvoice

examples/libritts/cosyvoice3/local

@@ -0,0 +1 @@
+../cosyvoice/local

examples/libritts/cosyvoice3/path.sh

@@ -0,0 +1 @@
+../cosyvoice/path.sh

examples/libritts/cosyvoice3/run.sh

@@ -0,0 +1,112 @@
+#!/bin/bash
+# Copyright 2024 Alibaba Inc. All Rights Reserved.
+. ./path.sh || exit 1;
+
+stage=-1
+stop_stage=3
+
+data_url=www.openslr.org/resources/60
+data_dir=/mnt/lyuxiang.lx/data/tts/openslr/libritts
+pretrained_model_dir=../../../pretrained_models/CosyVoice3-0.5B
+
+if [ ${stage} -le -1 ] && [ ${stop_stage} -ge -1 ]; then
+  echo "Data Download"
+  for part in dev-clean test-clean dev-other test-other train-clean-100 train-clean-360 train-other-500; do
+    local/download_and_untar.sh ${data_dir} ${data_url} ${part}
+  done
+fi
+
+if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
+  echo "Data preparation, prepare wav.scp/text/utt2spk/spk2utt"
+  for x in train-clean-100 train-clean-360 train-other-500 dev-clean dev-other test-clean test-other; do
+    mkdir -p data/$x
+    python local/prepare_data.py --src_dir $data_dir/LibriTTS/$x --des_dir data/$x --instruct
+  done
+fi
+
+if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
+  echo "Extract campplus speaker embedding, you will get spk2embedding.pt and utt2embedding.pt in data/$x dir"
+  for x in train-clean-100 train-clean-360 train-other-500 dev-clean dev-other test-clean test-other; do
+    tools/extract_embedding.py --dir data/$x \
+      --onnx_path $pretrained_model_dir/campplus.onnx
+  done
+fi
+
+if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
+  echo "Extract discrete speech token, you will get utt2speech_token.pt in data/$x dir"
+  for x in train-clean-100 train-clean-360 train-other-500 dev-clean dev-other test-clean test-other; do
+    tools/extract_speech_token.py --dir data/$x \
+      --onnx_path $pretrained_model_dir/speech_tokenizer_v3.onnx
+  done
+fi
+
+if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
+  echo "Prepare required parquet format data, you should have prepared wav.scp/text/utt2spk/spk2utt/utt2embedding.pt/spk2embedding.pt/utt2speech_token.pt"
+  for x in train-clean-100 train-clean-360 train-other-500 dev-clean dev-other test-clean test-other; do
+    mkdir -p data/$x/parquet
+    tools/make_parquet_list.py --num_utts_per_parquet 1000 \
+      --num_processes 10 \
+      --instruct \
+      --src_dir data/$x \
+      --des_dir data/$x/parquet
+  done
+fi
+
+# train llm
+export CUDA_VISIBLE_DEVICES="0,1,2,3"
+num_gpus=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
+job_id=1986
+dist_backend="nccl"
+num_workers=2
+prefetch=100
+train_engine=torch_ddp
+if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
+  echo "Run train. We only support llm traning for now"
+  if [ $train_engine == 'deepspeed' ]; then
+    echo "Notice deepspeed has its own optimizer config. Modify conf/ds_stage2.json if necessary"
+  fi
+  cat data/{train-clean-100,train-clean-360,train-other-500}/parquet/data.list > data/train.data.list
+  cat data/{dev-clean,dev-other}/parquet/data.list > data/dev.data.list
+  # NOTE will update llm/hift training later
+  for model in llm flow hifigan; do
+    torchrun --nnodes=1 --nproc_per_node=$num_gpus \
+        --rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:1234" \
+      cosyvoice/bin/train.py \
+      --train_engine $train_engine \
+      --config conf/cosyvoice3.yaml \
+      --train_data data/train.data.list \
+      --cv_data data/dev.data.list \
+      --qwen_pretrain_path $pretrained_model_dir/CosyVoice-BlankEN \
+      --model $model \
+      --checkpoint $pretrained_model_dir/$model.pt \
+      --model_dir `pwd`/exp/cosyvoice3/$model/$train_engine \
+      --tensorboard_dir `pwd`/tensorboard/cosyvoice3/$model/$train_engine \
+      --ddp.dist_backend $dist_backend \
+      --num_workers ${num_workers} \
+      --prefetch ${prefetch} \
+      --pin_memory \
+      --use_amp \
+      --deepspeed_config ./conf/ds_stage2.json \
+      --deepspeed.save_states model+optimizer
+  done
+fi
+
+# average model
+average_num=5
+if [ ${stage} -le 6 ] && [ ${stop_stage} -ge 6 ]; then
+  for model in llm flow hifigan; do
+    decode_checkpoint=`pwd`/exp/cosyvoice/$model/$train_engine/${model}.pt
+    echo "do model average and final checkpoint is $decode_checkpoint"
+    python cosyvoice/bin/average_model.py \
+      --dst_model $decode_checkpoint \
+      --src_path `pwd`/exp/cosyvoice/$model/$train_engine  \
+      --num ${average_num} \
+      --val_best
+  done
+fi
+
+if [ ${stage} -le 7 ] && [ ${stop_stage} -ge 7 ]; then
+  echo "Export your model for inference speedup. Remember copy your llm or flow model to model_dir"
+  python cosyvoice/bin/export_jit.py --model_dir $pretrained_model_dir
+  python cosyvoice/bin/export_onnx.py --model_dir $pretrained_model_dir
+fi

examples/libritts/cosyvoice3/tools

@@ -0,0 +1 @@
+../../../tools

requirements.txt

@@ -29,9 +29,9 @@ pyworld==0.3.4
 rich==13.7.1
 soundfile==0.12.1
 tensorboard==2.14.0
-tensorrt-cu12==10.0.1; sys_platform == 'linux'
-tensorrt-cu12-bindings==10.0.1; sys_platform == 'linux'
-tensorrt-cu12-libs==10.0.1; sys_platform == 'linux'
+tensorrt-cu12==10.13.3.9; sys_platform == 'linux'
+tensorrt-cu12-bindings==10.13.3.9; sys_platform == 'linux'
+tensorrt-cu12-libs==10.13.3.9; sys_platform == 'linux'
 torch==2.3.1
 torchaudio==2.3.1
 transformers==4.51.3

tools/make_parquet_list.py

@@ -37,6 +37,8 @@ def job(utt_list, parquet_file, utt2parquet_file, spk2parquet_file):
     speech_token_list = [utt2speech_token.get(utt, []) for utt in utt_list]
     if args.dpo:
         reject_speech_token_list = [utt2reject_speech_token[utt] for utt in utt_list]
+    if args.instruct:
+        instruct_list = [utt2instruct[utt] for utt in utt_list]
 
     # 保存到parquet,utt2parquet_file,spk2parquet_file
     df = pd.DataFrame()
@@ -50,6 +52,8 @@ def job(utt_list, parquet_file, utt2parquet_file, spk2parquet_file):
     df['speech_token'] = speech_token_list
     if args.dpo:
         df['reject_speech_token'] = reject_speech_token_list
+    if args.instruct:
+        df['instruct'] = instruct_list
     df.to_parquet(parquet_file)
     with open(utt2parquet_file, 'w') as f:
         json.dump({k: parquet_file for k in utt_list}, f, ensure_ascii=False, indent=2)
@@ -68,6 +72,10 @@ if __name__ == "__main__":
                         type=int,
                         default=1,
                         help='num processes for make parquets')
+    parser.add_argument('--instruct',
+                        action='store_true',
+                        default=False,
+                        help='has instruct file or not')
     parser.add_argument('--src_dir',
                         type=str)
     parser.add_argument('--des_dir',
@@ -91,6 +99,11 @@ if __name__ == "__main__":
         for l in f:
             l = l.replace('\n', '').split()
             utt2spk[l[0]] = l[1]
+    if args.instruct is True:
+        with open('{}/instruct'.format(args.src_dir)) as f:
+            for l in f:
+                l = l.replace('\n', '').split()
+                utt2instruct[l[0]] = ' '.join(l[1:])
     utt2embedding = torch.load('{}/utt2embedding.pt'.format(args.src_dir))
     spk2embedding = torch.load('{}/spk2embedding.pt'.format(args.src_dir))
     utt2speech_token = torch.load('{}/utt2speech_token.pt'.format(args.src_dir))

vllm_example.py

@@ -4,20 +4,36 @@ from vllm import ModelRegistry
 from cosyvoice.vllm.cosyvoice2 import CosyVoice2ForCausalLM
 ModelRegistry.register_model("CosyVoice2ForCausalLM", CosyVoice2ForCausalLM)
 
-from cosyvoice.cli.cosyvoice import CosyVoice2
-from cosyvoice.utils.file_utils import load_wav
+from cosyvoice.cli.cosyvoice import AutoModel
 from cosyvoice.utils.common import set_all_random_seed
 from tqdm import tqdm
 
 
-def main():
-    cosyvoice = CosyVoice2('pretrained_models/CosyVoice2-0.5B', load_jit=True, load_trt=True, load_vllm=True, fp16=True)
-    prompt_speech_16k = load_wav('./asset/zero_shot_prompt.wav', 16000)
+def cosyvoice2_example():
+    """ CosyVoice2 vllm usage
+    """
+    cosyvoice = AutoModel(model_dir='pretrained_models/CosyVoice2-0.5B', load_jit=True, load_trt=True, load_vllm=True, fp16=True)
     for i in tqdm(range(100)):
         set_all_random_seed(i)
-        for _, _ in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '希望你以后能够做的比我还好呦。', prompt_speech_16k, stream=False)):
+        for _, _ in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', '希望你以后能够做的比我还好呦。', './asset/zero_shot_prompt.wav', stream=False)):
             continue
 
 
+def cosyvoice3_example():
+    """ CosyVoice3 vllm usage
+    """
+    cosyvoice = AutoModel(model_dir='pretrained_models/Fun-CosyVoice3-0.5B', load_trt=True, load_vllm=True, fp16=False)
+    for i in tqdm(range(100)):
+        set_all_random_seed(i)
+        for _, _ in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物，那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐，笑容如花儿般绽放。', 'You are a helpful assistant.<|endofprompt|>希望你以后能够做的比我还好呦。',
+                                                            './asset/zero_shot_prompt.wav', stream=False)):
+            continue
+
+
+def main():
+    # cosyvoice2_example()
+    cosyvoice3_example()
+
+
 if __name__ == '__main__':
     main()

webui.py

@@ -22,8 +22,8 @@ import random
 import librosa
 ROOT_DIR = os.path.dirname(os.path.abspath(__file__))
 sys.path.append('{}/third_party/Matcha-TTS'.format(ROOT_DIR))
-from cosyvoice.cli.cosyvoice import CosyVoice, CosyVoice2
-from cosyvoice.utils.file_utils import load_wav, logging
+from cosyvoice.cli.cosyvoice import AutoModel
+from cosyvoice.utils.file_utils import logging
 from cosyvoice.utils.common import set_all_random_seed
 
 inference_mode_list = ['预训练音色', '3s极速复刻', '跨语种复刻', '自然语言控制']
@@ -43,18 +43,6 @@ def generate_seed():
     }
 
 
-def postprocess(speech, top_db=60, hop_length=220, win_length=440):
-    speech, _ = librosa.effects.trim(
-        speech, top_db=top_db,
-        frame_length=win_length,
-        hop_length=hop_length
-    )
-    if speech.abs().max() > max_val:
-        speech = speech / speech.abs().max() * max_val
-    speech = torch.concat([speech, torch.zeros(1, int(cosyvoice.sample_rate * 0.2))], dim=1)
-    return speech
-
-
 def change_instruction(mode_checkbox_group):
     return instruct_dict[mode_checkbox_group]
 
@@ -118,15 +106,13 @@ def generate_audio(tts_text, mode_checkbox_group, sft_dropdown, prompt_text, pro
             yield (cosyvoice.sample_rate, i['tts_speech'].numpy().flatten())
     elif mode_checkbox_group == '3s极速复刻':
         logging.info('get zero_shot inference request')
-        prompt_speech_16k = postprocess(load_wav(prompt_wav, prompt_sr))
         set_all_random_seed(seed)
-        for i in cosyvoice.inference_zero_shot(tts_text, prompt_text, prompt_speech_16k, stream=stream, speed=speed):
+        for i in cosyvoice.inference_zero_shot(tts_text, prompt_text, prompt_wav, stream=stream, speed=speed):
             yield (cosyvoice.sample_rate, i['tts_speech'].numpy().flatten())
     elif mode_checkbox_group == '跨语种复刻':
         logging.info('get cross_lingual inference request')
-        prompt_speech_16k = postprocess(load_wav(prompt_wav, prompt_sr))
         set_all_random_seed(seed)
-        for i in cosyvoice.inference_cross_lingual(tts_text, prompt_speech_16k, stream=stream, speed=speed):
+        for i in cosyvoice.inference_cross_lingual(tts_text, prompt_wav, stream=stream, speed=speed):
             yield (cosyvoice.sample_rate, i['tts_speech'].numpy().flatten())
     else:
         logging.info('get instruct inference request')
@@ -181,16 +167,10 @@ if __name__ == '__main__':
                         default=8000)
     parser.add_argument('--model_dir',
                         type=str,
-                        default='pretrained_models/CosyVoice2-0.5B',
+                        default='pretrained_models/CosyVoice3-0.5B',
                         help='local path or modelscope repo id')
     args = parser.parse_args()
-    try:
-        cosyvoice = CosyVoice(args.model_dir)
-    except Exception:
-        try:
-            cosyvoice = CosyVoice2(args.model_dir)
-        except Exception:
-            raise TypeError('no valid model_type!')
+    cosyvoice = AutoModel(model_dir=args.model_dir)
 
     sft_spk = cosyvoice.list_available_spks()
     if len(sft_spk) == 0: