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

Dev/lyuxiang.lx
This commit is contained in:
Xiang Lyu
2025-12-15 12:17:39 +08:00
committed by GitHub
37 changed files with 2394 additions and 514 deletions
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@@ -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
+45 -107
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@@ -2,7 +2,7 @@
## 👉🏻 CosyVoice 👈🏻
**CosyVoice 3.0**: [Demos](https://funaudiollm.github.io/cosyvoice3/); [Paper](https://arxiv.org/abs/2505.17589); [CV3-Eval](https://github.com/FunAudioLLM/CV3-Eval)
**Fun-CosyVoice 3.0**: [Demos](https://funaudiollm.github.io/cosyvoice3/); [Paper](https://arxiv.org/abs/2505.17589); [Modelscope](https://www.modelscope.cn/studios/FunAudioLLM/Fun-CosyVoice3-0.5B); [CV3-Eval](https://github.com/FunAudioLLM/CV3-Eval)
**CosyVoice 2.0**: [Demos](https://funaudiollm.github.io/cosyvoice2/); [Paper](https://arxiv.org/abs/2412.10117); [Modelscope](https://www.modelscope.cn/studios/iic/CosyVoice2-0.5B); [HuggingFace](https://huggingface.co/spaces/FunAudioLLM/CosyVoice2-0.5B)
@@ -10,45 +10,43 @@
## Highlight🔥
**CosyVoice 2.0** has been released! Compared to version 1.0, the new version offers more accurate, more stable, faster, and better speech generation capabilities.
### Multilingual
- **Supported Language**: Chinese, English, Japanese, Korean, Chinese dialects (Cantonese, Sichuanese, Shanghainese, Tianjinese, Wuhanese, etc.)
- **Crosslingual & Mixlingual**:Support zero-shot voice cloning for cross-lingual and code-switching scenarios.
### Ultra-Low Latency
- **Bidirectional Streaming Support**: CosyVoice 2.0 integrates offline and streaming modeling technologies.
- **Rapid First Packet Synthesis**: Achieves latency as low as 150ms while maintaining high-quality audio output.
### High Accuracy
- **Improved Pronunciation**: Reduces pronunciation errors by 30% to 50% compared to CosyVoice 1.0.
- **Benchmark Achievements**: Attains the lowest character error rate on the hard test set of the Seed-TTS evaluation set.
### Strong Stability
- **Consistency in Timbre**: Ensures reliable voice consistency for zero-shot and cross-language speech synthesis.
- **Cross-language Synthesis**: Marked improvements compared to version 1.0.
### Natural Experience
- **Enhanced Prosody and Sound Quality**: Improved alignment of synthesized audio, raising MOS evaluation scores from 5.4 to 5.53.
- **Emotional and Dialectal Flexibility**: Now supports more granular emotional controls and accent adjustments.
**Fun-CosyVoice 3.0** is an advanced text-to-speech (TTS) system based on large language models (LLM), surpassing its predecessor (CosyVoice 2.0) in content consistency, speaker similarity, and prosody naturalness. It is designed for zero-shot multilingual speech synthesis in the wild.
### Key Features
- **Language Coverage**: Covers 9 common languages (Chinese, English, Japanese, Korean, German, Spanish, French, Italian, Russian), 18+ Chinese dialects/accents (Guangdong, Minnan, Sichuan, Dongbei, Shan3xi, Shan1xi, Shanghai, Tianjin, Shan1dong, Ningxia, Gansu, etc.) and meanwhile supports both multi-lingual/cross-lingual zero-shot voice cloning.
- **Content Consistency & Naturalness**: Achieves state-of-the-art performance in content consistency, speaker similarity, and prosody naturalness.
- **Pronunciation Inpainting**: Supports pronunciation inpainting of Chinese Pinyin and English CMU phonemes, providing more controllability and thus suitable for production use.
- **Text Normalization**: Supports reading of numbers, special symbols and various text formats without a traditional frontend module.
- **Bi-Streaming**: Support both text-in streaming and audio-out streaming, and achieves latency as low as 150ms while maintaining high-quality audio output.
- **Instruct Support**: Supports various instructions such as languages, dialects, emotions, speed, volume, etc.
## Roadmap
- [x] 2025/12
- [x] release Fun-CosyVoice3-0.5B-2512 base model, rl model and its training/inference script
- [x] release Fun-CosyVoice3-0.5B modelscope gradio space
- [x] 2025/08
- [x] Thanks to the contribution from NVIDIA Yuekai Zhang, add triton trtllm runtime support and cosyvoice2 grpo training support
- [x] 2025/07
- [x] release cosyvoice 3.0 eval set
- [x] release Fun-CosyVoice 3.0 eval set
- [x] 2025/05
- [x] add cosyvoice 2.0 vllm support
- [x] add CosyVoice2-0.5B vllm support
- [x] 2024/12
- [x] 25hz cosyvoice 2.0 released
- [x] 25hz CosyVoice2-0.5B released
- [x] 2024/09
- [x] 25hz cosyvoice base model
- [x] 25hz cosyvoice voice conversion model
- [x] 25hz CosyVoice-300M base model
- [x] 25hz CosyVoice-300M voice conversion function
- [x] 2024/08
@@ -61,6 +59,25 @@
- [x] WeTextProcessing support when ttsfrd is not available
- [x] Fastapi server and client
## Evaluation
| Model | CER (%) ↓ (test-zh) | WER (%) ↓ (test-en) | CER (%) ↓ (test-hard) |
|-----|------------------|------------------|------------------|
| Human | 1.26 | 2.14 | - |
| F5-TTS | 1.53 | 2.00 | 8.67 |
| SparkTTS | 1.20 | 1.98 | - |
| Seed-TTS | 1.12 | 2.25 | 7.59 |
| CosyVoice2 | 1.45 | 2.57 | 6.83 |
| FireRedTTS-2 | 1.14 | 1.95 | - |
| IndexTTS2 | 1.01 | 1.52 | 7.12 |
| VibeVoice | 1.16 | 3.04 | - |
| HiggsAudio | 1.79 | 2.44 | - |
| MiniMax-Speech | 0.83 | 1.65 | - |
| VoxPCM | 0.93 | 1.85 | 8.87 |
| GLM-TTS | 1.03 | - | - |
| GLM-TTS_RL | 0.89 | - | - |
| Fun-CosyVoice3-0.5B-2512 | 1.21 | 2.24 | 6.71 |
| Fun-CosyVoice3-0.5B-2512_RL | 0.81 | 1.68 | 5.44 |
## Install
@@ -91,11 +108,12 @@
### Model download
We strongly recommend that you download our pretrained `CosyVoice2-0.5B` `CosyVoice-300M` `CosyVoice-300M-SFT` `CosyVoice-300M-Instruct` model and `CosyVoice-ttsfrd` resource.
We strongly recommend that you download our pretrained `Fun-CosyVoice3-0.5B` `CosyVoice2-0.5B` `CosyVoice-300M` `CosyVoice-300M-SFT` `CosyVoice-300M-Instruct` model and `CosyVoice-ttsfrd` resource.
``` python
# SDK模型下载
from modelscope import snapshot_download
snapshot_download('FunAudioLLM/Fun-CosyVoice3-0.5B-2512', local_dir='pretrained_models/Fun-CosyVoice3-0.5B')
snapshot_download('iic/CosyVoice2-0.5B', local_dir='pretrained_models/CosyVoice2-0.5B')
snapshot_download('iic/CosyVoice-300M', local_dir='pretrained_models/CosyVoice-300M')
snapshot_download('iic/CosyVoice-300M-SFT', local_dir='pretrained_models/CosyVoice-300M-SFT')
@@ -103,16 +121,6 @@ snapshot_download('iic/CosyVoice-300M-Instruct', local_dir='pretrained_models/Co
snapshot_download('iic/CosyVoice-ttsfrd', local_dir='pretrained_models/CosyVoice-ttsfrd')
```
``` sh
# git模型下载,请确保已安装git lfs
mkdir -p pretrained_models
git clone https://www.modelscope.cn/iic/CosyVoice2-0.5B.git pretrained_models/CosyVoice2-0.5B
git clone https://www.modelscope.cn/iic/CosyVoice-300M.git pretrained_models/CosyVoice-300M
git clone https://www.modelscope.cn/iic/CosyVoice-300M-SFT.git pretrained_models/CosyVoice-300M-SFT
git clone https://www.modelscope.cn/iic/CosyVoice-300M-Instruct.git pretrained_models/CosyVoice-300M-Instruct
git clone https://www.modelscope.cn/iic/CosyVoice-ttsfrd.git pretrained_models/CosyVoice-ttsfrd
```
Optionally, you can unzip `ttsfrd` resource and install `ttsfrd` package for better text normalization performance.
Notice that this step is not necessary. If you do not install `ttsfrd` package, we will use wetext by default.
@@ -126,50 +134,10 @@ pip install ttsfrd-0.4.2-cp310-cp310-linux_x86_64.whl
### Basic Usage
We strongly recommend using `CosyVoice2-0.5B` for better performance.
Follow the code below for detailed usage of each model.
``` python
import sys
sys.path.append('third_party/Matcha-TTS')
from cosyvoice.cli.cosyvoice import CosyVoice, CosyVoice2
from cosyvoice.utils.file_utils import load_wav
import torchaudio
```
#### CosyVoice2 Usage
```python
cosyvoice = CosyVoice2('pretrained_models/CosyVoice2-0.5B', load_jit=False, load_trt=False, load_vllm=False, fp16=False)
# NOTE if you want to reproduce the results on https://funaudiollm.github.io/cosyvoice2, please add text_frontend=False during inference
# zero_shot usage
prompt_speech_16k = load_wav('./asset/zero_shot_prompt.wav', 16000)
for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物,那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐,笑容如花儿般绽放。', '希望你以后能够做的比我还好呦。', prompt_speech_16k, stream=False)):
torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
# save zero_shot spk for future usage
assert cosyvoice.add_zero_shot_spk('希望你以后能够做的比我还好呦。', prompt_speech_16k, 'my_zero_shot_spk') is True
for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物,那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐,笑容如花儿般绽放。', '', '', zero_shot_spk_id='my_zero_shot_spk', stream=False)):
torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
cosyvoice.save_spkinfo()
# fine grained control, for supported control, check cosyvoice/tokenizer/tokenizer.py#L248
for i, j in enumerate(cosyvoice.inference_cross_lingual('在他讲述那个荒诞故事的过程中,他突然[laughter]停下来,因为他自己也被逗笑了[laughter]。', prompt_speech_16k, stream=False)):
torchaudio.save('fine_grained_control_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
# instruct usage
for i, j in enumerate(cosyvoice.inference_instruct2('收到好友从远方寄来的生日礼物,那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐,笑容如花儿般绽放。', '用四川话说这句话', prompt_speech_16k, stream=False)):
torchaudio.save('instruct_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
# bistream usage, you can use generator as input, this is useful when using text llm model as input
# NOTE you should still have some basic sentence split logic because llm can not handle arbitrary sentence length
def text_generator():
yield '收到好友从远方寄来的生日礼物,'
yield '那份意外的惊喜与深深的祝福'
yield '让我心中充满了甜蜜的快乐,'
yield '笑容如花儿般绽放。'
for i, j in enumerate(cosyvoice.inference_zero_shot(text_generator(), '希望你以后能够做的比我还好呦。', prompt_speech_16k, stream=False)):
torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
We strongly recommend using `Fun-CosyVoice3-0.5B` for better performance.
Follow the code in `example.py` for detailed usage of each model.
```sh
python example.py
```
#### CosyVoice2 vllm Usage
@@ -184,36 +152,6 @@ pip install vllm==v0.9.0 transformers==4.51.3 -i https://mirrors.aliyun.com/pypi
python vllm_example.py
```
#### CosyVoice Usage
```python
cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M-SFT', load_jit=False, load_trt=False, fp16=False)
# sft usage
print(cosyvoice.list_available_spks())
# change stream=True for chunk stream inference
for i, j in enumerate(cosyvoice.inference_sft('你好,我是通义生成式语音大模型,请问有什么可以帮您的吗?', '中文女', stream=False)):
torchaudio.save('sft_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M')
# zero_shot usage, <|zh|><|en|><|jp|><|yue|><|ko|> for Chinese/English/Japanese/Cantonese/Korean
prompt_speech_16k = load_wav('./asset/zero_shot_prompt.wav', 16000)
for i, j in enumerate(cosyvoice.inference_zero_shot('收到好友从远方寄来的生日礼物,那份意外的惊喜与深深的祝福让我心中充满了甜蜜的快乐,笑容如花儿般绽放。', '希望你以后能够做的比我还好呦。', prompt_speech_16k, stream=False)):
torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
# cross_lingual usage
prompt_speech_16k = load_wav('./asset/cross_lingual_prompt.wav', 16000)
for i, j in enumerate(cosyvoice.inference_cross_lingual('<|en|>And then later on, fully acquiring that company. So keeping management in line, interest in line with the asset that\'s coming into the family is a reason why sometimes we don\'t buy the whole thing.', prompt_speech_16k, stream=False)):
torchaudio.save('cross_lingual_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
# vc usage
prompt_speech_16k = load_wav('./asset/zero_shot_prompt.wav', 16000)
source_speech_16k = load_wav('./asset/cross_lingual_prompt.wav', 16000)
for i, j in enumerate(cosyvoice.inference_vc(source_speech_16k, prompt_speech_16k, stream=False)):
torchaudio.save('vc_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M-Instruct')
# instruct usage, support <laughter></laughter><strong></strong>[laughter][breath]
for i, j in enumerate(cosyvoice.inference_instruct('在面对挑战时,他展现了非凡的<strong>勇气</strong>与<strong>智慧</strong>。', '中文男', 'Theo \'Crimson\', is a fiery, passionate rebel leader. Fights with fervor for justice, but struggles with impulsiveness.', stream=False)):
torchaudio.save('instruct_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
```
#### Start web demo
You can use our web demo page to get familiar with CosyVoice quickly.
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@@ -23,8 +23,10 @@ import torch
ROOT_DIR = os.path.dirname(os.path.abspath(__file__))
sys.path.append('{}/../..'.format(ROOT_DIR))
sys.path.append('{}/../../third_party/Matcha-TTS'.format(ROOT_DIR))
from cosyvoice.cli.cosyvoice import CosyVoice, CosyVoice2
from cosyvoice.cli.cosyvoice import AutoModel
from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model, CosyVoice3Model
from cosyvoice.utils.file_utils import logging
from cosyvoice.utils.class_utils import get_model_type
def get_args():
@@ -57,15 +59,17 @@ def main():
torch._C._jit_set_profiling_mode(False)
torch._C._jit_set_profiling_executor(False)
try:
model = CosyVoice(args.model_dir)
except Exception:
try:
model = CosyVoice2(args.model_dir)
except Exception:
raise TypeError('no valid model_type!')
model = AutoModel(model_dir=args.model_dir)
if not isinstance(model, CosyVoice2):
if get_model_type(model.model) == CosyVoiceModel:
# 1. export flow encoder
flow_encoder = model.model.flow.encoder
script = get_optimized_script(flow_encoder)
script.save('{}/flow.encoder.fp32.zip'.format(args.model_dir))
script = get_optimized_script(flow_encoder.half())
script.save('{}/flow.encoder.fp16.zip'.format(args.model_dir))
logging.info('successfully export flow_encoder')
elif get_model_type(model.model) == CosyVoice2Model:
# 1. export llm text_encoder
llm_text_encoder = model.model.llm.text_encoder
script = get_optimized_script(llm_text_encoder)
@@ -90,13 +94,7 @@ def main():
script.save('{}/flow.encoder.fp16.zip'.format(args.model_dir))
logging.info('successfully export flow_encoder')
else:
# 3. export flow encoder
flow_encoder = model.model.flow.encoder
script = get_optimized_script(flow_encoder)
script.save('{}/flow.encoder.fp32.zip'.format(args.model_dir))
script = get_optimized_script(flow_encoder.half())
script.save('{}/flow.encoder.fp16.zip'.format(args.model_dir))
logging.info('successfully export flow_encoder')
raise ValueError('unsupported model type')
if __name__ == '__main__':
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@@ -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
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@@ -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()
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@@ -19,7 +19,7 @@ from hyperpyyaml import load_hyperpyyaml
from modelscope import snapshot_download
import torch
from cosyvoice.cli.frontend import CosyVoiceFrontEnd
from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model
from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model, CosyVoice3Model
from cosyvoice.utils.file_utils import logging
from cosyvoice.utils.class_utils import get_model_type
@@ -27,7 +27,6 @@ from cosyvoice.utils.class_utils import get_model_type
class CosyVoice:
def __init__(self, model_dir, load_jit=False, load_trt=False, fp16=False, trt_concurrent=1):
self.instruct = True if '-Instruct' in model_dir else False
self.model_dir = model_dir
self.fp16 = fp16
if not os.path.exists(model_dir):
@@ -37,7 +36,7 @@ class CosyVoice:
raise ValueError('{} not found!'.format(hyper_yaml_path))
with open(hyper_yaml_path, 'r') as f:
configs = load_hyperpyyaml(f)
assert get_model_type(configs) != CosyVoice2Model, 'do not use {} for CosyVoice initialization!'.format(model_dir)
assert get_model_type(configs) == CosyVoiceModel, 'do not use {} for CosyVoice initialization!'.format(model_dir)
self.frontend = CosyVoiceFrontEnd(configs['get_tokenizer'],
configs['feat_extractor'],
'{}/campplus.onnx'.format(model_dir),
@@ -67,9 +66,9 @@ class CosyVoice:
spks = list(self.frontend.spk2info.keys())
return spks
def add_zero_shot_spk(self, prompt_text, prompt_speech_16k, zero_shot_spk_id):
def add_zero_shot_spk(self, prompt_text, prompt_wav, zero_shot_spk_id):
assert zero_shot_spk_id != '', 'do not use empty zero_shot_spk_id'
model_input = self.frontend.frontend_zero_shot('', prompt_text, prompt_speech_16k, self.sample_rate, '')
model_input = self.frontend.frontend_zero_shot('', prompt_text, prompt_wav, self.sample_rate, '')
del model_input['text']
del model_input['text_len']
self.frontend.spk2info[zero_shot_spk_id] = model_input
@@ -89,12 +88,12 @@ class CosyVoice:
yield model_output
start_time = time.time()
def inference_zero_shot(self, tts_text, prompt_text, prompt_speech_16k, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
def inference_zero_shot(self, tts_text, prompt_text, prompt_wav, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
prompt_text = self.frontend.text_normalize(prompt_text, split=False, text_frontend=text_frontend)
for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
if (not isinstance(i, Generator)) and len(i) < 0.5 * len(prompt_text):
logging.warning('synthesis text {} too short than prompt text {}, this may lead to bad performance'.format(i, prompt_text))
model_input = self.frontend.frontend_zero_shot(i, prompt_text, prompt_speech_16k, self.sample_rate, zero_shot_spk_id)
model_input = self.frontend.frontend_zero_shot(i, prompt_text, prompt_wav, self.sample_rate, zero_shot_spk_id)
start_time = time.time()
logging.info('synthesis text {}'.format(i))
for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
@@ -103,9 +102,9 @@ class CosyVoice:
yield model_output
start_time = time.time()
def inference_cross_lingual(self, tts_text, prompt_speech_16k, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
def inference_cross_lingual(self, tts_text, prompt_wav, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
model_input = self.frontend.frontend_cross_lingual(i, prompt_speech_16k, self.sample_rate, zero_shot_spk_id)
model_input = self.frontend.frontend_cross_lingual(i, prompt_wav, self.sample_rate, zero_shot_spk_id)
start_time = time.time()
logging.info('synthesis text {}'.format(i))
for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
@@ -116,8 +115,6 @@ class CosyVoice:
def inference_instruct(self, tts_text, spk_id, instruct_text, stream=False, speed=1.0, text_frontend=True):
assert isinstance(self.model, CosyVoiceModel), 'inference_instruct is only implemented for CosyVoice!'
if self.instruct is False:
raise ValueError('{} do not support instruct inference'.format(self.model_dir))
instruct_text = self.frontend.text_normalize(instruct_text, split=False, text_frontend=text_frontend)
for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
model_input = self.frontend.frontend_instruct(i, spk_id, instruct_text)
@@ -129,8 +126,8 @@ class CosyVoice:
yield model_output
start_time = time.time()
def inference_vc(self, source_speech_16k, prompt_speech_16k, stream=False, speed=1.0):
model_input = self.frontend.frontend_vc(source_speech_16k, prompt_speech_16k, self.sample_rate)
def inference_vc(self, source_wav, prompt_wav, stream=False, speed=1.0):
model_input = self.frontend.frontend_vc(source_wav, prompt_wav, self.sample_rate)
start_time = time.time()
for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
speech_len = model_output['tts_speech'].shape[1] / self.sample_rate
@@ -142,7 +139,6 @@ class CosyVoice:
class CosyVoice2(CosyVoice):
def __init__(self, model_dir, load_jit=False, load_trt=False, load_vllm=False, fp16=False, trt_concurrent=1):
self.instruct = True if '-Instruct' in model_dir else False
self.model_dir = model_dir
self.fp16 = fp16
if not os.path.exists(model_dir):
@@ -160,9 +156,9 @@ class CosyVoice2(CosyVoice):
'{}/spk2info.pt'.format(model_dir),
configs['allowed_special'])
self.sample_rate = configs['sample_rate']
if torch.cuda.is_available() is False and (load_jit is True or load_trt is True or fp16 is True):
load_jit, load_trt, fp16 = False, False, False
logging.warning('no cuda device, set load_jit/load_trt/fp16 to False')
if torch.cuda.is_available() is False and (load_jit is True or load_trt is True or load_vllm is True or fp16 is True):
load_jit, load_trt, load_vllm, fp16 = False, False, False, False
logging.warning('no cuda device, set load_jit/load_trt/load_vllm/fp16 to False')
self.model = CosyVoice2Model(configs['llm'], configs['flow'], configs['hift'], fp16)
self.model.load('{}/llm.pt'.format(model_dir),
'{}/flow.pt'.format(model_dir),
@@ -178,13 +174,9 @@ class CosyVoice2(CosyVoice):
self.fp16)
del configs
def inference_instruct(self, *args, **kwargs):
raise NotImplementedError('inference_instruct is not implemented for CosyVoice2!')
def inference_instruct2(self, tts_text, instruct_text, prompt_speech_16k, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
assert isinstance(self.model, CosyVoice2Model), 'inference_instruct2 is only implemented for CosyVoice2!'
def inference_instruct2(self, tts_text, instruct_text, prompt_wav, zero_shot_spk_id='', stream=False, speed=1.0, text_frontend=True):
for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
model_input = self.frontend.frontend_instruct2(i, instruct_text, prompt_speech_16k, self.sample_rate, zero_shot_spk_id)
model_input = self.frontend.frontend_instruct2(i, instruct_text, prompt_wav, self.sample_rate, zero_shot_spk_id)
start_time = time.time()
logging.info('synthesis text {}'.format(i))
for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
@@ -192,3 +184,55 @@ class CosyVoice2(CosyVoice):
logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
yield model_output
start_time = time.time()
class CosyVoice3(CosyVoice2):
def __init__(self, model_dir, load_trt=False, load_vllm=False, fp16=False, trt_concurrent=1):
self.model_dir = model_dir
self.fp16 = fp16
if not os.path.exists(model_dir):
model_dir = snapshot_download(model_dir)
hyper_yaml_path = '{}/cosyvoice3.yaml'.format(model_dir)
if not os.path.exists(hyper_yaml_path):
raise ValueError('{} not found!'.format(hyper_yaml_path))
with open(hyper_yaml_path, 'r') as f:
configs = load_hyperpyyaml(f, overrides={'qwen_pretrain_path': os.path.join(model_dir, 'CosyVoice-BlankEN')})
assert get_model_type(configs) == CosyVoice3Model, 'do not use {} for CosyVoice3 initialization!'.format(model_dir)
self.frontend = CosyVoiceFrontEnd(configs['get_tokenizer'],
configs['feat_extractor'],
'{}/campplus.onnx'.format(model_dir),
'{}/speech_tokenizer_v3.onnx'.format(model_dir),
'{}/spk2info.pt'.format(model_dir),
configs['allowed_special'])
self.sample_rate = configs['sample_rate']
if torch.cuda.is_available() is False and (load_trt is True or fp16 is True):
load_trt, fp16 = False, False
logging.warning('no cuda device, set load_trt/fp16 to False')
self.model = CosyVoice3Model(configs['llm'], configs['flow'], configs['hift'], fp16)
self.model.load('{}/llm.pt'.format(model_dir),
'{}/flow.pt'.format(model_dir),
'{}/hift.pt'.format(model_dir))
if load_vllm:
self.model.load_vllm('{}/vllm'.format(model_dir))
if load_trt:
if self.fp16 is True:
logging.warning('DiT tensorRT fp16 engine have some performance issue, use at caution!')
self.model.load_trt('{}/flow.decoder.estimator.{}.mygpu.plan'.format(model_dir, 'fp16' if self.fp16 is True else 'fp32'),
'{}/flow.decoder.estimator.fp32.onnx'.format(model_dir),
trt_concurrent,
self.fp16)
del configs
def AutoModel(**kwargs):
if not os.path.exists(kwargs['model_dir']):
kwargs['model_dir'] = snapshot_download(kwargs['model_dir'])
if os.path.exists('{}/cosyvoice.yaml'.format(kwargs['model_dir'])):
return CosyVoice(**kwargs)
elif os.path.exists('{}/cosyvoice2.yaml'.format(kwargs['model_dir'])):
return CosyVoice2(**kwargs)
elif os.path.exists('{}/cosyvoice3.yaml'.format(kwargs['model_dir'])):
return CosyVoice3(**kwargs)
else:
raise TypeError('No valid model type found!')
+23 -19
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@@ -32,7 +32,7 @@ except ImportError:
from wetext import Normalizer as ZhNormalizer
from wetext import Normalizer as EnNormalizer
use_ttsfrd = False
from cosyvoice.utils.file_utils import logging
from cosyvoice.utils.file_utils import logging, load_wav
from cosyvoice.utils.frontend_utils import contains_chinese, replace_blank, replace_corner_mark, remove_bracket, spell_out_number, split_paragraph, is_only_punctuation
@@ -89,7 +89,8 @@ class CosyVoiceFrontEnd:
for i in range(text_token.shape[1]):
yield text_token[:, i: i + 1]
def _extract_speech_token(self, speech):
def _extract_speech_token(self, prompt_wav):
speech = load_wav(prompt_wav, 16000)
assert speech.shape[1] / 16000 <= 30, 'do not support extract speech token for audio longer than 30s'
feat = whisper.log_mel_spectrogram(speech, n_mels=128)
speech_token = self.speech_tokenizer_session.run(None,
@@ -101,7 +102,8 @@ class CosyVoiceFrontEnd:
speech_token_len = torch.tensor([speech_token.shape[1]], dtype=torch.int32).to(self.device)
return speech_token, speech_token_len
def _extract_spk_embedding(self, speech):
def _extract_spk_embedding(self, prompt_wav):
speech = load_wav(prompt_wav, 16000)
feat = kaldi.fbank(speech,
num_mel_bins=80,
dither=0,
@@ -112,7 +114,8 @@ class CosyVoiceFrontEnd:
embedding = torch.tensor([embedding]).to(self.device)
return embedding
def _extract_speech_feat(self, speech):
def _extract_speech_feat(self, prompt_wav):
speech = load_wav(prompt_wav, 24000)
speech_feat = self.feat_extractor(speech).squeeze(dim=0).transpose(0, 1).to(self.device)
speech_feat = speech_feat.unsqueeze(dim=0)
speech_feat_len = torch.tensor([speech_feat.shape[1]], dtype=torch.int32).to(self.device)
@@ -122,6 +125,9 @@ class CosyVoiceFrontEnd:
if isinstance(text, Generator):
logging.info('get tts_text generator, will skip text_normalize!')
return [text]
# NOTE skip text_frontend when ssml symbol in text
if '<|' in text and '|>' in text:
text_frontend = False
if text_frontend is False or text == '':
return [text] if split is True else text
text = text.strip()
@@ -154,19 +160,18 @@ class CosyVoiceFrontEnd:
model_input = {'text': tts_text_token, 'text_len': tts_text_token_len, 'llm_embedding': embedding, 'flow_embedding': embedding}
return model_input
def frontend_zero_shot(self, tts_text, prompt_text, prompt_speech_16k, resample_rate, zero_shot_spk_id):
def frontend_zero_shot(self, tts_text, prompt_text, prompt_wav, resample_rate, zero_shot_spk_id):
tts_text_token, tts_text_token_len = self._extract_text_token(tts_text)
if zero_shot_spk_id == '':
prompt_text_token, prompt_text_token_len = self._extract_text_token(prompt_text)
prompt_speech_resample = torchaudio.transforms.Resample(orig_freq=16000, new_freq=resample_rate)(prompt_speech_16k)
speech_feat, speech_feat_len = self._extract_speech_feat(prompt_speech_resample)
speech_token, speech_token_len = self._extract_speech_token(prompt_speech_16k)
speech_feat, speech_feat_len = self._extract_speech_feat(prompt_wav)
speech_token, speech_token_len = self._extract_speech_token(prompt_wav)
if resample_rate == 24000:
# cosyvoice2, force speech_feat % speech_token = 2
token_len = min(int(speech_feat.shape[1] / 2), speech_token.shape[1])
speech_feat, speech_feat_len[:] = speech_feat[:, :2 * token_len], 2 * token_len
speech_token, speech_token_len[:] = speech_token[:, :token_len], token_len
embedding = self._extract_spk_embedding(prompt_speech_16k)
embedding = self._extract_spk_embedding(prompt_wav)
model_input = {'prompt_text': prompt_text_token, 'prompt_text_len': prompt_text_token_len,
'llm_prompt_speech_token': speech_token, 'llm_prompt_speech_token_len': speech_token_len,
'flow_prompt_speech_token': speech_token, 'flow_prompt_speech_token_len': speech_token_len,
@@ -178,8 +183,8 @@ class CosyVoiceFrontEnd:
model_input['text_len'] = tts_text_token_len
return model_input
def frontend_cross_lingual(self, tts_text, prompt_speech_16k, resample_rate, zero_shot_spk_id):
model_input = self.frontend_zero_shot(tts_text, '', prompt_speech_16k, resample_rate, zero_shot_spk_id)
def frontend_cross_lingual(self, tts_text, prompt_wav, resample_rate, zero_shot_spk_id):
model_input = self.frontend_zero_shot(tts_text, '', prompt_wav, resample_rate, zero_shot_spk_id)
# in cross lingual mode, we remove prompt in llm
del model_input['prompt_text']
del model_input['prompt_text_len']
@@ -191,22 +196,21 @@ class CosyVoiceFrontEnd:
model_input = self.frontend_sft(tts_text, spk_id)
# in instruct mode, we remove spk_embedding in llm due to information leakage
del model_input['llm_embedding']
instruct_text_token, instruct_text_token_len = self._extract_text_token(instruct_text + '<endofprompt>')
instruct_text_token, instruct_text_token_len = self._extract_text_token(instruct_text)
model_input['prompt_text'] = instruct_text_token
model_input['prompt_text_len'] = instruct_text_token_len
return model_input
def frontend_instruct2(self, tts_text, instruct_text, prompt_speech_16k, resample_rate, zero_shot_spk_id):
model_input = self.frontend_zero_shot(tts_text, instruct_text + '<|endofprompt|>', prompt_speech_16k, resample_rate, zero_shot_spk_id)
def frontend_instruct2(self, tts_text, instruct_text, prompt_wav, resample_rate, zero_shot_spk_id):
model_input = self.frontend_zero_shot(tts_text, instruct_text, prompt_wav, resample_rate, zero_shot_spk_id)
del model_input['llm_prompt_speech_token']
del model_input['llm_prompt_speech_token_len']
return model_input
def frontend_vc(self, source_speech_16k, prompt_speech_16k, resample_rate):
prompt_speech_token, prompt_speech_token_len = self._extract_speech_token(prompt_speech_16k)
prompt_speech_resample = torchaudio.transforms.Resample(orig_freq=16000, new_freq=resample_rate)(prompt_speech_16k)
prompt_speech_feat, prompt_speech_feat_len = self._extract_speech_feat(prompt_speech_resample)
embedding = self._extract_spk_embedding(prompt_speech_16k)
def frontend_vc(self, source_speech_16k, prompt_wav, resample_rate):
prompt_speech_token, prompt_speech_token_len = self._extract_speech_token(prompt_wav)
prompt_speech_feat, prompt_speech_feat_len = self._extract_speech_feat(prompt_wav)
embedding = self._extract_spk_embedding(prompt_wav)
source_speech_token, source_speech_token_len = self._extract_speech_token(source_speech_16k)
model_input = {'source_speech_token': source_speech_token, 'source_speech_token_len': source_speech_token_len,
'flow_prompt_speech_token': prompt_speech_token, 'flow_prompt_speech_token_len': prompt_speech_token_len,
+52 -8
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@@ -38,9 +38,6 @@ class CosyVoiceModel:
self.flow = flow
self.hift = hift
self.fp16 = fp16
if self.fp16 is True:
self.llm.half()
self.flow.half()
self.token_min_hop_len = 2 * self.flow.input_frame_rate
self.token_max_hop_len = 4 * self.flow.input_frame_rate
self.token_overlap_len = 20
@@ -129,7 +126,7 @@ class CosyVoiceModel:
def token2wav(self, token, prompt_token, prompt_feat, embedding, uuid, finalize=False, speed=1.0):
with torch.cuda.amp.autocast(self.fp16):
tts_mel, self.flow_cache_dict[uuid] = self.flow.inference(token=token.to(self.device),
tts_mel, self.flow_cache_dict[uuid] = self.flow.inference(token=token.to(self.device, dtype=torch.int32),
token_len=torch.tensor([token.shape[1]], dtype=torch.int32).to(self.device),
prompt_token=prompt_token.to(self.device),
prompt_token_len=torch.tensor([prompt_token.shape[1]], dtype=torch.int32).to(self.device),
@@ -249,9 +246,6 @@ class CosyVoice2Model(CosyVoiceModel):
self.flow = flow
self.hift = hift
self.fp16 = fp16
if self.fp16 is True:
self.llm.half()
self.flow.half()
# NOTE must matching training static_chunk_size
self.token_hop_len = 25
# hift cache
@@ -284,7 +278,7 @@ class CosyVoice2Model(CosyVoiceModel):
def token2wav(self, token, prompt_token, prompt_feat, embedding, token_offset, uuid, stream=False, finalize=False, speed=1.0):
with torch.cuda.amp.autocast(self.fp16):
tts_mel, _ = self.flow.inference(token=token.to(self.device),
tts_mel, _ = self.flow.inference(token=token.to(self.device, dtype=torch.int32),
token_len=torch.tensor([token.shape[1]], dtype=torch.int32).to(self.device),
prompt_token=prompt_token.to(self.device),
prompt_token_len=torch.tensor([prompt_token.shape[1]], dtype=torch.int32).to(self.device),
@@ -384,3 +378,53 @@ class CosyVoice2Model(CosyVoiceModel):
if torch.cuda.is_available():
torch.cuda.empty_cache()
torch.cuda.current_stream().synchronize()
class CosyVoice3Model(CosyVoice2Model):
def __init__(self,
llm: torch.nn.Module,
flow: torch.nn.Module,
hift: torch.nn.Module,
fp16: bool = False):
self.device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
self.llm = llm
self.flow = flow
self.hift = hift
self.fp16 = fp16
# NOTE must matching training static_chunk_size
self.token_hop_len = 25
# rtf and decoding related
self.llm_context = torch.cuda.stream(torch.cuda.Stream(self.device)) if torch.cuda.is_available() else nullcontext()
self.lock = threading.Lock()
# dict used to store session related variable
self.tts_speech_token_dict = {}
self.llm_end_dict = {}
self.hift_cache_dict = {}
def token2wav(self, token, prompt_token, prompt_feat, embedding, token_offset, uuid, stream=False, finalize=False, speed=1.0):
with torch.cuda.amp.autocast(self.fp16):
tts_mel, _ = self.flow.inference(token=token.to(self.device, dtype=torch.int32),
token_len=torch.tensor([token.shape[1]], dtype=torch.int32).to(self.device),
prompt_token=prompt_token.to(self.device),
prompt_token_len=torch.tensor([prompt_token.shape[1]], dtype=torch.int32).to(self.device),
prompt_feat=prompt_feat.to(self.device),
prompt_feat_len=torch.tensor([prompt_feat.shape[1]], dtype=torch.int32).to(self.device),
embedding=embedding.to(self.device),
streaming=stream,
finalize=finalize)
tts_mel = tts_mel[:, :, token_offset * self.flow.token_mel_ratio:]
# append mel cache
if self.hift_cache_dict[uuid] is not None:
hift_cache_mel = self.hift_cache_dict[uuid]['mel']
tts_mel = torch.concat([hift_cache_mel, tts_mel], dim=2)
self.hift_cache_dict[uuid]['mel'] = tts_mel
else:
self.hift_cache_dict[uuid] = {'mel': tts_mel, 'speech_offset': 0}
if speed != 1.0:
assert token_offset == 0 and finalize is True, 'speed change only support non-stream inference mode'
tts_mel = F.interpolate(tts_mel, size=int(tts_mel.shape[2] / speed), mode='linear')
tts_speech, _ = self.hift.inference(speech_feat=tts_mel, finalize=finalize)
tts_speech = tts_speech[:, self.hift_cache_dict[uuid]['speech_offset']:]
self.hift_cache_dict[uuid]['speech_offset'] += tts_speech.shape[1]
return tts_speech
+9
View File
@@ -242,6 +242,10 @@ def tokenize(data, get_tokenizer, allowed_special, mode='train'):
for sample in data:
assert 'text' in sample
sample['text_token'] = tokenizer.encode(sample['text'], allowed_special=allowed_special)
if 'instruct' in sample:
sample['instruct_token'] = tokenizer.encode(sample['instruct'], allowed_special=allowed_special)
else:
sample['instruct_token'] = tokenizer.encode('', allowed_special=allowed_special)
yield sample
@@ -390,6 +394,9 @@ def padding(data, use_spk_embedding, mode='train', gan=False, dpo=False):
text_token = [torch.tensor(sample[i]['text_token']) for i in order]
text_token_len = torch.tensor([i.size(0) for i in text_token], dtype=torch.int32)
text_token = pad_sequence(text_token, batch_first=True, padding_value=0)
instruct_token = [torch.tensor(sample[i]['instruct_token']) for i in order]
instruct_token_len = torch.tensor([i.size(0) for i in instruct_token], dtype=torch.int32)
instruct_token = pad_sequence(instruct_token, batch_first=True, padding_value=0)
utt_embedding = torch.stack([sample[i]['utt_embedding'] for i in order], dim=0)
spk_embedding = torch.stack([sample[i]['spk_embedding'] for i in order], dim=0)
batch = {
@@ -403,6 +410,8 @@ def padding(data, use_spk_embedding, mode='train', gan=False, dpo=False):
"text": text,
"text_token": text_token,
"text_token_len": text_token_len,
"instruct_token": instruct_token,
"instruct_token_len": instruct_token_len,
"utt_embedding": utt_embedding,
"spk_embedding": spk_embedding,
}
+176
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@@ -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
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"""
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
+159 -8
View File
@@ -37,14 +37,11 @@ class MaskedDiffWithXvec(torch.nn.Module):
'cfm_params': DictConfig({'sigma_min': 1e-06, 'solver': 'euler', 't_scheduler': 'cosine',
'training_cfg_rate': 0.2, 'inference_cfg_rate': 0.7, 'reg_loss_type': 'l1'}),
'decoder_params': {'channels': [256, 256], 'dropout': 0.0, 'attention_head_dim': 64,
'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}},
mel_feat_conf: Dict = {'n_fft': 1024, 'num_mels': 80, 'sampling_rate': 22050,
'hop_size': 256, 'win_size': 1024, 'fmin': 0, 'fmax': 8000}):
'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}}):
super().__init__()
self.input_size = input_size
self.output_size = output_size
self.decoder_conf = decoder_conf
self.mel_feat_conf = mel_feat_conf
self.vocab_size = vocab_size
self.output_type = output_type
self.input_frame_rate = input_frame_rate
@@ -165,14 +162,11 @@ class CausalMaskedDiffWithXvec(torch.nn.Module):
'cfm_params': DictConfig({'sigma_min': 1e-06, 'solver': 'euler', 't_scheduler': 'cosine',
'training_cfg_rate': 0.2, 'inference_cfg_rate': 0.7, 'reg_loss_type': 'l1'}),
'decoder_params': {'channels': [256, 256], 'dropout': 0.0, 'attention_head_dim': 64,
'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}},
mel_feat_conf: Dict = {'n_fft': 1024, 'num_mels': 80, 'sampling_rate': 22050,
'hop_size': 256, 'win_size': 1024, 'fmin': 0, 'fmax': 8000}):
'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}}):
super().__init__()
self.input_size = input_size
self.output_size = output_size
self.decoder_conf = decoder_conf
self.mel_feat_conf = mel_feat_conf
self.vocab_size = vocab_size
self.output_type = output_type
self.input_frame_rate = input_frame_rate
@@ -279,3 +273,160 @@ class CausalMaskedDiffWithXvec(torch.nn.Module):
feat = feat[:, :, mel_len1:]
assert feat.shape[2] == mel_len2
return feat.float(), None
class CausalMaskedDiffWithDiT(torch.nn.Module):
def __init__(self,
input_size: int = 512,
output_size: int = 80,
spk_embed_dim: int = 192,
output_type: str = "mel",
vocab_size: int = 4096,
input_frame_rate: int = 50,
only_mask_loss: bool = True,
token_mel_ratio: int = 2,
pre_lookahead_len: int = 3,
pre_lookahead_layer: torch.nn.Module = None,
decoder: torch.nn.Module = None,
decoder_conf: Dict = {'in_channels': 240, 'out_channel': 80, 'spk_emb_dim': 80, 'n_spks': 1,
'cfm_params': DictConfig({'sigma_min': 1e-06, 'solver': 'euler', 't_scheduler': 'cosine',
'training_cfg_rate': 0.2, 'inference_cfg_rate': 0.7, 'reg_loss_type': 'l1'}),
'decoder_params': {'channels': [256, 256], 'dropout': 0.0, 'attention_head_dim': 64,
'n_blocks': 4, 'num_mid_blocks': 12, 'num_heads': 8, 'act_fn': 'gelu'}}):
super().__init__()
self.input_size = input_size
self.output_size = output_size
self.decoder_conf = decoder_conf
self.vocab_size = vocab_size
self.output_type = output_type
self.input_frame_rate = input_frame_rate
logging.info(f"input frame rate={self.input_frame_rate}")
self.input_embedding = nn.Embedding(vocab_size, input_size)
self.spk_embed_affine_layer = torch.nn.Linear(spk_embed_dim, output_size)
self.pre_lookahead_len = pre_lookahead_len
self.pre_lookahead_layer = pre_lookahead_layer
self.decoder = decoder
self.only_mask_loss = only_mask_loss
self.token_mel_ratio = token_mel_ratio
def forward(
self,
batch: dict,
device: torch.device,
) -> Dict[str, Optional[torch.Tensor]]:
token = batch['speech_token'].to(device)
token_len = batch['speech_token_len'].to(device)
feat = batch['speech_feat'].to(device)
feat_len = batch['speech_feat_len'].to(device)
embedding = batch['embedding'].to(device)
# NOTE unified training, static_chunk_size > 0 or = 0
streaming = True if random.random() < 0.5 else False
# xvec projection
embedding = F.normalize(embedding, dim=1)
embedding = self.spk_embed_affine_layer(embedding)
# concat text and prompt_text
mask = (~make_pad_mask(token_len)).float().unsqueeze(-1).to(device)
token = self.input_embedding(torch.clamp(token, min=0)) * mask
# text encode
h, h_lengths = self.encoder(token, token_len, streaming=streaming)
h = self.encoder_proj(h)
# get conditions
conds = torch.zeros(feat.shape, device=token.device)
for i, j in enumerate(feat_len):
if random.random() < 0.5:
continue
index = random.randint(0, int(0.3 * j))
conds[i, :index] = feat[i, :index]
conds = conds.transpose(1, 2)
mask = (~make_pad_mask(h_lengths.sum(dim=-1).squeeze(dim=1))).to(h)
loss, _ = self.decoder.compute_loss(
feat.transpose(1, 2).contiguous(),
mask.unsqueeze(1),
h.transpose(1, 2).contiguous(),
embedding,
cond=conds,
streaming=streaming,
)
return {'loss': loss}
@torch.inference_mode()
def inference(self,
token,
token_len,
prompt_token,
prompt_token_len,
prompt_feat,
prompt_feat_len,
embedding,
streaming,
finalize):
assert token.shape[0] == 1
# xvec projection
embedding = F.normalize(embedding, dim=1)
embedding = self.spk_embed_affine_layer(embedding)
# concat text and prompt_text
token, token_len = torch.concat([prompt_token, token], dim=1), prompt_token_len + token_len
mask = (~make_pad_mask(token_len)).unsqueeze(-1).to(embedding)
token = self.input_embedding(torch.clamp(token, min=0)) * mask
# text encode
if finalize is True:
h = self.pre_lookahead_layer(token)
else:
h = self.pre_lookahead_layer(token[:, :-self.pre_lookahead_len], context=token[:, -self.pre_lookahead_len:])
h = h.repeat_interleave(self.token_mel_ratio, dim=1)
mel_len1, mel_len2 = prompt_feat.shape[1], h.shape[1] - prompt_feat.shape[1]
# get conditions
conds = torch.zeros([1, mel_len1 + mel_len2, self.output_size], device=token.device).to(h.dtype)
conds[:, :mel_len1] = prompt_feat
conds = conds.transpose(1, 2)
mask = (~make_pad_mask(torch.tensor([mel_len1 + mel_len2]))).to(h)
feat, _ = self.decoder(
mu=h.transpose(1, 2).contiguous(),
mask=mask.unsqueeze(1),
spks=embedding,
cond=conds,
n_timesteps=10,
streaming=streaming
)
feat = feat[:, :, mel_len1:]
assert feat.shape[2] == mel_len2
return feat.float(), None
if __name__ == '__main__':
torch.backends.cudnn.deterministic = True
torch.backends.cudnn.benchmark = False
from hyperpyyaml import load_hyperpyyaml
with open('./pretrained_models/Fun-CosyVoice3-0.5B/cosyvoice3.yaml', 'r') as f:
configs = load_hyperpyyaml(f, overrides={'llm': None, 'hift': None})
model = configs['flow']
device = 'cuda' if torch.cuda.is_available() else 'cpu'
model.to(device)
model.eval()
max_len = 10 * model.decoder.estimator.static_chunk_size
chunk_size = model.decoder.estimator.static_chunk_size
context_size = model.pre_lookahead_layer.pre_lookahead_len
token = torch.randint(0, 6561, size=(1, max_len)).to(device)
token_len = torch.tensor([max_len]).to(device)
prompt_token = torch.randint(0, 6561, size=(1, chunk_size)).to(device)
prompt_token_len = torch.tensor([chunk_size]).to(device)
prompt_feat = torch.rand(1, chunk_size * 2, 80).to(device)
prompt_feat_len = torch.tensor([chunk_size * 2]).to(device)
prompt_embedding = torch.rand(1, 192).to(device)
pred_gt, _ = model.inference(token, token_len, prompt_token, prompt_token_len, prompt_feat, prompt_feat_len, prompt_embedding, streaming=True, finalize=True)
for i in range(0, max_len, chunk_size):
finalize = True if i + chunk_size + context_size >= max_len else False
pred_chunk, _ = model.inference(token[:, :i + chunk_size + context_size], torch.tensor([token[:, :i + chunk_size + context_size].shape[1]]).to(device),
prompt_token, prompt_token_len, prompt_feat, prompt_feat_len, prompt_embedding, streaming=True, finalize=finalize)
pred_chunk = pred_chunk[:, :, i * model.token_mel_ratio:]
print((pred_gt[:, :, i * model.token_mel_ratio: i * model.token_mel_ratio + pred_chunk.shape[2]] - pred_chunk).abs().max().item())
+7 -6
View File
@@ -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
+45
View File
@@ -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))