Merge pull request #1814 from hexisyztem/main

[BUG FIX] 使用 float64 避免精度误差问题，弃用 CPU 计算，避免拖累性能

.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

@@ -1,50 +1,52 @@
-[![SVG Banners](https://svg-banners.vercel.app/api?type=origin&text1=CosyVoice🤠&text2=Text-to-Speech%20💖%20Large%20Language%20Model&width=800&height=210)](https://github.com/Akshay090/svg-banners)
+![SVG Banners](https://svg-banners.vercel.app/api?type=origin&text1=CosyVoice🤠&text2=Text-to-Speech%20💖%20Large%20Language%20Model&width=800&height=210)
 
 ## 👉🏻 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/pdf/2505.17589); [Modelscope](https://www.modelscope.cn/models/FunAudioLLM/Fun-CosyVoice3-0.5B-2512); [Huggingface](https://huggingface.co/FunAudioLLM/Fun-CosyVoice3-0.5B-2512); [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)
+**CosyVoice 2.0**: [Demos](https://funaudiollm.github.io/cosyvoice2/); [Paper](https://arxiv.org/pdf/2412.10117); [Modelscope](https://www.modelscope.cn/models/iic/CosyVoice2-0.5B); [HuggingFace](https://huggingface.co/FunAudioLLM/CosyVoice2-0.5B)
 
-**CosyVoice 1.0**: [Demos](https://fun-audio-llm.github.io); [Paper](https://funaudiollm.github.io/pdf/CosyVoice_v1.pdf); [Modelscope](https://www.modelscope.cn/studios/iic/CosyVoice-300M)
+**CosyVoice 1.0**: [Demos](https://fun-audio-llm.github.io); [Paper](https://funaudiollm.github.io/pdf/CosyVoice_v1.pdf); [Modelscope](https://www.modelscope.cn/models/iic/CosyVoice-300M); [HuggingFace](https://huggingface.co/FunAudioLLM/CosyVoice-300M)
 
 ## 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, Shandong, 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
 
@@ -57,6 +59,27 @@
     - [x] WeTextProcessing support when ttsfrd is not available
     - [x] Fastapi server and client
 
+## Evaluation
+
+| Model | Open-Source | Model Size | test-zh<br>CER (%) ↓ | test-zh<br>SS (%) ↑ | test-en<br>WER (%) ↓ | test-en<br>SS (%) ↑ | test-hard<br>CER (%) ↓ | test-hard<br>SS (%) ↑ |
+| :--- | :---: | :---: | :---: | :---: | :---: | :---: | :---: | :---: |
+| Human | - | - | 1.26 | 75.5 | 2.14 | 73.4 | - | - |
+| Seed-TTS | ❌ | - | 1.12 | 79.6 | 2.25 | 76.2 | 7.59 | 77.6 |
+| MiniMax-Speech | ❌ | - | 0.83 | 78.3 | 1.65 | 69.2 | - | - |
+| F5-TTS | ✅ | 0.3B | 1.52 | 74.1 | 2.00 | 64.7 | 8.67 | 71.3 |
+| Spark TTS | ✅ | 0.5B | 1.2 | 66.0 | 1.98 | 57.3 | - | - |
+| CosyVoice2 | ✅ | 0.5B | 1.45 | 75.7 | 2.57 | 65.9 | 6.83 | 72.4 |
+| FireRedTTS2 | ✅ | 1.5B | 1.14 | 73.2 | 1.95 | 66.5 | - | - |
+| Index-TTS2 | ✅ | 1.5B | 1.03 | 76.5 | 2.23 | 70.6 | 7.12 | 75.5 |
+| VibeVoice-1.5B | ✅ | 1.5B | 1.16 | 74.4 | 3.04 | 68.9 | - | - |
+| VibeVoice-Realtime | ✅ | 0.5B | - | - | 2.05 | 63.3 | - | - |
+| HiggsAudio-v2 | ✅ | 3B | 1.50 | 74.0 | 2.44 | 67.7 | - | - |
+| VoxCPM | ✅ | 0.5B | 0.93 | 77.2 | 1.85 | 72.9 | 8.87 | 73.0 |
+| GLM-TTS | ✅ | 1.5B | 1.03 | 76.1 | - | - | - | - |
+| GLM-TTS RL | ✅ | 1.5B | 0.89 | 76.4 | - | - | - | - |
+| Fun-CosyVoice3-0.5B-2512 | ✅ | 0.5B | 1.21 | 78.0 | 2.24 | 71.8 | 6.71 | 75.8 |
+| Fun-CosyVoice3-0.5B-2512_RL | ✅ | 0.5B | 0.81 | 77.4 | 1.68 | 69.5 | 5.44 | 75.0 |
+
 
 ## Install
 
@@ -87,26 +110,26 @@
 
 ### 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
+# modelscope SDK model download
 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')
 snapshot_download('iic/CosyVoice-300M-Instruct', local_dir='pretrained_models/CosyVoice-300M-Instruct')
 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
+# for oversea users, huggingface SDK model download
+from huggingface_hub import snapshot_download
+snapshot_download('FunAudioLLM/Fun-CosyVoice3-0.5B-2512', local_dir='pretrained_models/Fun-CosyVoice3-0.5B')
+snapshot_download('FunAudioLLM/CosyVoice2-0.5B', local_dir='pretrained_models/CosyVoice2-0.5B')
+snapshot_download('FunAudioLLM/CosyVoice-300M', local_dir='pretrained_models/CosyVoice-300M')
+snapshot_download('FunAudioLLM/CosyVoice-300M-SFT', local_dir='pretrained_models/CosyVoice-300M-SFT')
+snapshot_download('FunAudioLLM/CosyVoice-300M-Instruct', local_dir='pretrained_models/CosyVoice-300M-Instruct')
+snapshot_download('FunAudioLLM/CosyVoice-ttsfrd', local_dir='pretrained_models/CosyVoice-ttsfrd')
 ```
 
 Optionally, you can unzip `ttsfrd` resource and install `ttsfrd` package for better text normalization performance.
@@ -122,94 +145,28 @@ 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
+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 Usage
-```python
-cosyvoice = CosyVoice2('pretrained_models/CosyVoice2-0.5B', load_jit=False, load_trt=False, load_vllm=False, fp16=False)
+#### vLLM Usage
+CosyVoice2/3 now supports **vLLM 0.11.x+ (V1 engine)** and **vLLM 0.9.0 (legacy)**.
+Older vllm version(<0.9.0) do not support CosyVoice inference, and versions in between (e.g., 0.10.x) are not tested.
 
-# 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)
-```
-
-#### CosyVoice2 vllm Usage
-If you want to use vllm for inference, please install `vllm==v0.9.0`. Older vllm version do not support CosyVoice2 inference.
-
-Notice that `vllm==v0.9.0` has a lot of specific requirements, for example `torch==2.7.0`. You can create a new env to in case your hardward do not support vllm and old env is corrupted.
+Notice that `vllm` has a lot of specific requirements. You can create a new env to in case your hardward do not support vllm and old env is corrupted.
 
 ``` sh
 conda create -n cosyvoice_vllm --clone cosyvoice
 conda activate cosyvoice_vllm
-pip install vllm==v0.9.0 -i https://mirrors.aliyun.com/pypi/simple/ --trusted-host=mirrors.aliyun.com
+# for vllm==0.9.0
+pip install vllm==v0.9.0 transformers==4.51.3 numpy==1.26.4 -i https://mirrors.aliyun.com/pypi/simple/ --trusted-host=mirrors.aliyun.com
+# for vllm>=0.11.0
+pip install vllm==v0.11.0 transformers==4.57.1 numpy==1.26.4 -i https://mirrors.aliyun.com/pypi/simple/ --trusted-host=mirrors.aliyun.com
 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.
@@ -223,7 +180,7 @@ python3 webui.py --port 50000 --model_dir pretrained_models/CosyVoice-300M
 
 #### Advanced Usage
 
-For advanced users, we have provided training and inference scripts in `examples/libritts/cosyvoice/run.sh`.
+For advanced users, we have provided training and inference scripts in `examples/libritts`.
 
 #### Build for deployment
 
@@ -242,6 +199,17 @@ docker run -d --runtime=nvidia -p 50000:50000 cosyvoice:v1.0 /bin/bash -c "cd /o
 cd fastapi && python3 client.py --port 50000 --mode <sft|zero_shot|cross_lingual|instruct>
 ```
 
+#### Using Nvidia TensorRT-LLM for deployment
+
+Using TensorRT-LLM to accelerate cosyvoice2 llm could give 4x acceleration comparing with huggingface transformers implementation.
+To quick start:
+
+``` sh
+cd runtime/triton_trtllm
+docker compose up -d
+```
+For more details, you could check [here](https://github.com/FunAudioLLM/CosyVoice/tree/main/runtime/triton_trtllm)
+
 ## Discussion & Communication
 
 You can directly discuss on [Github Issues](https://github.com/FunAudioLLM/CosyVoice/issues).

cosyvoice/bin/export_jit.py

@@ -23,7 +23,7 @@ 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.utils.file_utils import logging
 
 
@@ -57,15 +57,9 @@ 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 model.__class__.__name__ == 'CosyVoice':
         # 1. export llm text_encoder
         llm_text_encoder = model.model.llm.text_encoder
         script = get_optimized_script(llm_text_encoder)
@@ -89,14 +83,16 @@ def main():
         script = get_optimized_script(flow_encoder.half())
         script.save('{}/flow.encoder.fp16.zip'.format(args.model_dir))
         logging.info('successfully export flow_encoder')
-    else:
-        # 3. export flow encoder
+    elif model.__class__.__name__ == 'CosyVoice2':
+        # 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')
+    else:
+        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/bin/train.py

@@ -49,6 +49,7 @@ def get_args():
     parser.add_argument('--train_data', required=True, help='train data file')
     parser.add_argument('--cv_data', required=True, help='cv data file')
     parser.add_argument('--qwen_pretrain_path', required=False, help='qwen pretrain path')
+    parser.add_argument('--onnx_path', required=False, help='onnx path, which is required for online feature extraction')
     parser.add_argument('--checkpoint', help='checkpoint model')
     parser.add_argument('--model_dir', required=True, help='save model dir')
     parser.add_argument('--tensorboard_dir',
@@ -96,6 +97,7 @@ def get_args():
 @record
 def main():
     args = get_args()
+    os.environ['onnx_path'] = args.onnx_path
     logging.basicConfig(level=logging.DEBUG,
                         format='%(asctime)s %(levelname)s %(message)s')
     # gan train has some special initialization logic
@@ -104,12 +106,10 @@ def main():
     override_dict = {k: None for k in ['llm', 'flow', 'hift', 'hifigan'] if k != args.model}
     if gan is True:
         override_dict.pop('hift')
-    try:
-        with open(args.config, 'r') as f:
-            configs = load_hyperpyyaml(f, overrides={**override_dict, 'qwen_pretrain_path': args.qwen_pretrain_path})
-    except Exception:
-        with open(args.config, 'r') as f:
-            configs = load_hyperpyyaml(f, overrides=override_dict)
+    if args.qwen_pretrain_path is not None:
+        override_dict['qwen_pretrain_path'] = args.qwen_pretrain_path
+    with open(args.config, 'r') as f:
+        configs = load_hyperpyyaml(f, overrides=override_dict)
     if gan is True:
         configs['train_conf'] = configs['train_conf_gan']
     configs['train_conf'].update(vars(args))

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 +88,14 @@ 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):
+        if self.__class__.__name__ == 'CosyVoice3' and '<|endofprompt|>' not in prompt_text + tts_text:
+            logging.warning('<|endofprompt|> not found in CosyVoice3 inference, check your input text')
         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 +104,9 @@ 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):
@@ -115,9 +116,7 @@ class CosyVoice:
                 start_time = time.time()
 
     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))
+        assert self.__class__.__name__ == 'CosyVoice', 'inference_instruct is only implemented for CosyVoice!'
         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 +128,8 @@ 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 +141,6 @@ 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 +158,9 @@ 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 +176,9 @@ 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 +186,55 @@ 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

@@ -20,19 +20,10 @@ import numpy as np
 import whisper
 from typing import Callable
 import torchaudio.compliance.kaldi as kaldi
-import torchaudio
 import os
 import re
 import inflect
-try:
-    import ttsfrd
-    use_ttsfrd = True
-except ImportError:
-    print("failed to import ttsfrd, use wetext instead")
-    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
 
 
@@ -56,21 +47,33 @@ class CosyVoiceFrontEnd:
                                                                      providers=["CUDAExecutionProvider" if torch.cuda.is_available() else
                                                                                 "CPUExecutionProvider"])
         if os.path.exists(spk2info):
-            self.spk2info = torch.load(spk2info, map_location=self.device)
+            self.spk2info = torch.load(spk2info, map_location=self.device, weights_only=True)
         else:
             self.spk2info = {}
         self.allowed_special = allowed_special
-        self.use_ttsfrd = use_ttsfrd
-        if self.use_ttsfrd:
+        self.inflect_parser = inflect.engine()
+        # NOTE compatible when no text frontend tool is avaliable
+        try:
+            import ttsfrd
             self.frd = ttsfrd.TtsFrontendEngine()
             ROOT_DIR = os.path.dirname(os.path.abspath(__file__))
             assert self.frd.initialize('{}/../../pretrained_models/CosyVoice-ttsfrd/resource'.format(ROOT_DIR)) is True, \
                 'failed to initialize ttsfrd resource'
             self.frd.set_lang_type('pinyinvg')
-        else:
-            self.zh_tn_model = ZhNormalizer(remove_erhua=False)
-            self.en_tn_model = EnNormalizer()
-            self.inflect_parser = inflect.engine()
+            self.text_frontend = 'ttsfrd'
+            logging.info('use ttsfrd frontend')
+        except:
+            try:
+                from wetext import Normalizer as ZhNormalizer
+                from wetext import Normalizer as EnNormalizer
+                self.zh_tn_model = ZhNormalizer(remove_erhua=False)
+                self.en_tn_model = EnNormalizer()
+                self.text_frontend = 'wetext'
+                logging.info('use wetext frontend')
+            except:
+                self.text_frontend = ''
+                logging.info('no frontend is avaliable')
+
 
     def _extract_text_token(self, text):
         if isinstance(text, Generator):
@@ -89,7 +92,8 @@ 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 +105,8 @@ 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 +117,8 @@ 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)
@@ -122,15 +128,19 @@ 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()
-        if self.use_ttsfrd:
+        if self.text_frontend == 'ttsfrd':
             texts = [i["text"] for i in json.loads(self.frd.do_voicegen_frd(text))["sentences"]]
             text = ''.join(texts)
         else:
             if contains_chinese(text):
-                text = self.zh_tn_model.normalize(text)
+                if self.text_frontend == 'wetext':
+                    text = self.zh_tn_model.normalize(text)
                 text = text.replace("\n", "")
                 text = replace_blank(text)
                 text = replace_corner_mark(text)
@@ -141,7 +151,8 @@ class CosyVoiceFrontEnd:
                 texts = list(split_paragraph(text, partial(self.tokenizer.encode, allowed_special=self.allowed_special), "zh", token_max_n=80,
                                              token_min_n=60, merge_len=20, comma_split=False))
             else:
-                text = self.en_tn_model.normalize(text)
+                if self.text_frontend == 'wetext':
+                    text = self.en_tn_model.normalize(text)
                 text = spell_out_number(text, self.inflect_parser)
                 texts = list(split_paragraph(text, partial(self.tokenizer.encode, allowed_special=self.allowed_special), "en", token_max_n=80,
                                              token_min_n=60, merge_len=20, comma_split=False))
@@ -154,32 +165,31 @@ 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,
                            'prompt_speech_feat': speech_feat, 'prompt_speech_feat_len': speech_feat_len,
                            'llm_embedding': embedding, 'flow_embedding': embedding}
         else:
-            model_input = self.spk2info[zero_shot_spk_id]
+            model_input = {**self.spk2info[zero_shot_spk_id]}
         model_input['text'] = tts_text_token
         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 +201,21 @@ 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
@@ -63,14 +60,15 @@ class CosyVoiceModel:
         self.mel_overlap_dict = {}
         self.flow_cache_dict = {}
         self.hift_cache_dict = {}
+        self.silent_tokens = []
 
     def load(self, llm_model, flow_model, hift_model):
-        self.llm.load_state_dict(torch.load(llm_model, map_location=self.device), strict=True)
+        self.llm.load_state_dict(torch.load(llm_model, map_location=self.device, weights_only=True), strict=True)
         self.llm.to(self.device).eval()
-        self.flow.load_state_dict(torch.load(flow_model, map_location=self.device), strict=True)
+        self.flow.load_state_dict(torch.load(flow_model, map_location=self.device, weights_only=True), strict=True)
         self.flow.to(self.device).eval()
         # in case hift_model is a hifigan model
-        hift_state_dict = {k.replace('generator.', ''): v for k, v in torch.load(hift_model, map_location=self.device).items()}
+        hift_state_dict = {k.replace('generator.', ''): v for k, v in torch.load(hift_model, map_location=self.device, weights_only=True).items()}
         self.hift.load_state_dict(hift_state_dict, strict=True)
         self.hift.to(self.device).eval()
 
@@ -101,26 +99,33 @@ class CosyVoiceModel:
         return {'min_shape': min_shape, 'opt_shape': opt_shape, 'max_shape': max_shape, 'input_names': input_names}
 
     def llm_job(self, text, prompt_text, llm_prompt_speech_token, llm_embedding, uuid):
+        cur_silent_token_num, max_silent_token_num = 0, 5
         with self.llm_context, torch.cuda.amp.autocast(self.fp16 is True and hasattr(self.llm, 'vllm') is False):
             if isinstance(text, Generator):
-                assert isinstance(self, CosyVoice2Model) and not hasattr(self.llm, 'vllm'), 'streaming input text is only implemented for CosyVoice2 and do not support vllm!'
-                for i in self.llm.inference_bistream(text=text,
+                assert (self.__class__.__name__ != 'CosyVoiceModel') and not hasattr(self.llm, 'vllm'), 'streaming input text is only implemented for CosyVoice2/3 and do not support vllm!'
+                token_generator = self.llm.inference_bistream(text=text,
+                                                              prompt_text=prompt_text.to(self.device),
+                                                              prompt_text_len=torch.tensor([prompt_text.shape[1]], dtype=torch.int32).to(self.device),
+                                                              prompt_speech_token=llm_prompt_speech_token.to(self.device),
+                                                              prompt_speech_token_len=torch.tensor([llm_prompt_speech_token.shape[1]], dtype=torch.int32).to(self.device),
+                                                              embedding=llm_embedding.to(self.device))
+            else:
+                token_generator = self.llm.inference(text=text.to(self.device),
+                                                     text_len=torch.tensor([text.shape[1]], dtype=torch.int32).to(self.device),
                                                      prompt_text=prompt_text.to(self.device),
                                                      prompt_text_len=torch.tensor([prompt_text.shape[1]], dtype=torch.int32).to(self.device),
                                                      prompt_speech_token=llm_prompt_speech_token.to(self.device),
                                                      prompt_speech_token_len=torch.tensor([llm_prompt_speech_token.shape[1]], dtype=torch.int32).to(self.device),
-                                                     embedding=llm_embedding.to(self.device)):
-                    self.tts_speech_token_dict[uuid].append(i)
-            else:
-                for i in self.llm.inference(text=text.to(self.device),
-                                            text_len=torch.tensor([text.shape[1]], dtype=torch.int32).to(self.device),
-                                            prompt_text=prompt_text.to(self.device),
-                                            prompt_text_len=torch.tensor([prompt_text.shape[1]], dtype=torch.int32).to(self.device),
-                                            prompt_speech_token=llm_prompt_speech_token.to(self.device),
-                                            prompt_speech_token_len=torch.tensor([llm_prompt_speech_token.shape[1]], dtype=torch.int32).to(self.device),
-                                            embedding=llm_embedding.to(self.device),
-                                            uuid=uuid):
-                    self.tts_speech_token_dict[uuid].append(i)
+                                                     embedding=llm_embedding.to(self.device),
+                                                     uuid=uuid)  
+            for i in token_generator:
+                if i in self.silent_tokens:
+                    cur_silent_token_num += 1
+                    if cur_silent_token_num > max_silent_token_num:
+                        continue
+                else:
+                    cur_silent_token_num = 0
+                self.tts_speech_token_dict[uuid].append(i)
         self.llm_end_dict[uuid] = True
 
     def vc_job(self, source_speech_token, uuid):
@@ -129,7 +134,7 @@ 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),
@@ -249,11 +254,12 @@ 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
+        # NOTE increase token_hop_len incrementally to avoid duplicate inference 
+        self.token_max_hop_len = 4 * self.token_hop_len
+        self.stream_scale_factor = 2
+        assert self.stream_scale_factor >= 1, 'stream_scale_factor should be greater than 1, change it according to your actual rtf'
         # hift cache
         self.mel_cache_len = 8
         self.source_cache_len = int(self.mel_cache_len * 480)
@@ -266,6 +272,7 @@ class CosyVoice2Model(CosyVoiceModel):
         self.tts_speech_token_dict = {}
         self.llm_end_dict = {}
         self.hift_cache_dict = {}
+        self.silent_tokens = []
 
     def load_jit(self, flow_encoder_model):
         flow_encoder = torch.jit.load(flow_encoder_model, map_location=self.device)
@@ -284,7 +291,7 @@ 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),
@@ -350,6 +357,7 @@ class CosyVoice2Model(CosyVoiceModel):
                                                      stream=stream,
                                                      finalize=False)
                     token_offset += this_token_hop_len
+                    self.token_hop_len = min(self.token_max_hop_len, self.token_hop_len * self.stream_scale_factor)
                     yield {'tts_speech': this_tts_speech.cpu()}
                 if self.llm_end_dict[this_uuid] is True and len(self.tts_speech_token_dict[this_uuid]) - token_offset < this_token_hop_len + self.flow.pre_lookahead_len:
                     break
@@ -384,3 +392,59 @@ 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
+        # NOTE increase token_hop_len incrementally to avoid duplicate inference 
+        self.token_max_hop_len = 4 * self.token_hop_len
+        self.stream_scale_factor = 2
+        assert self.stream_scale_factor >= 1, 'stream_scale_factor should be greater than 1, change it according to your actual rtf'
+        # 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 = {}
+        # FSQ silent and breath token
+        self.silent_tokens = [1, 2, 28, 29, 55, 248, 494, 2241, 2242, 2322, 2323]
+
+    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/dataset.py

@@ -145,7 +145,11 @@ def Dataset(data_list_file,
                        shuffle=shuffle,
                        partition=partition)
     # map partial arg to padding func
-    data_pipeline[-1] = partial(data_pipeline[-1], gan=gan, dpo=dpo)
+    for i in range(1, len(data_pipeline)):
+        if data_pipeline[i].func.__name__ == 'compute_fbank' and gan is True:
+            data_pipeline[i] = partial(data_pipeline[i], token_mel_ratio=0)
+        if data_pipeline[i].func.__name__ == 'padding':
+            data_pipeline[i] = partial(data_pipeline[i], gan=gan, dpo=dpo)
     for func in data_pipeline:
         dataset = Processor(dataset, func, mode=mode)
     return dataset

cosyvoice/dataset/processor.py

@@ -16,17 +16,19 @@ import random
 
 import pyarrow.parquet as pq
 from io import BytesIO
+import numpy as np
+import whisper
 import torch
 import torchaudio
 from torch.nn.utils.rnn import pad_sequence
 import torch.nn.functional as F
 import pyworld as pw
-
+from cosyvoice.utils.onnx import embedding_extractor, online_feature
 
 AUDIO_FORMAT_SETS = {'flac', 'mp3', 'm4a', 'ogg', 'opus', 'wav', 'wma'}
 
 
-def parquet_opener(data, mode='train', tts_data={}):
+def parquet_opener(data, mode='train'):
     """ Give url or local file, return file descriptor
         Inplace operation.
 
@@ -44,12 +46,8 @@ def parquet_opener(data, mode='train', tts_data={}):
                 df = df.to_pandas()
                 for i in range(len(df)):
                     sample.update(dict(df.loc[i]))
-                    if mode == 'train':
-                        # NOTE do not return sample directly, must initialize a new dict
-                        yield {**sample}
-                    else:
-                        for index, text in enumerate(tts_data[df.loc[i, 'utt']]):
-                            yield {**sample, 'tts_index': index, 'tts_text': text}
+                    # NOTE do not return sample directly, must initialize a new dict
+                    yield {**sample}
         except Exception as ex:
             logging.warning('Failed to open {}, ex info {}'.format(url, ex))
 
@@ -96,9 +94,9 @@ def filter(data,
             continue
         if len(sample['text_token']) > token_max_length:
             continue
-        if len(sample['speech_token']) == 0:
+        if online_feature is False and len(sample['speech_token']) == 0:
             continue
-        if 'reject_speech_token' in sample and len(sample['reject_speech_token']) == 0:
+        if online_feature is False and 'reject_speech_token' in sample and len(sample['reject_speech_token']) == 0:
             continue
         if num_frames != 0:
             if len(sample['text_token']) / num_frames < min_output_input_ratio:
@@ -159,7 +157,7 @@ def truncate(data, truncate_length=24576, mode='train'):
 
 def compute_fbank(data,
                   feat_extractor,
-                  token_mel_ratio=0,
+                  num_frames=-1,
                   mode='train'):
     """ Extract fbank
 
@@ -174,14 +172,28 @@ def compute_fbank(data,
         assert 'speech' in sample
         assert 'utt' in sample
         assert 'text_token' in sample
-        waveform = sample['speech']
-        feat = feat_extractor(waveform).squeeze(dim=0).transpose(0, 1)
-        if token_mel_ratio != 0:
-            # trim to align speech_token and speech_feat
-            token_len = int(min(feat.shape[0] / token_mel_ratio, sample["speech_token"].shape[0]))
-            feat = feat[:token_mel_ratio * token_len]
-            sample["speech_token"] = sample["speech_token"][:token_len]
-        sample['speech_feat'] = feat
+        # NOTE in cosyvoice2/3, we support online token extraction, so we need to align speech to 25hz first
+        if num_frames != -1:
+            index = int(np.ceil(sample['speech'].shape[1] / num_frames))
+            sample['speech'] = torch.concat([sample['speech'], torch.zeros(1, index * num_frames - sample['speech'].shape[1])], dim=1)
+        sample['speech_feat'] = feat_extractor(sample['speech']).squeeze(dim=0).transpose(0, 1)
+        yield sample
+
+
+def compute_whisper_fbank(data, num_frames=-1, mode='train'):
+    """ Extract whisper fbank 
+
+        Args:
+            data: Iterable[{key, wav, label, sample_rate}]
+
+        Returns:
+            Iterable[{key, feat, label}]
+    """
+    for sample in data:
+        if num_frames != -1:
+            assert sample['speech'].shape[1] % num_frames == 0, 'speech length is not aligned with speech_token'
+        sample['speech_16k'] = torchaudio.transforms.Resample(orig_freq=sample['sample_rate'], new_freq=16000)(sample['speech'])
+        sample['whisper_feat'] = whisper.log_mel_spectrogram(sample['speech_16k'], n_mels=128).squeeze(dim=0).transpose(0, 1)
         yield sample
 
 
@@ -220,8 +232,13 @@ def parse_embedding(data, normalize, mode='train'):
             Iterable[{key, feat, label}]
     """
     for sample in data:
-        sample['utt_embedding'] = torch.tensor(sample['utt_embedding'], dtype=torch.float32)
-        sample['spk_embedding'] = torch.tensor(sample['spk_embedding'], dtype=torch.float32)
+        if 'utt_embedding' not in sample and 'spk_embedding' not in sample:
+            sample['speech_16k'] = torchaudio.transforms.Resample(orig_freq=sample['sample_rate'], new_freq=16000)(sample['speech'])
+            embedding = embedding_extractor.inference(sample['speech_16k'])
+            sample['spk_embedding'] = sample['utt_embedding'] = embedding
+        else:
+            sample['utt_embedding'] = torch.tensor(sample['utt_embedding'], dtype=torch.float32)
+            sample['spk_embedding'] = torch.tensor(sample['spk_embedding'], dtype=torch.float32)
         if normalize:
             sample['utt_embedding'] = F.normalize(sample['utt_embedding'], dim=0)
             sample['spk_embedding'] = F.normalize(sample['spk_embedding'], dim=0)
@@ -242,6 +259,8 @@ 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)
         yield sample
 
 
@@ -256,13 +275,14 @@ def shuffle(data, shuffle_size=10000, mode='train'):
             Iterable[{key, feat, label}]
     """
     buf = []
+    yield_size = int(shuffle_size / 2)
     for sample in data:
         buf.append(sample)
         if len(buf) >= shuffle_size:
             random.shuffle(buf)
-            for x in buf:
+            for x in buf[:yield_size]:
                 yield x
-            buf = []
+            buf = buf[yield_size:]
     # The sample left over
     random.shuffle(buf)
     for x in buf:
@@ -368,65 +388,42 @@ def padding(data, use_spk_embedding, mode='train', gan=False, dpo=False):
     """
     for sample in data:
         assert isinstance(sample, list)
-        speech_feat_len = torch.tensor([x['speech_feat'].size(1) for x in sample],
-                                       dtype=torch.int32)
-        order = torch.argsort(speech_feat_len, descending=True)
-
-        utts = [sample[i]['utt'] for i in order]
-        speech = [sample[i]['speech'].squeeze(dim=0) for i in order]
-        speech_len = torch.tensor([i.size(0) for i in speech], dtype=torch.int32)
-        speech = pad_sequence(speech, batch_first=True, padding_value=0)
-        speech_token = [torch.tensor(sample[i]['speech_token']) for i in order]
-        speech_token_len = torch.tensor([i.size(0) for i in speech_token], dtype=torch.int32)
-        speech_token = pad_sequence(speech_token,
-                                    batch_first=True,
-                                    padding_value=0)
-        speech_feat = [sample[i]['speech_feat'] for i in order]
-        speech_feat_len = torch.tensor([i.size(0) for i in speech_feat], dtype=torch.int32)
-        speech_feat = pad_sequence(speech_feat,
-                                   batch_first=True,
-                                   padding_value=0)
-        text = [sample[i]['text'] for i in order]
+        order = torch.argsort(torch.tensor([x['speech'].size(1) for x in sample], dtype=torch.int32), descending=True)
+        batch = {}
+        batch['utts'] = [sample[i]['utt'] for i in order]
+        batch['text'] = [sample[i]['text'] for i in order]
         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)
-        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 = {
-            "utts": utts,
-            "speech": speech,
-            "speech_len": speech_len,
-            "speech_token": speech_token,
-            "speech_token_len": speech_token_len,
-            "speech_feat": speech_feat,
-            "speech_feat_len": speech_feat_len,
-            "text": text,
-            "text_token": text_token,
-            "text_token_len": text_token_len,
-            "utt_embedding": utt_embedding,
-            "spk_embedding": spk_embedding,
-        }
+        batch['text_token_len'] = torch.tensor([i.size(0) for i in text_token], dtype=torch.int32)
+        batch['text_token'] = pad_sequence(text_token, batch_first=True, padding_value=0)
+        speech_feat = [sample[i]['speech_feat'] for i in order]
+        batch['speech_feat_len'] = torch.tensor([i.size(0) for i in speech_feat], dtype=torch.int32)
+        batch['speech_feat'] = pad_sequence(speech_feat, batch_first=True, padding_value=0)
+        batch['utt_embedding'] = torch.stack([sample[i]['utt_embedding'] for i in order], dim=0)
+        batch['spk_embedding'] = torch.stack([sample[i]['spk_embedding'] for i in order], dim=0)
+        if torch.tensor(['instruct_token' in sample[i] for i in order]).all():
+            instruct_token = [torch.tensor(sample[i]['instruct_token']) for i in order]
+            batch['instruct_token_len'] = torch.tensor([i.size(0) for i in instruct_token], dtype=torch.int32)
+            batch['instruct_token'] = pad_sequence(instruct_token, batch_first=True, padding_value=0)
+        if torch.tensor(['whisper_feat' in sample[i] for i in order]).all():
+            whisper_feat = [sample[i]['whisper_feat'] for i in order]
+            batch['whisper_feat_len'] = torch.tensor([i.size(0) for i in whisper_feat], dtype=torch.int32)
+            batch['whisper_feat'] = pad_sequence(whisper_feat, batch_first=True, padding_value=0)
+        if torch.tensor(['speech_token' in sample[i] for i in order]).all():
+            speech_token = [torch.tensor(sample[i]['speech_token']) for i in order]
+            batch['speech_token_len'] = torch.tensor([i.size(0) for i in speech_token], dtype=torch.int32)
+            batch['speech_token'] = pad_sequence(speech_token, batch_first=True, padding_value=0)
         if gan is True:
-            # in gan train, we need pitch_feat
+            # in gan train, we need speech/pitch_feat
+            speech = [sample[i]['speech'].squeeze(dim=0) for i in order]
+            batch['speech_len'] = torch.tensor([i.size(0) for i in speech], dtype=torch.int32)
+            batch['speech'] = pad_sequence(speech, batch_first=True, padding_value=0)
             pitch_feat = [sample[i]['pitch_feat'] for i in order]
-            pitch_feat_len = torch.tensor([i.size(0) for i in pitch_feat], dtype=torch.int32)
-            pitch_feat = pad_sequence(pitch_feat,
-                                      batch_first=True,
-                                      padding_value=0)
-            batch["pitch_feat"] = pitch_feat
-            batch["pitch_feat_len"] = pitch_feat_len
-        else:
-            # only gan train needs speech, delete it to save memory
-            del batch["speech"]
-            del batch["speech_len"]
+            batch['pitch_feat_len'] = torch.tensor([i.size(0) for i in pitch_feat], dtype=torch.int32)
+            batch['pitch_feat'] = pad_sequence(pitch_feat, batch_first=True, padding_value=0)
         if dpo is True:
             reject_speech_token = [torch.tensor(sample[i]['reject_speech_token']) for i in order]
-            reject_speech_token_len = torch.tensor([i.size(0) for i in reject_speech_token], dtype=torch.int32)
-            reject_speech_token = pad_sequence(reject_speech_token,
-                                               batch_first=True,
-                                               padding_value=0)
-            batch['reject_speech_token'] = reject_speech_token
-            batch['reject_speech_token_len'] = reject_speech_token_len
+            batch['reject_speech_token_len'] = torch.tensor([i.size(0) for i in reject_speech_token], dtype=torch.int32)
+            batch['reject_speech_token'] = pad_sequence(reject_speech_token, batch_first=True, padding_value=0)
         if use_spk_embedding is True:
             batch["embedding"] = batch["spk_embedding"]
         else:

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

@@ -11,7 +11,7 @@
 # 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.
-import logging
+import os, logging
 import random
 from typing import Dict, Optional
 import torch
@@ -19,6 +19,7 @@ import torch.nn as nn
 from torch.nn import functional as F
 from omegaconf import DictConfig
 from cosyvoice.utils.mask import make_pad_mask
+from cosyvoice.utils.onnx import SpeechTokenExtractor, online_feature, onnx_path
 
 
 class MaskedDiffWithXvec(torch.nn.Module):
@@ -37,14 +38,11 @@ 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 +163,11 @@ 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
@@ -185,14 +180,19 @@ class CausalMaskedDiffWithXvec(torch.nn.Module):
         self.only_mask_loss = only_mask_loss
         self.token_mel_ratio = token_mel_ratio
         self.pre_lookahead_len = pre_lookahead_len
+        if online_feature is True:
+            self.speech_token_extractor = SpeechTokenExtractor(model_path=os.path.join(onnx_path, 'speech_tokenizer_v2.batch.onnx'))
 
     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)
+        if 'speech_token' not in batch:
+            token, token_len = self.speech_token_extractor.inference(batch['whisper_feat'], batch['whisper_feat_len'], device)
+        else:
+            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)
@@ -279,3 +279,165 @@ 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
+        if online_feature is True:
+            self.speech_token_extractor = SpeechTokenExtractor(model_path=os.path.join(onnx_path, 'speech_tokenizer_v3.batch.onnx'))
+
+    def forward(
+            self,
+            batch: dict,
+            device: torch.device,
+    ) -> Dict[str, Optional[torch.Tensor]]:
+        if 'speech_token' not in batch:
+            token, token_len = self.speech_token_extractor.inference(batch['whisper_feat'], batch['whisper_feat_len'], device)
+        else:
+            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 = self.pre_lookahead_layer(token)
+        h = h.repeat_interleave(self.token_mel_ratio, dim=1)
+        mask = mask.repeat_interleave(self.token_mel_ratio, dim=1).squeeze(dim=-1)
+
+        # 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)
+
+        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
@@ -173,8 +174,7 @@ class ConditionalCFM(BASECFM):
 
         # random timestep
         t = torch.rand([b, 1, 1], device=mu.device, dtype=mu.dtype)
-        if self.t_scheduler == 'cosine':
-            t = 1 - torch.cos(t * 0.5 * torch.pi)
+
         # sample noise p(x_0)
         z = torch.randn_like(x1)
 

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(torch.float64)
+        f0 = self.f0_predictor(speech_feat.to(torch.float64), 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

@@ -1,5 +1,5 @@
 # Copyright (c) 2024 Alibaba Inc (authors: Xiang Lyu, Zhihao Du)
-#               2025 Alibaba Inc (authors: Xiang Lyu, Yabin Li, Qihua)
+#               2025 Alibaba Inc (authors: Xiang Lyu, Yabin Li, Qihua, Shengqiang Li)
 #
 # Licensed under the Apache License, Version 2.0 (the "License");
 # you may not use this file except in compliance with the License.
@@ -12,11 +12,12 @@
 # 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.
-import queue
+import os, queue
 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
@@ -27,6 +28,7 @@ from cosyvoice.transformer.label_smoothing_loss import LabelSmoothingLoss
 from cosyvoice.utils.common import th_accuracy
 from cosyvoice.utils.file_utils import logging
 from cosyvoice.utils.mask import make_pad_mask
+from cosyvoice.utils.onnx import SpeechTokenExtractor, online_feature, onnx_path
 
 
 class TransformerLM(torch.nn.Module):
@@ -56,8 +58,9 @@ 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 +88,10 @@ 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 +129,15 @@ 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 +157,7 @@ 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 +196,13 @@ 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 +218,8 @@ 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 +276,10 @@ 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
@@ -300,19 +301,29 @@ class Qwen2LM(TransformerLM):
         # 5. vllm related
         self.stop_token_ids = [speech_token_size + i for i in range(3)]
         self.vllm_output_queue = {}
+        if online_feature is True:
+            self.speech_token_extractor = SpeechTokenExtractor(model_path=os.path.join(onnx_path, 'speech_tokenizer_v2.batch.onnx'))
 
-    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, instruct_token=None, instruct_token_emb=None, instruct_token_len=None):
         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)
         text_token_emb = unpad_sequence(text_token_emb, text_token_len.cpu(), batch_first=True)
         speech_token_emb = unpad_sequence(speech_token_emb, speech_token_len.cpu(), batch_first=True)
+        # NOTE add instruct_token in CosyVoice3
+        if instruct_token is not None and instruct_token_emb is not None and instruct_token_len is not None:
+            instruct_token = unpad_sequence(instruct_token, instruct_token_len.cpu(), batch_first=True)
+            instruct_token_emb = unpad_sequence(instruct_token_emb, instruct_token_len.cpu(), batch_first=True)
+        else:
+            instruct_token = [torch.empty(0).to(text_token[0])] * len(text_token)
+            instruct_token_emb = [torch.empty(0, 896).to(text_token_emb[0])] * len(text_token)
+            instruct_token_len = torch.zeros(len(text_token)).to(text_token_len)
         for i in range(len(text_token)):
             # bistream sequence
             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_target, this_lm_input = [IGNORE_ID], [sos_emb.squeeze(dim=0)]
+                this_lm_target += [IGNORE_ID] * instruct_token_len[i]
+                this_lm_input.append(instruct_token_emb[i])
                 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 +331,21 @@ 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 + instruct_token_len[i] + text_token_len[i]) + speech_token[i].tolist() + [self.eos_token])
+                this_lm_input = torch.concat([sos_emb.squeeze(dim=0), instruct_token_emb[i], 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)
@@ -357,17 +367,25 @@ class Qwen2LM(TransformerLM):
         """
         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)
+        if 'speech_token' not in batch:
+            speech_token, speech_token_len = self.speech_token_extractor.inference(batch['whisper_feat'], batch['whisper_feat_len'], device)
+        else:
+            speech_token = batch['speech_token'].to(device)
+            speech_token_len = batch['speech_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.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 +410,10 @@ 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 +423,8 @@ 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
@@ -420,8 +442,8 @@ class Qwen2LM(TransformerLM):
         rejected_lm_mask = rejected_lm_target == IGNORE_ID
         chosen_logps = torch.gather(chosen_logits.log_softmax(dim=-1), dim=2, index=chosen_lm_target.masked_fill(chosen_lm_mask, 0).unsqueeze(dim=-1)).squeeze(dim=-1)
         rejected_logps = torch.gather(rejected_logits.log_softmax(dim=-1), dim=2, index=rejected_lm_target.masked_fill(rejected_lm_mask, 0).unsqueeze(dim=-1)).squeeze(dim=-1)
-        chosen_logps = (chosen_logps * chosen_lm_mask).mean(dim=-1)
-        rejected_logps = (rejected_logps * chosen_lm_mask).mean(dim=-1)
+        chosen_logps = (chosen_logps * chosen_lm_mask).sum(dim=-1) / chosen_lm_mask.sum(dim=-1)
+        rejected_logps = (rejected_logps * rejected_lm_mask).sum(dim=-1) / rejected_lm_mask.sum(dim=-1)
         return {'loss': loss, 'acc': acc, 'chosen_logps': chosen_logps, 'rejected_logps': rejected_logps}
 
     @torch.inference_mode()
@@ -445,13 +467,13 @@ 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 +522,9 @@ 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 +546,20 @@ 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 +574,12 @@ 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 +594,16 @@ 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 +619,142 @@ 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 = {}
+        if online_feature is True:
+            self.speech_token_extractor = SpeechTokenExtractor(model_path=os.path.join(onnx_path, 'speech_tokenizer_v3.batch.onnx'))
+
+    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)
+        if 'speech_token' not in batch:
+            speech_token, speech_token_len = self.speech_token_extractor.inference(batch['whisper_feat'], batch['whisper_feat_len'], device)
+        else:
+            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)
+        instruct_token_emb = self.llm.model.model.embed_tokens(instruct_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, instruct_token, instruct_token_emb, instruct_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 + 200), 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.
@@ -112,6 +139,7 @@ def ras_sampling(weighted_scores, decoded_tokens, sampling, top_p=0.8, top_k=25,
     top_ids = nucleus_sampling(weighted_scores, top_p=top_p, top_k=top_k)
     rep_num = (torch.tensor(decoded_tokens[-win_size:]).to(weighted_scores.device) == top_ids).sum().item()
     if rep_num >= win_size * tau_r:
+        weighted_scores[top_ids] = -float('inf')
         top_ids = random_sampling(weighted_scores, decoded_tokens, sampling)
     return top_ids
 
@@ -130,12 +158,12 @@ 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/executor.py

@@ -166,7 +166,7 @@ class Executor:
             for k, v in info_dict['loss_dict'].items():
                 if k not in total_loss_dict:
                     total_loss_dict[k] = []
-                total_loss_dict[k].append(v.item() * num_utts)
+                total_loss_dict[k].append(v.mean().item() * num_utts)
             log_per_step(None, info_dict)
         for k, v in total_loss_dict.items():
             total_loss_dict[k] = sum(v) / total_num_utts

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)

cosyvoice/utils/onnx.py

@@ -0,0 +1,54 @@
+import onnxruntime
+import torch, random
+import os
+import torchaudio.compliance.kaldi as kaldi
+
+
+class SpeechTokenExtractor():
+    def __init__(self, model_path):
+        self.local_rank = int(os.environ.get("LOCAL_RANK", 0))
+        option = onnxruntime.SessionOptions()
+        option.graph_optimization_level = onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
+        option.intra_op_num_threads = 1
+        self.speech_tokenizer_session = onnxruntime.InferenceSession(model_path,
+                                                                     sess_options=option,
+                                                                     providers=[("CUDAExecutionProvider", {'device_id': self.local_rank})])
+
+    def inference(self, feat, feat_lengths, device):
+        speech_token = self.speech_tokenizer_session.run(None,
+                                                    {self.speech_tokenizer_session.get_inputs()[0].name:
+                                                    feat.transpose(1, 2).detach().cpu().numpy(),
+                                                    self.speech_tokenizer_session.get_inputs()[1].name:
+                                                    feat_lengths.detach().cpu().numpy()})[0]
+        return torch.tensor(speech_token).to(torch.int32).to(device), (feat_lengths / 4).to(torch.int32).to(device)
+
+
+class EmbeddingExtractor():
+    def __init__(self, model_path):
+        option = onnxruntime.SessionOptions()
+        option.graph_optimization_level = onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
+        option.intra_op_num_threads = 1
+        self.max_len = 10 * 16000
+        self.campplus_session = onnxruntime.InferenceSession(model_path,
+                                                             sess_options=option,
+                                                             providers=["CPUExecutionProvider"])
+
+    def inference(self, speech):
+        if speech.shape[1] > self.max_len:
+            start_index = random.randint(0, speech.shape[1] - self.max_len)
+            speech = speech[:, start_index: start_index + self.max_len]
+        feat = kaldi.fbank(speech,
+                           num_mel_bins=80,
+                           dither=0,
+                           sample_frequency=16000)
+        feat = feat - feat.mean(dim=0, keepdim=True)
+        embedding = self.campplus_session.run(None,
+                                              {self.campplus_session.get_inputs()[0].name: feat.unsqueeze(dim=0).cpu().numpy()})[0].flatten().tolist()
+        return torch.tensor(embedding).to(speech.device)
+
+# singleton mode, only initialized once
+onnx_path = os.environ.get('onnx_path')
+if onnx_path is not None:
+    embedding_extractor, online_feature = EmbeddingExtractor(model_path=os.path.join(onnx_path, 'campplus.onnx')), True
+else:
+    embedding_extractor, online_feature = None, False

cosyvoice/utils/train_utils.py

@@ -53,7 +53,7 @@ def init_distributed(args):
 def init_dataset_and_dataloader(args, configs, gan, dpo):
     data_pipeline = configs['data_pipeline_gan'] if gan is True else configs['data_pipeline']
     train_dataset = Dataset(args.train_data, data_pipeline=data_pipeline, mode='train', gan=gan, dpo=dpo, shuffle=True, partition=True)
-    cv_dataset = Dataset(args.cv_data, data_pipeline=data_pipeline, mode='train', gan=gan, dpo=dpo, shuffle=False, partition=False)
+    cv_dataset = Dataset(args.cv_data, data_pipeline=data_pipeline, mode='dev', gan=gan, dpo=dpo, shuffle=False, partition=False)
 
     # do not use persistent_workers=True, as whisper tokenizer opens tiktoken file each time when the for loop starts
     train_data_loader = DataLoader(train_dataset,
@@ -71,7 +71,7 @@ def init_dataset_and_dataloader(args, configs, gan, dpo):
 
 def check_modify_and_save_config(args, configs):
     if args.train_engine == "torch_ddp":
-        configs['train_conf']["dtype"] = 'fp32'
+        configs['train_conf']["dtype"] = 'bf16' if args.use_amp is True else 'fp32'
     else:
         with open(args.deepspeed_config, 'r') as fin:
             ds_configs = json.load(fin)
@@ -164,18 +164,18 @@ def init_optimizer_and_scheduler(args, configs, model, gan):
             raise ValueError("unknown scheduler: " + configs['train_conf'])
 
         if configs['train_conf']['optim_d'] == 'adam':
-            optimizer_d = optim.Adam(model.module.discriminator.parameters(), **configs['train_conf']['optim_conf'])
+            optimizer_d = optim.Adam(model.module.discriminator.parameters(), **configs['train_conf']['optim_conf_d'])
         elif configs['train_conf']['optim_d'] == 'adamw':
-            optimizer_d = optim.AdamW(model.module.discriminator.parameters(), **configs['train_conf']['optim_conf'])
+            optimizer_d = optim.AdamW(model.module.discriminator.parameters(), **configs['train_conf']['optim_conf_d'])
         else:
             raise ValueError("unknown optimizer: " + configs['train_conf'])
 
         if configs['train_conf']['scheduler_d'] == 'warmuplr':
             scheduler_type = WarmupLR
-            scheduler_d = WarmupLR(optimizer_d, **configs['train_conf']['scheduler_conf'])
+            scheduler_d = WarmupLR(optimizer_d, **configs['train_conf']['scheduler_d'])
         elif configs['train_conf']['scheduler_d'] == 'NoamHoldAnnealing':
             scheduler_type = NoamHoldAnnealing
-            scheduler_d = NoamHoldAnnealing(optimizer_d, **configs['train_conf']['scheduler_conf'])
+            scheduler_d = NoamHoldAnnealing(optimizer_d, **configs['train_conf']['scheduler_d'])
         elif configs['train_conf']['scheduler'] == 'constantlr':
             scheduler_type = ConstantLR
             scheduler_d = ConstantLR(optimizer_d)
@@ -247,7 +247,7 @@ def batch_forward(model, batch, scaler, info_dict, ref_model=None, dpo_loss=None
         dtype = torch.float32
 
     if info_dict['train_engine'] == 'torch_ddp':
-        autocast = torch.cuda.amp.autocast(enabled=scaler is not None)
+        autocast = torch.cuda.amp.autocast(enabled=scaler is not None, dtype=dtype)
     else:
         autocast = torch.cuda.amp.autocast(enabled=True, dtype=dtype, cache_enabled=False)
 

cosyvoice/vllm/cosyvoice2.py

@@ -23,6 +23,15 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 """Inference-only Qwen2 model compatible with HuggingFace weights."""
+from typing import Optional
+from packaging.version import parse as vparse
+import vllm
+
+# vLLM-0.11.0+ only support V1 engine
+VLLM_V1_ENGINE_ONLY: bool = vparse(vllm.__version__) >= vparse("0.11.0")
+if VLLM_V1_ENGINE_ONLY:
+    from vllm.v1.sample.metadata import SamplingMetadata
+
 from vllm.model_executor.models.qwen2 import *
 
 
@@ -87,10 +96,14 @@ class CosyVoice2ForCausalLM(nn.Module, SupportsLoRA, SupportsPP):
     def compute_logits(
         self,
         hidden_states: torch.Tensor,
-        sampling_metadata: SamplingMetadata,
+        sampling_metadata: Optional[SamplingMetadata] = None,
     ) -> Optional[torch.Tensor]:
-        logits = self.logits_processor(self.lm_head, hidden_states,
-                                       sampling_metadata, self.lm_head.bias)
+        if VLLM_V1_ENGINE_ONLY:
+            logits = self.logits_processor(self.lm_head, hidden_states, 
+                                           self.lm_head.bias)
+        else:
+            logits = self.logits_processor(self.lm_head, hidden_states,
+                                           sampling_metadata, self.lm_head.bias)
         return logits
 
     def load_weights(self, weights: Iterable[tuple[str,

docker/Dockerfile

@@ -4,7 +4,7 @@ ARG VENV_NAME="cosyvoice"
 ENV VENV=$VENV_NAME
 ENV LANG=C.UTF-8 LC_ALL=C.UTF-8
 
-ENV DEBIAN_FRONTEN=noninteractive
+ENV DEBIAN_FRONTEND=noninteractive
 ENV PYTHONUNBUFFERED=1
 SHELL ["/bin/bash", "--login", "-c"]
 
@@ -46,6 +46,6 @@ RUN git clone --recursive https://github.com/FunAudioLLM/CosyVoice.git
 
 RUN conda activate ${VENV} && conda install -y -c conda-forge pynini==2.1.5
 RUN conda activate ${VENV} && cd CosyVoice && \
-    pip install -r requirements.txt -i https://mirrors.aliyun.com/pypi/simple/ --trusted-host=mirrors.aliyun.com
+    pip install -r requirements.txt -i https://mirrors.aliyun.com/pypi/simple/ --trusted-host=mirrors.aliyun.com --no-cache-dir
 
 WORKDIR /workspace/CosyVoice

example.py

@@ -0,0 +1,112 @@
+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|><|ja|><|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)
+
+    # NOTE for Japanese usage, you must translate it to katakana.
+    # 歴史的世界においては、過去は単に過ぎ去ったものではない、プラトンのいう如く非有が有である。 -> レキシ テキ セカイ ニ オイ テ ワ、カコ ワ タンニ スギサッ タ モノ デ ワ ナイ、プラトン ノ イウ ゴトク ヒ ユー ガ ユー デ アル。
+    for i, j in enumerate(cosyvoice.inference_cross_lingual('You are a helpful assistant.<|endofprompt|>レキシ テキ セカイ ニ オイ テ ワ、カコ ワ タンニ スギサッ タ モノ デ ワ ナイ、プラトン ノ イウ ゴトク ヒ ユー ガ ユー デ アル。',
+                                                            './asset/zero_shot_prompt.wav', stream=False)):
+        torchaudio.save('japanese_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)    
+
+
+def main():
+    # cosyvoice_example()
+    # cosyvoice2_example()
+    cosyvoice3_example()
+
+
+if __name__ == '__main__':
+    main()

examples/grpo/cosyvoice2/Dockerfile

@@ -0,0 +1,6 @@
+FROM verlai/verl:app-verl0.4-vllm0.8.5-mcore0.12.2-te2.2
+COPY requirements.txt /myworkspace/requirements.txt
+RUN pip install -r /myworkspace/requirements.txt
+RUN pip install -U nvidia-pytriton
+RUN git clone https://github.com/yuekaizhang/verl.git /myworkspace/verl -b thread && cd /myworkspace/verl && pip install --no-deps -e .
+RUN git clone https://github.com/yuekaizhang/PytritonSenseVoice.git /myworkspace/PytritonSenseVoice && cd /myworkspace/PytritonSenseVoice && pip install -e .

examples/grpo/cosyvoice2/README.md

@@ -0,0 +1,125 @@
+# CosyVoice2 LLM Reinforcement Learning Recipe
+
+This recipe demonstrates how to fine-tune the **CosyVoice2** large language model with reinforcement learning algorithms—specifically **GRPO**—using the [veRL](https://github.com/volcengine/verl) framework. Our experiments show that applying GRPO reduces the character error rate (CER) on the CosyVoice3 `zero_shot_zh` set from 4.08% to 3.36%.
+
+## Table of Contents
+
+- [Environment Setup](#environment-setup)
+- [Data Preparation](#data-preparation)
+- [Reward Function & ASR Server](#reward-function--asr-server)
+- [Training](#training)
+- [Evaluation](#evaluation)
+- [Export Model](#export-model)
+- [Results](#results)
+- [Acknowledgement](#acknowledgement)
+
+## Environment Setup
+We recommend using the pre-built Docker image below. Alternatively, you can manually install the dependencies following the Dockerfile.
+```bash
+docker pull soar97/verl:app-verl0.4-vllm0.8.5-mcore0.12.2-te2.2
+```
+If Docker is not available, you can refer to `run.sh` `stage -2` to install the dependencies locally.
+
+## Data Preparation
+
+`prepare_data.py` expects a JSON/JSONL file with at least the following schema:
+
+```jsonc
+{
+  "text": "An example sentence to be synthesized."
+}
+```
+You can download the JSONL files from the metadata directory of the [SparkAudio/voxbox](https://huggingface.co/datasets/SparkAudio/voxbox/tree/main/metadata) dataset on Hugging Face.
+
+Stage `0` converts raw JSONL files into the parquet format expected by veRL:
+
+```bash
+bash run.sh 0 0
+```
+Create two JSONL files—`train.jsonl` and `test.jsonl`.
+The script will then generate two Parquet files:
+
+```
+data/parquet_tiny/train.parquet
+data/parquet_tiny/test.parquet
+```
+
+Each sample is automatically wrapped into a CosyVoice2-style prompt so that the LLM learns to output CosyVoice2 speech tokens.
+
+
+## Reward Function & ASR Server
+
+To compute rewards, we run a lightweight server that:
+
+1. Converts generated speech tokens back to a 16 kHz waveform with the **CosyVoice2** pretrained U-Net model.
+2. Transcribes the waveform with **SenseVoice** ASR.
+3. Calculates the pinyin-level error rate relative to the ground-truth text and maps it to a score between 0 and 1.
+
+Start the server (stage `1`) in a dedicated terminal or on a separate GPU:
+
+```bash
+bash run.sh 1 1
+# Triton server listens on ports 8000/8001/8002
+```
+
+The custom reward implementation is located in [`reward_tts.py`](./reward_tts.py) and calls the server to obtain the reward score.
+
+## Training
+
+Run stage `2` to start GRPO training:
+
+```bash
+bash run.sh 2 2
+```
+
+Key CLI arguments passed to `verl.trainer.main_ppo`:
+
+* `algorithm.adv_estimator=grpo` – use GRPO instead of PPO.
+* `data.train_files=data/parquet_aishell3/train.parquet` and `data.val_files=data/parquet_aishell3/test.parquet`
+* `custom_reward_function.path=reward_tts.py` – custom reward function described above.
+
+Adjust `CUDA_VISIBLE_DEVICES`, batch sizes, and other hyperparameters to match your hardware.
+> [!TIP]
+> Note: the lm_head bias is disabled during training to make the model compatible with VLLM and Transformers' Qwen model.
+
+## Evaluation
+
+After training is complete, collect the sharded FSDP weights and export a Hugging Face-style checkpoint (stage `3`):
+
+```bash
+bash run.sh 3 3   # merges weights into $llm_path/merged_hf_model
+```
+
+You can then evaluate the model on the CosyVoice3 zero-shot Chinese test set (stage `4`):
+
+```bash
+bash run.sh 4 4
+```
+
+This command launches distributed inference via `infer_dataset.py` and computes WER with `scripts/compute_wer.sh`.
+
+> [!TIP]
+> The script also supports the Seed-TTS test set by setting `dataset=test_zh`.
+
+## Export Model
+
+To use the RL-trained model with the official CosyVoice repository:
+
+```bash
+bash run.sh 5 5
+```
+
+The script converts the Hugging Face checkpoint back into the format expected by the CosyVoice repository.
+> [!TIP]
+>  However, we observed a slight accuracy drop when using the RL-trained model after conversion, compared with the Hugging Face format.
+
+## Results
+
+| Model | Seed-TTS `test_zh` CER | CosyVoice3 `zero_shot_zh` CER | Comment |
+|-------|------------------------|------------------------------|---------|
+| CosyVoice2 LLM (official) | 1.45% | 4.08% | See the [paper](https://arxiv.org/abs/2412.10117) |
+| CosyVoice2 LLM + GRPO | 1.37% | **3.36%** | See the [decoding results](yuekai/official-cosyvoice-llm-grpo-aishell3), Hugging Face-format model |
+
+## Acknowledgement
+
+This work was inspired by the implementation in [ch-tts-llasa-rl-grpo](https://github.com/channel-io/ch-tts-llasa-rl-grpo).

examples/grpo/cosyvoice2/huggingface_to_pretrained.py

@@ -0,0 +1,71 @@
+
+# SPDX-FileCopyrightText: Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+"""
+python3 hf2pretrained.py --hf-cosyvoice2-llm-path /workspace/rl-exp/checkpoint-400 --output-path /workspace/CosyVoice2-0.5B/llm-new.pt
+"""
+from argparse import ArgumentParser
+import torch
+from safetensors import safe_open
+from transformers import AutoTokenizer
+
+
+def get_args():
+    parser = ArgumentParser()
+
+    parser.add_argument(
+        "--hf-cosyvoice2-llm-path",
+        type=str,
+        default=None,
+        help="The RL trained CosyVoice2 model path in HuggingFace format",
+    )
+    parser.add_argument(
+        "--output-path",
+        type=str,
+        default="./llm.pt",
+        help="The path to save the llm.pt",
+    )
+    args = parser.parse_args()
+    return args
+
+
+if __name__ == "__main__":
+    args = get_args()
+
+    tokenizer = AutoTokenizer.from_pretrained(args.hf_cosyvoice2_llm_path)
+    speech_start_idx = tokenizer.convert_tokens_to_ids("<|s_0|>")
+    cosyvoice2_token_size = 6561 + 3
+    llm_embedding_vocab_size = 2
+
+    hf_tensors = {}
+    with safe_open(f"{args.hf_cosyvoice2_llm_path}/model.safetensors", framework="pt", device="cpu") as f:
+        for k in f.keys():
+            if k.startswith("lm_head.bias"):
+                # RL trained model disable bias for lm_head
+                continue
+            new_k = "llm.model." + k
+            hf_tensors[new_k] = f.get_tensor(k)
+            if k.startswith("lm_head"):
+                hf_tensors["llm_decoder.weight"] = f.get_tensor(k)[speech_start_idx:speech_start_idx + cosyvoice2_token_size]
+                hf_tensors["llm_decoder.bias"] = torch.zeros_like(hf_tensors["llm_decoder.weight"][:, 0])
+            if k.startswith("model.embed_tokens"):
+                hf_tensors["speech_embedding.weight"] = f.get_tensor(k)[speech_start_idx:speech_start_idx + cosyvoice2_token_size]
+                hf_tensors["llm_embedding.weight"] = f.get_tensor(k)[speech_start_idx + cosyvoice2_token_size:speech_start_idx + cosyvoice2_token_size + llm_embedding_vocab_size]
+
+        # use tie_word_embeddings=True
+        hf_tensors["llm.model.model.embed_tokens.weight"] = hf_tensors["llm.model.model.embed_tokens.weight"][:151936]
+        hf_tensors["llm.model.lm_head.weight"] = hf_tensors["llm.model.model.embed_tokens.weight"]
+
+    torch.save(hf_tensors, args.output_path)

examples/grpo/cosyvoice2/infer_dataset.py

@@ -0,0 +1,397 @@
+# SPDX-FileCopyrightText: Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+""" Example Usage
+dataset=zero_shot_zh
+output_dir=./outputs_rl_aishell3_step${step}_${dataset}_jit_trt_fp16_reward_tts
+
+token2wav_path=/workspace/CosyVoice2-0.5B
+CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
+torchrun --nproc_per_node=8 \
+    infer_dataset.py \
+    --output-dir $output_dir \
+    --llm-model-name-or-path $llm_path/merged_hf_model \
+    --token2wav-path $token2wav_path \
+    --split-name ${dataset} || exit 1
+"""
+
+import argparse
+import json
+import os
+import sys
+from pathlib import Path
+
+import torch
+import torch.distributed as dist
+import torch.nn.functional as F
+import torchaudio
+from cosyvoice.cli.cosyvoice import CosyVoice2
+from cosyvoice.utils.file_utils import load_wav
+from datasets import load_dataset
+from transformers import AutoTokenizer, AutoModelForCausalLM
+from torch.utils.data import DataLoader, Dataset, DistributedSampler
+from tqdm import tqdm
+import soundfile as sf
+import s3tokenizer
+from functools import partial
+
+sys.path.append("/workspace/CosyVoice/third_party/Matcha-TTS")
+try:
+    torch.multiprocessing.set_start_method("spawn")
+except RuntimeError:
+    pass
+
+
+TEMPLATE = "{% for message in messages %}{%- if message['role'] == 'user' %}{{- '<|im_start|>' + message['role'] + '\n' + 'Convert the text to speech: ' + message['content'] + '<|im_end|>\n'}}{%- elif message['role'] == 'assistant' %}{{- '<|im_start|>' + message['role'] + '\n' + '<|SPEECH_GENERATION_START|>' + message['content']}}{%- endif %}{%- endfor %}"  # noqa: E501
+
+
+def audio_decode_cosyvoice2(
+    audio_tokens, prompt_text, prompt_speech_16k, codec_decoder
+):
+    """
+    Generate audio from tokens with optional tone and prompt embedding.
+    """
+    model_inputs_dict = codec_decoder.frontend.frontend_zero_shot(
+        "empty", prompt_text, prompt_speech_16k, 24000
+    )
+    tts_mel, _ = codec_decoder.model.flow.inference(
+        token=audio_tokens.to(codec_decoder.model.device),
+        token_len=torch.tensor([audio_tokens.shape[1]], dtype=torch.int32).to(
+            codec_decoder.model.device
+        ),
+        prompt_token=model_inputs_dict["flow_prompt_speech_token"].to(
+            codec_decoder.model.device
+        ),
+        prompt_token_len=torch.tensor(
+            [model_inputs_dict["flow_prompt_speech_token_len"]], dtype=torch.int32
+        ).to(codec_decoder.model.device),
+        prompt_feat=model_inputs_dict["prompt_speech_feat"].to(
+            codec_decoder.model.device
+        ),
+        prompt_feat_len=model_inputs_dict["prompt_speech_feat_len"].to(
+            codec_decoder.model.device
+        ),
+        embedding=model_inputs_dict["flow_embedding"].to(codec_decoder.model.device),
+        finalize=True,
+    )
+
+    audio_hat, _ = codec_decoder.model.hift.inference(
+        speech_feat=tts_mel, cache_source=torch.zeros(1, 1, 0)
+    )
+
+    return audio_hat
+
+
+def extract_speech_ids(speech_tokens_str):
+    """Extract speech IDs from token strings like <|s_23456|>"""
+    speech_ids = []
+    for token_str in speech_tokens_str:
+        if token_str.startswith('<|s_') and token_str.endswith('|>'):
+            num_str = token_str[4:-2]
+            num = int(num_str)
+            speech_ids.append(num)
+        else:
+            print(f"Unexpected token: {token_str}")
+    return speech_ids
+
+
+def convert_cosy2_tokens_to_speech_id_str(cosy2_tokens):
+    """Convert CosyVoice2 tokens to speech IDs string like <|s_23456|>"""
+    speech_id_str = ""
+    for token in cosy2_tokens:
+        speech_id_str += f"<|s_{token}|>"
+    return speech_id_str
+
+
+def get_args():
+    parser = argparse.ArgumentParser(description="Speech generation using LLM + CosyVoice2")
+    parser.add_argument(
+        "--split-name",
+        type=str,
+        default="wenetspeech4tts",
+        help="huggingface dataset split name, see yuekai/CV3-Eval, yuekai/seed_tts_cosy2",
+    )
+    parser.add_argument(
+        "--output-dir", required=True, type=str, help="dir to save result"
+    )
+    parser.add_argument(
+        "--batch-size",
+        default=1,
+        type=int,
+        help="batch size (per-device) for inference",
+    )
+    parser.add_argument(
+        "--num-workers", type=int, default=1, help="workers for dataloader"
+    )
+    parser.add_argument(
+        "--prefetch", type=int, default=5, help="prefetch for dataloader"
+    )
+    parser.add_argument(
+        "--llm-model-name-or-path",
+        required=True,
+        type=str,
+        help="LLM model path (includes both model and tokenizer)",
+    )
+    parser.add_argument(
+        "--token2wav-path",
+        required=True,
+        type=str,
+        help="CosyVoice2 token2wav model path",
+    )
+    parser.add_argument(
+        "--prompt-text",
+        type=str,
+        default=None,
+        help="The prompt text for CosyVoice2",
+    )
+    parser.add_argument(
+        "--prompt-speech-path",
+        type=str,
+        default=None,
+        help="The path to the prompt speech for CosyVoice2",
+    )
+    parser.add_argument(
+        "--top-p",
+        type=float,
+        default=0.95,
+        help="top p for sampling",
+    )
+    parser.add_argument(
+        "--temperature",
+        type=float,
+        default=0.8,
+        help="temperature for sampling",
+    )
+    parser.add_argument(
+        "--top-k",
+        type=int,
+        default=50,
+        help="top k for sampling",
+    )
+    args = parser.parse_args()
+    return args
+
+
+def data_collator(batch, tokenizer, s3_tokenizer):
+    """Simplified data collator for batch_size=1 processing"""
+    target_sample_rate = 16000  # CosyVoice2 uses 16kHz for prompt audio
+    device = s3_tokenizer.device if s3_tokenizer is not None else torch.device("cpu")
+    input_ids_list, prompt_audio_list, prompt_text_list = [], [], []
+    mels, prompt_audio_cosy2tokens_list = [], []
+    for item in batch:
+        prompt_text, target_text = (
+            item["prompt_text"],
+            item["target_text"],
+        )
+        prompt_text_list.append(prompt_text)
+        # Combine prompt and target text
+        full_text = prompt_text + target_text
+
+        # get prompt audio for CosyVoice2 (convert to 16kHz)
+        ref_audio_org, ref_sr = (
+            item["prompt_audio"]["array"],
+            item["prompt_audio"]["sampling_rate"],
+        )
+        ref_audio_org = torch.from_numpy(ref_audio_org).float().unsqueeze(0)
+        # ref_audio_org = ref_audio_org.mean(dim=0, keepdim=True)
+        print(ref_audio_org.shape)
+
+        if ref_sr != target_sample_rate:
+            resampler = torchaudio.transforms.Resample(ref_sr, target_sample_rate)
+            ref_audio = resampler(ref_audio_org)
+        else:
+            ref_audio = ref_audio_org
+
+        prompt_audio_list.append(ref_audio)
+
+        if "prompt_audio_cosy2_tokens" in item:
+            prompt_audio_cosy2tokens = item["prompt_audio_cosy2_tokens"]
+            prompt_audio_cosy2tokens_list.append(prompt_audio_cosy2tokens)
+        else:
+            # convert to float first
+            mels.append(s3tokenizer.log_mel_spectrogram(ref_audio.squeeze(0)))
+
+    if len(mels) > 0:
+        mels, mels_lens = s3tokenizer.padding(mels)
+        codes, codes_lens = s3_tokenizer.quantize(mels.to(device), mels_lens.to(device))
+        for i in range(len(codes)):
+            prompt_audio_cosy2tokens_list.append(codes[i, :codes_lens[i].item()])
+    for prompt_audio_cosy2tokens in prompt_audio_cosy2tokens_list:
+        prompt_audio_cosy2_id_str = convert_cosy2_tokens_to_speech_id_str(prompt_audio_cosy2tokens)
+        # Create chat template for LLM generation
+        chat = [
+            {"role": "user", "content": full_text},
+            {"role": "assistant", "content": prompt_audio_cosy2_id_str}
+        ]
+        if 'system' in tokenizer.chat_template:
+            tokenizer.chat_template = TEMPLATE
+        input_ids = tokenizer.apply_chat_template(
+            chat,
+            tokenize=True,
+            return_tensors='pt',
+            continue_final_message=True
+        )
+        input_ids_list.append(input_ids.squeeze(0))
+
+    # For batch_size=1, no need to pad
+    if len(input_ids_list) == 1:
+        input_ids = input_ids_list[0].unsqueeze(0)
+    else:
+        # Handle batch > 1 if needed
+        max_len = max([len(input_ids) for input_ids in input_ids_list])
+        input_ids_list = [
+            torch.cat([torch.full((max_len - len(input_ids),), tokenizer.pad_token_id), input_ids])
+            for input_ids in input_ids_list
+        ]
+        input_ids = torch.stack(input_ids_list)
+
+    ids = [item["id"] for item in batch]
+
+    return {
+        "input_ids": input_ids,
+        "ids": ids,
+        "prompt_text": prompt_text_list,
+        "prompt_audio_list": prompt_audio_list,
+    }
+
+
+def init_distributed():
+    world_size = int(os.environ.get("WORLD_SIZE", 1))
+    local_rank = int(os.environ.get("LOCAL_RANK", 0))
+    rank = int(os.environ.get("RANK", 0))
+    print(
+        "Inference on multiple gpus, this gpu {}".format(local_rank)
+        + ", rank {}, world_size {}".format(rank, world_size)
+    )
+    torch.cuda.set_device(local_rank)
+    dist.init_process_group("nccl")
+    return world_size, local_rank, rank
+
+
+def main():
+    args = get_args()
+    os.makedirs(args.output_dir, exist_ok=True)
+
+    assert torch.cuda.is_available()
+    world_size, local_rank, rank = init_distributed()
+    device = torch.device(f"cuda:{local_rank}")
+
+    # Load LLM model and tokenizer directly
+    tokenizer = AutoTokenizer.from_pretrained(args.llm_model_name_or_path)
+    model = AutoModelForCausalLM.from_pretrained(args.llm_model_name_or_path)
+    model.eval()
+    model.to(device)
+
+    cosyvoice_codec = CosyVoice2(
+        args.token2wav_path, load_jit=True, load_trt=True, fp16=True
+    )
+    if args.prompt_speech_path:
+        prompt_speech_16k = load_wav(args.prompt_speech_path, 16000)
+    else:
+        prompt_speech_16k = None
+    s3_tokenizer = s3tokenizer.load_model("speech_tokenizer_v2_25hz").to(device) if 'zero' in args.split_name else None
+    dataset_name = "yuekai/CV3-Eval" if 'zero' in args.split_name else "yuekai/seed_tts_cosy2"
+    dataset = load_dataset(
+        dataset_name,
+        split=args.split_name,
+        trust_remote_code=True,
+    )
+
+    sampler = DistributedSampler(dataset, num_replicas=world_size, rank=rank)
+
+    dataloader = DataLoader(
+        dataset,
+        batch_size=args.batch_size,
+        sampler=sampler,
+        shuffle=False,
+        num_workers=args.num_workers,
+        prefetch_factor=args.prefetch,
+        collate_fn=partial(data_collator, tokenizer=tokenizer, s3_tokenizer=s3_tokenizer),
+    )
+
+    total_steps = len(dataset)
+
+    if rank == 0:
+        progress_bar = tqdm(total=total_steps, desc="Processing", unit="wavs")
+
+    for batch in dataloader:
+        with torch.no_grad():
+            input_ids = batch["input_ids"].to(device)
+
+            # Generate speech tokens using LLM
+            outputs = model.generate(
+                input_ids,
+                max_new_tokens=2048,  # Max length for generation
+                do_sample=True,
+                top_p=args.top_p,
+                temperature=args.temperature,
+                top_k=args.top_k,
+            )
+
+            # Process each sample in the batch
+            for i in range(len(batch["ids"])):
+                # Extract generated tokens (excluding input)
+                input_length = input_ids[i].shape[0]
+                generated_ids = outputs[i][input_length:-1]  # Remove last token if needed
+                speech_tokens_str = tokenizer.batch_decode(generated_ids, skip_special_tokens=True)
+
+                # Extract speech IDs from token strings like <|s_23456|>
+                speech_ids = extract_speech_ids(speech_tokens_str)
+
+                if len(speech_ids) == 0:
+                    print(f"Warning: No speech tokens generated for sample {batch['ids'][i]}, skipping")
+                    continue
+
+                # Convert to tensor for CosyVoice2
+                audio_tokens = torch.tensor(speech_ids, dtype=torch.long, device=device).unsqueeze(0)
+
+                if args.prompt_text is not None:
+                    current_prompt_text = args.prompt_text
+                    current_prompt_audio = prompt_speech_16k
+                else:
+                    current_prompt_text = batch["prompt_text"][i]
+                    current_prompt_audio = batch["prompt_audio_list"][i]
+
+                if current_prompt_audio is not None:
+                    # Generate audio using CosyVoice2
+                    audio_hat = audio_decode_cosyvoice2(
+                        audio_tokens,
+                        current_prompt_text,
+                        current_prompt_audio,
+                        cosyvoice_codec,
+                    )
+
+                    # Convert to numpy and save
+                    generated_wave = audio_hat.squeeze(0).cpu().numpy()
+                    target_sample_rate = 24000
+
+                    utt = batch["ids"][i]
+                    sf.write(f"{args.output_dir}/{utt}.wav", generated_wave, target_sample_rate)
+
+                    print(f"Generated audio for sample {utt} with {len(speech_ids)} tokens")
+                else:
+                    print(f"Warning: No prompt audio available for sample {batch['ids'][i]}, skipping")
+
+        if rank == 0:
+            progress_bar.update(world_size * len(batch["ids"]))
+
+    if rank == 0:
+        progress_bar.close()
+
+    dist.barrier()
+    dist.destroy_process_group()
+
+
+if __name__ == "__main__":
+    main()

examples/grpo/cosyvoice2/prepare_data.py

@@ -0,0 +1,86 @@
+# Copyright 2024 Bytedance Ltd. and/or its affiliates
+#
+# 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.
+"""
+Preprocess the Text to Speech dataset to parquet format
+"""
+
+import argparse
+import os
+import re
+
+import datasets
+
+from verl.utils.hdfs_io import copy, makedirs
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--train_file", required=True, help="Path to training JSON/JSONL file")
+    parser.add_argument("--test_file", required=True, help="Path to test JSON/JSONL file")
+    parser.add_argument("--local_dir", default=None, required=True)
+    parser.add_argument("--hdfs_dir", default=None)
+
+    args = parser.parse_args()
+
+    # Load datasets from local JSON files
+    train_dataset = datasets.load_dataset("json", data_files=args.train_file)['train']
+    test_dataset = datasets.load_dataset("json", data_files=args.test_file)['train']
+
+    # add a row to each data item that represents a unique id
+    def make_map_fn(split):
+        def process_fn(example, idx):
+            text = example.pop("text")
+
+            # use cosyvoice2 official huggingface compatible checkpoint template
+            question = text
+            answer = ""
+
+            data = {
+                "data_source": f"{args.train_file}_{args.test_file}",  # Use file names as data source
+                "prompt": [
+                    {
+                        "role": "user",
+                        "content": question,
+                    },
+                    {
+                        "role": "assistant",
+                        "content": answer,
+                    },
+                ],
+                "ability": "text-to-speech",
+                "reward_model": {"style": "rule", "ground_truth": text},
+                "extra_info": {
+                    "split": split,
+                    "index": idx,
+                    "text": text,
+                },
+            }
+            return data
+
+        return process_fn
+
+    train_dataset = train_dataset.map(function=make_map_fn("train"), with_indices=True)
+    test_dataset = test_dataset.map(function=make_map_fn("test"), with_indices=True)
+
+    local_dir = args.local_dir
+    hdfs_dir = args.hdfs_dir
+
+    print(train_dataset)
+    print(test_dataset)
+    train_dataset.to_parquet(os.path.join(local_dir, "train.parquet"))
+    test_dataset.to_parquet(os.path.join(local_dir, "test.parquet"))
+
+    if hdfs_dir is not None:
+        makedirs(hdfs_dir)
+
+        copy(src=local_dir, dst=hdfs_dir)

examples/grpo/cosyvoice2/pretrained_to_huggingface.py

@@ -0,0 +1,133 @@
+# SPDX-FileCopyrightText: Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+"""
+Usage: Instruct TTS
+  python3 infer.py \
+    --token2wav-path /workspace/CosyVoice2-0.5B \
+    --prompt-text "吃燕窝就选燕之屋，本节目由26年专注高品质燕窝的燕之屋冠名播出。豆奶牛奶换着喝，营养更均衡，本节目由豆本豆豆奶特约播出。" \
+    --prompt-speech-path ./assets/prompt_audio.wav \
+    --model-path ./transformers_cosyvoice2_llm \
+    --input-text "用四川话说<|endofprompt|>扁担长，板凳宽，扁担绑在板凳上。吃葡萄不吐葡萄皮，不吃葡萄倒吐葡萄皮。"
+"""
+from cosyvoice.cli.cosyvoice import CosyVoice2
+import sys
+from argparse import ArgumentParser
+from transformers import AutoTokenizer, AutoModelForCausalLM
+import torch
+
+sys.path.append("/workspace/CosyVoice/third_party/Matcha-TTS")
+
+
+def get_args():
+    parser = ArgumentParser()
+
+    parser.add_argument(
+        "--pretrained-cosyvoice2-path",
+        type=str,
+        default="/workspace/CosyVoice2-0.5B",
+        help="Token2Wav path, default to %(default)r",
+    )
+    parser.add_argument(
+        "--save-path",
+        type=str,
+        default='./transformers_cosyvoice2_llm',
+        help="The path to save the model",
+    )
+    args = parser.parse_args()
+    return args
+
+
+if __name__ == "__main__":
+    args = get_args()
+    cosy2_model = CosyVoice2(
+        args.pretrained_cosyvoice2_path, load_jit=False, load_trt=False, fp16=False
+    )
+
+    llm = cosy2_model.model.llm.llm.model
+
+    speech_embedding = cosy2_model.model.llm.speech_embedding
+    llm_decoder = cosy2_model.model.llm.llm_decoder
+    llm_embedding = cosy2_model.model.llm.llm_embedding
+
+    tokenizer = AutoTokenizer.from_pretrained(f"{args.pretrained_cosyvoice2_path}/CosyVoice-BlankEN")
+    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]"
+        ]
+    }
+    tokenizer.add_special_tokens(special_tokens)
+
+    original_tokenizer_vocab_size = len(tokenizer)
+    cosyvoice2_token_size = 6561
+    new_tokens = [f"<|s_{i}|>" for i in range(cosyvoice2_token_size)] + [
+        "<|eos1|>", "<|eos2|>", "<|eos3|>", "<|sos|>", "<|task_id|>"
+    ]
+    num_added_tokens = tokenizer.add_tokens(new_tokens)
+
+    llm.resize_token_embeddings(len(tokenizer), pad_to_multiple_of=128)
+    vocab_size = llm.get_input_embeddings().weight.shape[0]
+
+    feature_size = speech_embedding.embedding_dim
+    new_lm_head = torch.nn.Linear(in_features=feature_size, out_features=vocab_size, bias=True)
+
+    with torch.no_grad():
+        # set the weight and bias of the new lm_head to 0
+        new_lm_head.weight.data.zero_()
+        # make bias value -inf
+        new_lm_head.bias.data.fill_(-float('inf'))
+        new_lm_head.weight[original_tokenizer_vocab_size:original_tokenizer_vocab_size + cosyvoice2_token_size + 3] = llm_decoder.weight
+        new_lm_head.bias[original_tokenizer_vocab_size:original_tokenizer_vocab_size + cosyvoice2_token_size + 3] = llm_decoder.bias
+
+    llm.lm_head = new_lm_head
+    input_embeddings = llm.get_input_embeddings()
+
+    with torch.no_grad():
+        input_embeddings.weight[original_tokenizer_vocab_size:original_tokenizer_vocab_size + cosyvoice2_token_size + 3] = speech_embedding.weight
+        input_embeddings.weight[original_tokenizer_vocab_size + cosyvoice2_token_size + 3:original_tokenizer_vocab_size + cosyvoice2_token_size + 3 + 2] = llm_embedding.weight
+
+    eos_token_ids = [original_tokenizer_vocab_size + cosyvoice2_token_size,
+                     original_tokenizer_vocab_size + cosyvoice2_token_size + 1,
+                     original_tokenizer_vocab_size + cosyvoice2_token_size + 2]
+    llm.generation_config.eos_token_id = eos_token_ids
+    llm.generation_config.temperature = 1.0
+    llm.generation_config.top_p = 0.8
+    llm.generation_config.top_k = 25
+
+    llm.config.eos_token_id = original_tokenizer_vocab_size + cosyvoice2_token_size
+    llm.config.vocab_size = vocab_size
+    llm.config.tie_word_embeddings = False
+    llm.config.use_bias = True
+    llm.to(torch.bfloat16)
+    llm.save_pretrained(args.save_path)
+
+    TEMPLATE = (
+        "{%- for message in messages %}"
+        "{%- if message['role'] == 'user' %}"
+        "{{- '<|sos|>' + message['content'] + '<|task_id|>' }}"
+        "{%- elif message['role'] == 'assistant' %}"
+        "{{- message['content']}}"
+        "{%- endif %}"
+        "{%- endfor %}"
+    )
+    tokenizer.chat_template = TEMPLATE
+    tokenizer.save_pretrained(args.save_path)

examples/grpo/cosyvoice2/requirements.txt

@@ -0,0 +1,31 @@
+conformer==0.3.2
+diffusers==0.29.0
+gdown==5.1.0
+gradio
+hydra-core==1.3.2
+HyperPyYAML==1.2.2
+inflect==7.3.1
+librosa==0.10.2
+lightning==2.2.4
+matplotlib==3.7.5
+modelscope==1.15.0
+networkx==3.1
+omegaconf==2.3.0
+onnx==1.16.0
+onnxruntime-gpu==1.18.0
+protobuf==4.25
+pydantic==2.7.0
+pyworld==0.3.4
+rich==13.7.1
+soundfile==0.12.1
+tensorboard==2.14.0
+wget==3.2
+WeTextProcessing==1.0.3
+s3tokenizer
+tensorrt
+sherpa_onnx
+jiwer
+zhon
+numpy==1.25.2
+pypinyin
+openai-whisper

examples/grpo/cosyvoice2/reward_tts.py

@@ -0,0 +1,233 @@
+# SPDX-FileCopyrightText: Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+"""
+Reward calculation for CosyVoice2-0.5B.
+"""
+
+from __future__ import annotations
+
+import re
+import json
+import time
+import argparse
+from typing import List
+
+import numpy as np
+import requests
+
+
+REWARD_SERVER_URL = "http://localhost:8000/v2/models/token2wav_asr/infer"
+
+
+def _parse_ids(token_str: str) -> List[int]:
+    return [int(t) for t in re.findall(r"<\|s_(\d+)\|>", token_str)]
+
+
+def _remote_reward(tokens: List[int], ground_truth: str, timeout: float = 200.0) -> float:
+    """Send token IDs and ground-truth text to the Triton server and get reward."""
+
+    tokens_arr = np.array(tokens, dtype=np.int32).reshape(1, -1)
+    lens_arr = np.array([[tokens_arr.shape[1]]], dtype=np.int32)
+
+    gt_arr = np.array([ground_truth.encode("utf-8")], dtype=object)
+
+    payload = {
+        "inputs": [
+            {
+                "name": "TOKENS",
+                "shape": list(tokens_arr.shape),
+                "datatype": "INT32",
+                "data": tokens_arr.tolist(),
+            },
+            {
+                "name": "TOKEN_LENS",
+                "shape": list(lens_arr.shape),
+                "datatype": "INT32",
+                "data": lens_arr.tolist(),
+            },
+            {
+                "name": "GT_TEXT",
+                "shape": [1, 1],
+                "datatype": "BYTES",
+                "data": [ground_truth],
+            },
+        ]
+    }
+    rsp = requests.post(
+        REWARD_SERVER_URL,
+        headers={"Content-Type": "application/json"},
+        json=payload,
+        timeout=timeout,
+        verify=False,
+        params={"request_id": "0"},
+    )
+    rsp.raise_for_status()
+    result = rsp.json()
+
+    try:
+        # Reward is returned as the first output
+        return float(result["outputs"][0]["data"][0])
+    except (KeyError, IndexError, TypeError):
+        return 0.0
+
+
+def compute_score(
+    data_source: str,
+    solution_str: str,
+    ground_truth: str,
+    extra_info: dict | None = None,
+    *,
+    debug_dump: bool = False,
+) -> float:
+    """Return reward in [0, 1] using the Triton ASR service.
+
+    The reward is based on the pinyin-level WER between the ASR transcript
+    produced from *solution_str* and the provided *ground_truth* text.
+    """
+
+    # Decode token IDs
+    ids = _parse_ids(solution_str)
+
+    # Query remote server for reward
+    try:
+        reward = _remote_reward(ids, ground_truth)
+    except Exception as e:
+        reward = 0.0
+
+    if debug_dump:
+        print(
+            f"\033[92m[{data_source}] Remote reward: {reward:.4f}\033[0m"
+        )
+
+    return reward
+
+
+# CLI quick test
+if __name__ == "__main__":
+    import sys
+
+    def get_args():
+        """Parse command line arguments."""
+        parser = argparse.ArgumentParser(
+            description="Test TTS CER scoring with data from JSONL file",
+            formatter_class=argparse.ArgumentDefaultsHelpFormatter
+        )
+
+        parser.add_argument(
+            "--input", "-i",
+            type=str,
+            default="data/emilia_zh-cosy-tiny-test.jsonl",
+            help="Path to input JSONL file"
+        )
+
+        parser.add_argument(
+            "--max-samples", "-n",
+            type=int,
+            default=None,
+            help="Maximum number of samples to process (default: all)"
+        )
+
+        parser.add_argument(
+            "--no-interactive",
+            action="store_true",
+            help="Run in non-interactive mode (process all samples without prompts)"
+        )
+
+        parser.add_argument(
+            "--debug",
+            action="store_true",
+            help="Enable debug mode"
+        )
+
+        return parser.parse_args()
+
+    def load_jsonl(file_path: str):
+        """Load data from jsonl file."""
+        data = []
+        with open(file_path, 'r', encoding='utf-8') as f:
+            for line in f:
+                data.append(json.loads(line.strip()))
+        return data
+
+    def code_to_solution_str(code_list: List[int]) -> str:
+        """Convert code list to solution string format."""
+        return ''.join([f"<|s_{code}|>" for code in code_list])
+
+    # Parse command line arguments
+    args = get_args()
+
+    try:
+        # Load data from jsonl file
+        print(f"Loading data from: {args.input}")
+        data_list = load_jsonl(args.input)
+        print(f"Loaded {len(data_list)} samples")
+
+        # Limit samples if specified
+        if args.max_samples is not None:
+            data_list = data_list[:args.max_samples]
+            print(f"Processing first {len(data_list)} samples (limited by --max-samples)")
+
+        # Process each sample
+        begin_time = time.time()
+        for i, sample in enumerate(data_list):
+            print(f"\n--- Sample {i+1}/{len(data_list)} ---")
+            print(f"Index: {sample.get('index', 'unknown')}")
+            print(f"Text: {sample['text']}")
+
+            # Extract required fields
+            code_list = sample['code']
+            ground_truth = sample['text']
+            data_source = sample.get('index', f'sample_{i}')  # Use index as data_source
+
+            # Convert code list to solution string
+            solution_str = code_to_solution_str(code_list)
+            print(f"Solution tokens: {len(code_list)} tokens")
+            if args.debug:
+                print(f"Solution string: {solution_str}")
+            else:
+                print(f"Solution string preview: {solution_str[:100]}..." if len(solution_str) > 100 else f"Solution string: {solution_str}")
+
+            # Call compute_score function
+            try:
+                score = compute_score(
+                    data_source=data_source,
+                    solution_str=solution_str,
+                    ground_truth=ground_truth,
+                    extra_info=None,
+                    debug_dump=args.debug
+                )
+                print(f"Final Score: {score:.4f}")
+            except Exception as e:
+                print(f"Error computing score: {e}")
+
+            # Ask user if they want to continue (for interactive mode)
+            if not args.no_interactive and i < len(data_list) - 1:
+                try:
+                    response = input("\nPress Enter to continue or 'q' to quit: ").strip().lower()
+                    if response == 'q':
+                        break
+                except KeyboardInterrupt:
+                    print("\nStopped by user")
+                    break
+
+        print(f"\nProcessed {min(i+1, len(data_list))} samples")
+        end_time = time.time()
+        print(f"Time taken: {end_time - begin_time} seconds")
+    except FileNotFoundError:
+        print(f"Error: File not found - {args.input}")
+        print("Please check the file path or use --input to specify correct path")
+        print("Run with --help for usage information")
+    except Exception as e:
+        print(f"Error: {e}")

examples/grpo/cosyvoice2/run.sh

@@ -0,0 +1,159 @@
+#!/usr/bin/env bash
+
+set -eou pipefail
+
+stage=-1
+stop_stage=4
+
+log() {
+  # This function is from espnet
+  local fname=${BASH_SOURCE[1]##*/}
+  echo -e "$(date '+%Y-%m-%d %H:%M:%S') (${fname}:${BASH_LINENO[0]}:${FUNCNAME[1]}) $*"
+}
+
+export PYTHONPATH=/workspace/CosyVoice
+model_scope_model_path=./CosyVoice2-0.5B
+sft_model_path=./transformers_cosyvoice2_llm
+
+if [ $stage -le -2 ] && [ $stop_stage -ge -2 ]; then
+  log "stage -2: install dependencies locally if pre-built docker image is not available"
+  conda create -n cosyvoice2 python=3.10 -y
+  conda activate cosyvoice2
+    # install verl
+  git clone https://github.com/yuekaizhang/verl.git -b thread
+  cd verl
+  USE_MEGATRON=0 bash scripts/install_vllm_sglang_mcore.sh
+  pip install --no-deps -e .
+  cd -
+  # install requirements
+  pip install -r requirements.txt
+  pip install -U nvidia-pytriton
+  git clone https://github.com/yuekaizhang/PytritonSenseVoice.git && cd PytritonSenseVoice && pip install -e .
+fi
+
+if [ $stage -le -1 ] && [ $stop_stage -ge -1 ]; then
+  log "stage -1: download official CosyVoice2-0.5B LLM model and convert to huggingface compatible checkpoint"
+  modelscope download --model iic/CosyVoice2-0.5B --local_dir $model_scope_model_path
+  python3 pretrained_to_huggingface.py \
+    --pretrained-cosyvoice2-path $model_scope_model_path \
+    --save-path $sft_model_path
+
+  # Or, you could use the following command to download the huggingface compatible checkpoint
+  # huggingface-cli download --local-dir $sft_model_path yuekai/cosyvoice2_llm
+
+  # Note: we remove the lm_head's bias to make it compatible with the Qwen2.5-0.5B model in Transformers.
+fi
+
+data_dir=data/parquet_aishell3
+if [ $stage -le 0 ] && [ $stop_stage -ge 0 ]; then
+  log "stage 0: prepare data into verl format"
+  mkdir -p $data_dir
+  wget -O data/aishell-3.jsonl https://huggingface.co/datasets/SparkAudio/voxbox/resolve/main/metadata/aishell-3.jsonl
+  # total 88035 samples
+  head -n 80000 data/aishell-3.jsonl > data/train.jsonl
+  tail -n 100 data/aishell-3.jsonl > data/test.jsonl
+  python prepare_data.py \
+    --train_file data/train.jsonl \
+    --test_file data/test.jsonl \
+    --local_dir $data_dir
+fi
+
+if [ $stage -le 1 ] && [ $stop_stage -ge 1 ]; then
+  log "stage 1: start token2wav asr server for reward function"
+  python3 token2wav_asr_server.py --number-of-devices 8
+fi
+
+exp_name=official_llm_aishell3_grpo
+if [ $stage -le 2 ] && [ $stop_stage -ge 2 ]; then
+  log "stage 2: grpo train"
+  export CUDA_VISIBLE_DEVICES="0,1,2,3,4,5,6,7"
+  export MKL_SERVICE_FORCE_INTEL=TRUE
+  n_gpus_per_node=8
+  micro_batch_size=4
+  train_batch_size=32
+  python3 -m verl.trainer.main_ppo \
+      algorithm.adv_estimator=grpo \
+      data.train_files=$data_dir/train.parquet \
+      data.val_files=$data_dir/test.parquet \
+      data.train_batch_size=$train_batch_size \
+      data.max_prompt_length=1024 \
+      data.max_response_length=512 \
+      data.truncation='error' \
+      actor_rollout_ref.model.use_remove_padding=False \
+      actor_rollout_ref.model.path=$sft_model_path \
+      actor_rollout_ref.actor.optim.lr=1e-6 \
+      actor_rollout_ref.actor.ppo_mini_batch_size=32 \
+      actor_rollout_ref.actor.ppo_micro_batch_size_per_gpu=$micro_batch_size \
+      actor_rollout_ref.actor.use_kl_loss=False \
+      actor_rollout_ref.model.enable_gradient_checkpointing=True \
+      actor_rollout_ref.actor.fsdp_config.param_offload=False \
+      actor_rollout_ref.actor.fsdp_config.optimizer_offload=False \
+      actor_rollout_ref.rollout.log_prob_micro_batch_size_per_gpu=$micro_batch_size \
+      actor_rollout_ref.rollout.tensor_model_parallel_size=1 \
+      actor_rollout_ref.rollout.name=vllm \
+      actor_rollout_ref.rollout.gpu_memory_utilization=0.6 \
+      actor_rollout_ref.rollout.do_sample=true \
+      actor_rollout_ref.rollout.temperature=0.8 \
+      actor_rollout_ref.rollout.top_p=0.95 \
+      actor_rollout_ref.rollout.top_k=25 \
+      actor_rollout_ref.rollout.n=4 \
+      actor_rollout_ref.rollout.val_kwargs.do_sample=true \
+      actor_rollout_ref.rollout.val_kwargs.temperature=0.8 \
+      actor_rollout_ref.rollout.val_kwargs.top_p=0.95 \
+      actor_rollout_ref.rollout.val_kwargs.top_k=25 \
+      reward_model.reward_manager=prime \
+      custom_reward_function.path=reward_tts.py \
+      custom_reward_function.name=compute_score \
+      trainer.project_name='cosyvoice2_grpo' \
+      trainer.experiment_name=$exp_name \
+      trainer.logger=['console','wandb'] \
+      trainer.n_gpus_per_node=$n_gpus_per_node \
+      trainer.nnodes=1 \
+      trainer.save_freq=100 \
+      trainer.test_freq=100 \
+      trainer.resume_mode='auto' \
+      trainer.total_epochs=1 \
+      trainer.val_before_train=False
+fi
+
+steps=(100 200 300 400 500)
+for step in ${steps[@]}; do
+llm_path=./checkpoints/cosyvoice2_grpo/$exp_name/global_step_${step}
+if [ $stage -le 3 ] && [ $stop_stage -ge 3 ]; then
+  log "stage 3: merge the model"
+  python -m verl.model_merger merge \
+      --backend fsdp \
+      --local_dir $llm_path/actor \
+      --target_dir $llm_path/merged_hf_model || exit 1
+fi
+
+if [ $stage -le 4 ] && [ $stop_stage -ge 4 ]; then
+  log "stage 4: Test the model"
+  dataset=zero_shot_zh # from CosyVoice3 test set
+  # dataset=test_zh # from seed_tts test set
+  output_dir=./outputs_${exp_name}_${step}_${dataset}
+
+  token2wav_path=/workspace/CosyVoice2-0.5B
+  model_path=$llm_path/merged_hf_model
+
+  CUDA_VISIBLE_DEVICES=0,1,2,3,4,5,6,7 \
+  torchrun --nproc_per_node=8 \
+      infer_dataset.py \
+        --output-dir $output_dir \
+        --llm-model-name-or-path $model_path \
+        --token2wav-path $token2wav_path \
+        --split-name ${dataset} || exit 1
+
+  bash scripts/compute_wer.sh $output_dir ${dataset}
+fi
+done
+
+if [ $stage -le 5 ] && [ $stop_stage -ge 5 ]; then
+  log "stage 5: Convert the RL trained model to CosyVoice repo format"
+  python3 huggingface_to_pretrained.py \
+    --hf-cosyvoice2-llm-path $llm_path/merged_hf_model \
+    --output-path /workspace/CosyVoice2-0.5B/llm-new.pt
+  # You need to manually move the llm-new.pt to overwrite /workspace/CosyVoice2-0.5B/llm.pt
+  # However, we found that the RL trained model accuracy would slightly drop after this conversion.
+  # Please be careful or use the huggingface format inference code.
+fi

examples/grpo/cosyvoice2/scripts/compute_wer.sh

@@ -0,0 +1,33 @@
+wav_dir=$1
+wav_files=$(ls $wav_dir/*.wav)
+# if wav_files is empty, then exit
+if [ -z "$wav_files" ]; then
+    exit 1
+fi
+split_name=$2
+model_path=models/sherpa-onnx-paraformer-zh-2023-09-14
+
+if [ ! -d $model_path ]; then
+    pip install sherpa-onnx
+    wget -nc https://github.com/k2-fsa/sherpa-onnx/releases/download/asr-models/sherpa-onnx-paraformer-zh-2023-09-14.tar.bz2
+    mkdir models
+    tar xvf sherpa-onnx-paraformer-zh-2023-09-14.tar.bz2 -C models
+fi
+
+python3 scripts/offline-decode-files.py  \
+    --tokens=$model_path/tokens.txt \
+    --paraformer=$model_path/model.int8.onnx \
+    --num-threads=2 \
+    --decoding-method=greedy_search \
+    --debug=false \
+    --sample-rate=24000 \
+    --log-dir $wav_dir \
+    --feature-dim=80 \
+    --split-name $split_name \
+    --name sherpa_onnx \
+    $wav_files
+
+# python3 scripts/paraformer-pytriton-client.py  \
+#     --log-dir $wav_dir \
+#     --split-name $split_name \
+#     $wav_files

examples/grpo/cosyvoice2/scripts/offline-decode-files.py

@@ -0,0 +1,754 @@
+# Copyright (c)  2023 by manyeyes
+# Copyright (c)  2023  Xiaomi Corporation
+
+"""
+This file demonstrates how to use sherpa-onnx Python API to transcribe
+file(s) with a non-streaming model.
+
+(1) For paraformer
+
+    ./python-api-examples/offline-decode-files.py  \
+      --tokens=/path/to/tokens.txt \
+      --paraformer=/path/to/paraformer.onnx \
+      --num-threads=2 \
+      --decoding-method=greedy_search \
+      --debug=false \
+      --sample-rate=16000 \
+      --feature-dim=80 \
+      /path/to/0.wav \
+      /path/to/1.wav
+
+(2) For transducer models from icefall
+
+    ./python-api-examples/offline-decode-files.py  \
+      --tokens=/path/to/tokens.txt \
+      --encoder=/path/to/encoder.onnx \
+      --decoder=/path/to/decoder.onnx \
+      --joiner=/path/to/joiner.onnx \
+      --num-threads=2 \
+      --decoding-method=greedy_search \
+      --debug=false \
+      --sample-rate=16000 \
+      --feature-dim=80 \
+      /path/to/0.wav \
+      /path/to/1.wav
+
+(3) For CTC models from NeMo
+
+python3 ./python-api-examples/offline-decode-files.py \
+  --tokens=./sherpa-onnx-nemo-ctc-en-citrinet-512/tokens.txt \
+  --nemo-ctc=./sherpa-onnx-nemo-ctc-en-citrinet-512/model.onnx \
+  --num-threads=2 \
+  --decoding-method=greedy_search \
+  --debug=false \
+  ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/0.wav \
+  ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/1.wav \
+  ./sherpa-onnx-nemo-ctc-en-citrinet-512/test_wavs/8k.wav
+
+(4) For Whisper models
+
+python3 ./python-api-examples/offline-decode-files.py \
+  --whisper-encoder=./sherpa-onnx-whisper-base.en/base.en-encoder.int8.onnx \
+  --whisper-decoder=./sherpa-onnx-whisper-base.en/base.en-decoder.int8.onnx \
+  --tokens=./sherpa-onnx-whisper-base.en/base.en-tokens.txt \
+  --whisper-task=transcribe \
+  --num-threads=1 \
+  ./sherpa-onnx-whisper-base.en/test_wavs/0.wav \
+  ./sherpa-onnx-whisper-base.en/test_wavs/1.wav \
+  ./sherpa-onnx-whisper-base.en/test_wavs/8k.wav
+
+(5) For CTC models from WeNet
+
+python3 ./python-api-examples/offline-decode-files.py \
+  --wenet-ctc=./sherpa-onnx-zh-wenet-wenetspeech/model.onnx \
+  --tokens=./sherpa-onnx-zh-wenet-wenetspeech/tokens.txt \
+  ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/0.wav \
+  ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/1.wav \
+  ./sherpa-onnx-zh-wenet-wenetspeech/test_wavs/8k.wav
+
+(6) For tdnn models of the yesno recipe from icefall
+
+python3 ./python-api-examples/offline-decode-files.py \
+  --sample-rate=8000 \
+  --feature-dim=23 \
+  --tdnn-model=./sherpa-onnx-tdnn-yesno/model-epoch-14-avg-2.onnx \
+  --tokens=./sherpa-onnx-tdnn-yesno/tokens.txt \
+  ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_0_1_0_0_0_1.wav \
+  ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_1_0_0_0_1_0.wav \
+  ./sherpa-onnx-tdnn-yesno/test_wavs/0_0_1_0_0_1_1_1.wav
+
+Please refer to
+https://k2-fsa.github.io/sherpa/onnx/index.html
+to install sherpa-onnx and to download non-streaming pre-trained models
+used in this file.
+"""
+import argparse
+import time
+import wave
+from pathlib import Path
+from typing import List, Tuple, Dict, Iterable, TextIO, Union
+
+import numpy as np
+import sherpa_onnx
+import soundfile as sf
+from datasets import load_dataset
+import logging
+from collections import defaultdict
+import kaldialign
+from zhon.hanzi import punctuation
+import string
+punctuation_all = punctuation + string.punctuation
+Pathlike = Union[str, Path]
+
+
+def remove_punctuation(text: str) -> str:
+    for x in punctuation_all:
+        if x == '\'':
+            continue
+        text = text.replace(x, '')
+    return text
+
+
+def store_transcripts(
+    filename: Pathlike, texts: Iterable[Tuple[str, str, str]], char_level: bool = False
+) -> None:
+    """Save predicted results and reference transcripts to a file.
+
+    Args:
+      filename:
+        File to save the results to.
+      texts:
+        An iterable of tuples. The first element is the cur_id, the second is
+        the reference transcript and the third element is the predicted result.
+        If it is a multi-talker ASR system, the ref and hyp may also be lists of
+        strings.
+    Returns:
+      Return None.
+    """
+    with open(filename, "w", encoding="utf8") as f:
+        for cut_id, ref, hyp in texts:
+            if char_level:
+                ref = list("".join(ref))
+                hyp = list("".join(hyp))
+            print(f"{cut_id}:\tref={ref}", file=f)
+            print(f"{cut_id}:\thyp={hyp}", file=f)
+
+
+def write_error_stats(
+    f: TextIO,
+    test_set_name: str,
+    results: List[Tuple[str, str]],
+    enable_log: bool = True,
+    compute_CER: bool = False,
+    sclite_mode: bool = False,
+) -> float:
+    """Write statistics based on predicted results and reference transcripts.
+
+    It will write the following to the given file:
+
+        - WER
+        - number of insertions, deletions, substitutions, corrects and total
+          reference words. For example::
+
+              Errors: 23 insertions, 57 deletions, 212 substitutions, over 2606
+              reference words (2337 correct)
+
+        - The difference between the reference transcript and predicted result.
+          An instance is given below::
+
+            THE ASSOCIATION OF (EDISON->ADDISON) ILLUMINATING COMPANIES
+
+          The above example shows that the reference word is `EDISON`,
+          but it is predicted to `ADDISON` (a substitution error).
+
+          Another example is::
+
+            FOR THE FIRST DAY (SIR->*) I THINK
+
+          The reference word `SIR` is missing in the predicted
+          results (a deletion error).
+      results:
+        An iterable of tuples. The first element is the cut_id, the second is
+        the reference transcript and the third element is the predicted result.
+      enable_log:
+        If True, also print detailed WER to the console.
+        Otherwise, it is written only to the given file.
+    Returns:
+      Return None.
+    """
+    subs: Dict[Tuple[str, str], int] = defaultdict(int)
+    ins: Dict[str, int] = defaultdict(int)
+    dels: Dict[str, int] = defaultdict(int)
+
+    # `words` stores counts per word, as follows:
+    #   corr, ref_sub, hyp_sub, ins, dels
+    words: Dict[str, List[int]] = defaultdict(lambda: [0, 0, 0, 0, 0])
+    num_corr = 0
+    ERR = "*"
+
+    if compute_CER:
+        for i, res in enumerate(results):
+            cut_id, ref, hyp = res
+            ref = list("".join(ref))
+            hyp = list("".join(hyp))
+            results[i] = (cut_id, ref, hyp)
+
+    for _cut_id, ref, hyp in results:
+        ali = kaldialign.align(ref, hyp, ERR, sclite_mode=sclite_mode)
+        for ref_word, hyp_word in ali:
+            if ref_word == ERR:
+                ins[hyp_word] += 1
+                words[hyp_word][3] += 1
+            elif hyp_word == ERR:
+                dels[ref_word] += 1
+                words[ref_word][4] += 1
+            elif hyp_word != ref_word:
+                subs[(ref_word, hyp_word)] += 1
+                words[ref_word][1] += 1
+                words[hyp_word][2] += 1
+            else:
+                words[ref_word][0] += 1
+                num_corr += 1
+    ref_len = sum([len(r) for _, r, _ in results])
+    sub_errs = sum(subs.values())
+    ins_errs = sum(ins.values())
+    del_errs = sum(dels.values())
+    tot_errs = sub_errs + ins_errs + del_errs
+    tot_err_rate = "%.2f" % (100.0 * tot_errs / ref_len)
+
+    if enable_log:
+        logging.info(
+            f"[{test_set_name}] %WER {tot_errs / ref_len:.2%} "
+            f"[{tot_errs} / {ref_len}, {ins_errs} ins, "
+            f"{del_errs} del, {sub_errs} sub ]"
+        )
+
+    print(f"%WER = {tot_err_rate}", file=f)
+    print(
+        f"Errors: {ins_errs} insertions, {del_errs} deletions, "
+        f"{sub_errs} substitutions, over {ref_len} reference "
+        f"words ({num_corr} correct)",
+        file=f,
+    )
+    print(
+        "Search below for sections starting with PER-UTT DETAILS:, "
+        "SUBSTITUTIONS:, DELETIONS:, INSERTIONS:, PER-WORD STATS:",
+        file=f,
+    )
+
+    print("", file=f)
+    print("PER-UTT DETAILS: corr or (ref->hyp)  ", file=f)
+    for cut_id, ref, hyp in results:
+        ali = kaldialign.align(ref, hyp, ERR)
+        combine_successive_errors = True
+        if combine_successive_errors:
+            ali = [[[x], [y]] for x, y in ali]
+            for i in range(len(ali) - 1):
+                if ali[i][0] != ali[i][1] and ali[i + 1][0] != ali[i + 1][1]:
+                    ali[i + 1][0] = ali[i][0] + ali[i + 1][0]
+                    ali[i + 1][1] = ali[i][1] + ali[i + 1][1]
+                    ali[i] = [[], []]
+            ali = [
+                [
+                    list(filter(lambda a: a != ERR, x)),
+                    list(filter(lambda a: a != ERR, y)),
+                ]
+                for x, y in ali
+            ]
+            ali = list(filter(lambda x: x != [[], []], ali))
+            ali = [
+                [
+                    ERR if x == [] else " ".join(x),
+                    ERR if y == [] else " ".join(y),
+                ]
+                for x, y in ali
+            ]
+
+        print(
+            f"{cut_id}:\t"
+            + " ".join(
+                (
+                    ref_word if ref_word == hyp_word else f"({ref_word}->{hyp_word})"
+                    for ref_word, hyp_word in ali
+                )
+            ),
+            file=f,
+        )
+
+    print("", file=f)
+    print("SUBSTITUTIONS: count ref -> hyp", file=f)
+
+    for count, (ref, hyp) in sorted([(v, k) for k, v in subs.items()], reverse=True):
+        print(f"{count}   {ref} -> {hyp}", file=f)
+
+    print("", file=f)
+    print("DELETIONS: count ref", file=f)
+    for count, ref in sorted([(v, k) for k, v in dels.items()], reverse=True):
+        print(f"{count}   {ref}", file=f)
+
+    print("", file=f)
+    print("INSERTIONS: count hyp", file=f)
+    for count, hyp in sorted([(v, k) for k, v in ins.items()], reverse=True):
+        print(f"{count}   {hyp}", file=f)
+
+    print("", file=f)
+    print("PER-WORD STATS: word  corr tot_errs count_in_ref count_in_hyp", file=f)
+    for _, word, counts in sorted(
+        [(sum(v[1:]), k, v) for k, v in words.items()], reverse=True
+    ):
+        (corr, ref_sub, hyp_sub, ins, dels) = counts
+        tot_errs = ref_sub + hyp_sub + ins + dels
+        ref_count = corr + ref_sub + dels
+        hyp_count = corr + hyp_sub + ins
+
+        print(f"{word}   {corr} {tot_errs} {ref_count} {hyp_count}", file=f)
+    return float(tot_err_rate)
+
+
+def get_args():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--tokens",
+        type=str,
+        help="Path to tokens.txt",
+    )
+
+    parser.add_argument(
+        "--hotwords-file",
+        type=str,
+        default="",
+        help="""
+        The file containing hotwords, one words/phrases per line, like
+        HELLO WORLD
+        你好世界
+        """,
+    )
+
+    parser.add_argument(
+        "--hotwords-score",
+        type=float,
+        default=1.5,
+        help="""
+        The hotword score of each token for biasing word/phrase. Used only if
+        --hotwords-file is given.
+        """,
+    )
+
+    parser.add_argument(
+        "--modeling-unit",
+        type=str,
+        default="",
+        help="""
+        The modeling unit of the model, valid values are cjkchar, bpe, cjkchar+bpe.
+        Used only when hotwords-file is given.
+        """,
+    )
+
+    parser.add_argument(
+        "--bpe-vocab",
+        type=str,
+        default="",
+        help="""
+        The path to the bpe vocabulary, the bpe vocabulary is generated by
+        sentencepiece, you can also export the bpe vocabulary through a bpe model
+        by `scripts/export_bpe_vocab.py`. Used only when hotwords-file is given
+        and modeling-unit is bpe or cjkchar+bpe.
+        """,
+    )
+
+    parser.add_argument(
+        "--encoder",
+        default="",
+        type=str,
+        help="Path to the encoder model",
+    )
+
+    parser.add_argument(
+        "--decoder",
+        default="",
+        type=str,
+        help="Path to the decoder model",
+    )
+
+    parser.add_argument(
+        "--joiner",
+        default="",
+        type=str,
+        help="Path to the joiner model",
+    )
+
+    parser.add_argument(
+        "--paraformer",
+        default="",
+        type=str,
+        help="Path to the model.onnx from Paraformer",
+    )
+
+    parser.add_argument(
+        "--nemo-ctc",
+        default="",
+        type=str,
+        help="Path to the model.onnx from NeMo CTC",
+    )
+
+    parser.add_argument(
+        "--wenet-ctc",
+        default="",
+        type=str,
+        help="Path to the model.onnx from WeNet CTC",
+    )
+
+    parser.add_argument(
+        "--tdnn-model",
+        default="",
+        type=str,
+        help="Path to the model.onnx for the tdnn model of the yesno recipe",
+    )
+
+    parser.add_argument(
+        "--num-threads",
+        type=int,
+        default=1,
+        help="Number of threads for neural network computation",
+    )
+
+    parser.add_argument(
+        "--whisper-encoder",
+        default="",
+        type=str,
+        help="Path to whisper encoder model",
+    )
+
+    parser.add_argument(
+        "--whisper-decoder",
+        default="",
+        type=str,
+        help="Path to whisper decoder model",
+    )
+
+    parser.add_argument(
+        "--whisper-language",
+        default="",
+        type=str,
+        help="""It specifies the spoken language in the input audio file.
+        Example values: en, fr, de, zh, jp.
+        Available languages for multilingual models can be found at
+        https://github.com/openai/whisper/blob/main/whisper/tokenizer.py#L10
+        If not specified, we infer the language from the input audio file.
+        """,
+    )
+
+    parser.add_argument(
+        "--whisper-task",
+        default="transcribe",
+        choices=["transcribe", "translate"],
+        type=str,
+        help="""For multilingual models, if you specify translate, the output
+        will be in English.
+        """,
+    )
+
+    parser.add_argument(
+        "--whisper-tail-paddings",
+        default=-1,
+        type=int,
+        help="""Number of tail padding frames.
+        We have removed the 30-second constraint from whisper, so you need to
+        choose the amount of tail padding frames by yourself.
+        Use -1 to use a default value for tail padding.
+        """,
+    )
+
+    parser.add_argument(
+        "--blank-penalty",
+        type=float,
+        default=0.0,
+        help="""
+        The penalty applied on blank symbol during decoding.
+        Note: It is a positive value that would be applied to logits like
+        this `logits[:, 0] -= blank_penalty` (suppose logits.shape is
+        [batch_size, vocab] and blank id is 0).
+        """,
+    )
+
+    parser.add_argument(
+        "--decoding-method",
+        type=str,
+        default="greedy_search",
+        help="Valid values are greedy_search and modified_beam_search",
+    )
+    parser.add_argument(
+        "--debug",
+        type=bool,
+        default=False,
+        help="True to show debug messages",
+    )
+
+    parser.add_argument(
+        "--sample-rate",
+        type=int,
+        default=16000,
+        help="""Sample rate of the feature extractor. Must match the one
+        expected  by the model. Note: The input sound files can have a
+        different sample rate from this argument.""",
+    )
+
+    parser.add_argument(
+        "--feature-dim",
+        type=int,
+        default=80,
+        help="Feature dimension. Must match the one expected by the model",
+    )
+
+    parser.add_argument(
+        "sound_files",
+        type=str,
+        nargs="+",
+        help="The input sound file(s) to decode. Each file must be of WAVE"
+        "format with a single channel, and each sample has 16-bit, "
+        "i.e., int16_t. "
+        "The sample rate of the file can be arbitrary and does not need to "
+        "be 16 kHz",
+    )
+
+    parser.add_argument(
+        "--name",
+        type=str,
+        default="",
+        help="The directory containing the input sound files to decode",
+    )
+
+    parser.add_argument(
+        "--log-dir",
+        type=str,
+        default="",
+        help="The directory containing the input sound files to decode",
+    )
+
+    parser.add_argument(
+        "--label",
+        type=str,
+        default=None,
+        help="wav_base_name label",
+    )
+
+    # Dataset related arguments for loading labels when label file is not provided
+    parser.add_argument(
+        "--dataset-name",
+        type=str,
+        default="yuekai/seed_tts_cosy2",
+        help="Huggingface dataset name for loading labels",
+    )
+
+    parser.add_argument(
+        "--split-name",
+        type=str,
+        default="wenetspeech4tts",
+        help="Dataset split name for loading labels",
+    )
+
+    return parser.parse_args()
+
+
+def assert_file_exists(filename: str):
+    assert Path(filename).is_file(), (
+        f"{filename} does not exist!\n"
+        "Please refer to "
+        "https://k2-fsa.github.io/sherpa/onnx/pretrained_models/index.html to download it"
+    )
+
+
+def read_wave(wave_filename: str) -> Tuple[np.ndarray, int]:
+    """
+    Args:
+      wave_filename:
+        Path to a wave file. It should be single channel and can be of type
+        32-bit floating point PCM. Its sample rate does not need to be 24kHz.
+
+    Returns:
+      Return a tuple containing:
+       - A 1-D array of dtype np.float32 containing the samples,
+         which are normalized to the range [-1, 1].
+       - Sample rate of the wave file.
+    """
+
+    samples, sample_rate = sf.read(wave_filename, dtype="float32")
+    assert (
+        samples.ndim == 1
+    ), f"Expected single channel, but got {samples.ndim} channels."
+
+    samples_float32 = samples.astype(np.float32)
+
+    return samples_float32, sample_rate
+
+
+def normalize_text_alimeeting(text: str) -> str:
+    """
+    Text normalization similar to M2MeT challenge baseline.
+    See: https://github.com/yufan-aslp/AliMeeting/blob/main/asr/local/text_normalize.pl
+    """
+    import re
+    text = text.replace('\u00A0', '')  # test_hard
+    text = text.replace(" ", "")
+    text = text.replace("<sil>", "")
+    text = text.replace("<%>", "")
+    text = text.replace("<->", "")
+    text = text.replace("<$>", "")
+    text = text.replace("<#>", "")
+    text = text.replace("<_>", "")
+    text = text.replace("<space>", "")
+    text = text.replace("`", "")
+    text = text.replace("&", "")
+    text = text.replace(",", "")
+    if re.search("[a-zA-Z]", text):
+        text = text.upper()
+    text = text.replace("Ａ", "A")
+    text = text.replace("ａ", "A")
+    text = text.replace("ｂ", "B")
+    text = text.replace("ｃ", "C")
+    text = text.replace("ｋ", "K")
+    text = text.replace("ｔ", "T")
+    text = text.replace("，", "")
+    text = text.replace("丶", "")
+    text = text.replace("。", "")
+    text = text.replace("、", "")
+    text = text.replace("？", "")
+    text = remove_punctuation(text)
+    return text
+
+
+def main():
+    args = get_args()
+    assert_file_exists(args.tokens)
+    assert args.num_threads > 0, args.num_threads
+
+    assert len(args.nemo_ctc) == 0, args.nemo_ctc
+    assert len(args.wenet_ctc) == 0, args.wenet_ctc
+    assert len(args.whisper_encoder) == 0, args.whisper_encoder
+    assert len(args.whisper_decoder) == 0, args.whisper_decoder
+    assert len(args.tdnn_model) == 0, args.tdnn_model
+
+    assert_file_exists(args.paraformer)
+
+    recognizer = sherpa_onnx.OfflineRecognizer.from_paraformer(
+        paraformer=args.paraformer,
+        tokens=args.tokens,
+        num_threads=args.num_threads,
+        sample_rate=args.sample_rate,
+        feature_dim=args.feature_dim,
+        decoding_method=args.decoding_method,
+        debug=args.debug,
+    )
+
+    print("Started!")
+    start_time = time.time()
+
+    streams, results = [], []
+    total_duration = 0
+
+    for i, wave_filename in enumerate(args.sound_files):
+        assert_file_exists(wave_filename)
+        samples, sample_rate = read_wave(wave_filename)
+        duration = len(samples) / sample_rate
+        total_duration += duration
+        s = recognizer.create_stream()
+        s.accept_waveform(sample_rate, samples)
+
+        streams.append(s)
+        if i % 10 == 0:
+            recognizer.decode_streams(streams)
+            results += [s.result.text for s in streams]
+            streams = []
+            print(f"Processed {i} files")
+        # process the last batch
+    if streams:
+        recognizer.decode_streams(streams)
+        results += [s.result.text for s in streams]
+    end_time = time.time()
+    print("Done!")
+
+    results_dict = {}
+    for wave_filename, result in zip(args.sound_files, results):
+        print(f"{wave_filename}\n{result}")
+        print("-" * 10)
+        wave_basename = Path(wave_filename).stem
+        results_dict[wave_basename] = result
+
+    elapsed_seconds = end_time - start_time
+    rtf = elapsed_seconds / total_duration
+    print(f"num_threads: {args.num_threads}")
+    print(f"decoding_method: {args.decoding_method}")
+    print(f"Wave duration: {total_duration:.3f} s")
+    print(f"Elapsed time: {elapsed_seconds:.3f} s")
+    print(
+        f"Real time factor (RTF): {elapsed_seconds:.3f}/{total_duration:.3f} = {rtf:.3f}"
+    )
+
+    # Load labels either from file or from dataset
+    labels_dict = {}
+
+    if args.label:
+        # Load labels from file (original functionality)
+        print(f"Loading labels from file: {args.label}")
+        with open(args.label, "r") as f:
+            for line in f:
+                # fields = line.strip().split(" ")
+                # fields = [item for item in fields if item]
+                # assert len(fields) == 4
+                # prompt_text, prompt_audio, text, audio_path = fields
+
+                fields = line.strip().split("|")
+                fields = [item for item in fields if item]
+                assert len(fields) == 4
+                audio_path, prompt_text, prompt_audio, text = fields
+                labels_dict[Path(audio_path).stem] = normalize_text_alimeeting(text)
+    else:
+        # Load labels from dataset (new functionality)
+        print(f"Loading labels from dataset: {args.dataset_name}, split: {args.split_name}")
+        if 'zero' in args.split_name:
+            dataset_name = "yuekai/CV3-Eval"
+        else:
+            dataset_name = "yuekai/seed_tts_cosy2"
+        dataset = load_dataset(
+            dataset_name,
+            split=args.split_name,
+            trust_remote_code=True,
+        )
+
+        for item in dataset:
+            audio_id = item["id"]
+            labels_dict[audio_id] = normalize_text_alimeeting(item["target_text"])
+
+        print(f"Loaded {len(labels_dict)} labels from dataset")
+
+    # Perform evaluation if labels are available
+    if labels_dict:
+
+        final_results = []
+        for key, value in results_dict.items():
+            if key in labels_dict:
+                final_results.append((key, labels_dict[key], value))
+            else:
+                print(f"Warning: No label found for {key}, skipping...")
+
+        if final_results:
+            store_transcripts(
+                filename=f"{args.log_dir}/recogs-{args.name}.txt", texts=final_results
+            )
+            with open(f"{args.log_dir}/errs-{args.name}.txt", "w") as f:
+                write_error_stats(f, "test-set", final_results, enable_log=True)
+
+            with open(f"{args.log_dir}/errs-{args.name}.txt", "r") as f:
+                print(f.readline())  # WER
+                print(f.readline())  # Detailed errors
+        else:
+            print("No matching labels found for evaluation")
+    else:
+        print("No labels available for evaluation")
+
+
+if __name__ == "__main__":
+    main()

examples/grpo/cosyvoice2/token2wav_asr_server.py

@@ -0,0 +1,346 @@
+# SPDX-FileCopyrightText: Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+"""Pytriton server for token2wav conversion and ASR"""
+
+from datasets import load_dataset
+from cosyvoice.cli.cosyvoice import CosyVoice2
+from omnisense.models import OmniSenseVoiceSmall
+from pytriton.proxy.types import Request
+from pytriton.triton import Triton, TritonConfig
+from pytriton.model_config import DynamicBatcher, ModelConfig, Tensor
+from pytriton.decorators import batch
+import argparse
+import io
+import logging
+from typing import Any, List
+import numpy as np
+import torch
+from scipy.signal import resample
+import sys
+import random
+import re
+from jiwer import wer
+from pypinyin import lazy_pinyin, Style
+from tn.chinese.normalizer import Normalizer as ZhNormalizer
+
+# Chinese text normalizer (cached globally)
+zh_tn_model = ZhNormalizer(
+    cache_dir="./cache",
+    remove_erhua=False,
+    remove_interjections=False,
+    remove_puncts=True,
+    overwrite_cache=True,
+)
+
+
+sys.path.append("/workspace/CosyVoice/third_party/Matcha-TTS")
+
+logger = logging.getLogger("token2wav_asr_server")
+
+
+class _ASR_Server:
+    """Wraps a single OmniSenseVoiceSmall model instance for Triton."""
+
+    def __init__(self, device_id: int):
+        self._model = OmniSenseVoiceSmall("iic/SenseVoiceSmall", quantize=False, device_id=device_id)
+
+    @batch
+    def __call__(self, WAV: np.ndarray, WAV_LENS: np.ndarray, LANGUAGE: np.ndarray, TEXT_NORM: np.ndarray):
+        """
+        WAV: np.ndarray, WAV_LENS: np.ndarray
+        LANGUAGE: np.ndarray, TEXTNORM: np.ndarray for backward compatibility, not used
+        See: https://github.com/modelscope/FunASR/tree/main/runtime/triton_gpu
+        """
+        logger.debug("WAV: %s, WAV_LENS: %s, shapes: %s %s", type(WAV), type(WAV_LENS), WAV.shape, WAV_LENS.shape)
+        wavs = [WAV[i, :WAV_LENS[i, 0]] for i in range(len(WAV))]
+
+        results = self._model.transcribe_single_batch(
+            wavs,
+            language="zh",
+            textnorm="woitn",
+        )
+        texts = [result.text for result in results]
+        transcripts = np.char.encode(np.array(texts).reshape(-1, 1), "utf-8")
+        return {"TRANSCRIPTS": transcripts}
+
+
+def audio_decode_cosyvoice2(
+    audio_tokens, prompt_text, prompt_speech_16k, codec_decoder
+):
+    """
+    Generate audio from tokens with optional tone and prompt embedding.
+    """
+    model_inputs_dict = codec_decoder.frontend.frontend_zero_shot(
+        "empty", prompt_text, prompt_speech_16k, 24000
+    )
+    tts_mel, _ = codec_decoder.model.flow.inference(
+        token=audio_tokens.to(codec_decoder.model.device),
+        token_len=torch.tensor([audio_tokens.shape[1]], dtype=torch.int32).to(
+            codec_decoder.model.device
+        ),
+        prompt_token=model_inputs_dict["flow_prompt_speech_token"].to(
+            codec_decoder.model.device
+        ),
+        prompt_token_len=torch.tensor(
+            [model_inputs_dict["flow_prompt_speech_token_len"]], dtype=torch.int32
+        ).to(codec_decoder.model.device),
+        prompt_feat=model_inputs_dict["prompt_speech_feat"].to(
+            codec_decoder.model.device
+        ),
+        prompt_feat_len=model_inputs_dict["prompt_speech_feat_len"].to(
+            codec_decoder.model.device
+        ),
+        embedding=model_inputs_dict["flow_embedding"].to(codec_decoder.model.device),
+        finalize=True,
+    )
+
+    audio_hat, _ = codec_decoder.model.hift.inference(
+        speech_feat=tts_mel, cache_source=torch.zeros(1, 1, 0)
+    )
+
+    return audio_hat
+
+
+def get_random_prompt_from_dataset(dataset):
+    """
+    Get random prompt text and speech from the pre-loaded dataset.
+    Returns (prompt_text, prompt_speech_16k)
+    """
+    random_idx = random.randint(0, len(dataset) - 1)
+    sample = dataset[random_idx]
+
+    # Extract audio data
+    audio_data = sample["audio"]
+    audio_array = audio_data["array"]
+    sample_rate = audio_data["sampling_rate"]
+
+    # Convert audio to 16kHz if needed
+    if sample_rate != 16000:
+        num_samples = int(len(audio_array) * (16000 / sample_rate))
+        audio_array = resample(audio_array, num_samples)
+
+    # Convert to torch tensor
+    prompt_speech_16k = torch.from_numpy(audio_array).float().unsqueeze(0)
+    prompt_text = sample["text"]
+    # remove space in prompt_text
+    prompt_text = prompt_text.replace(" ", "")
+    return prompt_text, prompt_speech_16k
+
+
+class _Token2Wav_ASR:
+    """Wraps a single OmniSenseVoiceSmall model instance for Triton."""
+
+    def __init__(self, device_id: int):
+        self.asr_model = OmniSenseVoiceSmall("iic/SenseVoiceSmall", quantize=False, device_id=device_id)
+        self.dataset = load_dataset("yuekai/aishell", "test", trust_remote_code=True)["test"]
+
+        # Make sure the CosyVoice2 decoder lives on the same GPU as the ASR model
+        # CosyVoice2 internally uses generic "cuda" device, so we first switch the
+        # current CUDA context to the desired card before the object is created.
+        # Afterwards, all parameters loaded with the generic "cuda" device will
+        # reside on this GPU.  We keep the selected id in `self.device_id` and
+        # will set the context again for every forward call to avoid race
+        # conditions when several instances are used in the same process.
+
+        self.device_id = device_id
+
+        # Construct the TTS codec decoder under the correct CUDA device context
+        with torch.cuda.device(self.device_id):
+            self.codec_decoder = CosyVoice2(
+                "/workspace/CosyVoice2-0.5B", load_jit=True, load_trt=True, fp16=True
+            )
+
+    @batch
+    def __call__(self, TOKENS: np.ndarray, TOKEN_LENS: np.ndarray, GT_TEXT: np.ndarray):
+        """
+        WAV: np.ndarray, WAV_LENS: np.ndarray
+        LANGUAGE: np.ndarray, TEXTNORM: np.ndarray for backward compatibility, not used
+        See: https://github.com/modelscope/FunASR/tree/main/runtime/triton_gpu
+        """
+        # Ensure the default CUDA device is set correctly for this invocation
+        torch.cuda.set_device(self.device_id)
+
+        if self.device_id == 0:
+            print(f"device_id: {self.device_id}, TOKENS: {TOKENS.shape}, TOKEN_LENS: {TOKEN_LENS.shape}")
+
+        tokens_list = [TOKENS[i, :TOKEN_LENS[i, 0]] for i in range(len(TOKENS))]
+
+        # Decode ground-truth text strings (BYTES → str)
+        if GT_TEXT.ndim == 2:
+            gt_texts = [GT_TEXT[i, 0].decode("utf-8") for i in range(len(GT_TEXT))]
+        else:
+            gt_texts = [GT_TEXT[i].decode("utf-8") for i in range(len(GT_TEXT))]
+
+        wavs = []
+        for tokens in tokens_list:
+            prompt_text, prompt_speech_16k = get_random_prompt_from_dataset(self.dataset)
+            audio_tokens = torch.tensor(tokens, dtype=torch.long, device=self.asr_model.device).unsqueeze(0)
+            audio_hat = audio_decode_cosyvoice2(
+                audio_tokens,
+                prompt_text,
+                prompt_speech_16k,
+                self.codec_decoder,
+            )
+            # resample to 16000 using soundfile
+            audio_hat = audio_hat.squeeze(0).float().cpu()
+            audio_hat = audio_hat.numpy()
+            num_samples = int(len(audio_hat) * (16000 / 24000))
+            audio_hat = resample(audio_hat, num_samples)
+            wavs.append(audio_hat)
+
+        results = self.asr_model.transcribe_single_batch(
+            wavs,
+            language="zh",
+            textnorm="woitn",
+        )
+        texts = [result.text for result in results]
+
+        # ---------------- Reward computation ----------------
+        rewards = []
+        for gt_text, hyp_text in zip(gt_texts, texts):
+            gt_norm = zh_tn_model.normalize(gt_text).lower()
+            hyp_norm = zh_tn_model.normalize(hyp_text).lower()
+
+            gt_pinyin = lazy_pinyin(
+                gt_norm,
+                style=Style.TONE3,
+                tone_sandhi=True,
+                neutral_tone_with_five=True,
+            )
+            hyp_pinyin = lazy_pinyin(
+                hyp_norm,
+                style=Style.TONE3,
+                tone_sandhi=True,
+                neutral_tone_with_five=True,
+            )
+
+            c = float(wer(" ".join(gt_pinyin), " ".join(hyp_pinyin)))
+            reward_val = 1.0 - np.tanh(3.0 * c)
+            reward_val = max(0.0, min(1.0, reward_val))
+            rewards.append(reward_val)
+            print(f"gt_text: {gt_text}, hyp_text: {hyp_text}, reward_val: {reward_val}")
+
+        transcripts = np.char.encode(np.array(texts).reshape(-1, 1), "utf-8")
+        rewards_arr = np.array(rewards, dtype=np.float32).reshape(-1, 1)
+
+        return {"REWARDS": rewards_arr, "TRANSCRIPTS": transcripts}
+
+
+def _infer_function_factory(device_ids: List[int], model_name: str):
+    """Creates a list of inference functions, one for each requested device ID."""
+    infer_funcs = []
+    for device_id in device_ids:
+        if model_name == "sensevoice":
+            infer_funcs.append(_ASR_Server(device_id=device_id))
+        else:
+            infer_funcs.append(_Token2Wav_ASR(device_id=device_id))
+    return infer_funcs
+
+
+def main():
+    parser = argparse.ArgumentParser(description=__doc__)
+    parser.add_argument(
+        "--max-batch-size",
+        type=int,
+        default=32,
+        help="Batch size of request.",
+        required=False,
+    )
+    parser.add_argument(
+        "--verbose",
+        action="store_true",
+        default=False,
+    )
+    parser.add_argument(
+        "--number-of-instances-per-device",
+        type=int,
+        default=1,
+        help="Number of model instances to load.",
+        required=False,
+    )
+    parser.add_argument(
+        "--number-of-devices",
+        type=int,
+        default=8,
+        help="Number of devices to use.",
+    )
+    parser.add_argument(
+        "--model-name",
+        type=str,
+        default="token2wav_asr",
+        choices=["token2wav_asr", "sensevoice"],
+        help="Model name.",
+    )
+
+    args = parser.parse_args()
+
+    log_level = logging.DEBUG if args.verbose else logging.INFO
+    logging.basicConfig(level=log_level, format="%(asctime)s - %(levelname)s - %(name)s: %(message)s")
+
+    triton_config = TritonConfig(
+        http_port=8000,
+        grpc_port=8001,
+        metrics_port=8002,
+    )
+
+    device_ids = list(range(args.number_of_devices))
+    device_ids = device_ids * args.number_of_instances_per_device
+
+    with Triton(config=triton_config) as triton:
+        logger.info("Loading SenseVoice model on device ids: %s", device_ids)
+        if args.model_name == "sensevoice":
+            triton.bind(
+                model_name="sensevoice",
+                infer_func=_infer_function_factory(device_ids, args.model_name),
+                inputs=[
+                    Tensor(name="WAV", dtype=np.float32, shape=(-1,)),
+                    Tensor(name="WAV_LENS", dtype=np.int32, shape=(-1,)),
+                    Tensor(name="LANGUAGE", dtype=np.int32, shape=(-1,)),
+                    Tensor(name="TEXT_NORM", dtype=np.int32, shape=(-1,)),
+                ],
+                outputs=[
+                    Tensor(name="TRANSCRIPTS", dtype=bytes, shape=(-1,)),
+                ],
+                config=ModelConfig(
+                    max_batch_size=args.max_batch_size,
+                    batcher=DynamicBatcher(max_queue_delay_microseconds=10000),  # 10ms
+                ),
+                strict=True,
+            )
+        else:
+            triton.bind(
+                model_name="token2wav_asr",
+                infer_func=_infer_function_factory(device_ids, args.model_name),
+                inputs=[
+                    Tensor(name="TOKENS", dtype=np.int32, shape=(-1,)),
+                    Tensor(name="TOKEN_LENS", dtype=np.int32, shape=(-1,)),
+                    Tensor(name="GT_TEXT", dtype=bytes, shape=(-1,)),
+                ],
+                outputs=[
+                    Tensor(name="REWARDS", dtype=np.float32, shape=(-1,)),
+                    Tensor(name="TRANSCRIPTS", dtype=bytes, shape=(-1,)),
+                ],
+                config=ModelConfig(
+                    max_batch_size=args.max_batch_size,
+                    batcher=DynamicBatcher(max_queue_delay_microseconds=10000),  # 10ms
+                ),
+                strict=True,
+            )
+        logger.info("Serving inference")
+        triton.serve()
+
+
+if __name__ == "__main__":
+    main()

examples/libritts/cosyvoice/conf/cosyvoice.fromscratch.yaml

@@ -1,257 +0,0 @@
-# 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: 22050
-text_encoder_input_size: 512
-llm_input_size: 1024
-llm_output_size: 1024
-spk_embed_dim: 192
-
-# 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.TransformerLM
-    text_encoder_input_size: !ref <text_encoder_input_size>
-    llm_input_size: !ref <llm_input_size>
-    llm_output_size: !ref <llm_output_size>
-    text_token_size: 51866 # change to 60515 if you want to train with CosyVoice-300M-25Hz recipe
-    speech_token_size: 4096
-    length_normalized_loss: True
-    lsm_weight: 0
-    spk_embed_dim: !ref <spk_embed_dim>
-    text_encoder: !new:cosyvoice.transformer.encoder.ConformerEncoder
-        input_size: !ref <text_encoder_input_size>
-        output_size: 1024
-        attention_heads: 8
-        linear_units: 2048
-        num_blocks: 3
-        dropout_rate: 0.1
-        positional_dropout_rate: 0.1
-        attention_dropout_rate: 0.0
-        normalize_before: True
-        input_layer: 'linear'
-        pos_enc_layer_type: 'rel_pos_espnet'
-        selfattention_layer_type: 'rel_selfattn'
-        use_cnn_module: False
-        macaron_style: False
-        use_dynamic_chunk: False
-        use_dynamic_left_chunk: False
-        static_chunk_size: 1
-    llm: !new:cosyvoice.transformer.encoder.TransformerEncoder
-        input_size: !ref <llm_input_size>
-        output_size: !ref <llm_output_size>
-        attention_heads: 8
-        linear_units: 2048
-        num_blocks: 7
-        dropout_rate: 0.1
-        positional_dropout_rate: 0.1
-        attention_dropout_rate: 0.0
-        input_layer: 'linear_legacy'
-        pos_enc_layer_type: 'rel_pos_espnet'
-        selfattention_layer_type: 'rel_selfattn'
-        static_chunk_size: 1
-    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.MaskedDiffWithXvec
-    input_size: 512
-    output_size: 80
-    spk_embed_dim: !ref <spk_embed_dim>
-    output_type: 'mel'
-    vocab_size: 4096
-    input_frame_rate: 50 # change to 25 if you want to train with CosyVoice-300M-25Hz recipe
-    only_mask_loss: True
-    encoder: !new:cosyvoice.transformer.encoder.ConformerEncoder
-        output_size: 512
-        attention_heads: 4
-        linear_units: 1024
-        num_blocks: 3
-        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
-    length_regulator: !new:cosyvoice.flow.length_regulator.InterpolateRegulator
-        channels: 80
-        sampling_ratios: [1, 1, 1, 1]
-    decoder: !new:cosyvoice.flow.flow_matching.ConditionalCFM
-        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.ConditionalDecoder
-            in_channels: 320
-            out_channels: 80
-            channels: [256, 256]
-            dropout: 0.0
-            attention_head_dim: 64
-            n_blocks: 4
-            num_mid_blocks: 8
-            num_heads: 8
-            act_fn: 'gelu'
-
-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, 8]
-    upsample_kernel_sizes: [16, 16]
-    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, 11]
-    source_resblock_dilation_sizes: [[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: 1024
-    num_mels: 80
-    sampling_rate: !ref <sample_rate>
-    hop_size: 256
-    win_size: 1024
-    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:whisper.tokenizer.get_tokenizer # change to !name:cosyvoice.tokenizer.tokenizer.get_tokenizer if you want to train with CosyVoice-300M-25Hz recipe
-    multilingual: True
-    num_languages: 100
-    language: 'en'
-    task: 'transcribe'
-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: 0
-    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: 24576 # must be a multiplier of hop_size
-feat_extractor: !name:matcha.utils.audio.mel_spectrogram
-    n_fft: 1024
-    num_mels: 80
-    sampling_rate: !ref <sample_rate>
-    hop_size: 256
-    win_size: 1024
-    fmin: 0
-    fmax: 8000
-    center: False
-compute_fbank: !name:cosyvoice.dataset.processor.compute_fbank
-    feat_extractor: !ref <feat_extractor>
-compute_f0: !name:cosyvoice.dataset.processor.compute_f0
-    sample_rate: !ref <sample_rate>
-    hop_size: 256
-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: 12000
-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: 0.002 # change to 0.001 if you want to train flow from scratch
-    scheduler: warmuplr
-    scheduler_conf:
-        warmup_steps: 25000
-    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/cosyvoice/cosyvoice

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

examples/libritts/cosyvoice/local/prepare_data.py

@@ -40,6 +40,10 @@ 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 != '':
+        with open('{}/instruct'.format(args.des_dir), 'w') as f:
+            for k, v in utt2text.items():
+                f.write('{} {}\n'.format(k, args.instruct))
     return
 
 
@@ -49,7 +53,8 @@ if __name__ == "__main__":
                         type=str)
     parser.add_argument('--des_dir',
                         type=str)
-    parser.add_argument('--ref_model',
-                        type=str)
+    parser.add_argument('--instruct',
+                        type=str,
+                        default='')
     args = parser.parse_args()
     main()

examples/libritts/cosyvoice/run.sh

@@ -27,7 +27,7 @@ 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/$x \
+    ../../../tools/extract_embedding.py --dir data/$x \
       --onnx_path $pretrained_model_dir/campplus.onnx
   done
 fi
@@ -35,7 +35,7 @@ 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/$x \
+    ../../../tools/extract_speech_token.py --dir data/$x \
       --onnx_path $pretrained_model_dir/speech_tokenizer_v1.onnx
   done
 fi
@@ -44,7 +44,7 @@ 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 \
+    ../../../tools/make_parquet_list.py --num_utts_per_parquet 1000 \
       --num_processes 10 \
       --src_dir data/$x \
       --des_dir data/$x/parquet
@@ -60,7 +60,7 @@ 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 your want to train from scratch, please use conf/cosyvoice.fromscratch.yaml"
+  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
@@ -69,7 +69,7 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
   for model in llm flow hifigan; do
     torchrun --nnodes=1 --nproc_per_node=$num_gpus \
         --rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:1234" \
-      cosyvoice/bin/train.py \
+      ../../../cosyvoice/bin/train.py \
       --train_engine $train_engine \
       --config conf/cosyvoice.yaml \
       --train_data data/train.data.list \

examples/libritts/cosyvoice/tools

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

examples/libritts/cosyvoice2/conf/cosyvoice2.yaml

@@ -139,7 +139,7 @@ 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
+    max_length: 6000
     min_length: 100
     token_max_length: 200
     token_min_length: 1
@@ -158,7 +158,9 @@ feat_extractor: !name:matcha.utils.audio.mel_spectrogram
     center: False
 compute_fbank: !name:cosyvoice.dataset.processor.compute_fbank
     feat_extractor: !ref <feat_extractor>
-    token_mel_ratio: 2
+    num_frames: 960
+compute_whisper_fbank: !name:cosyvoice.dataset.processor.compute_whisper_fbank
+    num_frames: 960
 compute_f0: !name:cosyvoice.dataset.processor.compute_f0
     sample_rate: !ref <sample_rate>
     hop_size: 480
@@ -183,6 +185,7 @@ data_pipeline: [
     !ref <resample>,
     !ref <compute_fbank>,
     !ref <parse_embedding>,
+    !ref <compute_whisper_fbank>,
     !ref <shuffle>,
     !ref <sort>,
     !ref <batch>,

examples/libritts/cosyvoice2/cosyvoice

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

examples/libritts/cosyvoice2/run.sh

@@ -24,27 +24,12 @@ if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
   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_v2.onnx
-  done
-fi
-
+# NOTE embedding/token extraction is not necessary now as we support online feature extraction
 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 \
+    ../../../tools/make_parquet_list.py --num_utts_per_parquet 1000 \
       --num_processes 10 \
       --src_dir data/$x \
       --des_dir data/$x/parquet
@@ -60,22 +45,22 @@ 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 your want to train from scratch, please use conf/cosyvoice.fromscratch.yaml"
+  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 \
+      ../../../cosyvoice/bin/train.py \
       --train_engine $train_engine \
       --config conf/cosyvoice2.yaml \
       --train_data data/train.data.list \
       --cv_data data/dev.data.list \
       --qwen_pretrain_path $pretrained_model_dir/CosyVoice-BlankEN \
+      --onnx_path $pretrained_model_dir \
       --model $model \
       --checkpoint $pretrained_model_dir/$model.pt \
       --model_dir `pwd`/exp/cosyvoice2/$model/$train_engine \

examples/libritts/cosyvoice2/run_dpo.sh

@@ -36,7 +36,7 @@ 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/$x \
+    ../../../tools/extract_embedding.py --dir data/$x \
       --onnx_path $pretrained_model_dir/campplus.onnx
   done
 fi
@@ -44,7 +44,7 @@ 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 train-clean-100_reject train-clean-360_reject dev-clean dev-other test-clean test-other; do
-    tools/extract_speech_token.py --dir data/$x \
+    ../../../tools/extract_speech_token.py --dir data/$x \
       --onnx_path $pretrained_model_dir/speech_tokenizer_v2.onnx
   done
 fi
@@ -53,7 +53,7 @@ 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 \
+    ../../../tools/make_parquet_list.py --num_utts_per_parquet 1000 \
       --num_processes 10 \
       --dpo \
       --src_dir data/$x \
@@ -70,7 +70,7 @@ 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 your want to train from scratch, please use conf/cosyvoice.fromscratch.yaml"
+  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
@@ -80,11 +80,12 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
   for model in llm; do
     torchrun --nnodes=1 --nproc_per_node=$num_gpus \
         --rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:1234" \
-      cosyvoice/bin/train.py \
+      ../../../cosyvoice/bin/train.py \
       --train_engine $train_engine \
       --config conf/cosyvoice2.yaml \
       --train_data data/train.data.list \
       --cv_data data/dev.data.list \
+      --onnx_path $pretrained_model_dir \
       --qwen_pretrain_path $pretrained_model_dir/CosyVoice-BlankEN \
       --model $model \
       --checkpoint $pretrained_model_dir/$model.pt \

examples/libritts/cosyvoice2/tools

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

examples/libritts/cosyvoice3/conf/cosyvoice3.yaml

@@ -5,22 +5,28 @@ __set_seed3: !apply:torch.manual_seed [1986]
 __set_seed4: !apply:torch.cuda.manual_seed_all [1986]
 
 # fixed params
-sample_rate: 24000   # 16000 for llm, 24000 for cfm
+sample_rate: 24000
 llm_input_size: 896
 llm_output_size: 896
 spk_embed_dim: 192
-qwen_pretrain_path: 'CosyVoice2-0.5B/CosyVoice-BlankEN'
+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_dpo.Qwen2LM
+llm: !new:cosyvoice.llm.llm.CosyVoice3LM
     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
-    dpo: True
+    mix_ratio: [5, 15]
     llm: !new:cosyvoice.llm.llm.Qwen2Encoder
         pretrain_path: !ref <qwen_pretrain_path>
     sampling: !name:cosyvoice.utils.common.ras_sampling
@@ -28,31 +34,21 @@ llm: !new:cosyvoice.llm.llm_dpo.Qwen2LM
         top_k: 25
         win_size: 10
         tau_r: 0.1
-flow: !new:cosyvoice.flow.flow.CausalMaskedDiffWithXvec
-    input_size: 512
+
+flow: !new:cosyvoice.flow.flow.CausalMaskedDiffWithDiT
+    input_size: 80
     output_size: 80
     spk_embed_dim: !ref <spk_embed_dim>
     output_type: 'mel'
     vocab_size: 6561
-    input_frame_rate: 25
+    input_frame_rate: !ref <token_frame_rate>
     only_mask_loss: True
-    token_mel_ratio: 2
+    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
+    pre_lookahead_layer: !new:cosyvoice.transformer.upsample_encoder.PreLookaheadLayer
+        in_channels: 80
+        channels: 1024
+        pre_lookahead_len: 3
     decoder: !new:cosyvoice.flow.flow_matching.CausalConditionalCFM
         in_channels: 240
         n_spks: 1
@@ -65,19 +61,20 @@ flow: !new:cosyvoice.flow.flow.CausalMaskedDiffWithXvec
                 training_cfg_rate: 0.2
                 inference_cfg_rate: 0.7
                 reg_loss_type: 'l1'
-        estimator: !new:cosyvoice.flow.decoder.ConditionalDecoder
-            in_channels: 320
+        estimator: !new:cosyvoice.flow.DiT.dit.DiT
+            dim: 1024
+            depth: 22
+            heads: 16
+            dim_head: 64
+            ff_mult: 2
+            mel_dim: 80
+            mu_dim: 80
+            spk_dim: 80
             out_channels: 80
-            causal: True
-            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
+hift: !new:cosyvoice.hifigan.generator.CausalHiFTGenerator
     in_channels: 80
     base_channels: 512
     nb_harmonics: 8
@@ -96,18 +93,19 @@ hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
     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
+    conv_pre_look_right: 4
+    f0_predictor: !new:cosyvoice.hifigan.f0_predictor.CausalConvRNNF0Predictor
         num_class: 1
         in_channels: 80
         cond_channels: 512
 
 # gan related module
 mel_spec_transform1: !name:matcha.utils.audio.mel_spectrogram
-    n_fft: 1024
+    n_fft: 1920
     num_mels: 80
     sampling_rate: !ref <sample_rate>
-    hop_size: 256
-    win_size: 1024
+    hop_size: 480
+    win_size: 1920
     fmin: 0
     fmax: null
     center: False
@@ -115,45 +113,47 @@ 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.MultiResolutionDiscriminator
+        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:whisper.tokenizer.get_tokenizer # change to !name:cosyvoice.tokenizer.tokenizer.get_tokenizer if you want to train with CosyVoice-300M-25Hz recipe
-    multilingual: True
-    num_languages: 100
-    language: 'en'
-    task: 'transcribe'
+get_tokenizer: !name:cosyvoice.tokenizer.tokenizer.get_qwen_tokenizer
+    token_path: !ref <qwen_pretrain_path>
+    skip_special_tokens: True
+    version: cosyvoice3
 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: 0
+    max_length: 6000
+    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: 24576 # must be a multiplier of hop_size
+    truncate_length: 24960 # must be a multiplier of hop_size and token_mel_ratio
 feat_extractor: !name:matcha.utils.audio.mel_spectrogram
-    n_fft: 1024
+    n_fft: 1920
     num_mels: 80
     sampling_rate: !ref <sample_rate>
-    hop_size: 256
-    win_size: 1024
+    hop_size: 480
+    win_size: 1920
     fmin: 0
-    fmax: 8000
+    fmax: null
     center: False
 compute_fbank: !name:cosyvoice.dataset.processor.compute_fbank
     feat_extractor: !ref <feat_extractor>
+    num_frames: 960
+compute_whisper_fbank: !name:cosyvoice.dataset.processor.compute_whisper_fbank
+    num_frames: 960
 compute_f0: !name:cosyvoice.dataset.processor.compute_f0
     sample_rate: !ref <sample_rate>
-    hop_size: 256
+    hop_size: 480
 parse_embedding: !name:cosyvoice.dataset.processor.parse_embedding
     normalize: True
 shuffle: !name:cosyvoice.dataset.processor.shuffle
@@ -162,10 +162,10 @@ 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 # change to 1400 in gan train on v100 16g
+    max_frames_in_batch: 2000
 padding: !name:cosyvoice.dataset.processor.padding
-    use_spk_embedding: True # change to True during sft
-    dpo: True
+    use_spk_embedding: False # change to True during sft
+
 
 # dataset processor pipeline
 data_pipeline: [
@@ -175,6 +175,7 @@ data_pipeline: [
     !ref <resample>,
     !ref <compute_fbank>,
     !ref <parse_embedding>,
+    !ref <compute_whisper_fbank>,
     !ref <shuffle>,
     !ref <sort>,
     !ref <batch>,
@@ -199,10 +200,10 @@ data_pipeline_gan: [
 train_conf:
     optim: adam
     optim_conf:
-        lr: 0.00001 # change to 1e-5 during sft
-    scheduler: warmuplr # change to constantlr during sft
+        lr: 1e-5 # change to 1e-5 during sft
+    scheduler: constantlr # change to constantlr during sft
     scheduler_conf:
-        warmup_steps: 25000
+        warmup_steps: 2500
     max_epoch: 200
     grad_clip: 5
     accum_grad: 2
@@ -223,4 +224,4 @@ train_conf_gan:
     grad_clip: 5
     accum_grad: 1 # in gan training, accum_grad must be 1
     log_interval: 100
-    save_per_step: -1
+    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/local

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

examples/libritts/cosyvoice3/path.sh

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

examples/libritts/cosyvoice3/run.sh

@@ -0,0 +1,97 @@
+#!/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/Fun-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
+    # NOTE in CosyVoice3, we add instruct in sequence
+    python local/prepare_data.py --src_dir $data_dir/LibriTTS/$x --des_dir data/$x --instruct "You are a helpful assistant.<|endofprompt|>"
+  done
+fi
+
+# NOTE embedding/token extraction is not necessary now as we support online feature extraction
+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 \
+      --src_dir data/$x \
+      --des_dir data/$x/parquet
+  done
+fi
+
+# train llm
+export CUDA_VISIBLE_DEVICES="0"
+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 [ $train_engine == 'deepspeed' ]; then
+    echo "Notice deepspeed has its own optimizer config. Modify conf/ds_stage2.json if necessary"
+  fi
+  cat data/{train-clean-100,train-clean-360,train-other-500}/parquet/data.list > data/train.data.list
+  cat data/{dev-clean,dev-other}/parquet/data.list > data/dev.data.list
+  for model in llm flow hifigan; do
+    torchrun --nnodes=1 --nproc_per_node=$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 \
+      --onnx_path $pretrained_model_dir \
+      --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/magicdata-read/cosyvoice/cosyvoice

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

examples/magicdata-read/cosyvoice/run.sh

@@ -27,7 +27,7 @@ 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 dev test train; do
-    tools/extract_embedding.py --dir data/$x \
+    ../../../tools/extract_embedding.py --dir data/$x \
       --onnx_path $pretrained_model_dir/campplus.onnx
   done
 fi
@@ -35,7 +35,7 @@ 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 dev test train; do
-    tools/extract_speech_token.py --dir data/$x \
+    ../../../tools/extract_speech_token.py --dir data/$x \
       --onnx_path $pretrained_model_dir/speech_tokenizer_v1.onnx
   done
 fi
@@ -44,7 +44,7 @@ if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
   echo "Prepare required parquet format data, you should have prepared wav.scp/text/utt2spk/spk2utt/utt2embedding.pt/spk2embedding.pt/utt2speech_token.pt"
   for x in dev test train; do
     mkdir -p data/$x/parquet
-    tools/make_parquet_list.py --num_utts_per_parquet 1000 \
+    ../../../tools/make_parquet_list.py --num_utts_per_parquet 1000 \
       --num_processes 10 \
       --src_dir data/$x \
       --des_dir data/$x/parquet
@@ -69,7 +69,7 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
   for model in llm flow hifigan; do
     torchrun --nnodes=1 --nproc_per_node=$num_gpus \
         --rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:0" \
-      cosyvoice/bin/train.py \
+      ../../../cosyvoice/bin/train.py \
       --train_engine $train_engine \
       --config conf/cosyvoice.yaml \
       --train_data data/train.data.list \

examples/magicdata-read/cosyvoice/tools

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

requirements.txt

@@ -17,6 +17,7 @@ lightning==2.2.4
 matplotlib==3.7.5
 modelscope==1.20.0
 networkx==3.1
+numpy==1.26.4
 omegaconf==2.3.0
 onnx==1.16.0
 onnxruntime-gpu==1.18.0; sys_platform == 'linux'
@@ -29,12 +30,13 @@ 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==4.40.1
+transformers==4.51.3
+x-transformers==2.11.24
 uvicorn==0.30.0
 wetext==0.0.4
 wget==3.2

runtime/python/Dockerfile

@@ -9,5 +9,5 @@ RUN apt-get -y install git unzip git-lfs g++
 RUN git lfs install
 RUN git clone --recursive https://github.com/FunAudioLLM/CosyVoice.git
 # here we use python==3.10 because we cannot find an image which have both python3.8 and torch2.0.1-cu118 installed
-RUN cd CosyVoice && pip3 install -r requirements.txt -i https://mirrors.aliyun.com/pypi/simple/ --trusted-host=mirrors.aliyun.com
+RUN cd CosyVoice && pip3 install -r requirements.txt -i https://mirrors.aliyun.com/pypi/simple/ --trusted-host=mirrors.aliyun.com --no-cache-dir
 RUN cd CosyVoice/runtime/python/grpc && python3 -m grpc_tools.protoc -I. --python_out=. --grpc_python_out=. cosyvoice.proto

runtime/python/fastapi/server.py

@@ -24,7 +24,7 @@ import numpy as np
 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 load_wav
 
 app = FastAPI()
@@ -88,14 +88,8 @@ if __name__ == '__main__':
                         default=50000)
     parser.add_argument('--model_dir',
                         type=str,
-                        default='iic/CosyVoice-300M',
+                        default='iic/CosyVoice2-0.5B',
                         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)
     uvicorn.run(app, host="0.0.0.0", port=args.port)

runtime/python/grpc/server.py

@@ -25,7 +25,7 @@ import numpy as np
 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
 
 logging.basicConfig(level=logging.DEBUG,
                     format='%(asctime)s %(levelname)s %(message)s')
@@ -33,13 +33,7 @@ logging.basicConfig(level=logging.DEBUG,
 
 class CosyVoiceServiceImpl(cosyvoice_pb2_grpc.CosyVoiceServicer):
     def __init__(self, args):
-        try:
-            self.cosyvoice = CosyVoice(args.model_dir, trt_concurrent=args.max_conc)
-        except Exception:
-            try:
-                self.cosyvoice = CosyVoice2(args.model_dir, trt_concurrent=args.max_conc)
-            except Exception:
-                raise TypeError('no valid model_type!')
+        self.cosyvoice = AutoModel(model_dir=args.model_dir)
         logging.info('grpc service initialized')
 
     def Inference(self, request, context):
@@ -90,7 +84,7 @@ if __name__ == '__main__':
                         default=4)
     parser.add_argument('--model_dir',
                         type=str,
-                        default='iic/CosyVoice-300M',
+                        default='iic/CosyVoice2-0.5B',
                         help='local path or modelscope repo id')
     args = parser.parse_args()
     main()

runtime/triton_trtllm/Dockerfile.server

@@ -0,0 +1,8 @@
+FROM nvcr.io/nvidia/tritonserver:25.06-trtllm-python-py3
+LABEL maintainer="zhangyuekai@foxmail.com"
+
+RUN apt-get update && apt-get install -y cmake
+RUN git clone https://github.com/pytorch/audio.git && cd audio && git checkout c670ad8 && PATH=/usr/local/cuda/bin:$PATH python3 setup.py develop
+COPY ./requirements.txt /workspace/requirements.txt
+RUN pip install -r /workspace/requirements.txt
+WORKDIR /workspace

runtime/triton_trtllm/README.DIT.md

@@ -0,0 +1,141 @@
+## Accelerating CosyVoice with DiT-based Token2Wav, NVIDIA Triton Inference Server and TensorRT-LLM
+
+Contributed by Yuekai Zhang (NVIDIA).
+
+This document describes how to accelerate CosyVoice with a DiT-based Token2Wav module from Step-Audio2, using NVIDIA Triton Inference Server and TensorRT-LLM.
+
+### Quick Start
+
+Launch the service directly with Docker Compose:
+```sh
+docker compose -f docker-compose.dit.yml up
+```
+
+### Build the Docker Image
+
+To build the image from scratch:
+```sh
+docker build . -f Dockerfile.server -t soar97/triton-cosyvoice:25.06
+```
+
+### Run a Docker Container
+```sh
+your_mount_dir=/mnt:/mnt
+docker run -it --name "cosyvoice-server" --gpus all --net host -v $your_mount_dir --shm-size=2g soar97/triton-cosyvoice:25.06
+```
+
+### Understanding `run_stepaudio2_dit_token2wav.sh`
+
+The `run_stepaudio2_dit_token2wav.sh` script orchestrates the entire workflow through numbered stages.
+
+You can run a subset of stages with:
+```sh
+bash run_stepaudio2_dit_token2wav.sh <start_stage> <stop_stage>
+```
+- `<start_stage>`: The stage to start from.
+- `<stop_stage>`: The stage to stop after.
+
+**Stages:**
+
+- **Stage -1**: Clones the `Step-Audio2` and `CosyVoice` repositories.
+- **Stage 0**: Downloads the `cosyvoice2_llm`, `CosyVoice2-0.5B`, and `Step-Audio-2-mini` models.
+- **Stage 1**: Converts the HuggingFace checkpoint for the LLM to the TensorRT-LLM format and builds the TensorRT engines.
+- **Stage 2**: Creates the Triton model repository, including configurations for `cosyvoice2_dit` and `token2wav_dit`.
+- **Stage 3**: Launches the Triton Inference Server for Token2Wav module and uses `trtllm-serve` to deploy Cosyvoice2 LLM.
+- **Stage 4**: Runs the gRPC benchmark client for performance testing.
+- **Stage 5**: Runs the offline TTS inference benchmark test.
+- **Stage 6**: Runs a standalone inference script for the Step-Audio2-mini DiT Token2Wav model.
+- **Stage 7**: Launches servers in a disaggregated setup, with the LLM on GPU 0 and Token2Wav servers on GPUs 1-3.
+- **Stage 8**: Runs the benchmark client for the disaggregated server configuration.
+### Export Models and Launch Server
+
+Inside the Docker container, prepare the models and start the Triton server by running stages 0-3:
+```sh
+# This command runs stages 0, 1, 2, and 3
+bash run_stepaudio2_dit_token2wav.sh 0 3
+```
+
+### Benchmark with client-server mode
+
+To benchmark the running Triton server, run stage 4:
+```sh
+bash run_stepaudio2_dit_token2wav.sh 4 4
+
+# You can customize parameters such as the number of tasks inside the script.
+```
+The following results were obtained by decoding on a single L20 GPU with the `yuekai/seed_tts_cosy2` dataset.
+
+#### Total Request Latency
+
+| Concurrent Tasks | RTF    | Average (ms) | 50th Percentile (ms) | 90th Percentile (ms) | 95th Percentile (ms) | 99th Percentile (ms) |
+| ---------------- | ------ | ------------ | -------------------- | -------------------- | -------------------- | -------------------- |
+| 1                | 0.1228 | 833.66       | 779.98               | 1297.05              | 1555.97              | 1653.02              |
+| 2                | 0.0901 | 1166.23      | 1124.69              | 1762.76              | 1900.64              | 2204.14              |
+| 4                | 0.0741 | 1849.30      | 1759.42              | 2624.50              | 2822.20              | 3128.42              |
+| 6                | 0.0774 | 2936.13      | 3054.64              | 3849.60              | 3900.49              | 4245.79              |
+| 8                | 0.0691 | 3408.56      | 3434.98              | 4547.13              | 5047.76              | 5346.53              |
+| 10               | 0.0707 | 4306.56      | 4343.44              | 5769.64              | 5876.09              | 5939.79              |
+
+#### First Chunk Latency
+
+| Concurrent Tasks | Average (ms) | 50th Percentile (ms) | 90th Percentile (ms) | 95th Percentile (ms) | 99th Percentile (ms) |
+| ---------------- | ------------ | -------------------- | -------------------- | -------------------- | -------------------- |
+| 1                | 197.50       | 196.13               | 214.65               | 215.96               | 229.21               |
+| 2                |  281.15       | 278.20               | 345.18               | 361.79               | 395.97               |
+| 4                |  510.65       | 530.50               | 630.13               | 642.44               | 666.65               |
+| 6                |  921.54       | 918.86               | 1079.97              | 1265.22              | 1524.41              |
+| 8                |  1019.95      | 1085.26              | 1371.05              | 1402.24              | 1410.66              |
+| 10               |  1214.98      | 1293.54              | 1575.36              | 1654.51              | 2161.76              |
+
+### Benchmark with offline inference mode
+For offline inference mode benchmark, please run stage 5:
+```sh
+bash run_stepaudio2_dit_token2wav.sh 5 5
+```
+
+The following results were obtained by decoding on a single L20 GPU with the `yuekai/seed_tts_cosy2` dataset.
+
+#### Offline TTS (Cosyvoice2 0.5B LLM + StepAudio2 DiT Token2Wav)
+| Backend | Batch Size | llm_time_seconds  | total_time_seconds | RTF |
+|---------|------------|------------------|-----------------------|--|
+| TRTLLM | 16 | 2.01 |  5.03 | 0.0292 |
+
+
+### Disaggregated Server
+When the LLM and token2wav components are deployed on the same GPU, they compete for resources. To optimize performance, we use a disaggregated setup where the LLM is deployed on one dedicated L20 GPU, taking advantage of in-flight batching for inference. The token2wav module is deployed on separate, dedicated GPUs.
+
+The table below shows the first chunk latency results for this configuration. In our tests, we deploy two token2wav instances on each dedicated token2wav GPU.
+
+| token2wav_num_gpu | concurrent_task_per_instance | concurrent_tasks_per_gpu | avg (ms) | p50 (ms) | p90 (ms) | p99 (ms) |
+|---|---|---|---|---|---|---|
+| 1 | 1 | 1.00 | 218.53 | 217.86 | 254.07 | 296.49 |
+| 2 | 1 | 1.33 | 218.82 | 219.21 | 256.62 | 303.13 |
+| 3 | 1 | 1.50 | 229.08 | 223.27 | 302.13 | 324.41 |
+| 4 | 1 | 1.60 | 203.87 | 198.23 | 254.92 | 279.31 |
+| 1 | 2 | 2.00 | 293.46 | 280.53 | 370.81 | 407.40 |
+| 2 | 2 | 2.67 | 263.38 | 236.84 | 350.82 | 397.39 |
+| 3 | 2 | 3.00 | 308.09 | 275.48 | 385.22 | 521.45 |
+| 4 | 2 | 3.20 | 271.85 | 253.25 | 359.03 | 387.91 |
+| 1 | 3 | 3.00 | 389.15 | 373.01 | 469.22 | 542.89 |
+| 2 | 3 | 4.00 | 403.48 | 394.80 | 481.24 | 507.75 |
+| 3 | 3 | 4.50 | 406.33 | 391.28 | 495.43 | 571.29 |
+| 4 | 3 | 4.80 | 436.72 | 383.81 | 638.44 | 879.23 |
+| 1 | 4 | 4.00 | 520.12 | 493.98 | 610.38 | 739.85 |
+| 2 | 4 | 5.33 | 494.60 | 490.50 | 605.93 | 708.09 |
+| 3 | 4 | 6.00 | 538.23 | 508.33 | 687.62 | 736.96 |
+| 4 | 4 | 6.40 | 579.68 | 546.20 | 721.53 | 958.04 |
+| 1 | 5 | 5.00 | 635.02 | 623.30 | 786.85 | 819.84 |
+| 2 | 5 | 6.67 | 598.23 | 617.09 | 741.00 | 788.96 |
+| 3 | 5 | 7.50 | 644.78 | 684.40 | 786.45 | 1009.45 |
+| 4 | 5 | 8.00 | 733.92 | 642.26 | 1024.79 | 1281.55 |
+| 1 | 6 | 6.00 | 715.38 | 745.68 | 887.04 | 906.68 |
+| 2 | 6 | 8.00 | 748.31 | 753.94 | 873.59 | 1007.14 |
+| 3 | 6 | 9.00 | 900.27 | 822.28 | 1431.14 | 1800.23 |
+| 4 | 6 | 9.60 | 857.54 | 820.33 | 1150.30 | 1298.53 |
+
+The `concurrent_task_per_gpu` is calculated as:
+`concurrent_task_per_gpu = concurrent_task_per_instance * num_token2wav_instance_per_gpu (2) * token2wav_gpus / (token2wav_gpus + llm_gpus (1))`
+
+### Acknowledgements
+
+This work originates from the NVIDIA CISI project. For more multimodal resources, please see [mair-hub](https://github.com/nvidia-china-sae/mair-hub).

runtime/triton_trtllm/README.md

@@ -0,0 +1,146 @@
+## Accelerating CosyVoice with NVIDIA Triton Inference Server and TensorRT-LLM
+
+Contributed by Yuekai Zhang (NVIDIA).
+
+### Quick Start
+
+Launch the service directly with Docker Compose:
+```sh
+docker compose up
+```
+
+### Build the Docker Image
+
+To build the image from scratch:
+```sh
+docker build . -f Dockerfile.server -t soar97/triton-cosyvoice:25.06
+```
+
+### Run a Docker Container
+```sh
+your_mount_dir=/mnt:/mnt
+docker run -it --name "cosyvoice-server" --gpus all --net host -v $your_mount_dir --shm-size=2g soar97/triton-cosyvoice:25.06
+```
+
+### Understanding `run.sh`
+
+The `run.sh` script orchestrates the entire workflow through numbered stages.
+
+You can run a subset of stages with:
+```sh
+bash run.sh <start_stage> <stop_stage> [service_type]
+```
+- `<start_stage>`: The stage to start from (0-5).
+- `<stop_stage>`: The stage to stop after (0-5).
+
+**Stages:**
+
+- **Stage 0**: Downloads the `cosyvoice-2 0.5B` model from HuggingFace.
+- **Stage 1**: Converts the HuggingFace checkpoint to the TensorRT-LLM format and builds the TensorRT engines.
+- **Stage 2**: Creates the Triton model repository and configures the model files. The configuration is adjusted based on whether `Decoupled=True` (streaming) or `Decoupled=False` (offline) will be used.
+- **Stage 3**: Launches the Triton Inference Server.
+- **Stage 4**: Runs the single-utterance HTTP client for testing.
+- **Stage 5**: Runs the gRPC benchmark client.
+- **Stage 6**: Runs the offline inference benchmark test.
+
+### Export Models and Launch Server
+
+Inside the Docker container, prepare the models and start the Triton server by running stages 0-3:
+```sh
+# This command runs stages 0, 1, 2, and 3
+bash run.sh 0 3
+```
+> [!TIP]
+> Both streaming and offline (non-streaming) TTS modes are supported. For streaming TTS, set `Decoupled=True`. For offline TTS, set `Decoupled=False`. You need to rerun stage 2 if you switch between modes.
+
+### Single-Utterance HTTP Client
+
+Sends a single HTTP inference request. This is intended for testing the offline TTS mode (`Decoupled=False`):
+```sh
+bash run.sh 4 4
+```
+
+### Benchmark with client-server mode
+
+To benchmark the running Triton server, pass `streaming` or `offline` as the third argument:
+```sh
+bash run.sh 5 5 # [streaming|offline]
+
+# You can also customize parameters such as the number of tasks and the dataset split:
+# python3 client_grpc.py --num-tasks 2 --huggingface-dataset yuekai/seed_tts_cosy2 --split-name test_zh --mode [streaming|offline]
+```
+> [!TIP]
+> It is recommended to run the benchmark multiple times to get stable results after the initial server warm-up.
+
+### Benchmark with offline inference mode
+For offline inference mode benchmark, please check the below command:
+```sh
+# install FlashCosyVoice for token2wav batching
+# git clone https://github.com/yuekaizhang/FlashCosyVoice.git /workspace/FlashCosyVoice -b trt
+# cd /workspace/FlashCosyVoice
+# pip install -e .
+# cd -
+# wget https://huggingface.co/yuekai/cosyvoice2_flow_onnx/resolve/main/flow.decoder.estimator.fp32.dynamic_batch.onnx -O $model_scope_model_local_dir/flow.decoder.estimator.fp32.dynamic_batch.onnx
+
+bash run.sh 6 6
+
+# You can also switch to huggingface backend by setting backend=hf
+```
+
+
+### Benchmark Results
+The following results were obtained by decoding on a single L20 GPU with 26 prompt audio/target text pairs from the [yuekai/seed_tts](https://huggingface.co/datasets/yuekai/seed_tts) dataset (approximately 170 seconds of audio):
+
+**Client-Server Mode: Streaming TTS (First Chunk Latency)**
+| Mode | Concurrency | Avg Latency (ms) | P50 Latency (ms) | RTF |
+|---|---|---|---|---|
+| Streaming, use_spk2info_cache=False | 1 | 220.43 | 218.07 | 0.1237 |
+| Streaming, use_spk2info_cache=False | 2 | 476.97 | 369.25 | 0.1022 |
+| Streaming, use_spk2info_cache=False | 4 | 1107.34 | 1243.75| 0.0922 |
+| Streaming, use_spk2info_cache=True | 1 | 189.88 | 184.81 | 0.1155 |
+| Streaming, use_spk2info_cache=True | 2 | 323.04 | 316.83 | 0.0905 |
+| Streaming, use_spk2info_cache=True | 4 | 977.68 | 903.68| 0.0733 |
+
+> If your service only needs a fixed speaker, you can set `use_spk2info_cache=True` in `run.sh`. To add more speakers, refer to the instructions [here](https://github.com/qi-hua/async_cosyvoice?tab=readme-ov-file#9-spk2info-%E8%AF%B4%E6%98%8E).
+
+**Client-Server Mode: Offline TTS (Full Sentence Latency)**
+| Mode | Note | Concurrency | Avg Latency (ms) | P50 Latency (ms) | RTF |
+|---|---|---|---|---|---|
+| Offline, Decoupled=False, use_spk2info_cache=False | [Commit](https://github.com/yuekaizhang/CosyVoice/commit/b44f12110224cb11c03aee4084b1597e7b9331cb) | 1 | 758.04 | 615.79 | 0.0891 |
+| Offline, Decoupled=False, use_spk2info_cache=False | [Commit](https://github.com/yuekaizhang/CosyVoice/commit/b44f12110224cb11c03aee4084b1597e7b9331cb) | 2 | 1025.93 | 901.68 | 0.0657 |
+| Offline, Decoupled=False, use_spk2info_cache=False | [Commit](https://github.com/yuekaizhang/CosyVoice/commit/b44f12110224cb11c03aee4084b1597e7b9331cb) | 4 | 1914.13 | 1783.58 | 0.0610 |
+
+**Offline Inference Mode: Hugginface LLM V.S. TensorRT-LLM**
+| Backend | Batch Size | llm_time_seconds  | total_time_seconds | RTF |
+|---------|------------|------------------|-----------------------|--|
+| HF | 1 | 39.26 |  44.31 | 0.2494 |
+| HF | 2 | 30.54 | 35.62 | 0.2064 |
+| HF | 4 | 18.63 |  23.90 | 0.1421 |
+| HF | 8 | 11.22 | 16.45 | 0.0947 |
+| HF | 16 | 8.42 | 13.78 | 0.0821 |
+| TRTLLM | 1 | 12.46 | 17.31 | 0.0987 |
+| TRTLLM | 2 | 7.64 |12.65 | 0.0739 |
+| TRTLLM | 4 | 4.89 |  9.38 | 0.0539 |
+| TRTLLM | 8 | 2.92 |  7.23 | 0.0418 |
+| TRTLLM | 16 | 2.01 |  6.63 | 0.0386 |
+### OpenAI-Compatible Server
+
+To launch an OpenAI-compatible API service, run the following commands:
+```sh
+git clone https://github.com/yuekaizhang/Triton-OpenAI-Speech.git
+cd Triton-OpenAI-Speech
+pip install -r requirements.txt
+
+# After the Triton service is running, start the FastAPI bridge:
+python3 tts_server.py --url http://localhost:8000 --ref_audios_dir ./ref_audios/ --port 10086 --default_sample_rate 24000
+
+# Test the service with curl:
+bash test/test_cosyvoice.sh
+```
+> [!NOTE]
+> Currently, only the offline TTS mode is compatible with the OpenAI-compatible server.
+
+### Acknowledgements
+
+This work originates from the NVIDIA CISI project. For more multimodal resources, please see [mair-hub](https://github.com/nvidia-china-sae/mair-hub).
+

runtime/triton_trtllm/client_grpc.py

@@ -0,0 +1,922 @@
+# Copyright      2022  Xiaomi Corp.        (authors: Fangjun Kuang)
+#                2023  Nvidia              (authors: Yuekai Zhang)
+#                2023  Recurrent.ai        (authors: Songtao Shi)
+# See LICENSE for clarification regarding multiple authors
+#
+# 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.
+"""
+This script supports to load dataset from huggingface and sends it to the server
+for decoding, in parallel.
+
+Usage:
+num_task=2
+
+# For offline F5-TTS
+python3 client_grpc.py \
+    --server-addr localhost \
+    --model-name f5_tts \
+    --num-tasks $num_task \
+    --huggingface-dataset yuekai/seed_tts \
+    --split-name test_zh \
+    --log-dir ./log_concurrent_tasks_${num_task}
+
+# For offline Spark-TTS-0.5B
+python3 client_grpc.py \
+    --server-addr localhost \
+    --model-name spark_tts \
+    --num-tasks $num_task \
+    --huggingface-dataset yuekai/seed_tts \
+    --split-name wenetspeech4tts \
+    --log-dir ./log_concurrent_tasks_${num_task}
+"""
+
+import argparse
+import asyncio
+import json
+import queue
+import uuid
+import functools
+
+import os
+import time
+import types
+from pathlib import Path
+
+import numpy as np
+import soundfile as sf
+import tritonclient
+import tritonclient.grpc.aio as grpcclient_aio
+import tritonclient.grpc as grpcclient_sync
+from tritonclient.utils import np_to_triton_dtype, InferenceServerException
+
+
+class UserData:
+    def __init__(self):
+        self._completed_requests = queue.Queue()
+        self._first_chunk_time = None
+        self._second_chunk_time = None
+        self._start_time = None
+
+    def record_start_time(self):
+        self._start_time = time.time()
+
+    def get_first_chunk_latency(self):
+        if self._first_chunk_time and self._start_time:
+            return self._first_chunk_time - self._start_time
+        return None
+
+    def get_second_chunk_latency(self):
+        if self._first_chunk_time and self._second_chunk_time:
+            return self._second_chunk_time - self._first_chunk_time
+        return None
+
+
+def callback(user_data, result, error):
+    if not error:
+        if user_data._first_chunk_time is None:
+            user_data._first_chunk_time = time.time()
+        elif user_data._second_chunk_time is None:
+            user_data._second_chunk_time = time.time()
+
+    if error:
+        user_data._completed_requests.put(error)
+    else:
+        user_data._completed_requests.put(result)
+
+
+def stream_callback(user_data_map, result, error):
+    request_id = None
+    if error:
+        print(f"An error occurred in the stream callback: {error}")
+    else:
+        request_id = result.get_response().id
+
+    if request_id:
+        user_data = user_data_map.get(request_id)
+        if user_data:
+            callback(user_data, result, error)
+        else:
+            print(f"Warning: Could not find user_data for request_id {request_id}")
+
+
+def write_triton_stats(stats, summary_file):
+    with open(summary_file, "w") as summary_f:
+        model_stats = stats["model_stats"]
+        for model_state in model_stats:
+            if "last_inference" not in model_state:
+                continue
+            summary_f.write(f"model name is {model_state['name']} \n")
+            model_inference_stats = model_state["inference_stats"]
+            total_queue_time_s = int(model_inference_stats["queue"]["ns"]) / 1e9
+            total_infer_time_s = int(model_inference_stats["compute_infer"]["ns"]) / 1e9
+            total_input_time_s = int(model_inference_stats["compute_input"]["ns"]) / 1e9
+            total_output_time_s = int(model_inference_stats["compute_output"]["ns"]) / 1e9
+            summary_f.write(
+                f"queue time {total_queue_time_s:<5.2f} s, "
+                f"compute infer time {total_infer_time_s:<5.2f} s, "
+                f"compute input time {total_input_time_s:<5.2f} s, "
+                f"compute output time {total_output_time_s:<5.2f} s \n"
+            )
+            model_batch_stats = model_state["batch_stats"]
+            for batch in model_batch_stats:
+                batch_size = int(batch["batch_size"])
+                compute_input = batch["compute_input"]
+                compute_output = batch["compute_output"]
+                compute_infer = batch["compute_infer"]
+                batch_count = int(compute_infer["count"])
+                if batch_count == 0:
+                    continue
+                assert compute_infer["count"] == compute_output["count"] == compute_input["count"]
+                compute_infer_time_ms = int(compute_infer["ns"]) / 1e6
+                compute_input_time_ms = int(compute_input["ns"]) / 1e6
+                compute_output_time_ms = int(compute_output["ns"]) / 1e6
+                summary_f.write(
+                    f"execuate inference with batch_size {batch_size:<2} total {batch_count:<5} times, "
+                    f"total_infer_time {compute_infer_time_ms:<9.2f} ms, "
+                    f"avg_infer_time {compute_infer_time_ms:<9.2f}/{batch_count:<5}="
+                    f"{compute_infer_time_ms / batch_count:.2f} ms, "
+                    f"avg_infer_time_per_sample {compute_infer_time_ms:<9.2f}/{batch_count:<5}/{batch_size}="
+                    f"{compute_infer_time_ms / batch_count / batch_size:.2f} ms \n"
+                )
+                summary_f.write(
+                    f"input {compute_input_time_ms:<9.2f} ms, avg {compute_input_time_ms / batch_count:.2f} ms, "
+                )
+                summary_f.write(
+                    f"output {compute_output_time_ms:<9.2f} ms, avg {compute_output_time_ms / batch_count:.2f} ms \n"
+                )
+
+
+def subtract_stats(stats_after, stats_before):
+    """Subtracts two Triton inference statistics objects."""
+    stats_diff = json.loads(json.dumps(stats_after))
+
+    model_stats_before_map = {
+        s["name"]: {
+            "version": s["version"],
+            "last_inference": s.get("last_inference", 0),
+            "inference_count": s.get("inference_count", 0),
+            "execution_count": s.get("execution_count", 0),
+            "inference_stats": s.get("inference_stats", {}),
+            "batch_stats": s.get("batch_stats", []),
+        }
+        for s in stats_before["model_stats"]
+    }
+
+    for model_stat_after in stats_diff["model_stats"]:
+        model_name = model_stat_after["name"]
+        if model_name in model_stats_before_map:
+            model_stat_before = model_stats_before_map[model_name]
+
+            model_stat_after["inference_count"] = str(
+                int(model_stat_after.get("inference_count", 0)) - int(model_stat_before.get("inference_count", 0))
+            )
+            model_stat_after["execution_count"] = str(
+                int(model_stat_after.get("execution_count", 0)) - int(model_stat_before.get("execution_count", 0))
+            )
+
+            if "inference_stats" in model_stat_after and "inference_stats" in model_stat_before:
+                for key in ["success", "fail", "queue", "compute_input", "compute_infer", "compute_output", "cache_hit", "cache_miss"]:
+                    if key in model_stat_after["inference_stats"] and key in model_stat_before["inference_stats"]:
+                        if "ns" in model_stat_after["inference_stats"][key]:
+                            ns_after = int(model_stat_after["inference_stats"][key]["ns"])
+                            ns_before = int(model_stat_before["inference_stats"][key]["ns"])
+                            model_stat_after["inference_stats"][key]["ns"] = str(ns_after - ns_before)
+                        if "count" in model_stat_after["inference_stats"][key]:
+                            count_after = int(model_stat_after["inference_stats"][key]["count"])
+                            count_before = int(model_stat_before["inference_stats"][key]["count"])
+                            model_stat_after["inference_stats"][key]["count"] = str(count_after - count_before)
+
+            if "batch_stats" in model_stat_after and "batch_stats" in model_stat_before:
+                batch_stats_before_map = {b["batch_size"]: b for b in model_stat_before["batch_stats"]}
+                for batch_stat_after in model_stat_after["batch_stats"]:
+                    bs = batch_stat_after["batch_size"]
+                    if bs in batch_stats_before_map:
+                        batch_stat_before = batch_stats_before_map[bs]
+                        for key in ["compute_input", "compute_infer", "compute_output"]:
+                            if key in batch_stat_after and key in batch_stat_before:
+                                count_after = int(batch_stat_after[key]["count"])
+                                count_before = int(batch_stat_before[key]["count"])
+                                batch_stat_after[key]["count"] = str(count_after - count_before)
+
+                                ns_after = int(batch_stat_after[key]["ns"])
+                                ns_before = int(batch_stat_before[key]["ns"])
+                                batch_stat_after[key]["ns"] = str(ns_after - ns_before)
+    return stats_diff
+
+
+def get_args():
+    parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter)
+
+    parser.add_argument(
+        "--server-addr",
+        type=str,
+        default="localhost",
+        help="Address of the server",
+    )
+
+    parser.add_argument(
+        "--server-port",
+        type=int,
+        default=8001,
+        help="Grpc port of the triton server, default is 8001",
+    )
+
+    parser.add_argument(
+        "--reference-audio",
+        type=str,
+        default=None,
+        help="Path to a single audio file. It can't be specified at the same time with --manifest-dir",
+    )
+
+    parser.add_argument(
+        "--reference-text",
+        type=str,
+        default="",
+        help="",
+    )
+
+    parser.add_argument(
+        "--target-text",
+        type=str,
+        default="",
+        help="",
+    )
+
+    parser.add_argument(
+        "--huggingface-dataset",
+        type=str,
+        default="yuekai/seed_tts",
+        help="dataset name in huggingface dataset hub",
+    )
+
+    parser.add_argument(
+        "--split-name",
+        type=str,
+        default="wenetspeech4tts",
+        choices=["wenetspeech4tts", "test_zh", "test_en", "test_hard"],
+        help="dataset split name, default is 'test'",
+    )
+
+    parser.add_argument(
+        "--manifest-path",
+        type=str,
+        default=None,
+        help="Path to the manifest dir which includes wav.scp trans.txt files.",
+    )
+
+    parser.add_argument(
+        "--model-name",
+        type=str,
+        default="f5_tts",
+        choices=[
+            "f5_tts",
+            "spark_tts",
+            "cosyvoice2",
+            "cosyvoice2_dit"],
+        help="triton model_repo module name to request",
+    )
+
+    parser.add_argument(
+        "--num-tasks",
+        type=int,
+        default=1,
+        help="Number of concurrent tasks for sending",
+    )
+
+    parser.add_argument(
+        "--log-interval",
+        type=int,
+        default=5,
+        help="Controls how frequently we print the log.",
+    )
+
+    parser.add_argument(
+        "--compute-wer",
+        action="store_true",
+        default=False,
+        help="""True to compute WER.
+        """,
+    )
+
+    parser.add_argument(
+        "--log-dir",
+        type=str,
+        required=False,
+        default="./tmp",
+        help="log directory",
+    )
+
+    parser.add_argument(
+        "--mode",
+        type=str,
+        default="offline",
+        choices=["offline", "streaming"],
+        help="Select offline or streaming benchmark mode."
+    )
+    parser.add_argument(
+        "--chunk-overlap-duration",
+        type=float,
+        default=0.1,
+        help="Chunk overlap duration for streaming reconstruction (in seconds)."
+    )
+
+    parser.add_argument(
+        "--use-spk2info-cache",
+        type=str,
+        default="False",
+        help="Use spk2info cache for reference audio.",
+    )
+
+    return parser.parse_args()
+
+
+def load_audio(wav_path, target_sample_rate=16000):
+    assert target_sample_rate == 16000, "hard coding in server"
+    if isinstance(wav_path, dict):
+        waveform = wav_path["array"]
+        sample_rate = wav_path["sampling_rate"]
+    else:
+        waveform, sample_rate = sf.read(wav_path)
+    if sample_rate != target_sample_rate:
+        from scipy.signal import resample
+
+        num_samples = int(len(waveform) * (target_sample_rate / sample_rate))
+        waveform = resample(waveform, num_samples)
+    return waveform, target_sample_rate
+
+
+def prepare_request_input_output(
+    protocol_client,
+    waveform,
+    reference_text,
+    target_text,
+    sample_rate=16000,
+    padding_duration: int = None,
+    use_spk2info_cache: bool = False
+):
+    """Prepares inputs for Triton inference (offline or streaming)."""
+    assert len(waveform.shape) == 1, "waveform should be 1D"
+    lengths = np.array([[len(waveform)]], dtype=np.int32)
+
+    if padding_duration:
+        duration = len(waveform) / sample_rate
+        if reference_text:
+            estimated_target_duration = duration / len(reference_text) * len(target_text)
+        else:
+            estimated_target_duration = duration
+
+        required_total_samples = padding_duration * sample_rate * (
+            (int(estimated_target_duration + duration) // padding_duration) + 1
+        )
+        samples = np.zeros((1, required_total_samples), dtype=np.float32)
+        samples[0, : len(waveform)] = waveform
+    else:
+        samples = waveform.reshape(1, -1).astype(np.float32)
+
+    inputs = [
+        protocol_client.InferInput("reference_wav", samples.shape, np_to_triton_dtype(samples.dtype)),
+        protocol_client.InferInput(
+            "reference_wav_len", lengths.shape, np_to_triton_dtype(lengths.dtype)
+        ),
+        protocol_client.InferInput("reference_text", [1, 1], "BYTES"),
+        protocol_client.InferInput("target_text", [1, 1], "BYTES"),
+    ]
+    inputs[0].set_data_from_numpy(samples)
+    inputs[1].set_data_from_numpy(lengths)
+
+    input_data_numpy = np.array([reference_text], dtype=object)
+    input_data_numpy = input_data_numpy.reshape((1, 1))
+    inputs[2].set_data_from_numpy(input_data_numpy)
+
+    input_data_numpy = np.array([target_text], dtype=object)
+    input_data_numpy = input_data_numpy.reshape((1, 1))
+    inputs[3].set_data_from_numpy(input_data_numpy)
+
+    outputs = [protocol_client.InferRequestedOutput("waveform")]
+    if use_spk2info_cache:
+        inputs = inputs[-1:]
+    return inputs, outputs
+
+
+def run_sync_streaming_inference(
+    sync_triton_client: tritonclient.grpc.InferenceServerClient,
+    model_name: str,
+    inputs: list,
+    outputs: list,
+    request_id: str,
+    user_data: UserData,
+    chunk_overlap_duration: float,
+    save_sample_rate: int,
+    audio_save_path: str,
+):
+    """Helper function to run the blocking sync streaming call."""
+    start_time_total = time.time()
+    user_data.record_start_time()
+
+    sync_triton_client.async_stream_infer(
+        model_name,
+        inputs,
+        request_id=request_id,
+        outputs=outputs,
+        enable_empty_final_response=True,
+    )
+
+    audios = []
+    while True:
+        try:
+            result = user_data._completed_requests.get(timeout=200)
+            if isinstance(result, InferenceServerException):
+                print(f"Received InferenceServerException: {result}")
+                return None, None, None, None
+            response = result.get_response()
+            final = response.parameters["triton_final_response"].bool_param
+            if final is True:
+                break
+
+            audio_chunk = result.as_numpy("waveform").reshape(-1)
+            if audio_chunk.size > 0:
+                audios.append(audio_chunk)
+            else:
+                print("Warning: received empty audio chunk.")
+
+        except queue.Empty:
+            print(f"Timeout waiting for response for request id {request_id}")
+            return None, None, None, None
+
+    end_time_total = time.time()
+    total_request_latency = end_time_total - start_time_total
+    first_chunk_latency = user_data.get_first_chunk_latency()
+    second_chunk_latency = user_data.get_second_chunk_latency()
+
+    if audios:
+        if model_name == "spark_tts":
+            cross_fade_samples = int(chunk_overlap_duration * save_sample_rate)
+            fade_out = np.linspace(1, 0, cross_fade_samples)
+            fade_in = np.linspace(0, 1, cross_fade_samples)
+            reconstructed_audio = None
+
+            if not audios:
+                print("Warning: No audio chunks received.")
+                reconstructed_audio = np.array([], dtype=np.float32)
+            elif len(audios) == 1:
+                reconstructed_audio = audios[0]
+            else:
+                reconstructed_audio = audios[0][:-cross_fade_samples]
+                for i in range(1, len(audios)):
+                    cross_faded_overlap = (audios[i][:cross_fade_samples] * fade_in +
+                                           audios[i - 1][-cross_fade_samples:] * fade_out)
+                    middle_part = audios[i][cross_fade_samples:-cross_fade_samples]
+                    reconstructed_audio = np.concatenate([reconstructed_audio, cross_faded_overlap, middle_part])
+                reconstructed_audio = np.concatenate([reconstructed_audio, audios[-1][-cross_fade_samples:]])
+
+            if reconstructed_audio is not None and reconstructed_audio.size > 0:
+                actual_duration = len(reconstructed_audio) / save_sample_rate
+                sf.write(audio_save_path, reconstructed_audio, save_sample_rate, "PCM_16")
+            else:
+                print("Warning: No audio chunks received or reconstructed.")
+                actual_duration = 0
+        else:
+            reconstructed_audio = np.concatenate(audios)
+            actual_duration = len(reconstructed_audio) / save_sample_rate
+            sf.write(audio_save_path, reconstructed_audio, save_sample_rate, "PCM_16")
+
+    else:
+        print("Warning: No audio chunks received.")
+        actual_duration = 0
+
+    return total_request_latency, first_chunk_latency, second_chunk_latency, actual_duration
+
+
+async def send_streaming(
+    manifest_item_list: list,
+    name: str,
+    server_url: str,
+    protocol_client: types.ModuleType,
+    log_interval: int,
+    model_name: str,
+    audio_save_dir: str = "./",
+    save_sample_rate: int = 16000,
+    chunk_overlap_duration: float = 0.1,
+    padding_duration: int = None,
+    use_spk2info_cache: bool = False,
+):
+    total_duration = 0.0
+    latency_data = []
+    task_id = int(name[5:])
+    sync_triton_client = None
+    user_data_map = {}
+
+    try:
+        print(f"{name}: Initializing sync client for streaming...")
+        sync_triton_client = grpcclient_sync.InferenceServerClient(url=server_url, verbose=False)
+        sync_triton_client.start_stream(callback=functools.partial(stream_callback, user_data_map))
+
+        print(f"{name}: Starting streaming processing for {len(manifest_item_list)} items.")
+        for i, item in enumerate(manifest_item_list):
+            if i % log_interval == 0:
+                print(f"{name}: Processing item {i}/{len(manifest_item_list)}")
+
+            try:
+                waveform, sample_rate = load_audio(item["audio_filepath"], target_sample_rate=16000)
+                reference_text, target_text = item["reference_text"], item["target_text"]
+
+                inputs, outputs = prepare_request_input_output(
+                    protocol_client,
+                    waveform,
+                    reference_text,
+                    target_text,
+                    sample_rate,
+                    padding_duration=padding_duration,
+                    use_spk2info_cache=use_spk2info_cache
+                )
+
+                request_id = str(uuid.uuid4())
+                user_data = UserData()
+                user_data_map[request_id] = user_data
+
+                audio_save_path = os.path.join(audio_save_dir, f"{item['target_audio_path']}.wav")
+                total_request_latency, first_chunk_latency, second_chunk_latency, actual_duration = await asyncio.to_thread(
+                    run_sync_streaming_inference,
+                    sync_triton_client,
+                    model_name,
+                    inputs,
+                    outputs,
+                    request_id,
+                    user_data,
+                    chunk_overlap_duration,
+                    save_sample_rate,
+                    audio_save_path
+                )
+
+                if total_request_latency is not None:
+                    print(
+                        f"{name}: Item {i} - First Chunk Latency: {first_chunk_latency:.4f}s, "
+                        f"Second Chunk Latency: {second_chunk_latency if second_chunk_latency is not None else 'N/A'}, "
+                        f"Total Latency: {total_request_latency:.4f}s, Duration: {actual_duration:.4f}s"
+                    )
+                    latency_data.append((total_request_latency, first_chunk_latency, second_chunk_latency, actual_duration))
+                    total_duration += actual_duration
+                else:
+                    print(f"{name}: Item {i} failed.")
+
+                del user_data_map[request_id]
+
+            except FileNotFoundError:
+                print(f"Error: Audio file not found for item {i}: {item['audio_filepath']}")
+            except Exception as e:
+                print(f"Error processing item {i} ({item['target_audio_path']}): {e}")
+                import traceback
+                traceback.print_exc()
+
+    finally:
+        if sync_triton_client:
+            try:
+                print(f"{name}: Closing stream and sync client...")
+                sync_triton_client.stop_stream()
+                sync_triton_client.close()
+            except Exception as e:
+                print(f"{name}: Error closing sync client: {e}")
+
+    print(f"{name}: Finished streaming processing. Total duration synthesized: {total_duration:.4f}s")
+    return total_duration, latency_data
+
+
+async def send(
+    manifest_item_list: list,
+    name: str,
+    triton_client: tritonclient.grpc.aio.InferenceServerClient,
+    protocol_client: types.ModuleType,
+    log_interval: int,
+    model_name: str,
+    padding_duration: int = None,
+    audio_save_dir: str = "./",
+    save_sample_rate: int = 16000,
+    use_spk2info_cache: bool = False,
+):
+    total_duration = 0.0
+    latency_data = []
+    task_id = int(name[5:])
+
+    for i, item in enumerate(manifest_item_list):
+        if i % log_interval == 0:
+            print(f"{name}: {i}/{len(manifest_item_list)}")
+        waveform, sample_rate = load_audio(item["audio_filepath"], target_sample_rate=16000)
+        reference_text, target_text = item["reference_text"], item["target_text"]
+
+        inputs, outputs = prepare_request_input_output(
+            protocol_client,
+            waveform,
+            reference_text,
+            target_text,
+            sample_rate,
+            padding_duration=padding_duration,
+            use_spk2info_cache=use_spk2info_cache
+        )
+        sequence_id = 100000000 + i + task_id * 10
+        start = time.time()
+        response = await triton_client.infer(model_name, inputs, request_id=str(sequence_id), outputs=outputs)
+
+        audio = response.as_numpy("waveform").reshape(-1)
+        actual_duration = len(audio) / save_sample_rate
+
+        end = time.time() - start
+
+        audio_save_path = os.path.join(audio_save_dir, f"{item['target_audio_path']}.wav")
+        sf.write(audio_save_path, audio, save_sample_rate, "PCM_16")
+
+        latency_data.append((end, actual_duration))
+        total_duration += actual_duration
+
+    return total_duration, latency_data
+
+
+def load_manifests(manifest_path):
+    with open(manifest_path, "r") as f:
+        manifest_list = []
+        for line in f:
+            assert len(line.strip().split("|")) == 4
+            utt, prompt_text, prompt_wav, gt_text = line.strip().split("|")
+            utt = Path(utt).stem
+            if not os.path.isabs(prompt_wav):
+                prompt_wav = os.path.join(os.path.dirname(manifest_path), prompt_wav)
+            manifest_list.append(
+                {
+                    "audio_filepath": prompt_wav,
+                    "reference_text": prompt_text,
+                    "target_text": gt_text,
+                    "target_audio_path": utt,
+                }
+            )
+    return manifest_list
+
+
+def split_data(data, k):
+    n = len(data)
+    if n < k:
+        print(f"Warning: the length of the input list ({n}) is less than k ({k}). Setting k to {n}.")
+        k = n
+
+    quotient = n // k
+    remainder = n % k
+
+    result = []
+    start = 0
+    for i in range(k):
+        if i < remainder:
+            end = start + quotient + 1
+        else:
+            end = start + quotient
+
+        result.append(data[start:end])
+        start = end
+
+    return result
+
+
+async def main():
+    args = get_args()
+    url = f"{args.server_addr}:{args.server_port}"
+
+    triton_client = None
+    protocol_client = None
+    if args.mode == "offline":
+        print("Initializing gRPC client for offline mode...")
+        triton_client = grpcclient_aio.InferenceServerClient(url=url, verbose=False)
+        protocol_client = grpcclient_aio
+    elif args.mode == "streaming":
+        print("Initializing gRPC client for streaming mode...")
+        protocol_client = grpcclient_sync
+    else:
+        raise ValueError(f"Invalid mode: {args.mode}")
+
+    if args.reference_audio:
+        args.num_tasks = 1
+        args.log_interval = 1
+        manifest_item_list = [
+            {
+                "reference_text": args.reference_text,
+                "target_text": args.target_text,
+                "audio_filepath": args.reference_audio,
+                "target_audio_path": "test",
+            }
+        ]
+    elif args.huggingface_dataset:
+        import datasets
+
+        dataset = datasets.load_dataset(
+            args.huggingface_dataset,
+            split=args.split_name,
+            trust_remote_code=True,
+        )
+        manifest_item_list = []
+        for i in range(len(dataset)):
+            manifest_item_list.append(
+                {
+                    "audio_filepath": dataset[i]["prompt_audio"],
+                    "reference_text": dataset[i]["prompt_text"],
+                    "target_audio_path": dataset[i]["id"],
+                    "target_text": dataset[i]["target_text"],
+                }
+            )
+    else:
+        manifest_item_list = load_manifests(args.manifest_path)
+
+    stats_client = None
+    stats_before = None
+    try:
+        print("Initializing temporary async client for fetching stats...")
+        stats_client = grpcclient_aio.InferenceServerClient(url=url, verbose=False)
+        print("Fetching inference statistics before running tasks...")
+        stats_before = await stats_client.get_inference_statistics(model_name="", as_json=True)
+    except Exception as e:
+        print(f"Could not retrieve statistics before running tasks: {e}")
+
+    num_tasks = min(args.num_tasks, len(manifest_item_list))
+    manifest_item_list = split_data(manifest_item_list, num_tasks)
+
+    os.makedirs(args.log_dir, exist_ok=True)
+    args.use_spk2info_cache = args.use_spk2info_cache == "True" or args.use_spk2info_cache == "true"
+    tasks = []
+    start_time = time.time()
+    for i in range(num_tasks):
+        if args.mode == "offline":
+            task = asyncio.create_task(
+                send(
+                    manifest_item_list[i],
+                    name=f"task-{i}",
+                    triton_client=triton_client,
+                    protocol_client=protocol_client,
+                    log_interval=args.log_interval,
+                    model_name=args.model_name,
+                    audio_save_dir=args.log_dir,
+                    padding_duration=1,
+                    save_sample_rate=16000 if args.model_name == "spark_tts" else 24000,
+                    use_spk2info_cache=args.use_spk2info_cache,
+                )
+            )
+        elif args.mode == "streaming":
+            task = asyncio.create_task(
+                send_streaming(
+                    manifest_item_list[i],
+                    name=f"task-{i}",
+                    server_url=url,
+                    protocol_client=protocol_client,
+                    log_interval=args.log_interval,
+                    model_name=args.model_name,
+                    audio_save_dir=args.log_dir,
+                    padding_duration=10,
+                    save_sample_rate=16000 if args.model_name == "spark_tts" else 24000,
+                    chunk_overlap_duration=args.chunk_overlap_duration,
+                    use_spk2info_cache=args.use_spk2info_cache,
+                )
+            )
+        tasks.append(task)
+
+    ans_list = await asyncio.gather(*tasks)
+
+    end_time = time.time()
+    elapsed = end_time - start_time
+
+    total_duration = 0.0
+    latency_data = []
+    for ans in ans_list:
+        if ans:
+            total_duration += ans[0]
+            latency_data.extend(ans[1])
+        else:
+            print("Warning: A task returned None, possibly due to an error.")
+
+    if total_duration == 0:
+        print("Total synthesized duration is zero. Cannot calculate RTF or latency percentiles.")
+        rtf = float('inf')
+    else:
+        rtf = elapsed / total_duration
+
+    s = f"Mode: {args.mode}\n"
+    s += f"RTF: {rtf:.4f}\n"
+    s += f"total_duration: {total_duration:.3f} seconds\n"
+    s += f"({total_duration / 3600:.2f} hours)\n"
+    s += f"processing time: {elapsed:.3f} seconds ({elapsed / 3600:.2f} hours)\n"
+
+    if latency_data:
+        if args.mode == "offline":
+            latency_list = [chunk_end for (chunk_end, chunk_duration) in latency_data]
+            if latency_list:
+                latency_ms = sum(latency_list) / float(len(latency_list)) * 1000.0
+                latency_variance = np.var(latency_list, dtype=np.float64) * 1000.0
+                s += f"latency_variance: {latency_variance:.2f}\n"
+                s += f"latency_50_percentile_ms: {np.percentile(latency_list, 50) * 1000.0:.2f}\n"
+                s += f"latency_90_percentile_ms: {np.percentile(latency_list, 90) * 1000.0:.2f}\n"
+                s += f"latency_95_percentile_ms: {np.percentile(latency_list, 95) * 1000.0:.2f}\n"
+                s += f"latency_99_percentile_ms: {np.percentile(latency_list, 99) * 1000.0:.2f}\n"
+                s += f"average_latency_ms: {latency_ms:.2f}\n"
+            else:
+                s += "No latency data collected for offline mode.\n"
+
+        elif args.mode == "streaming":
+            total_latency_list = [total for (total, first, second, duration) in latency_data if total is not None]
+            first_chunk_latency_list = [first for (total, first, second, duration) in latency_data if first is not None]
+            second_chunk_latency_list = [second for (total, first, second, duration) in latency_data if second is not None]
+
+            s += "\n--- Total Request Latency ---\n"
+            if total_latency_list:
+                avg_total_latency_ms = sum(total_latency_list) / len(total_latency_list) * 1000.0
+                variance_total_latency = np.var(total_latency_list, dtype=np.float64) * 1000.0
+                s += f"total_request_latency_variance: {variance_total_latency:.2f}\n"
+                s += f"total_request_latency_50_percentile_ms: {np.percentile(total_latency_list, 50) * 1000.0:.2f}\n"
+                s += f"total_request_latency_90_percentile_ms: {np.percentile(total_latency_list, 90) * 1000.0:.2f}\n"
+                s += f"total_request_latency_95_percentile_ms: {np.percentile(total_latency_list, 95) * 1000.0:.2f}\n"
+                s += f"total_request_latency_99_percentile_ms: {np.percentile(total_latency_list, 99) * 1000.0:.2f}\n"
+                s += f"average_total_request_latency_ms: {avg_total_latency_ms:.2f}\n"
+            else:
+                s += "No total request latency data collected.\n"
+
+            s += "\n--- First Chunk Latency ---\n"
+            if first_chunk_latency_list:
+                avg_first_chunk_latency_ms = sum(first_chunk_latency_list) / len(first_chunk_latency_list) * 1000.0
+                variance_first_chunk_latency = np.var(first_chunk_latency_list, dtype=np.float64) * 1000.0
+                s += f"first_chunk_latency_variance: {variance_first_chunk_latency:.2f}\n"
+                s += f"first_chunk_latency_50_percentile_ms: {np.percentile(first_chunk_latency_list, 50) * 1000.0:.2f}\n"
+                s += f"first_chunk_latency_90_percentile_ms: {np.percentile(first_chunk_latency_list, 90) * 1000.0:.2f}\n"
+                s += f"first_chunk_latency_95_percentile_ms: {np.percentile(first_chunk_latency_list, 95) * 1000.0:.2f}\n"
+                s += f"first_chunk_latency_99_percentile_ms: {np.percentile(first_chunk_latency_list, 99) * 1000.0:.2f}\n"
+                s += f"average_first_chunk_latency_ms: {avg_first_chunk_latency_ms:.2f}\n"
+            else:
+                s += "No first chunk latency data collected (check for errors or if all requests failed before first chunk).\n"
+
+            s += "\n--- Second Chunk Latency ---\n"
+            if second_chunk_latency_list:
+                avg_second_chunk_latency_ms = sum(second_chunk_latency_list) / len(second_chunk_latency_list) * 1000.0
+                variance_second_chunk_latency = np.var(second_chunk_latency_list, dtype=np.float64) * 1000.0
+                s += f"second_chunk_latency_variance: {variance_second_chunk_latency:.2f}\n"
+                s += f"second_chunk_latency_50_percentile_ms: {np.percentile(second_chunk_latency_list, 50) * 1000.0:.2f}\n"
+                s += f"second_chunk_latency_90_percentile_ms: {np.percentile(second_chunk_latency_list, 90) * 1000.0:.2f}\n"
+                s += f"second_chunk_latency_95_percentile_ms: {np.percentile(second_chunk_latency_list, 95) * 1000.0:.2f}\n"
+                s += f"second_chunk_latency_99_percentile_ms: {np.percentile(second_chunk_latency_list, 99) * 1000.0:.2f}\n"
+                s += f"average_second_chunk_latency_ms: {avg_second_chunk_latency_ms:.2f}\n"
+            else:
+                s += "No second chunk latency data collected (check for errors or if all requests failed before second chunk).\n"
+    else:
+        s += "No latency data collected.\n"
+
+    print(s)
+    if args.manifest_path:
+        name = Path(args.manifest_path).stem
+    elif args.split_name:
+        name = args.split_name
+    elif args.reference_audio:
+        name = Path(args.reference_audio).stem
+    else:
+        name = "results"
+    with open(f"{args.log_dir}/rtf-{name}.txt", "w") as f:
+        f.write(s)
+
+    try:
+        if stats_client and stats_before:
+            print("Fetching inference statistics after running tasks...")
+            stats_after = await stats_client.get_inference_statistics(model_name="", as_json=True)
+
+            print("Calculating statistics difference...")
+            stats = subtract_stats(stats_after, stats_before)
+
+            print("Fetching model config...")
+            metadata = await stats_client.get_model_config(model_name=args.model_name, as_json=True)
+
+            write_triton_stats(stats, f"{args.log_dir}/stats_summary-{name}.txt")
+
+            with open(f"{args.log_dir}/model_config-{name}.json", "w") as f:
+                json.dump(metadata, f, indent=4)
+        else:
+            print("Stats client not available or initial stats were not fetched. Skipping stats reporting.")
+
+    except Exception as e:
+        print(f"Could not retrieve statistics or config: {e}")
+    finally:
+        if stats_client:
+            try:
+                print("Closing temporary async stats client...")
+                await stats_client.close()
+            except Exception as e:
+                print(f"Error closing async stats client: {e}")
+
+
+if __name__ == "__main__":
+    async def run_main():
+        try:
+            await main()
+        except Exception as e:
+            print(f"An error occurred in main: {e}")
+            import traceback
+            traceback.print_exc()
+
+    asyncio.run(run_main())

runtime/triton_trtllm/client_http.py

@@ -0,0 +1,172 @@
+# Copyright 2025, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions
+# are met:
+#  * Redistributions of source code must retain the above copyright
+#    notice, this list of conditions and the following disclaimer.
+#  * Redistributions in binary form must reproduce the above copyright
+#    notice, this list of conditions and the following disclaimer in the
+#    documentation and/or other materials provided with the distribution.
+#  * Neither the name of NVIDIA CORPORATION nor the names of its
+#    contributors may be used to endorse or promote products derived
+#    from this software without specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS ``AS IS'' AND ANY
+# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+# PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+# OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+# (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+import requests
+import soundfile as sf
+import numpy as np
+import argparse
+
+
+def get_args():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--server-url",
+        type=str,
+        default="localhost:8000",
+        help="Address of the server",
+    )
+
+    parser.add_argument(
+        "--reference-audio",
+        type=str,
+        default="../../example/prompt_audio.wav",
+        help="Path to a single audio file. It can't be specified at the same time with --manifest-dir",
+    )
+
+    parser.add_argument(
+        "--reference-text",
+        type=str,
+        default="吃燕窝就选燕之屋，本节目由26年专注高品质燕窝的燕之屋冠名播出。豆奶牛奶换着喝，营养更均衡，本节目由豆本豆豆奶特约播出。",
+        help="",
+    )
+
+    parser.add_argument(
+        "--target-text",
+        type=str,
+        default="身临其境，换新体验。塑造开源语音合成新范式，让智能语音更自然。",
+        help="",
+    )
+
+    parser.add_argument(
+        "--model-name",
+        type=str,
+        default="spark_tts",
+        choices=[
+            "f5_tts",
+            "spark_tts",
+            "cosyvoice2"],
+        help="triton model_repo module name to request",
+    )
+
+    parser.add_argument(
+        "--output-audio",
+        type=str,
+        default="output.wav",
+        help="Path to save the output audio",
+    )
+    return parser.parse_args()
+
+
+def prepare_request(
+    waveform,
+    reference_text,
+    target_text,
+    sample_rate=16000,
+    padding_duration: int = None,
+    audio_save_dir: str = "./",
+):
+    assert len(waveform.shape) == 1, "waveform should be 1D"
+    lengths = np.array([[len(waveform)]], dtype=np.int32)
+    if padding_duration:
+        # padding to nearset 10 seconds
+        samples = np.zeros(
+            (
+                1,
+                padding_duration
+                * sample_rate
+                * ((int(len(waveform) / sample_rate) // padding_duration) + 1),
+            ),
+            dtype=np.float32,
+        )
+
+        samples[0, : len(waveform)] = waveform
+    else:
+        samples = waveform
+
+    samples = samples.reshape(1, -1).astype(np.float32)
+
+    data = {
+        "inputs": [
+            {
+                "name": "reference_wav",
+                "shape": samples.shape,
+                "datatype": "FP32",
+                "data": samples.tolist()
+            },
+            {
+                "name": "reference_wav_len",
+                "shape": lengths.shape,
+                "datatype": "INT32",
+                "data": lengths.tolist(),
+            },
+            {
+                "name": "reference_text",
+                "shape": [1, 1],
+                "datatype": "BYTES",
+                "data": [reference_text]
+            },
+            {
+                "name": "target_text",
+                "shape": [1, 1],
+                "datatype": "BYTES",
+                "data": [target_text]
+            }
+        ]
+    }
+
+    return data
+
+
+if __name__ == "__main__":
+    args = get_args()
+    server_url = args.server_url
+    if not server_url.startswith(("http://", "https://")):
+        server_url = f"http://{server_url}"
+
+    url = f"{server_url}/v2/models/{args.model_name}/infer"
+    waveform, sr = sf.read(args.reference_audio)
+    assert sr == 16000, "sample rate hardcoded in server"
+
+    samples = np.array(waveform, dtype=np.float32)
+    data = prepare_request(samples, args.reference_text, args.target_text)
+
+    rsp = requests.post(
+        url,
+        headers={"Content-Type": "application/json"},
+        json=data,
+        verify=False,
+        params={"request_id": '0'}
+    )
+    result = rsp.json()
+    audio = result["outputs"][0]["data"]
+    audio = np.array(audio, dtype=np.float32)
+    if args.model_name == "spark_tts":
+        sample_rate = 16000
+    else:
+        sample_rate = 24000
+    sf.write(args.output_audio, audio, sample_rate, "PCM_16")

runtime/triton_trtllm/docker-compose.dit.yml

@@ -0,0 +1,20 @@
+services:
+  tts:
+    image: soar97/triton-cosyvoice:25.06
+    shm_size: '1gb'
+    ports:
+      - "8000:8000"
+      - "8001:8001"
+      - "8002:8002"
+    environment:
+      - PYTHONIOENCODING=utf-8
+      - MODEL_ID=${MODEL_ID}
+    deploy:
+      resources:
+        reservations:
+          devices:
+            - driver: nvidia
+              device_ids: ['0']
+              capabilities: [gpu]
+    command: >
+      /bin/bash -c "pip install modelscope && cd /workspace && git clone https://github.com/yuekaizhang/Step-Audio2.git -b trt && git clone https://github.com/yuekaizhang/CosyVoice.git -b streaming && cd CosyVoice && git submodule update --init --recursive && cd runtime/triton_trtllm && bash run_stepaudio2_dit_token2wav.sh 0 3"

runtime/triton_trtllm/docker-compose.yml

@@ -0,0 +1,20 @@
+services:
+  tts:
+    image: soar97/triton-cosyvoice:25.06
+    shm_size: '1gb'
+    ports:
+      - "8000:8000"
+      - "8001:8001"
+      - "8002:8002"
+    environment:
+      - PYTHONIOENCODING=utf-8
+      - MODEL_ID=${MODEL_ID}
+    deploy:
+      resources:
+        reservations:
+          devices:
+            - driver: nvidia
+              device_ids: ['0']
+              capabilities: [gpu]
+    command: >
+      /bin/bash -c "pip install modelscope && cd /workspace && git clone https://github.com/FunAudioLLM/CosyVoice.git && cd CosyVoice && git submodule update --init --recursive && cd runtime/triton_trtllm && bash run.sh 0 3"

runtime/triton_trtllm/model_repo/audio_tokenizer/1/model.py

@@ -0,0 +1,97 @@
+# Copyright 2025, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions
+# are met:
+#  * Redistributions of source code must retain the above copyright
+#    notice, this list of conditions and the following disclaimer.
+#  * Redistributions in binary form must reproduce the above copyright
+#    notice, this list of conditions and the following disclaimer in the
+#    documentation and/or other materials provided with the distribution.
+#  * Neither the name of NVIDIA CORPORATION nor the names of its
+#    contributors may be used to endorse or promote products derived
+#    from this software without specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS ``AS IS'' AND ANY
+# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+# PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+# OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+# (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+import json
+import torch
+from torch.utils.dlpack import to_dlpack
+
+import triton_python_backend_utils as pb_utils
+
+import os
+import numpy as np
+import s3tokenizer
+torch.set_num_threads(1)
+ORIGINAL_VOCAB_SIZE = 151663
+
+
+class TritonPythonModel:
+    """Triton Python model for audio tokenization.
+
+    This model takes reference audio input and extracts semantic tokens
+    using s3tokenizer.
+    """
+
+    def initialize(self, args):
+        """Initialize the model.
+
+        Args:
+            args: Dictionary containing model configuration
+        """
+        # Parse model parameters
+        parameters = json.loads(args['model_config'])['parameters']
+        model_params = {k: v["string_value"] for k, v in parameters.items()}
+
+        self.device = torch.device("cuda")
+        model_path = os.path.join(model_params["model_dir"], "speech_tokenizer_v2.onnx")
+        self.audio_tokenizer = s3tokenizer.load_model(model_path).to(self.device)
+
+    def execute(self, requests):
+        """Execute inference on the batched requests.
+
+        Args:
+            requests: List of inference requests
+
+        Returns:
+            List of inference responses containing tokenized outputs
+        """
+        mels = []
+
+        # Process each request in batch
+        for request in requests:
+            # Extract input tensors
+            wav_array = pb_utils.get_input_tensor_by_name(
+                request, "reference_wav").as_numpy()
+            wav_len = pb_utils.get_input_tensor_by_name(
+                request, "reference_wav_len").as_numpy().item()
+
+            wav_array = torch.from_numpy(wav_array).to(self.device)
+            # Prepare inputs
+            wav = wav_array[:, :wav_len].squeeze(0)
+            mels.append(s3tokenizer.log_mel_spectrogram(wav))
+
+        mels, mels_lens = s3tokenizer.padding(mels)
+        codes, codes_lens = self.audio_tokenizer.quantize(mels.to(self.device), mels_lens.to(self.device))
+        codes = codes.clone() + ORIGINAL_VOCAB_SIZE
+
+        responses = []
+        for i in range(len(requests)):
+            prompt_speech_tokens = codes[i, :codes_lens[i].item()]
+            prompt_speech_tokens_tensor = pb_utils.Tensor.from_dlpack(
+                "prompt_speech_tokens", to_dlpack(prompt_speech_tokens))
+            inference_response = pb_utils.InferenceResponse(
+                output_tensors=[prompt_speech_tokens_tensor])
+            responses.append(inference_response)
+
+        return responses

runtime/triton_trtllm/model_repo/audio_tokenizer/config.pbtxt

@@ -0,0 +1,53 @@
+# Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+#
+# 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.
+
+name: "audio_tokenizer"
+backend: "python"
+max_batch_size: ${triton_max_batch_size}
+dynamic_batching {
+    max_queue_delay_microseconds: ${max_queue_delay_microseconds}
+}
+parameters [
+  {
+   key: "model_dir",
+   value: {string_value:"${model_dir}"}
+  }
+]
+
+input [
+  {
+    name: "reference_wav"
+    data_type: TYPE_FP32
+    dims: [-1]
+  },
+  {
+    name: "reference_wav_len"
+    data_type: TYPE_INT32
+    dims: [1]
+  }
+]
+output [
+  {
+    name: "prompt_speech_tokens"
+    data_type: TYPE_INT32
+    dims: [-1]
+  }
+]
+
+instance_group [
+  {
+    count: 1
+    kind: KIND_CPU
+  }
+]

runtime/triton_trtllm/model_repo/cosyvoice2/1/model.py

@@ -0,0 +1,454 @@
+# Copyright 2025, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions
+# are met:
+#  * Redistributions of source code must retain the above copyright
+#    notice, this list of conditions and the following disclaimer.
+#  * Redistributions in binary form must reproduce the above copyright
+#    notice, this list of conditions and the following disclaimer in the
+#    documentation and/or other materials provided with the distribution.
+#  * Neither the name of NVIDIA CORPORATION nor the names of its
+#    contributors may be used to endorse or promote products derived
+#    from this software without specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS ``AS IS'' AND ANY
+# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+# PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+# OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+# (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+import json
+import os
+import threading
+import time
+from uuid import uuid4
+
+import numpy as np
+import torch
+from torch.utils.dlpack import from_dlpack, to_dlpack
+import triton_python_backend_utils as pb_utils
+from transformers import AutoTokenizer
+
+import torchaudio
+
+
+from matcha.utils.audio import mel_spectrogram
+
+ORIGINAL_VOCAB_SIZE = 151663
+torch.set_num_threads(1)
+
+
+class TritonPythonModel:
+    """Triton Python model for Spark TTS.
+
+    This model orchestrates the end-to-end TTS pipeline by coordinating
+    between audio tokenizer, LLM, and vocoder components.
+    """
+
+    def initialize(self, args):
+        """Initialize the model.
+
+        Args:
+            args: Dictionary containing model configuration
+        """
+        self.logger = pb_utils.Logger
+        # Parse model parameters
+        self.model_config = json.loads(args['model_config'])
+        parameters = self.model_config['parameters']
+        model_params = {k: v["string_value"] for k, v in parameters.items()}
+        self.logger.log_info(f"model_params:{model_params}")
+        self.dynamic_chunk_strategy = model_params.get("dynamic_chunk_strategy", "exponential")  # "exponential" or "time_based"
+        self.logger.log_info(f"Using dynamic chunk strategy: {self.dynamic_chunk_strategy}")
+
+        # Initialize tokenizer
+        llm_tokenizer_dir = model_params["llm_tokenizer_dir"]
+        self.tokenizer = AutoTokenizer.from_pretrained(llm_tokenizer_dir)
+        self.prompt_template = "<|sos|>{input_text}<|task_id|>"
+        self.eos_token_id = self.tokenizer.convert_tokens_to_ids("<|eos1|>")
+
+        self.device = torch.device("cuda")
+        self.decoupled = pb_utils.using_decoupled_model_transaction_policy(self.model_config)
+
+        self.token_frame_rate = 25
+        self.flow_pre_lookahead_len = 3
+        self.token_hop_len = 15
+
+        spk_info_path = os.path.join(model_params["model_dir"], "spk2info.pt")
+        if not os.path.exists(spk_info_path):
+            raise ValueError(f"spk2info.pt not found in {model_params['model_dir']}")
+        spk_info = torch.load(spk_info_path, map_location="cpu", weights_only=False)
+        self.default_spk_info = spk_info["001"]
+
+    def forward_llm(self, input_ids):
+        """
+        Prepares the response from the language model based on the provided
+        inputs. Creates a `pb_utils.InferenceRequest` object with passed
+        `llm_request_inputs` to send to a decoupled TensorRTLLM model.
+        For each response from the language model:
+            - Checks for errors and raise an exception if any are found.
+            - Extracts the "output_ids" tensor from the response.
+            - Determines the finish reason based on the presence of the
+              end-of-sequence token or reaching the maximum length.
+            - Appends the generated token IDs to `output_ids`.
+            - If the finish reason is determined, decodes the output IDs to text
+              and prepares the final response.
+
+        The final response includes the generated text, finish reason,
+        completion tokens, prompt tokens, and total tokens.
+
+        Parameters
+        ----------
+        - llm_request_inputs (dict): A dictionary containing the inputs for the language model.
+
+        Returns
+        -------
+        - pb_utils.InferenceResponse: The response object containing the generated text and additional metadata.
+        """
+        # convert input_ids to numpy, with shape [1, sequence_length]
+        input_ids = input_ids.cpu().numpy()
+        max_tokens = 750
+        input_dict = {
+            "request_output_len": np.array([[max_tokens]], dtype=np.int32),
+            "end_id": np.array([[self.eos_token_id]], dtype=np.int32),
+            "pad_id": np.array([[self.eos_token_id]], dtype=np.int32),
+            "streaming": np.array([[self.decoupled]], dtype=np.bool_),
+            "runtime_top_p": np.array([[0.95]], dtype=np.float32),
+            "runtime_top_k": np.array([[50]], dtype=np.int32),
+            "temperature": np.array([[0.8]], dtype=np.float32),
+            "repetition_penalty": np.array([[1.1]], dtype=np.float32),
+            "random_seed": np.array([[42]], dtype=np.uint64),
+            "input_ids": input_ids,
+            "input_lengths": np.array([[input_ids.shape[1]]], dtype=np.int32),
+        }
+
+        # Convert inputs to Triton tensors
+        input_tensor_list = [
+            pb_utils.Tensor(k, v) for k, v in input_dict.items()
+        ]
+
+        # Create and execute inference request
+        llm_request = pb_utils.InferenceRequest(
+            model_name="tensorrt_llm",
+            requested_output_names=["output_ids", "sequence_length"],
+            inputs=input_tensor_list,
+        )
+
+        llm_responses = llm_request.exec(decoupled=self.decoupled)
+        if self.decoupled:
+            for llm_response in llm_responses:
+                if llm_response.has_error():
+                    raise pb_utils.TritonModelException(llm_response.error().message())
+
+                # Extract and process output
+                output_ids = pb_utils.get_output_tensor_by_name(
+                    llm_response, "output_ids").as_numpy()
+                seq_lens = pb_utils.get_output_tensor_by_name(
+                    llm_response, "sequence_length").as_numpy()
+
+                # Get actual output IDs up to the sequence length
+                actual_output_ids = output_ids[0][0][:seq_lens[0][0]]
+
+                yield actual_output_ids
+        else:
+            llm_response = llm_responses
+            if llm_response.has_error():
+                raise pb_utils.TritonModelException(llm_response.error().message())
+
+            # Extract and process output
+            output_ids = pb_utils.get_output_tensor_by_name(
+                llm_response, "output_ids").as_numpy()
+            seq_lens = pb_utils.get_output_tensor_by_name(
+                llm_response, "sequence_length").as_numpy()
+
+            # Get actual output IDs up to the sequence length
+            actual_output_ids = output_ids[0][0][:seq_lens[0][0]]
+
+            yield actual_output_ids
+
+    def forward_audio_tokenizer(self, wav, wav_len):
+        """Forward pass through the audio tokenizer component.
+
+        Args:
+            wav: Input waveform tensor
+            wav_len: Waveform length tensor
+
+        Returns:
+            Tuple of global and semantic tokens
+        """
+        inference_request = pb_utils.InferenceRequest(
+            model_name='audio_tokenizer',
+            requested_output_names=['prompt_speech_tokens'],
+            inputs=[wav, wav_len]
+        )
+
+        inference_response = inference_request.exec()
+        if inference_response.has_error():
+            raise pb_utils.TritonModelException(inference_response.error().message())
+
+        # Extract and convert output tensors
+        prompt_speech_tokens = pb_utils.get_output_tensor_by_name(inference_response, 'prompt_speech_tokens')
+        prompt_speech_tokens = torch.utils.dlpack.from_dlpack(prompt_speech_tokens.to_dlpack()).cpu()
+
+        return prompt_speech_tokens
+
+    def forward_speaker_embedding(self, wav):
+        """Forward pass through the speaker embedding component.
+
+        Args:
+            wav: Input waveform tensor
+
+        Returns:
+            Prompt speaker embedding tensor
+        """
+        inference_request = pb_utils.InferenceRequest(
+            model_name='speaker_embedding',
+            requested_output_names=['prompt_spk_embedding'],
+            inputs=[pb_utils.Tensor.from_dlpack("reference_wav", to_dlpack(wav))]
+        )
+
+        inference_response = inference_request.exec()
+        if inference_response.has_error():
+            raise pb_utils.TritonModelException(inference_response.error().message())
+
+        # Extract and convert output tensors
+        prompt_spk_embedding = pb_utils.get_output_tensor_by_name(inference_response, 'prompt_spk_embedding')
+        prompt_spk_embedding = torch.utils.dlpack.from_dlpack(prompt_spk_embedding.to_dlpack())
+
+        return prompt_spk_embedding
+
+    def forward_token2wav(
+            self,
+            target_speech_tokens: torch.Tensor,
+            request_id: str,
+            prompt_speech_tokens: torch.Tensor = None,
+            prompt_speech_feat: torch.Tensor = None,
+            prompt_spk_embedding: torch.Tensor = None,
+            token_offset: int = None,
+            finalize: bool = None) -> torch.Tensor:
+        """Forward pass through the vocoder component.
+
+        Args:
+            prompt_speech_tokens: Prompt speech tokens tensor
+            prompt_speech_feat: Prompt speech feat tensor
+            prompt_spk_embedding: Prompt spk embedding tensor
+            target_speech_tokens: Target speech tokens tensor
+
+        Returns:
+            Generated waveform tensor
+        """
+        target_speech_tokens_tensor = pb_utils.Tensor.from_dlpack("target_speech_tokens", to_dlpack(target_speech_tokens))
+
+        inputs_tensor = [target_speech_tokens_tensor]
+
+        if token_offset is not None:
+            assert finalize is not None
+            token_offset_tensor = pb_utils.Tensor("token_offset", np.array([[token_offset]], dtype=np.int32))
+            finalize_tensor = pb_utils.Tensor("finalize", np.array([[finalize]], dtype=np.bool_))
+            inputs_tensor.append(token_offset_tensor)
+            inputs_tensor.append(finalize_tensor)
+
+        if prompt_spk_embedding is not None:
+            assert prompt_speech_feat is not None
+            prompt_speech_tokens_tensor = pb_utils.Tensor.from_dlpack("prompt_speech_tokens", to_dlpack(prompt_speech_tokens))
+            prompt_speech_feat_tensor = pb_utils.Tensor.from_dlpack("prompt_speech_feat", to_dlpack(prompt_speech_feat))
+            prompt_spk_embedding_tensor = pb_utils.Tensor.from_dlpack("prompt_spk_embedding", to_dlpack(prompt_spk_embedding))
+            inputs_tensor.extend([prompt_speech_tokens_tensor, prompt_speech_feat_tensor, prompt_spk_embedding_tensor])
+
+        # Create and execute inference request
+        inference_request = pb_utils.InferenceRequest(
+            model_name='token2wav',
+            requested_output_names=['waveform'],
+            inputs=inputs_tensor,
+            request_id=request_id,
+        )
+
+        inference_response = inference_request.exec()
+        if inference_response.has_error():
+            raise pb_utils.TritonModelException(inference_response.error().message())
+
+        # Extract and convert output waveform
+        waveform = pb_utils.get_output_tensor_by_name(inference_response, 'waveform')
+        waveform = torch.utils.dlpack.from_dlpack(waveform.to_dlpack()).cpu()
+
+        return waveform
+
+    def parse_input(self, text, prompt_text, prompt_speech_tokens):
+        total_text = f"{prompt_text}{text}"
+        prompt = self.prompt_template.format(input_text=total_text)
+        input_ids = self.tokenizer.encode(prompt)
+        input_ids = torch.tensor([input_ids], dtype=torch.int32)
+        input_ids = torch.cat([input_ids, prompt_speech_tokens], dim=1)
+        return input_ids
+
+    def _extract_speech_feat(self, speech):
+        speech_feat = mel_spectrogram(
+            speech,
+            n_fft=1920,
+            num_mels=80,
+            sampling_rate=24000,
+            hop_size=480,
+            win_size=1920,
+            fmin=0,
+            fmax=8000).squeeze(
+            dim=0).transpose(
+            0,
+            1).to(
+                self.device)
+        speech_feat = speech_feat.unsqueeze(dim=0)
+        return speech_feat
+
+    def _llm_gen_thread(self, generated_ids_iter, semantic_token_ids_arr, llm_is_done_flag):
+        for generated_ids in generated_ids_iter:
+            generated_ids = generated_ids.tolist()
+            if len(generated_ids) == 0:
+                break
+            semantic_token_ids_arr.extend(generated_ids)
+        llm_is_done_flag[0] = True
+
+    def execute(self, requests):
+        """Execute inference on the batched requests.
+
+        Args:
+            requests: List of inference requests
+
+        Returns:
+            List of inference responses containing generated audio
+        """
+        responses = []
+
+        for request in requests:
+            request_id = request.request_id()
+            # Extract input tensors
+            wav = pb_utils.get_input_tensor_by_name(request, "reference_wav")
+
+            # Process reference audio through audio tokenizer
+            if wav is not None:
+                wav_len = pb_utils.get_input_tensor_by_name(request, "reference_wav_len")
+                prompt_speech_tokens = self.forward_audio_tokenizer(wav, wav_len)
+                prompt_speech_tokens = prompt_speech_tokens.unsqueeze(0)
+
+                wav_tensor = wav.as_numpy()
+                wav_tensor = torch.from_numpy(wav_tensor)[:, :wav_len.as_numpy()[0][0]]
+                prompt_speech_resample = torchaudio.transforms.Resample(orig_freq=16000, new_freq=24000)(wav_tensor)
+                speech_feat = self._extract_speech_feat(prompt_speech_resample)
+                token_len = min(int(speech_feat.shape[1] / 2), prompt_speech_tokens.shape[-1])
+                prompt_speech_feat = speech_feat[:, :2 * token_len].contiguous().half()
+                prompt_speech_tokens = prompt_speech_tokens[:, :token_len].contiguous()
+
+                reference_text = pb_utils.get_input_tensor_by_name(request, "reference_text").as_numpy()
+                reference_text = reference_text[0][0].decode('utf-8')
+                prompt_spk_embedding = self.forward_speaker_embedding(wav_tensor)
+            else:
+                # using pre-cached reference text
+                reference_text = self.default_spk_info["prompt_text"]
+                prompt_speech_tokens = self.default_spk_info["speech_token"] + ORIGINAL_VOCAB_SIZE
+                prompt_speech_feat = None
+                prompt_spk_embedding = None
+
+            target_text = pb_utils.get_input_tensor_by_name(request, "target_text").as_numpy()
+            target_text = target_text[0][0].decode('utf-8')
+
+            # Prepare prompt for LLM
+            input_ids = self.parse_input(
+                text=target_text,
+                prompt_text=reference_text,
+                prompt_speech_tokens=prompt_speech_tokens,
+            )
+
+            # Generate semantic tokens with LLM
+            generated_ids_iter = self.forward_llm(input_ids)
+
+            token2wav_request_id = request_id or str(uuid4())
+            if self.decoupled:
+                response_sender = request.get_response_sender()
+
+                semantic_token_ids_arr = []
+                llm_is_done_flag = [False]
+
+                llm_thread = threading.Thread(
+                    target=self._llm_gen_thread,
+                    args=(generated_ids_iter, semantic_token_ids_arr, llm_is_done_flag)
+                )
+
+                llm_thread.start()
+
+                token_offset, chunk_index = 0, 0
+                start_time = time.time()
+                this_token_hop_len = self.token_hop_len
+
+                while True:
+                    pending_num = len(semantic_token_ids_arr) - token_offset
+
+                    if llm_is_done_flag[0]:
+                        break
+
+                    if pending_num >= this_token_hop_len + self.flow_pre_lookahead_len:
+                        this_tts_speech_token = semantic_token_ids_arr[:token_offset + this_token_hop_len + self.flow_pre_lookahead_len]
+                        this_tts_speech_token = torch.tensor(this_tts_speech_token).unsqueeze(dim=0).to(torch.int32).to(self.device)
+
+                        sub_tts_speech = self.forward_token2wav(
+                            this_tts_speech_token, token2wav_request_id, prompt_speech_tokens,
+                            prompt_speech_feat, prompt_spk_embedding, token_offset, False
+                        )
+
+                        audio_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(sub_tts_speech))
+                        inference_response = pb_utils.InferenceResponse(output_tensors=[audio_tensor])
+                        response_sender.send(inference_response)
+
+                        token_offset += this_token_hop_len
+                        self.logger.log_info(f"chunk_index: {chunk_index}, current_token_hop_len: {this_token_hop_len}")
+
+                        if self.dynamic_chunk_strategy == "exponential":
+                            this_token_hop_len = self.token_frame_rate * (2 ** chunk_index)
+                        elif self.dynamic_chunk_strategy == "time_based":
+                            # see https://github.com/qi-hua/async_cosyvoice/blob/main/model.py#L306
+                            cost_time = time.time() - start_time
+                            duration = token_offset / self.token_frame_rate
+                            if chunk_index > 0 and cost_time > 0:
+                                avg_chunk_processing_time = cost_time / (chunk_index + 1)
+                                if avg_chunk_processing_time > 0:
+                                    multiples = (duration - cost_time) / avg_chunk_processing_time
+                                    self.logger.log_info(f"multiples: {multiples}")
+                                    next_pending_num = len(semantic_token_ids_arr) - token_offset
+                                    if multiples > 4:
+                                        this_token_hop_len = (next_pending_num // self.token_hop_len + 1) * self.token_hop_len
+                                    elif multiples > 2:
+                                        this_token_hop_len = (next_pending_num // self.token_hop_len) * self.token_hop_len
+                                    else:
+                                        this_token_hop_len = self.token_hop_len
+                                    this_token_hop_len = max(self.token_hop_len, this_token_hop_len)
+                        chunk_index += 1
+                    else:
+                        time.sleep(0.02)
+
+                this_tts_speech_token = torch.tensor(semantic_token_ids_arr).unsqueeze(dim=0).to(torch.int32).to(self.device)
+                sub_tts_speech = self.forward_token2wav(this_tts_speech_token, token2wav_request_id, prompt_speech_tokens, prompt_speech_feat, prompt_spk_embedding, token_offset, True)
+                audio_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(sub_tts_speech))
+                inference_response = pb_utils.InferenceResponse(output_tensors=[audio_tensor])
+                response_sender.send(inference_response)
+
+                llm_thread.join()
+                response_sender.send(flags=pb_utils.TRITONSERVER_RESPONSE_COMPLETE_FINAL)
+                self.logger.log_info("send tritonserver_response_complete_final to end")
+            else:
+                generated_ids = next(generated_ids_iter)
+                generated_ids = torch.tensor(generated_ids).unsqueeze(0).to(self.device)
+                if generated_ids is None or len(generated_ids) == 0:
+                    raise pb_utils.TritonModelException("Generated IDs is None or empty")
+
+                audio = self.forward_token2wav(generated_ids, token2wav_request_id, prompt_speech_tokens, prompt_speech_feat, prompt_spk_embedding)
+
+                # Prepare response
+                audio_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(audio))
+                inference_response = pb_utils.InferenceResponse(output_tensors=[audio_tensor])
+                responses.append(inference_response)
+
+        if not self.decoupled:
+            return responses

runtime/triton_trtllm/model_repo/cosyvoice2/config.pbtxt

@@ -0,0 +1,73 @@
+# Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+#
+# 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.
+
+name: "cosyvoice2"
+backend: "python"
+max_batch_size: ${triton_max_batch_size}
+dynamic_batching {
+    max_queue_delay_microseconds: ${max_queue_delay_microseconds}
+}
+model_transaction_policy {
+  decoupled: ${decoupled_mode}
+}
+parameters [
+  {
+   key: "llm_tokenizer_dir",
+   value: {string_value:"${llm_tokenizer_dir}"}
+  },
+  {
+   key: "model_dir",
+   value: {string_value:"${model_dir}"}
+  }
+]
+
+input [
+  {
+    name: "reference_wav"
+    data_type: TYPE_FP32
+    dims: [-1]
+    optional: true
+  },
+  {
+    name: "reference_wav_len"
+    data_type: TYPE_INT32
+    dims: [1]
+    optional: true
+  },
+  {
+    name: "reference_text"
+    data_type: TYPE_STRING
+    dims: [1]
+    optional: true
+  },
+  {
+    name: "target_text"
+    data_type: TYPE_STRING
+    dims: [1]
+  }
+]
+output [
+  {
+    name: "waveform"
+    data_type: TYPE_FP32
+    dims: [ -1 ]
+  }
+]
+
+instance_group [
+  {
+    count: ${bls_instance_num}
+    kind: KIND_CPU
+  }
+]

runtime/triton_trtllm/model_repo/cosyvoice2_dit/1/model.py

@@ -0,0 +1,394 @@
+# Copyright 2025, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions
+# are met:
+#  * Redistributions of source code must retain the above copyright
+#    notice, this list of conditions and the following disclaimer.
+#  * Redistributions in binary form must reproduce the above copyright
+#    notice, this list of conditions and the following disclaimer in the
+#    documentation and/or other materials provided with the distribution.
+#  * Neither the name of NVIDIA CORPORATION nor the names of its
+#    contributors may be used to endorse or promote products derived
+#    from this software without specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS ``AS IS'' AND ANY
+# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+# PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+# OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+# (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+import json
+import math
+import os
+import re
+import time
+from typing import Dict, List, Tuple, Optional, Union
+import asyncio
+import httpx
+
+import numpy as np
+import torch
+from torch.utils.dlpack import from_dlpack, to_dlpack
+import triton_python_backend_utils as pb_utils
+from transformers import AutoTokenizer
+
+import torchaudio
+
+
+from matcha.utils.audio import mel_spectrogram
+
+
+ORIGINAL_VOCAB_SIZE = 151663
+torch.set_num_threads(1)
+
+
+def parse_speech_token_string(response_text: str) -> List[int]:
+    """
+    Parses a string of speech tokens (e.g., "<|s_123|><|s_456|>") into a list of integer IDs.
+    """
+    speech_tokens = response_text.strip().split('><')
+    if len(speech_tokens) > 1:
+        # Add back the missing '<' and '>' for proper parsing
+        speech_tokens = ['<' + t if not t.startswith('<') else t for t in speech_tokens]
+        speech_tokens = [t + '>' if not t.endswith('>') else t for t in speech_tokens]
+
+    speech_ids = []
+    for token_str in speech_tokens:
+        match = re.match(r'<\|s_(\d+)\|>', token_str)
+        if match:
+            speech_ids.append(int(match.group(1)))
+    return speech_ids
+
+
+class TritonPythonModel:
+    """Triton Python model for Spark TTS.
+
+    This model orchestrates the end-to-end TTS pipeline by coordinating
+    between audio tokenizer, LLM, and vocoder components.
+    """
+
+    def initialize(self, args):
+        """Initialize the model.
+
+        Args:
+            args: Dictionary containing model configuration
+        """
+        self.logger = pb_utils.Logger
+        # Parse model parameters
+        self.model_config = json.loads(args['model_config'])
+        parameters = self.model_config['parameters']
+        model_params = {k: v["string_value"] for k, v in parameters.items()}
+        self.dynamic_chunk_strategy = model_params.get("dynamic_chunk_strategy", "exponential")  # "exponential" or "time_based"
+        self.logger.log_info(f"Using dynamic chunk strategy: {self.dynamic_chunk_strategy}")
+
+        # Initialize tokenizer
+        llm_tokenizer_dir = model_params["llm_tokenizer_dir"]
+        self.tokenizer = AutoTokenizer.from_pretrained(llm_tokenizer_dir)
+        self.prompt_template = "<|sos|>{input_text}<|task_id|>"
+        self.eos_token_id = self.tokenizer.convert_tokens_to_ids("<|eos1|>")
+
+        self.device = torch.device("cuda")
+        self.decoupled = pb_utils.using_decoupled_model_transaction_policy(self.model_config)
+
+        self.token_frame_rate = 25
+        self.flow_pre_lookahead_len = 3
+        self.token_hop_len = 15
+
+        self.http_client = httpx.AsyncClient()
+        self.api_base = "http://localhost:8000/v1/chat/completions"
+        self.speaker_cache = {}
+
+    def _convert_speech_tokens_to_str(self, speech_tokens: Union[torch.Tensor, List]) -> str:
+        """Converts a tensor or list of speech token IDs to a string representation."""
+        if isinstance(speech_tokens, torch.Tensor):
+            # Ensure tensor is on CPU and flattened
+            speech_tokens = speech_tokens.cpu().numpy().flatten().tolist()
+
+        speech_id_str = ""
+        for token_id in speech_tokens:
+            # Convert token ID back to the speech number N
+            token_num = token_id - ORIGINAL_VOCAB_SIZE
+            speech_id_str += f"<|s_{token_num}|>"
+        return speech_id_str
+
+    async def forward_llm_async(self, target_text: str, reference_text: str, prompt_speech_tokens: Union[torch.Tensor, List]):
+        """
+        Asynchronously sends a request to the TRTLLM-serve endpoint and processes the streaming response.
+        """
+        full_text = f"{reference_text}{target_text}"
+        prompt_speech_tokens_str = self._convert_speech_tokens_to_str(prompt_speech_tokens)
+
+        chat = [
+            {"role": "user", "content": full_text},
+            {"role": "assistant", "content": prompt_speech_tokens_str}
+        ]
+
+        payload = {
+            "model": "trt_engines_bfloat16",
+            "messages": chat,
+            "max_tokens": 750,
+            "temperature": 0.8,
+            "top_p": 0.95,
+            "top_k": 50,
+            "repetition_penalty": 1.1,
+            "stop": ["<|eos1|>", "<|eos|>"],
+            "stream": True,
+        }
+
+        buffer = ""
+        async with self.http_client.stream("POST", self.api_base, json=payload, timeout=None) as response:
+            response.raise_for_status()
+            async for line in response.aiter_lines():
+                if line.startswith("data: "):
+                    line_data = line[len("data: "):].strip()
+                    if line_data == "[DONE]":
+                        break
+                    try:
+                        json_data = json.loads(line_data)
+                        content = json_data.get("choices", [{}])[0].get("delta", {}).get("content")
+                        if content:
+                            buffer += content
+                            while True:
+                                match = re.search(r"<\|s_(\d+)\|>", buffer)
+                                if not match:
+                                    break
+
+                                token_num = int(match.group(1))
+                                final_id = token_num + ORIGINAL_VOCAB_SIZE
+                                yield final_id
+                                buffer = buffer[match.end():]
+                    except json.JSONDecodeError:
+                        self.logger.log_info(f"Skipping non-JSON line: {line_data}")
+                        continue
+
+        # Process any remaining complete tokens in the buffer after the stream ends
+        while True:
+            match = re.search(r"<\|s_(\d+)\|>", buffer)
+            if not match:
+                break
+            token_num = int(match.group(1))
+            final_id = token_num + ORIGINAL_VOCAB_SIZE
+            yield final_id
+            buffer = buffer[match.end():]
+
+    def forward_audio_tokenizer(self, wav, wav_len):
+        """Forward pass through the audio tokenizer component.
+
+        Args:
+            wav: Input waveform tensor
+            wav_len: Waveform length tensor
+
+        Returns:
+            Tuple of global and semantic tokens
+        """
+        inference_request = pb_utils.InferenceRequest(
+            model_name='audio_tokenizer',
+            requested_output_names=['prompt_speech_tokens'],
+            inputs=[wav, wav_len]
+        )
+
+        inference_response = inference_request.exec()
+        if inference_response.has_error():
+            raise pb_utils.TritonModelException(inference_response.error().message())
+
+        # Extract and convert output tensors
+        prompt_speech_tokens = pb_utils.get_output_tensor_by_name(inference_response, 'prompt_speech_tokens')
+        prompt_speech_tokens = torch.utils.dlpack.from_dlpack(prompt_speech_tokens.to_dlpack()).cpu()
+
+        return prompt_speech_tokens
+
+    def forward_speaker_embedding(self, wav):
+        """Forward pass through the speaker embedding component.
+
+        Args:
+            wav: Input waveform tensor
+
+        Returns:
+            Prompt speaker embedding tensor
+        """
+        inference_request = pb_utils.InferenceRequest(
+            model_name='speaker_embedding',
+            requested_output_names=['prompt_spk_embedding'],
+            inputs=[pb_utils.Tensor.from_dlpack("reference_wav", to_dlpack(wav))]
+        )
+
+        inference_response = inference_request.exec()
+        if inference_response.has_error():
+            raise pb_utils.TritonModelException(inference_response.error().message())
+
+        # Extract and convert output tensors
+        prompt_spk_embedding = pb_utils.get_output_tensor_by_name(inference_response, 'prompt_spk_embedding')
+        prompt_spk_embedding = torch.utils.dlpack.from_dlpack(prompt_spk_embedding.to_dlpack())
+
+        return prompt_spk_embedding
+
+    async def forward_token2wav(
+            self,
+            index: int,
+            target_speech_tokens: torch.Tensor,
+            request_id: str,
+            reference_wav: object,
+            reference_wav_len: object,
+            finalize: bool = None) -> torch.Tensor:
+        """Forward pass through the vocoder component.
+
+        Args:
+            index: Index of the request
+            target_speech_tokens: Target speech tokens tensor
+            request_id: Request ID
+            reference_wav: Reference waveform tensor
+            reference_wav_len: Reference waveform length tensor
+            finalize: Whether to finalize the request
+
+        Returns:
+            Generated waveform tensor
+        """
+        target_speech_tokens_tensor = pb_utils.Tensor.from_dlpack("target_speech_tokens", to_dlpack(target_speech_tokens))
+        finalize_tensor = pb_utils.Tensor("finalize", np.array([[finalize]], dtype=np.bool_))
+        inputs_tensor = [target_speech_tokens_tensor, reference_wav, reference_wav_len, finalize_tensor]
+
+        # Create and execute inference request
+        inference_request = pb_utils.InferenceRequest(
+            model_name='token2wav_dit',
+            requested_output_names=[
+                "waveform",
+            ],
+            inputs=inputs_tensor,
+            request_id=request_id,
+            parameters={"priority": index + 1},
+        )
+
+        inference_response = await inference_request.async_exec()
+        if inference_response.has_error():
+            raise pb_utils.TritonModelException(inference_response.error().message())
+
+        # Extract and convert output waveform
+        waveform = pb_utils.get_output_tensor_by_name(inference_response, 'waveform')
+        waveform = torch.utils.dlpack.from_dlpack(waveform.to_dlpack()).cpu()
+
+        return waveform
+
+    def _extract_speech_feat(self, speech):
+        speech_feat = mel_spectrogram(
+            speech,
+            n_fft=1920,
+            num_mels=80,
+            sampling_rate=24000,
+            hop_size=480,
+            win_size=1920,
+            fmin=0,
+            fmax=8000).squeeze(
+            dim=0).transpose(
+            0,
+            1).to(
+                self.device)
+        speech_feat = speech_feat.unsqueeze(dim=0)
+        return speech_feat
+
+    async def _process_request(self, request):
+        request_id = request.request_id()
+
+        reference_text = pb_utils.get_input_tensor_by_name(request, "reference_text").as_numpy()
+        reference_text = reference_text[0][0].decode('utf-8')
+
+        wav = pb_utils.get_input_tensor_by_name(request, "reference_wav")
+        wav_len = pb_utils.get_input_tensor_by_name(request, "reference_wav_len")
+
+        if reference_text not in self.speaker_cache:
+            self.speaker_cache[reference_text] = self.forward_audio_tokenizer(wav, wav_len).unsqueeze(0)
+        prompt_speech_tokens = self.speaker_cache[reference_text]
+
+        target_text = pb_utils.get_input_tensor_by_name(request, "target_text").as_numpy()
+        target_text = target_text[0][0].decode('utf-8')
+
+        if self.decoupled:
+            response_sender = request.get_response_sender()
+
+            semantic_token_ids_arr = []
+            token_offset, chunk_index = 0, 0
+            start_time = time.time()
+            this_token_hop_len = self.token_hop_len
+            async for generated_ids in self.forward_llm_async(
+                target_text=target_text,
+                reference_text=reference_text,
+                prompt_speech_tokens=prompt_speech_tokens,
+            ):
+                if not generated_ids:
+                    break
+                semantic_token_ids_arr.append(generated_ids)
+                while True:
+                    pending_num = len(semantic_token_ids_arr) - token_offset
+                    if pending_num >= this_token_hop_len + self.flow_pre_lookahead_len:
+                        this_tts_speech_token = semantic_token_ids_arr[token_offset:token_offset + this_token_hop_len + self.flow_pre_lookahead_len]
+                        this_tts_speech_token = torch.tensor(this_tts_speech_token).unsqueeze(dim=0).to(torch.int32).to(self.device)
+                        sub_tts_speech = await self.forward_token2wav(
+                            chunk_index,
+                            this_tts_speech_token, request_id, wav, wav_len, False
+                        )
+                        audio_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(sub_tts_speech))
+                        inference_response = pb_utils.InferenceResponse(output_tensors=[audio_tensor])
+                        response_sender.send(inference_response)
+
+                        token_offset += this_token_hop_len
+
+                        if self.dynamic_chunk_strategy == "exponential":
+                            this_token_hop_len = self.token_frame_rate * (2 ** chunk_index)
+                        elif self.dynamic_chunk_strategy == "equal":
+                            this_token_hop_len = self.token_hop_len
+                        elif self.dynamic_chunk_strategy == "time_based":
+                            # see https://github.com/qi-hua/async_cosyvoice/blob/main/model.py#L306
+                            cost_time = time.time() - start_time
+                            duration = token_offset / self.token_frame_rate
+                            if chunk_index > 0 and cost_time > 0:
+                                avg_chunk_processing_time = cost_time / (chunk_index + 1)
+                                if avg_chunk_processing_time > 0:
+                                    multiples = (duration - cost_time) / avg_chunk_processing_time
+                                    next_pending_num = len(semantic_token_ids_arr) - token_offset
+                                    if multiples > 4:
+                                        this_token_hop_len = (next_pending_num // self.token_hop_len + 1) * self.token_hop_len
+                                    elif multiples > 2:
+                                        this_token_hop_len = (next_pending_num // self.token_hop_len) * self.token_hop_len
+                                    else:
+                                        this_token_hop_len = self.token_hop_len
+                                    this_token_hop_len = max(self.token_hop_len, this_token_hop_len)
+                        chunk_index += 1
+                    else:
+                        break
+
+            this_tts_speech_token = torch.tensor(semantic_token_ids_arr[token_offset:]).unsqueeze(dim=0).to(torch.int32).to(self.device)
+            sub_tts_speech = await self.forward_token2wav(chunk_index, this_tts_speech_token, request_id, wav, wav_len, True)
+            audio_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(sub_tts_speech))
+            inference_response = pb_utils.InferenceResponse(output_tensors=[audio_tensor])
+            response_sender.send(inference_response)
+
+            response_sender.send(flags=pb_utils.TRITONSERVER_RESPONSE_COMPLETE_FINAL)
+        else:
+            raise NotImplementedError("Offline TTS mode is not supported")
+
+    async def execute(self, requests):
+        """Execute inference on the batched requests.
+
+        Args:
+            requests: List of inference requests
+
+        Returns:
+            List of inference responses containing generated audio
+        """
+        tasks = [
+            asyncio.create_task(self._process_request(request))
+            for request in requests
+        ]
+        await asyncio.gather(*tasks)
+        return None
+
+    def finalize(self):
+        self.logger.log_info("Finalizing CosyVoice DIT model")
+        if hasattr(self, "http_client"):
+            asyncio.run(self.http_client.aclose())

runtime/triton_trtllm/model_repo/cosyvoice2_dit/config.pbtxt

@@ -0,0 +1,73 @@
+# Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+#
+# 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.
+
+name: "cosyvoice2_dit"
+backend: "python"
+max_batch_size: ${triton_max_batch_size}
+dynamic_batching {
+    max_queue_delay_microseconds: ${max_queue_delay_microseconds}
+}
+model_transaction_policy {
+  decoupled: ${decoupled_mode}
+}
+parameters [
+  {
+   key: "llm_tokenizer_dir",
+   value: {string_value:"${llm_tokenizer_dir}"}
+  },
+  {
+   key: "model_dir",
+   value: {string_value:"${model_dir}"}
+  }
+]
+
+input [
+  {
+    name: "reference_wav"
+    data_type: TYPE_FP32
+    dims: [-1]
+    optional: true
+  },
+  {
+    name: "reference_wav_len"
+    data_type: TYPE_INT32
+    dims: [1]
+    optional: true
+  },
+  {
+    name: "reference_text"
+    data_type: TYPE_STRING
+    dims: [1]
+    optional: true
+  },
+  {
+    name: "target_text"
+    data_type: TYPE_STRING
+    dims: [1]
+  }
+]
+output [
+  {
+    name: "waveform"
+    data_type: TYPE_FP32
+    dims: [ -1 ]
+  }
+]
+
+instance_group [
+  {
+    count: ${bls_instance_num}
+    kind: KIND_CPU
+  }
+]

runtime/triton_trtllm/model_repo/speaker_embedding/1/model.py

@@ -0,0 +1,153 @@
+# Copyright 2025, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions
+# are met:
+#  * Redistributions of source code must retain the above copyright
+#    notice, this list of conditions and the following disclaimer.
+#  * Redistributions in binary form must reproduce the above copyright
+#    notice, this list of conditions and the following disclaimer in the
+#    documentation and/or other materials provided with the distribution.
+#  * Neither the name of NVIDIA CORPORATION nor the names of its
+#    contributors may be used to endorse or promote products derived
+#    from this software without specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS ``AS IS'' AND ANY
+# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+# PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+# OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+# (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+import json
+import torch
+from torch.utils.dlpack import to_dlpack
+
+import triton_python_backend_utils as pb_utils
+
+import os
+import numpy as np
+import torchaudio.compliance.kaldi as kaldi
+from cosyvoice.utils.file_utils import convert_onnx_to_trt
+from cosyvoice.utils.common import TrtContextWrapper
+import onnxruntime
+
+
+class TritonPythonModel:
+    """Triton Python model for audio tokenization.
+
+    This model takes reference audio input and extracts semantic tokens
+    using s3tokenizer.
+    """
+
+    def initialize(self, args):
+        """Initialize the model.
+
+        Args:
+            args: Dictionary containing model configuration
+        """
+        # Parse model parameters
+        parameters = json.loads(args['model_config'])['parameters']
+        model_params = {k: v["string_value"] for k, v in parameters.items()}
+
+        self.device = torch.device("cuda")
+
+        model_dir = model_params["model_dir"]
+        gpu = "l20"
+        enable_trt = True
+        if enable_trt:
+            self.load_spk_trt(f'{model_dir}/campplus.{gpu}.fp32.trt',
+                              f'{model_dir}/campplus.onnx',
+                              1,
+                              False)
+        else:
+            campplus_model = f'{model_dir}/campplus.onnx'
+            option = onnxruntime.SessionOptions()
+            option.graph_optimization_level = onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
+            option.intra_op_num_threads = 1
+            self.spk_model = onnxruntime.InferenceSession(campplus_model, sess_options=option, providers=["CPUExecutionProvider"])
+
+    def load_spk_trt(self, spk_model, spk_onnx_model, trt_concurrent=1, fp16=True):
+        if not os.path.exists(spk_model) or os.path.getsize(spk_model) == 0:
+            trt_kwargs = self.get_spk_trt_kwargs()
+            convert_onnx_to_trt(spk_model, trt_kwargs, spk_onnx_model, fp16)
+        import tensorrt as trt
+        with open(spk_model, 'rb') as f:
+            spk_engine = trt.Runtime(trt.Logger(trt.Logger.INFO)).deserialize_cuda_engine(f.read())
+        assert spk_engine is not None, 'failed to load trt {}'.format(spk_model)
+        self.spk_model = TrtContextWrapper(spk_engine, trt_concurrent=trt_concurrent, device=self.device)
+
+    def get_spk_trt_kwargs(self):
+        min_shape = [(1, 4, 80)]
+        opt_shape = [(1, 500, 80)]
+        max_shape = [(1, 3000, 80)]
+        input_names = ["input"]
+        return {'min_shape': min_shape, 'opt_shape': opt_shape, 'max_shape': max_shape, 'input_names': input_names}
+
+    def _extract_spk_embedding(self, speech):
+        feat = kaldi.fbank(speech,
+                           num_mel_bins=80,
+                           dither=0,
+                           sample_frequency=16000)
+        spk_feat = feat - feat.mean(dim=0, keepdim=True)
+
+        if isinstance(self.spk_model, onnxruntime.InferenceSession):
+            embedding = self.spk_model.run(
+                None, {self.spk_model.get_inputs()[0].name: spk_feat.unsqueeze(dim=0).cpu().numpy()}
+            )[0].flatten().tolist()
+            embedding = torch.tensor([embedding]).to(self.device)
+        else:
+            [spk_model, stream], trt_engine = self.spk_model.acquire_estimator()
+            # NOTE need to synchronize when switching stream
+            with torch.cuda.device(self.device):
+                torch.cuda.current_stream().synchronize()
+                spk_feat = spk_feat.unsqueeze(dim=0).to(self.device)
+                batch_size = spk_feat.size(0)
+
+                with stream:
+                    spk_model.set_input_shape('input', (batch_size, spk_feat.size(1), 80))
+                    embedding = torch.empty((batch_size, 192), device=spk_feat.device)
+
+                    data_ptrs = [spk_feat.contiguous().data_ptr(),
+                                 embedding.contiguous().data_ptr()]
+                    for i, j in enumerate(data_ptrs):
+
+                        spk_model.set_tensor_address(trt_engine.get_tensor_name(i), j)
+                    # run trt engine
+                    assert spk_model.execute_async_v3(torch.cuda.current_stream().cuda_stream) is True
+                    torch.cuda.current_stream().synchronize()
+                self.spk_model.release_estimator(spk_model, stream)
+
+        return embedding.half()
+
+    def execute(self, requests):
+        """Execute inference on the batched requests.
+
+        Args:
+            requests: List of inference requests
+
+        Returns:
+            List of inference responses containing tokenized outputs
+        """
+        responses = []
+        # Process each request in batch
+        for request in requests:
+            # Extract input tensors
+            wav_array = pb_utils.get_input_tensor_by_name(
+                request, "reference_wav").as_numpy()
+            wav_array = torch.from_numpy(wav_array).to(self.device)
+
+            embedding = self._extract_spk_embedding(wav_array)
+
+            prompt_spk_embedding_tensor = pb_utils.Tensor.from_dlpack(
+                "prompt_spk_embedding", to_dlpack(embedding))
+            inference_response = pb_utils.InferenceResponse(
+                output_tensors=[prompt_spk_embedding_tensor])
+
+            responses.append(inference_response)
+
+        return responses

runtime/triton_trtllm/model_repo/speaker_embedding/config.pbtxt

@@ -0,0 +1,48 @@
+# Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+#
+# 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.
+
+name: "speaker_embedding"
+backend: "python"
+max_batch_size: ${triton_max_batch_size}
+dynamic_batching {
+    max_queue_delay_microseconds: ${max_queue_delay_microseconds}
+}
+parameters [
+  {
+   key: "model_dir",
+   value: {string_value:"${model_dir}"}
+  }
+]
+
+input [
+  {
+    name: "reference_wav"
+    data_type: TYPE_FP32
+    dims: [-1]
+  }
+]
+output [
+  {
+    name: "prompt_spk_embedding"
+    data_type: TYPE_FP16
+    dims: [-1]
+  }
+]
+
+instance_group [
+  {
+    count: 1
+    kind: KIND_CPU
+  }
+]

runtime/triton_trtllm/model_repo/tensorrt_llm/config.pbtxt

@@ -0,0 +1,857 @@
+# Copyright 2024, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions
+# are met:
+#  * Redistributions of source code must retain the above copyright
+#    notice, this list of conditions and the following disclaimer.
+#  * Redistributions in binary form must reproduce the above copyright
+#    notice, this list of conditions and the following disclaimer in the
+#    documentation and/or other materials provided with the distribution.
+#  * Neither the name of NVIDIA CORPORATION nor the names of its
+#    contributors may be used to endorse or promote products derived
+#    from this software without specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS ``AS IS'' AND ANY
+# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+# PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+# OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+# (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+name: "tensorrt_llm"
+backend: "${triton_backend}"
+max_batch_size: ${triton_max_batch_size}
+
+model_transaction_policy {
+  decoupled: ${decoupled_mode}
+}
+
+dynamic_batching {
+    preferred_batch_size: [ ${triton_max_batch_size} ]
+    max_queue_delay_microseconds: ${max_queue_delay_microseconds}
+    default_queue_policy: { max_queue_size: ${max_queue_size} }
+}
+
+input [
+  {
+    name: "input_ids"
+    data_type: TYPE_INT32
+    dims: [ -1 ]
+    allow_ragged_batch: true
+    optional: true
+  },
+  {
+    name: "encoder_input_features"
+    data_type: ${encoder_input_features_data_type}
+    dims: [ -1, -1 ]
+    allow_ragged_batch: true
+    optional: true
+  },
+  {
+    name: "encoder_output_lengths"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "input_lengths"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+  },
+  {
+    name: "request_output_len"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+  },
+  {
+    name: "num_return_sequences"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "draft_input_ids"
+    data_type: TYPE_INT32
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "decoder_input_ids"
+    data_type: TYPE_INT32
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "decoder_input_lengths"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    optional: true
+    reshape: { shape: [ ] }
+  },
+  {
+    name: "draft_logits"
+    data_type: ${logits_datatype}
+    dims: [ -1, -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "draft_acceptance_threshold"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "end_id"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "pad_id"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "stop_words_list"
+    data_type: TYPE_INT32
+    dims: [ 2, -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "bad_words_list"
+    data_type: TYPE_INT32
+    dims: [ 2, -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "embedding_bias"
+    data_type: TYPE_FP32
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "beam_width"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "temperature"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "runtime_top_k"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "runtime_top_p"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "runtime_top_p_min"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "runtime_top_p_decay"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "runtime_top_p_reset_ids"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "len_penalty"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "early_stopping"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "repetition_penalty"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "min_length"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "beam_search_diversity_rate"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "presence_penalty"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "frequency_penalty"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "random_seed"
+    data_type: TYPE_UINT64
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "return_log_probs"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "return_context_logits"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "return_generation_logits"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "return_perf_metrics"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "exclude_input_in_output"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "stop"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "streaming"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "prompt_embedding_table"
+    data_type: TYPE_FP16
+    dims: [ -1, -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "prompt_table_extra_ids"
+    data_type: TYPE_UINT64
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "prompt_vocab_size"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  # cross_attention_mask shape `[bs, seq_len, num_images*num_tiles]`
+  {
+    name: "cross_attention_mask"
+    data_type: TYPE_BOOL
+    dims: [ -1, -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  # Mrope param when mrope is used
+  {
+    name: "mrope_rotary_cos_sin"
+    data_type: TYPE_FP32
+    dims: [ -1 ]
+    optional: true
+  },
+  {
+    name: "mrope_position_deltas"
+    data_type: TYPE_INT64
+    dims: [ 1 ]
+    optional: true
+  },
+  # the unique task ID for the given LoRA.
+  # To perform inference with a specific LoRA for the first time `lora_task_id` `lora_weights` and `lora_config` must all be given.
+  # The LoRA will be cached, so that subsequent requests for the same task only require `lora_task_id`.
+  # If the cache is full the oldest LoRA will be evicted to make space for new ones.  An error is returned if `lora_task_id` is not cached.
+  {
+    name: "lora_task_id"
+    data_type: TYPE_UINT64
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  # weights for a lora adapter shape [ num_lora_modules_layers, D x Hi + Ho x D ]
+  # where the last dimension holds the in / out adapter weights for the associated module (e.g. attn_qkv) and model layer
+  # each of the in / out tensors are first flattened and then concatenated together in the format above.
+  # D=adapter_size (R value), Hi=hidden_size_in, Ho=hidden_size_out.
+  {
+    name: "lora_weights"
+    data_type: TYPE_FP16
+    dims: [ -1, -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  # module identifier (same size a first dimension of lora_weights)
+  # See LoraModule::ModuleType for model id mapping
+  #
+  # "attn_qkv": 0     # compbined qkv adapter
+  # "attn_q": 1       # q adapter
+  # "attn_k": 2       # k adapter
+  # "attn_v": 3       # v adapter
+  # "attn_dense": 4   # adapter for the dense layer in attention
+  # "mlp_h_to_4h": 5  # for llama2 adapter for gated mlp layer after attention / RMSNorm: up projection
+  # "mlp_4h_to_h": 6  # for llama2 adapter for gated mlp layer after attention / RMSNorm: down projection
+  # "mlp_gate": 7     # for llama2 adapter for gated mlp later after attention / RMSNorm: gate
+  #
+  # last dim holds [ module_id, layer_idx, adapter_size (D aka R value) ]
+  {
+    name: "lora_config"
+    data_type: TYPE_INT32
+    dims: [ -1, 3 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "context_phase_params"
+    data_type: TYPE_UINT8
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  # skip_cross_attn_blocks shape `[bs, 1]`, only used in mllama
+  {
+    name: "skip_cross_attn_blocks"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "retention_token_range_starts"
+    data_type: TYPE_INT32
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "retention_token_range_ends"
+    data_type: TYPE_INT32
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "retention_token_range_priorities"
+    data_type: TYPE_INT32
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "retention_token_range_durations_ms"
+    data_type: TYPE_INT32
+    dims: [ -1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "retention_decode_priority"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "retention_decode_duration_ms"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "guided_decoding_guide_type"
+    data_type: TYPE_STRING
+    dims: [ 1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "guided_decoding_guide"
+    data_type: TYPE_STRING
+    dims: [ 1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "lookahead_window_size"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "lookahead_ngram_size"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    optional: true
+    allow_ragged_batch: true
+  },
+  {
+    name: "lookahead_verification_set_size"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    optional: true
+    allow_ragged_batch: true
+  }
+]
+output [
+  {
+    name: "output_ids"
+    data_type: TYPE_INT32
+    dims: [ -1, -1 ]
+  },
+  {
+    name: "sequence_length"
+    data_type: TYPE_INT32
+    dims: [ -1 ]
+  },
+  {
+    name: "cum_log_probs"
+    data_type: TYPE_FP32
+    dims: [ -1 ]
+  },
+  {
+    name: "output_log_probs"
+    data_type: TYPE_FP32
+    dims: [ -1, -1 ]
+  },
+  {
+    name: "context_logits"
+    data_type: ${logits_datatype}
+    dims: [ -1, -1 ]
+  },
+  {
+    name: "generation_logits"
+    data_type: ${logits_datatype}
+    dims: [ -1, -1, -1 ]
+  },
+  {
+    name: "batch_index"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+  },
+  {
+    name: "sequence_index"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+  },
+  {
+    name: "context_phase_params"
+    data_type: TYPE_UINT8
+    dims: [ -1 ]
+  },
+  {
+    name: "kv_cache_alloc_new_blocks"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+  },
+  {
+    name: "kv_cache_reused_blocks"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+  },
+  {
+    name: "kv_cache_alloc_total_blocks"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+  },
+  {
+    name: "arrival_time_ns"
+    data_type: TYPE_INT64
+    dims: [ 1 ]
+  },
+  {
+    name: "first_scheduled_time_ns"
+    data_type: TYPE_INT64
+    dims: [ 1 ]
+  },
+  {
+    name: "first_token_time_ns"
+    data_type: TYPE_INT64
+    dims: [ 1 ]
+  },
+  {
+    name: "last_token_time_ns"
+    data_type: TYPE_INT64
+    dims: [ 1 ]
+  },
+  {
+    name: "acceptance_rate"
+    data_type: TYPE_FP32
+    dims: [ 1 ]
+  },
+  {
+    name: "total_accepted_draft_tokens"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+  },
+  {
+    name: "total_draft_tokens"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+  }
+]
+instance_group [
+  {
+    count: 1
+    kind : KIND_CPU
+  }
+]
+parameters: {
+  key: "max_beam_width"
+  value: {
+    string_value: "${max_beam_width}"
+  }
+}
+parameters: {
+  key: "FORCE_CPU_ONLY_INPUT_TENSORS"
+  value: {
+    string_value: "no"
+  }
+}
+parameters: {
+  key: "gpt_model_type"
+  value: {
+    string_value: "${batching_strategy}"
+  }
+}
+parameters: {
+  key: "gpt_model_path"
+  value: {
+    string_value: "${engine_dir}"
+  }
+}
+parameters: {
+  key: "encoder_model_path"
+  value: {
+    string_value: "${encoder_engine_dir}"
+  }
+}
+parameters: {
+  key: "max_tokens_in_paged_kv_cache"
+  value: {
+    string_value: "${max_tokens_in_paged_kv_cache}"
+  }
+}
+parameters: {
+  key: "max_attention_window_size"
+  value: {
+    string_value: "${max_attention_window_size}"
+  }
+}
+parameters: {
+  key: "sink_token_length"
+  value: {
+    string_value: "${sink_token_length}"
+  }
+}
+parameters: {
+  key: "batch_scheduler_policy"
+  value: {
+    string_value: "${batch_scheduler_policy}"
+  }
+}
+parameters: {
+  key: "kv_cache_free_gpu_mem_fraction"
+  value: {
+    string_value: "${kv_cache_free_gpu_mem_fraction}"
+  }
+}
+parameters: {
+  key: "cross_kv_cache_fraction"
+  value: {
+    string_value: "${cross_kv_cache_fraction}"
+  }
+}
+parameters: {
+  key: "kv_cache_host_memory_bytes"
+  value: {
+    string_value: "${kv_cache_host_memory_bytes}"
+  }
+}
+# kv_cache_onboard_blocks is for internal implementation.
+parameters: {
+  key: "kv_cache_onboard_blocks"
+  value: {
+    string_value: "${kv_cache_onboard_blocks}"
+  }
+}
+# enable_trt_overlap is deprecated and doesn't have any effect on the runtime
+# parameters: {
+#   key: "enable_trt_overlap"
+#   value: {
+#     string_value: "${enable_trt_overlap}"
+#   }
+# }
+parameters: {
+  key: "exclude_input_in_output"
+  value: {
+    string_value: "${exclude_input_in_output}"
+  }
+}
+parameters: {
+  key: "cancellation_check_period_ms"
+  value: {
+    string_value: "${cancellation_check_period_ms}"
+  }
+}
+parameters: {
+  key: "stats_check_period_ms"
+  value: {
+    string_value: "${stats_check_period_ms}"
+  }
+}
+parameters: {
+  key: "iter_stats_max_iterations"
+  value: {
+    string_value: "${iter_stats_max_iterations}"
+  }
+}
+parameters: {
+  key: "request_stats_max_iterations"
+  value: {
+    string_value: "${request_stats_max_iterations}"
+  }
+}
+parameters: {
+  key: "enable_kv_cache_reuse"
+  value: {
+    string_value: "${enable_kv_cache_reuse}"
+  }
+}
+parameters: {
+  key: "normalize_log_probs"
+  value: {
+    string_value: "${normalize_log_probs}"
+  }
+}
+parameters: {
+  key: "enable_chunked_context"
+  value: {
+    string_value: "${enable_chunked_context}"
+  }
+}
+parameters: {
+  key: "gpu_device_ids"
+  value: {
+    string_value: "${gpu_device_ids}"
+  }
+}
+parameters: {
+  key: "participant_ids"
+  value: {
+    string_value: "${participant_ids}"
+  }
+}
+parameters: {
+  key: "lora_cache_optimal_adapter_size"
+  value: {
+    string_value: "${lora_cache_optimal_adapter_size}"
+  }
+}
+parameters: {
+  key: "lora_cache_max_adapter_size"
+  value: {
+    string_value: "${lora_cache_max_adapter_size}"
+  }
+}
+parameters: {
+  key: "lora_cache_gpu_memory_fraction"
+  value: {
+    string_value: "${lora_cache_gpu_memory_fraction}"
+  }
+}
+parameters: {
+  key: "lora_cache_host_memory_bytes"
+  value: {
+    string_value: "${lora_cache_host_memory_bytes}"
+  }
+}
+parameters: {
+  key: "lora_prefetch_dir"
+  value: {
+    string_value: "${lora_prefetch_dir}"
+  }
+}
+parameters: {
+  key: "decoding_mode"
+  value: {
+    string_value: "${decoding_mode}"
+  }
+}
+parameters: {
+  key: "executor_worker_path"
+  value: {
+    string_value: "/opt/tritonserver/backends/tensorrtllm/trtllmExecutorWorker"
+  }
+}
+parameters: {
+  key: "lookahead_window_size"
+    value: {
+      string_value: "${lookahead_window_size}"
+  }
+}
+parameters: {
+  key: "lookahead_ngram_size"
+    value: {
+      string_value: "${lookahead_ngram_size}"
+  }
+}
+parameters: {
+  key: "lookahead_verification_set_size"
+    value: {
+      string_value: "${lookahead_verification_set_size}"
+  }
+}
+parameters: {
+  key: "medusa_choices"
+    value: {
+      string_value: "${medusa_choices}"
+  }
+}
+parameters: {
+  key: "eagle_choices"
+    value: {
+      string_value: "${eagle_choices}"
+  }
+}
+parameters: {
+  key: "gpu_weights_percent"
+    value: {
+      string_value: "${gpu_weights_percent}"
+  }
+}
+parameters: {
+  key: "enable_context_fmha_fp32_acc"
+  value: {
+    string_value: "${enable_context_fmha_fp32_acc}"
+  }
+}
+parameters: {
+  key: "multi_block_mode"
+  value: {
+    string_value: "${multi_block_mode}"
+  }
+}
+parameters: {
+  key: "cuda_graph_mode"
+  value: {
+    string_value: "${cuda_graph_mode}"
+  }
+}
+parameters: {
+  key: "cuda_graph_cache_size"
+  value: {
+    string_value: "${cuda_graph_cache_size}"
+  }
+}
+parameters: {
+  key: "speculative_decoding_fast_logits"
+  value: {
+    string_value: "${speculative_decoding_fast_logits}"
+  }
+}
+parameters: {
+  key: "tokenizer_dir"
+  value: {
+    string_value: "${tokenizer_dir}"
+  }
+}
+parameters: {
+  key: "guided_decoding_backend"
+  value: {
+    string_value: "${guided_decoding_backend}"
+  }
+}
+parameters: {
+  key: "xgrammar_tokenizer_info_path"
+  value: {
+    string_value: "${xgrammar_tokenizer_info_path}"
+  }
+}

runtime/triton_trtllm/model_repo/token2wav/1/model.py

@@ -0,0 +1,277 @@
+# Copyright 2025, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions
+# are met:
+#  * Redistributions of source code must retain the above copyright
+#    notice, this list of conditions and the following disclaimer.
+#  * Redistributions in binary form must reproduce the above copyright
+#    notice, this list of conditions and the following disclaimer in the
+#    documentation and/or other materials provided with the distribution.
+#  * Neither the name of NVIDIA CORPORATION nor the names of its
+#    contributors may be used to endorse or promote products derived
+#    from this software without specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS ``AS IS'' AND ANY
+# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+# PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+# OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+# (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+import json
+import os
+
+import logging
+
+import torch
+from torch.utils.dlpack import to_dlpack
+from torch.nn import functional as F
+
+import triton_python_backend_utils as pb_utils
+
+from hyperpyyaml import load_hyperpyyaml
+from cosyvoice.utils.common import fade_in_out
+from cosyvoice.utils.file_utils import convert_onnx_to_trt, export_cosyvoice2_vllm
+from cosyvoice.utils.common import TrtContextWrapper
+from collections import defaultdict
+import numpy as np
+
+logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(name)s - %(levelname)s - %(message)s')
+logger = logging.getLogger(__name__)
+
+ORIGINAL_VOCAB_SIZE = 151663
+torch.set_num_threads(1)
+
+
+class CosyVoice2:
+
+    def __init__(self, model_dir, load_jit=False, load_trt=False, fp16=False, trt_concurrent=1, device='cuda'):
+
+        self.model_dir = model_dir
+        self.fp16 = fp16
+
+        hyper_yaml_path = '{}/cosyvoice2.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')})
+        self.model = CosyVoice2Model(configs['flow'], configs['hift'], fp16, device)
+        self.model.load('{}/flow.pt'.format(model_dir), '{}/hift.pt'.format(model_dir))
+        if load_jit:
+            self.model.load_jit('{}/flow.encoder.{}.zip'.format(model_dir, 'fp16' if self.fp16 is True else 'fp32'))
+        if load_trt:
+            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)
+
+
+class CosyVoice2Model:
+
+    def __init__(self,
+                 flow: torch.nn.Module,
+                 hift: torch.nn.Module,
+                 fp16: bool = False,
+                 device: str = 'cuda'):
+        self.device = device
+        self.flow = flow
+        self.hift = hift
+        self.fp16 = fp16
+        if self.fp16 is True:
+            self.flow.half()
+
+        # streaming tts config
+        self.token_hop_len = 25
+        self.mel_cache_len = 8
+        self.source_cache_len = int(self.mel_cache_len * 480)
+        self.speech_window = np.hamming(2 * self.source_cache_len)
+        self.hift_cache_dict = defaultdict(lambda: None)
+
+    def load_jit(self, flow_encoder_model):
+        flow_encoder = torch.jit.load(flow_encoder_model, map_location=self.device)
+        self.flow.encoder = flow_encoder
+
+    def load(self, flow_model, hift_model):
+        self.flow.load_state_dict(torch.load(flow_model, map_location=self.device), strict=True)
+        self.flow.to(self.device).eval()
+        # in case hift_model is a hifigan model
+        hift_state_dict = {k.replace('generator.', ''): v for k, v in torch.load(hift_model, map_location=self.device).items()}
+        self.hift.load_state_dict(hift_state_dict, strict=True)
+        self.hift.to(self.device).eval()
+
+    def load_trt(self, flow_decoder_estimator_model, flow_decoder_onnx_model, trt_concurrent, fp16):
+        assert torch.cuda.is_available(), 'tensorrt only supports gpu!'
+        if not os.path.exists(flow_decoder_estimator_model) or os.path.getsize(flow_decoder_estimator_model) == 0:
+            convert_onnx_to_trt(flow_decoder_estimator_model, self.get_trt_kwargs(), flow_decoder_onnx_model, fp16)
+        del self.flow.decoder.estimator
+        import tensorrt as trt
+        with open(flow_decoder_estimator_model, 'rb') as f:
+            estimator_engine = trt.Runtime(trt.Logger(trt.Logger.INFO)).deserialize_cuda_engine(f.read())
+        assert estimator_engine is not None, 'failed to load trt {}'.format(flow_decoder_estimator_model)
+        self.flow.decoder.estimator = TrtContextWrapper(estimator_engine, trt_concurrent=trt_concurrent, device=self.device)
+
+    def get_trt_kwargs(self):
+        min_shape = [(2, 80, 4), (2, 1, 4), (2, 80, 4), (2, 80, 4)]
+        opt_shape = [(2, 80, 500), (2, 1, 500), (2, 80, 500), (2, 80, 500)]
+        max_shape = [(2, 80, 3000), (2, 1, 3000), (2, 80, 3000), (2, 80, 3000)]
+        input_names = ["x", "mask", "mu", "cond"]
+        return {'min_shape': min_shape, 'opt_shape': opt_shape, 'max_shape': max_shape, 'input_names': input_names}
+
+    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),
+                                             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 hift cache
+        if self.hift_cache_dict[uuid] is not None:
+            hift_cache_mel, hift_cache_source = self.hift_cache_dict[uuid]['mel'], self.hift_cache_dict[uuid]['source']
+            tts_mel = torch.concat([hift_cache_mel, tts_mel], dim=2)
+        else:
+            hift_cache_source = torch.zeros(1, 1, 0)
+        # keep overlap mel and hift cache
+        if finalize is False:
+            tts_speech, tts_source = self.hift.inference(speech_feat=tts_mel, cache_source=hift_cache_source)
+            if self.hift_cache_dict[uuid] is not None:
+                tts_speech = fade_in_out(tts_speech, self.hift_cache_dict[uuid]['speech'], self.speech_window)
+            self.hift_cache_dict[uuid] = {'mel': tts_mel[:, :, -self.mel_cache_len:],
+                                          'source': tts_source[:, :, -self.source_cache_len:],
+                                          'speech': tts_speech[:, -self.source_cache_len:]}
+            tts_speech = tts_speech[:, :-self.source_cache_len]
+        else:
+            if speed != 1.0:
+                assert self.hift_cache_dict[uuid] is None, '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, tts_source = self.hift.inference(speech_feat=tts_mel, cache_source=hift_cache_source)
+            if self.hift_cache_dict[uuid] is not None:
+                tts_speech = fade_in_out(tts_speech, self.hift_cache_dict[uuid]['speech'], self.speech_window)
+        return tts_speech
+
+
+class TritonPythonModel:
+    """Triton Python model for vocoder.
+
+    This model takes global and semantic tokens as input and generates audio waveforms
+    using the BiCodec vocoder.
+    """
+
+    def initialize(self, args):
+        """Initialize the model.
+
+        Args:
+            args: Dictionary containing model configuration
+        """
+        # Parse model parameters
+        parameters = json.loads(args['model_config'])['parameters']
+        model_params = {key: value["string_value"] for key, value in parameters.items()}
+        model_dir = model_params["model_dir"]
+
+        # Initialize device and vocoder
+        self.device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
+        logger.info(f"Initializing vocoder from {model_dir} on {self.device}")
+
+        self.token2wav_model = CosyVoice2(
+            model_dir, load_jit=False, load_trt=True, fp16=True, device=self.device
+        )
+
+        spk_info_path = os.path.join(model_dir, "spk2info.pt")
+        if not os.path.exists(spk_info_path):
+            raise ValueError(f"spk2info.pt not found in {model_dir}")
+        spk_info = torch.load(spk_info_path, map_location="cpu", weights_only=False)
+        self.default_spk_info = spk_info["001"]
+
+        logger.info("Token2Wav initialized successfully")
+
+    def execute(self, requests):
+        """Execute inference on the batched requests.
+
+        Args:
+            requests: List of inference requests
+
+        Returns:
+            List of inference responses containing generated waveforms
+        """
+        responses = []
+        # Process each request in batch
+        for request in requests:
+            target_speech_tokens_tensor = pb_utils.get_input_tensor_by_name(request, "target_speech_tokens").as_numpy()
+            target_speech_tokens = torch.from_numpy(target_speech_tokens_tensor).to(self.device)
+
+            prompt_speech_tokens_tensor = pb_utils.get_input_tensor_by_name(request, "prompt_speech_tokens")
+            if prompt_speech_tokens_tensor is not None:
+                prompt_speech_tokens_tensor = prompt_speech_tokens_tensor.as_numpy()
+                prompt_speech_feat_tensor = pb_utils.get_input_tensor_by_name(request, "prompt_speech_feat").as_numpy()
+                prompt_spk_embedding_tensor = pb_utils.get_input_tensor_by_name(request, "prompt_spk_embedding").as_numpy()
+                prompt_speech_tokens = torch.from_numpy(prompt_speech_tokens_tensor).to(self.device)
+                prompt_speech_feat = torch.from_numpy(prompt_speech_feat_tensor).to(self.device)
+                prompt_spk_embedding = torch.from_numpy(prompt_spk_embedding_tensor).to(self.device)
+                prompt_speech_tokens = prompt_speech_tokens - ORIGINAL_VOCAB_SIZE
+            else:
+                prompt_speech_tokens = self.default_spk_info["speech_token"].to(self.device)
+                prompt_speech_feat = self.default_spk_info["speech_feat"].to(torch.float16).to(self.device)
+                prompt_spk_embedding = self.default_spk_info["embedding"].to(torch.float16).to(self.device)
+
+            # shift the speech tokens according to the original vocab size
+            target_speech_tokens = target_speech_tokens - ORIGINAL_VOCAB_SIZE
+
+            # We set token_offset as an optional input to support streaming/offline tts. It has to be None when offline tts.
+            token_offset = pb_utils.get_input_tensor_by_name(request, "token_offset")
+            if token_offset is not None:
+                token_offset = token_offset.as_numpy().item()
+                finalize = pb_utils.get_input_tensor_by_name(request, "finalize").as_numpy().item()
+                if not finalize:
+                    stream = True
+                else:
+                    stream = False
+                request_id = request.request_id()
+                audio_hat = self.token2wav_model.model.token2wav(token=target_speech_tokens,
+                                                                 prompt_token=prompt_speech_tokens,
+                                                                 prompt_feat=prompt_speech_feat,
+                                                                 embedding=prompt_spk_embedding,
+                                                                 token_offset=token_offset,
+                                                                 uuid=request_id,
+                                                                 stream=stream,
+                                                                 finalize=finalize)
+                if finalize:
+                    self.token2wav_model.model.hift_cache_dict.pop(request_id)
+
+            else:
+                tts_mel, _ = self.token2wav_model.model.flow.inference(
+                    token=target_speech_tokens,
+                    token_len=torch.tensor([target_speech_tokens.shape[1]], dtype=torch.int32).to(
+                        self.device
+                    ),
+                    prompt_token=prompt_speech_tokens,
+                    prompt_token_len=torch.tensor(
+                        [prompt_speech_tokens.shape[1]], dtype=torch.int32
+                    ).to(self.device),
+                    prompt_feat=prompt_speech_feat,
+                    prompt_feat_len=torch.tensor([prompt_speech_feat.shape[1]], dtype=torch.int32).to(self.device),
+                    embedding=prompt_spk_embedding,
+                    streaming=False,
+                    finalize=True,
+                )
+
+                audio_hat, _ = self.token2wav_model.model.hift.inference(
+                    speech_feat=tts_mel, cache_source=torch.zeros(1, 1, 0)
+                )
+
+            generated_wave = audio_hat.squeeze(0).cpu().numpy()
+
+            wav_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(audio_hat))
+            inference_response = pb_utils.InferenceResponse(output_tensors=[wav_tensor])
+            responses.append(inference_response)
+
+        return responses

runtime/triton_trtllm/model_repo/token2wav/config.pbtxt

@@ -0,0 +1,80 @@
+# Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+#
+# 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.
+
+name: "token2wav"
+backend: "python"
+max_batch_size: ${triton_max_batch_size}
+dynamic_batching {
+    max_queue_delay_microseconds: ${max_queue_delay_microseconds}
+}
+parameters [
+  {
+   key: "model_dir",
+   value: {string_value:"${model_dir}"}
+  }
+]
+
+input [
+  {
+    name: "target_speech_tokens"
+    data_type: TYPE_INT32
+    dims: [-1]
+  },
+  {
+    name: "prompt_speech_tokens"
+    data_type: TYPE_INT32
+    dims: [-1]
+    optional: true
+  },
+  {
+    name: "prompt_speech_feat"
+    data_type: TYPE_FP16
+    dims: [-1, 80]
+    optional: true
+  },
+  {
+    name: "prompt_spk_embedding"
+    data_type: TYPE_FP16
+    dims: [-1]
+    optional: true
+  },
+  {
+    name: "token_offset"
+    data_type: TYPE_INT32
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  },
+  {
+    name: "finalize"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  }
+]
+output [
+  {
+    name: "waveform"
+    data_type: TYPE_FP32
+    dims: [ -1 ]
+  }
+]
+
+instance_group [
+  {
+    count: 1
+    kind: KIND_CPU
+  }
+]

runtime/triton_trtllm/model_repo/token2wav_dit/1/model.py

@@ -0,0 +1,142 @@
+# Copyright 2025, NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+#
+# Redistribution and use in source and binary forms, with or without
+# modification, are permitted provided that the following conditions
+# are met:
+#  * Redistributions of source code must retain the above copyright
+#    notice, this list of conditions and the following disclaimer.
+#  * Redistributions in binary form must reproduce the above copyright
+#    notice, this list of conditions and the following disclaimer in the
+#    documentation and/or other materials provided with the distribution.
+#  * Neither the name of NVIDIA CORPORATION nor the names of its
+#    contributors may be used to endorse or promote products derived
+#    from this software without specific prior written permission.
+#
+# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS ``AS IS'' AND ANY
+# EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+# PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+# CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+# EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+# PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY
+# OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+# (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+import json
+import os
+
+import logging
+from typing import List, Dict
+
+import torch
+from torch.utils.dlpack import to_dlpack
+from torch.nn import functional as F
+
+import triton_python_backend_utils as pb_utils
+
+from hyperpyyaml import load_hyperpyyaml
+from cosyvoice.utils.common import fade_in_out
+from cosyvoice.utils.file_utils import convert_onnx_to_trt, export_cosyvoice2_vllm
+from cosyvoice.utils.common import TrtContextWrapper
+from collections import defaultdict
+import numpy as np
+from .token2wav_dit import CosyVoice2_Token2Wav
+import hashlib
+
+logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(name)s - %(levelname)s - %(message)s')
+logger = logging.getLogger(__name__)
+
+
+ORIGINAL_VOCAB_SIZE = 151663
+torch.set_num_threads(1)
+
+
+def get_spk_id_from_prompt_audio(tensor: torch.Tensor) -> str:
+    """
+    Generates a unique ID for a torch.Tensor.
+    Tensors with the same elements and properties will have the same ID.
+    """
+    # Convert tensor to a byte string
+    tensor_bytes = tensor.numpy().tobytes()
+
+    # Create a SHA-256 hash of the byte string
+    hasher = hashlib.sha256()
+    hasher.update(tensor_bytes)
+
+    return hasher.hexdigest()
+
+
+class TritonPythonModel:
+    """Triton Python model for vocoder.
+
+    This model takes global and semantic tokens as input and generates audio waveforms
+    using the BiCodec vocoder.
+    """
+
+    def initialize(self, args):
+        """Initialize the model.
+
+        Args:
+            args: Dictionary containing model configuration
+        """
+        # Parse model parameters
+        parameters = json.loads(args['model_config'])['parameters']
+        model_params = {key: value["string_value"] for key, value in parameters.items()}
+        model_dir = model_params["model_dir"]
+
+        # Initialize device and vocoder
+        self.device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
+        logger.info(f"Initializing vocoder from {model_dir} on {self.device}")
+
+        # FIXME: device id settings
+        self.token2wav_model = CosyVoice2_Token2Wav(
+            model_dir, enable_trt=True, streaming=True
+        )
+        logger.info("Token2Wav initialized successfully")
+
+    def execute(self, requests):
+        """Execute inference on the batched requests.
+
+        Args:
+            requests: List of inference requests
+
+        Returns:
+            List of inference responses containing generated waveforms
+        """
+        responses = []
+        # Process each request in batch
+        for request in requests:
+            target_speech_tokens_tensor = pb_utils.get_input_tensor_by_name(request, "target_speech_tokens").as_numpy()
+            target_speech_tokens = torch.from_numpy(target_speech_tokens_tensor)
+            target_speech_tokens = target_speech_tokens - ORIGINAL_VOCAB_SIZE
+            target_speech_tokens = target_speech_tokens.squeeze().tolist()
+
+            finalize = pb_utils.get_input_tensor_by_name(request, "finalize").as_numpy().item()
+
+            request_id = request.request_id()
+
+            wav_array = pb_utils.get_input_tensor_by_name(
+                request, "reference_wav").as_numpy()
+            wav_len = pb_utils.get_input_tensor_by_name(
+                request, "reference_wav_len").as_numpy().item()
+
+            wav_array = torch.from_numpy(wav_array)
+            wav = wav_array[:, :wav_len].squeeze(0)
+
+            spk_id = get_spk_id_from_prompt_audio(wav)
+
+            audio_hat = self.token2wav_model.forward_streaming(
+                target_speech_tokens, finalize, request_id=request_id,
+                speaker_id=f"{spk_id}", prompt_audio=wav, prompt_audio_sample_rate=16000
+            )
+
+            outputs = []
+
+            wav_tensor = pb_utils.Tensor.from_dlpack("waveform", to_dlpack(audio_hat))
+            outputs.append(wav_tensor)
+            inference_response = pb_utils.InferenceResponse(output_tensors=outputs)
+            responses.append(inference_response)
+
+        return responses

runtime/triton_trtllm/model_repo/token2wav_dit/1/token2wav_dit.py

@@ -0,0 +1,510 @@
+# SPDX-FileCopyrightText: Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+""" Example Usage
+    CUDA_VISIBLE_DEVICES=0 \
+        python3 token2wav.py --enable-trt || exit 1
+"""
+import torch
+# from flashcosyvoice.modules.flow import CausalMaskedDiffWithXvec
+from flashcosyvoice.modules.hifigan import HiFTGenerator
+from flashcosyvoice.utils.audio import mel_spectrogram
+import torchaudio.compliance.kaldi as kaldi
+import onnxruntime
+import s3tokenizer
+from torch.utils.data import DataLoader
+from datasets import load_dataset
+import torchaudio
+import os
+import logging
+import argparse
+import queue
+import time
+import numpy as np
+from hyperpyyaml import load_hyperpyyaml
+
+
+def fade_in_out(fade_in_mel: torch.Tensor, fade_out_mel: torch.Tensor, window: torch.Tensor):
+    """perform fade_in_out in tensor style
+    """
+    mel_overlap_len = int(window.shape[0] / 2)
+    fade_in_mel = fade_in_mel.clone()
+    fade_in_mel[..., :mel_overlap_len] = \
+        fade_in_mel[..., :mel_overlap_len] * window[:mel_overlap_len] + \
+        fade_out_mel[..., -mel_overlap_len:] * window[mel_overlap_len:]
+    return fade_in_mel
+
+
+def convert_onnx_to_trt(trt_model, trt_kwargs, onnx_model, dtype):
+    import tensorrt as trt
+    logging.info("Converting onnx to trt...")
+    network_flags = 1 << int(trt.NetworkDefinitionCreationFlag.EXPLICIT_BATCH)
+    logger = trt.Logger(trt.Logger.INFO)
+    builder = trt.Builder(logger)
+    network = builder.create_network(network_flags)
+    parser = trt.OnnxParser(network, logger)
+    config = builder.create_builder_config()
+    # config.set_memory_pool_limit(trt.MemoryPoolType.WORKSPACE, 1 << 32)  # 4GB
+    if dtype == torch.float16:
+        config.set_flag(trt.BuilderFlag.FP16)
+
+    profile = builder.create_optimization_profile()
+    # load onnx model
+    with open(onnx_model, "rb") as f:
+        if not parser.parse(f.read()):
+            for error in range(parser.num_errors):
+                print(parser.get_error(error))
+            raise ValueError('failed to parse {}'.format(onnx_model))
+    # set input shapes
+    for i in range(len(trt_kwargs['input_names'])):
+        profile.set_shape(trt_kwargs['input_names'][i], trt_kwargs['min_shape'][i], trt_kwargs['opt_shape'][i], trt_kwargs['max_shape'][i])
+    if dtype == torch.float16:
+        tensor_dtype = trt.DataType.HALF
+    elif dtype == torch.bfloat16:
+        tensor_dtype = trt.DataType.BF16
+    elif dtype == torch.float32:
+        tensor_dtype = trt.DataType.FLOAT
+    else:
+        raise ValueError('invalid dtype {}'.format(dtype))
+    # set input and output data type
+    for i in range(network.num_inputs):
+        input_tensor = network.get_input(i)
+        input_tensor.dtype = tensor_dtype
+    for i in range(network.num_outputs):
+        output_tensor = network.get_output(i)
+        output_tensor.dtype = tensor_dtype
+    config.add_optimization_profile(profile)
+    engine_bytes = builder.build_serialized_network(network, config)
+    # save trt engine
+    with open(trt_model, "wb") as f:
+        f.write(engine_bytes)
+    logging.info("Succesfully convert onnx to trt...")
+
+
+class TrtContextWrapper:
+    def __init__(self, trt_engine, trt_concurrent=1, device='cuda:0'):
+        self.trt_context_pool = queue.Queue(maxsize=trt_concurrent)
+        self.trt_engine = trt_engine
+        self.device = device
+        for _ in range(trt_concurrent):
+            trt_context = trt_engine.create_execution_context()
+            trt_stream = torch.cuda.stream(torch.cuda.Stream(torch.device(device)))
+            assert trt_context is not None, 'failed to create trt context, maybe not enough CUDA memory, try reduce current trt concurrent {}'.format(trt_concurrent)
+            self.trt_context_pool.put([trt_context, trt_stream])
+        assert self.trt_context_pool.empty() is False, 'no avaialbe estimator context'
+
+    def acquire_estimator(self):
+        return self.trt_context_pool.get(), self.trt_engine
+
+    def release_estimator(self, context, stream):
+        self.trt_context_pool.put([context, stream])
+
+
+class CosyVoice2_Token2Wav(torch.nn.Module):
+    def __init__(self, model_dir: str, enable_trt: bool = False, device_id: int = 0, streaming: bool = False, dtype: torch.dtype = torch.float16):
+        super().__init__()
+        self.device_id = device_id
+        self.device = f"cuda:{device_id}"
+        with open(f"{model_dir}/flow.yaml", "r") as f:
+            configs = load_hyperpyyaml(f)
+            self.flow = configs['flow']
+
+        self.dtype = dtype
+        self.flow.to(self.dtype)
+
+        self.flow.load_state_dict(torch.load(f"{model_dir}/flow.pt", map_location="cpu", weights_only=True), strict=True)
+        self.flow.to(self.device).eval()
+
+        self.hift = HiFTGenerator()
+        hift_state_dict = {k.replace('generator.', ''): v for k, v in torch.load(f"{model_dir}/hift.pt", map_location="cpu", weights_only=True).items()}
+        self.hift.load_state_dict(hift_state_dict, strict=True)
+        self.hift.to(self.device).eval()
+
+        option = onnxruntime.SessionOptions()
+        option.graph_optimization_level = onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
+        option.intra_op_num_threads = 1
+        self.spk_model = onnxruntime.InferenceSession(
+            f"{model_dir}/campplus.onnx", sess_options=option,
+            providers=["CPUExecutionProvider"])
+        self.audio_tokenizer = s3tokenizer.load_model(f"{model_dir}/speech_tokenizer_v2_25hz.onnx").to(self.device).eval()
+
+        gpu = "l20"
+        if enable_trt:
+            if streaming:
+                self.load_trt(
+                    f'{model_dir}/flow.decoder.estimator.{self.dtype}.dynamic_batch.chunk.{gpu}.plan',
+                    f'{model_dir}/flow.decoder.estimator.chunk.fp32.dynamic_batch.simplify.onnx',
+                    1,
+                    self.dtype, streaming
+                )
+            else:
+                self.load_trt(
+                    f'{model_dir}/flow.decoder.estimator.{self.dtype}.dynamic_batch.{gpu}.plan',
+                    f'{model_dir}/flow.decoder.estimator.fp32.dynamic_batch.onnx',
+                    1,
+                    self.dtype
+                )
+            self.load_spk_trt(
+                f'{model_dir}/campplus.{gpu}.fp32.trt',
+                f'{model_dir}/campplus.onnx',
+                1,
+                False
+            )
+
+        self.streaming_flow_cache = {}
+        self.speaker_cache = {}
+
+        self.mel_cache_len = 8  # hard-coded, 160ms
+        self.source_cache_len = int(self.mel_cache_len * 480)   # 50hz mel -> 24kHz wave
+        self.speech_window = torch.from_numpy(np.hamming(2 * self.source_cache_len)).cuda()
+
+        # hifigan cache for streaming tts
+        self.hift_cache_dict = {}
+
+    def forward_spk_embedding(self, spk_feat):
+        if isinstance(self.spk_model, onnxruntime.InferenceSession):
+            return self.spk_model.run(
+                None, {self.spk_model.get_inputs()[0].name: spk_feat.unsqueeze(dim=0).cpu().numpy()}
+            )[0].flatten().tolist()
+        else:
+            [spk_model, stream], trt_engine = self.spk_model.acquire_estimator()
+            # NOTE need to synchronize when switching stream
+            with torch.cuda.device(self.device_id):
+                torch.cuda.current_stream().synchronize()
+                spk_feat = spk_feat.unsqueeze(dim=0).to(self.device)
+                batch_size = spk_feat.size(0)
+
+                with stream:
+                    spk_model.set_input_shape('input', (batch_size, spk_feat.size(1), 80))
+                    output_tensor = torch.empty((batch_size, 192), device=spk_feat.device)
+
+                    data_ptrs = [spk_feat.contiguous().data_ptr(),
+                                 output_tensor.contiguous().data_ptr()]
+                    for i, j in enumerate(data_ptrs):
+
+                        spk_model.set_tensor_address(trt_engine.get_tensor_name(i), j)
+                    # run trt engine
+                    assert spk_model.execute_async_v3(torch.cuda.current_stream().cuda_stream) is True
+                    torch.cuda.current_stream().synchronize()
+                self.spk_model.release_estimator(spk_model, stream)
+
+            return output_tensor.cpu().numpy().flatten().tolist()
+
+    def load_spk_trt(self, spk_model, spk_onnx_model, trt_concurrent=1, fp16=True):
+        if not os.path.exists(spk_model) or os.path.getsize(spk_model) == 0:
+            trt_kwargs = self.get_spk_trt_kwargs()
+            convert_onnx_to_trt(spk_model, trt_kwargs, spk_onnx_model, torch.float32)
+        import tensorrt as trt
+        with open(spk_model, 'rb') as f:
+            spk_engine = trt.Runtime(trt.Logger(trt.Logger.INFO)).deserialize_cuda_engine(f.read())
+        assert spk_engine is not None, 'failed to load trt {}'.format(spk_model)
+        self.spk_model = TrtContextWrapper(spk_engine, trt_concurrent=trt_concurrent, device=self.device)
+
+    def get_spk_trt_kwargs(self):
+        min_shape = [(1, 4, 80)]
+        opt_shape = [(1, 500, 80)]
+        max_shape = [(1, 3000, 80)]
+        input_names = ["input"]
+        return {'min_shape': min_shape, 'opt_shape': opt_shape, 'max_shape': max_shape, 'input_names': input_names}
+
+    def load_trt(self, flow_decoder_estimator_model, flow_decoder_onnx_model, trt_concurrent=1, dtype=torch.float16, streaming=False):
+        assert torch.cuda.is_available(), 'tensorrt only supports gpu!'
+        if not os.path.exists(flow_decoder_estimator_model) or os.path.getsize(flow_decoder_estimator_model) == 0:
+            opt_batch_size = 2
+            max_batch_size = 16
+            if streaming:
+                opt_batch_size, max_batch_size = 1, 1  # only support batch size 1 for streaming tts
+            trt_kwargs = self.get_trt_kwargs_dynamic_batch(opt_batch_size=opt_batch_size, max_batch_size=max_batch_size, streaming=streaming)
+            convert_onnx_to_trt(flow_decoder_estimator_model, trt_kwargs, flow_decoder_onnx_model, dtype)
+        del self.flow.decoder.estimator
+        import tensorrt as trt
+        with open(flow_decoder_estimator_model, 'rb') as f:
+            estimator_engine = trt.Runtime(trt.Logger(trt.Logger.INFO)).deserialize_cuda_engine(f.read())
+        assert estimator_engine is not None, 'failed to load trt {}'.format(flow_decoder_estimator_model)
+        self.flow.decoder.estimator = TrtContextWrapper(estimator_engine, trt_concurrent=trt_concurrent, device=self.device)
+
+    def get_trt_kwargs_dynamic_batch(self, opt_batch_size=2, max_batch_size=64, streaming=False):
+        if streaming:
+            min_shape = [(2, 80, 4), (2, 80, 4), (2, 80, 4), (2,), (2, 80), (16, 2, 1024, 2), (16, 2, 8, 0, 128)]
+            opt_shape = [
+                (opt_batch_size * 2, 80, 500), (opt_batch_size * 2, 80, 500), (opt_batch_size * 2, 80, 500),
+                (opt_batch_size * 2,), (opt_batch_size * 2, 80), (16, opt_batch_size * 2, 1024, 2),
+                (16, opt_batch_size * 2, 8, 100, 128)
+            ]
+            max_shape = [
+                (max_batch_size * 2, 80, 3000), (max_batch_size * 2, 80, 3000), (max_batch_size * 2, 80, 3000),
+                (max_batch_size * 2,), (max_batch_size * 2, 80), (16, max_batch_size * 2, 1024, 2),
+                (16, max_batch_size * 2, 8, 1000, 128)
+            ]
+            input_names = ["x", "mu", "cond", "t", "spks", "cnn_cache", "att_cache"]
+        else:
+            min_shape = [(2, 80, 4), (2, 1, 4), (2, 80, 4), (2, 80, 4), (2,), (2, 80)]
+            opt_shape = [
+                (opt_batch_size * 2, 80, 500), (opt_batch_size * 2, 1, 500), (opt_batch_size * 2, 80, 500),
+                (opt_batch_size * 2, 80, 500), (opt_batch_size * 2,), (opt_batch_size * 2, 80)
+            ]
+            max_shape = [
+                (max_batch_size * 2, 80, 3000), (max_batch_size * 2, 1, 3000), (max_batch_size * 2, 80, 3000),
+                (max_batch_size * 2, 80, 3000), (max_batch_size * 2,), (max_batch_size * 2, 80)
+            ]
+            input_names = ["x", "mask", "mu", "cond", "t", "spks"]
+        return {'min_shape': min_shape, 'opt_shape': opt_shape, 'max_shape': max_shape, 'input_names': input_names}
+
+    def prompt_audio_tokenization(self, prompt_audios_list: list[torch.Tensor]) -> list[list[int]]:
+        prompt_speech_tokens_list, prompt_speech_mels_list = [], []
+        for audio in prompt_audios_list:
+            assert len(audio.shape) == 1
+            log_mel = s3tokenizer.log_mel_spectrogram(audio)  # [num_mels, T]
+            prompt_speech_mels_list.append(log_mel)
+        prompt_mels_for_llm, prompt_mels_lens_for_llm = s3tokenizer.padding(prompt_speech_mels_list)
+        prompt_speech_tokens, prompt_speech_tokens_lens = self.audio_tokenizer.quantize(
+            prompt_mels_for_llm.to(self.device), prompt_mels_lens_for_llm.to(self.device)
+        )
+        for i in range(len(prompt_speech_tokens)):
+            speech_tokens_i = prompt_speech_tokens[i, :prompt_speech_tokens_lens[i].item()].tolist()
+            prompt_speech_tokens_list.append(speech_tokens_i)
+        return prompt_speech_tokens_list
+
+    def get_spk_emb(self, prompt_audios_list: list[torch.Tensor]) -> torch.Tensor:
+        spk_emb_for_flow = []
+        for audio in prompt_audios_list:
+            assert len(audio.shape) == 1
+            spk_feat = kaldi.fbank(audio.unsqueeze(0), num_mel_bins=80, dither=0, sample_frequency=16000)
+            spk_feat = spk_feat - spk_feat.mean(dim=0, keepdim=True)
+            spk_emb = self.forward_spk_embedding(spk_feat)
+
+            spk_emb_for_flow.append(spk_emb)
+        spk_emb_for_flow = torch.tensor(spk_emb_for_flow)
+        if self.dtype != torch.float32:
+            spk_emb_for_flow = spk_emb_for_flow.to(self.dtype)
+        return spk_emb_for_flow
+
+    def get_prompt_mels(self, prompt_audios_list: list[torch.Tensor], prompt_audios_sample_rate: list[int]):
+        prompt_mels_for_flow = []
+        prompt_mels_lens_for_flow = []
+        for audio, sample_rate in zip(prompt_audios_list, prompt_audios_sample_rate):
+            assert len(audio.shape) == 1
+            audio = audio.unsqueeze(0)
+            if sample_rate != 24000:
+                audio = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=24000)(audio)
+            mel = mel_spectrogram(audio).transpose(1, 2).squeeze(0)  # [T, num_mels]
+            mel_len = mel.shape[0]
+            prompt_mels_for_flow.append(mel)
+            prompt_mels_lens_for_flow.append(mel_len)
+        prompt_mels_for_flow = torch.nn.utils.rnn.pad_sequence(
+            prompt_mels_for_flow, batch_first=True, padding_value=0
+        )  # [B, T', num_mels=80]
+        prompt_mels_lens_for_flow = torch.tensor(prompt_mels_lens_for_flow)
+        return prompt_mels_for_flow, prompt_mels_lens_for_flow
+
+    def forward_flow(self, prompt_speech_tokens_list: list[list[int]],
+                     generated_speech_tokens_list: list[list[int]],
+                     prompt_mels_for_flow: torch.Tensor,
+                     prompt_mels_lens_for_flow: torch.Tensor,
+                     spk_emb_for_flow: torch.Tensor):
+        batch_size = prompt_mels_for_flow.shape[0]
+        flow_inputs = []
+        flow_inputs_lens = []
+        for prompt_speech_tokens, generated_speech_tokens in zip(prompt_speech_tokens_list, generated_speech_tokens_list):
+            flow_inputs.append(torch.tensor(prompt_speech_tokens + generated_speech_tokens))
+            flow_inputs_lens.append(len(prompt_speech_tokens) + len(generated_speech_tokens))
+
+        flow_inputs = torch.nn.utils.rnn.pad_sequence(flow_inputs, batch_first=True, padding_value=0)
+        flow_inputs_lens = torch.tensor(flow_inputs_lens)
+
+        with torch.amp.autocast(self.device, dtype=torch.float16):
+            generated_mels, generated_mels_lens = self.flow.inference(
+                flow_inputs.to(self.device), flow_inputs_lens.to(self.device),
+                prompt_mels_for_flow.to(self.device), prompt_mels_lens_for_flow.to(self.device), spk_emb_for_flow.to(self.device), 10
+            )
+
+        return generated_mels, generated_mels_lens
+
+    def forward_hift(self, generated_mels: torch.Tensor, generated_mels_lens: torch.Tensor, prompt_mels_lens_for_flow: torch.Tensor):
+        batch_size = generated_mels.shape[0]
+        generated_wavs = []
+        for i in range(batch_size):
+            mel = generated_mels[i, :, prompt_mels_lens_for_flow[i].item():generated_mels_lens[i].item()].unsqueeze(0)
+            wav, _ = self.hift(speech_feat=mel)
+            generated_wavs.append(wav)
+        return generated_wavs
+
+    @torch.inference_mode()
+    def forward(
+        self, generated_speech_tokens_list: list[list[int]], prompt_audios_list: list[torch.Tensor], prompt_audios_sample_rate: list[int]
+    ):
+        assert all(sample_rate == 16000 for sample_rate in prompt_audios_sample_rate)
+
+        prompt_speech_tokens_list, prompt_mels_for_flow, prompt_mels_lens_for_flow, spk_emb_for_flow = self.prepare_prompt_audio(prompt_audios_list, prompt_audios_sample_rate)
+
+        generated_mels, generated_mels_lens = self.forward_flow(
+            prompt_speech_tokens_list, generated_speech_tokens_list,
+            prompt_mels_for_flow, prompt_mels_lens_for_flow, spk_emb_for_flow
+        )
+
+        generated_wavs = self.forward_hift(generated_mels, generated_mels_lens, prompt_mels_lens_for_flow)
+        return generated_wavs
+
+    def prepare_prompt_audio(
+        self, prompt_audios_list: list[torch.Tensor], prompt_audios_sample_rate: list[int]
+    ):
+        assert all(sample_rate == 16000 for sample_rate in prompt_audios_sample_rate)
+
+        prompt_speech_tokens_list = self.prompt_audio_tokenization(prompt_audios_list)
+
+        prompt_mels_for_flow, prompt_mels_lens_for_flow = self.get_prompt_mels(prompt_audios_list, prompt_audios_sample_rate)
+
+        spk_emb_for_flow = self.get_spk_emb(prompt_audios_list)
+        return prompt_speech_tokens_list, prompt_mels_for_flow, prompt_mels_lens_for_flow, spk_emb_for_flow
+
+    def get_prompt_audio_cache_for_streaming_tts(
+        self, prompt_speech_tokens_list, prompt_mels_for_flow, prompt_mels_lens_for_flow, spk_emb_for_flow
+    ):
+        assert len(prompt_speech_tokens_list) == 1, "only support batch size 1 for streaming tts"
+        for i, prompt_speech_tokens in enumerate(prompt_speech_tokens_list):
+            prompt_speech_tokens_list[i] = torch.tensor(prompt_speech_tokens + prompt_speech_tokens_list[i][:3])
+        prompt_speech_tokens_tensor = torch.nn.utils.rnn.pad_sequence(prompt_speech_tokens_list, batch_first=True, padding_value=0)
+
+        cache = self.flow.setup_cache(
+            prompt_speech_tokens_tensor.to(self.device),
+            prompt_mels_for_flow.to(self.device),
+            spk_emb_for_flow.to(self.device),
+            n_timesteps=10
+        )
+        new_cache = {k: v.clone() for k, v in cache.items()}
+        # Hack: this is a hack to avoid in-place changes to the cache['estimator_att_cache'] and cache['estimator_cnn_cache']
+        return new_cache
+
+    @torch.inference_mode()
+    def forward_streaming(
+        self, generated_speech_tokens: list[int], last_chunk: bool, request_id: str, speaker_id: str, prompt_audio: torch.Tensor = None, prompt_audio_sample_rate: int = 16000
+    ):
+        if speaker_id not in self.speaker_cache:
+            assert prompt_audio is not None, "prompt_audio is required for new speaker"
+            assert prompt_audio_sample_rate == 16000
+
+            prompt_speech_tokens_list, prompt_mels_for_flow, prompt_mels_lens_for_flow, spk_emb_for_flow = self.prepare_prompt_audio([prompt_audio], [prompt_audio_sample_rate])
+
+            token_len = min(int(prompt_mels_for_flow.shape[1] / 2), len(prompt_speech_tokens_list[0]))
+            prompt_mels_for_flow = prompt_mels_for_flow[:, :2 * token_len].contiguous()
+            prompt_speech_tokens_list[0] = prompt_speech_tokens_list[0][:token_len]
+
+            prompt_audio_dict = {'spk_emb_for_flow': spk_emb_for_flow, 'prompt_mels_for_flow': prompt_mels_for_flow}
+
+            cache_dict = self.get_prompt_audio_cache_for_streaming_tts(prompt_speech_tokens_list, prompt_mels_for_flow, prompt_mels_lens_for_flow, spk_emb_for_flow)
+            self.speaker_cache[speaker_id] = {'prompt_audio_dict': prompt_audio_dict, 'cache_dict': cache_dict}
+
+        if request_id not in self.streaming_flow_cache:
+            self.streaming_flow_cache[request_id] = {k: v.clone() for k, v in self.speaker_cache[speaker_id]['cache_dict'].items()}
+            self.hift_cache_dict[request_id] = dict(
+                mel=torch.zeros(1, 80, 0, device='cuda'),
+                source=torch.zeros(1, 1, 0, device='cuda'),
+                speech=torch.zeros(1, 0, device='cuda'),
+            )
+
+        current_request_cache = self.streaming_flow_cache[request_id]
+
+        current_prompt_audio_dict = self.speaker_cache[speaker_id]['prompt_audio_dict']
+        generated_speech_tokens = torch.tensor([generated_speech_tokens], dtype=torch.int32, device='cuda')
+
+        chunk_mel, new_streaming_flow_cache = self.flow.inference_chunk(
+            token=generated_speech_tokens,
+            spk=current_prompt_audio_dict['spk_emb_for_flow'].to(self.device),
+            cache=current_request_cache,
+            last_chunk=last_chunk,
+            n_timesteps=10,
+        )
+
+        self.streaming_flow_cache[request_id] = new_streaming_flow_cache
+
+        if self.streaming_flow_cache[request_id]['estimator_att_cache'].shape[4] > (current_prompt_audio_dict['prompt_mels_for_flow'].shape[1] + 100):
+            self.streaming_flow_cache[request_id]['estimator_att_cache'] = torch.cat([
+                self.streaming_flow_cache[request_id]['estimator_att_cache'][:, :, :, :, :current_prompt_audio_dict['prompt_mels_for_flow'].shape[1]],
+                self.streaming_flow_cache[request_id]['estimator_att_cache'][:, :, :, :, -100:],
+            ], dim=4)
+
+        hift_cache_mel = self.hift_cache_dict[request_id]['mel'].clone()
+        hift_cache_source = self.hift_cache_dict[request_id]['source'].clone()
+        hift_cache_speech = self.hift_cache_dict[request_id]['speech'].clone()
+        mel = torch.concat([hift_cache_mel, chunk_mel], dim=2).clone()
+
+        speech, source = self.hift(mel, hift_cache_source)
+
+        # overlap speech smooth
+        if hift_cache_speech.shape[-1] > 0:
+            speech = fade_in_out(speech, hift_cache_speech, self.speech_window)
+
+        # update vocoder cache
+        self.hift_cache_dict[request_id] = dict(
+            mel=mel[..., -self.mel_cache_len:].clone().detach(),
+            source=source[:, :, -self.source_cache_len:].clone().detach(),
+            speech=speech[:, -self.source_cache_len:].clone().detach(),
+        )
+        if not last_chunk:
+            speech = speech[:, :-self.source_cache_len]
+
+        if last_chunk:
+            assert request_id in self.streaming_flow_cache
+            self.streaming_flow_cache.pop(request_id)
+            self.hift_cache_dict.pop(request_id)
+
+        return speech
+
+
+def collate_fn(batch):
+    ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate = [], [], [], []
+    for item in batch:
+        generated_speech_tokens_list.append(item['target_audio_cosy2_tokens'])
+        audio = torch.from_numpy(item['prompt_audio']['array']).float()
+        prompt_audios_list.append(audio)
+        prompt_audios_sample_rate.append(item['prompt_audio']['sampling_rate'])
+        ids.append(item['id'])
+
+    return ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate
+
+
+def get_args():
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--enable-trt", action="store_true")
+    parser.add_argument("--model-dir", type=str, default="./Step-Audio-2-mini/token2wav")
+    parser.add_argument("--batch-size", type=int, default=1)
+    parser.add_argument("--output-dir", type=str, default="generated_wavs")
+    parser.add_argument("--huggingface-dataset-split", type=str, default="wenetspeech4tts")
+    parser.add_argument("--warmup", type=int, default=3, help="Number of warmup epochs, performance statistics will only be collected from the last epoch")
+    return parser.parse_args()
+
+
+if __name__ == "__main__":
+    args = get_args()
+    model = CosyVoice2_Token2Wav(model_dir=args.model_dir, enable_trt=args.enable_trt)
+    if not os.path.exists(args.output_dir):
+        os.makedirs(args.output_dir)
+    dataset_name = "yuekai/seed_tts_cosy2"
+
+    dataset = load_dataset(dataset_name, split=args.huggingface_dataset_split, trust_remote_code=True)
+
+    data_loader = DataLoader(dataset, batch_size=args.batch_size, shuffle=False, collate_fn=collate_fn, num_workers=0)
+
+    for _ in range(args.warmup):
+        start_time = time.time()
+        for batch in data_loader:
+            ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate = batch
+
+            generated_wavs = model(generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate)
+
+            for id, wav in zip(ids, generated_wavs):
+                torchaudio.save(f"{args.output_dir}/{id}.wav", wav.cpu(), 24000)
+        end_time = time.time()
+        epoch_time = end_time - start_time
+        print(f"Measurement epoch time taken: {epoch_time:.4f} seconds")

runtime/triton_trtllm/model_repo/token2wav_dit/config.pbtxt

@@ -0,0 +1,69 @@
+# Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+#
+# 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.
+
+name: "token2wav_dit"
+backend: "python"
+max_batch_size: ${triton_max_batch_size}
+
+dynamic_batching {
+    max_queue_delay_microseconds: ${max_queue_delay_microseconds}
+    priority_levels: 10
+    default_priority_level: 10
+}
+
+parameters [
+  {
+   key: "model_dir",
+   value: {string_value:"${model_dir}"}
+  }
+]
+
+input [
+  {
+    name: "target_speech_tokens"
+    data_type: TYPE_INT32
+    dims: [-1]
+  },
+  {
+    name: "reference_wav"
+    data_type: TYPE_FP32
+    dims: [-1]
+  },
+  {
+    name: "reference_wav_len"
+    data_type: TYPE_INT32
+    dims: [1]
+  },
+  {
+    name: "finalize"
+    data_type: TYPE_BOOL
+    dims: [ 1 ]
+    reshape: { shape: [ ] }
+    optional: true
+  }
+]
+output [
+  {
+    name: "waveform"
+    data_type: TYPE_FP32
+    dims: [ -1 ]
+  }
+]
+
+instance_group [
+  {
+    count: 1
+    kind: KIND_CPU
+  }
+]

runtime/triton_trtllm/offline_inference.py

@@ -0,0 +1,652 @@
+# SPDX-FileCopyrightText: Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+""" Example Usage
+    CUDA_VISIBLE_DEVICES=0 \
+        python3 offline_inference.py \
+            --output-dir $output_dir \
+            --llm-model-name-or-path $huggingface_model_local_dir \
+            --token2wav-path $model_scope_model_local_dir \
+            --backend $backend \
+            --batch-size $batch_size --token2wav-batch-size $token2wav_batch_size \
+            --engine-dir $trt_engines_dir \
+            --split-name ${dataset} || exit 1
+"""
+
+import argparse
+import json
+import os
+import sys
+
+import torch
+import torch.distributed as dist
+import torch.nn.functional as F
+import torchaudio
+from cosyvoice.utils.file_utils import load_wav
+from datasets import load_dataset
+from transformers import AutoTokenizer
+from torch.utils.data import DataLoader, Dataset
+from tqdm import tqdm
+import soundfile as sf
+import s3tokenizer
+from functools import partial
+import time
+import requests
+import asyncio
+import httpx
+
+sys.path.append("/workspace/CosyVoice/third_party/Matcha-TTS")
+try:
+    torch.multiprocessing.set_start_method("spawn")
+except RuntimeError:
+    pass
+
+
+async def send_request_async(client, url, payload):
+    response = await client.post(url, json=payload, timeout=None)
+    response.raise_for_status()
+    response_json = response.json()
+    return response_json['choices'][0]['message']['content']
+
+
+async def send_batch_requests_async(api_base, model_name, chats, temperature, top_p, top_k):
+    async with httpx.AsyncClient() as client:
+        tasks = []
+        for chat in chats:
+            payload = {
+                "model": model_name,
+                "messages": chat,
+                "max_tokens": 2048,
+                "temperature": temperature,
+                "top_p": top_p,
+                "top_k": top_k,
+                "repetition_penalty": 1.1,
+                "stop": ["<|eos1|>", "<|eos|>"],
+                "stream": False,
+            }
+            tasks.append(send_request_async(client, api_base, payload))
+        return await asyncio.gather(*tasks)
+
+
+def extract_speech_ids(speech_tokens_str):
+    """Extract speech IDs from token strings like <|s_23456|>"""
+    speech_ids = []
+    for token_str in speech_tokens_str:
+        if token_str.startswith('<|s_') and token_str.endswith('|>'):
+            num_str = token_str[4:-2]
+            num = int(num_str)
+            speech_ids.append(num)
+        else:
+            print(f"Unexpected token: {token_str}")
+    return speech_ids
+
+
+def convert_cosy2_tokens_to_speech_id_str(cosy2_tokens):
+    """Convert CosyVoice2 tokens to speech IDs string like <|s_23456|>"""
+    speech_id_str = ""
+    for token in cosy2_tokens:
+        speech_id_str += f"<|s_{token}|>"
+    return speech_id_str
+
+
+def get_args():
+    parser = argparse.ArgumentParser(description="Speech generation using LLM + CosyVoice2")
+    parser.add_argument(
+        "--split-name",
+        type=str,
+        default="wenetspeech4tts",
+        help="huggingface dataset split name, see yuekai/CV3-Eval, yuekai/seed_tts_cosy2",
+    )
+    parser.add_argument(
+        "--output-dir", required=True, type=str, help="dir to save result"
+    )
+    parser.add_argument(
+        "--batch-size",
+        default=1,
+        type=int,
+        help="batch size (per-device) for inference",
+    )
+    parser.add_argument(
+        "--token2wav-batch-size",
+        default=1,
+        type=int,
+        help="batch size (per-device) for inference",
+    )
+    parser.add_argument(
+        "--num-workers", type=int, default=0, help="workers for dataloader"
+    )
+    parser.add_argument(
+        "--prefetch", type=int, default=None, help="prefetch for dataloader"
+    )
+    parser.add_argument(
+        "--llm-model-name-or-path",
+        required=True,
+        type=str,
+        help="LLM model path (includes both model and tokenizer)",
+    )
+    parser.add_argument(
+        "--token2wav-path",
+        required=True,
+        type=str,
+        help="CosyVoice2 token2wav model path",
+    )
+    parser.add_argument(
+        "--prompt-text",
+        type=str,
+        default=None,
+        help="The prompt text for CosyVoice2",
+    )
+    parser.add_argument(
+        "--prompt-speech-path",
+        type=str,
+        default=None,
+        help="The path to the prompt speech for CosyVoice2",
+    )
+    parser.add_argument(
+        "--top-p",
+        type=float,
+        default=0.95,
+        help="top p for sampling",
+    )
+    parser.add_argument(
+        "--temperature",
+        type=float,
+        default=0.8,
+        help="temperature for sampling",
+    )
+    parser.add_argument(
+        "--top-k",
+        type=int,
+        default=50,
+        help="top k for sampling",
+    )
+    parser.add_argument(
+        "--backend",
+        type=str,
+        default="hf",
+        choices=["hf", "trtllm", "vllm", "trtllm-serve"],
+        help="Backend to use for LLM inference: 'hf' for HuggingFace, 'trtllm' for TensorRT-LLM, 'vllm' for VLLM",
+    )
+    parser.add_argument(
+        "--engine-dir",
+        type=str,
+        default=None,
+        help="TensorRT-LLM engine directory (required when backend is 'trtllm')",
+    )
+    parser.add_argument(
+        "--kv-cache-free-gpu-memory-fraction",
+        type=float,
+        default=0.6,
+        help="Fraction of GPU memory to free for KV cache (TensorRT-LLM only)",
+    )
+    parser.add_argument(
+        "--openai-api-base",
+        type=str,
+        default="http://localhost:8000/v1/chat/completions",
+        help="OpenAI API base URL (for trtllm-serve backend)",
+    )
+    parser.add_argument(
+        "--openai-model-name",
+        type=str,
+        default="trt_engines_bfloat16",
+        help="Model name to use with OpenAI API (for trtllm-serve backend)",
+    )
+    args = parser.parse_args()
+    return args
+
+
+def data_collator(batch, tokenizer, s3_tokenizer):
+    """Simplified data collator for batch_size=1 processing"""
+    collator_start_time = time.time()
+    total_audio_processing_time = 0
+    total_speech_tokenization_time = 0
+    total_text_tokenization_time = 0
+
+    target_sample_rate = 16000  # CosyVoice2 uses 16kHz for prompt audio
+    device = s3_tokenizer.device if s3_tokenizer is not None else torch.device("cpu")
+    input_ids_list, prompt_audio_list, prompt_text_list = [], [], []
+    prompt_text_after_apply_template_list = []
+    mels, prompt_audio_cosy2tokens_list, full_text_list = [], [], []
+    chat_list = []
+    for _, item in enumerate(batch):
+        audio_processing_start_time = time.time()
+        prompt_text, target_text = (
+            item["prompt_text"],
+            item["target_text"],
+        )
+        prompt_text_list.append(prompt_text)
+        full_text = prompt_text + target_text
+        full_text_list.append(full_text)
+        # remove the unnecessary punctuation for cosyvoice3 zero_shot_zh dataset
+        puncts = ['"', '(', ')', '“', '”', '‘', '（', '）', '\'']
+        for p in puncts:
+            if p in full_text:
+                full_text = full_text.replace(p, '')
+                print(f"removed {p} from {full_text}")
+
+        # get prompt audio for CosyVoice2 (convert to 16kHz)
+        ref_audio_org, ref_sr = (
+            item["prompt_audio"]["array"],
+            item["prompt_audio"]["sampling_rate"],
+        )
+        ref_audio_org = torch.from_numpy(ref_audio_org).float().unsqueeze(0)
+        print(ref_audio_org.shape)
+
+        if ref_sr != target_sample_rate:
+            resampler = torchaudio.transforms.Resample(ref_sr, target_sample_rate)
+            ref_audio = resampler(ref_audio_org)
+        else:
+            ref_audio = ref_audio_org
+
+        prompt_audio_list.append(ref_audio)
+        audio_processing_end_time = time.time()
+        total_audio_processing_time += audio_processing_end_time - audio_processing_start_time
+
+        speech_tokenization_start_time = time.time()
+        if "prompt_audio_cosy2_tokens" in item:
+            prompt_audio_cosy2tokens = item["prompt_audio_cosy2_tokens"]
+            prompt_audio_cosy2tokens_list.append(prompt_audio_cosy2tokens)
+        else:
+            mels.append(s3tokenizer.log_mel_spectrogram(ref_audio.squeeze(0)))
+
+    if len(mels) > 0:
+        mels, mels_lens = s3tokenizer.padding(mels)
+        codes, codes_lens = s3_tokenizer.quantize(mels.to(device), mels_lens.to(device))
+        for i in range(len(codes)):
+            prompt_audio_cosy2tokens_list.append(codes[i, :codes_lens[i].item()])
+    speech_tokenization_end_time = time.time()
+    total_speech_tokenization_time += speech_tokenization_end_time - speech_tokenization_start_time
+
+    for i, prompt_audio_cosy2tokens in enumerate(prompt_audio_cosy2tokens_list):
+        text_tokenization_start_time = time.time()
+        prompt_audio_cosy2_id_str = convert_cosy2_tokens_to_speech_id_str(prompt_audio_cosy2tokens)
+        # Create chat template for LLM generation
+        chat = [
+            {"role": "user", "content": full_text_list[i]},
+            {"role": "assistant", "content": prompt_audio_cosy2_id_str}
+        ]
+        chat_list.append(chat)
+
+        assert 'system' not in tokenizer.chat_template, "system is not allowed in the chat template"
+
+        input_ids = tokenizer.apply_chat_template(
+            chat,
+            tokenize=True,
+            return_tensors='pt',
+            continue_final_message=True
+        )
+        input_ids_list.append(input_ids.squeeze(0))
+
+        prompt_text_after_apply_template = f"<|sos|>{full_text_list[i]}<|task_id|>{prompt_audio_cosy2_id_str}"
+
+        prompt_text_after_apply_template_list.append(prompt_text_after_apply_template)
+        text_tokenization_end_time = time.time()
+        total_text_tokenization_time += text_tokenization_end_time - text_tokenization_start_time
+
+    ids = [item["id"] for item in batch]
+
+    return {
+        "input_ids": input_ids_list,
+        "ids": ids,
+        "prompt_text": prompt_text_list,
+        "prompt_audio_list": prompt_audio_list,
+        "prompt_text_after_apply_template": prompt_text_after_apply_template_list,
+        "audio_processing_time": total_audio_processing_time,
+        "speech_tokenization_time": total_speech_tokenization_time,
+        "text_tokenization_time": total_text_tokenization_time,
+        "chat_list": chat_list
+    }
+
+
+def init_distributed():
+    world_size = int(os.environ.get("WORLD_SIZE", 1))
+    local_rank = int(os.environ.get("LOCAL_RANK", 0))
+    rank = int(os.environ.get("RANK", 0))
+    print(
+        "Inference on multiple gpus, this gpu {}".format(local_rank)
+        + ", rank {}, world_size {}".format(rank, world_size)
+    )
+    torch.cuda.set_device(local_rank)
+    dist.init_process_group("nccl")
+    return world_size, local_rank, rank
+
+
+def main(args):
+    os.makedirs(args.output_dir, exist_ok=True)
+
+    assert torch.cuda.is_available()
+    local_rank, world_size, rank = 0, 1, 0
+    device = torch.device(f"cuda:{local_rank}")
+
+    tokenizer = AutoTokenizer.from_pretrained(args.llm_model_name_or_path)
+
+    if args.backend == "hf":
+        model = AutoModelForCausalLM.from_pretrained(args.llm_model_name_or_path)
+        model.eval()
+        model.to(device)
+        runner = None
+    elif args.backend == "trtllm":
+        if args.engine_dir is None:
+            raise ValueError("--engine-dir is required when backend is 'trtllm'")
+
+        runtime_rank = tensorrt_llm.mpi_rank()
+        model = None
+
+        runner_kwargs = dict(
+            engine_dir=args.engine_dir,
+            rank=runtime_rank,
+            max_output_len=2048,
+            enable_context_fmha_fp32_acc=False,
+            max_batch_size=args.batch_size,
+            max_input_len=512,
+            kv_cache_free_gpu_memory_fraction=args.kv_cache_free_gpu_memory_fraction,
+            cuda_graph_mode=False,
+            gather_generation_logits=False,
+        )
+
+        runner = ModelRunnerCpp.from_dir(**runner_kwargs)
+    elif args.backend == "vllm":
+        model = LLM(model=args.llm_model_name_or_path, gpu_memory_utilization=0.4)
+        runner = None
+    elif args.backend == "trtllm-serve":
+        model = None
+        runner = None
+    else:
+        raise ValueError(f"Unsupported backend: {args.backend}")
+    if 'Step-Audio-2-mini' in args.token2wav_path:
+        from token2wav_dit import CosyVoice2_Token2Wav
+    else:
+        assert 'CosyVoice2-0.5B' in args.token2wav_path
+        from token2wav import CosyVoice2_Token2Wav
+    token2wav_model = CosyVoice2_Token2Wav(
+        model_dir=args.token2wav_path, enable_trt=True, device_id=local_rank
+    )
+    if args.prompt_speech_path:
+        prompt_speech_16k = load_wav(args.prompt_speech_path, 16000)
+    else:
+        prompt_speech_16k = None
+    s3_tokenizer = s3tokenizer.load_model(f"{args.token2wav_path}/speech_tokenizer_v2.onnx").to(device) if 'zero' in args.split_name else None
+    dataset_name = "yuekai/CV3-Eval" if 'zero' in args.split_name else "yuekai/seed_tts_cosy2"
+    dataset = load_dataset(
+        dataset_name,
+        split=args.split_name,
+        trust_remote_code=True,
+    )
+
+    sampler = None
+    dataloader = DataLoader(
+        dataset,
+        batch_size=args.batch_size,
+        sampler=sampler,
+        shuffle=False,
+        num_workers=args.num_workers,
+        prefetch_factor=args.prefetch,
+        collate_fn=partial(data_collator, tokenizer=tokenizer, s3_tokenizer=s3_tokenizer),
+    )
+    for _ in range(3):
+        print(f"Running {_} times")
+        total_llm_time = 0
+        total_token2wav_time = 0
+        total_data_load_time = 0
+        total_llm_post_processing_time = 0
+        total_audio_save_time = 0
+        total_audio_processing_time_in_collator = 0
+        total_speech_tokenization_time_in_collator = 0
+        total_text_tokenization_time_in_collator = 0
+        total_audio_samples = 0
+        start_time = time.time()
+        total_steps = len(dataset)
+
+        if rank == 0:
+            progress_bar = tqdm(total=total_steps, desc="Processing", unit="wavs")
+
+        last_batch_end_time = time.time()
+        for batch in dataloader:
+            data_loaded_time = time.time()
+            total_data_load_time += data_loaded_time - last_batch_end_time
+            total_audio_processing_time_in_collator += batch["audio_processing_time"]
+            total_speech_tokenization_time_in_collator += batch["speech_tokenization_time"]
+            total_text_tokenization_time_in_collator += batch["text_tokenization_time"]
+            with torch.no_grad():
+                llm_start_time = time.time()
+                if args.backend == "hf":
+                    input_ids_list = batch["input_ids"]
+                    if len(input_ids_list) == 1:
+                        input_ids = input_ids_list[0].unsqueeze(0)
+                        attention_mask = torch.ones_like(input_ids)
+                    else:
+                        max_len = max([len(input_ids) for input_ids in input_ids_list])
+                        input_ids_list_new = [
+                            torch.cat([input_ids, torch.full((max_len - len(input_ids),), tokenizer.pad_token_id)])
+                            for input_ids in input_ids_list
+                        ]
+                        input_ids = torch.stack(input_ids_list_new)
+                        attention_mask = torch.zeros_like(input_ids)
+                        for i in range(len(input_ids_list)):
+                            attention_mask[i, :len(input_ids_list[i])] = 1
+
+                    input_ids = input_ids.to(device)
+
+                    outputs = model.generate(
+                        input_ids=input_ids.to(device),
+                        attention_mask=attention_mask.to(device),
+                        max_new_tokens=2048,
+                        do_sample=True,
+                        top_p=args.top_p,
+                        temperature=args.temperature,
+                        repetition_penalty=1.1,
+                        top_k=args.top_k,
+                    )
+                    torch.cuda.synchronize()
+                elif args.backend == "trtllm":
+                    batch_input_ids = list(batch["input_ids"])
+                    input_lengths = [x.size(0) for x in batch_input_ids]
+
+                    end_id = tokenizer.convert_tokens_to_ids("<|eos1|>") if "<|eos1|>" in tokenizer.get_vocab() else tokenizer.eos_token_id
+                    print(f"end_id: {end_id}, tokenizer.eos_token_id: {tokenizer.eos_token_id} ========================")
+                    outputs = runner.generate(
+                        batch_input_ids=batch_input_ids,
+                        max_new_tokens=2048,
+                        end_id=end_id,
+                        pad_id=end_id,
+                        temperature=args.temperature,
+                        top_k=args.top_k,
+                        top_p=args.top_p,
+                        repetition_penalty=1.1,
+                        num_return_sequences=1,
+                        streaming=False,
+                        output_sequence_lengths=True,
+                        output_generation_logits=False,
+                        return_dict=True,
+                        return_all_generated_tokens=False
+                    )
+                    torch.cuda.synchronize()
+                    output_ids, sequence_lengths = outputs["output_ids"], outputs["sequence_lengths"]
+                    num_output_sents, num_beams, _ = output_ids.size()
+                    assert num_beams == 1
+                    beam = 0
+                    batch_size = len(batch["input_ids"])
+                    num_return_sequences = num_output_sents // batch_size
+                    assert num_return_sequences == 1
+                    outputs = []
+                    for i in range(batch_size * num_return_sequences):
+                        batch_idx = i // num_return_sequences
+                        seq_idx = i % num_return_sequences
+                        output_begin = input_lengths[batch_idx]
+                        output_end = sequence_lengths[i][beam]
+                        outputs_i = output_ids[i][beam][:output_end].tolist()
+                        outputs.append(outputs_i)
+                elif args.backend == "vllm":
+                    input_ids_list = [ids.tolist() for ids in batch["input_ids"]]
+                    sampling_params = SamplingParams(
+                        temperature=args.temperature,
+                        top_p=args.top_p,
+                        top_k=args.top_k,
+                        repetition_penalty=1.1,
+                        max_tokens=2048,
+                    )
+                    outputs = model.generate(prompt_token_ids=input_ids_list, sampling_params=sampling_params)
+                    print(outputs)
+                    for j, output in enumerate(outputs):
+                        outputs[j] = input_ids_list[j] + output.outputs[0].token_ids
+                elif args.backend == "trtllm-serve":
+                    if args.batch_size > 1:
+                        outputs = asyncio.run(send_batch_requests_async(
+                            args.openai_api_base,
+                            args.openai_model_name,
+                            batch["chat_list"],
+                            args.temperature,
+                            args.top_p,
+                            args.top_k,
+                        ))
+                    else:
+                        outputs = []
+                        for chat in batch["chat_list"]:
+                            payload = {
+                                "model": args.openai_model_name,
+                                "messages": chat,
+                                "max_tokens": 2048,
+                                "temperature": args.temperature,
+                                "top_p": args.top_p,
+                                "top_k": args.top_k,
+                                "repetition_penalty": 1.1,
+                                "stop": ["<|eos1|>", "<|eos|>"],
+                                "stream": False,
+                            }
+                            response = requests.post(args.openai_api_base, json=payload)
+                            response.raise_for_status()
+                            response_json = response.json()
+                            generated_content = response_json['choices'][0]['message']['content']
+                            outputs.append(generated_content)
+
+                llm_end_time = time.time()
+                total_llm_time += (llm_end_time - llm_start_time)
+
+                items_for_token_2wav = []
+                for i in range(len(batch["ids"])):
+                    llm_post_processing_start_time = time.time()
+                    if args.backend == "trtllm-serve":
+                        speech_tokens_str = outputs[i].strip().split('><')
+                        if len(speech_tokens_str) > 1:
+                            speech_tokens_str = [
+                                t if t.startswith('<') else '<' + t for t in speech_tokens_str
+                            ]
+                            speech_tokens_str = [
+                                t if t.endswith('>') else t + '>' for t in speech_tokens_str
+                            ]
+                        speech_ids = extract_speech_ids(speech_tokens_str)
+                    else:
+                        input_length = len(batch["input_ids"][i])
+                        generated_ids = outputs[i][input_length:]
+                        speech_tokens_str = tokenizer.batch_decode(generated_ids, skip_special_tokens=True)
+                        speech_ids = extract_speech_ids(speech_tokens_str)
+                    print(i, speech_ids)
+                    if len(speech_ids) == 0:
+                        print(f"Warning: No speech tokens generated for sample {batch['ids'][i]}, skipping")
+                        continue
+
+                    if args.prompt_text is not None:
+                        current_prompt_text = args.prompt_text
+                        current_prompt_audio = prompt_speech_16k
+                    else:
+                        current_prompt_text = batch["prompt_text"][i]
+                        current_prompt_audio = batch["prompt_audio_list"][i]
+
+                    llm_post_processing_end_time = time.time()
+                    total_llm_post_processing_time += llm_post_processing_end_time - llm_post_processing_start_time
+                    if current_prompt_audio is not None:
+                        items_for_token_2wav.append({
+                            "speech_ids": speech_ids,
+                            "prompt_audio": current_prompt_audio.squeeze(0),
+                            "id": batch["ids"][i]
+                        })
+                    else:
+                        print(f"Warning: No prompt audio available for sample {batch['ids'][i]}, skipping")
+
+                for i in range(0, len(items_for_token_2wav), args.token2wav_batch_size):
+                    t2w_batch = items_for_token_2wav[i:i + args.token2wav_batch_size]
+                    if not t2w_batch:
+                        continue
+
+                    t2w_generated_speech_tokens_list = [item["speech_ids"] for item in t2w_batch]
+                    t2w_prompt_audios_list = [item["prompt_audio"] for item in t2w_batch]
+                    t2w_prompt_audios_sample_rate = [16000] * len(t2w_batch)
+                    t2w_ids = [item["id"] for item in t2w_batch]
+
+                    token2wav_start_time = time.time()
+                    generated_wavs = token2wav_model(
+                        t2w_generated_speech_tokens_list,
+                        t2w_prompt_audios_list,
+                        t2w_prompt_audios_sample_rate,
+                    )
+                    token2wav_end_time = time.time()
+                    total_token2wav_time += (token2wav_end_time - token2wav_start_time)
+
+                    audio_save_start_time = time.time()
+                    for j, audio_hat in enumerate(generated_wavs):
+                        generated_wave = audio_hat.squeeze().cpu().numpy()
+                        total_audio_samples += len(generated_wave)
+                        target_sample_rate = 24000
+
+                        utt = t2w_ids[j]
+                        sf.write(f"{args.output_dir}/{utt}.wav", generated_wave, target_sample_rate)
+                        print(f"Generated audio for sample {utt} with {len(t2w_generated_speech_tokens_list[j])} tokens")
+                    audio_save_end_time = time.time()
+                    total_audio_save_time += audio_save_end_time - audio_save_start_time
+
+            if rank == 0:
+                progress_bar.update(world_size * len(batch["ids"]))
+
+            last_batch_end_time = time.time()
+        if rank == 0:
+            progress_bar.close()
+            end_time = time.time()
+            target_sample_rate = 24000
+            total_audio_duration_seconds = total_audio_samples / target_sample_rate
+
+            log_file_path = os.path.join(args.output_dir, "log.txt")
+            with open(log_file_path, 'w') as f:
+                args_dict = vars(args)
+                log_data = {
+                    "args": args_dict,
+                    "data_load_time_seconds": total_data_load_time,
+                    "audio_processing_time_in_collator_seconds": total_audio_processing_time_in_collator,
+                    "speech_tokenization_time_in_collator_seconds": total_speech_tokenization_time_in_collator,
+                    "text_tokenization_time_in_collator_seconds": total_text_tokenization_time_in_collator,
+                    "llm_time_seconds": total_llm_time,
+                    "llm_post_processing_time_seconds": total_llm_post_processing_time,
+                    "token2wav_time_seconds": total_token2wav_time,
+                    "audio_save_time_seconds": total_audio_save_time,
+                    "total_audio_duration_seconds": total_audio_duration_seconds,
+                    "pipeline_time_seconds": end_time - start_time,
+                }
+                print(log_data)
+                f.write(json.dumps(log_data, indent=4))
+            print(f"Metrics logged to {log_file_path}")
+
+
+if __name__ == "__main__":
+    args = get_args()
+    if args.backend == "vllm":
+        from vllm import LLM, SamplingParams
+    elif args.backend == "trtllm":
+        import tensorrt_llm
+        from tensorrt_llm.runtime import ModelRunnerCpp
+    elif args.backend == "hf":
+        from transformers import AutoModelForCausalLM
+    elif args.backend == "trtllm-serve":
+        pass
+    else:
+        raise ValueError(f"Unsupported backend: {args.backend}")
+    main(args)

runtime/triton_trtllm/requirements.txt

@@ -0,0 +1,14 @@
+hyperpyyaml
+s3tokenizer
+onnxruntime-gpu
+omegaconf
+conformer
+hydra-core
+lightning
+gdown
+wget
+librosa
+pyworld
+openai-whisper
+tritonclient
+modelscope

runtime/triton_trtllm/run.sh

@@ -0,0 +1,142 @@
+#!/bin/bash
+# Copyright (c) 2025 NVIDIA (authors: Yuekai Zhang)
+export CUDA_VISIBLE_DEVICES=0
+cosyvoice_path=/workspace/CosyVoice
+export PYTHONPATH=${cosyvoice_path}:$PYTHONPATH
+export PYTHONPATH=${cosyvoice_path}/third_party/Matcha-TTS:$PYTHONPATH
+stage=$1
+stop_stage=$2
+
+huggingface_model_local_dir=./cosyvoice2_llm
+model_scope_model_local_dir=./CosyVoice2-0.5B
+trt_dtype=bfloat16
+trt_weights_dir=./trt_weights_${trt_dtype}
+trt_engines_dir=./trt_engines_${trt_dtype}
+
+model_repo=./model_repo_cosyvoice2
+
+use_spk2info_cache=False
+
+if [ $stage -le -1 ] && [ $stop_stage -ge -1 ]; then
+    echo "Cloning CosyVoice"
+    git clone --recursive https://github.com/FunAudioLLM/CosyVoice.git $cosyvoice_path
+    cd $cosyvoice_path
+    git submodule update --init --recursive
+    cd runtime/triton_trtllm
+fi
+
+if [ $stage -le 0 ] && [ $stop_stage -ge 0 ]; then
+    echo "Downloading CosyVoice2-0.5B"
+    # see https://github.com/nvidia-china-sae/mair-hub/blob/main/rl-tutorial/cosyvoice_llm/pretrained_to_huggingface.py
+    huggingface-cli download --local-dir $huggingface_model_local_dir yuekai/cosyvoice2_llm
+    modelscope download --model iic/CosyVoice2-0.5B --local_dir $model_scope_model_local_dir
+    # download spk2info.pt to directly use cached speech tokens, speech feats, and embeddings
+    wget https://raw.githubusercontent.com/qi-hua/async_cosyvoice/main/CosyVoice2-0.5B/spk2info.pt -O $model_scope_model_local_dir/spk2info.pt
+fi
+
+
+if [ $stage -le 1 ] && [ $stop_stage -ge 1 ]; then
+    echo "Converting checkpoint to TensorRT weights"
+    python3 scripts/convert_checkpoint.py --model_dir $huggingface_model_local_dir \
+                                --output_dir $trt_weights_dir \
+                                --dtype $trt_dtype || exit 1
+
+    echo "Building TensorRT engines"
+    trtllm-build --checkpoint_dir $trt_weights_dir \
+                --output_dir $trt_engines_dir \
+                --max_batch_size 16 \
+                --max_num_tokens 32768 \
+                --gemm_plugin $trt_dtype || exit 1
+
+    echo "Testing TensorRT engines"
+    python3 ./scripts/test_llm.py --input_text "你好，请问你叫什么？" \
+                    --tokenizer_dir $huggingface_model_local_dir \
+                    --top_k 50 --top_p 0.95 --temperature 0.8 \
+                    --engine_dir=$trt_engines_dir  || exit 1
+fi
+
+if [ $stage -le 2 ] && [ $stop_stage -ge 2 ]; then
+    echo "Creating model repository"
+    rm -rf $model_repo
+    mkdir -p $model_repo
+    cosyvoice2_dir="cosyvoice2"
+
+    cp -r ./model_repo/${cosyvoice2_dir} $model_repo
+    cp -r ./model_repo/tensorrt_llm $model_repo
+    cp -r ./model_repo/token2wav $model_repo
+    if [ $use_spk2info_cache == "False" ]; then
+        cp -r ./model_repo/audio_tokenizer $model_repo
+        cp -r ./model_repo/speaker_embedding $model_repo
+    fi
+
+    ENGINE_PATH=$trt_engines_dir
+    MAX_QUEUE_DELAY_MICROSECONDS=0
+    MODEL_DIR=$model_scope_model_local_dir
+    LLM_TOKENIZER_DIR=$huggingface_model_local_dir
+    BLS_INSTANCE_NUM=4
+    TRITON_MAX_BATCH_SIZE=16
+    DECOUPLED_MODE=True # True for streaming, False for offline
+
+    python3 scripts/fill_template.py -i ${model_repo}/token2wav/config.pbtxt model_dir:${MODEL_DIR},triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS}
+    python3 scripts/fill_template.py -i ${model_repo}/${cosyvoice2_dir}/config.pbtxt model_dir:${MODEL_DIR},bls_instance_num:${BLS_INSTANCE_NUM},llm_tokenizer_dir:${LLM_TOKENIZER_DIR},triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},decoupled_mode:${DECOUPLED_MODE},max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS}
+    python3 scripts/fill_template.py -i ${model_repo}/tensorrt_llm/config.pbtxt triton_backend:tensorrtllm,triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},decoupled_mode:${DECOUPLED_MODE},max_beam_width:1,engine_dir:${ENGINE_PATH},max_tokens_in_paged_kv_cache:2560,max_attention_window_size:2560,kv_cache_free_gpu_mem_fraction:0.5,exclude_input_in_output:True,enable_kv_cache_reuse:False,batching_strategy:inflight_fused_batching,max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS},encoder_input_features_data_type:TYPE_FP16,logits_datatype:TYPE_FP32
+    if [ $use_spk2info_cache == "False" ]; then
+        python3 scripts/fill_template.py -i ${model_repo}/audio_tokenizer/config.pbtxt model_dir:${MODEL_DIR},triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS}
+        python3 scripts/fill_template.py -i ${model_repo}/speaker_embedding/config.pbtxt model_dir:${MODEL_DIR},triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS}
+    fi
+fi
+
+if [ $stage -le 3 ] && [ $stop_stage -ge 3 ]; then
+   echo "Starting Triton server"
+   tritonserver --model-repository $model_repo
+fi
+
+if [ $stage -le 4 ] && [ $stop_stage -ge 4 ]; then
+    echo "Single request test http, only work for offline TTS mode"
+    python3 client_http.py \
+        --reference-audio ./assets/prompt_audio.wav \
+        --reference-text "吃燕窝就选燕之屋，本节目由26年专注高品质燕窝的燕之屋冠名播出。豆奶牛奶换着喝，营养更均衡，本节目由豆本豆豆奶特约播出。" \
+        --target-text "身临其境，换新体验。塑造开源语音合成新范式，让智能语音更自然。" \
+        --model-name cosyvoice2
+fi
+
+if [ $stage -le 5 ] && [ $stop_stage -ge 5 ]; then
+    echo "Running benchmark client grpc"
+    num_task=4
+
+    mode=streaming
+    BLS_INSTANCE_NUM=4
+
+    python3 client_grpc.py \
+        --server-addr localhost \
+        --model-name cosyvoice2 \
+        --num-tasks $num_task \
+        --mode $mode \
+        --use-spk2info-cache $use_spk2info_cache \
+        --huggingface-dataset yuekai/seed_tts_cosy2 \
+        --log-dir ./log_concurrent_tasks_${num_task}_${mode}_bls_${BLS_INSTANCE_NUM}_spk_cache_${use_spk2info_cache}
+fi
+
+if [ $stage -le 6 ] && [ $stop_stage -ge 6 ]; then
+  echo "stage 6: Offline inference benchmark"
+  n_gpus=1
+  datasets=(wenetspeech4tts) # wenetspeech4tts, test_zh, zero_shot_zh
+  backend=trtllm # hf, trtllm, vllm
+
+  batch_sizes=(16 8 4 2 1)
+  token2wav_batch_size=1
+  for batch_size in ${batch_sizes[@]}; do
+    for dataset in ${datasets[@]}; do
+    output_dir=./${dataset}_${backend}_llm_batch_size_${batch_size}_token2wav_batch_size_${token2wav_batch_size}
+    CUDA_VISIBLE_DEVICES=0 \
+        python3 offline_inference.py \
+            --output-dir $output_dir \
+            --llm-model-name-or-path $huggingface_model_local_dir \
+            --token2wav-path $model_scope_model_local_dir \
+            --backend $backend \
+            --batch-size $batch_size --token2wav-batch-size $token2wav_batch_size \
+            --engine-dir $trt_engines_dir \
+            --split-name ${dataset} || exit 1
+    done
+  done
+fi

runtime/triton_trtllm/run_stepaudio2_dit_token2wav.sh

@@ -0,0 +1,225 @@
+#!/bin/bash
+# Copyright (c) 2025 NVIDIA (authors: Yuekai Zhang)
+export CUDA_VISIBLE_DEVICES=0
+cosyvoice_path=/workspace/CosyVoice
+stepaudio2_path=/workspace/Step-Audio2
+
+export PYTHONPATH=${stepaudio2_path}:$PYTHONPATH
+export PYTHONPATH=${cosyvoice_path}:$PYTHONPATH
+export PYTHONPATH=${cosyvoice_path}/third_party/Matcha-TTS:$PYTHONPATH
+
+stage=$1
+stop_stage=$2
+
+huggingface_model_local_dir=./cosyvoice2_llm
+model_scope_model_local_dir=./CosyVoice2-0.5B
+step_audio_model_dir=./Step-Audio-2-mini
+
+trt_dtype=bfloat16
+trt_weights_dir=./trt_weights_${trt_dtype}
+trt_engines_dir=./trt_engines_${trt_dtype}
+
+model_repo=./model_repo_cosyvoice2_dit
+bls_instance_num=10
+
+if [ $stage -le -1 ] && [ $stop_stage -ge -1 ]; then
+
+    echo "Cloning Step-Audio2-mini"
+    git clone https://github.com/yuekaizhang/Step-Audio2.git -b trt $stepaudio2_path
+
+    echo "Cloning CosyVoice"
+    git clone --recursive https://github.com/FunAudioLLM/CosyVoice.git $cosyvoice_path
+    cd $cosyvoice_path
+    git submodule update --init --recursive
+    cd runtime/triton_trtllm
+fi
+
+if [ $stage -le 0 ] && [ $stop_stage -ge 0 ]; then
+    echo "Downloading CosyVoice2-0.5B"
+    # see https://github.com/nvidia-china-sae/mair-hub/blob/main/rl-tutorial/cosyvoice_llm/pretrained_to_huggingface.py
+    huggingface-cli download --local-dir $huggingface_model_local_dir yuekai/cosyvoice2_llm
+    modelscope download --model iic/CosyVoice2-0.5B --local_dir $model_scope_model_local_dir
+
+    echo "Step-Audio2-mini"
+    huggingface-cli download --local-dir $step_audio_model_dir stepfun-ai/Step-Audio-2-mini
+    cd $step_audio_model_dir/token2wav
+    wget https://huggingface.co/yuekai/cosyvoice2_dit_flow_matching_onnx/resolve/main/flow.decoder.estimator.fp32.dynamic_batch.onnx -O flow.decoder.estimator.fp32.dynamic_batch.onnx
+    wget https://huggingface.co/yuekai/cosyvoice2_dit_flow_matching_onnx/resolve/main/flow.decoder.estimator.chunk.fp32.dynamic_batch.simplify.onnx -O flow.decoder.estimator.chunk.fp32.dynamic_batch.simplify.onnx
+    cd -
+fi
+
+
+if [ $stage -le 1 ] && [ $stop_stage -ge 1 ]; then
+    echo "Converting checkpoint to TensorRT weights"
+    python3 scripts/convert_checkpoint.py --model_dir $huggingface_model_local_dir \
+                                --output_dir $trt_weights_dir \
+                                --dtype $trt_dtype || exit 1
+
+    echo "Building TensorRT engines"
+    trtllm-build --checkpoint_dir $trt_weights_dir \
+                --output_dir $trt_engines_dir \
+                --max_batch_size 64 \
+                --max_num_tokens 32768 \
+                --gemm_plugin $trt_dtype || exit 1
+
+    echo "Testing TensorRT engines"
+    python3 ./scripts/test_llm.py --input_text "你好，请问你叫什么？" \
+                    --tokenizer_dir $huggingface_model_local_dir \
+                    --top_k 50 --top_p 0.95 --temperature 0.8 \
+                    --engine_dir=$trt_engines_dir  || exit 1
+fi
+
+if [ $stage -le 2 ] && [ $stop_stage -ge 2 ]; then
+    echo "Creating model repository async mode"
+    rm -rf $model_repo
+    mkdir -p $model_repo
+    cosyvoice2_dir="cosyvoice2_dit"
+    token2wav_dir="token2wav_dit"
+
+    cp -r ./model_repo/${cosyvoice2_dir} $model_repo
+    cp -r ./model_repo/${token2wav_dir} $model_repo
+    cp -r ./model_repo/audio_tokenizer $model_repo
+    cp -r ./model_repo/speaker_embedding $model_repo
+
+
+    ENGINE_PATH=$trt_engines_dir
+    MAX_QUEUE_DELAY_MICROSECONDS=0
+    MODEL_DIR=$model_scope_model_local_dir
+    LLM_TOKENIZER_DIR=$huggingface_model_local_dir
+    BLS_INSTANCE_NUM=$bls_instance_num
+    TRITON_MAX_BATCH_SIZE=1
+    DECOUPLED_MODE=True # Only streaming TTS mode is supported using Nvidia Triton for now
+    STEP_AUDIO_MODEL_DIR=$step_audio_model_dir/token2wav
+
+    python3 scripts/fill_template.py -i ${model_repo}/${token2wav_dir}/config.pbtxt model_dir:${STEP_AUDIO_MODEL_DIR},triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS}
+    python3 scripts/fill_template.py -i ${model_repo}/${cosyvoice2_dir}/config.pbtxt model_dir:${MODEL_DIR},bls_instance_num:${BLS_INSTANCE_NUM},llm_tokenizer_dir:${LLM_TOKENIZER_DIR},triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},decoupled_mode:${DECOUPLED_MODE},max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS}
+    python3 scripts/fill_template.py -i ${model_repo}/audio_tokenizer/config.pbtxt model_dir:${MODEL_DIR},triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS}
+    python3 scripts/fill_template.py -i ${model_repo}/speaker_embedding/config.pbtxt model_dir:${MODEL_DIR},triton_max_batch_size:${TRITON_MAX_BATCH_SIZE},max_queue_delay_microseconds:${MAX_QUEUE_DELAY_MICROSECONDS}
+
+fi
+
+if [ $stage -le 3 ] && [ $stop_stage -ge 3 ]; then
+   echo "Starting Token2wav Triton server and Cosyvoice2 llm using trtllm-serve"
+   mpirun -np 1 --allow-run-as-root --oversubscribe trtllm-serve serve --tokenizer $huggingface_model_local_dir $trt_engines_dir --max_batch_size 64  --kv_cache_free_gpu_memory_fraction 0.4 &
+   tritonserver --model-repository $model_repo --http-port 18000 &
+   wait
+    # Test using curl
+    # curl http://localhost:8000/v1/chat/completions \
+    #     -H "Content-Type: application/json" \
+    #     -d '{
+    #         "model": "",
+    #         "messages":[{"role": "user", "content": "Where is New York?"},
+    #                     {"role": "assistant", "content": "<|s_1708|><|s_2050|><|s_2159|>"}],
+    #         "max_tokens": 512,
+    #         "temperature": 0.8,
+    #         "top_p": 0.95,
+    #         "top_k": 50,
+    #         "stop": ["<|eos1|>"],
+    #         "repetition_penalty": 1.2,
+    #         "stream": false
+    #     }'
+fi
+
+if [ $stage -le 4 ] && [ $stop_stage -ge 4 ]; then
+    echo "Running benchmark client"
+    num_task=4
+    mode=streaming
+    BLS_INSTANCE_NUM=$bls_instance_num
+
+    python3 client_grpc.py \
+        --server-addr localhost \
+        --server-port 8001 \
+        --model-name cosyvoice2_dit \
+        --num-tasks $num_task \
+        --mode $mode \
+        --huggingface-dataset yuekai/seed_tts_cosy2 \
+        --log-dir ./log_single_gpu_concurrent_tasks_${num_task}_${mode}_bls_${BLS_INSTANCE_NUM}
+
+fi
+
+if [ $stage -le 5 ] && [ $stop_stage -ge 5 ]; then
+  echo "stage 5: Offline TTS (Cosyvoice2 LLM + Step-Audio2-mini DiT Token2Wav) inference using a single python script"
+
+  datasets=(wenetspeech4tts) # wenetspeech4tts, test_zh, zero_shot_zh
+  backend=trtllm # hf, trtllm, vllm, trtllm-serve
+
+  batch_sizes=(16)
+  token2wav_batch_size=1
+
+  for batch_size in ${batch_sizes[@]}; do
+    for dataset in ${datasets[@]}; do
+    output_dir=./${dataset}_${backend}_llm_batch_size_${batch_size}_token2wav_batch_size_${token2wav_batch_size}
+    CUDA_VISIBLE_DEVICES=1 \
+        python3 offline_inference.py \
+            --output-dir $output_dir \
+            --llm-model-name-or-path $huggingface_model_local_dir \
+            --token2wav-path $step_audio_model_dir/token2wav \
+            --backend $backend \
+            --batch-size $batch_size --token2wav-batch-size $token2wav_batch_size \
+            --engine-dir $trt_engines_dir \
+            --split-name ${dataset} || exit 1
+    done
+  done
+fi
+
+if [ $stage -le 6 ] && [ $stop_stage -ge 6 ]; then
+   echo "Running Step-Audio2-mini DiT Token2Wav inference using a single python script"
+   export CUDA_VISIBLE_DEVICES=1
+   # Note: Using pre-computed cosyvoice2 tokens
+   python3 streaming_inference.py --enable-trt --strategy equal # equal, exponential
+   # Offline Token2wav inference
+   python3 token2wav_dit.py --enable-trt
+fi
+
+
+if [ $stage -le 7 ] && [ $stop_stage -ge 7 ]; then
+   echo "Disaggregated Server: LLM and Token2wav on different GPUs"
+   echo "Starting LLM server on GPU 0"
+   export CUDA_VISIBLE_DEVICES=0
+   mpirun -np 1 --allow-run-as-root --oversubscribe trtllm-serve serve --tokenizer $huggingface_model_local_dir $trt_engines_dir --max_batch_size 64  --kv_cache_free_gpu_memory_fraction 0.4 &
+   echo "Starting Token2wav server on GPUs 1-3"
+   Token2wav_num_gpus=3
+   http_port=17000
+   grpc_port=18000
+   metrics_port=16000
+   for i in $(seq 0 $(($Token2wav_num_gpus - 1))); do
+       echo "Starting server on GPU $i"
+       http_port=$((http_port + 1))
+       grpc_port=$((grpc_port + 1))
+       metrics_port=$((metrics_port + 1))
+       # Two instances of Token2wav server on the same GPU
+       CUDA_VISIBLE_DEVICES=$(($i + 1)) tritonserver --model-repository $model_repo --http-port $http_port --grpc-port $grpc_port --metrics-port $metrics_port &
+       http_port=$((http_port + 1))
+       grpc_port=$((grpc_port + 1))
+       metrics_port=$((metrics_port + 1))
+       CUDA_VISIBLE_DEVICES=$(($i + 1)) tritonserver --model-repository $model_repo --http-port $http_port --grpc-port $grpc_port --metrics-port $metrics_port &
+   done
+   wait
+fi
+
+if [ $stage -le 8 ] && [ $stop_stage -ge 8 ]; then
+    echo "Running benchmark client for Disaggregated Server"
+    per_gpu_instances=2
+    mode=streaming
+    BLS_INSTANCE_NUM=$bls_instance_num
+    Token2wav_num_gpus=(1 2 3)
+    concurrent_tasks=(1 2 3 4 5 6)
+    for n_gpu in ${Token2wav_num_gpus[@]}; do
+        echo "Test 1 GPU for LLM server and $n_gpu GPUs for Token2wav servers"
+        for concurrent_task in ${concurrent_tasks[@]}; do
+            num_instances=$((per_gpu_instances * n_gpu))
+            for i in $(seq 1 $num_instances); do
+                port=$(($i + 18000))
+                python3 client_grpc.py \
+                    --server-addr localhost \
+                    --server-port $port \
+                    --model-name cosyvoice2_dit \
+                    --num-tasks $concurrent_task \
+                    --mode $mode \
+                    --huggingface-dataset yuekai/seed_tts_cosy2 \
+                    --log-dir ./log_disagg_concurrent_tasks_${concurrent_task}_per_instance_total_token2wav_instances_${num_instances}_port_${port} &
+            done
+            wait
+        done
+    done
+fi

runtime/triton_trtllm/scripts/convert_checkpoint.py

@@ -0,0 +1,330 @@
+import argparse
+import os
+import time
+import traceback
+from concurrent.futures import ThreadPoolExecutor, as_completed
+
+from transformers import AutoConfig
+
+import tensorrt_llm
+from tensorrt_llm._utils import release_gc
+from tensorrt_llm.logger import logger
+from tensorrt_llm.mapping import Mapping
+from tensorrt_llm.models import QWenForCausalLM
+from tensorrt_llm.models.modeling_utils import QuantConfig
+from tensorrt_llm.quantization import QuantAlgo
+
+
+def parse_arguments():
+    parser = argparse.ArgumentParser()
+    parser.add_argument('--model_dir', type=str, default=None, required=True)
+    parser.add_argument('--tp_size',
+                        type=int,
+                        default=1,
+                        help='N-way tensor parallelism size')
+    parser.add_argument('--pp_size',
+                        type=int,
+                        default=1,
+                        help='N-way pipeline parallelism size')
+    parser.add_argument('--cp_size',
+                        type=int,
+                        default=1,
+                        help='N-way context parallelism size')
+    parser.add_argument(
+        '--dtype',
+        type=str,
+        default='auto',
+        choices=['auto', 'float16', 'bfloat16', 'float32'],
+        help="The data type for the model weights and activations if not quantized. "
+        "If 'auto', the data type is automatically inferred from the source model; "
+        "however, if the source dtype is float32, it is converted to float16.")
+    parser.add_argument(
+        '--use_weight_only',
+        default=False,
+        action="store_true",
+        help='Quantize weights for the various GEMMs to INT4/INT8.'
+        'See --weight_only_precision to set the precision')
+    parser.add_argument(
+        '--disable_weight_only_quant_plugin',
+        default=False,
+        action="store_true",
+        help='By default, using plugin implementation for weight quantization. Enabling disable_weight_only_quant_plugin flag will use ootb implementation instead of plugin.'
+        'You must also use --use_weight_only for that argument to have an impact.'
+    )
+    parser.add_argument(
+        '--weight_only_precision',
+        const='int8',
+        type=str,
+        nargs='?',
+        default='int8',
+        choices=['int8', 'int4', 'int4_gptq'],
+        help='Define the precision for the weights when using weight-only quantization.'
+        'You must also use --use_weight_only for that argument to have an impact.'
+    )
+    parser.add_argument(
+        '--calib_dataset',
+        type=str,
+        default='ccdv/cnn_dailymail',
+        help="The huggingface dataset name or the local directory of the dataset for calibration."
+    )
+    parser.add_argument(
+        "--smoothquant",
+        "-sq",
+        type=float,
+        default=None,
+        help="Set the α parameter (see https://arxiv.org/pdf/2211.10438.pdf)"
+        " to Smoothquant the model, and output int8 weights."
+        " A good first try is 0.5. Must be in [0, 1]")
+    parser.add_argument(
+        '--per_channel',
+        action="store_true",
+        default=False,
+        help='By default, we use a single static scaling factor for the GEMM\'s result. '
+        'per_channel instead uses a different static scaling factor for each channel. '
+        'The latter is usually more accurate, but a little slower.')
+    parser.add_argument(
+        '--per_token',
+        action="store_true",
+        default=False,
+        help='By default, we use a single static scaling factor to scale activations in the int8 range. '
+        'per_token chooses at run time, and for each token, a custom scaling factor. '
+        'The latter is usually more accurate, but a little slower.')
+    parser.add_argument(
+        '--int8_kv_cache',
+        default=False,
+        action="store_true",
+        help='By default, we use dtype for KV cache. int8_kv_cache chooses int8 quantization for KV'
+    )
+    parser.add_argument(
+        '--per_group',
+        default=False,
+        action="store_true",
+        help='By default, we use a single static scaling factor to scale weights in the int4 range. '
+        'per_group chooses at run time, and for each group, a custom scaling factor. '
+        'The flag is built for GPTQ/AWQ quantization.')
+
+    parser.add_argument('--group_size',
+                        type=int,
+                        default=128,
+                        help='Group size used in GPTQ quantization.')
+
+    parser.add_argument("--load_model_on_cpu", action="store_true")
+    parser.add_argument(
+        '--use_parallel_embedding',
+        action="store_true",
+        default=False,
+        help='By default embedding parallelism is disabled. By setting this flag, embedding parallelism is enabled'
+    )
+    parser.add_argument(
+        '--embedding_sharding_dim',
+        type=int,
+        default=0,
+        choices=[0, 1],
+        help='By default the embedding lookup table is sharded along vocab dimension (embedding_sharding_dim=0). '
+        'To shard it along hidden dimension, set embedding_sharding_dim=1'
+        'Note: embedding sharing is only enabled when embedding_sharding_dim = 0'
+    )
+    parser.add_argument('--output_dir',
+                        type=str,
+                        default='tllm_checkpoint',
+                        help='The path to save the TensorRT-LLM checkpoint')
+    parser.add_argument(
+        '--workers',
+        type=int,
+        default=1,
+        help='The number of workers for converting checkpoint in parallel')
+    parser.add_argument(
+        '--moe_tp_size',
+        type=int,
+        default=-1,
+        help='N-way tensor parallelism size for MOE, default is tp_size, which will do tp-only for MoE'
+    )
+    parser.add_argument(
+        '--moe_ep_size',
+        type=int,
+        default=-1,
+        help='N-way expert parallelism size for MOE, default is 1, which will do tp-only for MoE'
+    )
+    args = parser.parse_args()
+    return args
+
+
+def args_to_quant_config(args: argparse.Namespace) -> QuantConfig:
+    '''return config dict with quantization info based on the command line args
+    '''
+    quant_config = QuantConfig()
+    if args.use_weight_only:
+        if args.weight_only_precision == 'int8':
+            quant_config.quant_algo = QuantAlgo.W8A16
+        elif args.weight_only_precision == 'int4':
+            quant_config.quant_algo = QuantAlgo.W4A16
+    elif args.smoothquant:
+        quant_config.smoothquant_val = args.smoothquant
+        if args.per_channel:
+            if args.per_token:
+                quant_config.quant_algo = QuantAlgo.W8A8_SQ_PER_CHANNEL_PER_TOKEN_PLUGIN
+            else:
+                quant_config.quant_algo = QuantAlgo.W8A8_SQ_PER_CHANNEL_PER_TENSOR_PLUGIN
+        else:
+            if args.per_token:
+                quant_config.quant_algo = QuantAlgo.W8A8_SQ_PER_TENSOR_PER_TOKEN_PLUGIN
+            else:
+                quant_config.quant_algo = QuantAlgo.W8A8_SQ_PER_TENSOR_PLUGIN
+
+    if args.int8_kv_cache:
+        quant_config.kv_cache_quant_algo = QuantAlgo.INT8
+
+    if args.weight_only_precision == 'int4_gptq':
+        quant_config.group_size = args.group_size
+        quant_config.has_zero_point = True
+        quant_config.pre_quant_scale = False
+        quant_config.quant_algo = QuantAlgo.W4A16_GPTQ
+
+    return quant_config
+
+
+def update_quant_config_from_hf(quant_config, hf_config,
+                                override_fields) -> tuple[QuantConfig, dict]:
+    hf_config_dict = hf_config.to_dict()
+    if hf_config_dict.get('quantization_config'):
+        # update the quant_algo, and clamp_val.
+        if hf_config_dict['quantization_config'].get('quant_method') == 'awq':
+            logger.info(
+                "Load quantization configs from huggingface model_config.")
+            quant_config.quant_algo = QuantAlgo.W4A16_GPTQ
+            quant_config.group_size = hf_config_dict['quantization_config'].get(
+                'group_size', 128)
+            quant_config.has_zero_point = hf_config_dict[
+                'quantization_config'].get('zero_point', False)
+            override_fields.update({"use_autoawq": True})
+        elif hf_config_dict['quantization_config'].get(
+                'quant_method') == 'gptq':
+            logger.info(
+                "Load quantization configs from huggingface model_config.")
+            desc_act = hf_config_dict['quantization_config'].get(
+                'desc_act', False)
+            if desc_act:
+                raise ValueError("GPTQ with desc_act=True is not implemented!")
+            quant_config.quant_algo = QuantAlgo.W4A16_GPTQ
+            quant_config.group_size = hf_config_dict['quantization_config'].get(
+                'group_size', 128)
+            quant_config.has_zero_point = hf_config_dict[
+                'quantization_config'].get('sym', False)
+    return quant_config, override_fields
+
+
+def args_to_build_options(args):
+    return {
+        'use_parallel_embedding': args.use_parallel_embedding,
+        'embedding_sharding_dim': args.embedding_sharding_dim,
+        'disable_weight_only_quant_plugin':
+        args.disable_weight_only_quant_plugin
+    }
+
+
+def convert_and_save_hf(args):
+    model_dir = args.model_dir
+    world_size = args.tp_size * args.pp_size
+    # Need to convert the cli args to the kay-value pairs and override them in the generate config dict.
+    # Ideally these fields will be moved out of the config and pass them into build API, keep them here for compatibility purpose for now,
+    # before the refactor is done.
+    override_fields = {}
+    override_fields.update(args_to_build_options(args))
+    quant_config = args_to_quant_config(args)
+
+    try:
+        hf_config = AutoConfig.from_pretrained(model_dir,
+                                               trust_remote_code=True)
+        quant_config, override_fields = update_quant_config_from_hf(
+            quant_config, hf_config, override_fields)
+    except BaseException:
+        logger.warning("AutoConfig cannot load the huggingface config.")
+
+    if args.smoothquant is not None or args.int8_kv_cache:
+        mapping = Mapping(world_size=world_size,
+                          tp_size=args.tp_size,
+                          pp_size=args.pp_size,
+                          moe_tp_size=args.moe_tp_size,
+                          moe_ep_size=args.moe_ep_size,
+                          cp_size=args.cp_size)
+        QWenForCausalLM.quantize(args.model_dir,
+                                 args.output_dir,
+                                 dtype=args.dtype,
+                                 mapping=mapping,
+                                 quant_config=quant_config,
+                                 calib_dataset=args.calib_dataset,
+                                 **override_fields)
+    else:
+
+        def convert_and_save_rank(args, rank):
+            mapping = Mapping(world_size=world_size,
+                              rank=rank,
+                              tp_size=args.tp_size,
+                              pp_size=args.pp_size,
+                              moe_tp_size=args.moe_tp_size,
+                              moe_ep_size=args.moe_ep_size)
+            qwen = QWenForCausalLM.from_hugging_face(model_dir,
+                                                     args.dtype,
+                                                     mapping=mapping,
+                                                     quant_config=quant_config,
+                                                     **override_fields)
+            qwen.config.mapping.cp_size = args.cp_size
+            qwen.config.mapping.attn_tp_size = -1
+            qwen.config.mapping.attn_cp_size = -1
+            qwen.config.mapping.world_size *= args.cp_size
+            qwen.save_checkpoint(args.output_dir, save_config=(rank == 0))
+            del qwen
+
+        execute(args.workers, [convert_and_save_rank] * world_size, args)
+        release_gc()
+
+
+def execute(workers, func, args):
+    if workers == 1:
+        for rank, f in enumerate(func):
+            f(args, rank)
+    else:
+        with ThreadPoolExecutor(max_workers=workers) as p:
+            futures = [p.submit(f, args, rank) for rank, f in enumerate(func)]
+            exceptions = []
+            for future in as_completed(futures):
+                try:
+                    future.result()
+                except Exception as e:
+                    traceback.print_exc()
+                    exceptions.append(e)
+            assert len(
+                exceptions
+            ) == 0, "Checkpoint conversion failed, please check error log."
+
+
+def main():
+    print(tensorrt_llm.__version__)
+    args = parse_arguments()
+
+    if (args.moe_tp_size == -1 and args.moe_ep_size == -1):
+        # moe default to tp-only
+        args.moe_tp_size = args.tp_size
+        args.moe_ep_size = 1
+    elif (args.moe_tp_size == -1):
+        args.moe_tp_size = args.tp_size // args.moe_ep_size
+    elif (args.moe_ep_size == -1):
+        args.moe_ep_size = args.tp_size // args.moe_tp_size
+    assert (args.moe_tp_size * args.moe_ep_size == args.tp_size
+            ), "moe_tp_size * moe_ep_size must equal to tp_size"
+
+    tik = time.time()
+
+    if not os.path.exists(args.output_dir):
+        os.makedirs(args.output_dir)
+
+    assert args.model_dir is not None
+    convert_and_save_hf(args)
+
+    tok = time.time()
+    t = time.strftime('%H:%M:%S', time.gmtime(tok - tik))
+    print(f'Total time of converting checkpoints: {t}')
+
+
+if __name__ == '__main__':
+    main()

runtime/triton_trtllm/scripts/fill_template.py

@@ -0,0 +1,69 @@
+# /usr/bin/env python3
+from argparse import ArgumentParser
+from string import Template
+
+
+def split(string, delimiter):
+    """Split a string using delimiter. Supports escaping.
+
+    Args:
+        string (str): The string to split.
+        delimiter (str): The delimiter to split the string with.
+
+    Returns:
+        list: A list of strings.
+    """
+    result = []
+    current = ""
+    escape = False
+    for char in string:
+        if escape:
+            current += char
+            escape = False
+        elif char == delimiter:
+            result.append(current)
+            current = ""
+        elif char == "\\":
+            escape = True
+        else:
+            current += char
+    result.append(current)
+    return result
+
+
+def main(file_path, substitutions, in_place):
+    with open(file_path) as f:
+        pbtxt = Template(f.read())
+
+    sub_dict = {
+        "max_queue_size": 0,
+        'max_queue_delay_microseconds': 0,
+    }
+    for sub in split(substitutions, ","):
+        key, value = split(sub, ":")
+        sub_dict[key] = value
+
+        assert key in pbtxt.template, f"key '{key}' does not exist in the file {file_path}."
+
+    pbtxt = pbtxt.safe_substitute(sub_dict)
+
+    if in_place:
+        with open(file_path, "w") as f:
+            f.write(pbtxt)
+    else:
+        print(pbtxt)
+
+
+if __name__ == "__main__":
+    parser = ArgumentParser()
+    parser.add_argument("file_path", help="path of the .pbtxt to modify")
+    parser.add_argument(
+        "substitutions",
+        help="substitutions to perform, in the format variable_name_1:value_1,variable_name_2:value_2..."
+    )
+    parser.add_argument("--in_place",
+                        "-i",
+                        action="store_true",
+                        help="do the operation in-place")
+    args = parser.parse_args()
+    main(**vars(args))

runtime/triton_trtllm/scripts/test_llm.py

@@ -0,0 +1,138 @@
+
+# SPDX-FileCopyrightText: Copyright (c) 2022-2024 NVIDIA CORPORATION & AFFILIATES. All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+
+import argparse
+
+import numpy as np
+import torch
+
+import tensorrt_llm
+from tensorrt_llm.logger import logger
+
+from tensorrt_llm.runtime import ModelRunnerCpp
+from transformers import AutoTokenizer
+
+
+def parse_arguments(args=None):
+    parser = argparse.ArgumentParser()
+    parser.add_argument(
+        '--input_text',
+        type=str,
+        nargs='+',
+        default=["Born in north-east France, Soyer trained as a"])
+    parser.add_argument('--tokenizer_dir', type=str, default="meta-llama/Meta-Llama-3-8B-Instruct")
+    parser.add_argument('--engine_dir', type=str, default="meta-llama/Meta-Llama-3-8B-Instruct")
+    parser.add_argument('--log_level', type=str, default="debug")
+    parser.add_argument('--kv_cache_free_gpu_memory_fraction', type=float, default=0.6)
+    parser.add_argument('--temperature', type=float, default=0.8)
+    parser.add_argument('--top_k', type=int, default=50)
+    parser.add_argument('--top_p', type=float, default=0.95)
+
+    return parser.parse_args(args=args)
+
+
+def parse_input(tokenizer,
+                input_text=None,
+                prompt_template=None):
+    batch_input_ids = []
+    for curr_text in input_text:
+        if prompt_template is not None:
+            curr_text = prompt_template.format(input_text=curr_text)
+        input_ids = tokenizer.encode(
+            curr_text)
+        batch_input_ids.append(input_ids)
+
+    batch_input_ids = [
+        torch.tensor(x, dtype=torch.int32) for x in batch_input_ids
+    ]
+
+    logger.debug(f"Input token ids (batch_size = {len(batch_input_ids)}):")
+    for i, input_ids in enumerate(batch_input_ids):
+        logger.debug(f"Request {i}: {input_ids.tolist()}")
+
+    return batch_input_ids
+
+
+def main(args):
+    runtime_rank = tensorrt_llm.mpi_rank()
+    logger.set_level(args.log_level)
+
+    tokenizer = AutoTokenizer.from_pretrained(args.tokenizer_dir)
+    prompt_template = "<|sos|>{input_text}<|task_id|>"
+    end_id = tokenizer.convert_tokens_to_ids("<|eos1|>")
+
+    batch_input_ids = parse_input(tokenizer=tokenizer,
+                                  input_text=args.input_text,
+                                  prompt_template=prompt_template)
+
+    input_lengths = [x.size(0) for x in batch_input_ids]
+
+    runner_kwargs = dict(
+        engine_dir=args.engine_dir,
+        rank=runtime_rank,
+        max_output_len=1024,
+        enable_context_fmha_fp32_acc=False,
+        max_batch_size=len(batch_input_ids),
+        max_input_len=max(input_lengths),
+        kv_cache_free_gpu_memory_fraction=args.kv_cache_free_gpu_memory_fraction,
+        cuda_graph_mode=False,
+        gather_generation_logits=False,
+    )
+
+    runner = ModelRunnerCpp.from_dir(**runner_kwargs)
+
+    with torch.no_grad():
+        outputs = runner.generate(
+            batch_input_ids=batch_input_ids,
+            max_new_tokens=1024,
+            end_id=end_id,
+            pad_id=end_id,
+            temperature=args.temperature,
+            top_k=args.top_k,
+            top_p=args.top_p,
+            num_return_sequences=1,
+            repetition_penalty=1.1,
+            random_seed=42,
+            streaming=False,
+            output_sequence_lengths=True,
+            output_generation_logits=False,
+            return_dict=True,
+            return_all_generated_tokens=False)
+        torch.cuda.synchronize()
+        output_ids, sequence_lengths = outputs["output_ids"], outputs["sequence_lengths"]
+        num_output_sents, num_beams, _ = output_ids.size()
+        assert num_beams == 1
+        beam = 0
+        batch_size = len(input_lengths)
+        num_return_sequences = num_output_sents // batch_size
+        assert num_return_sequences == 1
+        for i in range(batch_size * num_return_sequences):
+            batch_idx = i // num_return_sequences
+            seq_idx = i % num_return_sequences
+            inputs = output_ids[i][0][:input_lengths[batch_idx]].tolist()
+            input_text = tokenizer.decode(inputs)
+            print(f'Input [Text {batch_idx}]: \"{input_text}\"')
+            output_begin = input_lengths[batch_idx]
+            output_end = sequence_lengths[i][beam]
+            outputs = output_ids[i][beam][output_begin:output_end].tolist()
+            output_text = tokenizer.decode(outputs)
+            print(f'Output [Text {batch_idx}]: \"{output_text}\"')
+            logger.debug(str(outputs))
+
+
+if __name__ == '__main__':
+    args = parse_arguments()
+    main(args)

runtime/triton_trtllm/streaming_inference.py

@@ -0,0 +1,122 @@
+import torch
+import os
+import argparse
+from datasets import load_dataset
+from torch.utils.data import DataLoader
+import numpy as np
+import torchaudio
+import time
+from token2wav_dit import CosyVoice2_Token2Wav
+import soundfile as sf
+
+
+def collate_fn(batch):
+    ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate = [], [], [], []
+    prompt_speech_tokens_list, prompt_text_list = [], []
+    for item in batch:
+        generated_speech_tokens_list.append(item['target_audio_cosy2_tokens'])
+        audio = torch.from_numpy(item['prompt_audio']['array']).float()
+        prompt_audios_list.append(audio)
+        prompt_audios_sample_rate.append(item['prompt_audio']['sampling_rate'])
+        ids.append(item['id'])
+        prompt_speech_tokens_list.append(item['prompt_audio_cosy2_tokens'])
+        prompt_text_list.append(item['prompt_text'])
+
+    return ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate, prompt_speech_tokens_list, prompt_text_list
+
+
+def get_args():
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--enable-trt", action="store_true")
+    parser.add_argument("--model-dir", type=str, default="./Step-Audio-2-mini/token2wav")
+    parser.add_argument("--batch-size", type=int, default=1)
+    parser.add_argument("--output-dir", type=str, default="generated_wavs")
+    parser.add_argument("--huggingface-dataset-split", type=str, default="wenetspeech4tts")
+    parser.add_argument("--dataset-name", type=str, default="yuekai/seed_tts_cosy2")
+    parser.add_argument("--strategy", type=str, default="equal", choices=["equal", "exponential"])
+    return parser.parse_args()
+
+
+if __name__ == "__main__":
+    args = get_args()
+
+    if not os.path.exists(args.output_dir):
+        os.makedirs(args.output_dir)
+
+    dataset_name = args.dataset_name
+    dataset = load_dataset(dataset_name, split=args.huggingface_dataset_split, trust_remote_code=True)
+    data_loader = DataLoader(dataset, batch_size=args.batch_size, shuffle=False, collate_fn=collate_fn, num_workers=0)
+
+    token2wav_model = CosyVoice2_Token2Wav(model_dir=args.model_dir, enable_trt=args.enable_trt, streaming=True)
+
+    CHUNK_SIZE = 25
+    token_frame_rate = 25
+    OVERLAP_SIZE = 0
+
+    warmup_times = 3
+    for _ in range(warmup_times):
+        start_time = time.time()
+        total_forward_count = 0
+        for batch in data_loader:
+            tts_speech_list = []
+            ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate, prompt_speech_tokens_list, prompt_text_list = batch
+
+            id, generated_speech_tokens, prompt_audio, prompt_audio_sample_rate = ids[0], generated_speech_tokens_list[0], prompt_audios_list[0], prompt_audios_sample_rate[0]
+
+            assert prompt_audio_sample_rate == 16000
+
+            prompt_text = prompt_text_list[0]
+            prompt_speech_tokens = prompt_speech_tokens_list[0]
+
+            semantic_token_ids_arr, token_offset = [], 0
+            flow_prompt_speech_token_len = len(prompt_speech_tokens)
+
+            buffer = generated_speech_tokens
+            output_wavs = []
+            chunk_index = 0
+            while True:
+                if args.strategy == "equal":
+                    this_chunk_size = CHUNK_SIZE
+                elif args.strategy == "exponential":
+                    this_chunk_size = token_frame_rate * (2 ** chunk_index)
+
+                if len(buffer) >= this_chunk_size + token2wav_model.flow.pre_lookahead_len:
+                    wavs = token2wav_model.forward_streaming(
+                        buffer[:this_chunk_size + token2wav_model.flow.pre_lookahead_len],
+                        False, request_id=id, speaker_id=f"{id}", prompt_audio=prompt_audio,
+                        prompt_audio_sample_rate=prompt_audio_sample_rate
+                    )
+                    buffer = buffer[this_chunk_size - OVERLAP_SIZE:]
+
+                    output_wavs.append(wavs)
+                    total_forward_count += 1
+                    chunk_index += 1
+
+                else:
+                    wavs = token2wav_model.forward_streaming(
+                        buffer, True, request_id=id, speaker_id=f"{id}",
+                        prompt_audio=prompt_audio, prompt_audio_sample_rate=prompt_audio_sample_rate
+                    )
+                    output_wavs.append(wavs)
+                    total_forward_count += 1
+                    # chunk_index += 1
+                    break
+
+            for i, wav in enumerate(output_wavs):
+                output_wavs[i] = wav.cpu().numpy().squeeze()
+
+            audios = output_wavs
+            reconstructed_audio = np.concatenate(audios)
+            sf.write(os.path.join(args.output_dir, f"{id}.wav"), reconstructed_audio, 24000, "PCM_16")
+
+        end_time = time.time()
+
+        if _ == 0:
+            token2wav_model.speaker_cache = {}
+            print(f"Warmup time: {end_time - start_time} seconds")
+            print("clear speaker cache")
+        elif _ == 1:
+            print(f"Cost time without speaker cache: {end_time - start_time} seconds")
+        else:
+            print(f"Cost time with speaker cache: {end_time - start_time} seconds")
+            print(f"Total flow matching forward calls: {total_forward_count}")

runtime/triton_trtllm/token2wav.py

@@ -0,0 +1,335 @@
+# SPDX-FileCopyrightText: Copyright (c) 2025, NVIDIA CORPORATION.  All rights reserved.
+# SPDX-License-Identifier: Apache-2.0
+#
+# 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.
+""" Example Usage
+    CUDA_VISIBLE_DEVICES=0 \
+        python3 token2wav.py --enable-trt || exit 1
+"""
+import torch
+from flashcosyvoice.modules.flow import CausalMaskedDiffWithXvec
+from flashcosyvoice.modules.hifigan import HiFTGenerator
+from flashcosyvoice.utils.audio import mel_spectrogram
+import torchaudio.compliance.kaldi as kaldi
+import onnxruntime
+import s3tokenizer
+from torch.utils.data import DataLoader
+from datasets import load_dataset
+import torchaudio
+import os
+import logging
+import argparse
+import queue
+import time
+
+
+def convert_onnx_to_trt(trt_model, trt_kwargs, onnx_model, fp16):
+    import tensorrt as trt
+    logging.info("Converting onnx to trt...")
+    network_flags = 1 << int(trt.NetworkDefinitionCreationFlag.EXPLICIT_BATCH)
+    logger = trt.Logger(trt.Logger.INFO)
+    builder = trt.Builder(logger)
+    network = builder.create_network(network_flags)
+    parser = trt.OnnxParser(network, logger)
+    config = builder.create_builder_config()
+    # config.set_memory_pool_limit(trt.MemoryPoolType.WORKSPACE, 1 << 32)  # 4GB
+    if fp16:
+        config.set_flag(trt.BuilderFlag.FP16)
+    profile = builder.create_optimization_profile()
+    # load onnx model
+    with open(onnx_model, "rb") as f:
+        if not parser.parse(f.read()):
+            for error in range(parser.num_errors):
+                print(parser.get_error(error))
+            raise ValueError('failed to parse {}'.format(onnx_model))
+    # set input shapes
+    for i in range(len(trt_kwargs['input_names'])):
+        profile.set_shape(trt_kwargs['input_names'][i], trt_kwargs['min_shape'][i], trt_kwargs['opt_shape'][i], trt_kwargs['max_shape'][i])
+    tensor_dtype = trt.DataType.HALF if fp16 else trt.DataType.FLOAT
+    # set input and output data type
+    for i in range(network.num_inputs):
+        input_tensor = network.get_input(i)
+        input_tensor.dtype = tensor_dtype
+    for i in range(network.num_outputs):
+        output_tensor = network.get_output(i)
+        output_tensor.dtype = tensor_dtype
+    config.add_optimization_profile(profile)
+    engine_bytes = builder.build_serialized_network(network, config)
+    # save trt engine
+    with open(trt_model, "wb") as f:
+        f.write(engine_bytes)
+    logging.info("Succesfully convert onnx to trt...")
+
+
+class TrtContextWrapper:
+    def __init__(self, trt_engine, trt_concurrent=1, device='cuda:0'):
+        self.trt_context_pool = queue.Queue(maxsize=trt_concurrent)
+        self.trt_engine = trt_engine
+        self.device = device
+        for _ in range(trt_concurrent):
+            trt_context = trt_engine.create_execution_context()
+            trt_stream = torch.cuda.stream(torch.cuda.Stream(torch.device(device)))
+            assert trt_context is not None, 'failed to create trt context, maybe not enough CUDA memory, try reduce current trt concurrent {}'.format(trt_concurrent)
+            self.trt_context_pool.put([trt_context, trt_stream])
+        assert self.trt_context_pool.empty() is False, 'no avaialbe estimator context'
+
+    def acquire_estimator(self):
+        return self.trt_context_pool.get(), self.trt_engine
+
+    def release_estimator(self, context, stream):
+        self.trt_context_pool.put([context, stream])
+
+
+class CosyVoice2_Token2Wav(torch.nn.Module):
+    def __init__(self, model_dir: str = "./CosyVoice2-0.5B", enable_trt: bool = False, device_id: int = 0):
+        super().__init__()
+        self.device_id = device_id
+        self.device = f"cuda:{device_id}"
+
+        self.flow = CausalMaskedDiffWithXvec()
+        self.flow.half()
+        self.flow.load_state_dict(torch.load(f"{model_dir}/flow.pt", map_location="cpu", weights_only=True), strict=True)
+        self.flow.to(self.device).eval()
+
+        self.hift = HiFTGenerator()
+        hift_state_dict = {k.replace('generator.', ''): v for k, v in torch.load(f"{model_dir}/hift.pt", map_location="cpu", weights_only=True).items()}
+        self.hift.load_state_dict(hift_state_dict, strict=True)
+        self.hift.to(self.device).eval()
+
+        option = onnxruntime.SessionOptions()
+        option.graph_optimization_level = onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
+        option.intra_op_num_threads = 1
+        self.spk_model = onnxruntime.InferenceSession(f"{model_dir}/campplus.onnx", sess_options=option, providers=["CPUExecutionProvider"])
+
+        self.audio_tokenizer = s3tokenizer.load_model(f"{model_dir}/speech_tokenizer_v2.onnx").to(self.device).eval()
+
+        gpu = "l20"
+        if enable_trt:
+            self.load_trt(f'{model_dir}/flow.decoder.estimator.fp16.dynamic_batch.{gpu}.plan',
+                          f'{model_dir}/flow.decoder.estimator.fp32.dynamic_batch.onnx',
+                          1,
+                          True)
+            self.load_spk_trt(f'{model_dir}/campplus.{gpu}.fp32.trt',
+                              f'{model_dir}/campplus.onnx',
+                              1,
+                              False)
+
+    def forward_spk_embedding(self, spk_feat):
+        if isinstance(self.spk_model, onnxruntime.InferenceSession):
+            return self.spk_model.run(
+                None, {self.spk_model.get_inputs()[0].name: spk_feat.unsqueeze(dim=0).cpu().numpy()}
+            )[0].flatten().tolist()
+        else:
+            [spk_model, stream], trt_engine = self.spk_model.acquire_estimator()
+            # NOTE need to synchronize when switching stream
+            with torch.cuda.device(self.device_id):
+                torch.cuda.current_stream().synchronize()
+                spk_feat = spk_feat.unsqueeze(dim=0).to(self.device)
+                batch_size = spk_feat.size(0)
+
+                with stream:
+                    spk_model.set_input_shape('input', (batch_size, spk_feat.size(1), 80))
+                    output_tensor = torch.empty((batch_size, 192), device=spk_feat.device)
+
+                    data_ptrs = [spk_feat.contiguous().data_ptr(),
+                                 output_tensor.contiguous().data_ptr()]
+                    for i, j in enumerate(data_ptrs):
+
+                        spk_model.set_tensor_address(trt_engine.get_tensor_name(i), j)
+                    # run trt engine
+                    assert spk_model.execute_async_v3(torch.cuda.current_stream().cuda_stream) is True
+                    torch.cuda.current_stream().synchronize()
+                self.spk_model.release_estimator(spk_model, stream)
+
+            return output_tensor.cpu().numpy().flatten().tolist()
+
+    def load_spk_trt(self, spk_model, spk_onnx_model, trt_concurrent=1, fp16=True):
+        if not os.path.exists(spk_model) or os.path.getsize(spk_model) == 0:
+            trt_kwargs = self.get_spk_trt_kwargs()
+            convert_onnx_to_trt(spk_model, trt_kwargs, spk_onnx_model, fp16)
+        import tensorrt as trt
+        with open(spk_model, 'rb') as f:
+            spk_engine = trt.Runtime(trt.Logger(trt.Logger.INFO)).deserialize_cuda_engine(f.read())
+        assert spk_engine is not None, 'failed to load trt {}'.format(spk_model)
+        self.spk_model = TrtContextWrapper(spk_engine, trt_concurrent=trt_concurrent, device=self.device)
+
+    def get_spk_trt_kwargs(self):
+        min_shape = [(1, 4, 80)]
+        opt_shape = [(1, 500, 80)]
+        max_shape = [(1, 3000, 80)]
+        input_names = ["input"]
+        return {'min_shape': min_shape, 'opt_shape': opt_shape, 'max_shape': max_shape, 'input_names': input_names}
+
+    def load_trt(self, flow_decoder_estimator_model, flow_decoder_onnx_model, trt_concurrent=1, fp16=True):
+        assert torch.cuda.is_available(), 'tensorrt only supports gpu!'
+        if not os.path.exists(flow_decoder_estimator_model) or os.path.getsize(flow_decoder_estimator_model) == 0:
+            trt_kwargs = self.get_trt_kwargs_dynamic_batch(opt_bs=2, max_batch_size=16)
+            convert_onnx_to_trt(flow_decoder_estimator_model, trt_kwargs, flow_decoder_onnx_model, fp16)
+        del self.flow.decoder.estimator
+        import tensorrt as trt
+        with open(flow_decoder_estimator_model, 'rb') as f:
+            estimator_engine = trt.Runtime(trt.Logger(trt.Logger.INFO)).deserialize_cuda_engine(f.read())
+        assert estimator_engine is not None, 'failed to load trt {}'.format(flow_decoder_estimator_model)
+        self.flow.decoder.estimator = TrtContextWrapper(estimator_engine, trt_concurrent=trt_concurrent, device=self.device)
+
+    def get_trt_kwargs_dynamic_batch(self, opt_bs=2, max_batch_size=64):
+        min_shape = [(2, 80, 4), (2, 1, 4), (2, 80, 4), (2, 80, 4), (2,), (2, 80)]
+        opt_shape = [(opt_bs * 2, 80, 500), (opt_bs * 2, 1, 500), (opt_bs * 2, 80, 500), (opt_bs * 2, 80, 500), (opt_bs * 2,), (opt_bs * 2, 80)]
+        max_shape = [(max_batch_size * 2, 80, 3000), (max_batch_size * 2, 1, 3000), (max_batch_size * 2, 80, 3000), (max_batch_size * 2, 80, 3000), (max_batch_size * 2,),
+                     (max_batch_size * 2, 80)]
+        input_names = ["x", "mask", "mu", "cond", "t", "spks"]
+        return {'min_shape': min_shape, 'opt_shape': opt_shape, 'max_shape': max_shape, 'input_names': input_names}
+
+    def prompt_audio_tokenization(self, prompt_audios_list: list[torch.Tensor]) -> list[list[int]]:
+        prompt_speech_tokens_list, prompt_speech_mels_list = [], []
+        for audio in prompt_audios_list:
+            assert len(audio.shape) == 1
+            log_mel = s3tokenizer.log_mel_spectrogram(audio)  # [num_mels, T]
+            prompt_speech_mels_list.append(log_mel)
+        prompt_mels_for_llm, prompt_mels_lens_for_llm = s3tokenizer.padding(prompt_speech_mels_list)
+        prompt_speech_tokens, prompt_speech_tokens_lens = self.audio_tokenizer.quantize(
+            prompt_mels_for_llm.to(self.device), prompt_mels_lens_for_llm.to(self.device)
+        )
+        for i in range(len(prompt_speech_tokens)):
+            speech_tokens_i = prompt_speech_tokens[i, :prompt_speech_tokens_lens[i].item()].tolist()
+            prompt_speech_tokens_list.append(speech_tokens_i)
+        return prompt_speech_tokens_list
+
+    def get_spk_emb(self, prompt_audios_list: list[torch.Tensor]) -> torch.Tensor:
+        spk_emb_for_flow = []
+        for audio in prompt_audios_list:
+            assert len(audio.shape) == 1
+            spk_feat = kaldi.fbank(audio.unsqueeze(0), num_mel_bins=80, dither=0, sample_frequency=16000)
+            spk_feat = spk_feat - spk_feat.mean(dim=0, keepdim=True)
+            spk_emb = self.forward_spk_embedding(spk_feat)
+
+            spk_emb_for_flow.append(spk_emb)
+        spk_emb_for_flow = torch.tensor(spk_emb_for_flow)
+        return spk_emb_for_flow
+
+    def get_prompt_mels(self, prompt_audios_list: list[torch.Tensor], prompt_audios_sample_rate: list[int]):
+        prompt_mels_for_flow = []
+        prompt_mels_lens_for_flow = []
+        for audio, sample_rate in zip(prompt_audios_list, prompt_audios_sample_rate):
+            assert len(audio.shape) == 1
+            audio = audio.unsqueeze(0)
+            if sample_rate != 24000:
+                audio = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=24000)(audio)
+            mel = mel_spectrogram(audio).transpose(1, 2).squeeze(0)  # [T, num_mels]
+            mel_len = mel.shape[0]
+            prompt_mels_for_flow.append(mel)
+            prompt_mels_lens_for_flow.append(mel_len)
+        prompt_mels_for_flow = torch.nn.utils.rnn.pad_sequence(prompt_mels_for_flow, batch_first=True, padding_value=0)  # [B, T', num_mels=80]
+        prompt_mels_lens_for_flow = torch.tensor(prompt_mels_lens_for_flow)
+        return prompt_mels_for_flow, prompt_mels_lens_for_flow
+
+    def forward_flow(self, prompt_speech_tokens_list: list[list[int]], generated_speech_tokens_list: list[list[int]], prompt_mels_for_flow: torch.Tensor,
+                     prompt_mels_lens_for_flow: torch.Tensor, spk_emb_for_flow: torch.Tensor):
+        batch_size = prompt_mels_for_flow.shape[0]
+        flow_inputs = []
+        flow_inputs_lens = []
+        for prompt_speech_tokens, generated_speech_tokens in zip(prompt_speech_tokens_list, generated_speech_tokens_list):
+            flow_inputs.append(torch.tensor(prompt_speech_tokens + generated_speech_tokens))
+            flow_inputs_lens.append(len(prompt_speech_tokens) + len(generated_speech_tokens))
+
+        flow_inputs = torch.nn.utils.rnn.pad_sequence(flow_inputs, batch_first=True, padding_value=0)
+        flow_inputs_lens = torch.tensor(flow_inputs_lens)
+
+        with torch.amp.autocast(self.device, dtype=torch.float16):
+            generated_mels, generated_mels_lens = self.flow(
+                flow_inputs.to(self.device), flow_inputs_lens.to(self.device),
+                prompt_mels_for_flow.to(self.device), prompt_mels_lens_for_flow.to(self.device), spk_emb_for_flow.to(self.device),
+                streaming=False, finalize=True
+            )
+
+        return generated_mels, generated_mels_lens
+
+    def forward_hift(self, generated_mels: torch.Tensor, generated_mels_lens: torch.Tensor, prompt_mels_lens_for_flow: torch.Tensor):
+        batch_size = generated_mels.shape[0]
+        generated_wavs = []
+        for i in range(batch_size):
+            mel = generated_mels[i, :, prompt_mels_lens_for_flow[i].item():generated_mels_lens[i].item()].unsqueeze(0)
+            wav, _ = self.hift(speech_feat=mel)
+            generated_wavs.append(wav)
+        return generated_wavs
+
+    @torch.inference_mode()
+    def forward(
+        self, generated_speech_tokens_list: list[list[int]], prompt_audios_list: list[torch.Tensor], prompt_audios_sample_rate: list[int]
+    ):
+        # assert all item in prompt_audios_sample_rate is 16000
+        assert all(sample_rate == 16000 for sample_rate in prompt_audios_sample_rate)
+
+        prompt_speech_tokens_list = self.prompt_audio_tokenization(prompt_audios_list)
+
+        prompt_mels_for_flow, prompt_mels_lens_for_flow = self.get_prompt_mels(prompt_audios_list, prompt_audios_sample_rate)
+
+        spk_emb_for_flow = self.get_spk_emb(prompt_audios_list)
+
+        generated_mels, generated_mels_lens = self.forward_flow(
+            prompt_speech_tokens_list, generated_speech_tokens_list, prompt_mels_for_flow, prompt_mels_lens_for_flow, spk_emb_for_flow)
+
+        generated_wavs = self.forward_hift(generated_mels, generated_mels_lens, prompt_mels_lens_for_flow)
+
+        return generated_wavs
+
+
+def collate_fn(batch):
+    ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate = [], [], [], []
+    for _, item in enumerate(batch):
+        generated_speech_tokens_list.append(item['target_audio_cosy2_tokens'])
+        audio = torch.from_numpy(item['prompt_audio']['array']).float()
+        prompt_audios_list.append(audio)
+        prompt_audios_sample_rate.append(item['prompt_audio']['sampling_rate'])
+        ids.append(item['id'])
+
+    return ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate
+
+
+def get_args():
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--enable-trt", action="store_true")
+    parser.add_argument("--model-dir", type=str, default="./CosyVoice2-0.5B")
+    parser.add_argument("--batch-size", type=int, default=4)
+    parser.add_argument("--output-dir", type=str, default="generated_wavs")
+    parser.add_argument("--huggingface-dataset-split", type=str, default="wenetspeech4tts")
+    parser.add_argument("--warmup", type=int, default=3, help="Number of warmup epochs, performance statistics will only be collected from the last epoch")
+    return parser.parse_args()
+
+
+if __name__ == "__main__":
+    args = get_args()
+    model = CosyVoice2_Token2Wav(model_dir=args.model_dir, enable_trt=args.enable_trt)
+    # mkdir output_dir if not exists
+    if not os.path.exists(args.output_dir):
+        os.makedirs(args.output_dir)
+    dataset_name = "yuekai/seed_tts_cosy2"
+
+    dataset = load_dataset(dataset_name, split=args.huggingface_dataset_split, trust_remote_code=True)
+
+    data_loader = DataLoader(dataset, batch_size=args.batch_size, shuffle=False, collate_fn=collate_fn, num_workers=0)
+
+    for _ in range(args.warmup):
+        start_time = time.time()
+
+        for batch in data_loader:
+            ids, generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate = batch
+
+            generated_wavs = model(generated_speech_tokens_list, prompt_audios_list, prompt_audios_sample_rate)
+
+            for id, wav in zip(ids, generated_wavs):
+                torchaudio.save(f"{args.output_dir}/{id}.wav", wav.cpu(), 24000)
+
+        end_time = time.time()
+        epoch_time = end_time - start_time
+        print(f"Measurement epoch time taken: {epoch_time:.4f} seconds")

runtime/triton_trtllm/token2wav_dit.py

@@ -0,0 +1 @@
+model_repo/token2wav_dit/1/token2wav_dit.py

tools/make_parquet_list.py

@@ -29,27 +29,24 @@ def job(utt_list, parquet_file, utt2parquet_file, spk2parquet_file):
     for utt in tqdm(utt_list):
         data = open(utt2wav[utt], 'rb').read()
         data_list.append(data)
-    wav_list = [utt2wav[utt] for utt in utt_list]
-    text_list = [utt2text[utt] for utt in utt_list]
-    spk_list = [utt2spk[utt] for utt in utt_list]
-    uttembedding_list = [utt2embedding[utt] for utt in utt_list]
-    spkembedding_list = [spk2embedding[utt2spk[utt]] for utt in utt_list]
-    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]
 
     # 保存到parquet,utt2parquet_file,spk2parquet_file
     df = pd.DataFrame()
     df['utt'] = utt_list
-    df['wav'] = wav_list
     df['audio_data'] = data_list
-    df['text'] = text_list
-    df['spk'] = spk_list
-    df['utt_embedding'] = uttembedding_list
-    df['spk_embedding'] = spkembedding_list
-    df['speech_token'] = speech_token_list
+    df['wav'] = [utt2wav[utt] for utt in utt_list]
+    df['text'] = [utt2text[utt] for utt in utt_list]
+    df['spk'] = [utt2spk[utt] for utt in utt_list]
+    if utt2embedding is not None:
+        df['utt_embedding'] = [utt2embedding[utt] for utt in utt_list]
+    if spk2embedding is not None:
+        df['spk_embedding'] = [spk2embedding[utt2spk[utt]] for utt in utt_list]
+    if utt2speech_token is not None:
+        df['speech_token'] = [utt2speech_token[utt] for utt in utt_list]
+    if utt2instruct is not None:
+        df['instruct'] = [utt2instruct[utt] for utt in utt_list]
     if args.dpo:
-        df['reject_speech_token'] = reject_speech_token_list
+        df['reject_speech_token'] = [utt2reject_speech_token.get(utt, None) for utt in utt_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)
@@ -91,11 +88,19 @@ if __name__ == "__main__":
         for l in f:
             l = l.replace('\n', '').split()
             utt2spk[l[0]] = 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))
+    if os.path.exists('{}/instruct'.format(args.src_dir)):
+        utt2instruct = {}
+        with open('{}/instruct'.format(args.src_dir)) as f:
+            for l in f:
+                l = l.replace('\n', '').split()
+                utt2instruct[l[0]] = ' '.join(l[1:])
+    else:
+        utt2instruct = None
+    utt2embedding = torch.load('{}/utt2embedding.pt'.format(args.src_dir)) if os.path.exists('{}/utt2embedding.pt'.format(args.src_dir)) else None
+    spk2embedding = torch.load('{}/spk2embedding.pt'.format(args.src_dir)) if os.path.exists('{}/spk2embedding.pt'.format(args.src_dir)) else None
+    utt2speech_token = torch.load('{}/utt2speech_token.pt'.format(args.src_dir)) if os.path.exists('{}/utt2speech_token.pt'.format(args.src_dir)) else None
     if args.dpo:
-        utt2reject_speech_token = torch.load('{}_reject/utt2speech_token.pt'.format(args.src_dir))
+        utt2reject_speech_token = torch.load('{}_reject/utt2speech_token.pt'.format(args.src_dir)) if os.path.exists('{}_reject/utt2speech_token.pt'.format(args.src_dir)) else {}
     utts = list(utt2wav.keys())
 
     # Using process pool to speedup

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()