update requirements.txt

Convert audio to mono while extract speech token

.github/workflows/lint.yml

@@ -0,0 +1,56 @@
+name: Lint
+
+on:
+  pull_request:
+  push:
+
+jobs:
+  quick-checks:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Fetch CosyVoice
+        uses: actions/checkout@v1
+      - name: Checkout PR tip
+        run: |
+          set -eux
+          if [[ "${{ github.event_name }}" == "pull_request" ]]; then
+            # We are on a PR, so actions/checkout leaves us on a merge commit.
+            # Check out the actual tip of the branch.
+            git checkout ${{ github.event.pull_request.head.sha }}
+          fi
+          echo ::set-output name=commit_sha::$(git rev-parse HEAD)
+        id: get_pr_tip
+      - name: Ensure no tabs
+        run: |
+          (! git grep -I -l $'\t' -- . ':(exclude)*.txt' ':(exclude)*.svg' ':(exclude)**Makefile' ':(exclude)**/contrib/**' ':(exclude)third_party' ':(exclude).gitattributes' ':(exclude).gitmodules' || (echo "The above files have tabs; please convert them to spaces"; false))
+      - name: Ensure no trailing whitespace
+        run: |
+          (! git grep -I -n $' $' -- . ':(exclude)*.txt' ':(exclude)third_party' ':(exclude).gitattributes' ':(exclude).gitmodules' || (echo "The above files have trailing whitespace; please remove them"; false))
+
+  flake8-py3:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Setup Python
+        uses: actions/setup-python@v1
+        with:
+          python-version: 3.9
+          architecture: x64
+      - name: Fetch CosyVoice
+        uses: actions/checkout@v1
+      - name: Checkout PR tip
+        run: |
+          set -eux
+          if [[ "${{ github.event_name }}" == "pull_request" ]]; then
+            # We are on a PR, so actions/checkout leaves us on a merge commit.
+            # Check out the actual tip of the branch.
+            git checkout ${{ github.event.pull_request.head.sha }}
+          fi
+          echo ::set-output name=commit_sha::$(git rev-parse HEAD)
+        id: get_pr_tip
+      - name: Run flake8
+        run: |
+          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 --exclude ./third_party/,./runtime/python/grpc/cosyvoice_pb2*py
+          if [ $? != 0 ]; then exit 1; fi

.github/workflows/stale-issues.yml

@@ -0,0 +1,22 @@
+name: Close inactive issues
+on:
+  schedule:
+    - cron: "30 1 * * *"
+
+jobs:
+  close-issues:
+    runs-on: ubuntu-latest
+    permissions:
+      issues: write
+      pull-requests: write
+    steps:
+      - uses: actions/stale@v5
+        with:
+          days-before-issue-stale: 30
+          days-before-issue-close: 14
+          stale-issue-label: "stale"
+          stale-issue-message: "This issue is stale because it has been open for 30 days with no activity."
+          close-issue-message: "This issue was closed because it has been inactive for 14 days since being marked as stale."
+          days-before-pr-stale: -1
+          days-before-pr-close: -1
+          repo-token: ${{ secrets.GITHUB_TOKEN }}

.gitignore

@@ -43,7 +43,10 @@ compile_commands.json
 
 # train/inference files
 *.wav
+*.m4a
+*.aac
 *.pt
 pretrained_models/*
 *_pb2_grpc.py
-*_pb2.py
+*_pb2.py
+*.tar

README.md

@@ -1,38 +1,51 @@
-# CosyVoice
+[![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)
-## 👉🏻 [CosyVoice Demos](https://fun-audio-llm.github.io/) 👈🏻
-[[CosyVoice Paper](https://fun-audio-llm.github.io/pdf/CosyVoice_v1.pdf)][[CosyVoice Studio](https://www.modelscope.cn/studios/iic/CosyVoice-300M)][[CosyVoice Code](https://github.com/FunAudioLLM/CosyVoice)]
 
-For `SenseVoice`, visit [SenseVoice repo](https://github.com/FunAudioLLM/SenseVoice) and [SenseVoice space](https://www.modelscope.cn/studios/iic/SenseVoice).
+## 👉🏻 CosyVoice 👈🏻
+**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 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)
+
+## 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.
 
 ## Roadmap
 
+- [x] 2024/12
+
+    - [x] 25hz cosyvoice 2.0 released
+
+- [x] 2024/09
+
+    - [x] 25hz cosyvoice base model
+    - [x] 25hz cosyvoice voice conversion model
+
+- [x] 2024/08
+
+    - [x] Repetition Aware Sampling(RAS) inference for llm stability
+    - [x] Streaming inference mode support, including kv cache and sdpa for rtf optimization
+
 - [x] 2024/07
 
     - [x] Flow matching training support
-    - [x] WeTextProcessing support when ttsfrd is not avaliable
+    - [x] WeTextProcessing support when ttsfrd is not available
     - [x] Fastapi server and client
 
-- [ ] 2024/08
-
-    - [ ] Repetition Aware Sampling(RAS) inference for llm stability
-    - [ ] Streaming inference mode support, including kv cache and sdpa for rtf optimization
-
-- [ ] 2024/09
-
-    - [ ] 50hz llm model which supports 10 language
-
-- [ ] 2024/10
-
-    - [ ] 50hz llama based llm model which supports lora finetune
-
-- [ ] TBD
-
-    - [ ] Support more instruction mode
-    - [ ] Voice conversion
-    - [ ] Music generation
-    - [ ] Training script sample based on Mandarin
-    - [ ] CosyVoice-500M trained with more multi-lingual data
-    - [ ] More...
 
 ## Install
 
@@ -50,7 +63,7 @@ git submodule update --init --recursive
 - Create Conda env:
 
 ``` sh
-conda create -n cosyvoice python=3.8
+conda create -n cosyvoice -y python=3.10
 conda activate cosyvoice
 # pynini is required by WeTextProcessing, use conda to install it as it can be executed on all platform.
 conda install -y -c conda-forge pynini==2.1.5
@@ -65,14 +78,14 @@ sudo yum install sox sox-devel
 
 **Model download**
 
-We strongly recommend that you download our pretrained `CosyVoice-300M` `CosyVoice-300M-SFT` `CosyVoice-300M-Instruct` model and `CosyVoice-ttsfrd` resource.
+We strongly recommend that you download our pretrained `CosyVoice2-0.5B` `CosyVoice-300M` `CosyVoice-300M-SFT` `CosyVoice-300M-Instruct` model and `CosyVoice-ttsfrd` resource.
-
-If you are expert in this field, and you are only interested in training your own CosyVoice model from scratch, you can skip this step.
 
 ``` python
 # SDK模型下载
 from modelscope import snapshot_download
+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-25Hz', local_dir='pretrained_models/CosyVoice-300M-25Hz')
 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')
@@ -81,64 +94,100 @@ snapshot_download('iic/CosyVoice-ttsfrd', local_dir='pretrained_models/CosyVoice
 ``` 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-25Hz.git pretrained_models/CosyVoice-300M-25Hz
 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
 ```
 
-Optionaly, you can unzip `ttsfrd` resouce and install `ttsfrd` package for better text normalization performance.
+Optionally, you can unzip `ttsfrd` resouce 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 WeTextProcessing by default.
 
 ``` sh
 cd pretrained_models/CosyVoice-ttsfrd/
 unzip resource.zip -d .
-pip install ttsfrd-0.3.6-cp38-cp38-linux_x86_64.whl
+pip install ttsfrd_dependency-0.1-py3-none-any.whl
+pip install ttsfrd-0.4.2-cp310-cp310-linux_x86_64.whl
 ```
 
 **Basic Usage**
 
-For zero_shot/cross_lingual inference, please use `CosyVoice-300M` model.
+We strongly recommend using `CosyVoice2-0.5B` for better performance.
-For sft inference, please use `CosyVoice-300M-SFT` model.
+Follow code below for detailed usage of each model.
-For instruct inference, please use `CosyVoice-300M-Instruct` model.
-First, add `third_party/Matcha-TTS` to your `PYTHONPATH`.
-
-``` sh
-export PYTHONPATH=third_party/Matcha-TTS
-```
 
 ``` python
-from cosyvoice.cli.cosyvoice import CosyVoice
+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
+```
 
-cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M-SFT')
+**CosyVoice2 Usage**
+```python
+cosyvoice = CosyVoice2('pretrained_models/CosyVoice2-0.5B', load_jit=False, load_trt=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)
+
+# 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)
+```
+
+**CosyVoice Usage**
+```python
+cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M-SFT', load_jit=False, load_trt=False, fp16=False)
 # sft usage
-print(cosyvoice.list_avaliable_spks())
+print(cosyvoice.list_available_spks())
-output = 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')
+# change stream=True for chunk stream inference
-torchaudio.save('sft.wav', output['tts_speech'], 22050)
+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')
+cosyvoice = CosyVoice('pretrained_models/CosyVoice-300M') # or change to pretrained_models/CosyVoice-300M-25Hz for 25Hz inference
 # zero_shot usage, <|zh|><|en|><|jp|><|yue|><|ko|> for Chinese/English/Japanese/Cantonese/Korean
-prompt_speech_16k = load_wav('zero_shot_prompt.wav', 16000)
+prompt_speech_16k = load_wav('./asset/zero_shot_prompt.wav', 16000)
-output = 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)
+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', output['tts_speech'], 22050)
+    torchaudio.save('zero_shot_{}.wav'.format(i), j['tts_speech'], cosyvoice.sample_rate)
 # cross_lingual usage
-prompt_speech_16k = load_wav('cross_lingual_prompt.wav', 16000)
+prompt_speech_16k = load_wav('./asset/cross_lingual_prompt.wav', 16000)
-output = 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)
+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', output['tts_speech'], 22050)
+    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]
-output = 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.')
+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', output['tts_speech'], 22050)
+    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.
-We support sft/zero_shot/cross_lingual/instruct inference in web demo.
 
 Please see the demo website for details.
 
@@ -150,12 +199,11 @@ python3 webui.py --port 50000 --model_dir pretrained_models/CosyVoice-300M
 **Advanced Usage**
 
 For advanced user, we have provided train and inference scripts in `examples/libritts/cosyvoice/run.sh`.
-You can get familiar with CosyVoice following this recipie.
 
 **Build for deployment**
 
-Optionally, if you want to use grpc for service deployment,
+Optionally, if you want service deployment,
-you can run following steps. Otherwise, you can just ignore this step.
+you can run following steps.
 
 ``` sh
 cd runtime/python
@@ -163,10 +211,10 @@ docker build -t cosyvoice:v1.0 .
 # change iic/CosyVoice-300M to iic/CosyVoice-300M-Instruct if you want to use instruct inference
 # for grpc usage
 docker run -d --runtime=nvidia -p 50000:50000 cosyvoice:v1.0 /bin/bash -c "cd /opt/CosyVoice/CosyVoice/runtime/python/grpc && python3 server.py --port 50000 --max_conc 4 --model_dir iic/CosyVoice-300M && sleep infinity"
-python3 grpc/client.py --port 50000 --mode <sft|zero_shot|cross_lingual|instruct>
+cd grpc && python3 client.py --port 50000 --mode <sft|zero_shot|cross_lingual|instruct>
 # for fastapi usage
-docker run -d --runtime=nvidia -p 50000:50000 cosyvoice:v1.0 /bin/bash -c "cd /opt/CosyVoice/CosyVoice/runtime/python/fastapi && MODEL_DIR=iic/CosyVoice-300M fastapi dev --port 50000 server.py && sleep infinity"
+docker run -d --runtime=nvidia -p 50000:50000 cosyvoice:v1.0 /bin/bash -c "cd /opt/CosyVoice/CosyVoice/runtime/python/fastapi && python3 server.py --port 50000 --model_dir iic/CosyVoice-300M && sleep infinity"
-python3 fastapi/client.py --port 50000 --mode <sft|zero_shot|cross_lingual|instruct>
+cd fastapi && python3 client.py --port 50000 --mode <sft|zero_shot|cross_lingual|instruct>
 ```
 
 ## Discussion & Communication

cosyvoice/bin/average_model.py

@@ -0,0 +1,92 @@
+# Copyright (c) 2020 Mobvoi Inc (Di Wu)
+# 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.
+
+import os
+import argparse
+import glob
+
+import yaml
+import torch
+
+
+def get_args():
+    parser = argparse.ArgumentParser(description='average model')
+    parser.add_argument('--dst_model', required=True, help='averaged model')
+    parser.add_argument('--src_path',
+                        required=True,
+                        help='src model path for average')
+    parser.add_argument('--val_best',
+                        action="store_true",
+                        help='averaged model')
+    parser.add_argument('--num',
+                        default=5,
+                        type=int,
+                        help='nums for averaged model')
+
+    args = parser.parse_args()
+    print(args)
+    return args
+
+
+def main():
+    args = get_args()
+    val_scores = []
+    if args.val_best:
+        yamls = glob.glob('{}/*.yaml'.format(args.src_path))
+        yamls = [
+            f for f in yamls
+            if not (os.path.basename(f).startswith('train')
+                    or os.path.basename(f).startswith('init'))
+        ]
+        for y in yamls:
+            with open(y, 'r') as f:
+                dic_yaml = yaml.load(f, Loader=yaml.BaseLoader)
+                loss = float(dic_yaml['loss_dict']['loss'])
+                epoch = int(dic_yaml['epoch'])
+                step = int(dic_yaml['step'])
+                tag = dic_yaml['tag']
+                val_scores += [[epoch, step, loss, tag]]
+        sorted_val_scores = sorted(val_scores,
+                                   key=lambda x: x[2],
+                                   reverse=False)
+        print("best val (epoch, step, loss, tag) = " +
+              str(sorted_val_scores[:args.num]))
+        path_list = [
+            args.src_path + '/epoch_{}_whole.pt'.format(score[0])
+            for score in sorted_val_scores[:args.num]
+        ]
+    print(path_list)
+    avg = {}
+    num = args.num
+    assert num == len(path_list)
+    for path in path_list:
+        print('Processing {}'.format(path))
+        states = torch.load(path, map_location=torch.device('cpu'))
+        for k in states.keys():
+            if k not in avg.keys():
+                avg[k] = states[k].clone()
+            else:
+                avg[k] += states[k]
+    # average
+    for k in avg.keys():
+        if avg[k] is not None:
+            # pytorch 1.6 use true_divide instead of /=
+            avg[k] = torch.true_divide(avg[k], num)
+    print('Saving to {}'.format(args.dst_model))
+    torch.save(avg, args.dst_model)
+
+
+if __name__ == '__main__':
+    main()

cosyvoice/bin/export_jit.py

@@ -0,0 +1,91 @@
+# 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 sys
+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
+
+
+def get_args():
+    parser = argparse.ArgumentParser(description='export your model for deployment')
+    parser.add_argument('--model_dir',
+                        type=str,
+                        default='pretrained_models/CosyVoice-300M',
+                        help='local path')
+    args = parser.parse_args()
+    print(args)
+    return args
+
+
+def get_optimized_script(model, preserved_attrs=[]):
+    script = torch.jit.script(model)
+    if preserved_attrs != []:
+        script = torch.jit.freeze(script, preserved_attrs=preserved_attrs)
+    else:
+        script = torch.jit.freeze(script)
+    script = torch.jit.optimize_for_inference(script)
+    return script
+
+
+def main():
+    args = get_args()
+    logging.basicConfig(level=logging.DEBUG,
+                        format='%(asctime)s %(levelname)s %(message)s')
+
+    torch._C._jit_set_fusion_strategy([('STATIC', 1)])
+    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!')
+
+    if not isinstance(model, CosyVoice2):
+        # 1. export llm text_encoder
+        llm_text_encoder = model.model.llm.text_encoder
+        script = get_optimized_script(llm_text_encoder)
+        script.save('{}/llm.text_encoder.fp32.zip'.format(args.model_dir))
+        script = get_optimized_script(llm_text_encoder.half())
+        script.save('{}/llm.text_encoder.fp16.zip'.format(args.model_dir))
+
+        # 2. export llm llm
+        llm_llm = model.model.llm.llm
+        script = get_optimized_script(llm_llm, ['forward_chunk'])
+        script.save('{}/llm.llm.fp32.zip'.format(args.model_dir))
+        script = get_optimized_script(llm_llm.half(), ['forward_chunk'])
+        script.save('{}/llm.llm.fp16.zip'.format(args.model_dir))
+
+    # 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))
+
+
+if __name__ == '__main__':
+    main()

cosyvoice/bin/export_onnx.py

@@ -0,0 +1,116 @@
+# Copyright (c) 2024 Antgroup Inc (authors: Zhoubofan, hexisyztem@icloud.com)
+# 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 sys
+import onnxruntime
+import random
+import torch
+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
+
+
+def get_dummy_input(batch_size, seq_len, out_channels, device):
+    x = torch.rand((batch_size, out_channels, seq_len), dtype=torch.float32, device=device)
+    mask = torch.ones((batch_size, 1, seq_len), dtype=torch.float32, device=device)
+    mu = torch.rand((batch_size, out_channels, seq_len), dtype=torch.float32, device=device)
+    t = torch.rand((batch_size), dtype=torch.float32, device=device)
+    spks = torch.rand((batch_size, out_channels), dtype=torch.float32, device=device)
+    cond = torch.rand((batch_size, out_channels, seq_len), dtype=torch.float32, device=device)
+    return x, mask, mu, t, spks, cond
+
+
+def get_args():
+    parser = argparse.ArgumentParser(description='export your model for deployment')
+    parser.add_argument('--model_dir',
+                        type=str,
+                        default='pretrained_models/CosyVoice-300M',
+                        help='local path')
+    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')
+
+    try:
+        model = CosyVoice(args.model_dir)
+    except Exception:
+        try:
+            model = CosyVoice2(args.model_dir)
+        except Exception:
+            raise TypeError('no valid model_type!')
+
+    # 1. export flow decoder estimator
+    estimator = model.model.flow.decoder.estimator
+
+    device = model.model.device
+    batch_size, seq_len = 2, 256
+    out_channels = model.model.flow.decoder.estimator.out_channels
+    x, mask, mu, t, spks, cond = get_dummy_input(batch_size, seq_len, out_channels, device)
+    torch.onnx.export(
+        estimator,
+        (x, mask, mu, t, spks, cond),
+        '{}/flow.decoder.estimator.fp32.onnx'.format(args.model_dir),
+        export_params=True,
+        opset_version=18,
+        do_constant_folding=True,
+        input_names=['x', 'mask', 'mu', 't', 'spks', 'cond'],
+        output_names=['estimator_out'],
+        dynamic_axes={
+            'x': {2: 'seq_len'},
+            'mask': {2: 'seq_len'},
+            'mu': {2: 'seq_len'},
+            'cond': {2: 'seq_len'},
+            'estimator_out': {2: 'seq_len'},
+        }
+    )
+
+    # 2. test computation consistency
+    option = onnxruntime.SessionOptions()
+    option.graph_optimization_level = onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
+    option.intra_op_num_threads = 1
+    providers = ['CUDAExecutionProvider' if torch.cuda.is_available() else 'CPUExecutionProvider']
+    estimator_onnx = onnxruntime.InferenceSession('{}/flow.decoder.estimator.fp32.onnx'.format(args.model_dir),
+                                                  sess_options=option, providers=providers)
+
+    for _ in tqdm(range(10)):
+        x, mask, mu, t, spks, cond = get_dummy_input(batch_size, random.randint(16, 512), out_channels, device)
+        output_pytorch = estimator(x, mask, mu, t, spks, cond)
+        ort_inputs = {
+            'x': x.cpu().numpy(),
+            'mask': mask.cpu().numpy(),
+            'mu': mu.cpu().numpy(),
+            't': t.cpu().numpy(),
+            'spks': spks.cpu().numpy(),
+            'cond': cond.cpu().numpy()
+        }
+        output_onnx = estimator_onnx.run(None, ort_inputs)[0]
+        torch.testing.assert_allclose(output_pytorch, torch.from_numpy(output_onnx).to(device), rtol=1e-2, atol=1e-4)
+
+
+if __name__ == "__main__":
+    main()

cosyvoice/bin/export_trt.sh

@@ -0,0 +1,10 @@
+#!/bin/bash
+# Copyright 2024 Alibaba Inc. All Rights Reserved.
+# download tensorrt from https://developer.nvidia.com/tensorrt/download/10x, check your system and cuda for compatibability
+# for example for linux + cuda12.4, you can download https://developer.nvidia.com/downloads/compute/machine-learning/tensorrt/10.0.1/tars/TensorRT-10.0.1.6.Linux.x86_64-gnu.cuda-12.4.tar.gz
+TRT_DIR=<YOUR_TRT_DIR>
+MODEL_DIR=<COSYVOICE2_MODEL_DIR>
+
+export LD_LIBRARY_PATH=$LD_LIBRARY_PATH:$TRT_DIR/lib:/usr/local/cuda/lib64
+$TRT_DIR/bin/trtexec --onnx=$MODEL_DIR/flow.decoder.estimator.fp32.onnx --saveEngine=$MODEL_DIR/flow.decoder.estimator.fp32.mygpu.plan --minShapes=x:2x80x4,mask:2x1x4,mu:2x80x4,cond:2x80x4 --optShapes=x:2x80x193,mask:2x1x193,mu:2x80x193,cond:2x80x193 --maxShapes=x:2x80x6800,mask:2x1x6800,mu:2x80x6800,cond:2x80x6800 --inputIOFormats=fp32:chw,fp32:chw,fp32:chw,fp32:chw,fp32:chw,fp32:chw --outputIOFormats=fp32:chw
+$TRT_DIR/bin/trtexec --onnx=$MODEL_DIR/flow.decoder.estimator.fp32.onnx --saveEngine=$MODEL_DIR/flow.decoder.estimator.fp16.mygpu.plan --fp16 --minShapes=x:2x80x4,mask:2x1x4,mu:2x80x4,cond:2x80x4 --optShapes=x:2x80x193,mask:2x1x193,mu:2x80x193,cond:2x80x193 --maxShapes=x:2x80x6800,mask:2x1x6800,mu:2x80x6800,cond:2x80x6800 --inputIOFormats=fp16:chw,fp16:chw,fp16:chw,fp16:chw,fp16:chw,fp16:chw --outputIOFormats=fp16:chw

cosyvoice/bin/inference.py

@@ -18,16 +18,15 @@ 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
-
 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')
@@ -66,7 +65,8 @@ def main():
     model = CosyVoiceModel(configs['llm'], configs['flow'], configs['hift'])
     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_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)
 
     del configs
@@ -74,13 +74,11 @@ def main():
     fn = os.path.join(args.result_dir, 'wav.scp')
     f = open(fn, 'w')
     with torch.no_grad():
-        for batch_idx, batch in tqdm(enumerate(test_data_loader)):
+        for _, batch in tqdm(enumerate(test_data_loader)):
             utts = batch["utts"]
             assert len(utts) == 1, "inference mode only support batchsize 1"
-            text = batch["text"]
             text_token = batch["text_token"].to(device)
             text_token_len = batch["text_token_len"].to(device)
-            tts_text = batch["tts_text"]
             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)
@@ -100,10 +98,13 @@ def main():
                                '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}
-            model_output = model.inference(**model_input)
+            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, model_output['tts_speech'], sample_rate=22050)
+            torchaudio.save(tts_fn, tts_speeches, sample_rate=22050)
             f.write('{} {}\n'.format(tts_key, tts_fn))
             f.flush()
     f.close()

cosyvoice/bin/train.py

@@ -18,6 +18,7 @@ import datetime
 import logging
 logging.getLogger('matplotlib').setLevel(logging.WARNING)
 from copy import deepcopy
+import os
 import torch
 import torch.distributed as dist
 import deepspeed
@@ -67,13 +68,17 @@ def get_args():
                         action='store_true',
                         default=False,
                         help='Use pinned memory buffers used for reading')
+    parser.add_argument('--use_amp',
+                        action='store_true',
+                        default=False,
+                        help='Use automatic mixed precision training')
     parser.add_argument('--deepspeed.save_states',
                         dest='save_states',
                         default='model_only',
                         choices=['model_only', 'model+optimizer'],
                         help='save model/optimizer states')
     parser.add_argument('--timeout',
-                        default=30,
+                        default=60,
                         type=int,
                         help='timeout (in seconds) of cosyvoice_join.')
     parser = deepspeed.add_config_arguments(parser)
@@ -86,10 +91,16 @@ def main():
     args = get_args()
     logging.basicConfig(level=logging.DEBUG,
                         format='%(asctime)s %(levelname)s %(message)s')
+    # gan train has some special initialization logic
+    gan = True if args.model == 'hifigan' else False
 
-    override_dict = {k: None for k in ['llm', 'flow', 'hift'] if k != args.model}
+    override_dict = {k: None for k in ['llm', 'flow', 'hift', 'hifigan'] if k != args.model}
+    if gan is True:
+        override_dict.pop('hift')
     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))
 
     # Init env for ddp
@@ -97,7 +108,7 @@ def main():
 
     # Get dataset & dataloader
     train_dataset, cv_dataset, train_data_loader, cv_data_loader = \
-        init_dataset_and_dataloader(args, configs)
+        init_dataset_and_dataloader(args, configs, gan)
 
     # Do some sanity checks and save config to arsg.model_dir
     configs = check_modify_and_save_config(args, configs)
@@ -107,30 +118,53 @@ def main():
 
     # load checkpoint
     model = configs[args.model]
+    start_step, start_epoch = 0, -1
     if args.checkpoint is not None:
-        model.load_state_dict(torch.load(args.checkpoint, map_location='cpu'))
+        if os.path.exists(args.checkpoint):
+            state_dict = torch.load(args.checkpoint, map_location='cpu')
+            model.load_state_dict(state_dict, strict=False)
+            if 'step' in state_dict:
+                start_step = state_dict['step']
+            if 'epoch' in state_dict:
+                start_epoch = state_dict['epoch']
+        else:
+            logging.warning('checkpoint {} do not exsist!'.format(args.checkpoint))
 
     # Dispatch model from cpu to gpu
     model = wrap_cuda_model(args, model)
 
     # Get optimizer & scheduler
-    model, optimizer, scheduler = init_optimizer_and_scheduler(args, configs, model)
+    model, optimizer, scheduler, optimizer_d, scheduler_d = init_optimizer_and_scheduler(args, configs, model, gan)
+    scheduler.set_step(start_step)
+    if scheduler_d is not None:
+        scheduler_d.set_step(start_step)
 
     # Save init checkpoints
     info_dict = deepcopy(configs['train_conf'])
+    info_dict['step'] = start_step
+    info_dict['epoch'] = start_epoch
     save_model(model, 'init', info_dict)
 
     # Get executor
-    executor = Executor()
+    executor = Executor(gan=gan)
+    executor.step = start_step
 
+    # Init scaler, used for pytorch amp mixed precision training
+    scaler = torch.cuda.amp.GradScaler() if args.use_amp else None
+    print('start step {} start epoch {}'.format(start_step, start_epoch))
     # Start training loop
-    for epoch in range(info_dict['max_epoch']):
+    for epoch in range(start_epoch + 1, info_dict['max_epoch']):
         executor.epoch = epoch
         train_dataset.set_epoch(epoch)
         dist.barrier()
         group_join = dist.new_group(backend="gloo", timeout=datetime.timedelta(seconds=args.timeout))
-        executor.train_one_epoc(model, optimizer, scheduler, train_data_loader, cv_data_loader, writer, info_dict, group_join)
+        if gan is True:
+            executor.train_one_epoc_gan(model, optimizer, scheduler, optimizer_d, scheduler_d, train_data_loader, cv_data_loader,
+                                        writer, info_dict, scaler, group_join)
+        else:
+            executor.train_one_epoc(model, optimizer, scheduler, train_data_loader, cv_data_loader, writer, info_dict, scaler, group_join)
         dist.destroy_process_group(group_join)
 
+
 if __name__ == '__main__':
     main()

cosyvoice/cli/cosyvoice.py

@@ -12,72 +12,173 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 import os
-import torch
+import time
+from typing import Generator
+from tqdm import tqdm
 from hyperpyyaml import load_hyperpyyaml
 from modelscope import snapshot_download
+import torch
 from cosyvoice.cli.frontend import CosyVoiceFrontEnd
-from cosyvoice.cli.model import CosyVoiceModel
+from cosyvoice.cli.model import CosyVoiceModel, CosyVoice2Model
+from cosyvoice.utils.file_utils import logging
+from cosyvoice.utils.class_utils import get_model_type
+
 
 class CosyVoice:
 
-    def __init__(self, model_dir):
+    def __init__(self, model_dir, load_jit=False, load_trt=False, fp16=False):
-        instruct = True if '-Instruct' in model_dir else False
+        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):
             model_dir = snapshot_download(model_dir)
         with open('{}/cosyvoice.yaml'.format(model_dir), 'r') as f:
             configs = load_hyperpyyaml(f)
+        assert get_model_type(configs) != CosyVoice2Model, 'do not use {} for CosyVoice initialization!'.format(model_dir)
         self.frontend = CosyVoiceFrontEnd(configs['get_tokenizer'],
                                           configs['feat_extractor'],
                                           '{}/campplus.onnx'.format(model_dir),
                                           '{}/speech_tokenizer_v1.onnx'.format(model_dir),
                                           '{}/spk2info.pt'.format(model_dir),
-                                          instruct,
                                           configs['allowed_special'])
-        self.model = CosyVoiceModel(configs['llm'], configs['flow'], configs['hift'])
+        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')
+        self.model = CosyVoiceModel(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))
+        self.vllm_codec_engine = None
+        if load_jit:
+            self.model.load_jit('{}/llm.text_encoder.{}.zip'.format(model_dir, 'fp16' if self.fp16 is True else 'fp32'),
+                                '{}/llm.llm.{}.zip'.format(model_dir, 'fp16' if self.fp16 is True else 'fp32'),
+                                '{}/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),
+                                self.fp16)
         del configs
 
-    def list_avaliable_spks(self):
+    def list_available_spks(self):
         spks = list(self.frontend.spk2info.keys())
         return spks
 
-    def inference_sft(self, tts_text, spk_id):
+    def inference_sft(self, tts_text, spk_id, stream=False, speed=1.0, text_frontend=True):
-        tts_speeches = []
+        for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
-        for i in self.frontend.text_normalize(tts_text, split=True):
             model_input = self.frontend.frontend_sft(i, spk_id)
-            model_output = self.model.inference(**model_input)
+            start_time = time.time()
-            tts_speeches.append(model_output['tts_speech'])
+            logging.info('synthesis text {}'.format(i))
-        return {'tts_speech': torch.concat(tts_speeches, dim=1)}
+            for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
+                speech_len = model_output['tts_speech'].shape[1] / self.sample_rate
+                logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
+                yield model_output
+                start_time = time.time()
 
-    def inference_zero_shot(self, tts_text, prompt_text, prompt_speech_16k):
+    def inference_zero_shot(self, tts_text, prompt_text, prompt_speech_16k, stream=False, speed=1.0, text_frontend=True):
-        prompt_text = self.frontend.text_normalize(prompt_text, split=False)
+        prompt_text = self.frontend.text_normalize(prompt_text, split=False, text_frontend=text_frontend)
-        tts_speeches = []
+        for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
-        for i in self.frontend.text_normalize(tts_text, split=True):
+            if (not isinstance(i, Generator)) and len(i) < 0.5 * len(prompt_text):
-            model_input = self.frontend.frontend_zero_shot(i, prompt_text, prompt_speech_16k)
+                logging.warning('synthesis text {} too short than prompt text {}, this may lead to bad performance'.format(i, prompt_text))
-            model_output = self.model.inference(**model_input)
+            model_input = self.frontend.frontend_zero_shot(i, prompt_text, prompt_speech_16k, self.sample_rate)
-            tts_speeches.append(model_output['tts_speech'])
+            start_time = time.time()
-        return {'tts_speech': torch.concat(tts_speeches, dim=1)}
+            logging.info('synthesis text {}'.format(i))
+            for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
+                speech_len = model_output['tts_speech'].shape[1] / self.sample_rate
+                logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
+                yield model_output
+                start_time = time.time()
 
-    def inference_cross_lingual(self, tts_text, prompt_speech_16k):
+    def inference_cross_lingual(self, tts_text, prompt_speech_16k, stream=False, speed=1.0, text_frontend=True):
-        if self.frontend.instruct is True:
+        for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
-            raise ValueError('{} do not support cross_lingual inference'.format(self.model_dir))
+            model_input = self.frontend.frontend_cross_lingual(i, prompt_speech_16k, self.sample_rate)
-        tts_speeches = []
+            start_time = time.time()
-        for i in self.frontend.text_normalize(tts_text, split=True):
+            logging.info('synthesis text {}'.format(i))
-            model_input = self.frontend.frontend_cross_lingual(i, prompt_speech_16k)
+            for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
-            model_output = self.model.inference(**model_input)
+                speech_len = model_output['tts_speech'].shape[1] / self.sample_rate
-            tts_speeches.append(model_output['tts_speech'])
+                logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
-        return {'tts_speech': torch.concat(tts_speeches, dim=1)}
+                yield model_output
+                start_time = time.time()
 
-    def inference_instruct(self, tts_text, spk_id, instruct_text):
+    def inference_instruct(self, tts_text, spk_id, instruct_text, stream=False, speed=1.0, text_frontend=True):
-        if self.frontend.instruct is False:
+        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)
+        instruct_text = self.frontend.text_normalize(instruct_text, split=False, text_frontend=text_frontend)
-        tts_speeches = []
+        for i in tqdm(self.frontend.text_normalize(tts_text, split=True, text_frontend=text_frontend)):
-        for i in self.frontend.text_normalize(tts_text, split=True):
             model_input = self.frontend.frontend_instruct(i, spk_id, instruct_text)
-            model_output = self.model.inference(**model_input)
+            start_time = time.time()
-            tts_speeches.append(model_output['tts_speech'])
+            logging.info('synthesis text {}'.format(i))
-        return {'tts_speech': torch.concat(tts_speeches, dim=1)}
+            for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
+                speech_len = model_output['tts_speech'].shape[1] / self.sample_rate
+                logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
+                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)
+        start_time = time.time()
+        for model_output in self.model.vc(**model_input, stream=stream, speed=speed):
+            speech_len = model_output['tts_speech'].shape[1] / self.sample_rate
+            logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
+            yield model_output
+            start_time = time.time()
+
+
+class CosyVoice2(CosyVoice):
+
+    def __init__(self, model_dir, load_jit=False, load_trt=False, fp16=False, use_vllm=False):
+        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):
+            model_dir = snapshot_download(model_dir)
+        with open('{}/cosyvoice.yaml'.format(model_dir), 'r') as f:
+            configs = load_hyperpyyaml(f, overrides={'qwen_pretrain_path': os.path.join(model_dir, 'CosyVoice-BlankEN')})
+        assert get_model_type(configs) == CosyVoice2Model, 'do not use {} for CosyVoice2 initialization!'.format(model_dir)
+        self.frontend = CosyVoiceFrontEnd(configs['get_tokenizer'],
+                                          configs['feat_extractor'],
+                                          '{}/campplus.onnx'.format(model_dir),
+                                          '{}/speech_tokenizer_v2.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_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')
+        self.model = CosyVoice2Model(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))
+        self.vllm_codec_engine = None
+        if use_vllm:
+            from vllm import EngineArgs, LLMEngine
+            self.model.export_codec_vllm(''.join([model_dir, '/codec_vllm_model']))
+            engine_args = EngineArgs(model=''.join([model_dir, '/codec_vllm_model']),
+                         skip_tokenizer_init=True,
+                         gpu_memory_utilization=0.2)
+            self.vllm_codec_engine = LLMEngine.from_engine_args(engine_args)
+            self.model.vllm_codec_engine = self.vllm_codec_engine
+
+        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),
+                                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, stream=False, speed=1.0, text_frontend=True):
+        assert isinstance(self.model, CosyVoice2Model), 'inference_instruct2 is only implemented for CosyVoice2!'
+        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)
+            start_time = time.time()
+            logging.info('synthesis text {}'.format(i))
+            for model_output in self.model.tts(**model_input, stream=stream, speed=speed):
+                speech_len = model_output['tts_speech'].shape[1] / self.sample_rate
+                logging.info('yield speech len {}, rtf {}'.format(speech_len, (time.time() - start_time) / speech_len))
+                yield model_output
+                start_time = time.time()

cosyvoice/cli/frontend.py

@@ -12,6 +12,8 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 from functools import partial
+from typing import Generator
+import json
 import onnxruntime
 import torch
 import numpy as np
@@ -30,7 +32,8 @@ except ImportError:
     from tn.chinese.normalizer import Normalizer as ZhNormalizer
     from tn.english.normalizer import Normalizer as EnNormalizer
     use_ttsfrd = False
-from cosyvoice.utils.frontend_utils import contains_chinese, replace_blank, replace_corner_mark, remove_bracket, spell_out_number, split_paragraph
+from cosyvoice.utils.file_utils import logging
+from cosyvoice.utils.frontend_utils import contains_chinese, replace_blank, replace_corner_mark, remove_bracket, spell_out_number, split_paragraph, is_only_punctuation
 
 
 class CosyVoiceFrontEnd:
@@ -41,7 +44,6 @@ class CosyVoiceFrontEnd:
                  campplus_model: str,
                  speech_tokenizer_model: str,
                  spk2info: str = '',
-                 instruct: bool = False,
                  allowed_special: str = 'all'):
         self.tokenizer = get_tokenizer()
         self.feat_extractor = feat_extractor
@@ -50,34 +52,51 @@ class CosyVoiceFrontEnd:
         option.graph_optimization_level = onnxruntime.GraphOptimizationLevel.ORT_ENABLE_ALL
         option.intra_op_num_threads = 1
         self.campplus_session = onnxruntime.InferenceSession(campplus_model, sess_options=option, providers=["CPUExecutionProvider"])
-        self.speech_tokenizer_session = onnxruntime.InferenceSession(speech_tokenizer_model, sess_options=option, providers=["CUDAExecutionProvider"if torch.cuda.is_available() else "CPUExecutionProvider"])
+        self.speech_tokenizer_session = onnxruntime.InferenceSession(speech_tokenizer_model, sess_options=option,
+                                                                     providers=["CUDAExecutionProvider" if torch.cuda.is_available() else
+                                                                                "CPUExecutionProvider"])
         if os.path.exists(spk2info):
             self.spk2info = torch.load(spk2info, map_location=self.device)
-        self.instruct = instruct
+        else:
+            self.spk2info = {}
         self.allowed_special = allowed_special
-        self.inflect_parser = inflect.engine()
         self.use_ttsfrd = use_ttsfrd
         if self.use_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'
+            assert self.frd.initialize('{}/../../pretrained_models/CosyVoice-ttsfrd/resource'.format(ROOT_DIR)) is True, \
-            self.frd.set_lang_type('pinyin')
+                'failed to initialize ttsfrd resource'
-            self.frd.enable_pinyin_mix(True)
+            self.frd.set_lang_type('pinyinvg')
-            self.frd.set_breakmodel_index(1)
         else:
-            self.zh_tn_model = ZhNormalizer(remove_erhua=False, full_to_half=False)
+            self.zh_tn_model = ZhNormalizer(remove_erhua=False, full_to_half=False, overwrite_cache=True)
             self.en_tn_model = EnNormalizer()
+            self.inflect_parser = inflect.engine()
 
     def _extract_text_token(self, text):
-        text_token = self.tokenizer.encode(text, allowed_special=self.allowed_special)
+        if isinstance(text, Generator):
-        text_token = torch.tensor([text_token], dtype=torch.int32).to(self.device)
+            logging.info('get tts_text generator, will return _extract_text_token_generator!')
-        text_token_len = torch.tensor([text_token.shape[1]], dtype=torch.int32).to(self.device)
+            # NOTE add a dummy text_token_len for compatibility
-        return text_token, text_token_len
+            return self._extract_text_token_generator(text), torch.tensor([0], dtype=torch.int32).to(self.device)
+        else:
+            text_token = self.tokenizer.encode(text, allowed_special=self.allowed_special)
+            text_token = torch.tensor([text_token], dtype=torch.int32).to(self.device)
+            text_token_len = torch.tensor([text_token.shape[1]], dtype=torch.int32).to(self.device)
+            return text_token, text_token_len
+
+    def _extract_text_token_generator(self, text_generator):
+        for text in text_generator:
+            text_token, _ = self._extract_text_token(text)
+            for i in range(text_token.shape[1]):
+                yield text_token[:, i: i + 1]
 
     def _extract_speech_token(self, speech):
+        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, {self.speech_tokenizer_session.get_inputs()[0].name: feat.detach().cpu().numpy(),
+        speech_token = self.speech_tokenizer_session.run(None,
-                                                                self.speech_tokenizer_session.get_inputs()[1].name: np.array([feat.shape[2]], dtype=np.int32)})[0].flatten().tolist()
+                                                         {self.speech_tokenizer_session.get_inputs()[0].name:
+                                                          feat.detach().cpu().numpy(),
+                                                          self.speech_tokenizer_session.get_inputs()[1].name:
+                                                          np.array([feat.shape[2]], dtype=np.int32)})[0].flatten().tolist()
         speech_token = torch.tensor([speech_token], dtype=torch.int32).to(self.device)
         speech_token_len = torch.tensor([speech_token.shape[1]], dtype=torch.int32).to(self.device)
         return speech_token, speech_token_len
@@ -88,7 +107,8 @@ class CosyVoiceFrontEnd:
                            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()
+        embedding = self.campplus_session.run(None,
+                                              {self.campplus_session.get_inputs()[0].name: feat.unsqueeze(dim=0).cpu().numpy()})[0].flatten().tolist()
         embedding = torch.tensor([embedding]).to(self.device)
         return embedding
 
@@ -98,35 +118,35 @@ class CosyVoiceFrontEnd:
         speech_feat_len = torch.tensor([speech_feat.shape[1]], dtype=torch.int32).to(self.device)
         return speech_feat, speech_feat_len
 
-    def text_normalize(self, text, split=True):
+    def text_normalize(self, text, split=True, text_frontend=True):
+        if isinstance(text, Generator):
+            logging.info('get tts_text generator, will skip text_normalize!')
+            return [text]
+        if text_frontend is False:
+            return [text] if split is True else text
         text = text.strip()
-        if contains_chinese(text):
+        if self.use_ttsfrd:
-            if self.use_ttsfrd:
+            texts = [i["text"] for i in json.loads(self.frd.do_voicegen_frd(text))["sentences"]]
-                text = self.frd.get_frd_extra_info(text, 'input')
+            text = ''.join(texts)
-            else:
-                text = self.zh_tn_model.normalize(text)
-            text = text.replace("\n", "")
-            text = replace_blank(text)
-            text = replace_corner_mark(text)
-            text = text.replace(".", "、")
-            text = text.replace(" - ", "，")
-            text = remove_bracket(text)
-            text = re.sub(r'[，,]+$', '。', text)
-            texts = [i for i in split_paragraph(text, partial(self.tokenizer.encode, allowed_special=self.allowed_special), "zh", token_max_n=80,
-                                                token_min_n=60, merge_len=20,
-                                                comma_split=False)]
         else:
-            if self.use_ttsfrd:
+            if contains_chinese(text):
-                text = self.frd.get_frd_extra_info(text, 'input')
+                text = self.zh_tn_model.normalize(text)
+                text = text.replace("\n", "")
+                text = replace_blank(text)
+                text = replace_corner_mark(text)
+                text = text.replace(".", "。")
+                text = text.replace(" - ", "，")
+                text = remove_bracket(text)
+                text = re.sub(r'[，,、]+$', '。', text)
+                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)
-            text = spell_out_number(text, self.inflect_parser)
+                text = spell_out_number(text, self.inflect_parser)
-            texts = [i for i in split_paragraph(text, partial(self.tokenizer.encode, allowed_special=self.allowed_special), "en", token_max_n=80,
+                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,
+                                             token_min_n=60, merge_len=20, comma_split=False))
-                                                comma_split=False)]
+        texts = [i for i in texts if not is_only_punctuation(i)]
-        if split is False:
+        return texts if split is True else text
-            return text
-        return texts
 
     def frontend_sft(self, tts_text, spk_id):
         tts_text_token, tts_text_token_len = self._extract_text_token(tts_text)
@@ -134,12 +154,17 @@ 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):
+    def frontend_zero_shot(self, tts_text, prompt_text, prompt_speech_16k, resample_rate):
         tts_text_token, tts_text_token_len = self._extract_text_token(tts_text)
         prompt_text_token, prompt_text_token_len = self._extract_text_token(prompt_text)
-        prompt_speech_22050 = torchaudio.transforms.Resample(orig_freq=16000, new_freq=22050)(prompt_speech_16k)
+        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_22050)
+        speech_feat, speech_feat_len = self._extract_speech_feat(prompt_speech_resample)
         speech_token, speech_token_len = self._extract_speech_token(prompt_speech_16k)
+        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)
         model_input = {'text': tts_text_token, 'text_len': tts_text_token_len,
                        'prompt_text': prompt_text_token, 'prompt_text_len': prompt_text_token_len,
@@ -149,8 +174,8 @@ class CosyVoiceFrontEnd:
                        'llm_embedding': embedding, 'flow_embedding': embedding}
         return model_input
 
-    def frontend_cross_lingual(self, tts_text, prompt_speech_16k):
+    def frontend_cross_lingual(self, tts_text, prompt_speech_16k, resample_rate):
-        model_input = self.frontend_zero_shot(tts_text, '', prompt_speech_16k)
+        model_input = self.frontend_zero_shot(tts_text, '', prompt_speech_16k, resample_rate)
         # in cross lingual mode, we remove prompt in llm
         del model_input['prompt_text']
         del model_input['prompt_text_len']
@@ -166,3 +191,21 @@ class CosyVoiceFrontEnd:
         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):
+        model_input = self.frontend_zero_shot(tts_text, instruct_text + '<|endofprompt|>', prompt_speech_16k, resample_rate)
+        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)
+        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,
+                       'prompt_speech_feat': prompt_speech_feat, 'prompt_speech_feat_len': prompt_speech_feat_len,
+                       'flow_embedding': embedding}
+        return model_input

cosyvoice/cli/model.py

@@ -11,50 +11,454 @@
 # 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 os
+from typing import Generator
 import torch
+import numpy as np
+import threading
+import time
+from torch import nn
+from torch.nn import functional as F
+from contextlib import nullcontext
+import uuid
+from cosyvoice.utils.common import fade_in_out
+from cosyvoice.utils.file_utils import convert_onnx_to_trt
+
 
 class CosyVoiceModel:
 
     def __init__(self,
                  llm: torch.nn.Module,
                  flow: torch.nn.Module,
-                 hift: torch.nn.Module):
+                 hift: torch.nn.Module,
+                 fp16: bool):
         self.device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
         self.llm = llm
         self.flow = flow
         self.hift = hift
+        self.fp16 = fp16
+        self.llm.fp16 = fp16
+        self.flow.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
+        # here we fix set flow.decoder.estimator.static_chunk_size = 0 for compatibability
+        self.flow.decoder.estimator.static_chunk_size = 0
+        # mel fade in out
+        self.mel_overlap_len = int(self.token_overlap_len / self.flow.input_frame_rate * 22050 / 256)
+        self.mel_window = np.hamming(2 * self.mel_overlap_len)
+        # hift cache
+        self.mel_cache_len = 20
+        self.source_cache_len = int(self.mel_cache_len * 256)
+        # speech fade in out
+        self.speech_window = np.hamming(2 * self.source_cache_len)
+        # rtf and decoding related
+        self.stream_scale_factor = 1
+        assert self.stream_scale_factor >= 1, 'stream_scale_factor should be greater than 1, change it according to your actual rtf'
+        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.mel_overlap_dict = {}
+        self.flow_cache_dict = {}
+        self.hift_cache_dict = {}
+        self.vllm_codec_engine = None
 
     def load(self, llm_model, flow_model, hift_model):
-        self.llm.load_state_dict(torch.load(llm_model, map_location=self.device))
+        self.llm.load_state_dict(torch.load(llm_model, map_location=self.device), strict=True)
         self.llm.to(self.device).eval()
-        self.flow.load_state_dict(torch.load(flow_model, map_location=self.device))
+        self.flow.load_state_dict(torch.load(flow_model, map_location=self.device), strict=True)
         self.flow.to(self.device).eval()
-        self.hift.load_state_dict(torch.load(hift_model, map_location=self.device))
+        # 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 inference(self, text, text_len, flow_embedding, llm_embedding=torch.zeros(0, 192),
+    def load_jit(self, llm_text_encoder_model, llm_llm_model, flow_encoder_model):
-                  prompt_text=torch.zeros(1, 0, dtype=torch.int32), prompt_text_len=torch.zeros(1, dtype=torch.int32),
+        llm_text_encoder = torch.jit.load(llm_text_encoder_model, map_location=self.device)
-                  llm_prompt_speech_token=torch.zeros(1, 0, dtype=torch.int32), llm_prompt_speech_token_len=torch.zeros(1, dtype=torch.int32),
+        self.llm.text_encoder = llm_text_encoder
-                  flow_prompt_speech_token=torch.zeros(1, 0, dtype=torch.int32), flow_prompt_speech_token_len=torch.zeros(1, dtype=torch.int32),
+        llm_llm = torch.jit.load(llm_llm_model, map_location=self.device)
-                  prompt_speech_feat=torch.zeros(1, 0, 80), prompt_speech_feat_len=torch.zeros(1, dtype=torch.int32)):
+        self.llm.llm = llm_llm
-        tts_speech_token = self.llm.inference(text=text.to(self.device),
+        flow_encoder = torch.jit.load(flow_encoder_model, map_location=self.device)
-                                              text_len=text_len.to(self.device),
+        self.flow.encoder = flow_encoder
-                                              prompt_text=prompt_text.to(self.device),
+
-                                              prompt_text_len=prompt_text_len.to(self.device),
+    def load_trt(self, flow_decoder_estimator_model, flow_decoder_onnx_model, fp16):
-                                              prompt_speech_token=llm_prompt_speech_token.to(self.device),
+        assert torch.cuda.is_available(), 'tensorrt only supports gpu!'
-                                              prompt_speech_token_len=llm_prompt_speech_token_len.to(self.device),
+        if not os.path.exists(flow_decoder_estimator_model):
-                                              embedding=llm_embedding.to(self.device),
+            convert_onnx_to_trt(flow_decoder_estimator_model, flow_decoder_onnx_model, fp16)
-                                              beam_size=1,
+        if os.path.getsize(flow_decoder_estimator_model) == 0:
-                                              sampling=25,
+            raise ValueError('{} is empty file, delete it and export again!'.format(flow_decoder_estimator_model))
-                                              max_token_text_ratio=30,
+        del self.flow.decoder.estimator
-                                              min_token_text_ratio=3)
+        import tensorrt as trt
-        tts_mel = self.flow.inference(token=tts_speech_token,
+        with open(flow_decoder_estimator_model, 'rb') as f:
-                                      token_len=torch.tensor([tts_speech_token.size(1)], dtype=torch.int32).to(self.device),
+            self.flow.decoder.estimator_engine = trt.Runtime(trt.Logger(trt.Logger.INFO)).deserialize_cuda_engine(f.read())
-                                      prompt_token=flow_prompt_speech_token.to(self.device),
+        if self.flow.decoder.estimator_engine is None:
-                                      prompt_token_len=flow_prompt_speech_token_len.to(self.device),
+            raise ValueError('failed to load trt {}'.format(flow_decoder_estimator_model))
-                                      prompt_feat=prompt_speech_feat.to(self.device),
+        self.flow.decoder.estimator = self.flow.decoder.estimator_engine.create_execution_context()
-                                      prompt_feat_len=prompt_speech_feat_len.to(self.device),
+
-                                      embedding=flow_embedding.to(self.device))
+    def llm_job(self, text, prompt_text, llm_prompt_speech_token, llm_embedding, uuid):
-        tts_speech = self.hift.inference(mel=tts_mel).cpu()
+        with self.llm_context:
+            if isinstance(text, Generator):
+                assert isinstance(self, CosyVoice2Model), 'streaming input text is only implemented for CosyVoice2!'
+                if self.vllm_codec_engine is None:
+                    for i in 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)):
+                        self.tts_speech_token_dict[uuid].append(i)
+                else:
+                    for i in self.llm.inference_bistream_vllm(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),
+                                                        vllm_codec_engine=self.vllm_codec_engine):
+                        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),
+                                            vllm_codec_engine=self.vllm_codec_engine):
+                    self.tts_speech_token_dict[uuid].append(i)
+        self.llm_end_dict[uuid] = True
+
+    def token2wav(self, token, prompt_token, prompt_feat, embedding, uuid, finalize=False, speed=1.0):
+        tts_mel, flow_cache = 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),
+                                                  flow_cache=self.flow_cache_dict[uuid])
+        self.flow_cache_dict[uuid] = flow_cache
+
+        # mel overlap fade in out
+        if self.mel_overlap_dict[uuid].shape[2] != 0:
+            tts_mel = fade_in_out(tts_mel, self.mel_overlap_dict[uuid], self.mel_window)
+        # 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:
+            self.mel_overlap_dict[uuid] = tts_mel[:, :, -self.mel_overlap_len:]
+            tts_mel = tts_mel[:, :, :-self.mel_overlap_len]
+            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
+
+    def tts(self, text, flow_embedding, llm_embedding=torch.zeros(0, 192),
+            prompt_text=torch.zeros(1, 0, dtype=torch.int32),
+            llm_prompt_speech_token=torch.zeros(1, 0, dtype=torch.int32),
+            flow_prompt_speech_token=torch.zeros(1, 0, dtype=torch.int32),
+            prompt_speech_feat=torch.zeros(1, 0, 80), stream=False, speed=1.0, **kwargs):
+        # this_uuid is used to track variables related to this inference thread
+        this_uuid = str(uuid.uuid1())
+        with self.lock:
+            self.tts_speech_token_dict[this_uuid], self.llm_end_dict[this_uuid] = [], False
+            self.hift_cache_dict[this_uuid] = None
+            self.mel_overlap_dict[this_uuid] = torch.zeros(1, 80, 0)
+            self.flow_cache_dict[this_uuid] = torch.zeros(1, 80, 0, 2)
+        p = threading.Thread(target=self.llm_job, args=(text, prompt_text, llm_prompt_speech_token, llm_embedding, this_uuid))
+        p.start()
+        if stream is True:
+            token_hop_len = self.token_min_hop_len
+            while True:
+                time.sleep(0.1)
+                if len(self.tts_speech_token_dict[this_uuid]) >= token_hop_len + self.token_overlap_len:
+                    this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid][:token_hop_len + self.token_overlap_len]) \
+                        .unsqueeze(dim=0)
+                    this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                                     prompt_token=flow_prompt_speech_token,
+                                                     prompt_feat=prompt_speech_feat,
+                                                     embedding=flow_embedding,
+                                                     uuid=this_uuid,
+                                                     finalize=False)
+                    yield {'tts_speech': this_tts_speech.cpu()}
+                    with self.lock:
+                        self.tts_speech_token_dict[this_uuid] = self.tts_speech_token_dict[this_uuid][token_hop_len:]
+                    # increase token_hop_len for better speech quality
+                    token_hop_len = min(self.token_max_hop_len, int(token_hop_len * self.stream_scale_factor))
+                if self.llm_end_dict[this_uuid] is True and len(self.tts_speech_token_dict[this_uuid]) < token_hop_len + self.token_overlap_len:
+                    break
+            p.join()
+            # deal with remain tokens, make sure inference remain token len equals token_hop_len when cache_speech is not None
+            this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid]).unsqueeze(dim=0)
+            this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                             prompt_token=flow_prompt_speech_token,
+                                             prompt_feat=prompt_speech_feat,
+                                             embedding=flow_embedding,
+                                             uuid=this_uuid,
+                                             finalize=True)
+            yield {'tts_speech': this_tts_speech.cpu()}
+        else:
+            # deal with all tokens
+            p.join()
+            this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid]).unsqueeze(dim=0)
+            this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                             prompt_token=flow_prompt_speech_token,
+                                             prompt_feat=prompt_speech_feat,
+                                             embedding=flow_embedding,
+                                             uuid=this_uuid,
+                                             finalize=True,
+                                             speed=speed)
+            yield {'tts_speech': this_tts_speech.cpu()}
+        with self.lock:
+            self.tts_speech_token_dict.pop(this_uuid)
+            self.llm_end_dict.pop(this_uuid)
+            self.mel_overlap_dict.pop(this_uuid)
+            self.hift_cache_dict.pop(this_uuid)
+            self.flow_cache_dict.pop(this_uuid)
+        torch.cuda.empty_cache()
+
+    def vc(self, source_speech_token, flow_prompt_speech_token, prompt_speech_feat, flow_embedding, stream=False, speed=1.0, **kwargs):
+        # this_uuid is used to track variables related to this inference thread
+        this_uuid = str(uuid.uuid1())
+        with self.lock:
+            self.tts_speech_token_dict[this_uuid], self.llm_end_dict[this_uuid] = source_speech_token.flatten().tolist(), True
+            self.hift_cache_dict[this_uuid] = None
+            self.mel_overlap_dict[this_uuid] = torch.zeros(1, 80, 0)
+            self.flow_cache_dict[this_uuid] = torch.zeros(1, 80, 0, 2)
+        if stream is True:
+            token_hop_len = self.token_min_hop_len
+            while True:
+                if len(self.tts_speech_token_dict[this_uuid]) >= token_hop_len + self.token_overlap_len:
+                    this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid][:token_hop_len + self.token_overlap_len]) \
+                        .unsqueeze(dim=0)
+                    this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                                     prompt_token=flow_prompt_speech_token,
+                                                     prompt_feat=prompt_speech_feat,
+                                                     embedding=flow_embedding,
+                                                     uuid=this_uuid,
+                                                     finalize=False)
+                    yield {'tts_speech': this_tts_speech.cpu()}
+                    with self.lock:
+                        self.tts_speech_token_dict[this_uuid] = self.tts_speech_token_dict[this_uuid][token_hop_len:]
+                    # increase token_hop_len for better speech quality
+                    token_hop_len = min(self.token_max_hop_len, int(token_hop_len * self.stream_scale_factor))
+                if self.llm_end_dict[this_uuid] is True and len(self.tts_speech_token_dict[this_uuid]) < token_hop_len + self.token_overlap_len:
+                    break
+            # deal with remain tokens, make sure inference remain token len equals token_hop_len when cache_speech is not None
+            this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid]).unsqueeze(dim=0)
+            this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                             prompt_token=flow_prompt_speech_token,
+                                             prompt_feat=prompt_speech_feat,
+                                             embedding=flow_embedding,
+                                             uuid=this_uuid,
+                                             finalize=True)
+            yield {'tts_speech': this_tts_speech.cpu()}
+        else:
+            # deal with all tokens
+            this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid]).unsqueeze(dim=0)
+            this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                             prompt_token=flow_prompt_speech_token,
+                                             prompt_feat=prompt_speech_feat,
+                                             embedding=flow_embedding,
+                                             uuid=this_uuid,
+                                             finalize=True,
+                                             speed=speed)
+            yield {'tts_speech': this_tts_speech.cpu()}
+        with self.lock:
+            self.tts_speech_token_dict.pop(this_uuid)
+            self.llm_end_dict.pop(this_uuid)
+            self.mel_overlap_dict.pop(this_uuid)
+            self.hift_cache_dict.pop(this_uuid)
+        torch.cuda.empty_cache()
+
+
+class CosyVoice2Model(CosyVoiceModel):
+
+    def __init__(self,
+                 llm: torch.nn.Module,
+                 flow: torch.nn.Module,
+                 hift: torch.nn.Module,
+                 fp16: bool):
+        self.device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
+        self.llm = llm
+        self.flow = flow
+        self.hift = hift
+        self.fp16 = fp16
+        self.llm.fp16 = fp16
+        self.flow.fp16 = fp16
+        if self.fp16 is True:
+            self.llm.half()
+            self.flow.half()
+        self.token_hop_len = 2 * self.flow.input_frame_rate
+        # here we fix flow encoder/decoder decoding_chunk_size, in the future we will send it as arguments, or use cache
+        self.flow.encoder.static_chunk_size = 2 * self.flow.input_frame_rate
+        self.flow.decoder.estimator.static_chunk_size = 2 * self.flow.input_frame_rate * self.flow.token_mel_ratio
+        # hift cache
+        self.mel_cache_len = 8
+        self.source_cache_len = int(self.mel_cache_len * 480)
+        # speech fade in out
+        self.speech_window = np.hamming(2 * self.source_cache_len)
+        # rtf and decoding related
+        self.stream_scale_factor = 1
+        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 = {}
+        self.vllm_codec_engine = 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 export_codec_vllm(self, model_path):
+        if os.path.exists(model_path):
+            return
+        pad_to = DEFAULT_VOCAB_PADDING_SIZE = 64
+        vocab_size = self.llm.speech_embedding.num_embeddings
+        feature_size = self.llm.speech_embedding.embedding_dim
+        pad_vocab_size = ((vocab_size + pad_to - 1) // pad_to) * pad_to
+
+        dtype = torch.bfloat16
+        # lm_head
+        new_lm_head = nn.Linear(in_features=feature_size, out_features=pad_vocab_size, bias=True)
+        with torch.no_grad():
+            new_lm_head.weight[:vocab_size] = self.llm.llm_decoder.weight
+            new_lm_head.bias[:vocab_size] = self.llm.llm_decoder.bias
+            new_lm_head.weight[vocab_size:] = 0
+            new_lm_head.bias[vocab_size:] = 0
+        self.llm.llm.model.lm_head = new_lm_head
+        new_codec_embed = nn.Linear(in_features=feature_size, out_features=pad_vocab_size)
+        # embed_tokens
+        embed_tokens = self.llm.llm.model.model.embed_tokens
+        with torch.no_grad():
+            new_codec_embed.weight[:vocab_size] = self.llm.speech_embedding.weight
+            new_codec_embed.weight[vocab_size:] = 0
+        self.llm.llm.model.set_input_embeddings(new_codec_embed)
+        self.llm.llm.model.to(self.device)
+        self.llm.llm.model.to(dtype)
+        tmp_vocab_size = self.llm.llm.model.config.vocab_size
+        tmp_tie_embedding = self.llm.llm.model.config.tie_word_embeddings
+        del self.llm.llm.model.generation_config.eos_token_id
+        del self.llm.llm.model.config.bos_token_id
+        del self.llm.llm.model.config.eos_token_id
+        self.llm.llm.model.config.vocab_size = pad_vocab_size
+        self.llm.llm.model.config.tie_word_embeddings = False
+        self.llm.llm.model.config.use_bias = True
+        self.llm.llm.model.save_pretrained(model_path)
+        self.llm.llm.model.config.vocab_size = tmp_vocab_size
+        self.llm.llm.model.config.tie_word_embeddings = tmp_tie_embedding
+        self.llm.llm.model.set_input_embeddings(embed_tokens)
+
+    def token2wav(self, token, prompt_token, prompt_feat, embedding, uuid, token_offset, finalize=False, speed=1.0):
+        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),
+                                         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
+
+    def tts(self, text, flow_embedding, llm_embedding=torch.zeros(0, 192),
+            prompt_text=torch.zeros(1, 0, dtype=torch.int32),
+            llm_prompt_speech_token=torch.zeros(1, 0, dtype=torch.int32),
+            flow_prompt_speech_token=torch.zeros(1, 0, dtype=torch.int32),
+            prompt_speech_feat=torch.zeros(1, 0, 80), stream=False, speed=1.0, **kwargs):
+        # this_uuid is used to track variables related to this inference thread
+        this_uuid = str(uuid.uuid1())
+        with self.lock:
+            self.tts_speech_token_dict[this_uuid], self.llm_end_dict[this_uuid] = [], False
+            self.hift_cache_dict[this_uuid] = None
+        p = threading.Thread(target=self.llm_job, args=(text, prompt_text, llm_prompt_speech_token, llm_embedding, this_uuid))
+        p.start()
+        if stream is True:
+            token_offset = 0
+            while True:
+                time.sleep(0.1)
+                if len(self.tts_speech_token_dict[this_uuid]) - token_offset >= self.token_hop_len + self.flow.pre_lookahead_len:
+                    this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid][:token_offset + self.token_hop_len + self.flow.pre_lookahead_len]).unsqueeze(dim=0)
+                    this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                                     prompt_token=flow_prompt_speech_token,
+                                                     prompt_feat=prompt_speech_feat,
+                                                     embedding=flow_embedding,
+                                                     uuid=this_uuid,
+                                                     token_offset=token_offset,
+                                                     finalize=False)
+                    token_offset += self.token_hop_len
+                    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 < self.token_hop_len + self.flow.pre_lookahead_len:
+                    break
+            p.join()
+            # deal with remain tokens, make sure inference remain token len equals token_hop_len when cache_speech is not None
+            this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid]).unsqueeze(dim=0)
+            this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                             prompt_token=flow_prompt_speech_token,
+                                             prompt_feat=prompt_speech_feat,
+                                             embedding=flow_embedding,
+                                             uuid=this_uuid,
+                                             token_offset=token_offset,
+                                             finalize=True)
+            yield {'tts_speech': this_tts_speech.cpu()}
+        else:
+            # deal with all tokens
+            p.join()
+            this_tts_speech_token = torch.tensor(self.tts_speech_token_dict[this_uuid]).unsqueeze(dim=0)
+            this_tts_speech = self.token2wav(token=this_tts_speech_token,
+                                             prompt_token=flow_prompt_speech_token,
+                                             prompt_feat=prompt_speech_feat,
+                                             embedding=flow_embedding,
+                                             uuid=this_uuid,
+                                             token_offset=0,
+                                             finalize=True,
+                                             speed=speed)
+            yield {'tts_speech': this_tts_speech.cpu()}
+        with self.lock:
+            self.tts_speech_token_dict.pop(this_uuid)
+            self.llm_end_dict.pop(this_uuid)
         torch.cuda.empty_cache()
-        return {'tts_speech': tts_speech}

cosyvoice/dataset/dataset.py

@@ -126,6 +126,7 @@ class DataList(IterableDataset):
 def Dataset(data_list_file,
             data_pipeline,
             mode='train',
+            gan=False,
             shuffle=True,
             partition=True,
             tts_file='',
@@ -148,13 +149,16 @@ def Dataset(data_list_file,
             tts_data = json.load(f)
         utt2lists = read_json_lists(prompt_utt2data)
         # filter unnecessary file in inference mode
-        lists = list(set([utt2lists[utt] for utt in tts_data.keys() if utt2lists[utt] in lists]))
+        lists = list({utt2lists[utt] for utt in tts_data.keys() if utt2lists[utt] in lists})
     dataset = DataList(lists,
                        shuffle=shuffle,
                        partition=partition)
     if mode == 'inference':
-        # map partial arg tts_data in inference mode
+        # map partial arg to parquet_opener func in inference mode
         data_pipeline[0] = partial(data_pipeline[0], tts_data=tts_data)
+    if gan is True:
+        # map partial arg to padding func in gan mode
+        data_pipeline[-1] = partial(data_pipeline[-1], gan=gan)
     for func in data_pipeline:
         dataset = Processor(dataset, func, mode=mode)
     return dataset

cosyvoice/dataset/processor.py

@@ -20,10 +20,10 @@ import torch
 import torchaudio
 from torch.nn.utils.rnn import pad_sequence
 import torch.nn.functional as F
+import pyworld as pw
 
-torchaudio.set_audio_backend('soundfile')
 
-AUDIO_FORMAT_SETS = set(['flac', 'mp3', 'm4a', 'ogg', 'opus', 'wav', 'wma'])
+AUDIO_FORMAT_SETS = {'flac', 'mp3', 'm4a', 'ogg', 'opus', 'wav', 'wma'}
 
 
 def parquet_opener(data, mode='train', tts_data={}):
@@ -40,20 +40,22 @@ def parquet_opener(data, mode='train', tts_data={}):
         assert 'src' in sample
         url = sample['src']
         try:
-            df = pq.read_table(url).to_pandas()
+            for df in pq.ParquetFile(url).iter_batches(batch_size=64):
-            for i in range(len(df)):
+                df = df.to_pandas()
-                if mode == 'inference' and df.loc[i, 'utt'] not in tts_data:
+                for i in range(len(df)):
-                    continue
+                    if mode == 'inference' and df.loc[i, 'utt'] not in tts_data:
-                sample.update(dict(df.loc[i]))
+                        continue
-                if mode == 'train':
+                    sample.update(dict(df.loc[i]))
-                    # NOTE do not return sample directly, must initialize a new dict
+                    if mode == 'train':
-                    yield {**sample}
+                        # NOTE do not return sample directly, must initialize a new dict
-                else:
+                        yield {**sample}
-                    for index, text in enumerate(tts_data[df.loc[i, 'utt']]):
+                    else:
-                        yield {**sample, 'tts_index': index, 'tts_text': text}
+                        for index, text in enumerate(tts_data[df.loc[i, 'utt']]):
+                            yield {**sample, 'tts_index': index, 'tts_text': text}
         except Exception as ex:
             logging.warning('Failed to open {}, ex info {}'.format(url, ex))
 
+
 def filter(data,
            max_length=10240,
            min_length=10,
@@ -84,6 +86,7 @@ def filter(data,
     """
     for sample in data:
         sample['speech'], sample['sample_rate'] = torchaudio.load(BytesIO(sample['audio_data']))
+        sample['speech'] = sample['speech'].mean(dim=0, keepdim=True)
         del sample['audio_data']
         # sample['wav'] is torch.Tensor, we have 100 frames every second
         num_frames = sample['speech'].size(1) / sample['sample_rate'] * 100
@@ -133,6 +136,27 @@ def resample(data, resample_rate=22050, min_sample_rate=16000, mode='train'):
         yield sample
 
 
+def truncate(data, truncate_length=24576, mode='train'):
+    """ Truncate data.
+
+        Args:
+            data: Iterable[{key, wav, label, sample_rate}]
+            truncate_length: truncate length
+
+        Returns:
+            Iterable[{key, wav, label, sample_rate}]
+    """
+    for sample in data:
+        waveform = sample['speech']
+        if waveform.shape[1] > truncate_length:
+            start = random.randint(0, waveform.shape[1] - truncate_length)
+            waveform = waveform[:, start: start + truncate_length]
+        else:
+            waveform = torch.concat([waveform, torch.zeros(1, truncate_length - waveform.shape[1])], dim=1)
+        sample['speech'] = waveform
+        yield sample
+
+
 def compute_fbank(data,
                   feat_extractor,
                   mode='train'):
@@ -152,7 +176,31 @@ def compute_fbank(data,
         waveform = sample['speech']
         mat = feat_extractor(waveform).squeeze(dim=0).transpose(0, 1)
         sample['speech_feat'] = mat
-        del sample['speech']
+        yield sample
+
+
+def compute_f0(data, sample_rate, hop_size, mode='train'):
+    """ Extract f0
+
+        Args:
+            data: Iterable[{key, wav, label, sample_rate}]
+
+        Returns:
+            Iterable[{key, feat, label}]
+    """
+    frame_period = hop_size * 1000 / sample_rate
+    for sample in data:
+        assert 'sample_rate' in sample
+        assert 'speech' in sample
+        assert 'utt' in sample
+        assert 'text_token' in sample
+        waveform = sample['speech']
+        _f0, t = pw.harvest(waveform.squeeze(dim=0).numpy().astype('double'), sample_rate, frame_period=frame_period)
+        if sum(_f0 != 0) < 5: # this happens when the algorithm fails
+            _f0, t = pw.dio(waveform.squeeze(dim=0).numpy().astype('double'), sample_rate, frame_period=frame_period) # if harvest fails, try dio
+        f0 = pw.stonemask(waveform.squeeze(dim=0).numpy().astype('double'), _f0, t, sample_rate)
+        f0 = F.interpolate(torch.from_numpy(f0).view(1, 1, -1), size=sample['speech_feat'].shape[0], mode='linear').view(-1)
+        sample['pitch_feat'] = f0
         yield sample
 
 
@@ -308,7 +356,7 @@ def batch(data, batch_type='static', batch_size=16, max_frames_in_batch=12000, m
             logging.fatal('Unsupported batch type {}'.format(batch_type))
 
 
-def padding(data, use_spk_embedding, mode='train'):
+def padding(data, use_spk_embedding, mode='train', gan=False):
     """ Padding the data into training data
 
         Args:
@@ -324,6 +372,9 @@ def padding(data, use_spk_embedding, mode='train'):
         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,
@@ -342,6 +393,8 @@ def padding(data, use_spk_embedding, mode='train'):
         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,
@@ -352,6 +405,19 @@ def padding(data, use_spk_embedding, mode='train'):
             "utt_embedding": utt_embedding,
             "spk_embedding": spk_embedding,
         }
+        if gan is True:
+            # in gan train, we need pitch_feat
+            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"]
         if mode == 'inference':
             tts_text = [sample[i]['tts_text'] for i in order]
             tts_index = [sample[i]['tts_index'] for i in order]

cosyvoice/flow/decoder.py

@@ -13,16 +13,83 @@
 # limitations under the License.
 import torch
 import torch.nn as nn
+import torch.nn.functional as F
 from einops import pack, rearrange, repeat
+from cosyvoice.utils.common import mask_to_bias
+from cosyvoice.utils.mask import add_optional_chunk_mask
 from matcha.models.components.decoder import SinusoidalPosEmb, Block1D, ResnetBlock1D, Downsample1D, TimestepEmbedding, Upsample1D
 from matcha.models.components.transformer import BasicTransformerBlock
 
 
+class Transpose(torch.nn.Module):
+    def __init__(self, dim0: int, dim1: int):
+        super().__init__()
+        self.dim0 = dim0
+        self.dim1 = dim1
+
+    def forward(self, x: torch.Tensor):
+        x = torch.transpose(x, self.dim0, self.dim1)
+        return x
+
+
+class CausalBlock1D(Block1D):
+    def __init__(self, dim: int, dim_out: int):
+        super(CausalBlock1D, self).__init__(dim, dim_out)
+        self.block = torch.nn.Sequential(
+            CausalConv1d(dim, dim_out, 3),
+            Transpose(1, 2),
+            nn.LayerNorm(dim_out),
+            Transpose(1, 2),
+            nn.Mish(),
+        )
+
+    def forward(self, x: torch.Tensor, mask: torch.Tensor):
+        output = self.block(x * mask)
+        return output * mask
+
+
+class CausalResnetBlock1D(ResnetBlock1D):
+    def __init__(self, dim: int, dim_out: int, time_emb_dim: int, groups: int = 8):
+        super(CausalResnetBlock1D, self).__init__(dim, dim_out, time_emb_dim, groups)
+        self.block1 = CausalBlock1D(dim, dim_out)
+        self.block2 = CausalBlock1D(dim_out, dim_out)
+
+
+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',
+        device=None,
+        dtype=None
+    ) -> None:
+        super(CausalConv1d, 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
+        self.causal_padding = (kernel_size - 1, 0)
+
+    def forward(self, x: torch.Tensor):
+        x = F.pad(x, self.causal_padding)
+        x = super(CausalConv1d, self).forward(x)
+        return x
+
+
 class ConditionalDecoder(nn.Module):
     def __init__(
         self,
         in_channels,
         out_channels,
+        causal=False,
         channels=(256, 256),
         dropout=0.05,
         attention_head_dim=64,
@@ -39,7 +106,7 @@ class ConditionalDecoder(nn.Module):
         channels = tuple(channels)
         self.in_channels = in_channels
         self.out_channels = out_channels
-
+        self.causal = causal
         self.time_embeddings = SinusoidalPosEmb(in_channels)
         time_embed_dim = channels[0] * 4
         self.time_mlp = TimestepEmbedding(
@@ -56,7 +123,8 @@ class ConditionalDecoder(nn.Module):
             input_channel = output_channel
             output_channel = channels[i]
             is_last = i == len(channels) - 1
-            resnet = ResnetBlock1D(dim=input_channel, dim_out=output_channel, time_emb_dim=time_embed_dim)
+            resnet = CausalResnetBlock1D(dim=input_channel, dim_out=output_channel, time_emb_dim=time_embed_dim) if self.causal else \
+                ResnetBlock1D(dim=input_channel, dim_out=output_channel, time_emb_dim=time_embed_dim)
             transformer_blocks = nn.ModuleList(
                 [
                     BasicTransformerBlock(
@@ -70,14 +138,16 @@ class ConditionalDecoder(nn.Module):
                 ]
             )
             downsample = (
-                Downsample1D(output_channel) if not is_last else nn.Conv1d(output_channel, output_channel, 3, padding=1)
+                Downsample1D(output_channel) if not is_last else
+                CausalConv1d(output_channel, output_channel, 3) if self.causal else nn.Conv1d(output_channel, output_channel, 3, padding=1)
             )
             self.down_blocks.append(nn.ModuleList([resnet, transformer_blocks, downsample]))
 
-        for i in range(num_mid_blocks):
+        for _ in range(num_mid_blocks):
             input_channel = channels[-1]
             out_channels = channels[-1]
-            resnet = ResnetBlock1D(dim=input_channel, dim_out=output_channel, time_emb_dim=time_embed_dim)
+            resnet = CausalResnetBlock1D(dim=input_channel, dim_out=output_channel, time_emb_dim=time_embed_dim) if self.causal else \
+                ResnetBlock1D(dim=input_channel, dim_out=output_channel, time_emb_dim=time_embed_dim)
 
             transformer_blocks = nn.ModuleList(
                 [
@@ -99,7 +169,11 @@ class ConditionalDecoder(nn.Module):
             input_channel = channels[i] * 2
             output_channel = channels[i + 1]
             is_last = i == len(channels) - 2
-            resnet = ResnetBlock1D(
+            resnet = CausalResnetBlock1D(
+                dim=input_channel,
+                dim_out=output_channel,
+                time_emb_dim=time_embed_dim,
+            ) if self.causal else ResnetBlock1D(
                 dim=input_channel,
                 dim_out=output_channel,
                 time_emb_dim=time_embed_dim,
@@ -119,14 +193,13 @@ class ConditionalDecoder(nn.Module):
             upsample = (
                 Upsample1D(output_channel, use_conv_transpose=True)
                 if not is_last
-                else nn.Conv1d(output_channel, output_channel, 3, padding=1)
+                else CausalConv1d(output_channel, output_channel, 3) if self.causal else nn.Conv1d(output_channel, output_channel, 3, padding=1)
             )
             self.up_blocks.append(nn.ModuleList([resnet, transformer_blocks, upsample]))
-        self.final_block = Block1D(channels[-1], channels[-1])
+        self.final_block = CausalBlock1D(channels[-1], channels[-1]) if self.causal else Block1D(channels[-1], channels[-1])
         self.final_proj = nn.Conv1d(channels[-1], self.out_channels, 1)
         self.initialize_weights()
 
-
     def initialize_weights(self):
         for m in self.modules():
             if isinstance(m, nn.Conv1d):
@@ -159,7 +232,7 @@ class ConditionalDecoder(nn.Module):
             _type_: _description_
         """
 
-        t = self.time_embeddings(t)
+        t = self.time_embeddings(t).to(t.dtype)
         t = self.time_mlp(t)
 
         x = pack([x, mu], "b * t")[0]
@@ -176,7 +249,9 @@ class ConditionalDecoder(nn.Module):
             mask_down = masks[-1]
             x = resnet(x, mask_down, t)
             x = rearrange(x, "b c t -> b t c").contiguous()
-            attn_mask = torch.matmul(mask_down.transpose(1, 2).contiguous(), mask_down)
+            # attn_mask = torch.matmul(mask_down.transpose(1, 2).contiguous(), mask_down)
+            attn_mask = add_optional_chunk_mask(x, mask_down.bool(), False, False, 0, self.static_chunk_size, -1)
+            attn_mask = mask_to_bias(attn_mask == 1, x.dtype)
             for transformer_block in transformer_blocks:
                 x = transformer_block(
                     hidden_states=x,
@@ -193,7 +268,9 @@ class ConditionalDecoder(nn.Module):
         for resnet, transformer_blocks in self.mid_blocks:
             x = resnet(x, mask_mid, t)
             x = rearrange(x, "b c t -> b t c").contiguous()
-            attn_mask = torch.matmul(mask_mid.transpose(1, 2).contiguous(), mask_mid)
+            # attn_mask = torch.matmul(mask_mid.transpose(1, 2).contiguous(), mask_mid)
+            attn_mask = add_optional_chunk_mask(x, mask_mid.bool(), False, False, 0, self.static_chunk_size, -1)
+            attn_mask = mask_to_bias(attn_mask == 1, x.dtype)
             for transformer_block in transformer_blocks:
                 x = transformer_block(
                     hidden_states=x,
@@ -208,7 +285,9 @@ class ConditionalDecoder(nn.Module):
             x = pack([x[:, :, :skip.shape[-1]], skip], "b * t")[0]
             x = resnet(x, mask_up, t)
             x = rearrange(x, "b c t -> b t c").contiguous()
-            attn_mask = torch.matmul(mask_up.transpose(1, 2).contiguous(), mask_up)
+            # attn_mask = torch.matmul(mask_up.transpose(1, 2).contiguous(), mask_up)
+            attn_mask = add_optional_chunk_mask(x, mask_up.bool(), False, False, 0, self.static_chunk_size, -1)
+            attn_mask = mask_to_bias(attn_mask == 1, x.dtype)
             for transformer_block in transformer_blocks:
                 x = transformer_block(
                     hidden_states=x,

cosyvoice/flow/flow.py

@@ -33,8 +33,13 @@ class MaskedDiffWithXvec(torch.nn.Module):
                  encoder: torch.nn.Module = None,
                  length_regulator: 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'}},
+                 decoder_conf: Dict = {'in_channels': 240, 'out_channel': 80, 'spk_emb_dim': 80, 'n_spks': 1,
-                 mel_feat_conf: Dict = {'n_fft': 1024, 'num_mels': 80, 'sampling_rate': 22050, 'hop_size': 256, 'win_size': 1024, 'fmin': 0, 'fmax': 8000}):
+                                       '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}):
         super().__init__()
         self.input_size = input_size
         self.output_size = output_size
@@ -104,7 +109,101 @@ class MaskedDiffWithXvec(torch.nn.Module):
                   prompt_token_len,
                   prompt_feat,
                   prompt_feat_len,
-                  embedding):
+                  embedding,
+                  flow_cache):
+        if self.fp16 is True:
+            prompt_feat = prompt_feat.half()
+            embedding = embedding.half()
+
+        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_len1, token_len2 = prompt_token.shape[1], token.shape[1]
+        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
+        h, h_lengths = self.encoder(token, token_len)
+        h = self.encoder_proj(h)
+        mel_len1, mel_len2 = prompt_feat.shape[1], int(token_len2 / self.input_frame_rate * 22050 / 256)
+        h, h_lengths = self.length_regulator.inference(h[:, :token_len1], h[:, token_len1:], mel_len1, mel_len2, self.input_frame_rate)
+
+        # 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, flow_cache = self.decoder(
+            mu=h.transpose(1, 2).contiguous(),
+            mask=mask.unsqueeze(1),
+            spks=embedding,
+            cond=conds,
+            n_timesteps=10,
+            prompt_len=mel_len1,
+            flow_cache=flow_cache
+        )
+        feat = feat[:, :, mel_len1:]
+        assert feat.shape[2] == mel_len2
+        return feat.float(), flow_cache
+
+
+class CausalMaskedDiffWithXvec(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,
+                 encoder: 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'}},
+                 mel_feat_conf: Dict = {'n_fft': 1024, 'num_mels': 80, 'sampling_rate': 22050,
+                                        'hop_size': 256, 'win_size': 1024, 'fmin': 0, 'fmax': 8000}):
+        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
+        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.encoder = encoder
+        self.encoder_proj = torch.nn.Linear(self.encoder.output_size(), output_size)
+        self.decoder = decoder
+        self.only_mask_loss = only_mask_loss
+        self.token_mel_ratio = token_mel_ratio
+        self.pre_lookahead_len = pre_lookahead_len
+
+    @torch.inference_mode()
+    def inference(self,
+                  token,
+                  token_len,
+                  prompt_token,
+                  prompt_token_len,
+                  prompt_feat,
+                  prompt_feat_len,
+                  embedding,
+                  finalize):
+        if self.fp16 is True:
+            prompt_feat = prompt_feat.half()
+            embedding = embedding.half()
+
         assert token.shape[0] == 1
         # xvec projection
         embedding = F.normalize(embedding, dim=1)
@@ -112,30 +211,29 @@ class MaskedDiffWithXvec(torch.nn.Module):
 
         # 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)).float().unsqueeze(-1).to(embedding)
+        mask = (~make_pad_mask(token_len)).unsqueeze(-1).to(embedding)
         token = self.input_embedding(torch.clamp(token, min=0)) * mask
 
         # text encode
         h, h_lengths = self.encoder(token, token_len)
+        if finalize is False:
+            h = h[:, :-self.pre_lookahead_len * self.token_mel_ratio]
+        mel_len1, mel_len2 = prompt_feat.shape[1], h.shape[1] - prompt_feat.shape[1]
         h = self.encoder_proj(h)
-        feat_len = (token_len / 50 * 22050 / 256).int()
-        h, h_lengths = self.length_regulator(h, feat_len)
 
         # get conditions
-        conds = torch.zeros([1, feat_len.max().item(), self.output_size], device=token.device)
+        conds = torch.zeros([1, mel_len1 + mel_len2, self.output_size], device=token.device).to(h.dtype)
-        if prompt_feat.shape[1] != 0:
+        conds[:, :mel_len1] = prompt_feat
-            for i, j in enumerate(prompt_feat_len):
-                conds[i, :j] = prompt_feat[i]
         conds = conds.transpose(1, 2)
 
-        mask = (~make_pad_mask(feat_len)).to(h)
+        mask = (~make_pad_mask(torch.tensor([mel_len1 + mel_len2]))).to(h)
-        feat = self.decoder(
+        feat, _ = self.decoder(
             mu=h.transpose(1, 2).contiguous(),
             mask=mask.unsqueeze(1),
             spks=embedding,
             cond=conds,
             n_timesteps=10
         )
-        if prompt_feat.shape[1] != 0:
+        feat = feat[:, :, mel_len1:]
-            feat = feat[:, :, prompt_feat.shape[1]:]
+        assert feat.shape[2] == mel_len2
-        return feat
+        return feat.float(), None

cosyvoice/flow/flow_matching.py

@@ -11,10 +11,12 @@
 # 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 threading
 import torch
 import torch.nn.functional as F
 from matcha.models.components.flow_matching import BASECFM
 
+
 class ConditionalCFM(BASECFM):
     def __init__(self, in_channels, cfm_params, n_spks=1, spk_emb_dim=64, estimator: torch.nn.Module = None):
         super().__init__(
@@ -29,9 +31,10 @@ class ConditionalCFM(BASECFM):
         in_channels = in_channels + (spk_emb_dim if n_spks > 0 else 0)
         # Just change the architecture of the estimator here
         self.estimator = estimator
+        self.lock = threading.Lock()
 
     @torch.inference_mode()
-    def forward(self, mu, mask, n_timesteps, temperature=1.0, spks=None, cond=None):
+    def forward(self, mu, mask, n_timesteps, temperature=1.0, spks=None, cond=None, prompt_len=0, flow_cache=torch.zeros(1, 80, 0, 2)):
         """Forward diffusion
 
         Args:
@@ -49,11 +52,21 @@ class ConditionalCFM(BASECFM):
             sample: generated mel-spectrogram
                 shape: (batch_size, n_feats, mel_timesteps)
         """
-        z = torch.randn_like(mu) * temperature
+
-        t_span = torch.linspace(0, 1, n_timesteps + 1, device=mu.device)
+        z = torch.randn_like(mu).to(mu.device).to(mu.dtype) * temperature
+        cache_size = flow_cache.shape[2]
+        # fix prompt and overlap part mu and z
+        if cache_size != 0:
+            z[:, :, :cache_size] = flow_cache[:, :, :, 0]
+            mu[:, :, :cache_size] = flow_cache[:, :, :, 1]
+        z_cache = torch.concat([z[:, :, :prompt_len], z[:, :, -34:]], dim=2)
+        mu_cache = torch.concat([mu[:, :, :prompt_len], mu[:, :, -34:]], dim=2)
+        flow_cache = torch.stack([z_cache, mu_cache], dim=-1)
+
+        t_span = torch.linspace(0, 1, n_timesteps + 1, device=mu.device, dtype=mu.dtype)
         if self.t_scheduler == 'cosine':
             t_span = 1 - torch.cos(t_span * 0.5 * torch.pi)
-        return self.solve_euler(z, t_span=t_span, mu=mu, mask=mask, spks=spks, cond=cond)
+        return self.solve_euler(z, t_span=t_span, mu=mu, mask=mask, spks=spks, cond=cond), flow_cache
 
     def solve_euler(self, x, t_span, mu, mask, spks, cond):
         """
@@ -71,30 +84,63 @@ class ConditionalCFM(BASECFM):
             cond: Not used but kept for future purposes
         """
         t, _, dt = t_span[0], t_span[-1], t_span[1] - t_span[0]
+        t = t.unsqueeze(dim=0)
 
         # I am storing this because I can later plot it by putting a debugger here and saving it to a file
         # Or in future might add like a return_all_steps flag
         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)
         for step in range(1, len(t_span)):
-            dphi_dt = self.estimator(x, mask, mu, t, spks, cond)
             # Classifier-Free Guidance inference introduced in VoiceBox
-            if self.inference_cfg_rate > 0:
+            x_in[:] = x
-                cfg_dphi_dt = self.estimator(
+            mask_in[:] = mask
-                    x, mask,
+            mu_in[0] = mu
-                    torch.zeros_like(mu), t,
+            t_in[:] = t.unsqueeze(0)
-                    torch.zeros_like(spks) if spks is not None else None,
+            spks_in[0] = spks
-                    torch.zeros_like(cond)
+            cond_in[0] = cond
-                )
+            dphi_dt = self.forward_estimator(
-                dphi_dt = ((1.0 + self.inference_cfg_rate) * dphi_dt -
+                x_in, mask_in,
-                           self.inference_cfg_rate * cfg_dphi_dt)
+                mu_in, t_in,
+                spks_in,
+                cond_in
+            )
+            dphi_dt, cfg_dphi_dt = torch.split(dphi_dt, [x.size(0), x.size(0)], dim=0)
+            dphi_dt = ((1.0 + self.inference_cfg_rate) * dphi_dt - self.inference_cfg_rate * cfg_dphi_dt)
             x = x + dt * dphi_dt
             t = t + dt
             sol.append(x)
             if step < len(t_span) - 1:
                 dt = t_span[step + 1] - t
 
-        return sol[-1]
+        return sol[-1].float()
+
+    def forward_estimator(self, x, mask, mu, t, spks, cond):
+        if isinstance(self.estimator, torch.nn.Module):
+            return self.estimator.forward(x, mask, mu, t, spks, cond)
+        else:
+            with self.lock:
+                self.estimator.set_input_shape('x', (2, 80, x.size(2)))
+                self.estimator.set_input_shape('mask', (2, 1, x.size(2)))
+                self.estimator.set_input_shape('mu', (2, 80, x.size(2)))
+                self.estimator.set_input_shape('t', (2,))
+                self.estimator.set_input_shape('spks', (2, 80))
+                self.estimator.set_input_shape('cond', (2, 80, x.size(2)))
+                # run trt engine
+                self.estimator.execute_v2([x.contiguous().data_ptr(),
+                                           mask.contiguous().data_ptr(),
+                                           mu.contiguous().data_ptr(),
+                                           t.contiguous().data_ptr(),
+                                           spks.contiguous().data_ptr(),
+                                           cond.contiguous().data_ptr(),
+                                           x.data_ptr()])
+            return x
 
     def compute_loss(self, x1, mask, mu, spks=None, cond=None):
         """Computes diffusion loss
@@ -136,3 +182,36 @@ class ConditionalCFM(BASECFM):
         pred = self.estimator(y, mask, mu, t.squeeze(), spks, cond)
         loss = F.mse_loss(pred * mask, u * mask, reduction="sum") / (torch.sum(mask) * u.shape[1])
         return loss, y
+
+
+class CausalConditionalCFM(ConditionalCFM):
+    def __init__(self, in_channels, cfm_params, n_spks=1, spk_emb_dim=64, estimator: torch.nn.Module = None):
+        super().__init__(in_channels, cfm_params, n_spks, spk_emb_dim, estimator)
+        self.rand_noise = torch.randn([1, 80, 50 * 300])
+
+    @torch.inference_mode()
+    def forward(self, mu, mask, n_timesteps, temperature=1.0, spks=None, cond=None):
+        """Forward diffusion
+
+        Args:
+            mu (torch.Tensor): output of encoder
+                shape: (batch_size, n_feats, mel_timesteps)
+            mask (torch.Tensor): output_mask
+                shape: (batch_size, 1, mel_timesteps)
+            n_timesteps (int): number of diffusion steps
+            temperature (float, optional): temperature for scaling noise. Defaults to 1.0.
+            spks (torch.Tensor, optional): speaker ids. Defaults to None.
+                shape: (batch_size, spk_emb_dim)
+            cond: Not used but kept for future purposes
+
+        Returns:
+            sample: generated mel-spectrogram
+                shape: (batch_size, n_feats, mel_timesteps)
+        """
+
+        z = self.rand_noise[:, :, :mu.size(2)].to(mu.device).to(mu.dtype) * temperature
+        # fix prompt and overlap part mu and z
+        t_span = torch.linspace(0, 1, n_timesteps + 1, device=mu.device, dtype=mu.dtype)
+        if self.t_scheduler == 'cosine':
+            t_span = 1 - torch.cos(t_span * 0.5 * torch.pi)
+        return self.solve_euler(z, t_span=t_span, mu=mu, mask=mask, spks=spks, cond=cond), None

cosyvoice/flow/length_regulator.py

@@ -13,6 +13,7 @@
 # limitations under the License.
 from typing import Tuple
 import torch.nn as nn
+import torch
 from torch.nn import functional as F
 from cosyvoice.utils.mask import make_pad_mask
 
@@ -43,7 +44,26 @@ class InterpolateRegulator(nn.Module):
     def forward(self, x, ylens=None):
         # x in (B, T, D)
         mask = (~make_pad_mask(ylens)).to(x).unsqueeze(-1)
-        x = F.interpolate(x.transpose(1, 2).contiguous(), size=ylens.max(), mode='nearest')
+        x = F.interpolate(x.transpose(1, 2).contiguous(), size=ylens.max(), mode='linear')
         out = self.model(x).transpose(1, 2).contiguous()
         olens = ylens
         return out * mask, olens
+
+    def inference(self, x1, x2, mel_len1, mel_len2, input_frame_rate=50):
+        # in inference mode, interploate prompt token and token(head/mid/tail) seprately, so we can get a clear separation point of mel
+        # x in (B, T, D)
+        if x2.shape[1] > 40:
+            x2_head = F.interpolate(x2[:, :20].transpose(1, 2).contiguous(), size=int(20 / input_frame_rate * 22050 / 256), mode='linear')
+            x2_mid = F.interpolate(x2[:, 20:-20].transpose(1, 2).contiguous(), size=mel_len2 - int(20 / input_frame_rate * 22050 / 256) * 2,
+                                   mode='linear')
+            x2_tail = F.interpolate(x2[:, -20:].transpose(1, 2).contiguous(), size=int(20 / input_frame_rate * 22050 / 256), mode='linear')
+            x2 = torch.concat([x2_head, x2_mid, x2_tail], dim=2)
+        else:
+            x2 = F.interpolate(x2.transpose(1, 2).contiguous(), size=mel_len2, mode='linear')
+        if x1.shape[1] != 0:
+            x1 = F.interpolate(x1.transpose(1, 2).contiguous(), size=mel_len1, mode='linear')
+            x = torch.concat([x1, x2], dim=2)
+        else:
+            x = x2
+        out = self.model(x).transpose(1, 2).contiguous()
+        return out, mel_len1 + mel_len2

cosyvoice/hifigan/discriminator.py

@@ -0,0 +1,140 @@
+import torch
+import torch.nn as nn
+from torch.nn.utils.parametrizations import weight_norm
+from typing import List, Optional, Tuple
+from einops import rearrange
+from torchaudio.transforms import Spectrogram
+
+
+class MultipleDiscriminator(nn.Module):
+    def __init__(
+            self, mpd: nn.Module, mrd: nn.Module
+    ):
+        super().__init__()
+        self.mpd = mpd
+        self.mrd = mrd
+
+    def forward(self, y: torch.Tensor, y_hat: torch.Tensor):
+        y_d_rs, y_d_gs, fmap_rs, fmap_gs = [], [], [], []
+        this_y_d_rs, this_y_d_gs, this_fmap_rs, this_fmap_gs = self.mpd(y.unsqueeze(dim=1), y_hat.unsqueeze(dim=1))
+        y_d_rs += this_y_d_rs
+        y_d_gs += this_y_d_gs
+        fmap_rs += this_fmap_rs
+        fmap_gs += this_fmap_gs
+        this_y_d_rs, this_y_d_gs, this_fmap_rs, this_fmap_gs = self.mrd(y, y_hat)
+        y_d_rs += this_y_d_rs
+        y_d_gs += this_y_d_gs
+        fmap_rs += this_fmap_rs
+        fmap_gs += this_fmap_gs
+        return y_d_rs, y_d_gs, fmap_rs, fmap_gs
+
+
+class MultiResolutionDiscriminator(nn.Module):
+    def __init__(
+        self,
+        fft_sizes: Tuple[int, ...] = (2048, 1024, 512),
+        num_embeddings: Optional[int] = None,
+    ):
+        """
+        Multi-Resolution Discriminator module adapted from https://github.com/descriptinc/descript-audio-codec.
+        Additionally, it allows incorporating conditional information with a learned embeddings table.
+
+        Args:
+            fft_sizes (tuple[int]): Tuple of window lengths for FFT. Defaults to (2048, 1024, 512).
+            num_embeddings (int, optional): Number of embeddings. None means non-conditional discriminator.
+                Defaults to None.
+        """
+
+        super().__init__()
+        self.discriminators = nn.ModuleList(
+            [DiscriminatorR(window_length=w, num_embeddings=num_embeddings) for w in fft_sizes]
+        )
+
+    def forward(
+        self, y: torch.Tensor, y_hat: torch.Tensor, bandwidth_id: torch.Tensor = None
+    ) -> Tuple[List[torch.Tensor], List[torch.Tensor], List[List[torch.Tensor]], List[List[torch.Tensor]]]:
+        y_d_rs = []
+        y_d_gs = []
+        fmap_rs = []
+        fmap_gs = []
+
+        for d in self.discriminators:
+            y_d_r, fmap_r = d(x=y, cond_embedding_id=bandwidth_id)
+            y_d_g, fmap_g = d(x=y_hat, cond_embedding_id=bandwidth_id)
+            y_d_rs.append(y_d_r)
+            fmap_rs.append(fmap_r)
+            y_d_gs.append(y_d_g)
+            fmap_gs.append(fmap_g)
+
+        return y_d_rs, y_d_gs, fmap_rs, fmap_gs
+
+
+class DiscriminatorR(nn.Module):
+    def __init__(
+        self,
+        window_length: int,
+        num_embeddings: Optional[int] = None,
+        channels: int = 32,
+        hop_factor: float = 0.25,
+        bands: Tuple[Tuple[float, float], ...] = ((0.0, 0.1), (0.1, 0.25), (0.25, 0.5), (0.5, 0.75), (0.75, 1.0)),
+    ):
+        super().__init__()
+        self.window_length = window_length
+        self.hop_factor = hop_factor
+        self.spec_fn = Spectrogram(
+            n_fft=window_length, hop_length=int(window_length * hop_factor), win_length=window_length, power=None
+        )
+        n_fft = window_length // 2 + 1
+        bands = [(int(b[0] * n_fft), int(b[1] * n_fft)) for b in bands]
+        self.bands = bands
+        convs = lambda: nn.ModuleList(
+            [
+                weight_norm(nn.Conv2d(2, channels, (3, 9), (1, 1), padding=(1, 4))),
+                weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
+                weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
+                weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
+                weight_norm(nn.Conv2d(channels, channels, (3, 3), (1, 1), padding=(1, 1))),
+            ]
+        )
+        self.band_convs = nn.ModuleList([convs() for _ in range(len(self.bands))])
+
+        if num_embeddings is not None:
+            self.emb = torch.nn.Embedding(num_embeddings=num_embeddings, embedding_dim=channels)
+            torch.nn.init.zeros_(self.emb.weight)
+
+        self.conv_post = weight_norm(nn.Conv2d(channels, 1, (3, 3), (1, 1), padding=(1, 1)))
+
+    def spectrogram(self, x):
+        # Remove DC offset
+        x = x - x.mean(dim=-1, keepdims=True)
+        # Peak normalize the volume of input audio
+        x = 0.8 * x / (x.abs().max(dim=-1, keepdim=True)[0] + 1e-9)
+        x = self.spec_fn(x)
+        x = torch.view_as_real(x)
+        x = rearrange(x, "b f t c -> b c t f")
+        # Split into bands
+        x_bands = [x[..., b[0]: b[1]] for b in self.bands]
+        return x_bands
+
+    def forward(self, x: torch.Tensor, cond_embedding_id: torch.Tensor = None):
+        x_bands = self.spectrogram(x)
+        fmap = []
+        x = []
+        for band, stack in zip(x_bands, self.band_convs):
+            for i, layer in enumerate(stack):
+                band = layer(band)
+                band = torch.nn.functional.leaky_relu(band, 0.1)
+                if i > 0:
+                    fmap.append(band)
+            x.append(band)
+        x = torch.cat(x, dim=-1)
+        if cond_embedding_id is not None:
+            emb = self.emb(cond_embedding_id)
+            h = (emb.view(1, -1, 1, 1) * x).sum(dim=1, keepdims=True)
+        else:
+            h = 0
+        x = self.conv_post(x)
+        fmap.append(x)
+        x += h
+
+        return x, fmap

cosyvoice/hifigan/f0_predictor.py

@@ -13,7 +13,7 @@
 # limitations under the License.
 import torch
 import torch.nn as nn
-from torch.nn.utils import weight_norm
+from torch.nn.utils.parametrizations import weight_norm
 
 
 class ConvRNNF0Predictor(nn.Module):

cosyvoice/hifigan/generator.py

@@ -14,7 +14,7 @@
 
 """HIFI-GAN"""
 
-import typing as tp
+from typing import Dict, Optional, List
 import numpy as np
 from scipy.signal import get_window
 import torch
@@ -23,7 +23,7 @@ import torch.nn.functional as F
 from torch.nn import Conv1d
 from torch.nn import ConvTranspose1d
 from torch.nn.utils import remove_weight_norm
-from torch.nn.utils import weight_norm
+from torch.nn.utils.parametrizations import weight_norm
 from torch.distributions.uniform import Uniform
 
 from cosyvoice.transformer.activation import Snake
@@ -38,13 +38,15 @@ This code is modified from https://github.com/jik876/hifi-gan
  https://github.com/NVIDIA/BigVGAN
 
 """
+
+
 class ResBlock(torch.nn.Module):
     """Residual block module in HiFiGAN/BigVGAN."""
     def __init__(
         self,
         channels: int = 512,
         kernel_size: int = 3,
-        dilations: tp.List[int] = [1, 3, 5],
+        dilations: List[int] = [1, 3, 5],
     ):
         super(ResBlock, self).__init__()
         self.convs1 = nn.ModuleList()
@@ -100,6 +102,7 @@ class ResBlock(torch.nn.Module):
             remove_weight_norm(self.convs1[idx])
             remove_weight_norm(self.convs2[idx])
 
+
 class SineGen(torch.nn.Module):
     """ Definition of sine generator
     SineGen(samp_rate, harmonic_num = 0,
@@ -231,13 +234,13 @@ class HiFTGenerator(nn.Module):
             nsf_alpha: float = 0.1,
             nsf_sigma: float = 0.003,
             nsf_voiced_threshold: float = 10,
-            upsample_rates: tp.List[int] = [8, 8],
+            upsample_rates: List[int] = [8, 8],
-            upsample_kernel_sizes: tp.List[int] = [16, 16],
+            upsample_kernel_sizes: List[int] = [16, 16],
-            istft_params: tp.Dict[str, int] = {"n_fft": 16, "hop_len": 4},
+            istft_params: Dict[str, int] = {"n_fft": 16, "hop_len": 4},
-            resblock_kernel_sizes: tp.List[int] = [3, 7, 11],
+            resblock_kernel_sizes: List[int] = [3, 7, 11],
-            resblock_dilation_sizes: tp.List[tp.List[int]] = [[1, 3, 5], [1, 3, 5], [1, 3, 5]],
+            resblock_dilation_sizes: List[List[int]] = [[1, 3, 5], [1, 3, 5], [1, 3, 5]],
-            source_resblock_kernel_sizes: tp.List[int] = [7, 11],
+            source_resblock_kernel_sizes: List[int] = [7, 11],
-            source_resblock_dilation_sizes: tp.List[tp.List[int]] = [[1, 3, 5], [1, 3, 5]],
+            source_resblock_dilation_sizes: List[List[int]] = [[1, 3, 5], [1, 3, 5]],
             lrelu_slope: float = 0.1,
             audio_limit: float = 0.99,
             f0_predictor: torch.nn.Module = None,
@@ -286,8 +289,7 @@ class HiFTGenerator(nn.Module):
         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,
+        for i, (u, k, d) in enumerate(zip(downsample_cum_rates[::-1], source_resblock_kernel_sizes, source_resblock_dilation_sizes)):
-                                          source_resblock_dilation_sizes)):
             if u == 1:
                 self.source_downs.append(
                     Conv1d(istft_params["n_fft"] + 2, base_channels // (2 ** (i + 1)), 1, 1)
@@ -304,7 +306,7 @@ class HiFTGenerator(nn.Module):
         self.resblocks = nn.ModuleList()
         for i in range(len(self.ups)):
             ch = base_channels // (2**(i + 1))
-            for j, (k, d) in enumerate(zip(resblock_kernel_sizes, resblock_dilation_sizes)):
+            for _, (k, d) in enumerate(zip(resblock_kernel_sizes, resblock_dilation_sizes)):
                 self.resblocks.append(ResBlock(ch, k, d))
 
         self.conv_post = weight_norm(Conv1d(ch, istft_params["n_fft"] + 2, 7, 1, padding=3))
@@ -314,11 +316,19 @@ class HiFTGenerator(nn.Module):
         self.stft_window = torch.from_numpy(get_window("hann", istft_params["n_fft"], fftbins=True).astype(np.float32))
         self.f0_predictor = f0_predictor
 
-    def _f02source(self, f0: torch.Tensor) -> torch.Tensor:
+    def remove_weight_norm(self):
-        f0 = self.f0_upsamp(f0[:, None]).transpose(1, 2)  # bs,n,t
+        print('Removing weight norm...')
-
+        for l in self.ups:
-        har_source, _, _ = self.m_source(f0)
+            remove_weight_norm(l)
-        return har_source.transpose(1, 2)
+        for l in self.resblocks:
+            l.remove_weight_norm()
+        remove_weight_norm(self.conv_pre)
+        remove_weight_norm(self.conv_post)
+        self.m_source.remove_weight_norm()
+        for l in self.source_downs:
+            remove_weight_norm(l)
+        for l in self.source_resblocks:
+            l.remove_weight_norm()
 
     def _stft(self, x):
         spec = torch.stft(
@@ -332,13 +342,11 @@ class HiFTGenerator(nn.Module):
         magnitude = torch.clip(magnitude, max=1e2)
         real = magnitude * torch.cos(phase)
         img = magnitude * torch.sin(phase)
-        inverse_transform = torch.istft(torch.complex(real, img), self.istft_params["n_fft"], self.istft_params["hop_len"], self.istft_params["n_fft"], window=self.stft_window.to(magnitude.device))
+        inverse_transform = torch.istft(torch.complex(real, img), self.istft_params["n_fft"], self.istft_params["hop_len"],
+                                        self.istft_params["n_fft"], window=self.stft_window.to(magnitude.device))
         return inverse_transform
 
-    def forward(self, x: torch.Tensor) -> torch.Tensor:
+    def decode(self, x: torch.Tensor, s: torch.Tensor = torch.zeros(1, 1, 0)) -> torch.Tensor:
-        f0 = self.f0_predictor(x)
-        s = self._f02source(f0)
-
         s_stft_real, s_stft_imag = self._stft(s.squeeze(1))
         s_stft = torch.cat([s_stft_real, s_stft_imag], dim=1)
 
@@ -372,20 +380,32 @@ class HiFTGenerator(nn.Module):
         x = torch.clamp(x, -self.audio_limit, self.audio_limit)
         return x
 
-    def remove_weight_norm(self):
+    def forward(
-        print('Removing weight norm...')
+            self,
-        for l in self.ups:
+            batch: dict,
-            remove_weight_norm(l)
+            device: torch.device,
-        for l in self.resblocks:
+    ) -> Dict[str, Optional[torch.Tensor]]:
-            l.remove_weight_norm()
+        speech_feat = batch['speech_feat'].transpose(1, 2).to(device)
-        remove_weight_norm(self.conv_pre)
+        # mel->f0
-        remove_weight_norm(self.conv_post)
+        f0 = self.f0_predictor(speech_feat)
-        self.source_module.remove_weight_norm()
+        # f0->source
-        for l in self.source_downs:
+        s = self.f0_upsamp(f0[:, None]).transpose(1, 2)  # bs,n,t
-            remove_weight_norm(l)
+        s, _, _ = self.m_source(s)
-        for l in self.source_resblocks:
+        s = s.transpose(1, 2)
-            l.remove_weight_norm()
+        # mel+source->speech
+        generated_speech = self.decode(x=speech_feat, s=s)
+        return generated_speech, f0
 
     @torch.inference_mode()
-    def inference(self, mel: torch.Tensor) -> torch.Tensor:
+    def inference(self, speech_feat: torch.Tensor, cache_source: torch.Tensor = torch.zeros(1, 1, 0)) -> torch.Tensor:
-        return self.forward(x=mel)
+        # mel->f0
+        f0 = self.f0_predictor(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)
+        # use cache_source to avoid glitch
+        if cache_source.shape[2] != 0:
+            s[:, :, :cache_source.shape[2]] = cache_source
+        generated_speech = self.decode(x=speech_feat, s=s)
+        return generated_speech, s

cosyvoice/hifigan/hifigan.py

@@ -0,0 +1,67 @@
+from typing import Dict, Optional
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from matcha.hifigan.models import feature_loss, generator_loss, discriminator_loss
+from cosyvoice.utils.losses import tpr_loss, mel_loss
+
+
+class HiFiGan(nn.Module):
+    def __init__(self, generator, discriminator, mel_spec_transform,
+                 multi_mel_spectral_recon_loss_weight=45, feat_match_loss_weight=2.0,
+                 tpr_loss_weight=1.0, tpr_loss_tau=0.04):
+        super(HiFiGan, self).__init__()
+        self.generator = generator
+        self.discriminator = discriminator
+        self.mel_spec_transform = mel_spec_transform
+        self.multi_mel_spectral_recon_loss_weight = multi_mel_spectral_recon_loss_weight
+        self.feat_match_loss_weight = feat_match_loss_weight
+        self.tpr_loss_weight = tpr_loss_weight
+        self.tpr_loss_tau = tpr_loss_tau
+
+    def forward(
+            self,
+            batch: dict,
+            device: torch.device,
+    ) -> Dict[str, Optional[torch.Tensor]]:
+        if batch['turn'] == 'generator':
+            return self.forward_generator(batch, device)
+        else:
+            return self.forward_discriminator(batch, device)
+
+    def forward_generator(self, batch, device):
+        real_speech = batch['speech'].to(device)
+        pitch_feat = batch['pitch_feat'].to(device)
+        # 1. calculate generator outputs
+        generated_speech, generated_f0 = self.generator(batch, device)
+        # 2. calculate discriminator outputs
+        y_d_rs, y_d_gs, fmap_rs, fmap_gs = self.discriminator(real_speech, generated_speech)
+        # 3. calculate generator losses, feature loss, mel loss, tpr losses [Optional]
+        loss_gen, _ = generator_loss(y_d_gs)
+        loss_fm = feature_loss(fmap_rs, fmap_gs)
+        loss_mel = mel_loss(real_speech, generated_speech, self.mel_spec_transform)
+        if self.tpr_loss_weight != 0:
+            loss_tpr = tpr_loss(y_d_rs, y_d_gs, self.tpr_loss_tau)
+        else:
+            loss_tpr = torch.zeros(1).to(device)
+        loss_f0 = F.l1_loss(generated_f0, pitch_feat)
+        loss = loss_gen + self.feat_match_loss_weight * loss_fm + \
+            self.multi_mel_spectral_recon_loss_weight * loss_mel + \
+            self.tpr_loss_weight * loss_tpr + loss_f0
+        return {'loss': loss, 'loss_gen': loss_gen, 'loss_fm': loss_fm, 'loss_mel': loss_mel, 'loss_tpr': loss_tpr, 'loss_f0': loss_f0}
+
+    def forward_discriminator(self, batch, device):
+        real_speech = batch['speech'].to(device)
+        # 1. calculate generator outputs
+        with torch.no_grad():
+            generated_speech, generated_f0 = self.generator(batch, device)
+        # 2. calculate discriminator outputs
+        y_d_rs, y_d_gs, fmap_rs, fmap_gs = self.discriminator(real_speech, generated_speech)
+        # 3. calculate discriminator losses, tpr losses [Optional]
+        loss_disc, _, _ = discriminator_loss(y_d_rs, y_d_gs)
+        if self.tpr_loss_weight != 0:
+            loss_tpr = tpr_loss(y_d_rs, y_d_gs, self.tpr_loss_tau)
+        else:
+            loss_tpr = torch.zeros(1).to(device)
+        loss = loss_disc + self.tpr_loss_weight * loss_tpr
+        return {'loss': loss, 'loss_disc': loss_disc, 'loss_tpr': loss_tpr}

cosyvoice/llm/llm.py

@@ -11,14 +11,16 @@
 # 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 typing import Dict, Optional, Union
+from typing import Dict, Optional, Callable, List, Generator
 import torch
 from torch import nn
 import torch.nn.functional as F
+from transformers import Qwen2ForCausalLM
 from torch.nn.utils.rnn import pad_sequence, unpad_sequence
 from cosyvoice.utils.common import IGNORE_ID
 from cosyvoice.transformer.label_smoothing_loss import LabelSmoothingLoss
 from cosyvoice.utils.common import th_accuracy
+from cosyvoice.utils.file_utils import logging
 
 
 class TransformerLM(torch.nn.Module):
@@ -31,6 +33,7 @@ class TransformerLM(torch.nn.Module):
             speech_token_size: int,
             text_encoder: torch.nn.Module,
             llm: torch.nn.Module,
+            sampling: Callable,
             length_normalized_loss: bool = True,
             lsm_weight: float = 0.0,
             spk_embed_dim: int = 192,
@@ -63,6 +66,9 @@ class TransformerLM(torch.nn.Module):
         self.speech_embedding = torch.nn.Embedding(speech_token_size, llm_input_size)
         self.spk_embed_affine_layer = torch.nn.Linear(spk_embed_dim, llm_input_size)
 
+        # 4. sampling method
+        self.sampling = sampling
+
     def encode(
             self,
             text: torch.Tensor,
@@ -76,7 +82,8 @@ class TransformerLM(torch.nn.Module):
     def pad_unpad_sequence(self, sos_eos_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) for i in range(len(text_token))]
+        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)
+                    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)
         return lm_input, lm_input_len
@@ -100,7 +107,8 @@ class TransformerLM(torch.nn.Module):
         embedding = batch['embedding'].to(device)
 
         # 1. prepare llm_target
-        lm_target = [torch.tensor([IGNORE_ID] * (2 + text_token_len[i]) + speech_token[i, :speech_token_len[i]].tolist() + [self.speech_token_size]) for i in range(text_token.size(0))]
+        lm_target = [torch.tensor([IGNORE_ID] * (2 + text_token_len[i]) + speech_token[i, :speech_token_len[i]].tolist() +
+                                  [self.speech_token_size]) for i in range(text_token.size(0))]
         lm_target = pad_sequence(lm_target, batch_first=True, padding_value=IGNORE_ID).to(device)
 
         # 1. encode text_token
@@ -120,7 +128,8 @@ class TransformerLM(torch.nn.Module):
         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, task_id_emb, speech_token, speech_token_len)
+        lm_input, lm_input_len = self.pad_unpad_sequence(sos_eos_emb, embedding, text_token, text_token_len,
+                                                         task_id_emb, speech_token, speech_token_len)
 
         # 6. run lm forward
         lm_output, lm_output_mask = self.llm(lm_input, lm_input_len.to(device))
@@ -132,16 +141,18 @@ class TransformerLM(torch.nn.Module):
     def sampling_ids(
             self,
             weighted_scores: torch.Tensor,
-            sampling: Union[bool, int, float] = True,
+            decoded_tokens: List,
-            beam_size: int = 1,
+            sampling: int,
             ignore_eos: bool = True,
     ):
+        num_trials, max_trials = 0, 100
         while True:
-            prob, indices = weighted_scores.softmax(dim=-1).topk(sampling)
+            top_ids = self.sampling(weighted_scores, decoded_tokens, sampling)
-            top_ids = prob.multinomial(beam_size, replacement=True)
-            top_ids = indices[top_ids]
             if (not ignore_eos) or (self.speech_token_size not in top_ids):
                 break
+            num_trials += 1
+            if num_trials > max_trials:
+                raise RuntimeError('sampling reaches max_trials {} and still get eos when ignore_eos is True, check your input!'.format(max_trials))
         return top_ids
 
     @torch.inference_mode()
@@ -154,11 +165,13 @@ class TransformerLM(torch.nn.Module):
             prompt_speech_token: torch.Tensor,
             prompt_speech_token_len: torch.Tensor,
             embedding: torch.Tensor,
-            beam_size: int = 1,
             sampling: int = 25,
             max_token_text_ratio: float = 20,
             min_token_text_ratio: float = 2,
-    ) -> torch.Tensor:
+    ) -> Generator[torch.Tensor, None, None]:
+        if self.fp16 is True:
+            embedding = embedding.half()
+
         device = text.device
         text = torch.concat([prompt_text, text], dim=1)
         text_len += prompt_text_len
@@ -173,7 +186,7 @@ class TransformerLM(torch.nn.Module):
             embedding = self.spk_embed_affine_layer(embedding)
             embedding = embedding.unsqueeze(dim=1)
         else:
-            embedding = torch.zeros(1, 0, self.llm_input_size).to(device)
+            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)
@@ -181,7 +194,7 @@ class TransformerLM(torch.nn.Module):
         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).to(device)
+            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)
 
         # 4. cal min/max_length
@@ -193,14 +206,385 @@ class TransformerLM(torch.nn.Module):
         offset = 0
         att_cache, cnn_cache = torch.zeros((0, 0, 0, 0), device=lm_input.device), torch.zeros((0, 0, 0, 0), device=lm_input.device)
         for i in range(max_len):
-            y_pred, att_cache, cnn_cache = self.llm.forward_chunk(lm_input, offset=0, required_cache_size=-1, att_cache=att_cache, cnn_cache=cnn_cache,
+            y_pred, att_cache, cnn_cache = self.llm.forward_chunk(lm_input, offset=offset, required_cache_size=-1,
-                                                                  att_mask=torch.tril(torch.ones((1, lm_input.shape[1], lm_input.shape[1]), device=lm_input.device)).to(torch.bool))
+                                                                  att_cache=att_cache, cnn_cache=cnn_cache,
+                                                                  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)
-            top_ids = self.sampling_ids(logp.squeeze(dim=0), sampling, beam_size, ignore_eos=True if i < min_len else False).item()
+            # 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:
                 break
+            # in stream mode, yield token one by one
+            yield top_ids
             out_tokens.append(top_ids)
             offset += lm_input.size(1)
             lm_input = self.speech_embedding.weight[top_ids].reshape(1, 1, -1)
 
-        return torch.tensor([out_tokens], dtype=torch.int64, device=device)
+
+class Qwen2Encoder(torch.nn.Module):
+    def __init__(self, pretrain_path):
+        super().__init__()
+        self.model = Qwen2ForCausalLM.from_pretrained(pretrain_path)
+
+    def forward_one_step(self, xs, masks, cache=None):
+        input_masks = masks[:, -1, :]
+        outs = self.model(
+            inputs_embeds=xs,
+            attention_mask=input_masks,
+            output_hidden_states=True,
+            return_dict=True,
+            use_cache=True,
+            past_key_values=cache,
+        )
+        xs = outs.hidden_states[-1]
+        new_cache = outs.past_key_values
+        return xs, new_cache
+
+
+class Qwen2LM(TransformerLM):
+    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_eos = 0
+        self.task_id = 1
+        self.fill_token = 2
+
+        self.llm_embedding = torch.nn.Embedding(2, llm_input_size)
+        self.llm = llm
+        self.llm_decoder = nn.Linear(llm_output_size, speech_token_size + 3)
+        self.criterion_ce = LabelSmoothingLoss(
+            size=speech_token_size + 3,
+            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 + 3, llm_input_size)
+
+        # 4. sampling method
+        self.sampling = sampling
+        self.mix_ratio = mix_ratio
+
+    @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,
+            vllm_codec_engine=None,
+    ) -> 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_eos_emb = self.llm_embedding.weight[self.sos_eos].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)
+
+        # 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
+        if vllm_codec_engine is None:
+            out_tokens = []
+            cache = None
+            for i in range(max_len):
+                y_pred, cache = self.llm.forward_one_step(lm_input,
+                                                        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:
+                    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)
+                lm_input = self.speech_embedding.weight[top_ids].reshape(1, 1, -1)
+        else:
+            from vllm import SamplingParams, RequestOutput
+            import uuid
+            sampling_params = SamplingParams(top_k=sampling,
+                                            stop_token_ids=[6561, 6563],
+                                            min_tokens=min_len,
+                                            max_tokens=max_len)
+            request_id = uuid.uuid4()
+            vllm_codec_engine.add_request(request_id,
+                                        {"prompt_embeds": lm_input.squeeze(0).to(torch.bfloat16).to(device)},
+                                        sampling_params)
+            while True:
+                speech_token_break = False
+                request_outputs: List[RequestOutput] = vllm_codec_engine.step()
+                for request_output in request_outputs:
+                    if str(request_output.request_id) != str(request_id):
+                        continue
+                    # print(f"request output: {request_output}")
+                    top_ids = list(request_output.outputs[0].token_ids)[-1]
+                    if top_ids == self.speech_token_size:
+                        speech_token_break = True
+                        break
+                    if top_ids > self.speech_token_size:
+                        continue
+                    yield top_ids
+
+                if not vllm_codec_engine.has_unfinished_requests() or speech_token_break:
+                    break
+
+    @torch.inference_mode()
+    def inference_bistream(
+            self,
+            text: Generator,
+            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,
+    ) -> Generator[torch.Tensor, None, None]:
+
+        device = prompt_text.device
+        # 1. prepare input
+        sos_eos_emb = self.llm_embedding.weight[self.sos_eos].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)
+
+        # 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
+        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
+            while prompt_speech_token_emb.size(1) != 0:
+                if text_cache.size(1) >= self.mix_ratio[0]:
+                    lm_input_text, lm_input_speech = text_cache[:, :self.mix_ratio[0]], prompt_speech_token_emb[:, :self.mix_ratio[1]]
+                    logging.info('append {} text token {} speech token'.format(lm_input_text.size(1), lm_input_speech.size(1)))
+                    lm_input = torch.concat([lm_input, lm_input_text, lm_input_speech], dim=1)
+                    text_cache, prompt_speech_token_emb = text_cache[:, self.mix_ratio[0]:], prompt_speech_token_emb[:, self.mix_ratio[1]:]
+                else:
+                    logging.info('not enough text token to decode, wait for more')
+                    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):
+                    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:
+                            lm_input = lm_input_text
+                        else:
+                            lm_input = torch.concat([lm_input, lm_input_text], dim=1)
+                        text_cache = text_cache[:, self.mix_ratio[0]:]
+                    else:
+                        logging.info('not enough text token to decode, wait for more')
+                        continue
+                while True:
+                    seq_len = lm_input.shape[1] if cache is None else lm_input.shape[1] + cache[0][0].size(2)
+                    y_pred, cache = self.llm.forward_one_step(lm_input,
+                                                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)
+                    if next_fill_index != -1 and len(out_tokens) == next_fill_index:
+                        top_ids = self.speech_token_size + 2
+                        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:
+                        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:
+                            break
+                        else:
+                            raise ValueError('should not get token {}'.format(top_ids))
+                    yield top_ids
+                    lm_input = self.speech_embedding.weight[top_ids].reshape(1, 1, -1)
+
+        # 3. final decode
+        lm_input = torch.concat([lm_input, text_cache, task_id_emb], dim=1)
+        logging.info('no more text token, decode until met eos')
+        while True:
+            seq_len = lm_input.shape[1] if cache is None else lm_input.shape[1] + cache[0][0].size(2)
+            y_pred, cache = self.llm.forward_one_step(lm_input,
+                                                      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()
+            out_tokens.append(top_ids)
+            if top_ids >= self.speech_token_size:
+                if top_ids == self.speech_token_size:
+                    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)
+    @torch.inference_mode()
+    def inference_bistream_vllm(
+            self,
+            text: Generator,
+            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,
+            vllm_codec_engine=None,
+    ) -> Generator[torch.Tensor, None, None]:
+
+        device = prompt_text.device
+        # 1. prepare input
+        sos_eos_emb = self.llm_embedding.weight[self.sos_eos].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)
+
+        # 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
+        from vllm import SamplingParams, RequestOutput
+        import uuid
+        sampling_params = SamplingParams(top_k=sampling,
+                                        stop_token_ids=[6561, 6563],
+                                        max_tokens=10000)
+        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
+            while prompt_speech_token_emb.size(1) != 0:
+                if text_cache.size(1) >= self.mix_ratio[0]:
+                    lm_input_text, lm_input_speech = text_cache[:, :self.mix_ratio[0]], prompt_speech_token_emb[:, :self.mix_ratio[1]]
+                    logging.info('append {} text token {} speech token'.format(lm_input_text.size(1), lm_input_speech.size(1)))
+                    lm_input = torch.concat([lm_input, lm_input_text, lm_input_speech], dim=1)
+                    text_cache, prompt_speech_token_emb = text_cache[:, self.mix_ratio[0]:], prompt_speech_token_emb[:, self.mix_ratio[1]:]
+                else:
+                    logging.info('not enough text token to decode, wait for more')
+                    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):
+                    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 vllm_codec_engine is None and len(out_tokens) != 0 and out_tokens[-1] == self.speech_token_size + 2:
+                            lm_input = lm_input_text
+                        else:
+                            lm_input = torch.concat([lm_input, lm_input_text], dim=1)
+                        text_cache = text_cache[:, self.mix_ratio[0]:]
+                    else:
+                        logging.info('not enough text token to decode, wait for more')
+                        continue
+                request_id = uuid.uuid4()
+                vllm_codec_engine.add_request(request_id,
+                                            {"prompt_embeds": lm_input.squeeze(0).to(torch.bfloat16).to(device)},
+                                            sampling_params)
+                ## generator
+                while True:
+                    speech_token_break = False
+                    request_outputs: List[RequestOutput] = vllm_codec_engine.step()
+                    for request_output in request_outputs:
+                        if str(request_output.request_id) != str(request_id):
+                            continue
+                        # print(f"request output: {request_output}")
+                        out_token = list(request_output.outputs[0].token_ids)[-1]
+                        if next_fill_index != -1 and len(out_tokens) == next_fill_index:
+                            top_ids = self.speech_token_size + 2
+                            next_fill_index += (self.mix_ratio[1] + 1)
+                        else:
+                            top_ids = out_token
+                        if top_ids == self.speech_token_size + 2:
+                            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:
+                                speech_token_break = True
+                                break
+                            else:
+                                raise ValueError('should not get token {}'.format(top_ids))
+                        yield top_ids
+                        token_embedding = self.speech_embedding.weight[top_ids].reshape(1, 1, -1)
+                        lm_input = torch.concat([lm_input, token_embedding], dim=1)
+
+                    if not vllm_codec_engine.has_unfinished_requests() or speech_token_break:
+                        break             
+
+        # 3. final decode
+        lm_input = torch.concat([lm_input, text_cache, task_id_emb], dim=1)
+        logging.info('no more text token, decode until met eos')
+        request_id = uuid.uuid4()
+        vllm_codec_engine.add_request(request_id,
+                                    {"prompt_embeds": lm_input.squeeze(0).to(torch.bfloat16).to(device)},
+                                    sampling_params)
+        ## generator
+        while True:
+            speech_token_break = False
+            request_outputs: List[RequestOutput] = vllm_codec_engine.step()
+            for request_output in request_outputs:
+                if str(request_output.request_id) != str(request_id):
+                    continue
+                # print(f"request output: {request_output}")
+                top_ids = list(request_output.outputs[0].token_ids)[-1]
+                out_tokens.append(top_ids)
+                if top_ids >= self.speech_token_size:
+                    if top_ids == self.speech_token_size:
+                        speech_token_break = True
+                        break
+                    else:
+                        raise ValueError('should not get token {}'.format(top_ids))
+                # in stream mode, yield token one by one
+                yield top_ids
+
+            if not vllm_codec_engine.has_unfinished_requests() or speech_token_break:
+                break

cosyvoice/tokenizer/tokenizer.py

@@ -0,0 +1,279 @@
+import base64
+import os
+from functools import lru_cache
+from typing import Optional
+import torch
+from transformers import AutoTokenizer
+from whisper.tokenizer import Tokenizer
+
+import tiktoken
+
+LANGUAGES = {
+    "en": "english",
+    "zh": "chinese",
+    "de": "german",
+    "es": "spanish",
+    "ru": "russian",
+    "ko": "korean",
+    "fr": "french",
+    "ja": "japanese",
+    "pt": "portuguese",
+    "tr": "turkish",
+    "pl": "polish",
+    "ca": "catalan",
+    "nl": "dutch",
+    "ar": "arabic",
+    "sv": "swedish",
+    "it": "italian",
+    "id": "indonesian",
+    "hi": "hindi",
+    "fi": "finnish",
+    "vi": "vietnamese",
+    "he": "hebrew",
+    "uk": "ukrainian",
+    "el": "greek",
+    "ms": "malay",
+    "cs": "czech",
+    "ro": "romanian",
+    "da": "danish",
+    "hu": "hungarian",
+    "ta": "tamil",
+    "no": "norwegian",
+    "th": "thai",
+    "ur": "urdu",
+    "hr": "croatian",
+    "bg": "bulgarian",
+    "lt": "lithuanian",
+    "la": "latin",
+    "mi": "maori",
+    "ml": "malayalam",
+    "cy": "welsh",
+    "sk": "slovak",
+    "te": "telugu",
+    "fa": "persian",
+    "lv": "latvian",
+    "bn": "bengali",
+    "sr": "serbian",
+    "az": "azerbaijani",
+    "sl": "slovenian",
+    "kn": "kannada",
+    "et": "estonian",
+    "mk": "macedonian",
+    "br": "breton",
+    "eu": "basque",
+    "is": "icelandic",
+    "hy": "armenian",
+    "ne": "nepali",
+    "mn": "mongolian",
+    "bs": "bosnian",
+    "kk": "kazakh",
+    "sq": "albanian",
+    "sw": "swahili",
+    "gl": "galician",
+    "mr": "marathi",
+    "pa": "punjabi",
+    "si": "sinhala",
+    "km": "khmer",
+    "sn": "shona",
+    "yo": "yoruba",
+    "so": "somali",
+    "af": "afrikaans",
+    "oc": "occitan",
+    "ka": "georgian",
+    "be": "belarusian",
+    "tg": "tajik",
+    "sd": "sindhi",
+    "gu": "gujarati",
+    "am": "amharic",
+    "yi": "yiddish",
+    "lo": "lao",
+    "uz": "uzbek",
+    "fo": "faroese",
+    "ht": "haitian creole",
+    "ps": "pashto",
+    "tk": "turkmen",
+    "nn": "nynorsk",
+    "mt": "maltese",
+    "sa": "sanskrit",
+    "lb": "luxembourgish",
+    "my": "myanmar",
+    "bo": "tibetan",
+    "tl": "tagalog",
+    "mg": "malagasy",
+    "as": "assamese",
+    "tt": "tatar",
+    "haw": "hawaiian",
+    "ln": "lingala",
+    "ha": "hausa",
+    "ba": "bashkir",
+    "jw": "javanese",
+    "su": "sundanese",
+    "yue": "cantonese",
+    "minnan": "minnan",
+    "wuyu": "wuyu",
+    "dialect": "dialect",
+    "zh/en": "zh/en",
+    "en/zh": "en/zh",
+}
+
+# language code lookup by name, with a few language aliases
+TO_LANGUAGE_CODE = {
+    **{language: code for code, language in LANGUAGES.items()},
+    "burmese": "my",
+    "valencian": "ca",
+    "flemish": "nl",
+    "haitian": "ht",
+    "letzeburgesch": "lb",
+    "pushto": "ps",
+    "panjabi": "pa",
+    "moldavian": "ro",
+    "moldovan": "ro",
+    "sinhalese": "si",
+    "castilian": "es",
+    "mandarin": "zh",
+}
+
+AUDIO_EVENT = {
+    "ASR": "ASR",
+    "AED": "AED",
+    "SER": "SER",
+    "Speech": "Speech",
+    "/Speech": "/Speech",
+    "BGM": "BGM",
+    "/BGM": "/BGM",
+    "Laughter": "Laughter",
+    "/Laughter": "/Laughter",
+    "Applause": "Applause",
+    "/Applause": "/Applause",
+}
+
+EMOTION = {
+    "HAPPY": "HAPPY",
+    "SAD": "SAD",
+    "ANGRY": "ANGRY",
+    "NEUTRAL": "NEUTRAL",
+}
+
+TTS_Vocal_Token = {
+    "TTS/B": "TTS/B",
+    "TTS/O": "TTS/O",
+    "TTS/Q": "TTS/Q",
+    "TTS/A": "TTS/A",
+    "TTS/CO": "TTS/CO",
+    "TTS/CL": "TTS/CL",
+    "TTS/H": "TTS/H",
+    **{f"TTS/SP{i:02d}": f"TTS/SP{i:02d}" for i in range(1, 14)}
+}
+
+
+@lru_cache(maxsize=None)
+def get_encoding(name: str = "gpt2", num_languages: int = 99):
+    vocab_path = os.path.join(os.path.dirname(__file__), "assets", f"{name}.tiktoken")
+    ranks = {
+        base64.b64decode(token): int(rank)
+        for token, rank in (line.split() for line in open(vocab_path) if line)
+    }
+    n_vocab = len(ranks)
+    special_tokens = {}
+
+    specials = [
+        "<|endoftext|>",
+        "<|startoftranscript|>",
+        *[f"<|{lang}|>" for lang in list(LANGUAGES.keys())[:num_languages]],
+        *[f"<|{audio_event}|>" for audio_event in list(AUDIO_EVENT.keys())],
+        *[f"<|{emotion}|>" for emotion in list(EMOTION.keys())],
+        "<|translate|>",
+        "<|transcribe|>",
+        "<|startoflm|>",
+        "<|startofprev|>",
+        "<|nospeech|>",
+        "<|notimestamps|>",
+        *[f"<|SPECIAL_TOKEN_{i}|>" for i in range(1, 31)],        # register special tokens for ASR
+        *[f"<|{tts}|>" for tts in list(TTS_Vocal_Token.keys())],  # register special tokens for TTS
+        *[f"<|{i * 0.02:.2f}|>" for i in range(1501)],
+    ]
+
+    for token in specials:
+        special_tokens[token] = n_vocab
+        n_vocab += 1
+
+    return tiktoken.Encoding(
+        name=os.path.basename(vocab_path),
+        explicit_n_vocab=n_vocab,
+        pat_str=r"""'s|'t|'re|'ve|'m|'ll|'d| ?\p{L}+| ?\p{N}+| ?[^\s\p{L}\p{N}]+|\s+(?!\S)|\s+""",
+        mergeable_ranks=ranks,
+        special_tokens=special_tokens,
+    )
+
+
+@lru_cache(maxsize=None)
+def get_tokenizer(
+    multilingual: bool,
+    *,
+    num_languages: int = 99,
+    language: Optional[str] = None,
+    task: Optional[str] = None,  # Literal["transcribe", "translate", None]
+) -> Tokenizer:
+    if language is not None:
+        language = language.lower()
+        if language not in LANGUAGES:
+            if language in TO_LANGUAGE_CODE:
+                language = TO_LANGUAGE_CODE[language]
+            else:
+                raise ValueError(f"Unsupported language: {language}")
+
+    if multilingual:
+        encoding_name = "multilingual_zh_ja_yue_char_del"
+        language = language or "en"
+        task = task or "transcribe"
+    else:
+        encoding_name = "gpt2"
+        language = None
+        task = None
+
+    encoding = get_encoding(name=encoding_name, num_languages=num_languages)
+
+    return Tokenizer(
+        encoding=encoding, num_languages=num_languages, language=language, task=task
+    )
+
+
+class QwenTokenizer():
+    def __init__(self, token_path, skip_special_tokens=True):
+        super().__init__()
+        # 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]"
+            ]
+        }
+        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
+
+    def encode(self, text, **kwargs):
+        tokens = self.tokenizer([text], return_tensors="pt")
+        tokens = tokens["input_ids"][0].cpu().tolist()
+        return tokens
+
+    def decode(self, tokens):
+        tokens = torch.tensor(tokens, dtype=torch.int64)
+        text = self.tokenizer.batch_decode([tokens], skip_special_tokens=self.skip_special_tokens)[0]
+        return text
+
+
+@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)

cosyvoice/transformer/attention.py

@@ -222,7 +222,7 @@ class RelPositionMultiHeadedAttention(MultiHeadedAttention):
         torch.nn.init.xavier_uniform_(self.pos_bias_u)
         torch.nn.init.xavier_uniform_(self.pos_bias_v)
 
-    def rel_shift(self, x):
+    def rel_shift(self, x: torch.Tensor) -> torch.Tensor:
         """Compute relative positional encoding.
 
         Args:
@@ -233,10 +233,14 @@ class RelPositionMultiHeadedAttention(MultiHeadedAttention):
             torch.Tensor: Output tensor.
 
         """
-        zero_pad = torch.zeros((*x.size()[:3], 1), device=x.device, dtype=x.dtype)
+        zero_pad = torch.zeros((x.size()[0], x.size()[1], x.size()[2], 1),
+                               device=x.device,
+                               dtype=x.dtype)
         x_padded = torch.cat([zero_pad, x], dim=-1)
 
-        x_padded = x_padded.view(*x.size()[:2], x.size(3) + 1, x.size(2))
+        x_padded = x_padded.view(x.size()[0],
+                                 x.size()[1],
+                                 x.size(3) + 1, x.size(2))
         x = x_padded[:, :, 1:].view_as(x)[
             :, :, :, : x.size(-1) // 2 + 1
         ]  # only keep the positions from 0 to time2

cosyvoice/transformer/decoder.py

@@ -174,7 +174,7 @@ class TransformerDecoder(torch.nn.Module):
                                                      memory_mask)
         return x
 
-    @torch.jit.ignore(drop=True)
+    @torch.jit.unused
     def forward_layers_checkpointed(self, x: torch.Tensor,
                                     tgt_mask: torch.Tensor,
                                     memory: torch.Tensor,

cosyvoice/transformer/embedding.py

@@ -212,7 +212,7 @@ class EspnetRelPositionalEncoding(torch.nn.Module):
 
     """
 
-    def __init__(self, d_model, dropout_rate, max_len=5000):
+    def __init__(self, d_model: int, dropout_rate: float, max_len: int = 5000):
         """Construct an PositionalEncoding object."""
         super(EspnetRelPositionalEncoding, self).__init__()
         self.d_model = d_model
@@ -221,7 +221,7 @@ class EspnetRelPositionalEncoding(torch.nn.Module):
         self.pe = None
         self.extend_pe(torch.tensor(0.0).expand(1, max_len))
 
-    def extend_pe(self, x):
+    def extend_pe(self, x: torch.Tensor):
         """Reset the positional encodings."""
         if self.pe is not None:
             # self.pe contains both positive and negative parts
@@ -253,7 +253,8 @@ class EspnetRelPositionalEncoding(torch.nn.Module):
         pe = torch.cat([pe_positive, pe_negative], dim=1)
         self.pe = pe.to(device=x.device, dtype=x.dtype)
 
-    def forward(self, x: torch.Tensor, offset: Union[int, torch.Tensor] = 0):
+    def forward(self, x: torch.Tensor, offset: Union[int, torch.Tensor] = 0) \
+            -> Tuple[torch.Tensor, torch.Tensor]:
         """Add positional encoding.
 
         Args:
@@ -288,6 +289,6 @@ class EspnetRelPositionalEncoding(torch.nn.Module):
         """
         pos_emb = self.pe[
             :,
-            self.pe.size(1) // 2 - size + 1 : self.pe.size(1) // 2 + size,
+            self.pe.size(1) // 2 - size + 1: self.pe.size(1) // 2 + size,
         ]
         return pos_emb

cosyvoice/transformer/encoder.py

@@ -169,7 +169,7 @@ class BaseEncoder(torch.nn.Module):
             xs, chunk_masks, _, _ = layer(xs, chunk_masks, pos_emb, mask_pad)
         return xs
 
-    @torch.jit.ignore(drop=True)
+    @torch.jit.unused
     def forward_layers_checkpointed(self, xs: torch.Tensor,
                                     chunk_masks: torch.Tensor,
                                     pos_emb: torch.Tensor,
@@ -180,6 +180,7 @@ class BaseEncoder(torch.nn.Module):
                                                     mask_pad)
         return xs
 
+    @torch.jit.export
     def forward_chunk(
         self,
         xs: torch.Tensor,
@@ -270,6 +271,7 @@ class BaseEncoder(torch.nn.Module):
 
         return (xs, r_att_cache, r_cnn_cache)
 
+    @torch.jit.unused
     def forward_chunk_by_chunk(
         self,
         xs: torch.Tensor,

cosyvoice/transformer/encoder_layer.py

@@ -49,8 +49,8 @@ class TransformerEncoderLayer(nn.Module):
         super().__init__()
         self.self_attn = self_attn
         self.feed_forward = feed_forward
-        self.norm1 = nn.LayerNorm(size, eps=1e-5)
+        self.norm1 = nn.LayerNorm(size, eps=1e-12)
-        self.norm2 = nn.LayerNorm(size, eps=1e-5)
+        self.norm2 = nn.LayerNorm(size, eps=1e-12)
         self.dropout = nn.Dropout(dropout_rate)
         self.size = size
         self.normalize_before = normalize_before
@@ -142,17 +142,17 @@ class ConformerEncoderLayer(nn.Module):
         self.feed_forward = feed_forward
         self.feed_forward_macaron = feed_forward_macaron
         self.conv_module = conv_module
-        self.norm_ff = nn.LayerNorm(size, eps=1e-5)  # for the FNN module
+        self.norm_ff = nn.LayerNorm(size, eps=1e-12)  # for the FNN module
-        self.norm_mha = nn.LayerNorm(size, eps=1e-5)  # for the MHA module
+        self.norm_mha = nn.LayerNorm(size, eps=1e-12)  # for the MHA module
         if feed_forward_macaron is not None:
-            self.norm_ff_macaron = nn.LayerNorm(size, eps=1e-5)
+            self.norm_ff_macaron = nn.LayerNorm(size, eps=1e-12)
             self.ff_scale = 0.5
         else:
             self.ff_scale = 1.0
         if self.conv_module is not None:
-            self.norm_conv = nn.LayerNorm(size, eps=1e-5)  # for the CNN module
+            self.norm_conv = nn.LayerNorm(size, eps=1e-12)  # for the CNN module
             self.norm_final = nn.LayerNorm(
-                size, eps=1e-5)  # for the final output of the block
+                size, eps=1e-12)  # for the final output of the block
         self.dropout = nn.Dropout(dropout_rate)
         self.size = size
         self.normalize_before = normalize_before

cosyvoice/transformer/upsample_encoder.py

@@ -0,0 +1,318 @@
+# Copyright (c) 2021 Mobvoi Inc (Binbin Zhang, Di Wu)
+#               2022 Xingchen Song (sxc19@mails.tsinghua.edu.cn)
+#               2024 Alibaba Inc (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.
+# Modified from ESPnet(https://github.com/espnet/espnet)
+"""Encoder definition."""
+from typing import Tuple
+
+import torch
+from torch import nn
+from torch.nn import functional as F
+
+from cosyvoice.transformer.convolution import ConvolutionModule
+from cosyvoice.transformer.encoder_layer import ConformerEncoderLayer
+from cosyvoice.transformer.positionwise_feed_forward import PositionwiseFeedForward
+from cosyvoice.utils.class_utils import (
+    COSYVOICE_EMB_CLASSES,
+    COSYVOICE_SUBSAMPLE_CLASSES,
+    COSYVOICE_ATTENTION_CLASSES,
+    COSYVOICE_ACTIVATION_CLASSES,
+)
+from cosyvoice.utils.mask import make_pad_mask
+from cosyvoice.utils.mask import add_optional_chunk_mask
+
+
+class Upsample1D(nn.Module):
+    """A 1D upsampling layer with an optional convolution.
+
+    Parameters:
+        channels (`int`):
+            number of channels in the inputs and outputs.
+        use_conv (`bool`, default `False`):
+            option to use a convolution.
+        use_conv_transpose (`bool`, default `False`):
+            option to use a convolution transpose.
+        out_channels (`int`, optional):
+            number of output channels. Defaults to `channels`.
+    """
+
+    def __init__(self, channels: int, out_channels: int, stride: int = 2):
+        super().__init__()
+        self.channels = channels
+        self.out_channels = out_channels
+        self.stride = stride
+        # In this mode, first repeat interpolate, than conv with stride=1
+        self.conv = nn.Conv1d(self.channels, self.out_channels, stride * 2 + 1, stride=1, padding=0)
+
+    def forward(self, inputs: torch.Tensor, input_lengths: torch.Tensor):
+        outputs = F.interpolate(inputs, scale_factor=float(self.stride), mode="nearest")
+        outputs = F.pad(outputs, (self.stride * 2, 0), value=0.0)
+        outputs = self.conv(outputs)
+        return outputs, input_lengths * self.stride
+
+
+class PreLookaheadLayer(nn.Module):
+    def __init__(self, channels: int, pre_lookahead_len: int = 1):
+        super().__init__()
+        self.channels = channels
+        self.pre_lookahead_len = pre_lookahead_len
+        self.conv1 = nn.Conv1d(
+            channels, channels,
+            kernel_size=pre_lookahead_len + 1,
+            stride=1, padding=0,
+        )
+        self.conv2 = nn.Conv1d(
+            channels, channels,
+            kernel_size=3, stride=1, padding=0,
+        )
+
+    def forward(self, inputs: torch.Tensor) -> torch.Tensor:
+        """
+        inputs: (batch_size, seq_len, channels)
+        """
+        outputs = inputs.transpose(1, 2).contiguous()
+        # look ahead
+        outputs = F.pad(outputs, (0, self.pre_lookahead_len), mode='constant', value=0.0)
+        outputs = F.leaky_relu(self.conv1(outputs))
+        # outputs
+        outputs = F.pad(outputs, (2, 0), mode='constant', value=0.0)
+        outputs = self.conv2(outputs)
+        outputs = outputs.transpose(1, 2).contiguous()
+
+        # residual connection
+        outputs = outputs + inputs
+        return outputs
+
+
+class UpsampleConformerEncoder(torch.nn.Module):
+
+    def __init__(
+        self,
+        input_size: int,
+        output_size: int = 256,
+        attention_heads: int = 4,
+        linear_units: int = 2048,
+        num_blocks: int = 6,
+        dropout_rate: float = 0.1,
+        positional_dropout_rate: float = 0.1,
+        attention_dropout_rate: float = 0.0,
+        input_layer: str = "conv2d",
+        pos_enc_layer_type: str = "rel_pos",
+        normalize_before: bool = True,
+        static_chunk_size: int = 0,
+        use_dynamic_chunk: bool = False,
+        global_cmvn: torch.nn.Module = None,
+        use_dynamic_left_chunk: bool = False,
+        positionwise_conv_kernel_size: int = 1,
+        macaron_style: bool = True,
+        selfattention_layer_type: str = "rel_selfattn",
+        activation_type: str = "swish",
+        use_cnn_module: bool = True,
+        cnn_module_kernel: int = 15,
+        causal: bool = False,
+        cnn_module_norm: str = "batch_norm",
+        key_bias: bool = True,
+        gradient_checkpointing: bool = False,
+    ):
+        """
+        Args:
+            input_size (int): input dim
+            output_size (int): dimension of attention
+            attention_heads (int): the number of heads of multi head attention
+            linear_units (int): the hidden units number of position-wise feed
+                forward
+            num_blocks (int): the number of decoder blocks
+            dropout_rate (float): dropout rate
+            attention_dropout_rate (float): dropout rate in attention
+            positional_dropout_rate (float): dropout rate after adding
+                positional encoding
+            input_layer (str): input layer type.
+                optional [linear, conv2d, conv2d6, conv2d8]
+            pos_enc_layer_type (str): Encoder positional encoding layer type.
+                opitonal [abs_pos, scaled_abs_pos, rel_pos, no_pos]
+            normalize_before (bool):
+                True: use layer_norm before each sub-block of a layer.
+                False: use layer_norm after each sub-block of a layer.
+            static_chunk_size (int): chunk size for static chunk training and
+                decoding
+            use_dynamic_chunk (bool): whether use dynamic chunk size for
+                training or not, You can only use fixed chunk(chunk_size > 0)
+                or dyanmic chunk size(use_dynamic_chunk = True)
+            global_cmvn (Optional[torch.nn.Module]): Optional GlobalCMVN module
+            use_dynamic_left_chunk (bool): whether use dynamic left chunk in
+                dynamic chunk training
+            key_bias: whether use bias in attention.linear_k, False for whisper models.
+            gradient_checkpointing: rerunning a forward-pass segment for each
+                checkpointed segment during backward.
+        """
+        super().__init__()
+        self._output_size = output_size
+
+        self.global_cmvn = global_cmvn
+        self.embed = COSYVOICE_SUBSAMPLE_CLASSES[input_layer](
+            input_size,
+            output_size,
+            dropout_rate,
+            COSYVOICE_EMB_CLASSES[pos_enc_layer_type](output_size,
+                                                      positional_dropout_rate),
+        )
+
+        self.normalize_before = normalize_before
+        self.after_norm = torch.nn.LayerNorm(output_size, eps=1e-5)
+        self.static_chunk_size = static_chunk_size
+        self.use_dynamic_chunk = use_dynamic_chunk
+        self.use_dynamic_left_chunk = use_dynamic_left_chunk
+        self.gradient_checkpointing = gradient_checkpointing
+        activation = COSYVOICE_ACTIVATION_CLASSES[activation_type]()
+        # self-attention module definition
+        encoder_selfattn_layer_args = (
+            attention_heads,
+            output_size,
+            attention_dropout_rate,
+            key_bias,
+        )
+        # feed-forward module definition
+        positionwise_layer_args = (
+            output_size,
+            linear_units,
+            dropout_rate,
+            activation,
+        )
+        # 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.encoders = torch.nn.ModuleList([
+            ConformerEncoderLayer(
+                output_size,
+                COSYVOICE_ATTENTION_CLASSES[selfattention_layer_type](
+                    *encoder_selfattn_layer_args),
+                PositionwiseFeedForward(*positionwise_layer_args),
+                PositionwiseFeedForward(
+                    *positionwise_layer_args) if macaron_style else None,
+                ConvolutionModule(
+                    *convolution_layer_args) if use_cnn_module else None,
+                dropout_rate,
+                normalize_before,
+            ) for _ in range(num_blocks)
+        ])
+        self.up_layer = Upsample1D(channels=512, out_channels=512, stride=2)
+        self.up_embed = COSYVOICE_SUBSAMPLE_CLASSES[input_layer](
+            input_size,
+            output_size,
+            dropout_rate,
+            COSYVOICE_EMB_CLASSES[pos_enc_layer_type](output_size,
+                                                      positional_dropout_rate),
+        )
+        self.up_encoders = torch.nn.ModuleList([
+            ConformerEncoderLayer(
+                output_size,
+                COSYVOICE_ATTENTION_CLASSES[selfattention_layer_type](
+                    *encoder_selfattn_layer_args),
+                PositionwiseFeedForward(*positionwise_layer_args),
+                PositionwiseFeedForward(
+                    *positionwise_layer_args) if macaron_style else None,
+                ConvolutionModule(
+                    *convolution_layer_args) if use_cnn_module else None,
+                dropout_rate,
+                normalize_before,
+            ) for _ in range(4)
+        ])
+
+    def output_size(self) -> int:
+        return self._output_size
+
+    def forward(
+        self,
+        xs: torch.Tensor,
+        xs_lens: torch.Tensor,
+        decoding_chunk_size: int = 0,
+        num_decoding_left_chunks: int = -1,
+    ) -> Tuple[torch.Tensor, torch.Tensor]:
+        """Embed positions in tensor.
+
+        Args:
+            xs: padded input tensor (B, T, D)
+            xs_lens: input length (B)
+            decoding_chunk_size: decoding chunk size for dynamic chunk
+                0: default for training, use random dynamic chunk.
+                <0: for decoding, use full chunk.
+                >0: for decoding, use fixed chunk size as set.
+            num_decoding_left_chunks: number of left chunks, this is for decoding,
+            the chunk size is decoding_chunk_size.
+                >=0: use num_decoding_left_chunks
+                <0: use all left chunks
+        Returns:
+            encoder output tensor xs, and subsampled masks
+            xs: padded output tensor (B, T' ~= T/subsample_rate, D)
+            masks: torch.Tensor batch padding mask after subsample
+                (B, 1, T' ~= T/subsample_rate)
+        NOTE(xcsong):
+            We pass the `__call__` method of the modules instead of `forward` to the
+            checkpointing API because `__call__` attaches all the hooks of the module.
+            https://discuss.pytorch.org/t/any-different-between-model-input-and-model-forward-input/3690/2
+        """
+        T = xs.size(1)
+        masks = ~make_pad_mask(xs_lens, T).unsqueeze(1)  # (B, 1, T)
+        if self.global_cmvn is not None:
+            xs = self.global_cmvn(xs)
+        xs, pos_emb, masks = self.embed(xs, masks)
+        mask_pad = masks  # (B, 1, T/subsample_rate)
+        chunk_masks = add_optional_chunk_mask(xs, masks,
+                                              self.use_dynamic_chunk,
+                                              self.use_dynamic_left_chunk,
+                                              decoding_chunk_size,
+                                              self.static_chunk_size,
+                                              num_decoding_left_chunks)
+        # lookahead + conformer encoder
+        xs = self.pre_lookahead_layer(xs)
+        xs = self.forward_layers(xs, chunk_masks, pos_emb, mask_pad)
+
+        # upsample + conformer encoder
+        xs = xs.transpose(1, 2).contiguous()
+        xs, xs_lens = self.up_layer(xs, xs_lens)
+        xs = xs.transpose(1, 2).contiguous()
+        T = xs.size(1)
+        masks = ~make_pad_mask(xs_lens, T).unsqueeze(1)  # (B, 1, T)
+        xs, pos_emb, masks = self.up_embed(xs, masks)
+        mask_pad = masks  # (B, 1, T/subsample_rate)
+        chunk_masks = add_optional_chunk_mask(xs, masks,
+                                              self.use_dynamic_chunk,
+                                              self.use_dynamic_left_chunk,
+                                              decoding_chunk_size,
+                                              self.static_chunk_size * self.up_layer.stride,
+                                              num_decoding_left_chunks)
+        xs = self.forward_up_layers(xs, chunk_masks, pos_emb, mask_pad)
+
+        if self.normalize_before:
+            xs = self.after_norm(xs)
+        # Here we assume the mask is not changed in encoder layers, so just
+        # return the masks before encoder layers, and the masks will be used
+        # for cross attention with decoder later
+        return xs, masks
+
+    def forward_layers(self, xs: torch.Tensor, chunk_masks: torch.Tensor,
+                       pos_emb: torch.Tensor,
+                       mask_pad: torch.Tensor) -> torch.Tensor:
+        for layer in self.encoders:
+            xs, chunk_masks, _, _ = layer(xs, chunk_masks, pos_emb, mask_pad)
+        return xs
+
+    def forward_up_layers(self, xs: torch.Tensor, chunk_masks: torch.Tensor,
+                          pos_emb: torch.Tensor,
+                          mask_pad: torch.Tensor) -> torch.Tensor:
+        for layer in self.up_encoders:
+            xs, chunk_masks, _, _ = layer(xs, chunk_masks, pos_emb, mask_pad)
+        return xs

cosyvoice/utils/class_utils.py

@@ -32,6 +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
 
 
 COSYVOICE_ACTIVATION_CLASSES = {
@@ -68,3 +72,12 @@ COSYVOICE_ATTENTION_CLASSES = {
     "selfattn": MultiHeadedAttention,
     "rel_selfattn": RelPositionMultiHeadedAttention,
 }
+
+
+def get_model_type(configs):
+    # NOTE CosyVoice2Model inherits CosyVoiceModel
+    if isinstance(configs['llm'], TransformerLM) and isinstance(configs['flow'], MaskedDiffWithXvec) and isinstance(configs['hift'], HiFTGenerator):
+        return CosyVoiceModel
+    if isinstance(configs['llm'], Qwen2LM) and isinstance(configs['flow'], CausalMaskedDiffWithXvec) and isinstance(configs['hift'], HiFTGenerator):
+        return CosyVoice2Model
+    raise TypeError('No valid model type found!')

cosyvoice/utils/common.py

@@ -15,8 +15,10 @@
 # Modified from ESPnet(https://github.com/espnet/espnet)
 """Unility functions for Transformer."""
 
+import random
 from typing import List
 
+import numpy as np
 import torch
 
 IGNORE_ID = -1
@@ -101,3 +103,64 @@ def init_weights(m, mean=0.0, std=0.01):
     classname = m.__class__.__name__
     if classname.find("Conv") != -1:
         m.weight.data.normal_(mean, std)
+
+
+# Repetition Aware Sampling in VALL-E 2
+def ras_sampling(weighted_scores, decoded_tokens, sampling, top_p=0.8, top_k=25, win_size=10, tau_r=0.1):
+    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:
+        top_ids = random_sampling(weighted_scores, decoded_tokens, sampling)
+    return top_ids
+
+
+def nucleus_sampling(weighted_scores, top_p=0.8, top_k=25):
+    prob, indices = [], []
+    cum_prob = 0.0
+    sorted_value, sorted_idx = weighted_scores.softmax(dim=0).sort(descending=True, stable=True)
+    for i in range(len(sorted_idx)):
+        # sampling both top-p and numbers.
+        if cum_prob < top_p and len(prob) < top_k:
+            cum_prob += sorted_value[i]
+            prob.append(sorted_value[i])
+            indices.append(sorted_idx[i])
+        else:
+            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)]
+    return top_ids
+
+
+def random_sampling(weighted_scores, decoded_tokens, sampling):
+    top_ids = weighted_scores.softmax(dim=0).multinomial(1, replacement=True)
+    return top_ids
+
+
+def fade_in_out(fade_in_mel, fade_out_mel, window):
+    device = fade_in_mel.device
+    fade_in_mel, fade_out_mel = fade_in_mel.cpu(), fade_out_mel.cpu()
+    mel_overlap_len = int(window.shape[0] / 2)
+    if fade_in_mel.device == torch.device('cpu'):
+        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.to(device)
+
+
+def set_all_random_seed(seed):
+    random.seed(seed)
+    np.random.seed(seed)
+    torch.manual_seed(seed)
+    torch.cuda.manual_seed_all(seed)
+
+
+def mask_to_bias(mask: torch.Tensor, dtype: torch.dtype) -> torch.Tensor:
+    assert mask.dtype == torch.bool
+    assert dtype in [torch.float32, torch.bfloat16, torch.float16]
+    mask = mask.to(dtype)
+    # attention mask bias
+    # NOTE(Mddct): torch.finfo jit issues
+    #     chunk_masks = (1.0 - chunk_masks) * torch.finfo(dtype).min
+    mask = (1.0 - mask) * -1.0e+10
+    return mask

cosyvoice/utils/executor.py

@@ -25,13 +25,14 @@ from cosyvoice.utils.train_utils import update_parameter_and_lr, log_per_step, l
 
 class Executor:
 
-    def __init__(self):
+    def __init__(self, gan: bool = False):
+        self.gan = gan
         self.step = 0
         self.epoch = 0
         self.rank = int(os.environ.get('RANK', 0))
         self.device = torch.device('cuda:{}'.format(self.rank))
 
-    def train_one_epoc(self, model, optimizer, scheduler, train_data_loader, cv_data_loader, writer, info_dict, group_join):
+    def train_one_epoc(self, model, optimizer, scheduler, train_data_loader, cv_data_loader, writer, info_dict, scaler, group_join):
         ''' Train one epoch
         '''
 
@@ -64,13 +65,72 @@ class Executor:
                     context = nullcontext
 
                 with context():
-                    info_dict = batch_forward(model, batch_dict, info_dict)
+                    info_dict = batch_forward(model, batch_dict, scaler, info_dict)
-                    info_dict = batch_backward(model, info_dict)
+                    info_dict = batch_backward(model, scaler, info_dict)
 
-                info_dict = update_parameter_and_lr(model, optimizer, scheduler, info_dict)
+                info_dict = update_parameter_and_lr(model, optimizer, scheduler, scaler, info_dict)
                 log_per_step(writer, info_dict)
                 # NOTE specify save_per_step in cosyvoice.yaml if you want to enable step save
-                if info_dict['save_per_step'] > 0 and (self.step + 1) % info_dict['save_per_step'] == 0 and (batch_idx + 1) % info_dict["accum_grad"] == 0:
+                if info_dict['save_per_step'] > 0 and (self.step + 1) % info_dict['save_per_step'] == 0 and \
+                   (batch_idx + 1) % info_dict["accum_grad"] == 0:
+                    dist.barrier()
+                    self.cv(model, cv_data_loader, writer, info_dict, on_batch_end=False)
+                    model.train()
+                if (batch_idx + 1) % info_dict["accum_grad"] == 0:
+                    self.step += 1
+        dist.barrier()
+        self.cv(model, cv_data_loader, writer, info_dict, on_batch_end=True)
+
+    def train_one_epoc_gan(self, model, optimizer, scheduler, optimizer_d, scheduler_d, train_data_loader, cv_data_loader,
+                           writer, info_dict, scaler, group_join):
+        ''' Train one epoch
+        '''
+
+        lr = optimizer.param_groups[0]['lr']
+        logging.info('Epoch {} TRAIN info lr {} rank {}'.format(self.epoch, lr, self.rank))
+        logging.info('using accumulate grad, new batch size is {} times'
+                     ' larger than before'.format(info_dict['accum_grad']))
+        # A context manager to be used in conjunction with an instance of
+        # torch.nn.parallel.DistributedDataParallel to be able to train
+        # with uneven inputs across participating processes.
+        model.train()
+        model_context = model.join if info_dict['train_engine'] == 'torch_ddp' else nullcontext
+        with model_context():
+            for batch_idx, batch_dict in enumerate(train_data_loader):
+                info_dict["tag"] = "TRAIN"
+                info_dict["step"] = self.step
+                info_dict["epoch"] = self.epoch
+                info_dict["batch_idx"] = batch_idx
+                if cosyvoice_join(group_join, info_dict):
+                    break
+
+                # Disable gradient synchronizations across DDP processes.
+                # Within this context, gradients will be accumulated on module
+                # variables, which will later be synchronized.
+                if info_dict['train_engine'] == 'torch_ddp' and (batch_idx + 1) % info_dict["accum_grad"] != 0:
+                    context = model.no_sync
+                # Used for single gpu training and DDP gradient synchronization
+                # processes.
+                else:
+                    context = nullcontext
+
+                with context():
+                    batch_dict['turn'] = 'discriminator'
+                    info_dict = batch_forward(model, batch_dict, scaler, info_dict)
+                    info_dict = batch_backward(model, scaler, info_dict)
+                info_dict = update_parameter_and_lr(model, optimizer_d, scheduler_d, scaler, info_dict)
+                optimizer.zero_grad()
+                log_per_step(writer, info_dict)
+                with context():
+                    batch_dict['turn'] = 'generator'
+                    info_dict = batch_forward(model, batch_dict, scaler, info_dict)
+                    info_dict = batch_backward(model, scaler, info_dict)
+                info_dict = update_parameter_and_lr(model, optimizer, scheduler, scaler, info_dict)
+                optimizer_d.zero_grad()
+                log_per_step(writer, info_dict)
+                # NOTE specify save_per_step in cosyvoice.yaml if you want to enable step save
+                if info_dict['save_per_step'] > 0 and (self.step + 1) % info_dict['save_per_step'] == 0 and \
+                   (batch_idx + 1) % info_dict["accum_grad"] == 0:
                     dist.barrier()
                     self.cv(model, cv_data_loader, writer, info_dict, on_batch_end=False)
                     model.train()
@@ -95,7 +155,9 @@ class Executor:
             num_utts = len(batch_dict["utts"])
             total_num_utts += num_utts
 
-            info_dict = batch_forward(model, batch_dict, info_dict)
+            if self.gan is True:
+                batch_dict['turn'] = 'generator'
+            info_dict = batch_forward(model, batch_dict, None, info_dict)
 
             for k, v in info_dict['loss_dict'].items():
                 if k not in total_loss_dict:

cosyvoice/utils/file_utils.py

@@ -1,5 +1,5 @@
 # Copyright (c) 2021 Mobvoi Inc. (authors: Binbin Zhang)
-#               2024 Alibaba Inc (authors: Xiang Lyu)
+#               2024 Alibaba Inc (authors: Xiang Lyu, Zetao Hu)
 #
 # Licensed under the Apache License, Version 2.0 (the "License");
 # you may not use this file except in compliance with the License.
@@ -15,6 +15,10 @@
 
 import json
 import torchaudio
+import logging
+logging.getLogger('matplotlib').setLevel(logging.WARNING)
+logging.basicConfig(level=logging.DEBUG,
+                    format='%(asctime)s %(levelname)s %(message)s')
 
 
 def read_lists(list_file):
@@ -24,6 +28,7 @@ def read_lists(list_file):
             lists.append(line.strip())
     return lists
 
+
 def read_json_lists(list_file):
     lists = read_lists(list_file)
     results = {}
@@ -32,22 +37,53 @@ def read_json_lists(list_file):
             results.update(json.load(fin))
     return results
 
+
 def load_wav(wav, target_sr):
-    speech, sample_rate = torchaudio.load(wav)
+    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)
         speech = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=target_sr)(speech)
     return speech
 
-def speed_change(waveform, sample_rate, speed_factor: str):
+
-    effects = [
+def convert_onnx_to_trt(trt_model, onnx_model, fp16):
-        ["tempo", speed_factor],  # speed_factor
+    import tensorrt as trt
-        ["rate", f"{sample_rate}"]
+    _min_shape = [(2, 80, 4), (2, 1, 4), (2, 80, 4), (2,), (2, 80), (2, 80, 4)]
-    ]
+    _opt_shape = [(2, 80, 193), (2, 1, 193), (2, 80, 193), (2,), (2, 80), (2, 80, 193)]
-    augmented_waveform, new_sample_rate = torchaudio.sox_effects.apply_effects_tensor(
+    _max_shape = [(2, 80, 6800), (2, 1, 6800), (2, 80, 6800), (2,), (2, 80), (2, 80, 6800)]
-        waveform,
+    input_names = ["x", "mask", "mu", "t", "spks", "cond"]
-        sample_rate,
+
-        effects
+    logging.info("Converting onnx to trt...")
-    )
+    network_flags = 1 << int(trt.NetworkDefinitionCreationFlag.EXPLICIT_BATCH)
-    return augmented_waveform, new_sample_rate
+    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 << 33)  # 8GB
+    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(input_names)):
+        profile.set_shape(input_names[i], _min_shape[i], _opt_shape[i], _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)

cosyvoice/utils/frontend_utils.py

@@ -13,8 +13,10 @@
 # limitations under the License.
 
 import re
+import regex
 chinese_char_pattern = re.compile(r'[\u4e00-\u9fff]+')
 
+
 # whether contain chinese character
 def contains_chinese(text):
     return bool(chinese_char_pattern.search(text))
@@ -79,6 +81,13 @@ def split_paragraph(text: str, tokenize, lang="zh", token_max_n=80, token_min_n=
         pounc = ['.', '?', '!', ';', ':']
     if comma_split:
         pounc.extend(['，', ','])
+
+    if text[-1] not in pounc:
+        if lang == "zh":
+            text += "。"
+        else:
+            text += "."
+
     st = 0
     utts = []
     for i, c in enumerate(text):
@@ -91,11 +100,7 @@ def split_paragraph(text: str, tokenize, lang="zh", token_max_n=80, token_min_n=
                 st = i + 2
             else:
                 st = i + 1
-    if len(utts) == 0:
+
-        if lang == "zh":
-            utts.append(text + '。')
-        else:
-            utts.append(text + '.')
     final_utts = []
     cur_utt = ""
     for utt in utts:
@@ -123,3 +128,9 @@ def replace_blank(text: str):
         else:
             out_str.append(c)
     return "".join(out_str)
+
+
+def is_only_punctuation(text):
+    # Regular expression: Match strings that consist only of punctuation marks or are empty.
+    punctuation_pattern = r'^[\p{P}\p{S}]*$'
+    return bool(regex.fullmatch(punctuation_pattern, text))

cosyvoice/utils/losses.py

@@ -0,0 +1,20 @@
+import torch
+import torch.nn.functional as F
+
+
+def tpr_loss(disc_real_outputs, disc_generated_outputs, tau):
+    loss = 0
+    for dr, dg in zip(disc_real_outputs, disc_generated_outputs):
+        m_DG = torch.median((dr - dg))
+        L_rel = torch.mean((((dr - dg) - m_DG) ** 2)[dr < dg + m_DG])
+        loss += tau - F.relu(tau - L_rel)
+    return loss
+
+
+def mel_loss(real_speech, generated_speech, mel_transforms):
+    loss = 0
+    for transform in mel_transforms:
+        mel_r = transform(real_speech)
+        mel_g = transform(generated_speech)
+        loss += F.l1_loss(mel_g, mel_r)
+    return loss

cosyvoice/utils/mask.py

@@ -15,6 +15,7 @@
 # limitations under the License.
 
 import torch
+from cosyvoice.utils.file_utils import logging
 '''
 def subsequent_mask(
         size: int,
@@ -86,7 +87,7 @@ def subsequent_mask(
     return mask
 
 
-def subsequent_chunk_mask(
+def subsequent_chunk_mask_deprecated(
         size: int,
         chunk_size: int,
         num_left_chunks: int = -1,
@@ -124,6 +125,41 @@ def subsequent_chunk_mask(
     return ret
 
 
+def subsequent_chunk_mask(
+        size: int,
+        chunk_size: int,
+        num_left_chunks: int = -1,
+        device: torch.device = torch.device("cpu"),
+) -> torch.Tensor:
+    """Create mask for subsequent steps (size, size) with chunk size,
+       this is for streaming encoder
+
+    Args:
+        size (int): size of mask
+        chunk_size (int): size of chunk
+        num_left_chunks (int): number of left chunks
+            <0: use full chunk
+            >=0: use num_left_chunks
+        device (torch.device): "cpu" or "cuda" or torch.Tensor.device
+
+    Returns:
+        torch.Tensor: mask
+
+    Examples:
+        >>> subsequent_chunk_mask(4, 2)
+        [[1, 1, 0, 0],
+         [1, 1, 0, 0],
+         [1, 1, 1, 1],
+         [1, 1, 1, 1]]
+    """
+    # NOTE this modified implementation meets onnx export requirements, but it doesn't support num_left_chunks
+    # actually this is not needed after we have inference cache implemented, will remove it later
+    pos_idx = torch.arange(size, device=device)
+    block_value = (torch.div(pos_idx, chunk_size, rounding_mode='trunc') + 1) * chunk_size
+    ret = pos_idx.unsqueeze(0) < block_value.unsqueeze(1)
+    return ret
+
+
 def add_optional_chunk_mask(xs: torch.Tensor,
                             masks: torch.Tensor,
                             use_dynamic_chunk: bool,
@@ -195,6 +231,10 @@ def add_optional_chunk_mask(xs: torch.Tensor,
         chunk_masks = masks & chunk_masks  # (B, L, L)
     else:
         chunk_masks = masks
+    assert chunk_masks.dtype == torch.bool
+    if (chunk_masks.sum(dim=-1) == 0).sum().item() != 0:
+        logging.warning('get chunk_masks all false at some timestep, force set to true, make sure they are masked in futuer computation!')
+        chunk_masks[chunk_masks.sum(dim=-1)==0] = True
     return chunk_masks
 
 

cosyvoice/utils/scheduler.py

@@ -567,8 +567,7 @@ class NoamAnnealing(_LRScheduler):
                  min_lr=0.0,
                  last_epoch=-1):
         self._normalize = d_model**(-0.5)
-        assert not (warmup_steps is not None
+        assert not (warmup_steps is not None and warmup_ratio is not None), \
-                    and warmup_ratio is not None), \
             "Either use particular number of step or ratio"
         assert warmup_ratio is None or max_steps is not None, \
             "If there is a ratio, there should be a total steps"

cosyvoice/utils/train_utils.py

@@ -14,7 +14,6 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 
-from contextlib import nullcontext
 import logging
 import os
 import torch
@@ -51,9 +50,10 @@ def init_distributed(args):
     return world_size, local_rank, rank
 
 
-def init_dataset_and_dataloader(args, configs):
+def init_dataset_and_dataloader(args, configs, gan):
-    train_dataset = Dataset(args.train_data, data_pipeline=configs['data_pipeline'], mode='train', shuffle=True, partition=True)
+    data_pipeline = configs['data_pipeline_gan'] if gan is True else configs['data_pipeline']
-    cv_dataset = Dataset(args.cv_data, data_pipeline=configs['data_pipeline'], mode='train', shuffle=False, partition=False)
+    train_dataset = Dataset(args.train_data, data_pipeline=data_pipeline, mode='train', gan=gan, shuffle=True, partition=True)
+    cv_dataset = Dataset(args.cv_data, data_pipeline=data_pipeline, mode='train', gan=gan, 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,
@@ -69,7 +69,6 @@ def init_dataset_and_dataloader(args, configs):
     return train_dataset, cv_dataset, train_data_loader, cv_data_loader
 
 
-
 def check_modify_and_save_config(args, configs):
     if args.train_engine == "torch_ddp":
         configs['train_conf']["dtype"] = 'fp32'
@@ -84,7 +83,8 @@ def check_modify_and_save_config(args, configs):
             configs['train_conf']["dtype"] = "fp32"
         assert ds_configs["train_micro_batch_size_per_gpu"] == 1
         # if use deepspeed, override ddp config
-        configs['train_conf']['save_per_step'] = int(configs['train_conf']['save_per_step'] * configs['train_conf']['accum_grad'] / ds_configs["gradient_accumulation_steps"])
+        configs['train_conf']['save_per_step'] = int(configs['train_conf']['save_per_step'] *
+                                                     configs['train_conf']['accum_grad'] / ds_configs["gradient_accumulation_steps"])
         configs['train_conf']['accum_grad'] = ds_configs["gradient_accumulation_steps"]
         configs['train_conf']['grad_clip'] = ds_configs["gradient_clipping"]
         configs['train_conf']['log_interval'] = ds_configs["steps_per_print"]
@@ -108,38 +108,80 @@ def wrap_cuda_model(args, model):
     return model
 
 
-def init_optimizer_and_scheduler(args, configs, model):
+def init_optimizer_and_scheduler(args, configs, model, gan):
-    if configs['train_conf']['optim'] == 'adam':
+    if gan is False:
-        optimizer = optim.Adam(model.parameters(), **configs['train_conf']['optim_conf'])
+        if configs['train_conf']['optim'] == 'adam':
-    elif configs['train_conf']['optim'] == 'adamw':
+            optimizer = optim.Adam(model.parameters(), **configs['train_conf']['optim_conf'])
-        optimizer = optim.AdamW(model.parameters(), **configs['train_conf']['optim_conf'])
+        elif configs['train_conf']['optim'] == 'adamw':
+            optimizer = optim.AdamW(model.parameters(), **configs['train_conf']['optim_conf'])
+        else:
+            raise ValueError("unknown optimizer: " + configs['train_conf'])
+
+        if configs['train_conf']['scheduler'] == 'warmuplr':
+            scheduler_type = WarmupLR
+            scheduler = WarmupLR(optimizer, **configs['train_conf']['scheduler_conf'])
+        elif configs['train_conf']['scheduler'] == 'NoamHoldAnnealing':
+            scheduler_type = NoamHoldAnnealing
+            scheduler = NoamHoldAnnealing(optimizer, **configs['train_conf']['scheduler_conf'])
+        elif configs['train_conf']['scheduler'] == 'constantlr':
+            scheduler_type = ConstantLR
+            scheduler = ConstantLR(optimizer)
+        else:
+            raise ValueError("unknown scheduler: " + configs['train_conf'])
+
+        # use deepspeed optimizer for speedup
+        if args.train_engine == "deepspeed":
+            def scheduler(opt):
+                return scheduler_type(opt, **configs['train_conf']['scheduler_conf'])
+            model, optimizer, _, scheduler = deepspeed.initialize(
+                args=args,
+                model=model,
+                optimizer=None,
+                lr_scheduler=scheduler,
+                model_parameters=model.parameters())
+
+        optimizer_d, scheduler_d = None, None
+
     else:
-        raise ValueError("unknown optimizer: " + configs['train_conf'])
+        # currently we wrap generator and discriminator in one model, so we cannot use deepspeed
+        if configs['train_conf']['optim'] == 'adam':
+            optimizer = optim.Adam(model.module.generator.parameters(), **configs['train_conf']['optim_conf'])
+        elif configs['train_conf']['optim'] == 'adamw':
+            optimizer = optim.AdamW(model.module.generator.parameters(), **configs['train_conf']['optim_conf'])
+        else:
+            raise ValueError("unknown optimizer: " + configs['train_conf'])
 
-    if configs['train_conf']['scheduler'] == 'warmuplr':
+        if configs['train_conf']['scheduler'] == 'warmuplr':
-        scheduler_type = WarmupLR
+            scheduler_type = WarmupLR
-        scheduler = WarmupLR(optimizer, **configs['train_conf']['scheduler_conf'])
+            scheduler = WarmupLR(optimizer, **configs['train_conf']['scheduler_conf'])
-    elif configs['train_conf']['scheduler'] == 'NoamHoldAnnealing':
+        elif configs['train_conf']['scheduler'] == 'NoamHoldAnnealing':
-        scheduler_type = NoamHoldAnnealing
+            scheduler_type = NoamHoldAnnealing
-        scheduler = NoamHoldAnnealing(optimizer, **configs['train_conf']['scheduler_conf'])
+            scheduler = NoamHoldAnnealing(optimizer, **configs['train_conf']['scheduler_conf'])
-    elif configs['train_conf']['scheduler'] == 'constantlr':
+        elif configs['train_conf']['scheduler'] == 'constantlr':
-        scheduler_type = ConstantLR
+            scheduler_type = ConstantLR
-        scheduler = ConstantLR(optimizer)
+            scheduler = ConstantLR(optimizer)
-    else:
+        else:
-        raise ValueError("unknown scheduler: " + configs['train_conf'])
+            raise ValueError("unknown scheduler: " + configs['train_conf'])
 
-    # use deepspeed optimizer for speedup
+        if configs['train_conf']['optim_d'] == 'adam':
-    if args.train_engine == "deepspeed":
+            optimizer_d = optim.Adam(model.module.discriminator.parameters(), **configs['train_conf']['optim_conf'])
-        def scheduler(opt):
+        elif configs['train_conf']['optim_d'] == 'adamw':
-            return scheduler_type(opt, **configs['train_conf']['scheduler_conf'])
+            optimizer_d = optim.AdamW(model.module.discriminator.parameters(), **configs['train_conf']['optim_conf'])
-        model, optimizer, _, scheduler = deepspeed.initialize(
+        else:
-            args=args,
+            raise ValueError("unknown optimizer: " + configs['train_conf'])
-            model=model,
-            optimizer=None,
-            lr_scheduler=scheduler,
-            model_parameters=model.parameters())
 
-    return model, optimizer, scheduler
+        if configs['train_conf']['scheduler_d'] == 'warmuplr':
+            scheduler_type = WarmupLR
+            scheduler_d = WarmupLR(optimizer_d, **configs['train_conf']['scheduler_conf'])
+        elif configs['train_conf']['scheduler_d'] == 'NoamHoldAnnealing':
+            scheduler_type = NoamHoldAnnealing
+            scheduler_d = NoamHoldAnnealing(optimizer_d, **configs['train_conf']['scheduler_conf'])
+        elif configs['train_conf']['scheduler'] == 'constantlr':
+            scheduler_type = ConstantLR
+            scheduler_d = ConstantLR(optimizer_d)
+        else:
+            raise ValueError("unknown scheduler: " + configs['train_conf'])
+    return model, optimizer, scheduler, optimizer_d, scheduler_d
 
 
 def init_summarywriter(args):
@@ -157,7 +199,7 @@ def save_model(model, model_name, info_dict):
 
     if info_dict["train_engine"] == "torch_ddp":
         if rank == 0:
-            torch.save(model.module.state_dict(), save_model_path)
+            torch.save({**model.module.state_dict(), 'epoch': info_dict['epoch'], 'step': info_dict['step']}, save_model_path)
     else:
         with torch.no_grad():
             model.save_checkpoint(save_dir=model_dir,
@@ -193,7 +235,7 @@ def cosyvoice_join(group_join, info_dict):
         return False
 
 
-def batch_forward(model, batch, info_dict):
+def batch_forward(model, batch, scaler, info_dict):
     device = int(os.environ.get('LOCAL_RANK', 0))
 
     dtype = info_dict["dtype"]
@@ -205,7 +247,7 @@ def batch_forward(model, batch, info_dict):
         dtype = torch.float32
 
     if info_dict['train_engine'] == 'torch_ddp':
-        autocast = nullcontext()
+        autocast = torch.cuda.amp.autocast(enabled=scaler is not None)
     else:
         autocast = torch.cuda.amp.autocast(enabled=True, dtype=dtype, cache_enabled=False)
 
@@ -214,27 +256,41 @@ def batch_forward(model, batch, info_dict):
     return info_dict
 
 
-def batch_backward(model, info_dict):
+def batch_backward(model, scaler, info_dict):
     if info_dict["train_engine"] == "deepspeed":
         scaled_loss = model.backward(info_dict['loss_dict']['loss'])
     else:
         scaled_loss = info_dict['loss_dict']['loss'] / info_dict['accum_grad']
-        scaled_loss.backward()
+        if scaler is not None:
+            scaler.scale(scaled_loss).backward()
+        else:
+            scaled_loss.backward()
 
     info_dict['loss_dict']['loss'] = scaled_loss
     return info_dict
 
 
-def update_parameter_and_lr(model, optimizer, scheduler, info_dict):
+def update_parameter_and_lr(model, optimizer, scheduler, scaler, info_dict):
     grad_norm = 0.0
     if info_dict['train_engine'] == "deepspeed":
         info_dict["is_gradient_accumulation_boundary"] = model.is_gradient_accumulation_boundary()
         model.step()
         grad_norm = model.get_global_grad_norm()
     elif (info_dict['batch_idx'] + 1) % info_dict["accum_grad"] == 0:
-        grad_norm = clip_grad_norm_(model.parameters(), info_dict['grad_clip'])
+        # Use mixed precision training
-        if torch.isfinite(grad_norm):
+        if scaler is not None:
-            optimizer.step()
+            scaler.unscale_(optimizer)
+            grad_norm = clip_grad_norm_(model.parameters(), info_dict['grad_clip'])
+            # We don't check grad here since that if the gradient
+            # has inf/nan values, scaler.step will skip
+            # optimizer.step().
+            if torch.isfinite(grad_norm):
+                scaler.step(optimizer)
+            scaler.update()
+        else:
+            grad_norm = clip_grad_norm_(model.parameters(), info_dict['grad_clip'])
+            if torch.isfinite(grad_norm):
+                optimizer.step()
         optimizer.zero_grad()
         scheduler.step()
     info_dict["lr"] = optimizer.param_groups[0]['lr']

docker/Dockerfile

@@ -0,0 +1,51 @@
+FROM nvidia/cuda:12.4.1-cudnn-devel-ubuntu22.04
+
+ARG VENV_NAME="cosyvoice"
+ENV VENV=$VENV_NAME
+ENV LANG=C.UTF-8 LC_ALL=C.UTF-8
+
+ENV DEBIAN_FRONTEN=noninteractive
+ENV PYTHONUNBUFFERED=1
+SHELL ["/bin/bash", "--login", "-c"]
+
+RUN apt-get update -y --fix-missing
+RUN apt-get install -y git build-essential curl wget ffmpeg unzip git git-lfs sox libsox-dev && \
+    apt-get clean && \
+    git lfs install
+
+# ==================================================================
+# conda install and conda forge channel as default
+# ------------------------------------------------------------------
+# Install miniforge
+RUN wget --quiet https://github.com/conda-forge/miniforge/releases/latest/download/Miniforge3-Linux-x86_64.sh -O ~/miniforge.sh && \
+    /bin/bash ~/miniforge.sh -b -p /opt/conda && \
+    rm ~/miniforge.sh && \
+    ln -s /opt/conda/etc/profile.d/conda.sh /etc/profile.d/conda.sh && \
+    echo "source /opt/conda/etc/profile.d/conda.sh" >> /opt/nvidia/entrypoint.d/100.conda.sh && \
+    echo "source /opt/conda/etc/profile.d/conda.sh" >> ~/.bashrc && \
+    echo "conda activate ${VENV}" >> /opt/nvidia/entrypoint.d/110.conda_default_env.sh && \
+    echo "conda activate ${VENV}" >> $HOME/.bashrc
+
+ENV PATH /opt/conda/bin:$PATH
+
+RUN conda config --add channels conda-forge && \
+    conda config --set channel_priority strict
+# ------------------------------------------------------------------
+# ~conda
+# ==================================================================
+
+RUN conda create -y -n ${VENV} python=3.10
+ENV CONDA_DEFAULT_ENV=${VENV}
+ENV PATH /opt/conda/bin:/opt/conda/envs/${VENV}/bin:$PATH
+
+WORKDIR /workspace
+
+ENV PYTHONPATH="${PYTHONPATH}:/workspace/CosyVoice:/workspace/CosyVoice/third_party/Matcha-TTS"
+
+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
+
+WORKDIR /workspace/CosyVoice

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

@@ -18,7 +18,7 @@ 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
+    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
@@ -31,7 +31,7 @@ llm: !new:cosyvoice.llm.llm.TransformerLM
         num_blocks: 3
         dropout_rate: 0.1
         positional_dropout_rate: 0.1
-        attention_dropout_rate: 0
+        attention_dropout_rate: 0.0
         normalize_before: True
         input_layer: 'linear'
         pos_enc_layer_type: 'rel_pos_espnet'
@@ -49,11 +49,16 @@ llm: !new:cosyvoice.llm.llm.TransformerLM
         num_blocks: 7
         dropout_rate: 0.1
         positional_dropout_rate: 0.1
-        attention_dropout_rate: 0
+        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
@@ -61,7 +66,7 @@ flow: !new:cosyvoice.flow.flow.MaskedDiffWithXvec
     spk_embed_dim: !ref <spk_embed_dim>
     output_type: 'mel'
     vocab_size: 4096
-    input_frame_rate: 50
+    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
@@ -97,7 +102,7 @@ flow: !new:cosyvoice.flow.flow.MaskedDiffWithXvec
             in_channels: 320
             out_channels: 80
             channels: [256, 256]
-            dropout: 0
+            dropout: 0.0
             attention_head_dim: 64
             n_blocks: 4
             num_mid_blocks: 8
@@ -128,9 +133,28 @@ hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
         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.MultiResolutionDiscriminator
+    mel_spec_transform: [
+        !ref <mel_spec_transform1>
+    ]
+
 # processor functions
 parquet_opener: !name:cosyvoice.dataset.processor.parquet_opener
-get_tokenizer: !name:whisper.tokenizer.get_tokenizer
+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'
@@ -146,6 +170,8 @@ filter: !name:cosyvoice.dataset.processor.filter
     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
@@ -157,6 +183,9 @@ feat_extractor: !name:matcha.utils.audio.mel_spectrogram
     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
@@ -182,8 +211,22 @@ data_pipeline: [
     !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>,
+]
 
-# train conf
+# llm flow train conf
 train_conf:
     optim: adam
     optim_conf:
@@ -195,4 +238,20 @@ train_conf:
     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/conf/cosyvoice.yaml

@@ -18,7 +18,7 @@ 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
+    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
@@ -31,7 +31,7 @@ llm: !new:cosyvoice.llm.llm.TransformerLM
         num_blocks: 6
         dropout_rate: 0.1
         positional_dropout_rate: 0.1
-        attention_dropout_rate: 0
+        attention_dropout_rate: 0.0
         normalize_before: True
         input_layer: 'linear'
         pos_enc_layer_type: 'rel_pos_espnet'
@@ -49,11 +49,16 @@ llm: !new:cosyvoice.llm.llm.TransformerLM
         num_blocks: 14
         dropout_rate: 0.1
         positional_dropout_rate: 0.1
-        attention_dropout_rate: 0
+        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
@@ -61,7 +66,7 @@ flow: !new:cosyvoice.flow.flow.MaskedDiffWithXvec
     spk_embed_dim: !ref <spk_embed_dim>
     output_type: 'mel'
     vocab_size: 4096
-    input_frame_rate: 50
+    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
@@ -97,7 +102,7 @@ flow: !new:cosyvoice.flow.flow.MaskedDiffWithXvec
             in_channels: 320
             out_channels: 80
             channels: [256, 256]
-            dropout: 0
+            dropout: 0.0
             attention_head_dim: 64
             n_blocks: 4
             num_mid_blocks: 12
@@ -128,9 +133,28 @@ hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
         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.MultiResolutionDiscriminator
+    mel_spec_transform: [
+        !ref <mel_spec_transform1>
+    ]
+
 # processor functions
 parquet_opener: !name:cosyvoice.dataset.processor.parquet_opener
-get_tokenizer: !name:whisper.tokenizer.get_tokenizer
+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'
@@ -146,6 +170,8 @@ filter: !name:cosyvoice.dataset.processor.filter
     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
@@ -157,6 +183,9 @@ feat_extractor: !name:matcha.utils.audio.mel_spectrogram
     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
@@ -165,7 +194,7 @@ 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
+    max_frames_in_batch: 2000 # change to 1400 in gan train on v100 16g
 padding: !name:cosyvoice.dataset.processor.padding
     use_spk_embedding: False # change to True during sft
 
@@ -182,8 +211,22 @@ data_pipeline: [
     !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>,
+]
 
-# train conf
+# llm flow train conf
 train_conf:
     optim: adam
     optim_conf:
@@ -195,4 +238,20 @@ train_conf:
     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/local/prepare_data.py

@@ -7,6 +7,7 @@ from tqdm import tqdm
 
 logger = logging.getLogger()
 
+
 def main():
     wavs = list(glob.glob('{}/*/*/*wav'.format(args.src_dir)))
 
@@ -41,6 +42,7 @@ def main():
             f.write('{} {}\n'.format(k, ' '.join(v)))
     return
 
+
 if __name__ == "__main__":
     parser = argparse.ArgumentParser()
     parser.add_argument('--src_dir',

examples/libritts/cosyvoice/run.sh

@@ -83,9 +83,9 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
   fi
   cat data/{train-clean-100,train-clean-360,train-other-500}/parquet/data.list > data/train.data.list
   cat data/{dev-clean,dev-other}/parquet/data.list > data/dev.data.list
-  for model in llm; do
+  for model in llm flow hifigan; do
     torchrun --nnodes=1 --nproc_per_node=$num_gpus \
-        --rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:0" \
+        --rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:1234" \
       cosyvoice/bin/train.py \
       --train_engine $train_engine \
       --config conf/cosyvoice.yaml \
@@ -99,7 +99,28 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
       --num_workers ${num_workers} \
       --prefetch ${prefetch} \
       --pin_memory \
+      --use_amp \
       --deepspeed_config ./conf/ds_stage2.json \
       --deepspeed.save_states model+optimizer
   done
+fi
+
+# average model
+average_num=5
+if [ ${stage} -le 6 ] && [ ${stop_stage} -ge 6 ]; then
+  for model in llm flow hifigan; do
+    decode_checkpoint=`pwd`/exp/cosyvoice/$model/$train_engine/${model}.pt
+    echo "do model average and final checkpoint is $decode_checkpoint"
+    python cosyvoice/bin/average_model.py \
+      --dst_model $decode_checkpoint \
+      --src_path `pwd`/exp/cosyvoice/$model/$train_engine  \
+      --num ${average_num} \
+      --val_best
+  done
+fi
+
+if [ ${stage} -le 7 ] && [ ${stop_stage} -ge 7 ]; then
+  echo "Export your model for inference speedup. Remember copy your llm or flow model to model_dir"
+  python cosyvoice/bin/export_jit.py --model_dir $pretrained_model_dir
+  python cosyvoice/bin/export_onnx.py --model_dir $pretrained_model_dir
 fi

examples/libritts/cosyvoice2/cosyvoice

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

examples/libritts/cosyvoice2/tools

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

examples/magicdata-read/cosyvoice/conf/cosyvoice.fromscratch.yaml

@@ -18,7 +18,7 @@ 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
+    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
@@ -54,6 +54,11 @@ llm: !new:cosyvoice.llm.llm.TransformerLM
         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
@@ -61,7 +66,7 @@ flow: !new:cosyvoice.flow.flow.MaskedDiffWithXvec
     spk_embed_dim: !ref <spk_embed_dim>
     output_type: 'mel'
     vocab_size: 4096
-    input_frame_rate: 50
+    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
@@ -130,7 +135,7 @@ hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
 
 # processor functions
 parquet_opener: !name:cosyvoice.dataset.processor.parquet_opener
-get_tokenizer: !name:whisper.tokenizer.get_tokenizer
+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'

examples/magicdata-read/cosyvoice/conf/cosyvoice.yaml

@@ -18,7 +18,7 @@ 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
+    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
@@ -54,6 +54,11 @@ llm: !new:cosyvoice.llm.llm.TransformerLM
         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
@@ -61,7 +66,7 @@ flow: !new:cosyvoice.flow.flow.MaskedDiffWithXvec
     spk_embed_dim: !ref <spk_embed_dim>
     output_type: 'mel'
     vocab_size: 4096
-    input_frame_rate: 50
+    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
@@ -130,7 +135,7 @@ hift: !new:cosyvoice.hifigan.generator.HiFTGenerator
 
 # processor functions
 parquet_opener: !name:cosyvoice.dataset.processor.parquet_opener
-get_tokenizer: !name:whisper.tokenizer.get_tokenizer
+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'

examples/magicdata-read/cosyvoice/local/prepare_data.py

@@ -6,6 +6,7 @@ from tqdm import tqdm
 
 logger = logging.getLogger()
 
+
 def main():
     utt2wav, utt2text, utt2spk, spk2utt = {}, {}, {}, {}
     with open(os.path.join(args.src_dir, "TRANS.txt"), "r") as f:
@@ -40,6 +41,7 @@ def main():
             f.write('{} {}\n'.format(k, ' '.join(v)))
     return
 
+
 if __name__ == "__main__":
     parser = argparse.ArgumentParser()
     parser.add_argument('--src_dir',

examples/magicdata-read/cosyvoice/run.sh

@@ -83,7 +83,7 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
   fi
   cp data/train/parquet/data.list data/train.data.list
   cp data/dev/parquet/data.list data/dev.data.list
-  for model in llm; do
+  for model in llm flow; do
     torchrun --nnodes=1 --nproc_per_node=$num_gpus \
         --rdzv_id=$job_id --rdzv_backend="c10d" --rdzv_endpoint="localhost:0" \
       cosyvoice/bin/train.py \
@@ -102,4 +102,10 @@ if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
       --deepspeed_config ./conf/ds_stage2.json \
       --deepspeed.save_states model+optimizer
   done
+fi
+
+if [ ${stage} -le 6 ] && [ ${stop_stage} -ge 6 ]; 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

requirements.txt

@@ -1,9 +1,10 @@
---extra-index-url https://download.pytorch.org/whl/cu118
+--extra-index-url https://download.pytorch.org/whl/cu121
+--extra-index-url https://aiinfra.pkgs.visualstudio.com/PublicPackages/_packaging/onnxruntime-cuda-12/pypi/simple/ # https://github.com/microsoft/onnxruntime/issues/21684
 conformer==0.3.2
 deepspeed==0.14.2; sys_platform == 'linux'
-diffusers==0.27.2
+diffusers==0.29.0
 gdown==5.1.0
-gradio==4.32.2
+gradio==5.4.0
 grpcio==1.57.0
 grpcio-tools==1.57.0
 hydra-core==1.3.2
@@ -15,17 +16,24 @@ matplotlib==3.7.5
 modelscope==1.15.0
 networkx==3.1
 omegaconf==2.3.0
-onnxruntime-gpu==1.16.0; sys_platform == 'linux'
+onnx==1.16.0
-onnxruntime==1.16.0; sys_platform == 'darwin' or sys_platform == 'windows'
+onnxruntime-gpu==1.18.0; sys_platform == 'linux'
+onnxruntime==1.18.0; sys_platform == 'darwin' or sys_platform == 'windows'
 openai-whisper==20231117
 protobuf==4.25
 pydantic==2.7.0
+pyworld==0.3.4
 rich==13.7.1
 soundfile==0.12.1
 tensorboard==2.14.0
-torch==2.0.1
+tensorrt-cu12==10.0.1; sys_platform == 'linux'
-torchaudio==2.0.2
+tensorrt-cu12-bindings==10.0.1; sys_platform == 'linux'
+tensorrt-cu12-libs==10.0.1; sys_platform == 'linux'
+torch==2.4.0
+torchaudio==2.4.0
+transformers==4.40.1
+uvicorn==0.30.0
 wget==3.2
-fastapi==0.111.0
+fastapi==0.115.6
 fastapi-cli==0.0.4
 WeTextProcessing==1.0.3

runtime/python/fastapi/client.py

@@ -1,56 +1,69 @@
+# 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.
 import argparse
 import logging
 import requests
+import torch
+import torchaudio
+import numpy as np
 
-def saveResponse(path, response):
-    # 以二进制写入模式打开文件
-    with open(path, 'wb') as file:
-        # 将响应的二进制内容写入文件
-        file.write(response.content)
 
 def main():
-    api = args.api_base
+    url = "http://{}:{}/inference_{}".format(args.host, args.port, args.mode)
     if args.mode == 'sft':
-        url = api + "/api/inference/sft"
-        payload={
-            'tts': args.tts_text,
-            'role': args.spk_id
-        }
-        response = requests.request("POST", url, data=payload)
-        saveResponse(args.tts_wav, response)
-    elif args.mode == 'zero_shot':
-        url = api + "/api/inference/zero-shot"
-        payload={
-            'tts': args.tts_text,
-            'prompt': args.prompt_text
-        }
-        files=[('audio', ('prompt_audio.wav', open(args.prompt_wav,'rb'), 'application/octet-stream'))]
-        response = requests.request("POST", url, data=payload, files=files)
-        saveResponse(args.tts_wav, response)
-    elif args.mode == 'cross_lingual':
-        url = api + "/api/inference/cross-lingual"
-        payload={
-            'tts': args.tts_text,
-        }
-        files=[('audio', ('prompt_audio.wav', open(args.prompt_wav,'rb'), 'application/octet-stream'))]
-        response = requests.request("POST", url, data=payload, files=files)
-        saveResponse(args.tts_wav, response)
-    else:
-        url = api + "/api/inference/instruct"
         payload = {
-            'tts': args.tts_text,
+            'tts_text': args.tts_text,
-            'role': args.spk_id,
+            'spk_id': args.spk_id
-            'instruct': args.instruct_text
         }
-        response = requests.request("POST", url, data=payload)
+        response = requests.request("GET", url, data=payload, stream=True)
-        saveResponse(args.tts_wav, response)
+    elif args.mode == 'zero_shot':
-    logging.info("Response save to {}", args.tts_wav)
+        payload = {
+            'tts_text': args.tts_text,
+            'prompt_text': args.prompt_text
+        }
+        files = [('prompt_wav', ('prompt_wav', open(args.prompt_wav, 'rb'), 'application/octet-stream'))]
+        response = requests.request("GET", url, data=payload, files=files, stream=True)
+    elif args.mode == 'cross_lingual':
+        payload = {
+            'tts_text': args.tts_text,
+        }
+        files = [('prompt_wav', ('prompt_wav', open(args.prompt_wav, 'rb'), 'application/octet-stream'))]
+        response = requests.request("GET", url, data=payload, files=files, stream=True)
+    else:
+        payload = {
+            'tts_text': args.tts_text,
+            'spk_id': args.spk_id,
+            'instruct_text': args.instruct_text
+        }
+        response = requests.request("GET", url, data=payload, stream=True)
+    tts_audio = b''
+    for r in response.iter_content(chunk_size=16000):
+        tts_audio += r
+    tts_speech = torch.from_numpy(np.array(np.frombuffer(tts_audio, dtype=np.int16))).unsqueeze(dim=0)
+    logging.info('save response to {}'.format(args.tts_wav))
+    torchaudio.save(args.tts_wav, tts_speech, target_sr)
+    logging.info('get response')
+
 
 if __name__ == "__main__":
     parser = argparse.ArgumentParser()
-    parser.add_argument('--api_base',
+    parser.add_argument('--host',
                         type=str,
-                        default='http://127.0.0.1:6006')
+                        default='0.0.0.0')
+    parser.add_argument('--port',
+                        type=int,
+                        default='50000')
     parser.add_argument('--mode',
                         default='sft',
                         choices=['sft', 'zero_shot', 'cross_lingual', 'instruct'],
@@ -66,10 +79,11 @@ if __name__ == "__main__":
                         default='希望你以后能够做的比我还好呦。')
     parser.add_argument('--prompt_wav',
                         type=str,
-                        default='../../../zero_shot_prompt.wav')
+                        default='../../../asset/zero_shot_prompt.wav')
     parser.add_argument('--instruct_text',
                         type=str,
-                        default='Theo \'Crimson\', is a fiery, passionate rebel leader. Fights with fervor for justice, but struggles with impulsiveness.')
+                        default='Theo \'Crimson\', is a fiery, passionate rebel leader. \
+                                 Fights with fervor for justice, but struggles with impulsiveness.')
     parser.add_argument('--tts_wav',
                         type=str,
                         default='demo.wav')

runtime/python/fastapi/server.py

@@ -1,119 +1,101 @@
-# Set inference model
+# Copyright (c) 2024 Alibaba Inc (authors: Xiang Lyu)
-# export MODEL_DIR=pretrained_models/CosyVoice-300M-Instruct
+#
-# For development
+# Licensed under the Apache License, Version 2.0 (the "License");
-# fastapi dev --port 6006 fastapi_server.py
+# you may not use this file except in compliance with the License.
-# For production deployment
+# You may obtain a copy of the License at
-# fastapi run --port 6006 fastapi_server.py
+#
-
+#   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 os
 import sys
-import io,time
+import argparse
-from fastapi import FastAPI, Response, File, UploadFile, Form
+import logging
-from fastapi.responses import HTMLResponse
+logging.getLogger('matplotlib').setLevel(logging.WARNING)
-from fastapi.middleware.cors import CORSMiddleware  #引入 CORS中间件模块
+from fastapi import FastAPI, UploadFile, Form, File
-from contextlib import asynccontextmanager
+from fastapi.responses import StreamingResponse
+from fastapi.middleware.cors import CORSMiddleware
+import uvicorn
+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
+from cosyvoice.cli.cosyvoice import CosyVoice, CosyVoice2
 from cosyvoice.utils.file_utils import load_wav
-import numpy as np
-import torch
-import torchaudio
-import logging
-logging.getLogger('matplotlib').setLevel(logging.WARNING)
 
-class LaunchFailed(Exception):
+app = FastAPI()
-    pass
+# set cross region allowance
-
-@asynccontextmanager
-async def lifespan(app: FastAPI):
-    model_dir = os.getenv("MODEL_DIR", "pretrained_models/CosyVoice-300M-SFT")
-    if model_dir:
-        logging.info("MODEL_DIR is {}", model_dir)
-        app.cosyvoice = CosyVoice(model_dir)
-        # sft usage
-        logging.info("Avaliable speakers {}", app.cosyvoice.list_avaliable_spks())
-    else:
-        raise LaunchFailed("MODEL_DIR environment must set")
-    yield
-
-app = FastAPI(lifespan=lifespan)
-
-#设置允许访问的域名
-origins = ["*"]  #"*"，即为所有,也可以改为允许的特定ip。
 app.add_middleware(
-    CORSMiddleware, 
+    CORSMiddleware,
-    allow_origins=origins,  #设置允许的origins来源
+    allow_origins=["*"],
     allow_credentials=True,
-    allow_methods=["*"],  # 设置允许跨域的http方法，比如 get、post、put等。
+    allow_methods=["*"],
-    allow_headers=["*"])  #允许跨域的headers，可以用来鉴别来源等作用。
+    allow_headers=["*"])
 
-def buildResponse(output):
-    buffer = io.BytesIO()
-    torchaudio.save(buffer, output, 22050, format="wav")
-    buffer.seek(0)
-    return Response(content=buffer.read(-1), media_type="audio/wav")
 
-@app.post("/api/inference/sft")
+def generate_data(model_output):
-@app.get("/api/inference/sft")
+    for i in model_output:
-async def sft(tts: str = Form(), role: str = Form()):
+        tts_audio = (i['tts_speech'].numpy() * (2 ** 15)).astype(np.int16).tobytes()
-    start = time.process_time()
+        yield tts_audio
-    output = app.cosyvoice.inference_sft(tts, role)
-    end = time.process_time()
-    logging.info("infer time is {} seconds", end-start)
-    return buildResponse(output['tts_speech'])
 
-@app.post("/api/inference/zero-shot")
-async def zeroShot(tts: str = Form(), prompt: str = Form(), audio: UploadFile = File()):
-    start = time.process_time()
-    prompt_speech = load_wav(audio.file, 16000)
-    prompt_audio = (prompt_speech.numpy() * (2**15)).astype(np.int16).tobytes()
-    prompt_speech_16k = torch.from_numpy(np.array(np.frombuffer(prompt_audio, dtype=np.int16))).unsqueeze(dim=0)
-    prompt_speech_16k = prompt_speech_16k.float() / (2**15)
 
-    output = app.cosyvoice.inference_zero_shot(tts, prompt, prompt_speech_16k)
+@app.get("/inference_sft")
-    end = time.process_time()
+@app.post("/inference_sft")
-    logging.info("infer time is {} seconds", end-start)
+async def inference_sft(tts_text: str = Form(), spk_id: str = Form()):
-    return buildResponse(output['tts_speech'])
+    model_output = cosyvoice.inference_sft(tts_text, spk_id)
+    return StreamingResponse(generate_data(model_output))
 
-@app.post("/api/inference/cross-lingual")
-async def crossLingual(tts: str = Form(), audio: UploadFile = File()):
-    start = time.process_time()
-    prompt_speech = load_wav(audio.file, 16000)
-    prompt_audio = (prompt_speech.numpy() * (2**15)).astype(np.int16).tobytes()
-    prompt_speech_16k = torch.from_numpy(np.array(np.frombuffer(prompt_audio, dtype=np.int16))).unsqueeze(dim=0)
-    prompt_speech_16k = prompt_speech_16k.float() / (2**15)
 
-    output = app.cosyvoice.inference_cross_lingual(tts, prompt_speech_16k)
+@app.get("/inference_zero_shot")
-    end = time.process_time()
+@app.post("/inference_zero_shot")
-    logging.info("infer time is {} seconds", end-start)
+async def inference_zero_shot(tts_text: str = Form(), prompt_text: str = Form(), prompt_wav: UploadFile = File()):
-    return buildResponse(output['tts_speech'])
+    prompt_speech_16k = load_wav(prompt_wav.file, 16000)
+    model_output = cosyvoice.inference_zero_shot(tts_text, prompt_text, prompt_speech_16k)
+    return StreamingResponse(generate_data(model_output))
 
-@app.post("/api/inference/instruct")
-@app.get("/api/inference/instruct")
-async def instruct(tts: str = Form(), role: str = Form(), instruct: str = Form()):
-    start = time.process_time()
-    output = app.cosyvoice.inference_instruct(tts, role, instruct)
-    end = time.process_time()
-    logging.info("infer time is {} seconds", end-start)
-    return buildResponse(output['tts_speech'])
 
-@app.get("/api/roles")
+@app.get("/inference_cross_lingual")
-async def roles():
+@app.post("/inference_cross_lingual")
-    return {"roles": app.cosyvoice.list_avaliable_spks()}
+async def inference_cross_lingual(tts_text: str = Form(), prompt_wav: UploadFile = File()):
+    prompt_speech_16k = load_wav(prompt_wav.file, 16000)
+    model_output = cosyvoice.inference_cross_lingual(tts_text, prompt_speech_16k)
+    return StreamingResponse(generate_data(model_output))
 
-@app.get("/", response_class=HTMLResponse)
+
-async def root():
+@app.get("/inference_instruct")
-    return """
+@app.post("/inference_instruct")
-    <!DOCTYPE html>
+async def inference_instruct(tts_text: str = Form(), spk_id: str = Form(), instruct_text: str = Form()):
-    <html lang=zh-cn>
+    model_output = cosyvoice.inference_instruct(tts_text, spk_id, instruct_text)
-        <head>
+    return StreamingResponse(generate_data(model_output))
-            <meta charset=utf-8>
+
-            <title>Api information</title>
+@app.get("/inference_instruct2")
-        </head>
+@app.post("/inference_instruct2")
-        <body>
+async def inference_instruct2(tts_text: str = Form(), instruct_text: str = Form(), prompt_wav: UploadFile = File()):
-            Get the supported tones from the Roles API first, then enter the tones and textual content in the TTS API for synthesis. <a href='./docs'>Documents of API</a>
+    prompt_speech_16k = load_wav(prompt_wav.file, 16000)
-        </body>
+    model_output = cosyvoice.inference_instruct2(tts_text, instruct_text, prompt_speech_16k)
-    </html>
+    return StreamingResponse(generate_data(model_output))
-    """
+
+
+
+if __name__ == '__main__':
+    parser = argparse.ArgumentParser()
+    parser.add_argument('--port',
+                        type=int,
+                        default=50000)
+    parser.add_argument('--model_dir',
+                        type=str,
+                        default='iic/CosyVoice-300M',
+                        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!')
+    uvicorn.run(app, host="0.0.0.0", port=args.port)

runtime/python/grpc/client.py

@@ -61,8 +61,11 @@ def main():
             request.instruct_request.CopyFrom(instruct_request)
 
         response = stub.Inference(request)
+        tts_audio = b''
+        for r in response:
+            tts_audio += r.tts_audio
+        tts_speech = torch.from_numpy(np.array(np.frombuffer(tts_audio, dtype=np.int16))).unsqueeze(dim=0)
         logging.info('save response to {}'.format(args.tts_wav))
-        tts_speech = torch.from_numpy(np.array(np.frombuffer(response.tts_audio, dtype=np.int16))).unsqueeze(dim=0)
         torchaudio.save(args.tts_wav, tts_speech, target_sr)
         logging.info('get response')
 
@@ -90,10 +93,11 @@ if __name__ == "__main__":
                         default='希望你以后能够做的比我还好呦。')
     parser.add_argument('--prompt_wav',
                         type=str,
-                        default='../../../zero_shot_prompt.wav')
+                        default='../../../asset/zero_shot_prompt.wav')
     parser.add_argument('--instruct_text',
                         type=str,
-                        default='Theo \'Crimson\', is a fiery, passionate rebel leader. Fights with fervor for justice, but struggles with impulsiveness.')
+                        default='Theo \'Crimson\', is a fiery, passionate rebel leader. \
+                                 Fights with fervor for justice, but struggles with impulsiveness.')
     parser.add_argument('--tts_wav',
                         type=str,
                         default='demo.wav')

runtime/python/grpc/cosyvoice.proto

@@ -4,7 +4,7 @@ package cosyvoice;
 option go_package = "protos/";
 
 service CosyVoice{
-  rpc Inference(Request) returns (Response) {}
+  rpc Inference(Request) returns (stream Response) {}
 }
 
 message Request{

runtime/python/grpc/server.py

@@ -13,9 +13,6 @@
 # limitations under the License.
 import os
 import sys
-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 concurrent import futures
 import argparse
 import cosyvoice_pb2
@@ -25,14 +22,24 @@ logging.getLogger('matplotlib').setLevel(logging.WARNING)
 import grpc
 import torch
 import numpy as np
-from cosyvoice.cli.cosyvoice import CosyVoice
+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
 
 logging.basicConfig(level=logging.DEBUG,
                     format='%(asctime)s %(levelname)s %(message)s')
 
+
 class CosyVoiceServiceImpl(cosyvoice_pb2_grpc.CosyVoiceServicer):
     def __init__(self, args):
-        self.cosyvoice = CosyVoice(args.model_dir)
+        try:
+            self.cosyvoice = CosyVoice(args.model_dir)
+        except Exception:
+            try:
+                self.cosyvoice = CosyVoice2(args.model_dir)
+            except Exception:
+                raise TypeError('no valid model_type!')
         logging.info('grpc service initialized')
 
     def Inference(self, request, context):
@@ -43,7 +50,9 @@ class CosyVoiceServiceImpl(cosyvoice_pb2_grpc.CosyVoiceServicer):
             logging.info('get zero_shot inference request')
             prompt_speech_16k = torch.from_numpy(np.array(np.frombuffer(request.zero_shot_request.prompt_audio, dtype=np.int16))).unsqueeze(dim=0)
             prompt_speech_16k = prompt_speech_16k.float() / (2**15)
-            model_output = self.cosyvoice.inference_zero_shot(request.zero_shot_request.tts_text, request.zero_shot_request.prompt_text, prompt_speech_16k)
+            model_output = self.cosyvoice.inference_zero_shot(request.zero_shot_request.tts_text,
+                                                              request.zero_shot_request.prompt_text,
+                                                              prompt_speech_16k)
         elif request.HasField('cross_lingual_request'):
             logging.info('get cross_lingual inference request')
             prompt_speech_16k = torch.from_numpy(np.array(np.frombuffer(request.cross_lingual_request.prompt_audio, dtype=np.int16))).unsqueeze(dim=0)
@@ -51,12 +60,16 @@ class CosyVoiceServiceImpl(cosyvoice_pb2_grpc.CosyVoiceServicer):
             model_output = self.cosyvoice.inference_cross_lingual(request.cross_lingual_request.tts_text, prompt_speech_16k)
         else:
             logging.info('get instruct inference request')
-            model_output = self.cosyvoice.inference_instruct(request.instruct_request.tts_text, request.instruct_request.spk_id, request.instruct_request.instruct_text)
+            model_output = self.cosyvoice.inference_instruct(request.instruct_request.tts_text,
+                                                             request.instruct_request.spk_id,
+                                                             request.instruct_request.instruct_text)
 
         logging.info('send inference response')
-        response = cosyvoice_pb2.Response()
+        for i in model_output:
-        response.tts_audio = (model_output['tts_speech'].numpy() * (2 ** 15)).astype(np.int16).tobytes()
+            response = cosyvoice_pb2.Response()
-        return response
+            response.tts_audio = (i['tts_speech'].numpy() * (2 ** 15)).astype(np.int16).tobytes()
+            yield response
+
 
 def main():
     grpcServer = grpc.server(futures.ThreadPoolExecutor(max_workers=args.max_conc), maximum_concurrent_rpcs=args.max_conc)

tools/extract_embedding.py

@@ -13,14 +13,50 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 import argparse
+from concurrent.futures import ThreadPoolExecutor, as_completed
+import onnxruntime
 import torch
 import torchaudio
-from tqdm import tqdm
-import onnxruntime
 import torchaudio.compliance.kaldi as kaldi
+from tqdm import tqdm
+
+
+def single_job(utt):
+    audio, sample_rate = torchaudio.load(utt2wav[utt])
+    if sample_rate != 16000:
+        audio = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(audio)
+    feat = kaldi.fbank(audio,
+                       num_mel_bins=80,
+                       dither=0,
+                       sample_frequency=16000)
+    feat = feat - feat.mean(dim=0, keepdim=True)
+    embedding = ort_session.run(None, {ort_session.get_inputs()[0].name: feat.unsqueeze(dim=0).cpu().numpy()})[0].flatten().tolist()
+    return utt, embedding
 
 
 def main(args):
+    all_task = [executor.submit(single_job, utt) for utt in utt2wav.keys()]
+    utt2embedding, spk2embedding = {}, {}
+    for future in tqdm(as_completed(all_task)):
+        utt, embedding = future.result()
+        utt2embedding[utt] = embedding
+        spk = utt2spk[utt]
+        if spk not in spk2embedding:
+            spk2embedding[spk] = []
+        spk2embedding[spk].append(embedding)
+    for k, v in spk2embedding.items():
+        spk2embedding[k] = torch.tensor(v).mean(dim=0).tolist()
+    torch.save(utt2embedding, "{}/utt2embedding.pt".format(args.dir))
+    torch.save(spk2embedding, "{}/spk2embedding.pt".format(args.dir))
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--dir", type=str)
+    parser.add_argument("--onnx_path", type=str)
+    parser.add_argument("--num_thread", type=int, default=8)
+    args = parser.parse_args()
+
     utt2wav, utt2spk = {}, {}
     with open('{}/wav.scp'.format(args.dir)) as f:
         for l in f:
@@ -36,34 +72,6 @@ def main(args):
     option.intra_op_num_threads = 1
     providers = ["CPUExecutionProvider"]
     ort_session = onnxruntime.InferenceSession(args.onnx_path, sess_options=option, providers=providers)
+    executor = ThreadPoolExecutor(max_workers=args.num_thread)
 
-    utt2embedding, spk2embedding = {}, {}
-    for utt in tqdm(utt2wav.keys()):
-        audio, sample_rate = torchaudio.load(utt2wav[utt])
-        if sample_rate != 16000:
-            audio = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(audio)
-        feat = kaldi.fbank(audio,
-                           num_mel_bins=80,
-                           dither=0,
-                           sample_frequency=16000)
-        feat = feat - feat.mean(dim=0, keepdim=True)
-        embedding = ort_session.run(None, {ort_session.get_inputs()[0].name: feat.unsqueeze(dim=0).cpu().numpy()})[0].flatten().tolist()
-        utt2embedding[utt] = embedding
-        spk = utt2spk[utt]
-        if spk not in spk2embedding:
-            spk2embedding[spk] = []
-        spk2embedding[spk].append(embedding)
-    for k, v in spk2embedding.items():
-        spk2embedding[k] = torch.tensor(v).mean(dim=0).tolist()
-
-    torch.save(utt2embedding, '{}/utt2embedding.pt'.format(args.dir))
-    torch.save(spk2embedding, '{}/spk2embedding.pt'.format(args.dir))
-
-if __name__ == "__main__":
-    parser = argparse.ArgumentParser()
-    parser.add_argument('--dir',
-                        type=str)
-    parser.add_argument('--onnx_path',
-                        type=str)
-    args = parser.parse_args()
     main(args)

tools/extract_speech_token.py

@@ -13,6 +13,7 @@
 # See the License for the specific language governing permissions and
 # limitations under the License.
 import argparse
+from concurrent.futures import ThreadPoolExecutor, as_completed
 import logging
 import torch
 from tqdm import tqdm
@@ -22,7 +23,39 @@ import torchaudio
 import whisper
 
 
+def single_job(utt):
+    audio, sample_rate = torchaudio.load(utt2wav[utt], backend='soundfile')
+    if sample_rate != 16000:
+        audio = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(audio)
+    # Convert audio to mono
+    if audio.shape[0] > 1:
+        audio = audio.mean(dim=0, keepdim=True)
+    if audio.shape[1] / 16000 > 30:
+        logging.warning('do not support extract speech token for audio longer than 30s')
+        speech_token = []
+    else:
+        feat = whisper.log_mel_spectrogram(audio, n_mels=128)
+        speech_token = ort_session.run(None, {ort_session.get_inputs()[0].name: feat.detach().cpu().numpy(),
+                                              ort_session.get_inputs()[1].name: np.array([feat.shape[2]], dtype=np.int32)})[0].flatten().tolist()
+    return utt, speech_token
+
+
 def main(args):
+    all_task = [executor.submit(single_job, utt) for utt in utt2wav.keys()]
+    utt2speech_token = {}
+    for future in tqdm(as_completed(all_task)):
+        utt, speech_token = future.result()
+        utt2speech_token[utt] = speech_token
+    torch.save(utt2speech_token, '{}/utt2speech_token.pt'.format(args.dir))
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--dir", type=str)
+    parser.add_argument("--onnx_path", type=str)
+    parser.add_argument("--num_thread", type=int, default=8)
+    args = parser.parse_args()
+
     utt2wav = {}
     with open('{}/wav.scp'.format(args.dir)) as f:
         for l in f:
@@ -34,28 +67,6 @@ def main(args):
     option.intra_op_num_threads = 1
     providers = ["CUDAExecutionProvider"]
     ort_session = onnxruntime.InferenceSession(args.onnx_path, sess_options=option, providers=providers)
+    executor = ThreadPoolExecutor(max_workers=args.num_thread)
 
-    utt2speech_token = {}
-    for utt in tqdm(utt2wav.keys()):
-        audio, sample_rate = torchaudio.load(utt2wav[utt])
-        if sample_rate != 16000:
-            audio = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(audio)
-        if audio.shape[1] / 16000 > 30:
-            logging.warning('do not support extract speech token for audio longer than 30s')
-            speech_token = []
-        else:
-            feat = whisper.log_mel_spectrogram(audio, n_mels=128)
-            speech_token = ort_session.run(None, {ort_session.get_inputs()[0].name: feat.detach().cpu().numpy(),
-                                                  ort_session.get_inputs()[1].name: np.array([feat.shape[2]], dtype=np.int32)})[0].flatten().tolist()
-        utt2speech_token[utt] = speech_token
-    torch.save(utt2speech_token, '{}/utt2speech_token.pt'.format(args.dir))
-
-
-if __name__ == "__main__":
-    parser = argparse.ArgumentParser()
-    parser.add_argument('--dir',
-                        type=str)
-    parser.add_argument('--onnx_path',
-                        type=str)
-    args = parser.parse_args()
     main(args)

tools/make_parquet_list.py

@@ -53,6 +53,7 @@ def job(utt_list, parquet_file, utt2parquet_file, spk2parquet_file):
         json.dump({k: parquet_file for k in list(set(spk_list))}, f, ensure_ascii=False, indent=2)
     logging.info('spend time {}'.format(time.time() - start_time))
 
+
 if __name__ == "__main__":
     parser = argparse.ArgumentParser()
     parser.add_argument('--num_utts_per_parquet',

webui.py

@@ -13,9 +13,6 @@
 # limitations under the License.
 import os
 import sys
-ROOT_DIR = os.path.dirname(os.path.abspath(__file__))
-sys.path.append('{}/third_party/Matcha-TTS'.format(ROOT_DIR))
-
 import argparse
 import gradio as gr
 import numpy as np
@@ -23,15 +20,20 @@ import torch
 import torchaudio
 import random
 import librosa
+ROOT_DIR = os.path.dirname(os.path.abspath(__file__))
+sys.path.append('{}/third_party/Matcha-TTS'.format(ROOT_DIR))
+from cosyvoice.cli.cosyvoice import CosyVoice, CosyVoice2
+from cosyvoice.utils.file_utils import load_wav, logging
+from cosyvoice.utils.common import set_all_random_seed
 
-import logging
+inference_mode_list = ['预训练音色', '3s极速复刻', '跨语种复刻', '自然语言控制']
-logging.getLogger('matplotlib').setLevel(logging.WARNING)
+instruct_dict = {'预训练音色': '1. 选择预训练音色\n2. 点击生成音频按钮',
+                 '3s极速复刻': '1. 选择prompt音频文件，或录入prompt音频，注意不超过30s，若同时提供，优先选择prompt音频文件\n2. 输入prompt文本\n3. 点击生成音频按钮',
+                 '跨语种复刻': '1. 选择prompt音频文件，或录入prompt音频，注意不超过30s，若同时提供，优先选择prompt音频文件\n2. 点击生成音频按钮',
+                 '自然语言控制': '1. 选择预训练音色\n2. 输入instruct文本\n3. 点击生成音频按钮'}
+stream_mode_list = [('否', False), ('是', True)]
+max_val = 0.8
 
-from cosyvoice.cli.cosyvoice import CosyVoice
-from cosyvoice.utils.file_utils import load_wav, speed_change
-
-logging.basicConfig(level=logging.DEBUG,
-                    format='%(asctime)s %(levelname)s %(message)s')
 
 def generate_seed():
     seed = random.randint(1, 100000000)
@@ -40,13 +42,7 @@ def generate_seed():
         "value": seed
     }
 
-def set_all_random_seed(seed):
-    random.seed(seed)
-    np.random.seed(seed)
-    torch.manual_seed(seed)
-    torch.cuda.manual_seed_all(seed)
 
-max_val = 0.8
 def postprocess(speech, top_db=60, hop_length=220, win_length=440):
     speech, _ = librosa.effects.trim(
         speech, top_db=top_db,
@@ -55,18 +51,16 @@ def postprocess(speech, top_db=60, hop_length=220, win_length=440):
     )
     if speech.abs().max() > max_val:
         speech = speech / speech.abs().max() * max_val
-    speech = torch.concat([speech, torch.zeros(1, int(target_sr * 0.2))], dim=1)
+    speech = torch.concat([speech, torch.zeros(1, int(cosyvoice.sample_rate * 0.2))], dim=1)
     return speech
 
-inference_mode_list = ['预训练音色', '3s极速复刻', '跨语种复刻', '自然语言控制']
+
-instruct_dict = {'预训练音色': '1. 选择预训练音色\n2. 点击生成音频按钮',
-                 '3s极速复刻': '1. 选择prompt音频文件，或录入prompt音频，注意不超过30s，若同时提供，优先选择prompt音频文件\n2. 输入prompt文本\n3. 点击生成音频按钮',
-                 '跨语种复刻': '1. 选择prompt音频文件，或录入prompt音频，注意不超过30s，若同时提供，优先选择prompt音频文件\n2. 点击生成音频按钮',
-                 '自然语言控制': '1. 选择预训练音色\n2. 输入instruct文本\n3. 点击生成音频按钮'}
 def change_instruction(mode_checkbox_group):
     return instruct_dict[mode_checkbox_group]
 
-def generate_audio(tts_text, mode_checkbox_group, sft_dropdown, prompt_text, prompt_wav_upload, prompt_wav_record, instruct_text, seed, speed_factor):
+
+def generate_audio(tts_text, mode_checkbox_group, sft_dropdown, prompt_text, prompt_wav_upload, prompt_wav_record, instruct_text,
+                   seed, stream, speed):
     if prompt_wav_upload is not None:
         prompt_wav = prompt_wav_upload
     elif prompt_wav_record is not None:
@@ -75,86 +69,87 @@ def generate_audio(tts_text, mode_checkbox_group, sft_dropdown, prompt_text, pro
         prompt_wav = None
     # if instruct mode, please make sure that model is iic/CosyVoice-300M-Instruct and not cross_lingual mode
     if mode_checkbox_group in ['自然语言控制']:
-        if cosyvoice.frontend.instruct is False:
+        if cosyvoice.instruct is False:
             gr.Warning('您正在使用自然语言控制模式, {}模型不支持此模式, 请使用iic/CosyVoice-300M-Instruct模型'.format(args.model_dir))
-            return (target_sr, default_data)
+            yield (cosyvoice.sample_rate, default_data)
         if instruct_text == '':
             gr.Warning('您正在使用自然语言控制模式, 请输入instruct文本')
-            return (target_sr, default_data)
+            yield (cosyvoice.sample_rate, default_data)
         if prompt_wav is not None or prompt_text != '':
             gr.Info('您正在使用自然语言控制模式, prompt音频/prompt文本会被忽略')
     # if cross_lingual mode, please make sure that model is iic/CosyVoice-300M and tts_text prompt_text are different language
     if mode_checkbox_group in ['跨语种复刻']:
-        if cosyvoice.frontend.instruct is True:
+        if cosyvoice.instruct is True:
             gr.Warning('您正在使用跨语种复刻模式, {}模型不支持此模式, 请使用iic/CosyVoice-300M模型'.format(args.model_dir))
-            return (target_sr, default_data)
+            yield (cosyvoice.sample_rate, default_data)
         if instruct_text != '':
             gr.Info('您正在使用跨语种复刻模式, instruct文本会被忽略')
         if prompt_wav is None:
             gr.Warning('您正在使用跨语种复刻模式, 请提供prompt音频')
-            return (target_sr, default_data)
+            yield (cosyvoice.sample_rate, default_data)
         gr.Info('您正在使用跨语种复刻模式, 请确保合成文本和prompt文本为不同语言')
     # if in zero_shot cross_lingual, please make sure that prompt_text and prompt_wav meets requirements
     if mode_checkbox_group in ['3s极速复刻', '跨语种复刻']:
         if prompt_wav is None:
             gr.Warning('prompt音频为空，您是否忘记输入prompt音频？')
-            return (target_sr, default_data)
+            yield (cosyvoice.sample_rate, default_data)
         if torchaudio.info(prompt_wav).sample_rate < prompt_sr:
             gr.Warning('prompt音频采样率{}低于{}'.format(torchaudio.info(prompt_wav).sample_rate, prompt_sr))
-            return (target_sr, default_data)
+            yield (cosyvoice.sample_rate, default_data)
     # sft mode only use sft_dropdown
     if mode_checkbox_group in ['预训练音色']:
         if instruct_text != '' or prompt_wav is not None or prompt_text != '':
             gr.Info('您正在使用预训练音色模式，prompt文本/prompt音频/instruct文本会被忽略！')
+        if sft_dropdown == '':
+            gr.Warning('没有可用的预训练音色！')
+            yield (cosyvoice.sample_rate, default_data)
     # zero_shot mode only use prompt_wav prompt text
     if mode_checkbox_group in ['3s极速复刻']:
         if prompt_text == '':
             gr.Warning('prompt文本为空，您是否忘记输入prompt文本？')
-            return (target_sr, default_data)
+            yield (cosyvoice.sample_rate, default_data)
         if instruct_text != '':
             gr.Info('您正在使用3s极速复刻模式，预训练音色/instruct文本会被忽略！')
 
     if mode_checkbox_group == '预训练音色':
         logging.info('get sft inference request')
         set_all_random_seed(seed)
-        output = cosyvoice.inference_sft(tts_text, sft_dropdown)
+        for i in cosyvoice.inference_sft(tts_text, sft_dropdown, stream=stream, speed=speed):
+            yield (cosyvoice.sample_rate, i['tts_speech'].numpy().flatten())
     elif mode_checkbox_group == '3s极速复刻':
         logging.info('get zero_shot inference request')
         prompt_speech_16k = postprocess(load_wav(prompt_wav, prompt_sr))
         set_all_random_seed(seed)
-        output = cosyvoice.inference_zero_shot(tts_text, prompt_text, prompt_speech_16k)
+        for i in cosyvoice.inference_zero_shot(tts_text, prompt_text, prompt_speech_16k, stream=stream, speed=speed):
+            yield (cosyvoice.sample_rate, i['tts_speech'].numpy().flatten())
     elif mode_checkbox_group == '跨语种复刻':
         logging.info('get cross_lingual inference request')
         prompt_speech_16k = postprocess(load_wav(prompt_wav, prompt_sr))
         set_all_random_seed(seed)
-        output = cosyvoice.inference_cross_lingual(tts_text, prompt_speech_16k)
+        for i in cosyvoice.inference_cross_lingual(tts_text, prompt_speech_16k, stream=stream, speed=speed):
+            yield (cosyvoice.sample_rate, i['tts_speech'].numpy().flatten())
     else:
         logging.info('get instruct inference request')
         set_all_random_seed(seed)
-        output = cosyvoice.inference_instruct(tts_text, sft_dropdown, instruct_text)
+        for i in cosyvoice.inference_instruct(tts_text, sft_dropdown, instruct_text, stream=stream, speed=speed):
+            yield (cosyvoice.sample_rate, i['tts_speech'].numpy().flatten())
 
-    if speed_factor != 1.0:
-        try:
-            audio_data, sample_rate = speed_change(output["tts_speech"], target_sr, str(speed_factor))
-            audio_data = audio_data.numpy().flatten()
-        except Exception as e:
-            print(f"Failed to change speed of audio: \n{e}")
-    else:
-        audio_data = output['tts_speech'].numpy().flatten()
-
-    return (target_sr, audio_data)
 
 def main():
     with gr.Blocks() as demo:
-        gr.Markdown("### 代码库 [CosyVoice](https://github.com/FunAudioLLM/CosyVoice) 预训练模型 [CosyVoice-300M](https://www.modelscope.cn/models/iic/CosyVoice-300M) [CosyVoice-300M-Instruct](https://www.modelscope.cn/models/iic/CosyVoice-300M-Instruct) [CosyVoice-300M-SFT](https://www.modelscope.cn/models/iic/CosyVoice-300M-SFT)")
+        gr.Markdown("### 代码库 [CosyVoice](https://github.com/FunAudioLLM/CosyVoice) \
+                    预训练模型 [CosyVoice-300M](https://www.modelscope.cn/models/iic/CosyVoice-300M) \
+                    [CosyVoice-300M-Instruct](https://www.modelscope.cn/models/iic/CosyVoice-300M-Instruct) \
+                    [CosyVoice-300M-SFT](https://www.modelscope.cn/models/iic/CosyVoice-300M-SFT)")
         gr.Markdown("#### 请输入需要合成的文本，选择推理模式，并按照提示步骤进行操作")
 
         tts_text = gr.Textbox(label="输入合成文本", lines=1, value="我是通义实验室语音团队全新推出的生成式语音大模型，提供舒适自然的语音合成能力。")
-        speed_factor = gr.Slider(minimum=0.25, maximum=4, step=0.05, label="语速调节", value=1.0, interactive=True)
         with gr.Row():
             mode_checkbox_group = gr.Radio(choices=inference_mode_list, label='选择推理模式', value=inference_mode_list[0])
             instruction_text = gr.Text(label="操作步骤", value=instruct_dict[inference_mode_list[0]], scale=0.5)
             sft_dropdown = gr.Dropdown(choices=sft_spk, label='选择预训练音色', value=sft_spk[0], scale=0.25)
+            stream = gr.Radio(choices=stream_mode_list, label='是否流式推理', value=stream_mode_list[0][1])
+            speed = gr.Number(value=1, label="速度调节(仅支持非流式推理)", minimum=0.5, maximum=2.0, step=0.1)
             with gr.Column(scale=0.25):
                 seed_button = gr.Button(value="\U0001F3B2")
                 seed = gr.Number(value=0, label="随机推理种子")
@@ -167,16 +162,18 @@ def main():
 
         generate_button = gr.Button("生成音频")
 
-        audio_output = gr.Audio(label="合成音频")
+        audio_output = gr.Audio(label="合成音频", autoplay=True, streaming=True)
 
         seed_button.click(generate_seed, inputs=[], outputs=seed)
         generate_button.click(generate_audio,
-                              inputs=[tts_text, mode_checkbox_group, sft_dropdown, prompt_text, prompt_wav_upload, prompt_wav_record, instruct_text, seed, speed_factor],
+                              inputs=[tts_text, mode_checkbox_group, sft_dropdown, prompt_text, prompt_wav_upload, prompt_wav_record, instruct_text,
+                                      seed, stream, speed],
                               outputs=[audio_output])
         mode_checkbox_group.change(fn=change_instruction, inputs=[mode_checkbox_group], outputs=[instruction_text])
     demo.queue(max_size=4, default_concurrency_limit=2)
     demo.launch(server_name='0.0.0.0', server_port=args.port)
 
+
 if __name__ == '__main__':
     parser = argparse.ArgumentParser()
     parser.add_argument('--port',
@@ -184,11 +181,20 @@ if __name__ == '__main__':
                         default=8000)
     parser.add_argument('--model_dir',
                         type=str,
-                        default='iic/CosyVoice-300M',
+                        default='pretrained_models/CosyVoice2-0.5B',
                         help='local path or modelscope repo id')
     args = parser.parse_args()
-    cosyvoice = CosyVoice(args.model_dir)
+    try:
-    sft_spk = cosyvoice.list_avaliable_spks()
+        cosyvoice = CosyVoice(args.model_dir)
-    prompt_sr, target_sr = 16000, 22050
+    except Exception:
-    default_data = np.zeros(target_sr)
+        try:
+            cosyvoice = CosyVoice2(args.model_dir)
+        except Exception:
+            raise TypeError('no valid model_type!')
+
+    sft_spk = cosyvoice.list_available_spks()
+    if len(sft_spk) == 0:
+        sft_spk = ['']
+    prompt_sr = 16000
+    default_data = np.zeros(cosyvoice.sample_rate)
     main()