Downgrade Gradio verison

.gitattributes

@@ -1,35 +0,0 @@
-*.7z filter=lfs diff=lfs merge=lfs -text
-*.arrow filter=lfs diff=lfs merge=lfs -text
-*.bin filter=lfs diff=lfs merge=lfs -text
-*.bz2 filter=lfs diff=lfs merge=lfs -text
-*.ckpt filter=lfs diff=lfs merge=lfs -text
-*.ftz filter=lfs diff=lfs merge=lfs -text
-*.gz filter=lfs diff=lfs merge=lfs -text
-*.h5 filter=lfs diff=lfs merge=lfs -text
-*.joblib filter=lfs diff=lfs merge=lfs -text
-*.lfs.* filter=lfs diff=lfs merge=lfs -text
-*.mlmodel filter=lfs diff=lfs merge=lfs -text
-*.model filter=lfs diff=lfs merge=lfs -text
-*.msgpack filter=lfs diff=lfs merge=lfs -text
-*.npy filter=lfs diff=lfs merge=lfs -text
-*.npz filter=lfs diff=lfs merge=lfs -text
-*.onnx filter=lfs diff=lfs merge=lfs -text
-*.ot filter=lfs diff=lfs merge=lfs -text
-*.parquet filter=lfs diff=lfs merge=lfs -text
-*.pb filter=lfs diff=lfs merge=lfs -text
-*.pickle filter=lfs diff=lfs merge=lfs -text
-*.pkl filter=lfs diff=lfs merge=lfs -text
-*.pt filter=lfs diff=lfs merge=lfs -text
-*.pth filter=lfs diff=lfs merge=lfs -text
-*.rar filter=lfs diff=lfs merge=lfs -text
-*.safetensors filter=lfs diff=lfs merge=lfs -text
-saved_model/**/* filter=lfs diff=lfs merge=lfs -text
-*.tar.* filter=lfs diff=lfs merge=lfs -text
-*.tar filter=lfs diff=lfs merge=lfs -text
-*.tflite filter=lfs diff=lfs merge=lfs -text
-*.tgz filter=lfs diff=lfs merge=lfs -text
-*.wasm filter=lfs diff=lfs merge=lfs -text
-*.xz filter=lfs diff=lfs merge=lfs -text
-*.zip filter=lfs diff=lfs merge=lfs -text
-*.zst filter=lfs diff=lfs merge=lfs -text
-*tfevents* filter=lfs diff=lfs merge=lfs -text

.github/ISSUE_TEMPLATE/bug_report.yml

@@ -1,50 +0,0 @@
-name: "Bug Report"
-description: |
-  Please provide as much details to help address the issue, including logs and screenshots.
-labels:
-  - bug
-body:
-  - type: checkboxes
-    attributes:
-      label: Checks
-      description: "To ensure timely help, please confirm the following:"
-      options:
-        - label: This template is only for bug reports, usage problems go with 'Help Wanted'.
-          required: true
-        - label: I have thoroughly reviewed the project documentation but couldn't find information to solve my problem.
-          required: true
-        - label: I have searched for existing issues, including closed ones, and couldn't find a solution.
-          required: true
-        - label: I confirm that I am using English to submit this report in order to facilitate communication.
-          required: true
-  - type: textarea
-    attributes:
-      label: Environment Details
-      description: "Provide details such as OS, Python version, and any relevant software or dependencies."
-      placeholder: e.g., CentOS Linux 7, RTX 3090, Python 3.10, torch==2.3.0, cuda 11.8
-    validations:
-      required: true
-  - type: textarea
-    attributes:
-      label: Steps to Reproduce
-      description: |
-        Include detailed steps, screenshots, and logs. Use the correct markdown syntax for code blocks.
-      placeholder: |
-        1. Create a new conda environment.
-        2. Clone the repository, install as local editable and properly set up.
-        3. Run the command: `accelerate launch src/f5_tts/train/train.py`.
-        4. Have following error message... (attach logs).
-    validations:
-      required: true
-  - type: textarea
-    attributes:
-      label: ✔️ Expected Behavior
-      placeholder: Describe what you expected to happen.
-    validations:
-      required: false
-  - type: textarea
-    attributes:
-      label: ❌ Actual Behavior
-      placeholder: Describe what actually happened.
-    validations:
-      required: false

.github/ISSUE_TEMPLATE/config.yml

@@ -1 +0,0 @@
-blank_issues_enabled: false

.github/ISSUE_TEMPLATE/feature_request.yml

@@ -1,62 +0,0 @@
-name: "Feature Request"
-description: |
-  Some constructive suggestions and new ideas regarding current repo.
-labels:
-  - enhancement
-body:
-  - type: checkboxes
-    attributes:
-      label: Checks
-      description: "To help us grasp quickly, please confirm the following:"
-      options:
-        - label: This template is only for feature request.
-          required: true
-        - label: I have thoroughly reviewed the project documentation but couldn't find any relevant information that meets my needs.
-          required: true
-        - label: I have searched for existing issues, including closed ones, and found not discussion yet.
-          required: true
-        - label: I confirm that I am using English to submit this report in order to facilitate communication.
-          required: true
-  - type: textarea
-    attributes:
-      label: 1. Is this request related to a challenge you're experiencing? Tell us your story.
-      description: |
-        Describe the specific problem or scenario you're facing in detail. For example:
-        *"I was trying to use [feature] for [specific task], but encountered [issue]. This was frustrating because...."*
-      placeholder: Please describe the situation in as much detail as possible.
-    validations:
-      required: true
-
-  - type: textarea
-    attributes:
-      label: 2. What is your suggested solution?
-      description: |
-        Provide a clear description of the feature or enhancement you'd like to propose. 
-        How would this feature solve your issue or improve the project?
-      placeholder: Describe your idea or proposed solution here.
-    validations:
-      required: true
-
-  - type: textarea
-    attributes:
-      label: 3. Additional context or comments
-      description: |
-        Any other relevant information, links, documents, or screenshots that provide clarity. 
-        Use this section for anything not covered above.
-      placeholder: Add any extra details here.
-    validations:
-      required: false
-
-  - type: checkboxes
-    attributes:
-      label: 4. Can you help us with this feature?
-      description: |
-        Let us know if you're interested in contributing. This is not a commitment but a way to express interest in collaboration.
-      options:
-        - label: I am interested in contributing to this feature.
-          required: false
-
-  - type: markdown
-    attributes:
-      value: |
-        **Note:** Please submit only one request per issue to keep discussions focused and manageable.

.github/ISSUE_TEMPLATE/help_wanted.yml

@@ -1,50 +0,0 @@
-name: "Help Wanted"
-description: |
-  Please provide as much details to help address the issue, including logs and screenshots.
-labels:
-  - help wanted
-body:
-  - type: checkboxes
-    attributes:
-      label: Checks
-      description: "To ensure timely help, please confirm the following:"
-      options:
-        - label: This template is only for usage issues encountered.
-          required: true
-        - label: I have thoroughly reviewed the project documentation but couldn't find information to solve my problem.
-          required: true
-        - label: I have searched for existing issues, including closed ones, and couldn't find a solution.
-          required: true
-        - label: I confirm that I am using English to submit this report in order to facilitate communication.
-          required: true
-  - type: textarea
-    attributes:
-      label: Environment Details
-      description: "Provide details such as OS, Python version, and any relevant software or dependencies."
-      placeholder: e.g., macOS 13.5, Python 3.10, torch==2.3.0, Gradio 4.44.1
-    validations:
-      required: true
-  - type: textarea
-    attributes:
-      label: Steps to Reproduce
-      description: |
-        Include detailed steps, screenshots, and logs. Use the correct markdown syntax for code blocks.
-      placeholder: |
-        1. Create a new conda environment.
-        2. Clone the repository and install as pip package.
-        3. Run the command: `f5-tts_infer-gradio` with no ref_text provided.
-        4. Stuck there with the following message... (attach logs and also error msg e.g. after ctrl-c).
-    validations:
-      required: true
-  - type: textarea
-    attributes:
-      label: ✔️ Expected Behavior
-      placeholder: Describe what you expected to happen, e.g. output a generated audio
-    validations:
-      required: false
-  - type: textarea
-    attributes:
-      label: ❌ Actual Behavior
-      placeholder: Describe what actually happened, failure messages, etc.
-    validations:
-      required: false

.github/ISSUE_TEMPLATE/question.yml

@@ -1,26 +0,0 @@
-name: "Question"
-description: |
-  Pure question or inquiry about the project, usage issue goes with "help wanted".
-labels:
-  - question
-body:
-  - type: checkboxes
-    attributes:
-      label: Checks
-      description: "To help us grasp quickly, please confirm the following:"
-      options:
-        - label: This template is only for question, not feature requests or bug reports.
-          required: true
-        - label: I have thoroughly reviewed the project documentation and read the related paper(s).
-          required: true
-        - label: I have searched for existing issues, including closed ones, no similar questions.
-          required: true
-        - label: I confirm that I am using English to submit this report in order to facilitate communication.
-          required: true
-  - type: textarea
-    attributes:
-      label: Question details
-      description: |
-        Question details, clearly stated using proper markdown syntax.
-    validations:
-      required: true

.github/workflows/pre-commit.yaml

@@ -1,14 +0,0 @@
-name: pre-commit
-
-on:
-  pull_request:
-  push:
-    branches: [main]
-
-jobs:
-  pre-commit:
-    runs-on: ubuntu-latest
-    steps:
-      - uses: actions/checkout@v3
-      - uses: actions/setup-python@v3
-      - uses: pre-commit/action@v3.0.1

.github/workflows/publish-docker-image.yaml

@@ -1,60 +0,0 @@
-name: Create and publish a Docker image
-
-# Configures this workflow to run every time a change is pushed to the branch called `release`.
-on:
-  push:
-    branches: ['main']
-
-# Defines two custom environment variables for the workflow. These are used for the Container registry domain, and a name for the Docker image that this workflow builds.
-env:
-  REGISTRY: ghcr.io
-  IMAGE_NAME: ${{ github.repository }}
-
-# There is a single job in this workflow. It's configured to run on the latest available version of Ubuntu.
-jobs:
-  build-and-push-image:
-    runs-on: ubuntu-latest
-    # Sets the permissions granted to the `GITHUB_TOKEN` for the actions in this job.
-    permissions:
-      contents: read
-      packages: write
-      # 
-    steps:
-      - name: Checkout repository
-        uses: actions/checkout@v4
-      - name: Free Up GitHub Actions Ubuntu Runner Disk Space 🔧
-        uses: jlumbroso/free-disk-space@main
-        with:
-          # This might remove tools that are actually needed, if set to "true" but frees about 6 GB
-          tool-cache: false
-
-          # All of these default to true, but feel free to set to "false" if necessary for your workflow
-          android: true
-          dotnet: true
-          haskell: true
-          large-packages: false
-          swap-storage: false
-          docker-images: false
-      # Uses the `docker/login-action` action to log in to the Container registry registry using the account and password that will publish the packages. Once published, the packages are scoped to the account defined here.
-      - name: Log in to the Container registry
-        uses: docker/login-action@65b78e6e13532edd9afa3aa52ac7964289d1a9c1
-        with:
-          registry: ${{ env.REGISTRY }}
-          username: ${{ github.actor }}
-          password: ${{ secrets.GITHUB_TOKEN }}
-      # This step uses [docker/metadata-action](https://github.com/docker/metadata-action#about) to extract tags and labels that will be applied to the specified image. The `id` "meta" allows the output of this step to be referenced in a subsequent step. The `images` value provides the base name for the tags and labels.
-      - name: Extract metadata (tags, labels) for Docker
-        id: meta
-        uses: docker/metadata-action@9ec57ed1fcdbf14dcef7dfbe97b2010124a938b7
-        with:
-          images: ${{ env.REGISTRY }}/${{ env.IMAGE_NAME }}
-      # This step uses the `docker/build-push-action` action to build the image, based on your repository's `Dockerfile`. If the build succeeds, it pushes the image to GitHub Packages.
-      # It uses the `context` parameter to define the build's context as the set of files located in the specified path. For more information, see "[Usage](https://github.com/docker/build-push-action#usage)" in the README of the `docker/build-push-action` repository.
-      # It uses the `tags` and `labels` parameters to tag and label the image with the output from the "meta" step.
-      - name: Build and push Docker image
-        uses: docker/build-push-action@f2a1d5e99d037542a71f64918e516c093c6f3fc4
-        with:
-          context: .
-          push: true
-          tags: ${{ steps.meta.outputs.tags }}
-          labels: ${{ steps.meta.outputs.labels }}

.gitmodules

@@ -1,3 +0,0 @@
-[submodule "src/third_party/BigVGAN"]
-	path = src/third_party/BigVGAN
-	url = https://github.com/NVIDIA/BigVGAN.git

.pre-commit-config.yaml

@@ -1,14 +0,0 @@
-repos:
-  - repo: https://github.com/astral-sh/ruff-pre-commit
-    # Ruff version.
-    rev: v0.7.0
-    hooks:
-      # Run the linter.
-      - id: ruff
-        args: [--fix]
-      # Run the formatter.
-      - id: ruff-format
-  - repo: https://github.com/pre-commit/pre-commit-hooks
-    rev: v2.3.0
-    hooks:
-      - id: check-yaml

Dockerfile

@@ -10,17 +10,15 @@ RUN set -x \
     && apt-get update \
     && apt-get -y install wget curl man git less openssl libssl-dev unzip unar build-essential aria2 tmux vim \
     && apt-get install -y openssh-server sox libsox-fmt-all libsox-fmt-mp3 libsndfile1-dev ffmpeg \
-    && apt-get install -y librdmacm1 libibumad3 librdmacm-dev libibverbs1 libibverbs-dev ibverbs-utils ibverbs-providers \
     && rm -rf /var/lib/apt/lists/* \
     && apt-get clean
-    
+
 WORKDIR /workspace
 
 RUN git clone https://github.com/SWivid/F5-TTS.git \
     && cd F5-TTS \
-    && git submodule update --init --recursive \
-    && sed -i '7iimport sys\nsys.path.append(os.path.dirname(os.path.abspath(__file__)))' src/third_party/BigVGAN/bigvgan.py \
-    && pip install -e . --no-cache-dir
+    && pip install --no-cache-dir -r requirements.txt \
+    && pip install --no-cache-dir -r requirements_eval.txt
 
 ENV SHELL=/bin/bash
 

README_REPO.md

@@ -16,146 +16,188 @@
 
 ### Thanks to all the contributors !
 
-## News
-- **2024/10/08**: F5-TTS & E2 TTS base models on [🤗 Hugging Face](https://huggingface.co/SWivid/F5-TTS), [🤖 Model Scope](https://www.modelscope.cn/models/SWivid/F5-TTS_Emilia-ZH-EN), [🟣 Wisemodel](https://wisemodel.cn/models/SJTU_X-LANCE/F5-TTS_Emilia-ZH-EN).
-
 ## Installation
 
-```bash
-# Create a python 3.10 conda env (you could also use virtualenv)
-conda create -n f5-tts python=3.10
-conda activate f5-tts
+Clone the repository:
 
-# Install pytorch with your CUDA version, e.g.
-pip install torch==2.3.0+cu118 torchaudio==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
+```bash
+git clone https://github.com/SWivid/F5-TTS.git
+cd F5-TTS
 ```
 
-Then you can choose from a few options below:
-
-### 1. As a pip package (if just for inference)
+Install torch with your CUDA version, e.g. :
 
 ```bash
-pip install git+https://github.com/SWivid/F5-TTS.git
+pip install torch==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
+pip install torchaudio==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
 ```
 
-### 2. Local editable (if also do training, finetuning)
+Install other packages:
 
 ```bash
-git clone https://github.com/SWivid/F5-TTS.git
-cd F5-TTS
-# git submodule update --init --recursive  # (optional, if need bigvgan)
-pip install -e .
-```
-If initialize submodule, you should add the following code at the beginning of `src/third_party/BigVGAN/bigvgan.py`.
-```python
-import os
-import sys
-sys.path.append(os.path.dirname(os.path.abspath(__file__)))
+pip install -r requirements.txt
 ```
 
-### 3. Docker usage
+## Prepare Dataset
+
+Example data processing scripts for Emilia and Wenetspeech4TTS, and you may tailor your own one along with a Dataset class in `model/dataset.py`.
+
 ```bash
-# Build from Dockerfile
-docker build -t f5tts:v1 .
+# prepare custom dataset up to your need
+# download corresponding dataset first, and fill in the path in scripts
+
+# Prepare the Emilia dataset
+python scripts/prepare_emilia.py
 
-# Or pull from GitHub Container Registry
-docker pull ghcr.io/swivid/f5-tts:main
+# Prepare the Wenetspeech4TTS dataset
+python scripts/prepare_wenetspeech4tts.py
 ```
 
+## Training & Finetuning
+
+Once your datasets are prepared, you can start the training process.
+
+```bash
+# setup accelerate config, e.g. use multi-gpu ddp, fp16
+# will be to: ~/.cache/huggingface/accelerate/default_config.yaml     
+accelerate config
+accelerate launch train.py
+```
+An initial guidance on Finetuning [#57](https://github.com/SWivid/F5-TTS/discussions/57).
+
+Gradio UI finetuning with `finetune_gradio.py` see [#143](https://github.com/SWivid/F5-TTS/discussions/143).
 
 ## Inference
 
-### 1. Gradio App
+The pretrained model checkpoints can be reached at [🤗 Hugging Face](https://huggingface.co/SWivid/F5-TTS) and [🤖 Model Scope](https://www.modelscope.cn/models/SWivid/F5-TTS_Emilia-ZH-EN), or automatically downloaded with `inference-cli` and `gradio_app`.
 
-Currently supported features:
+Currently support 30s for a single generation, which is the **TOTAL** length of prompt audio and the generated. Batch inference with chunks is supported by `inference-cli` and `gradio_app`. 
+- To avoid possible inference failures, make sure you have seen through the following instructions.
+- A longer prompt audio allows shorter generated output. The part longer than 30s cannot be generated properly. Consider using a prompt audio <15s.
+- Uppercased letters will be uttered letter by letter, so use lowercased letters for normal words. 
+- Add some spaces (blank: " ") or punctuations (e.g. "," ".") to explicitly introduce some pauses. If first few words skipped in code-switched generation (cuz different speed with different languages), this might help.
+
+### CLI Inference
 
-- Basic TTS with Chunk Inference
-- Multi-Style / Multi-Speaker Generation
-- Voice Chat powered by Qwen2.5-3B-Instruct
-- [Custom inference with more language support](src/f5_tts/infer/SHARED.md)
+Either you can specify everything in `inference-cli.toml` or override with flags. Leave `--ref_text ""` will have ASR model transcribe the reference audio automatically (use extra GPU memory). If encounter network error, consider use local ckpt, just set `ckpt_path` in `inference-cli.py`
 
 ```bash
-# Launch a Gradio app (web interface)
-f5-tts_infer-gradio
+python inference-cli.py \
+--model "F5-TTS" \
+--ref_audio "tests/ref_audio/test_en_1_ref_short.wav" \
+--ref_text "Some call me nature, others call me mother nature." \
+--gen_text "I don't really care what you call me. I've been a silent spectator, watching species evolve, empires rise and fall. But always remember, I am mighty and enduring. Respect me and I'll nurture you; ignore me and you shall face the consequences."
+
+python inference-cli.py \
+--model "E2-TTS" \
+--ref_audio "tests/ref_audio/test_zh_1_ref_short.wav" \
+--ref_text "对，这就是我，万人敬仰的太乙真人。" \
+--gen_text "突然，身边一阵笑声。我看着他们，意气风发地挺���了胸膛，甩了甩那稍显肉感的双臂，轻笑道，我身上的肉，是为了掩饰我爆棚的魅力，否则，岂不吓坏了你们呢？"
+
+# Multi voice
+python inference-cli.py -c samples/story.toml
+```
 
-# Specify the port/host
-f5-tts_infer-gradio --port 7860 --host 0.0.0.0
+### Gradio App
+Currently supported features:
+- Chunk inference
+- Podcast Generation
+- Multiple Speech-Type Generation
 
-# Launch a share link
-f5-tts_infer-gradio --share
+You can launch a Gradio app (web interface) to launch a GUI for inference (will load ckpt from Huggingface, you may set `ckpt_path` to local file in `gradio_app.py`). Currently load ASR model, F5-TTS and E2 TTS all in once, thus use more GPU memory than `inference-cli`.
+
+```bash
+python gradio_app.py
 ```
 
-### 2. CLI Inference
+You can specify the port/host:
 
 ```bash
-# Run with flags
-# Leave --ref_text "" will have ASR model transcribe (extra GPU memory usage)
-f5-tts_infer-cli \
---model "F5-TTS" \
---ref_audio "ref_audio.wav" \
---ref_text "The content, subtitle or transcription of reference audio." \
---gen_text "Some text you want TTS model generate for you."
+python gradio_app.py --port 7860 --host 0.0.0.0
+```
 
-# Run with default setting. src/f5_tts/infer/examples/basic/basic.toml
-f5-tts_infer-cli
-# Or with your own .toml file
-f5-tts_infer-cli -c custom.toml
+Or launch a share link:
 
-# Multi voice. See src/f5_tts/infer/README.md
-f5-tts_infer-cli -c src/f5_tts/infer/examples/multi/story.toml
+```bash
+python gradio_app.py --share
 ```
 
-### 3. More instructions
+### Speech Editing
+
+To test speech editing capabilities, use the following command.
 
-- In order to have better generation results, take a moment to read [detailed guidance](src/f5_tts/infer).
-- The [Issues](https://github.com/SWivid/F5-TTS/issues?q=is%3Aissue) are very useful, please try to find the solution by properly searching the keywords of problem encountered. If no answer found, then feel free to open an issue.
+```bash
+python speech_edit.py
+```
+
+## Evaluation
 
+### Prepare Test Datasets
 
-## Training
+1. Seed-TTS test set: Download from [seed-tts-eval](https://github.com/BytedanceSpeech/seed-tts-eval).
+2. LibriSpeech test-clean: Download from [OpenSLR](http://www.openslr.org/12/).
+3. Unzip the downloaded datasets and place them in the data/ directory.
+4. Update the path for the test-clean data in `scripts/eval_infer_batch.py`
+5. Our filtered LibriSpeech-PC 4-10s subset is already under data/ in this repo
 
-### 1. Gradio App
+### Batch Inference for Test Set
 
-Read [training & finetuning guidance](src/f5_tts/train) for more instructions.
+To run batch inference for evaluations, execute the following commands:
 
 ```bash
-# Quick start with Gradio web interface
-f5-tts_finetune-gradio
+# batch inference for evaluations
+accelerate config  # if not set before
+bash scripts/eval_infer_batch.sh
 ```
 
+### Download Evaluation Model Checkpoints
 
-## [Evaluation](src/f5_tts/eval)
+1. Chinese ASR Model: [Paraformer-zh](https://huggingface.co/funasr/paraformer-zh)
+2. English ASR Model: [Faster-Whisper](https://huggingface.co/Systran/faster-whisper-large-v3)
+3. WavLM Model: Download from [Google Drive](https://drive.google.com/file/d/1-aE1NfzpRCLxA4GUxX9ITI3F9LlbtEGP/view).
 
+### Objective Evaluation
 
-## Development
+Install packages for evaluation:
 
-Use pre-commit to ensure code quality (will run linters and formatters automatically)
+```bash
+pip install -r requirements_eval.txt
+```
+
+**Some Notes**
+
+For faster-whisper with CUDA 11:
 
 ```bash
-pip install pre-commit
-pre-commit install
+pip install --force-reinstall ctranslate2==3.24.0
 ```
 
-When making a pull request, before each commit, run: 
+(Recommended) To avoid possible ASR failures, such as abnormal repetitions in output:
 
 ```bash
-pre-commit run --all-files
+pip install faster-whisper==0.10.1
 ```
 
-Note: Some model components have linting exceptions for E722 to accommodate tensor notation
+Update the path with your batch-inferenced results, and carry out WER / SIM evaluations:
+```bash
+# Evaluation for Seed-TTS test set
+python scripts/eval_seedtts_testset.py
 
+# Evaluation for LibriSpeech-PC test-clean (cross-sentence)
+python scripts/eval_librispeech_test_clean.py
+```
 
 ## Acknowledgements
 
 - [E2-TTS](https://arxiv.org/abs/2406.18009) brilliant work, simple and effective
-- [Emilia](https://arxiv.org/abs/2407.05361), [WenetSpeech4TTS](https://arxiv.org/abs/2406.05763), [LibriTTS](https://arxiv.org/abs/1904.02882), [LJSpeech](https://keithito.com/LJ-Speech-Dataset/) valuable datasets
+- [Emilia](https://arxiv.org/abs/2407.05361), [WenetSpeech4TTS](https://arxiv.org/abs/2406.05763) valuable datasets
 - [lucidrains](https://github.com/lucidrains) initial CFM structure with also [bfs18](https://github.com/bfs18) for discussion
 - [SD3](https://arxiv.org/abs/2403.03206) & [Hugging Face diffusers](https://github.com/huggingface/diffusers) DiT and MMDiT code structure
-- [torchdiffeq](https://github.com/rtqichen/torchdiffeq) as ODE solver, [Vocos](https://huggingface.co/charactr/vocos-mel-24khz) and [BigVGAN](https://github.com/NVIDIA/BigVGAN) as vocoder
-- [FunASR](https://github.com/modelscope/FunASR), [faster-whisper](https://github.com/SYSTRAN/faster-whisper), [UniSpeech](https://github.com/microsoft/UniSpeech), [SpeechMOS](https://github.com/tarepan/SpeechMOS) for evaluation tools
+- [torchdiffeq](https://github.com/rtqichen/torchdiffeq) as ODE solver, [Vocos](https://huggingface.co/charactr/vocos-mel-24khz) as vocoder
+- [FunASR](https://github.com/modelscope/FunASR), [faster-whisper](https://github.com/SYSTRAN/faster-whisper), [UniSpeech](https://github.com/microsoft/UniSpeech) for evaluation tools
 - [ctc-forced-aligner](https://github.com/MahmoudAshraf97/ctc-forced-aligner) for speech edit test
 - [mrfakename](https://x.com/realmrfakename) huggingface space demo ~
-- [f5-tts-mlx](https://github.com/lucasnewman/f5-tts-mlx/tree/main) Implementation with MLX framework by [Lucas Newman](https://github.com/lucasnewman)
-- [F5-TTS-ONNX](https://github.com/DakeQQ/F5-TTS-ONNX) ONNX Runtime version by [DakeQQ](https://github.com/DakeQQ)
+- [f5-tts-mlx](https://github.com/lucasnewman/f5-tts-mlx/tree/main) Implementation of F5-TTS, with the MLX framework.
 
 ## Citation
 If our work and codebase is useful for you, please cite as:

api.py

@@ -1,132 +0,0 @@
-import soundfile as sf
-import torch
-import tqdm
-from cached_path import cached_path
-
-from model import DiT, UNetT
-from model.utils import save_spectrogram
-
-from model.utils_infer import load_vocoder, load_model, infer_process, remove_silence_for_generated_wav
-from model.utils import seed_everything
-import random
-import sys
-
-
-class F5TTS:
-    def __init__(
-        self,
-        model_type="F5-TTS",
-        ckpt_file="",
-        vocab_file="",
-        ode_method="euler",
-        use_ema=True,
-        local_path=None,
-        device=None,
-    ):
-        # Initialize parameters
-        self.final_wave = None
-        self.target_sample_rate = 24000
-        self.n_mel_channels = 100
-        self.hop_length = 256
-        self.target_rms = 0.1
-        self.seed = -1
-
-        # Set device
-        self.device = device or (
-            "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu"
-        )
-
-        # Load models
-        self.load_vocoder_model(local_path)
-        self.load_ema_model(model_type, ckpt_file, vocab_file, ode_method, use_ema)
-
-    def load_vocoder_model(self, local_path):
-        self.vocos = load_vocoder(local_path is not None, local_path, self.device)
-
-    def load_ema_model(self, model_type, ckpt_file, vocab_file, ode_method, use_ema):
-        if model_type == "F5-TTS":
-            if not ckpt_file:
-                ckpt_file = str(cached_path("hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.safetensors"))
-            model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
-            model_cls = DiT
-        elif model_type == "E2-TTS":
-            if not ckpt_file:
-                ckpt_file = str(cached_path("hf://SWivid/E2-TTS/E2TTS_Base/model_1200000.safetensors"))
-            model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
-            model_cls = UNetT
-        else:
-            raise ValueError(f"Unknown model type: {model_type}")
-
-        self.ema_model = load_model(model_cls, model_cfg, ckpt_file, vocab_file, ode_method, use_ema, self.device)
-
-    def export_wav(self, wav, file_wave, remove_silence=False):
-        sf.write(file_wave, wav, self.target_sample_rate)
-
-        if remove_silence:
-            remove_silence_for_generated_wav(file_wave)
-
-    def export_spectrogram(self, spect, file_spect):
-        save_spectrogram(spect, file_spect)
-
-    def infer(
-        self,
-        ref_file,
-        ref_text,
-        gen_text,
-        show_info=print,
-        progress=tqdm,
-        target_rms=0.1,
-        cross_fade_duration=0.15,
-        sway_sampling_coef=-1,
-        cfg_strength=2,
-        nfe_step=32,
-        speed=1.0,
-        fix_duration=None,
-        remove_silence=False,
-        file_wave=None,
-        file_spect=None,
-        seed=-1,
-    ):
-        if seed == -1:
-            seed = random.randint(0, sys.maxsize)
-        seed_everything(seed)
-        self.seed = seed
-        wav, sr, spect = infer_process(
-            ref_file,
-            ref_text,
-            gen_text,
-            self.ema_model,
-            show_info=show_info,
-            progress=progress,
-            target_rms=target_rms,
-            cross_fade_duration=cross_fade_duration,
-            nfe_step=nfe_step,
-            cfg_strength=cfg_strength,
-            sway_sampling_coef=sway_sampling_coef,
-            speed=speed,
-            fix_duration=fix_duration,
-            device=self.device,
-        )
-
-        if file_wave is not None:
-            self.export_wav(wav, file_wave, remove_silence)
-
-        if file_spect is not None:
-            self.export_spectrogram(spect, file_spect)
-
-        return wav, sr, spect
-
-
-if __name__ == "__main__":
-    f5tts = F5TTS()
-
-    wav, sr, spect = f5tts.infer(
-        ref_file="tests/ref_audio/test_en_1_ref_short.wav",
-        ref_text="some call me nature, others call me mother nature.",
-        gen_text="""I don't really care what you call me. I've been a silent spectator, watching species evolve, empires rise and fall. But always remember, I am mighty and enduring. Respect me and I'll nurture you; ignore me and you shall face the consequences.""",
-        file_wave="tests/out.wav",
-        file_spect="tests/out.png",
-        seed=-1,  # random seed = -1
-    )
-
-    print("seed :", f5tts.seed)

app.py

@@ -1,193 +1,403 @@
-# ruff: noqa: E402
-# Above allows ruff to ignore E402: module level import not at top of file
-
-import json
 import re
-import tempfile
-from collections import OrderedDict
-from importlib.resources import files
-
-import click
+import torch
+import torchaudio
 import gradio as gr
 import numpy as np
-import soundfile as sf
-import torchaudio
+import tempfile
+from einops import rearrange
+from vocos import Vocos
+from pydub import AudioSegment, silence
+from model import CFM, UNetT, DiT, MMDiT
 from cached_path import cached_path
-from transformers import AutoModelForCausalLM, AutoTokenizer
+from model.utils import (
+    load_checkpoint,
+    get_tokenizer,
+    convert_char_to_pinyin,
+    save_spectrogram,
+)
+from transformers import pipeline
+import click
+import soundfile as sf
 
 try:
     import spaces
-
     USING_SPACES = True
 except ImportError:
     USING_SPACES = False
 
-
 def gpu_decorator(func):
     if USING_SPACES:
         return spaces.GPU(func)
     else:
         return func
 
-
-from f5_tts.model import DiT, UNetT
-from f5_tts.infer.utils_infer import (
-    load_vocoder,
-    load_model,
-    preprocess_ref_audio_text,
-    infer_process,
-    remove_silence_for_generated_wav,
-    save_spectrogram,
+device = (
+    "cuda"
+    if torch.cuda.is_available()
+    else "mps" if torch.backends.mps.is_available() else "cpu"
 )
 
+print(f"Using {device} device")
 
-DEFAULT_TTS_MODEL = "F5-TTS"
-tts_model_choice = DEFAULT_TTS_MODEL
-
-DEFAULT_TTS_MODEL_CFG = [
-    "hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.safetensors",
-    "hf://SWivid/F5-TTS/F5TTS_Base/vocab.txt",
-    json.dumps(dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)),
-]
+pipe = pipeline(
+    "automatic-speech-recognition",
+    model="openai/whisper-large-v3-turbo",
+    torch_dtype=torch.float16,
+    device=device,
+)
+vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
+
+# --------------------- Settings -------------------- #
+
+target_sample_rate = 24000
+n_mel_channels = 100
+hop_length = 256
+target_rms = 0.1
+nfe_step = 32  # 16, 32
+cfg_strength = 2.0
+ode_method = "euler"
+sway_sampling_coef = -1.0
+speed = 1.0
+fix_duration = None
+
+
+def load_model(repo_name, exp_name, model_cls, model_cfg, ckpt_step):
+    ckpt_path = str(cached_path(f"hf://SWivid/{repo_name}/{exp_name}/model_{ckpt_step}.safetensors"))
+    # ckpt_path = f"ckpts/{exp_name}/model_{ckpt_step}.pt"  # .pt | .safetensors
+    vocab_char_map, vocab_size = get_tokenizer("Emilia_ZH_EN", "pinyin")
+    model = CFM(
+        transformer=model_cls(
+            **model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels
+        ),
+        mel_spec_kwargs=dict(
+            target_sample_rate=target_sample_rate,
+            n_mel_channels=n_mel_channels,
+            hop_length=hop_length,
+        ),
+        odeint_kwargs=dict(
+            method=ode_method,
+        ),
+        vocab_char_map=vocab_char_map,
+    ).to(device)
+
+    model = load_checkpoint(model, ckpt_path, device, use_ema = True)
+
+    return model
 
 
 # load models
+F5TTS_model_cfg = dict(
+    dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4
+)
+E2TTS_model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
 
-vocoder = load_vocoder()
+F5TTS_ema_model = load_model(
+    "F5-TTS", "F5TTS_Base", DiT, F5TTS_model_cfg, 1200000
+)
+E2TTS_ema_model = load_model(
+    "E2-TTS", "E2TTS_Base", UNetT, E2TTS_model_cfg, 1200000
+)
 
+def chunk_text(text, max_chars=135):
+    """
+    Splits the input text into chunks, each with a maximum number of characters.
 
-def load_f5tts(ckpt_path=str(cached_path("hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.safetensors"))):
-    F5TTS_model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
-    return load_model(DiT, F5TTS_model_cfg, ckpt_path)
+    Args:
+        text (str): The text to be split.
+        max_chars (int): The maximum number of characters per chunk.
 
+    Returns:
+        List[str]: A list of text chunks.
+    """
+    chunks = []
+    current_chunk = ""
+    # Split the text into sentences based on punctuation followed by whitespace
+    sentences = re.split(r'(?<=[;:,.!?])\s+|(?<=[；：，。！？])', text)
+
+    for sentence in sentences:
+        if len(current_chunk.encode('utf-8')) + len(sentence.encode('utf-8')) <= max_chars:
+            current_chunk += sentence + " " if sentence and len(sentence[-1].encode('utf-8')) == 1 else sentence
+        else:
+            if current_chunk:
+                chunks.append(current_chunk.strip())
+            current_chunk = sentence + " " if sentence and len(sentence[-1].encode('utf-8')) == 1 else sentence
 
-def load_e2tts(ckpt_path=str(cached_path("hf://SWivid/E2-TTS/E2TTS_Base/model_1200000.safetensors"))):
-    E2TTS_model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
-    return load_model(UNetT, E2TTS_model_cfg, ckpt_path)
+    if current_chunk:
+        chunks.append(current_chunk.strip())
 
+    return chunks
 
-def load_custom(ckpt_path: str, vocab_path="", model_cfg=None):
-    ckpt_path, vocab_path = ckpt_path.strip(), vocab_path.strip()
-    if ckpt_path.startswith("hf://"):
-        ckpt_path = str(cached_path(ckpt_path))
-    if vocab_path.startswith("hf://"):
-        vocab_path = str(cached_path(vocab_path))
-    if model_cfg is None:
-        model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
-    return load_model(DiT, model_cfg, ckpt_path, vocab_file=vocab_path)
+@gpu_decorator
+def infer_batch(ref_audio, ref_text, gen_text_batches, exp_name, remove_silence, cross_fade_duration=0.15, progress=gr.Progress()):
+    if exp_name == "F5-TTS":
+        ema_model = F5TTS_ema_model
+    elif exp_name == "E2-TTS":
+        ema_model = E2TTS_ema_model
 
+    audio, sr = ref_audio
+    if audio.shape[0] > 1:
+        audio = torch.mean(audio, dim=0, keepdim=True)
+
+    rms = torch.sqrt(torch.mean(torch.square(audio)))
+    if rms < target_rms:
+        audio = audio * target_rms / rms
+    if sr != target_sample_rate:
+        resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
+        audio = resampler(audio)
+    audio = audio.to(device)
+
+    generated_waves = []
+    spectrograms = []
+
+    for i, gen_text in enumerate(progress.tqdm(gen_text_batches)):
+        # Prepare the text
+        if len(ref_text[-1].encode('utf-8')) == 1:
+            ref_text = ref_text + " "
+        text_list = [ref_text + gen_text]
+        final_text_list = convert_char_to_pinyin(text_list)
+
+        # Calculate duration
+        ref_audio_len = audio.shape[-1] // hop_length
+        zh_pause_punc = r"。，、；：？！"
+        ref_text_len = len(ref_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, ref_text))
+        gen_text_len = len(gen_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, gen_text))
+        duration = ref_audio_len + int(ref_audio_len / ref_text_len * gen_text_len / speed)
+
+        # inference
+        with torch.inference_mode():
+            generated, _ = ema_model.sample(
+                cond=audio,
+                text=final_text_list,
+                duration=duration,
+                steps=nfe_step,
+                cfg_strength=cfg_strength,
+                sway_sampling_coef=sway_sampling_coef,
+            )
 
-F5TTS_ema_model = load_f5tts()
-E2TTS_ema_model = load_e2tts() if USING_SPACES else None
-custom_ema_model, pre_custom_path = None, ""
+        generated = generated[:, ref_audio_len:, :]
+        generated_mel_spec = rearrange(generated, "1 n d -> 1 d n")
+        generated_wave = vocos.decode(generated_mel_spec.cpu())
+        if rms < target_rms:
+            generated_wave = generated_wave * rms / target_rms
+
+        # wav -> numpy
+        generated_wave = generated_wave.squeeze().cpu().numpy()
+        
+        generated_waves.append(generated_wave)
+        spectrograms.append(generated_mel_spec[0].cpu().numpy())
+
+    # Combine all generated waves with cross-fading
+    if cross_fade_duration <= 0:
+        # Simply concatenate
+        final_wave = np.concatenate(generated_waves)
+    else:
+        final_wave = generated_waves[0]
+        for i in range(1, len(generated_waves)):
+            prev_wave = final_wave
+            next_wave = generated_waves[i]
 
-chat_model_state = None
-chat_tokenizer_state = None
+            # Calculate cross-fade samples, ensuring it does not exceed wave lengths
+            cross_fade_samples = int(cross_fade_duration * target_sample_rate)
+            cross_fade_samples = min(cross_fade_samples, len(prev_wave), len(next_wave))
 
+            if cross_fade_samples <= 0:
+                # No overlap possible, concatenate
+                final_wave = np.concatenate([prev_wave, next_wave])
+                continue
 
-@gpu_decorator
-def generate_response(messages, model, tokenizer):
-    """Generate response using Qwen"""
-    text = tokenizer.apply_chat_template(
-        messages,
-        tokenize=False,
-        add_generation_prompt=True,
-    )
+            # Overlapping parts
+            prev_overlap = prev_wave[-cross_fade_samples:]
+            next_overlap = next_wave[:cross_fade_samples]
 
-    model_inputs = tokenizer([text], return_tensors="pt").to(model.device)
-    generated_ids = model.generate(
-        **model_inputs,
-        max_new_tokens=512,
-        temperature=0.7,
-        top_p=0.95,
-    )
+            # Fade out and fade in
+            fade_out = np.linspace(1, 0, cross_fade_samples)
+            fade_in = np.linspace(0, 1, cross_fade_samples)
 
-    generated_ids = [
-        output_ids[len(input_ids) :] for input_ids, output_ids in zip(model_inputs.input_ids, generated_ids)
-    ]
-    return tokenizer.batch_decode(generated_ids, skip_special_tokens=True)[0]
+            # Cross-faded overlap
+            cross_faded_overlap = prev_overlap * fade_out + next_overlap * fade_in
 
+            # Combine
+            new_wave = np.concatenate([
+                prev_wave[:-cross_fade_samples],
+                cross_faded_overlap,
+                next_wave[cross_fade_samples:]
+            ])
 
-@gpu_decorator
-def infer(
-    ref_audio_orig,
-    ref_text,
-    gen_text,
-    model,
-    remove_silence,
-    cross_fade_duration=0.15,
-    nfe_step=32,
-    speed=1,
-    show_info=gr.Info,
-):
-    if not ref_audio_orig:
-        gr.Warning("Please provide reference audio.")
-        return gr.update(), gr.update(), ref_text
-
-    if not gen_text.strip():
-        gr.Warning("Please enter text to generate.")
-        return gr.update(), gr.update(), ref_text
-
-    ref_audio, ref_text = preprocess_ref_audio_text(ref_audio_orig, ref_text, show_info=show_info)
-
-    if model == "F5-TTS":
-        ema_model = F5TTS_ema_model
-    elif model == "E2-TTS":
-        global E2TTS_ema_model
-        if E2TTS_ema_model is None:
-            show_info("Loading E2-TTS model...")
-            E2TTS_ema_model = load_e2tts()
-        ema_model = E2TTS_ema_model
-    elif isinstance(model, list) and model[0] == "Custom":
-        assert not USING_SPACES, "Only official checkpoints allowed in Spaces."
-        global custom_ema_model, pre_custom_path
-        if pre_custom_path != model[1]:
-            show_info("Loading Custom TTS model...")
-            custom_ema_model = load_custom(model[1], vocab_path=model[2], model_cfg=model[3])
-            pre_custom_path = model[1]
-        ema_model = custom_ema_model
-
-    final_wave, final_sample_rate, combined_spectrogram = infer_process(
-        ref_audio,
-        ref_text,
-        gen_text,
-        ema_model,
-        vocoder,
-        cross_fade_duration=cross_fade_duration,
-        nfe_step=nfe_step,
-        speed=speed,
-        show_info=show_info,
-        progress=gr.Progress(),
-    )
+            final_wave = new_wave
 
     # Remove silence
     if remove_silence:
         with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
-            sf.write(f.name, final_wave, final_sample_rate)
-            remove_silence_for_generated_wav(f.name)
+            sf.write(f.name, final_wave, target_sample_rate)
+            aseg = AudioSegment.from_file(f.name)
+            non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
+            non_silent_wave = AudioSegment.silent(duration=0)
+            for non_silent_seg in non_silent_segs:
+                non_silent_wave += non_silent_seg
+            aseg = non_silent_wave
+            aseg.export(f.name, format="wav")
             final_wave, _ = torchaudio.load(f.name)
         final_wave = final_wave.squeeze().cpu().numpy()
 
-    # Save the spectrogram
+    # Create a combined spectrogram
+    combined_spectrogram = np.concatenate(spectrograms, axis=1)
+    
     with tempfile.NamedTemporaryFile(suffix=".png", delete=False) as tmp_spectrogram:
         spectrogram_path = tmp_spectrogram.name
         save_spectrogram(combined_spectrogram, spectrogram_path)
 
-    return (final_sample_rate, final_wave), spectrogram_path, ref_text
+    return (target_sample_rate, final_wave), spectrogram_path
+
+@gpu_decorator
+def infer(ref_audio_orig, ref_text, gen_text, exp_name, remove_silence, cross_fade_duration=0.15):
+
+    print(gen_text)
 
+    gr.Info("Converting audio...")
+    with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
+        aseg = AudioSegment.from_file(ref_audio_orig)
+
+        non_silent_segs = silence.split_on_silence(
+            aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=1000
+        )
+        non_silent_wave = AudioSegment.silent(duration=0)
+        for non_silent_seg in non_silent_segs:
+            non_silent_wave += non_silent_seg
+        aseg = non_silent_wave
+
+        audio_duration = len(aseg)
+        if audio_duration > 15000:
+            gr.Warning("Audio is over 15s, clipping to only first 15s.")
+            aseg = aseg[:15000]
+        aseg.export(f.name, format="wav")
+        ref_audio = f.name
+
+    if not ref_text.strip():
+        gr.Info("No reference text provided, transcribing reference audio...")
+        ref_text = pipe(
+            ref_audio,
+            chunk_length_s=30,
+            batch_size=128,
+            generate_kwargs={"task": "transcribe"},
+            return_timestamps=False,
+        )["text"].strip()
+        gr.Info("Finished transcription")
+    else:
+        gr.Info("Using custom reference text...")
+
+    # Add the functionality to ensure it ends with ". "
+    if not ref_text.endswith(". "):
+        if ref_text.endswith("."):
+            ref_text += " "
+        else:
+            ref_text += ". "
+
+    audio, sr = torchaudio.load(ref_audio)
+
+    # Use the new chunk_text function to split gen_text
+    max_chars = int(len(ref_text.encode('utf-8')) / (audio.shape[-1] / sr) * (25 - audio.shape[-1] / sr))
+    gen_text_batches = chunk_text(gen_text, max_chars=max_chars)
+    print('ref_text', ref_text)
+    for i, batch_text in enumerate(gen_text_batches):
+        print(f'gen_text {i}', batch_text)
+    
+    gr.Info(f"Generating audio using {exp_name} in {len(gen_text_batches)} batches")
+    return infer_batch((audio, sr), ref_text, gen_text_batches, exp_name, remove_silence, cross_fade_duration)
+
+
+@gpu_decorator
+def generate_podcast(script, speaker1_name, ref_audio1, ref_text1, speaker2_name, ref_audio2, ref_text2, exp_name, remove_silence):
+    # Split the script into speaker blocks
+    speaker_pattern = re.compile(f"^({re.escape(speaker1_name)}|{re.escape(speaker2_name)}):", re.MULTILINE)
+    speaker_blocks = speaker_pattern.split(script)[1:]  # Skip the first empty element
+    
+    generated_audio_segments = []
+    
+    for i in range(0, len(speaker_blocks), 2):
+        speaker = speaker_blocks[i]
+        text = speaker_blocks[i+1].strip()
+        
+        # Determine which speaker is talking
+        if speaker == speaker1_name:
+            ref_audio = ref_audio1
+            ref_text = ref_text1
+        elif speaker == speaker2_name:
+            ref_audio = ref_audio2
+            ref_text = ref_text2
+        else:
+            continue  # Skip if the speaker is neither speaker1 nor speaker2
+        
+        # Generate audio for this block
+        audio, _ = infer(ref_audio, ref_text, text, exp_name, remove_silence)
+        
+        # Convert the generated audio to a numpy array
+        sr, audio_data = audio
+        
+        # Save the audio data as a WAV file
+        with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file:
+            sf.write(temp_file.name, audio_data, sr)
+            audio_segment = AudioSegment.from_wav(temp_file.name)
+        
+        generated_audio_segments.append(audio_segment)
+        
+        # Add a short pause between speakers
+        pause = AudioSegment.silent(duration=500)  # 500ms pause
+        generated_audio_segments.append(pause)
+    
+    # Concatenate all audio segments
+    final_podcast = sum(generated_audio_segments)
+    
+    # Export the final podcast
+    with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file:
+        podcast_path = temp_file.name
+        final_podcast.export(podcast_path, format="wav")
+    
+    return podcast_path
+
+def parse_speechtypes_text(gen_text):
+    # Pattern to find (Emotion)
+    pattern = r'\((.*?)\)'
+
+    # Split the text by the pattern
+    tokens = re.split(pattern, gen_text)
+
+    segments = []
+
+    current_emotion = 'Regular'
+
+    for i in range(len(tokens)):
+        if i % 2 == 0:
+            # This is text
+            text = tokens[i].strip()
+            if text:
+                segments.append({'emotion': current_emotion, 'text': text})
+        else:
+            # This is emotion
+            emotion = tokens[i].strip()
+            current_emotion = emotion
+
+    return segments
+
+def update_speed(new_speed):
+    global speed
+    speed = new_speed
+    return f"Speed set to: {speed}"
 
 with gr.Blocks() as app_credits:
     gr.Markdown("""
 # Credits
 
 * [mrfakename](https://github.com/fakerybakery) for the original [online demo](https://huggingface.co/spaces/mrfakename/E2-F5-TTS)
-* [RootingInLoad](https://github.com/RootingInLoad) for initial chunk generation and podcast app exploration
-* [jpgallegoar](https://github.com/jpgallegoar) for multiple speech-type generation & voice chat
+* [RootingInLoad](https://github.com/RootingInLoad) for the podcast generation
+* [jpgallegoar](https://github.com/jpgallegoar) for multiple speech-type generation
 """)
 with gr.Blocks() as app_tts:
     gr.Markdown("# Batched TTS")
     ref_audio_input = gr.Audio(label="Reference Audio", type="filepath")
     gen_text_input = gr.Textbox(label="Text to Generate", lines=10)
+    model_choice = gr.Radio(
+        choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS"
+    )
     generate_btn = gr.Button("Synthesize", variant="primary")
     with gr.Accordion("Advanced Settings", open=False):
         ref_text_input = gr.Textbox(
@@ -204,18 +414,10 @@ with gr.Blocks() as app_tts:
             label="Speed",
             minimum=0.3,
             maximum=2.0,
-            value=1.0,
+            value=speed,
             step=0.1,
             info="Adjust the speed of the audio.",
         )
-        nfe_slider = gr.Slider(
-            label="NFE Steps",
-            minimum=4,
-            maximum=64,
-            value=32,
-            step=2,
-            info="Set the number of denoising steps.",
-        )
         cross_fade_duration_slider = gr.Slider(
             label="Cross-Fade Duration (s)",
             minimum=0.0,
@@ -224,288 +426,301 @@ with gr.Blocks() as app_tts:
             step=0.01,
             info="Set the duration of the cross-fade between audio clips.",
         )
+    speed_slider.change(update_speed, inputs=speed_slider)
 
     audio_output = gr.Audio(label="Synthesized Audio")
     spectrogram_output = gr.Image(label="Spectrogram")
 
-    @gpu_decorator
-    def basic_tts(
-        ref_audio_input,
-        ref_text_input,
-        gen_text_input,
-        remove_silence,
-        cross_fade_duration_slider,
-        nfe_slider,
-        speed_slider,
-    ):
-        audio_out, spectrogram_path, ref_text_out = infer(
-            ref_audio_input,
-            ref_text_input,
-            gen_text_input,
-            tts_model_choice,
-            remove_silence,
-            cross_fade_duration=cross_fade_duration_slider,
-            nfe_step=nfe_slider,
-            speed=speed_slider,
-        )
-        return audio_out, spectrogram_path, ref_text_out
-
     generate_btn.click(
-        basic_tts,
+        infer,
         inputs=[
             ref_audio_input,
             ref_text_input,
             gen_text_input,
+            model_choice,
             remove_silence,
             cross_fade_duration_slider,
-            nfe_slider,
-            speed_slider,
         ],
-        outputs=[audio_output, spectrogram_output, ref_text_input],
+        outputs=[audio_output, spectrogram_output],
+    )
+    
+with gr.Blocks() as app_podcast:
+    gr.Markdown("# Podcast Generation")
+    speaker1_name = gr.Textbox(label="Speaker 1 Name")
+    ref_audio_input1 = gr.Audio(label="Reference Audio (Speaker 1)", type="filepath")
+    ref_text_input1 = gr.Textbox(label="Reference Text (Speaker 1)", lines=2)
+    
+    speaker2_name = gr.Textbox(label="Speaker 2 Name")
+    ref_audio_input2 = gr.Audio(label="Reference Audio (Speaker 2)", type="filepath")
+    ref_text_input2 = gr.Textbox(label="Reference Text (Speaker 2)", lines=2)
+    
+    script_input = gr.Textbox(label="Podcast Script", lines=10, 
+                                placeholder="Enter the script with speaker names at the start of each block, e.g.:\nSean: How did you start studying...\n\nMeghan: I came to my interest in technology...\nIt was a long journey...\n\nSean: That's fascinating. Can you elaborate...")
+    
+    podcast_model_choice = gr.Radio(
+        choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS"
+    )
+    podcast_remove_silence = gr.Checkbox(
+        label="Remove Silences",
+        value=True,
     )
+    generate_podcast_btn = gr.Button("Generate Podcast", variant="primary")
+    podcast_output = gr.Audio(label="Generated Podcast")
 
+    def podcast_generation(script, speaker1, ref_audio1, ref_text1, speaker2, ref_audio2, ref_text2, model, remove_silence):
+        return generate_podcast(script, speaker1, ref_audio1, ref_text1, speaker2, ref_audio2, ref_text2, model, remove_silence)
 
-def parse_speechtypes_text(gen_text):
-    # Pattern to find {speechtype}
-    pattern = r"\{(.*?)\}"
+    generate_podcast_btn.click(
+        podcast_generation,
+        inputs=[
+            script_input,
+            speaker1_name,
+            ref_audio_input1,
+            ref_text_input1,
+            speaker2_name,
+            ref_audio_input2,
+            ref_text_input2,
+            podcast_model_choice,
+            podcast_remove_silence,
+        ],
+        outputs=podcast_output,
+    )
+
+def parse_emotional_text(gen_text):
+    # Pattern to find (Emotion)
+    pattern = r'\((.*?)\)'
 
     # Split the text by the pattern
     tokens = re.split(pattern, gen_text)
 
     segments = []
 
-    current_style = "Regular"
+    current_emotion = 'Regular'
 
     for i in range(len(tokens)):
         if i % 2 == 0:
             # This is text
             text = tokens[i].strip()
             if text:
-                segments.append({"style": current_style, "text": text})
+                segments.append({'emotion': current_emotion, 'text': text})
         else:
-            # This is style
-            style = tokens[i].strip()
-            current_style = style
+            # This is emotion
+            emotion = tokens[i].strip()
+            current_emotion = emotion
 
     return segments
 
-
-with gr.Blocks() as app_multistyle:
-    # New section for multistyle generation
+with gr.Blocks() as app_emotional:
+    # New section for emotional generation
     gr.Markdown(
         """
     # Multiple Speech-Type Generation
 
-    This section allows you to generate multiple speech types or multiple people's voices. Enter your text in the format shown below, and the system will generate speech using the appropriate type. If unspecified, the model will use the regular speech type. The current speech type will be used until the next speech type is specified.
-    """
-    )
-
-    with gr.Row():
-        gr.Markdown(
-            """
-            **Example Input:**                                                                      
-            {Regular} Hello, I'd like to order a sandwich please.                                                         
-            {Surprised} What do you mean you're out of bread?                                                                      
-            {Sad} I really wanted a sandwich though...                                                              
-            {Angry} You know what, darn you and your little shop!                                                                       
-            {Whisper} I'll just go back home and cry now.                                                                           
-            {Shouting} Why me?!                                                                         
-            """
-        )
+    This section allows you to upload different audio clips for each speech type. 'Regular' emotion is mandatory. You can add additional speech types by clicking the "Add Speech Type" button. Enter your text in the format shown below, and the system will generate speech using the appropriate emotions. If unspecified, the model will use the regular speech type. The current speech type will be used until the next speech type is specified.
 
-        gr.Markdown(
-            """
-            **Example Input 2:**                                                                                
-            {Speaker1_Happy} Hello, I'd like to order a sandwich please.                                                            
-            {Speaker2_Regular} Sorry, we're out of bread.                                                                                
-            {Speaker1_Sad} I really wanted a sandwich though...                                                                             
-            {Speaker2_Whisper} I'll give you the last one I was hiding.                                                                     
-            """
-        )
+    **Example Input:**
 
-    gr.Markdown(
-        "Upload different audio clips for each speech type. The first speech type is mandatory. You can add additional speech types by clicking the 'Add Speech Type' button."
+    (Regular) Hello, I'd like to order a sandwich please. (Surprised) What do you mean you're out of bread? (Sad) I really wanted a sandwich though... (Angry) You know what, darn you and your little shop, you suck! (Whisper) I'll just go back home and cry now. (Shouting) Why me?!
+    """
     )
 
+    gr.Markdown("Upload different audio clips for each speech type. 'Regular' emotion is mandatory. You can add additional speech types by clicking the 'Add Speech Type' button.")
+
     # Regular speech type (mandatory)
-    with gr.Row() as regular_row:
-        with gr.Column():
-            regular_name = gr.Textbox(value="Regular", label="Speech Type Name")
-            regular_insert = gr.Button("Insert Label", variant="secondary")
-        regular_audio = gr.Audio(label="Regular Reference Audio", type="filepath")
-        regular_ref_text = gr.Textbox(label="Reference Text (Regular)", lines=2)
-
-    # Regular speech type (max 100)
+    with gr.Row():
+        regular_name = gr.Textbox(value='Regular', label='Speech Type Name', interactive=False)
+        regular_audio = gr.Audio(label='Regular Reference Audio', type='filepath')
+        regular_ref_text = gr.Textbox(label='Reference Text (Regular)', lines=2)
+
+    # Additional speech types (up to 99 more)
     max_speech_types = 100
-    speech_type_rows = [regular_row]
-    speech_type_names = [regular_name]
-    speech_type_audios = [regular_audio]
-    speech_type_ref_texts = [regular_ref_text]
-    speech_type_delete_btns = [None]
-    speech_type_insert_btns = [regular_insert]
-
-    # Additional speech types (99 more)
+    speech_type_names = []
+    speech_type_audios = []
+    speech_type_ref_texts = []
+    speech_type_delete_btns = []
+
     for i in range(max_speech_types - 1):
-        with gr.Row(visible=False) as row:
-            with gr.Column():
-                name_input = gr.Textbox(label="Speech Type Name")
-                delete_btn = gr.Button("Delete Type", variant="secondary")
-                insert_btn = gr.Button("Insert Label", variant="secondary")
-            audio_input = gr.Audio(label="Reference Audio", type="filepath")
-            ref_text_input = gr.Textbox(label="Reference Text", lines=2)
-        speech_type_rows.append(row)
+        with gr.Row():
+            name_input = gr.Textbox(label='Speech Type Name', visible=False)
+            audio_input = gr.Audio(label='Reference Audio', type='filepath', visible=False)
+            ref_text_input = gr.Textbox(label='Reference Text', lines=2, visible=False)
+            delete_btn = gr.Button("Delete", variant="secondary", visible=False)
         speech_type_names.append(name_input)
         speech_type_audios.append(audio_input)
         speech_type_ref_texts.append(ref_text_input)
         speech_type_delete_btns.append(delete_btn)
-        speech_type_insert_btns.append(insert_btn)
 
     # Button to add speech type
     add_speech_type_btn = gr.Button("Add Speech Type")
 
-    # Keep track of autoincrement of speech types, no roll back
-    speech_type_count = 1
+    # Keep track of current number of speech types
+    speech_type_count = gr.State(value=0)
 
     # Function to add a speech type
-    def add_speech_type_fn():
-        row_updates = [gr.update() for _ in range(max_speech_types)]
-        global speech_type_count
-        if speech_type_count < max_speech_types:
-            row_updates[speech_type_count] = gr.update(visible=True)
+    def add_speech_type_fn(speech_type_count):
+        if speech_type_count < max_speech_types - 1:
             speech_type_count += 1
+            # Prepare updates for the components
+            name_updates = []
+            audio_updates = []
+            ref_text_updates = []
+            delete_btn_updates = []
+            for i in range(max_speech_types - 1):
+                if i < speech_type_count:
+                    name_updates.append(gr.update(visible=True))
+                    audio_updates.append(gr.update(visible=True))
+                    ref_text_updates.append(gr.update(visible=True))
+                    delete_btn_updates.append(gr.update(visible=True))
+                else:
+                    name_updates.append(gr.update())
+                    audio_updates.append(gr.update())
+                    ref_text_updates.append(gr.update())
+                    delete_btn_updates.append(gr.update())
         else:
-            gr.Warning("Exhausted maximum number of speech types. Consider restart the app.")
-        return row_updates
-
-    add_speech_type_btn.click(add_speech_type_fn, outputs=speech_type_rows)
+            # Optionally, show a warning
+            # gr.Warning("Maximum number of speech types reached.")
+            name_updates = [gr.update() for _ in range(max_speech_types - 1)]
+            audio_updates = [gr.update() for _ in range(max_speech_types - 1)]
+            ref_text_updates = [gr.update() for _ in range(max_speech_types - 1)]
+            delete_btn_updates = [gr.update() for _ in range(max_speech_types - 1)]
+        return [speech_type_count] + name_updates + audio_updates + ref_text_updates + delete_btn_updates
+
+    add_speech_type_btn.click(
+        add_speech_type_fn,
+        inputs=speech_type_count,
+        outputs=[speech_type_count] + speech_type_names + speech_type_audios + speech_type_ref_texts + speech_type_delete_btns
+    )
 
     # Function to delete a speech type
-    def delete_speech_type_fn():
-        return gr.update(visible=False), None, None, None
-
-    # Update delete button clicks
-    for i in range(1, len(speech_type_delete_btns)):
-        speech_type_delete_btns[i].click(
-            delete_speech_type_fn,
-            outputs=[speech_type_rows[i], speech_type_names[i], speech_type_audios[i], speech_type_ref_texts[i]],
+    def make_delete_speech_type_fn(index):
+        def delete_speech_type_fn(speech_type_count):
+            # Prepare updates
+            name_updates = []
+            audio_updates = []
+            ref_text_updates = []
+            delete_btn_updates = []
+
+            for i in range(max_speech_types - 1):
+                if i == index:
+                    name_updates.append(gr.update(visible=False, value=''))
+                    audio_updates.append(gr.update(visible=False, value=None))
+                    ref_text_updates.append(gr.update(visible=False, value=''))
+                    delete_btn_updates.append(gr.update(visible=False))
+                else:
+                    name_updates.append(gr.update())
+                    audio_updates.append(gr.update())
+                    ref_text_updates.append(gr.update())
+                    delete_btn_updates.append(gr.update())
+
+            speech_type_count = max(0, speech_type_count - 1)
+
+            return [speech_type_count] + name_updates + audio_updates + ref_text_updates + delete_btn_updates
+
+        return delete_speech_type_fn
+
+    for i, delete_btn in enumerate(speech_type_delete_btns):
+        delete_fn = make_delete_speech_type_fn(i)
+        delete_btn.click(
+            delete_fn,
+            inputs=speech_type_count,
+            outputs=[speech_type_count] + speech_type_names + speech_type_audios + speech_type_ref_texts + speech_type_delete_btns
         )
 
     # Text input for the prompt
-    gen_text_input_multistyle = gr.Textbox(
-        label="Text to Generate",
-        lines=10,
-        placeholder="Enter the script with speaker names (or emotion types) at the start of each block, e.g.:\n\n{Regular} Hello, I'd like to order a sandwich please.\n{Surprised} What do you mean you're out of bread?\n{Sad} I really wanted a sandwich though...\n{Angry} You know what, darn you and your little shop!\n{Whisper} I'll just go back home and cry now.\n{Shouting} Why me?!",
-    )
+    gen_text_input_emotional = gr.Textbox(label="Text to Generate", lines=10)
 
-    def make_insert_speech_type_fn(index):
-        def insert_speech_type_fn(current_text, speech_type_name):
-            current_text = current_text or ""
-            speech_type_name = speech_type_name or "None"
-            updated_text = current_text + f"{{{speech_type_name}}} "
-            return updated_text
-
-        return insert_speech_type_fn
-
-    for i, insert_btn in enumerate(speech_type_insert_btns):
-        insert_fn = make_insert_speech_type_fn(i)
-        insert_btn.click(
-            insert_fn,
-            inputs=[gen_text_input_multistyle, speech_type_names[i]],
-            outputs=gen_text_input_multistyle,
-        )
+    # Model choice
+    model_choice_emotional = gr.Radio(
+        choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS"
+    )
 
     with gr.Accordion("Advanced Settings", open=False):
-        remove_silence_multistyle = gr.Checkbox(
+        remove_silence_emotional = gr.Checkbox(
             label="Remove Silences",
             value=True,
         )
 
     # Generate button
-    generate_multistyle_btn = gr.Button("Generate Multi-Style Speech", variant="primary")
+    generate_emotional_btn = gr.Button("Generate Emotional Speech", variant="primary")
 
     # Output audio
-    audio_output_multistyle = gr.Audio(label="Synthesized Audio")
-
+    audio_output_emotional = gr.Audio(label="Synthesized Audio")
     @gpu_decorator
-    def generate_multistyle_speech(
+    def generate_emotional_speech(
+        regular_audio,
+        regular_ref_text,
         gen_text,
         *args,
     ):
-        speech_type_names_list = args[:max_speech_types]
-        speech_type_audios_list = args[max_speech_types : 2 * max_speech_types]
-        speech_type_ref_texts_list = args[2 * max_speech_types : 3 * max_speech_types]
-        remove_silence = args[3 * max_speech_types]
+        num_additional_speech_types = max_speech_types - 1
+        speech_type_names_list = args[:num_additional_speech_types]
+        speech_type_audios_list = args[num_additional_speech_types:2 * num_additional_speech_types]
+        speech_type_ref_texts_list = args[2 * num_additional_speech_types:3 * num_additional_speech_types]
+        model_choice = args[3 * num_additional_speech_types]
+        remove_silence = args[3 * num_additional_speech_types + 1]
+
         # Collect the speech types and their audios into a dict
-        speech_types = OrderedDict()
+        speech_types = {'Regular': {'audio': regular_audio, 'ref_text': regular_ref_text}}
 
-        ref_text_idx = 0
-        for name_input, audio_input, ref_text_input in zip(
-            speech_type_names_list, speech_type_audios_list, speech_type_ref_texts_list
-        ):
+        for name_input, audio_input, ref_text_input in zip(speech_type_names_list, speech_type_audios_list, speech_type_ref_texts_list):
             if name_input and audio_input:
-                speech_types[name_input] = {"audio": audio_input, "ref_text": ref_text_input}
-            else:
-                speech_types[f"@{ref_text_idx}@"] = {"audio": "", "ref_text": ""}
-            ref_text_idx += 1
+                speech_types[name_input] = {'audio': audio_input, 'ref_text': ref_text_input}
 
         # Parse the gen_text into segments
         segments = parse_speechtypes_text(gen_text)
 
         # For each segment, generate speech
         generated_audio_segments = []
-        current_style = "Regular"
+        current_emotion = 'Regular'
 
         for segment in segments:
-            style = segment["style"]
-            text = segment["text"]
+            emotion = segment['emotion']
+            text = segment['text']
 
-            if style in speech_types:
-                current_style = style
+            if emotion in speech_types:
+                current_emotion = emotion
             else:
-                gr.Warning(f"Type {style} is not available, will use Regular as default.")
-                current_style = "Regular"
+                # If emotion not available, default to Regular
+                current_emotion = 'Regular'
 
-            try:
-                ref_audio = speech_types[current_style]["audio"]
-            except KeyError:
-                gr.Warning(f"Please provide reference audio for type {current_style}.")
-                return [None] + [speech_types[style]["ref_text"] for style in speech_types]
-            ref_text = speech_types[current_style].get("ref_text", "")
+            ref_audio = speech_types[current_emotion]['audio']
+            ref_text = speech_types[current_emotion].get('ref_text', '')
 
             # Generate speech for this segment
-            audio_out, _, ref_text_out = infer(
-                ref_audio, ref_text, text, tts_model_choice, remove_silence, 0, show_info=print
-            )  # show_info=print no pull to top when generating
-            sr, audio_data = audio_out
+            audio, _ = infer(ref_audio, ref_text, text, model_choice, remove_silence, 0)
+            sr, audio_data = audio
 
             generated_audio_segments.append(audio_data)
-            speech_types[current_style]["ref_text"] = ref_text_out
 
         # Concatenate all audio segments
         if generated_audio_segments:
             final_audio_data = np.concatenate(generated_audio_segments)
-            return [(sr, final_audio_data)] + [speech_types[style]["ref_text"] for style in speech_types]
+            return (sr, final_audio_data)
         else:
             gr.Warning("No audio generated.")
-            return [None] + [speech_types[style]["ref_text"] for style in speech_types]
+            return None
 
-    generate_multistyle_btn.click(
-        generate_multistyle_speech,
+    generate_emotional_btn.click(
+        generate_emotional_speech,
         inputs=[
-            gen_text_input_multistyle,
-        ]
-        + speech_type_names
-        + speech_type_audios
-        + speech_type_ref_texts
-        + [
-            remove_silence_multistyle,
+            regular_audio,
+            regular_ref_text,
+            gen_text_input_emotional,
+        ] + speech_type_names + speech_type_audios + speech_type_ref_texts + [
+            model_choice_emotional,
+            remove_silence_emotional,
         ],
-        outputs=[audio_output_multistyle] + speech_type_ref_texts,
+        outputs=audio_output_emotional,
     )
 
     # Validation function to disable Generate button if speech types are missing
-    def validate_speech_types(gen_text, regular_name, *args):
-        speech_type_names_list = args
+    def validate_speech_types(
+        gen_text,
+        regular_name,
+        *args
+    ):
+        num_additional_speech_types = max_speech_types - 1
+        speech_type_names_list = args[:num_additional_speech_types]
 
         # Collect the speech types names
         speech_types_available = set()
@@ -516,8 +731,8 @@ with gr.Blocks() as app_multistyle:
                 speech_types_available.add(name_input)
 
         # Parse the gen_text to get the speech types used
-        segments = parse_speechtypes_text(gen_text)
-        speech_types_in_text = set(segment["style"] for segment in segments)
+        segments = parse_emotional_text(gen_text)
+        speech_types_in_text = set(segment['emotion'] for segment in segments)
 
         # Check if all speech types in text are available
         missing_speech_types = speech_types_in_text - speech_types_available
@@ -529,221 +744,11 @@ with gr.Blocks() as app_multistyle:
             # Enable the generate button
             return gr.update(interactive=True)
 
-    gen_text_input_multistyle.change(
+    gen_text_input_emotional.change(
         validate_speech_types,
-        inputs=[gen_text_input_multistyle, regular_name] + speech_type_names,
-        outputs=generate_multistyle_btn,
+        inputs=[gen_text_input_emotional, regular_name] + speech_type_names,
+        outputs=generate_emotional_btn
     )
-
-
-with gr.Blocks() as app_chat:
-    gr.Markdown(
-        """
-# Voice Chat
-Have a conversation with an AI using your reference voice! 
-1. Upload a reference audio clip and optionally its transcript.
-2. Load the chat model.
-3. Record your message through your microphone.
-4. The AI will respond using the reference voice.
-"""
-    )
-
-    if not USING_SPACES:
-        load_chat_model_btn = gr.Button("Load Chat Model", variant="primary")
-
-        chat_interface_container = gr.Column(visible=False)
-
-        @gpu_decorator
-        def load_chat_model():
-            global chat_model_state, chat_tokenizer_state
-            if chat_model_state is None:
-                show_info = gr.Info
-                show_info("Loading chat model...")
-                model_name = "Qwen/Qwen2.5-3B-Instruct"
-                chat_model_state = AutoModelForCausalLM.from_pretrained(
-                    model_name, torch_dtype="auto", device_map="auto"
-                )
-                chat_tokenizer_state = AutoTokenizer.from_pretrained(model_name)
-                show_info("Chat model loaded.")
-
-            return gr.update(visible=False), gr.update(visible=True)
-
-        load_chat_model_btn.click(load_chat_model, outputs=[load_chat_model_btn, chat_interface_container])
-
-    else:
-        chat_interface_container = gr.Column()
-
-        if chat_model_state is None:
-            model_name = "Qwen/Qwen2.5-3B-Instruct"
-            chat_model_state = AutoModelForCausalLM.from_pretrained(model_name, torch_dtype="auto", device_map="auto")
-            chat_tokenizer_state = AutoTokenizer.from_pretrained(model_name)
-
-    with chat_interface_container:
-        with gr.Row():
-            with gr.Column():
-                ref_audio_chat = gr.Audio(label="Reference Audio", type="filepath")
-            with gr.Column():
-                with gr.Accordion("Advanced Settings", open=False):
-                    remove_silence_chat = gr.Checkbox(
-                        label="Remove Silences",
-                        value=True,
-                    )
-                    ref_text_chat = gr.Textbox(
-                        label="Reference Text",
-                        info="Optional: Leave blank to auto-transcribe",
-                        lines=2,
-                    )
-                    system_prompt_chat = gr.Textbox(
-                        label="System Prompt",
-                        value="You are not an AI assistant, you are whoever the user says you are. You must stay in character. Keep your responses concise since they will be spoken out loud.",
-                        lines=2,
-                    )
-
-        chatbot_interface = gr.Chatbot(label="Conversation")
-
-        with gr.Row():
-            with gr.Column():
-                audio_input_chat = gr.Microphone(
-                    label="Speak your message",
-                    type="filepath",
-                )
-                audio_output_chat = gr.Audio(autoplay=True)
-            with gr.Column():
-                text_input_chat = gr.Textbox(
-                    label="Type your message",
-                    lines=1,
-                )
-                send_btn_chat = gr.Button("Send Message")
-                clear_btn_chat = gr.Button("Clear Conversation")
-
-        conversation_state = gr.State(
-            value=[
-                {
-                    "role": "system",
-                    "content": "You are not an AI assistant, you are whoever the user says you are. You must stay in character. Keep your responses concise since they will be spoken out loud.",
-                }
-            ]
-        )
-
-        # Modify process_audio_input to use model and tokenizer from state
-        @gpu_decorator
-        def process_audio_input(audio_path, text, history, conv_state):
-            """Handle audio or text input from user"""
-
-            if not audio_path and not text.strip():
-                return history, conv_state, ""
-
-            if audio_path:
-                text = preprocess_ref_audio_text(audio_path, text)[1]
-
-            if not text.strip():
-                return history, conv_state, ""
-
-            conv_state.append({"role": "user", "content": text})
-            history.append((text, None))
-
-            response = generate_response(conv_state, chat_model_state, chat_tokenizer_state)
-
-            conv_state.append({"role": "assistant", "content": response})
-            history[-1] = (text, response)
-
-            return history, conv_state, ""
-
-        @gpu_decorator
-        def generate_audio_response(history, ref_audio, ref_text, remove_silence):
-            """Generate TTS audio for AI response"""
-            if not history or not ref_audio:
-                return None
-
-            last_user_message, last_ai_response = history[-1]
-            if not last_ai_response:
-                return None
-
-            audio_result, _, ref_text_out = infer(
-                ref_audio,
-                ref_text,
-                last_ai_response,
-                tts_model_choice,
-                remove_silence,
-                cross_fade_duration=0.15,
-                speed=1.0,
-                show_info=print,  # show_info=print no pull to top when generating
-            )
-            return audio_result, ref_text_out
-
-        def clear_conversation():
-            """Reset the conversation"""
-            return [], [
-                {
-                    "role": "system",
-                    "content": "You are not an AI assistant, you are whoever the user says you are. You must stay in character. Keep your responses concise since they will be spoken out loud.",
-                }
-            ]
-
-        def update_system_prompt(new_prompt):
-            """Update the system prompt and reset the conversation"""
-            new_conv_state = [{"role": "system", "content": new_prompt}]
-            return [], new_conv_state
-
-        # Handle audio input
-        audio_input_chat.stop_recording(
-            process_audio_input,
-            inputs=[audio_input_chat, text_input_chat, chatbot_interface, conversation_state],
-            outputs=[chatbot_interface, conversation_state],
-        ).then(
-            generate_audio_response,
-            inputs=[chatbot_interface, ref_audio_chat, ref_text_chat, remove_silence_chat],
-            outputs=[audio_output_chat, ref_text_chat],
-        ).then(
-            lambda: None,
-            None,
-            audio_input_chat,
-        )
-
-        # Handle text input
-        text_input_chat.submit(
-            process_audio_input,
-            inputs=[audio_input_chat, text_input_chat, chatbot_interface, conversation_state],
-            outputs=[chatbot_interface, conversation_state],
-        ).then(
-            generate_audio_response,
-            inputs=[chatbot_interface, ref_audio_chat, ref_text_chat, remove_silence_chat],
-            outputs=[audio_output_chat, ref_text_chat],
-        ).then(
-            lambda: None,
-            None,
-            text_input_chat,
-        )
-
-        # Handle send button
-        send_btn_chat.click(
-            process_audio_input,
-            inputs=[audio_input_chat, text_input_chat, chatbot_interface, conversation_state],
-            outputs=[chatbot_interface, conversation_state],
-        ).then(
-            generate_audio_response,
-            inputs=[chatbot_interface, ref_audio_chat, ref_text_chat, remove_silence_chat],
-            outputs=[audio_output_chat, ref_text_chat],
-        ).then(
-            lambda: None,
-            None,
-            text_input_chat,
-        )
-
-        # Handle clear button
-        clear_btn_chat.click(
-            clear_conversation,
-            outputs=[chatbot_interface, conversation_state],
-        )
-
-        # Handle system prompt change and reset conversation
-        system_prompt_chat.change(
-            update_system_prompt,
-            inputs=system_prompt_chat,
-            outputs=[chatbot_interface, conversation_state],
-        )
-
-
 with gr.Blocks() as app:
     gr.Markdown(
         """
@@ -754,108 +759,14 @@ This is a local web UI for F5 TTS with advanced batch processing support. This a
 * [F5-TTS](https://arxiv.org/abs/2410.06885) (A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching)
 * [E2 TTS](https://arxiv.org/abs/2406.18009) (Embarrassingly Easy Fully Non-Autoregressive Zero-Shot TTS)
 
-The checkpoints currently support English and Chinese.
+The checkpoints support English and Chinese.
 
-If you're having issues, try converting your reference audio to WAV or MP3, clipping it to 15s with  ✂  in the bottom right corner (otherwise might have non-optimal auto-trimmed result).
+If you're having issues, try converting your reference audio to WAV or MP3, clipping it to 15s, and shortening your prompt.
 
 **NOTE: Reference text will be automatically transcribed with Whisper if not provided. For best results, keep your reference clips short (<15s). Ensure the audio is fully uploaded before generating.**
 """
     )
-
-    last_used_custom = files("f5_tts").joinpath("infer/.cache/last_used_custom_model_info.txt")
-
-    def load_last_used_custom():
-        try:
-            custom = []
-            with open(last_used_custom, "r", encoding="utf-8") as f:
-                for line in f:
-                    custom.append(line.strip())
-            return custom
-        except FileNotFoundError:
-            last_used_custom.parent.mkdir(parents=True, exist_ok=True)
-            return DEFAULT_TTS_MODEL_CFG
-
-    def switch_tts_model(new_choice):
-        global tts_model_choice
-        if new_choice == "Custom":  # override in case webpage is refreshed
-            custom_ckpt_path, custom_vocab_path, custom_model_cfg = load_last_used_custom()
-            tts_model_choice = ["Custom", custom_ckpt_path, custom_vocab_path, json.loads(custom_model_cfg)]
-            return (
-                gr.update(visible=True, value=custom_ckpt_path),
-                gr.update(visible=True, value=custom_vocab_path),
-                gr.update(visible=True, value=custom_model_cfg),
-            )
-        else:
-            tts_model_choice = new_choice
-            return gr.update(visible=False), gr.update(visible=False), gr.update(visible=False)
-
-    def set_custom_model(custom_ckpt_path, custom_vocab_path, custom_model_cfg):
-        global tts_model_choice
-        tts_model_choice = ["Custom", custom_ckpt_path, custom_vocab_path, json.loads(custom_model_cfg)]
-        with open(last_used_custom, "w", encoding="utf-8") as f:
-            f.write(custom_ckpt_path + "\n" + custom_vocab_path + "\n" + custom_model_cfg + "\n")
-
-    with gr.Row():
-        if not USING_SPACES:
-            choose_tts_model = gr.Radio(
-                choices=[DEFAULT_TTS_MODEL, "E2-TTS", "Custom"], label="Choose TTS Model", value=DEFAULT_TTS_MODEL
-            )
-        else:
-            choose_tts_model = gr.Radio(
-                choices=[DEFAULT_TTS_MODEL, "E2-TTS"], label="Choose TTS Model", value=DEFAULT_TTS_MODEL
-            )
-        custom_ckpt_path = gr.Dropdown(
-            choices=[DEFAULT_TTS_MODEL_CFG[0]],
-            value=load_last_used_custom()[0],
-            allow_custom_value=True,
-            label="Model: local_path | hf://user_id/repo_id/model_ckpt",
-            visible=False,
-        )
-        custom_vocab_path = gr.Dropdown(
-            choices=[DEFAULT_TTS_MODEL_CFG[1]],
-            value=load_last_used_custom()[1],
-            allow_custom_value=True,
-            label="Vocab: local_path | hf://user_id/repo_id/vocab_file",
-            visible=False,
-        )
-        custom_model_cfg = gr.Dropdown(
-            choices=[
-                DEFAULT_TTS_MODEL_CFG[2],
-                json.dumps(dict(dim=768, depth=18, heads=12, ff_mult=2, text_dim=512, conv_layers=4)),
-            ],
-            value=load_last_used_custom()[2],
-            allow_custom_value=True,
-            label="Config: in a dictionary form",
-            visible=False,
-        )
-
-    choose_tts_model.change(
-        switch_tts_model,
-        inputs=[choose_tts_model],
-        outputs=[custom_ckpt_path, custom_vocab_path, custom_model_cfg],
-        show_progress="hidden",
-    )
-    custom_ckpt_path.change(
-        set_custom_model,
-        inputs=[custom_ckpt_path, custom_vocab_path, custom_model_cfg],
-        show_progress="hidden",
-    )
-    custom_vocab_path.change(
-        set_custom_model,
-        inputs=[custom_ckpt_path, custom_vocab_path, custom_model_cfg],
-        show_progress="hidden",
-    )
-    custom_model_cfg.change(
-        set_custom_model,
-        inputs=[custom_ckpt_path, custom_vocab_path, custom_model_cfg],
-        show_progress="hidden",
-    )
-
-    gr.TabbedInterface(
-        [app_tts, app_multistyle, app_chat, app_credits],
-        ["Basic-TTS", "Multi-Speech", "Voice-Chat", "Credits"],
-    )
-
+    gr.TabbedInterface([app_tts, app_podcast, app_emotional, app_credits], ["TTS", "Podcast", "Multi-Style", "Credits"])
 
 @click.command()
 @click.option("--port", "-p", default=None, type=int, help="Port to run the app on")
@@ -868,21 +779,13 @@ If you're having issues, try converting your reference audio to WAV or MP3, clip
     help="Share the app via Gradio share link",
 )
 @click.option("--api", "-a", default=True, is_flag=True, help="Allow API access")
-@click.option(
-    "--root_path",
-    "-r",
-    default=None,
-    type=str,
-    help='The root path (or "mount point") of the application, if it\'s not served from the root ("/") of the domain. Often used when the application is behind a reverse proxy that forwards requests to the application, e.g. set "/myapp" or full URL for application served at "https://example.com/myapp".',
-)
-def main(port, host, share, api, root_path):
+def main(port, host, share, api):
     global app
-    print("Starting app...")
-    app.queue(api_open=api).launch(server_name=host, server_port=port, share=share, show_api=api, root_path=root_path)
+    print(f"Starting app...")
+    app.queue(api_open=api).launch(
+        server_name=host, server_port=port, share=share, show_api=api
+    )
 
 
-if __name__ == "__main__":
-    if not USING_SPACES:
-        main()
-    else:
-        app.queue().launch()
+
+app.queue().launch()

app_local.py

@@ -0,0 +1,236 @@
+print("WARNING: You are running this unofficial E2/F5 TTS demo locally, it may not be as up-to-date as the hosted version (https://huggingface.co/spaces/mrfakename/E2-F5-TTS)")
+
+import os
+import re
+import torch
+import torchaudio
+import gradio as gr
+import numpy as np
+import tempfile
+from einops import rearrange
+from ema_pytorch import EMA
+from vocos import Vocos
+from pydub import AudioSegment, silence
+from model import CFM, UNetT, DiT, MMDiT
+from cached_path import cached_path
+from model.utils import (
+    get_tokenizer, 
+    convert_char_to_pinyin, 
+    save_spectrogram,
+)
+from transformers import pipeline
+import librosa
+import soundfile as sf
+from txtsplit import txtsplit
+
+device = "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu"
+
+pipe = pipeline(
+    "automatic-speech-recognition",
+    model="openai/whisper-large-v3-turbo",
+    torch_dtype=torch.float16,
+    device=device,
+)
+
+vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
+
+# --------------------- Settings -------------------- #
+
+target_sample_rate = 24000
+n_mel_channels = 100
+hop_length = 256
+target_rms = 0.1
+nfe_step = 32  # 16, 32
+cfg_strength = 2.0
+ode_method = 'euler'
+sway_sampling_coef = -1.0
+speed = 1.0
+# fix_duration = 27  # None or float (duration in seconds)
+fix_duration = None
+
+def load_model(repo_name, exp_name, model_cls, model_cfg, ckpt_step):
+    checkpoint = torch.load(str(cached_path(f"hf://SWivid/{repo_name}/{exp_name}/model_{ckpt_step}.pt")), map_location=device)
+    vocab_char_map, vocab_size = get_tokenizer("Emilia_ZH_EN", "pinyin")
+    model = CFM(
+        transformer=model_cls(
+            **model_cfg,
+            text_num_embeds=vocab_size,
+            mel_dim=n_mel_channels
+        ),
+        mel_spec_kwargs=dict(
+            target_sample_rate=target_sample_rate,
+            n_mel_channels=n_mel_channels,
+            hop_length=hop_length,
+        ),
+        odeint_kwargs=dict(
+            method=ode_method,
+        ),
+        vocab_char_map=vocab_char_map,
+    ).to(device)
+
+    ema_model = EMA(model, include_online_model=False).to(device)
+    ema_model.load_state_dict(checkpoint['ema_model_state_dict'])
+    ema_model.copy_params_from_ema_to_model()
+
+    return model
+
+# load models
+F5TTS_model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
+E2TTS_model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
+
+F5TTS_ema_model = load_model("F5-TTS", "F5TTS_Base", DiT, F5TTS_model_cfg, 1200000)
+E2TTS_ema_model = load_model("E2-TTS", "E2TTS_Base", UNetT, E2TTS_model_cfg, 1200000)
+
+def infer(ref_audio_orig, ref_text, gen_text, exp_name, remove_silence, progress = gr.Progress()):
+    print(gen_text)
+    gr.Info("Converting audio...")
+    with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
+        aseg = AudioSegment.from_file(ref_audio_orig)
+        # remove long silence in reference audio
+        non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
+        non_silent_wave = AudioSegment.silent(duration=0)
+        for non_silent_seg in non_silent_segs:
+            non_silent_wave += non_silent_seg
+        aseg = non_silent_wave
+        # Convert to mono
+        aseg = aseg.set_channels(1)
+        audio_duration = len(aseg)
+        if audio_duration > 15000:
+            gr.Warning("Audio is over 15s, clipping to only first 15s.")
+            aseg = aseg[:15000]
+        aseg.export(f.name, format="wav")
+        ref_audio = f.name
+    if exp_name == "F5-TTS":
+        ema_model = F5TTS_ema_model
+    elif exp_name == "E2-TTS":
+        ema_model = E2TTS_ema_model
+    
+    if not ref_text.strip():
+        gr.Info("No reference text provided, transcribing reference audio...")
+        ref_text = outputs = pipe(
+            ref_audio,
+            chunk_length_s=30,
+            batch_size=128,
+            generate_kwargs={"task": "transcribe"},
+            return_timestamps=False,
+        )['text'].strip()
+        gr.Info("Finished transcription")
+    else:
+        gr.Info("Using custom reference text...")
+    audio, sr = torchaudio.load(ref_audio)
+    max_chars = int(len(ref_text) / (audio.shape[-1] / sr) * (30 - audio.shape[-1] / sr))
+    # Audio
+    if audio.shape[0] > 1:
+        audio = torch.mean(audio, dim=0, keepdim=True)
+    rms = torch.sqrt(torch.mean(torch.square(audio)))
+    if rms < target_rms:
+        audio = audio * target_rms / rms
+    if sr != target_sample_rate:
+        resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
+        audio = resampler(audio)
+    audio = audio.to(device)
+    # Chunk
+    chunks = txtsplit(gen_text, 0.7*max_chars, 0.9*max_chars) # 100 chars preferred, 150 max
+    results = []
+    generated_mel_specs = []
+    for chunk in progress.tqdm(chunks):
+        # Prepare the text
+        text_list = [ref_text + chunk]
+        final_text_list = convert_char_to_pinyin(text_list)
+    
+        # Calculate duration
+        ref_audio_len = audio.shape[-1] // hop_length
+        # if fix_duration is not None:
+        #     duration = int(fix_duration * target_sample_rate / hop_length)
+        # else:
+        zh_pause_punc = r"。，、；：？！"
+        ref_text_len = len(ref_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, ref_text))
+        chunk = len(chunk.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, gen_text))
+        duration = ref_audio_len + int(ref_audio_len / ref_text_len * chunk / speed)
+    
+        # inference
+        gr.Info(f"Generating audio using {exp_name}")
+        with torch.inference_mode():
+            generated, _ = ema_model.sample(
+                cond=audio,
+                text=final_text_list,
+                duration=duration,
+                steps=nfe_step,
+                cfg_strength=cfg_strength,
+                sway_sampling_coef=sway_sampling_coef,
+            )
+    
+        generated = generated[:, ref_audio_len:, :]
+        generated_mel_spec = rearrange(generated, '1 n d -> 1 d n')
+        gr.Info("Running vocoder")
+        generated_wave = vocos.decode(generated_mel_spec.cpu())
+        if rms < target_rms:
+            generated_wave = generated_wave * rms / target_rms
+    
+        # wav -> numpy
+        generated_wave = generated_wave.squeeze().cpu().numpy()
+        results.append(generated_wave)
+    generated_wave = np.concatenate(results)
+    if remove_silence:
+        gr.Info("Removing audio silences... This may take a moment")
+        # non_silent_intervals = librosa.effects.split(generated_wave, top_db=30)
+        # non_silent_wave = np.array([])
+        # for interval in non_silent_intervals:
+        #     start, end = interval
+        #     non_silent_wave = np.concatenate([non_silent_wave, generated_wave[start:end]])
+        # generated_wave = non_silent_wave
+        with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
+            sf.write(f.name, generated_wave, target_sample_rate)
+            aseg = AudioSegment.from_file(f.name)
+            non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
+            non_silent_wave = AudioSegment.silent(duration=0)
+            for non_silent_seg in non_silent_segs:
+                non_silent_wave += non_silent_seg
+            aseg = non_silent_wave
+            aseg.export(f.name, format="wav")
+            generated_wave, _ = torchaudio.load(f.name)
+        generated_wave = generated_wave.squeeze().cpu().numpy()
+
+    # spectogram
+    # with tempfile.NamedTemporaryFile(suffix=".png", delete=False) as tmp_spectrogram:
+    #     spectrogram_path = tmp_spectrogram.name
+    #     save_spectrogram(generated_mel_spec[0].cpu().numpy(), spectrogram_path)
+
+    return (target_sample_rate, generated_wave)
+
+with gr.Blocks() as app:
+    gr.Markdown("""
+# E2/F5 TTS
+
+This is an unofficial E2/F5 TTS demo. This demo supports the following TTS models:
+
+* [E2-TTS](https://arxiv.org/abs/2406.18009) (Embarrassingly Easy Fully Non-Autoregressive Zero-Shot TTS)
+* [F5-TTS](https://arxiv.org/abs/2410.06885) (A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching)
+
+This demo is based on the [F5-TTS](https://github.com/SWivid/F5-TTS) codebase, which is based on an [unofficial E2-TTS implementation](https://github.com/lucidrains/e2-tts-pytorch).
+
+The checkpoints support English and Chinese.
+
+If you're having issues, try converting your reference audio to WAV or MP3, clipping it to 15s, and shortening your prompt. If you're still running into issues, please open a [community Discussion](https://huggingface.co/spaces/mrfakename/E2-F5-TTS/discussions).
+
+Long-form/batched inference + speech editing is coming soon!
+
+**NOTE: Reference text will be automatically transcribed with Whisper if not provided. For best results, keep your reference clips short (<15s). Ensure the audio is fully uploaded before generating.**
+""")
+
+    ref_audio_input = gr.Audio(label="Reference Audio", type="filepath")
+    gen_text_input = gr.Textbox(label="Text to Generate (longer text will use chunking)", lines=4)
+    model_choice = gr.Radio(choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS")
+    generate_btn = gr.Button("Synthesize", variant="primary")
+    with gr.Accordion("Advanced Settings", open=False):
+        ref_text_input = gr.Textbox(label="Reference Text", info="Leave blank to automatically transcribe the reference audio. If you enter text it will override automatic transcription.", lines=2)
+        remove_silence = gr.Checkbox(label="Remove Silences", info="The model tends to produce silences, especially on longer audio. We can manually remove silences if needed. Note that this is an experimental feature and may produce strange results. This will also increase generation time.", value=True)
+    
+    audio_output = gr.Audio(label="Synthesized Audio")
+    # spectrogram_output = gr.Image(label="Spectrogram")
+
+    generate_btn.click(infer, inputs=[ref_audio_input, ref_text_input, gen_text_input, model_choice, remove_silence], outputs=[audio_output])
+    gr.Markdown("Unofficial demo by [mrfakename](https://x.com/realmrfakename)")
+    
+
+app.queue().launch()

cog.py

@@ -0,0 +1,180 @@
+# Prediction interface for Cog ⚙️
+# https://cog.run/python
+
+from cog import BasePredictor, Input, Path
+
+import os
+import re
+import torch
+import torchaudio
+import numpy as np
+import tempfile
+from einops import rearrange
+from ema_pytorch import EMA
+from vocos import Vocos
+from pydub import AudioSegment
+from model import CFM, UNetT, DiT, MMDiT
+from cached_path import cached_path
+from model.utils import (
+    get_tokenizer,
+    convert_char_to_pinyin,
+    save_spectrogram,
+)
+from transformers import pipeline
+import librosa
+
+device = "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu"
+
+target_sample_rate = 24000
+n_mel_channels = 100
+hop_length = 256
+target_rms = 0.1
+nfe_step = 32  # 16, 32
+cfg_strength = 2.0
+ode_method = 'euler'
+sway_sampling_coef = -1.0
+speed = 1.0
+# fix_duration = 27  # None or float (duration in seconds)
+fix_duration = None
+
+
+class Predictor(BasePredictor):
+    def load_model(exp_name, model_cls, model_cfg, ckpt_step):
+        checkpoint = torch.load(str(cached_path(f"hf://SWivid/F5-TTS/{exp_name}/model_{ckpt_step}.pt")), map_location=device)
+        vocab_char_map, vocab_size = get_tokenizer("Emilia_ZH_EN", "pinyin")
+        model = CFM(
+            transformer=model_cls(
+                **model_cfg,
+                text_num_embeds=vocab_size,
+                mel_dim=n_mel_channels
+            ),
+            mel_spec_kwargs=dict(
+                target_sample_rate=target_sample_rate,
+                n_mel_channels=n_mel_channels,
+                hop_length=hop_length,
+            ),
+            odeint_kwargs=dict(
+                method=ode_method,
+            ),
+            vocab_char_map=vocab_char_map,
+        ).to(device)
+
+        ema_model = EMA(model, include_online_model=False).to(device)
+        ema_model.load_state_dict(checkpoint['ema_model_state_dict'])
+        ema_model.copy_params_from_ema_to_model()
+
+        return ema_model, model
+    def setup(self) -> None:
+        """Load the model into memory to make running multiple predictions efficient"""
+        # self.model = torch.load("./weights.pth")
+        print("Loading Whisper model...")
+        self.pipe = pipeline(
+            "automatic-speech-recognition",
+            model="openai/whisper-large-v3-turbo",
+            torch_dtype=torch.float16,
+            device=device,
+        )
+        print("Loading F5-TTS model...")
+
+        F5TTS_model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
+        self.F5TTS_ema_model, self.F5TTS_base_model = self.load_model("F5TTS_Base", DiT, F5TTS_model_cfg, 1200000)
+
+
+    def predict(
+        self,
+        gen_text: str = Input(description="Text to generate"),
+        ref_audio_orig: Path = Input(description="Reference audio"),
+        remove_silence: bool = Input(description="Remove silences", default=True),
+    ) -> Path:
+        """Run a single prediction on the model"""
+        model_choice = "F5-TTS"
+        print(gen_text)
+        if len(gen_text) > 200:
+            raise gr.Error("Please keep your text under 200 chars.")
+        gr.Info("Converting audio...")
+        with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
+            aseg = AudioSegment.from_file(ref_audio_orig)
+            audio_duration = len(aseg)
+            if audio_duration > 15000:
+                gr.Warning("Audio is over 15s, clipping to only first 15s.")
+                aseg = aseg[:15000]
+            aseg.export(f.name, format="wav")
+            ref_audio = f.name
+        ema_model = self.F5TTS_ema_model
+        base_model = self.F5TTS_base_model
+
+        if not ref_text.strip():
+            gr.Info("No reference text provided, transcribing reference audio...")
+            ref_text = outputs = self.pipe(
+                ref_audio,
+                chunk_length_s=30,
+                batch_size=128,
+                generate_kwargs={"task": "transcribe"},
+                return_timestamps=False,
+            )['text'].strip()
+            gr.Info("Finished transcription")
+        else:
+            gr.Info("Using custom reference text...")
+        audio, sr = torchaudio.load(ref_audio)
+
+        rms = torch.sqrt(torch.mean(torch.square(audio)))
+        if rms < target_rms:
+            audio = audio * target_rms / rms
+        if sr != target_sample_rate:
+            resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
+            audio = resampler(audio)
+        audio = audio.to(device)
+
+        # Prepare the text
+        text_list = [ref_text + gen_text]
+        final_text_list = convert_char_to_pinyin(text_list)
+
+        # Calculate duration
+        ref_audio_len = audio.shape[-1] // hop_length
+        # if fix_duration is not None:
+        #     duration = int(fix_duration * target_sample_rate / hop_length)
+        # else:
+        zh_pause_punc = r"。，、；：？！"
+        ref_text_len = len(ref_text) + len(re.findall(zh_pause_punc, ref_text))
+        gen_text_len = len(gen_text) + len(re.findall(zh_pause_punc, gen_text))
+        duration = ref_audio_len + int(ref_audio_len / ref_text_len * gen_text_len / speed)
+
+        # inference
+        gr.Info(f"Generating audio using F5-TTS")
+        with torch.inference_mode():
+            generated, _ = base_model.sample(
+                cond=audio,
+                text=final_text_list,
+                duration=duration,
+                steps=nfe_step,
+                cfg_strength=cfg_strength,
+                sway_sampling_coef=sway_sampling_coef,
+            )
+
+        generated = generated[:, ref_audio_len:, :]
+        generated_mel_spec = rearrange(generated, '1 n d -> 1 d n')
+        gr.Info("Running vocoder")
+        vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
+        generated_wave = vocos.decode(generated_mel_spec.cpu())
+        if rms < target_rms:
+            generated_wave = generated_wave * rms / target_rms
+
+        # wav -> numpy
+        generated_wave = generated_wave.squeeze().cpu().numpy()
+
+        if remove_silence:
+            gr.Info("Removing audio silences... This may take a moment")
+            non_silent_intervals = librosa.effects.split(generated_wave, top_db=30)
+            non_silent_wave = np.array([])
+            for interval in non_silent_intervals:
+                start, end = interval
+                non_silent_wave = np.concatenate([non_silent_wave, generated_wave[start:end]])
+            generated_wave = non_silent_wave
+
+
+        # spectogram
+        with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmp_wav:
+            wav_path = tmp_wav.name
+            torchaudio.save(wav_path, torch.tensor(generated_wave), target_sample_rate)
+
+        return wav_path

data/Emilia_ZH_EN_pinyin/vocab.txt

@@ -1,2545 +1,2545 @@
- 
-!
-"
-#
-$
-%
-&
-'
-(
-)
-*
-+
-,
--
-.
-/
-0
-1
-2
-3
-4
-5
-6
-7
-8
-9
-:
-;
-=
->
-?
-@
-A
-B
-C
-D
-E
-F
-G
-H
-I
-J
-K
-L
-M
-N
-O
-P
-Q
-R
-S
-T
-U
-V
-W
-X
-Y
-Z
-[
-\
-]
-_
-a
-a1
-ai1
-ai2
-ai3
-ai4
-an1
-an3
-an4
-ang1
-ang2
-ang4
-ao1
-ao2
-ao3
-ao4
-b
-ba
-ba1
-ba2
-ba3
-ba4
-bai1
-bai2
-bai3
-bai4
-ban1
-ban2
-ban3
-ban4
-bang1
-bang2
-bang3
-bang4
-bao1
-bao2
-bao3
-bao4
-bei
-bei1
-bei2
-bei3
-bei4
-ben1
-ben2
-ben3
-ben4
-beng
-beng1
-beng2
-beng3
-beng4
-bi1
-bi2
-bi3
-bi4
-bian1
-bian2
-bian3
-bian4
-biao1
-biao2
-biao3
-bie1
-bie2
-bie3
-bie4
-bin1
-bin4
-bing1
-bing2
-bing3
-bing4
-bo
-bo1
-bo2
-bo3
-bo4
-bu2
-bu3
-bu4
-c
-ca1
-cai1
-cai2
-cai3
-cai4
-can1
-can2
-can3
-can4
-cang1
-cang2
-cao1
-cao2
-cao3
-ce4
-cen1
-cen2
-ceng1
-ceng2
-ceng4
-cha1
-cha2
-cha3
-cha4
-chai1
-chai2
-chan1
-chan2
-chan3
-chan4
-chang1
-chang2
-chang3
-chang4
-chao1
-chao2
-chao3
-che1
-che2
-che3
-che4
-chen1
-chen2
-chen3
-chen4
-cheng1
-cheng2
-cheng3
-cheng4
-chi1
-chi2
-chi3
-chi4
-chong1
-chong2
-chong3
-chong4
-chou1
-chou2
-chou3
-chou4
-chu1
-chu2
-chu3
-chu4
-chua1
-chuai1
-chuai2
-chuai3
-chuai4
-chuan1
-chuan2
-chuan3
-chuan4
-chuang1
-chuang2
-chuang3
-chuang4
-chui1
-chui2
-chun1
-chun2
-chun3
-chuo1
-chuo4
-ci1
-ci2
-ci3
-ci4
-cong1
-cong2
-cou4
-cu1
-cu4
-cuan1
-cuan2
-cuan4
-cui1
-cui3
-cui4
-cun1
-cun2
-cun4
-cuo1
-cuo2
-cuo4
-d
-da
-da1
-da2
-da3
-da4
-dai1
-dai2
-dai3
-dai4
-dan1
-dan2
-dan3
-dan4
-dang1
-dang2
-dang3
-dang4
-dao1
-dao2
-dao3
-dao4
-de
-de1
-de2
-dei3
-den4
-deng1
-deng2
-deng3
-deng4
-di1
-di2
-di3
-di4
-dia3
-dian1
-dian2
-dian3
-dian4
-diao1
-diao3
-diao4
-die1
-die2
-die4
-ding1
-ding2
-ding3
-ding4
-diu1
-dong1
-dong3
-dong4
-dou1
-dou2
-dou3
-dou4
-du1
-du2
-du3
-du4
-duan1
-duan2
-duan3
-duan4
-dui1
-dui4
-dun1
-dun3
-dun4
-duo1
-duo2
-duo3
-duo4
-e
-e1
-e2
-e3
-e4
-ei2
-en1
-en4
-er
-er2
-er3
-er4
-f
-fa1
-fa2
-fa3
-fa4
-fan1
-fan2
-fan3
-fan4
-fang1
-fang2
-fang3
-fang4
-fei1
-fei2
-fei3
-fei4
-fen1
-fen2
-fen3
-fen4
-feng1
-feng2
-feng3
-feng4
-fo2
-fou2
-fou3
-fu1
-fu2
-fu3
-fu4
-g
-ga1
-ga2
-ga3
-ga4
-gai1
-gai2
-gai3
-gai4
-gan1
-gan2
-gan3
-gan4
-gang1
-gang2
-gang3
-gang4
-gao1
-gao2
-gao3
-gao4
-ge1
-ge2
-ge3
-ge4
-gei2
-gei3
-gen1
-gen2
-gen3
-gen4
-geng1
-geng3
-geng4
-gong1
-gong3
-gong4
-gou1
-gou2
-gou3
-gou4
-gu
-gu1
-gu2
-gu3
-gu4
-gua1
-gua2
-gua3
-gua4
-guai1
-guai2
-guai3
-guai4
-guan1
-guan2
-guan3
-guan4
-guang1
-guang2
-guang3
-guang4
-gui1
-gui2
-gui3
-gui4
-gun3
-gun4
-guo1
-guo2
-guo3
-guo4
-h
-ha1
-ha2
-ha3
-hai1
-hai2
-hai3
-hai4
-han1
-han2
-han3
-han4
-hang1
-hang2
-hang4
-hao1
-hao2
-hao3
-hao4
-he1
-he2
-he4
-hei1
-hen2
-hen3
-hen4
-heng1
-heng2
-heng4
-hong1
-hong2
-hong3
-hong4
-hou1
-hou2
-hou3
-hou4
-hu1
-hu2
-hu3
-hu4
-hua1
-hua2
-hua4
-huai2
-huai4
-huan1
-huan2
-huan3
-huan4
-huang1
-huang2
-huang3
-huang4
-hui1
-hui2
-hui3
-hui4
-hun1
-hun2
-hun4
-huo
-huo1
-huo2
-huo3
-huo4
-i
-j
-ji1
-ji2
-ji3
-ji4
-jia
-jia1
-jia2
-jia3
-jia4
-jian1
-jian2
-jian3
-jian4
-jiang1
-jiang2
-jiang3
-jiang4
-jiao1
-jiao2
-jiao3
-jiao4
-jie1
-jie2
-jie3
-jie4
-jin1
-jin2
-jin3
-jin4
-jing1
-jing2
-jing3
-jing4
-jiong3
-jiu1
-jiu2
-jiu3
-jiu4
-ju1
-ju2
-ju3
-ju4
-juan1
-juan2
-juan3
-juan4
-jue1
-jue2
-jue4
-jun1
-jun4
-k
-ka1
-ka2
-ka3
-kai1
-kai2
-kai3
-kai4
-kan1
-kan2
-kan3
-kan4
-kang1
-kang2
-kang4
-kao1
-kao2
-kao3
-kao4
-ke1
-ke2
-ke3
-ke4
-ken3
-keng1
-kong1
-kong3
-kong4
-kou1
-kou2
-kou3
-kou4
-ku1
-ku2
-ku3
-ku4
-kua1
-kua3
-kua4
-kuai3
-kuai4
-kuan1
-kuan2
-kuan3
-kuang1
-kuang2
-kuang4
-kui1
-kui2
-kui3
-kui4
-kun1
-kun3
-kun4
-kuo4
-l
-la
-la1
-la2
-la3
-la4
-lai2
-lai4
-lan2
-lan3
-lan4
-lang1
-lang2
-lang3
-lang4
-lao1
-lao2
-lao3
-lao4
-le
-le1
-le4
-lei
-lei1
-lei2
-lei3
-lei4
-leng1
-leng2
-leng3
-leng4
-li
-li1
-li2
-li3
-li4
-lia3
-lian2
-lian3
-lian4
-liang2
-liang3
-liang4
-liao1
-liao2
-liao3
-liao4
-lie1
-lie2
-lie3
-lie4
-lin1
-lin2
-lin3
-lin4
-ling2
-ling3
-ling4
-liu1
-liu2
-liu3
-liu4
-long1
-long2
-long3
-long4
-lou1
-lou2
-lou3
-lou4
-lu1
-lu2
-lu3
-lu4
-luan2
-luan3
-luan4
-lun1
-lun2
-lun4
-luo1
-luo2
-luo3
-luo4
-lv2
-lv3
-lv4
-lve3
-lve4
-m
-ma
-ma1
-ma2
-ma3
-ma4
-mai2
-mai3
-mai4
-man1
-man2
-man3
-man4
-mang2
-mang3
-mao1
-mao2
-mao3
-mao4
-me
-mei2
-mei3
-mei4
-men
-men1
-men2
-men4
-meng
-meng1
-meng2
-meng3
-meng4
-mi1
-mi2
-mi3
-mi4
-mian2
-mian3
-mian4
-miao1
-miao2
-miao3
-miao4
-mie1
-mie4
-min2
-min3
-ming2
-ming3
-ming4
-miu4
-mo1
-mo2
-mo3
-mo4
-mou1
-mou2
-mou3
-mu2
-mu3
-mu4
-n
-n2
-na1
-na2
-na3
-na4
-nai2
-nai3
-nai4
-nan1
-nan2
-nan3
-nan4
-nang1
-nang2
-nang3
-nao1
-nao2
-nao3
-nao4
-ne
-ne2
-ne4
-nei3
-nei4
-nen4
-neng2
-ni1
-ni2
-ni3
-ni4
-nian1
-nian2
-nian3
-nian4
-niang2
-niang4
-niao2
-niao3
-niao4
-nie1
-nie4
-nin2
-ning2
-ning3
-ning4
-niu1
-niu2
-niu3
-niu4
-nong2
-nong4
-nou4
-nu2
-nu3
-nu4
-nuan3
-nuo2
-nuo4
-nv2
-nv3
-nve4
-o
-o1
-o2
-ou1
-ou2
-ou3
-ou4
-p
-pa1
-pa2
-pa4
-pai1
-pai2
-pai3
-pai4
-pan1
-pan2
-pan4
-pang1
-pang2
-pang4
-pao1
-pao2
-pao3
-pao4
-pei1
-pei2
-pei4
-pen1
-pen2
-pen4
-peng1
-peng2
-peng3
-peng4
-pi1
-pi2
-pi3
-pi4
-pian1
-pian2
-pian4
-piao1
-piao2
-piao3
-piao4
-pie1
-pie2
-pie3
-pin1
-pin2
-pin3
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+ﷺ
+ﷻ
+！
+，
+？
+�
+𠮶

finetune-cli.py

@@ -1,57 +1,42 @@
 import argparse
-from model import CFM, UNetT, DiT, Trainer
+from model import CFM, UNetT, DiT, MMDiT, Trainer
 from model.utils import get_tokenizer
 from model.dataset import load_dataset
 from cached_path import cached_path
-import shutil
-import os
-
+import shutil,os
 # -------------------------- Dataset Settings --------------------------- #
 target_sample_rate = 24000
 n_mel_channels = 100
 hop_length = 256
 
+tokenizer = "pinyin"  # 'pinyin', 'char', or 'custom'
+tokenizer_path = None  # if tokenizer = 'custom', define the path to the tokenizer you want to use (should be vocab.txt)
 
 # -------------------------- Argument Parsing --------------------------- #
 def parse_args():
-    parser = argparse.ArgumentParser(description="Train CFM Model")
-
-    parser.add_argument(
-        "--exp_name", type=str, default="F5TTS_Base", choices=["F5TTS_Base", "E2TTS_Base"], help="Experiment name"
-    )
-    parser.add_argument("--dataset_name", type=str, default="Emilia_ZH_EN", help="Name of the dataset to use")
-    parser.add_argument("--learning_rate", type=float, default=1e-4, help="Learning rate for training")
-    parser.add_argument("--batch_size_per_gpu", type=int, default=256, help="Batch size per GPU")
-    parser.add_argument(
-        "--batch_size_type", type=str, default="frame", choices=["frame", "sample"], help="Batch size type"
-    )
-    parser.add_argument("--max_samples", type=int, default=16, help="Max sequences per batch")
-    parser.add_argument("--grad_accumulation_steps", type=int, default=1, help="Gradient accumulation steps")
-    parser.add_argument("--max_grad_norm", type=float, default=1.0, help="Max gradient norm for clipping")
-    parser.add_argument("--epochs", type=int, default=10, help="Number of training epochs")
-    parser.add_argument("--num_warmup_updates", type=int, default=5, help="Warmup steps")
-    parser.add_argument("--save_per_updates", type=int, default=10, help="Save checkpoint every X steps")
-    parser.add_argument("--last_per_steps", type=int, default=10, help="Save last checkpoint every X steps")
-    parser.add_argument("--finetune", type=bool, default=True, help="Use Finetune")
-
-    parser.add_argument(
-        "--tokenizer", type=str, default="pinyin", choices=["pinyin", "char", "custom"], help="Tokenizer type"
-    )
-    parser.add_argument(
-        "--tokenizer_path",
-        type=str,
-        default=None,
-        help="Path to custom tokenizer vocab file (only used if tokenizer = 'custom')",
-    )
-
+    parser = argparse.ArgumentParser(description='Train CFM Model')
+
+    parser.add_argument('--exp_name', type=str, default="F5TTS_Base", choices=["F5TTS_Base", "E2TTS_Base"],help='Experiment name')
+    parser.add_argument('--dataset_name', type=str, default="Emilia_ZH_EN", help='Name of the dataset to use')
+    parser.add_argument('--learning_rate', type=float, default=1e-4, help='Learning rate for training')
+    parser.add_argument('--batch_size_per_gpu', type=int, default=256, help='Batch size per GPU')
+    parser.add_argument('--batch_size_type', type=str, default="frame", choices=["frame", "sample"],help='Batch size type')
+    parser.add_argument('--max_samples', type=int, default=16, help='Max sequences per batch')
+    parser.add_argument('--grad_accumulation_steps', type=int, default=1,help='Gradient accumulation steps')
+    parser.add_argument('--max_grad_norm', type=float, default=1.0, help='Max gradient norm for clipping')
+    parser.add_argument('--epochs', type=int, default=10, help='Number of training epochs')
+    parser.add_argument('--num_warmup_updates', type=int, default=5, help='Warmup steps')
+    parser.add_argument('--save_per_updates', type=int, default=10, help='Save checkpoint every X steps')
+    parser.add_argument('--last_per_steps', type=int, default=10, help='Save last checkpoint every X steps')
+    parser.add_argument('--finetune', type=bool, default=True, help='Use Finetune')
+    
     return parser.parse_args()
 
-
 # -------------------------- Training Settings -------------------------- #
 
-
 def main():
     args = parse_args()
+    
 
     # Model parameters based on experiment name
     if args.exp_name == "F5TTS_Base":
@@ -59,31 +44,24 @@ def main():
         model_cls = DiT
         model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
         if args.finetune:
-            ckpt_path = str(cached_path("hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.pt"))
+           ckpt_path = str(cached_path(f"hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.pt"))
     elif args.exp_name == "E2TTS_Base":
         wandb_resume_id = None
         model_cls = UNetT
         model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
         if args.finetune:
-            ckpt_path = str(cached_path("hf://SWivid/E2-TTS/E2TTS_Base/model_1200000.pt"))
-
+           ckpt_path = str(cached_path(f"hf://SWivid/E2-TTS/E2TTS_Base/model_1200000.pt"))
+           
     if args.finetune:
-        path_ckpt = os.path.join("ckpts", args.dataset_name)
-        if not os.path.isdir(path_ckpt):
-            os.makedirs(path_ckpt, exist_ok=True)
-            shutil.copy2(ckpt_path, os.path.join(path_ckpt, os.path.basename(ckpt_path)))
-
-    checkpoint_path = os.path.join("ckpts", args.dataset_name)
-
-    # Use the tokenizer and tokenizer_path provided in the command line arguments
-    tokenizer = args.tokenizer
-    if tokenizer == "custom":
-        if not args.tokenizer_path:
-            raise ValueError("Custom tokenizer selected, but no tokenizer_path provided.")
-        tokenizer_path = args.tokenizer_path
-    else:
-        tokenizer_path = args.dataset_name
-
+       path_ckpt = os.path.join("ckpts",args.dataset_name)
+       if os.path.isdir(path_ckpt)==False:
+          os.makedirs(path_ckpt,exist_ok=True)
+          shutil.copy2(ckpt_path,os.path.join(path_ckpt,os.path.basename(ckpt_path)))
+    
+    checkpoint_path=os.path.join("ckpts",args.dataset_name)
+
+    # Use the dataset_name provided in the command line
+    tokenizer_path = args.dataset_name if tokenizer != "custom" else tokenizer_path
     vocab_char_map, vocab_size = get_tokenizer(tokenizer_path, tokenizer)
 
     mel_spec_kwargs = dict(
@@ -93,7 +71,11 @@ def main():
     )
 
     e2tts = CFM(
-        transformer=model_cls(**model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels),
+        transformer=model_cls(
+            **model_cfg,
+            text_num_embeds=vocab_size,
+            mel_dim=n_mel_channels
+        ),
         mel_spec_kwargs=mel_spec_kwargs,
         vocab_char_map=vocab_char_map,
     )
@@ -117,11 +99,10 @@ def main():
     )
 
     train_dataset = load_dataset(args.dataset_name, tokenizer, mel_spec_kwargs=mel_spec_kwargs)
-    trainer.train(
-        train_dataset,
-        resumable_with_seed=666,  # seed for shuffling dataset
-    )
+    trainer.train(train_dataset,
+                  resumable_with_seed=666  # seed for shuffling dataset
+                  )
 
 
-if __name__ == "__main__":
+if __name__ == '__main__':
     main()

finetune_gradio.py

@@ -1,12 +1,8 @@
-import os
-import sys
+import os,sys
 
-import tempfile
-import random
 from transformers import pipeline
 import gradio as gr
 import torch
-import gc
 import click
 import torchaudio
 from glob import glob
@@ -23,43 +19,35 @@ import psutil
 import platform
 import subprocess
 from datasets.arrow_writer import ArrowWriter
-from datasets import Dataset as Dataset_
-from api import F5TTS
 
+import json
 
-training_process = None
+training_process = None    
 system = platform.system()
 python_executable = sys.executable or "python"
-tts_api = None
-last_checkpoint = ""
-last_device = ""
 
-path_data = "data"
+path_data="data"
 
-device = "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu"
+device = (
+"cuda"
+    if torch.cuda.is_available()
+    else "mps" if torch.backends.mps.is_available() else "cpu"
+)
 
 pipe = None
 
-
 # Load metadata
 def get_audio_duration(audio_path):
     """Calculate the duration of an audio file."""
     audio, sample_rate = torchaudio.load(audio_path)
-    num_channels = audio.shape[0]
+    num_channels = audio.shape[0]  
     return audio.shape[1] / (sample_rate * num_channels)
 
-
 def clear_text(text):
     """Clean and prepare text by lowering the case and stripping whitespace."""
     return text.lower().strip()
 
-
-def get_rms(
-    y,
-    frame_length=2048,
-    hop_length=512,
-    pad_mode="constant",
-):  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.py
+def get_rms(y,frame_length=2048,hop_length=512,pad_mode="constant",): # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.py
     padding = (int(frame_length // 2), int(frame_length // 2))
     y = np.pad(y, padding, mode=pad_mode)
 
@@ -86,8 +74,7 @@ def get_rms(
 
     return np.sqrt(power)
 
-
-class Slicer:  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.py
+class Slicer: # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.py
     def __init__(
         self,
         sr: int,
@@ -98,9 +85,13 @@ class Slicer:  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.
         max_sil_kept: int = 2000,
     ):
         if not min_length >= min_interval >= hop_size:
-            raise ValueError("The following condition must be satisfied: min_length >= min_interval >= hop_size")
+            raise ValueError(
+                "The following condition must be satisfied: min_length >= min_interval >= hop_size"
+            )
         if not max_sil_kept >= hop_size:
-            raise ValueError("The following condition must be satisfied: max_sil_kept >= hop_size")
+            raise ValueError(
+                "The following condition must be satisfied: max_sil_kept >= hop_size"
+            )
         min_interval = sr * min_interval / 1000
         self.threshold = 10 ** (threshold / 20.0)
         self.hop_size = round(sr * hop_size / 1000)
@@ -111,9 +102,13 @@ class Slicer:  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.
 
     def _apply_slice(self, waveform, begin, end):
         if len(waveform.shape) > 1:
-            return waveform[:, begin * self.hop_size : min(waveform.shape[1], end * self.hop_size)]
+            return waveform[
+                :, begin * self.hop_size : min(waveform.shape[1], end * self.hop_size)
+            ]
         else:
-            return waveform[begin * self.hop_size : min(waveform.shape[0], end * self.hop_size)]
+            return waveform[
+                begin * self.hop_size : min(waveform.shape[0], end * self.hop_size)
+            ]
 
     # @timeit
     def slice(self, waveform):
@@ -123,7 +118,9 @@ class Slicer:  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.
             samples = waveform
         if samples.shape[0] <= self.min_length:
             return [waveform]
-        rms_list = get_rms(y=samples, frame_length=self.win_size, hop_length=self.hop_size).squeeze(0)
+        rms_list = get_rms(
+            y=samples, frame_length=self.win_size, hop_length=self.hop_size
+        ).squeeze(0)
         sil_tags = []
         silence_start = None
         clip_start = 0
@@ -139,7 +136,10 @@ class Slicer:  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.
                 continue
             # Clear recorded silence start if interval is not enough or clip is too short
             is_leading_silence = silence_start == 0 and i > self.max_sil_kept
-            need_slice_middle = i - silence_start >= self.min_interval and i - clip_start >= self.min_length
+            need_slice_middle = (
+                i - silence_start >= self.min_interval
+                and i - clip_start >= self.min_length
+            )
             if not is_leading_silence and not need_slice_middle:
                 silence_start = None
                 continue
@@ -152,10 +152,21 @@ class Slicer:  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.
                     sil_tags.append((pos, pos))
                 clip_start = pos
             elif i - silence_start <= self.max_sil_kept * 2:
-                pos = rms_list[i - self.max_sil_kept : silence_start + self.max_sil_kept + 1].argmin()
+                pos = rms_list[
+                    i - self.max_sil_kept : silence_start + self.max_sil_kept + 1
+                ].argmin()
                 pos += i - self.max_sil_kept
-                pos_l = rms_list[silence_start : silence_start + self.max_sil_kept + 1].argmin() + silence_start
-                pos_r = rms_list[i - self.max_sil_kept : i + 1].argmin() + i - self.max_sil_kept
+                pos_l = (
+                    rms_list[
+                        silence_start : silence_start + self.max_sil_kept + 1
+                    ].argmin()
+                    + silence_start
+                )
+                pos_r = (
+                    rms_list[i - self.max_sil_kept : i + 1].argmin()
+                    + i
+                    - self.max_sil_kept
+                )
                 if silence_start == 0:
                     sil_tags.append((0, pos_r))
                     clip_start = pos_r
@@ -163,8 +174,17 @@ class Slicer:  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.
                     sil_tags.append((min(pos_l, pos), max(pos_r, pos)))
                     clip_start = max(pos_r, pos)
             else:
-                pos_l = rms_list[silence_start : silence_start + self.max_sil_kept + 1].argmin() + silence_start
-                pos_r = rms_list[i - self.max_sil_kept : i + 1].argmin() + i - self.max_sil_kept
+                pos_l = (
+                    rms_list[
+                        silence_start : silence_start + self.max_sil_kept + 1
+                    ].argmin()
+                    + silence_start
+                )
+                pos_r = (
+                    rms_list[i - self.max_sil_kept : i + 1].argmin()
+                    + i
+                    - self.max_sil_kept
+                )
                 if silence_start == 0:
                     sil_tags.append((0, pos_r))
                 else:
@@ -173,39 +193,33 @@ class Slicer:  # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.
             silence_start = None
         # Deal with trailing silence.
         total_frames = rms_list.shape[0]
-        if silence_start is not None and total_frames - silence_start >= self.min_interval:
+        if (
+            silence_start is not None
+            and total_frames - silence_start >= self.min_interval
+        ):
             silence_end = min(total_frames, silence_start + self.max_sil_kept)
             pos = rms_list[silence_start : silence_end + 1].argmin() + silence_start
             sil_tags.append((pos, total_frames + 1))
         # Apply and return slices.
         ####音频+起始时间+终止时间
         if len(sil_tags) == 0:
-            return [[waveform, 0, int(total_frames * self.hop_size)]]
+            return [[waveform,0,int(total_frames*self.hop_size)]]
         else:
             chunks = []
             if sil_tags[0][0] > 0:
-                chunks.append([self._apply_slice(waveform, 0, sil_tags[0][0]), 0, int(sil_tags[0][0] * self.hop_size)])
+                chunks.append([self._apply_slice(waveform, 0, sil_tags[0][0]),0,int(sil_tags[0][0]*self.hop_size)])
             for i in range(len(sil_tags) - 1):
                 chunks.append(
-                    [
-                        self._apply_slice(waveform, sil_tags[i][1], sil_tags[i + 1][0]),
-                        int(sil_tags[i][1] * self.hop_size),
-                        int(sil_tags[i + 1][0] * self.hop_size),
-                    ]
+                    [self._apply_slice(waveform, sil_tags[i][1], sil_tags[i + 1][0]),int(sil_tags[i][1]*self.hop_size),int(sil_tags[i + 1][0]*self.hop_size)]
                 )
             if sil_tags[-1][1] < total_frames:
                 chunks.append(
-                    [
-                        self._apply_slice(waveform, sil_tags[-1][1], total_frames),
-                        int(sil_tags[-1][1] * self.hop_size),
-                        int(total_frames * self.hop_size),
-                    ]
+                    [self._apply_slice(waveform, sil_tags[-1][1], total_frames),int(sil_tags[-1][1]*self.hop_size),int(total_frames*self.hop_size)]
                 )
             return chunks
 
-
-# terminal
-def terminate_process_tree(pid, including_parent=True):
+#terminal
+def terminate_process_tree(pid, including_parent=True):  
     try:
         parent = psutil.Process(pid)
     except psutil.NoSuchProcess:
@@ -224,7 +238,6 @@ def terminate_process_tree(pid, including_parent=True):
         except OSError:
             pass
 
-
 def terminate_process(pid):
     if system == "Windows":
         cmd = f"taskkill /t /f /pid {pid}"
@@ -232,160 +245,130 @@ def terminate_process(pid):
     else:
         terminate_process_tree(pid)
 
+def start_training(dataset_name="",     
+    exp_name="F5TTS_Base",            
+    learning_rate=1e-4,               
+    batch_size_per_gpu=400,           
+    batch_size_type="frame",          
+    max_samples=64,                   
+    grad_accumulation_steps=1,        
+    max_grad_norm=1.0,                
+    epochs=11,                        
+    num_warmup_updates=200,           
+    save_per_updates=400,             
+    last_per_steps=800,               
+    finetune=True,                    
+    ):
 
-def start_training(
-    dataset_name="",
-    exp_name="F5TTS_Base",
-    learning_rate=1e-4,
-    batch_size_per_gpu=400,
-    batch_size_type="frame",
-    max_samples=64,
-    grad_accumulation_steps=1,
-    max_grad_norm=1.0,
-    epochs=11,
-    num_warmup_updates=200,
-    save_per_updates=400,
-    last_per_steps=800,
-    finetune=True,
-):
-    global training_process, tts_api
-
-    if tts_api is not None:
-        del tts_api
-        gc.collect()
-        torch.cuda.empty_cache()
-        tts_api = None
+
+    global training_process
 
     path_project = os.path.join(path_data, dataset_name + "_pinyin")
 
-    if not os.path.isdir(path_project):
-        yield (
-            f"There is not project with name {dataset_name}",
-            gr.update(interactive=True),
-            gr.update(interactive=False),
-        )
+    if os.path.isdir(path_project)==False:
+        yield f"There is not project with name {dataset_name}",gr.update(interactive=True),gr.update(interactive=False)
         return
 
-    file_raw = os.path.join(path_project, "raw.arrow")
-    if not os.path.isfile(file_raw):
-        yield f"There is no file {file_raw}", gr.update(interactive=True), gr.update(interactive=False)
-        return
+    file_raw = os.path.join(path_project,"raw.arrow")
+    if os.path.isfile(file_raw)==False:
+       yield f"There is no file {file_raw}",gr.update(interactive=True),gr.update(interactive=False)
+       return    
 
     # Check if a training process is already running
     if training_process is not None:
-        return "Train run already!", gr.update(interactive=False), gr.update(interactive=True)
+        return "Train run already!",gr.update(interactive=False),gr.update(interactive=True)
 
-    yield "start train", gr.update(interactive=False), gr.update(interactive=False)
+    yield "start train",gr.update(interactive=False),gr.update(interactive=False)
 
     # Command to run the training script with the specified arguments
-    cmd = (
-        f"accelerate launch finetune-cli.py --exp_name {exp_name} "
-        f"--learning_rate {learning_rate} "
-        f"--batch_size_per_gpu {batch_size_per_gpu} "
-        f"--batch_size_type {batch_size_type} "
-        f"--max_samples {max_samples} "
-        f"--grad_accumulation_steps {grad_accumulation_steps} "
-        f"--max_grad_norm {max_grad_norm} "
-        f"--epochs {epochs} "
-        f"--num_warmup_updates {num_warmup_updates} "
-        f"--save_per_updates {save_per_updates} "
-        f"--last_per_steps {last_per_steps} "
-        f"--dataset_name {dataset_name}"
-    )
-    if finetune:
-        cmd += f" --finetune {finetune}"
-
-    print(cmd)
-
+    cmd = f"accelerate launch finetune-cli.py --exp_name {exp_name} " \
+          f"--learning_rate {learning_rate} " \
+          f"--batch_size_per_gpu {batch_size_per_gpu} " \
+          f"--batch_size_type {batch_size_type} " \
+          f"--max_samples {max_samples} " \
+          f"--grad_accumulation_steps {grad_accumulation_steps} " \
+          f"--max_grad_norm {max_grad_norm} " \
+          f"--epochs {epochs} " \
+          f"--num_warmup_updates {num_warmup_updates} " \
+          f"--save_per_updates {save_per_updates} " \
+          f"--last_per_steps {last_per_steps} " \
+          f"--dataset_name {dataset_name}"  
+    if finetune:cmd += f" --finetune {finetune}"   
+    print(cmd)      
     try:
-        # Start the training process
-        training_process = subprocess.Popen(cmd, shell=True)
-
-        time.sleep(5)
-        yield "train start", gr.update(interactive=False), gr.update(interactive=True)
+      # Start the training process
+      training_process = subprocess.Popen(cmd, shell=True)
 
-        # Wait for the training process to finish
-        training_process.wait()
-        time.sleep(1)
-
-        if training_process is None:
-            text_info = "train stop"
-        else:
-            text_info = "train complete !"
+      time.sleep(5) 
+      yield "check terminal for wandb",gr.update(interactive=False),gr.update(interactive=True) 
+      
+      # Wait for the training process to finish
+      training_process.wait()
+      time.sleep(1)
+      
+      if training_process is None: 
+         text_info = 'train stop'
+      else:     
+         text_info = "train complete !"
 
     except Exception as e:  # Catch all exceptions
         # Ensure that we reset the training process variable in case of an error
-        text_info = f"An error occurred: {str(e)}"
-
-    training_process = None
-
-    yield text_info, gr.update(interactive=True), gr.update(interactive=False)
+        text_info=f"An error occurred: {str(e)}"
+    
+    training_process=None
 
+    yield text_info,gr.update(interactive=True),gr.update(interactive=False)
 
 def stop_training():
     global training_process
-    if training_process is None:
-        return "Train not run !", gr.update(interactive=True), gr.update(interactive=False)
+    if training_process is None:return f"Train not run !",gr.update(interactive=True),gr.update(interactive=False)
     terminate_process_tree(training_process.pid)
     training_process = None
-    return "train stop", gr.update(interactive=True), gr.update(interactive=False)
-
+    return 'train stop',gr.update(interactive=True),gr.update(interactive=False)
 
 def create_data_project(name):
-    name += "_pinyin"
-    os.makedirs(os.path.join(path_data, name), exist_ok=True)
-    os.makedirs(os.path.join(path_data, name, "dataset"), exist_ok=True)
-
-
-def transcribe(file_audio, language="english"):
+    name+="_pinyin"
+    os.makedirs(os.path.join(path_data,name),exist_ok=True)
+    os.makedirs(os.path.join(path_data,name,"dataset"),exist_ok=True)
+    
+def transcribe(file_audio,language="english"):
     global pipe
 
     if pipe is None:
-        pipe = pipeline(
-            "automatic-speech-recognition",
-            model="openai/whisper-large-v3-turbo",
-            torch_dtype=torch.float16,
-            device=device,
-        )
+       pipe = pipeline("automatic-speech-recognition",model="openai/whisper-large-v3-turbo", torch_dtype=torch.float16,device=device)
 
     text_transcribe = pipe(
         file_audio,
         chunk_length_s=30,
         batch_size=128,
-        generate_kwargs={"task": "transcribe", "language": language},
+        generate_kwargs={"task": "transcribe","language": language},
         return_timestamps=False,
     )["text"].strip()
     return text_transcribe
 
+def transcribe_all(name_project,audio_files,language,user=False,progress=gr.Progress()):
+    name_project+="_pinyin"
+    path_project= os.path.join(path_data,name_project)
+    path_dataset = os.path.join(path_project,"dataset")
+    path_project_wavs = os.path.join(path_project,"wavs")
+    file_metadata = os.path.join(path_project,"metadata.csv")
 
-def transcribe_all(name_project, audio_files, language, user=False, progress=gr.Progress()):
-    name_project += "_pinyin"
-    path_project = os.path.join(path_data, name_project)
-    path_dataset = os.path.join(path_project, "dataset")
-    path_project_wavs = os.path.join(path_project, "wavs")
-    file_metadata = os.path.join(path_project, "metadata.csv")
-
-    if audio_files is None:
-        return "You need to load an audio file."
+    if audio_files is None:return "You need to load an audio file."
 
     if os.path.isdir(path_project_wavs):
-        shutil.rmtree(path_project_wavs)
+       shutil.rmtree(path_project_wavs)
 
     if os.path.isfile(file_metadata):
-        os.remove(file_metadata)
-
-    os.makedirs(path_project_wavs, exist_ok=True)
+       os.remove(file_metadata)
 
+    os.makedirs(path_project_wavs,exist_ok=True)
+    
     if user:
-        file_audios = [
-            file
-            for format in ("*.wav", "*.ogg", "*.opus", "*.mp3", "*.flac")
-            for file in glob(os.path.join(path_dataset, format))
-        ]
-        if file_audios == []:
-            return "No audio file was found in the dataset."
+       file_audios = [file for format in ('*.wav', '*.ogg', '*.opus', '*.mp3', '*.flac') for file in glob(os.path.join(path_dataset, format))]
+       if file_audios==[]:return "No audio file was found in the dataset."
     else:
-        file_audios = audio_files
+       file_audios = audio_files
+       
 
     alpha = 0.5
     _max = 1.0
@@ -393,213 +376,179 @@ def transcribe_all(name_project, audio_files, language, user=False, progress=gr.
 
     num = 0
     error_num = 0
-    data = ""
-    for file_audio in progress.tqdm(file_audios, desc="transcribe files", total=len((file_audios))):
-        audio, _ = librosa.load(file_audio, sr=24000, mono=True)
-
-        list_slicer = slicer.slice(audio)
-        for chunk, start, end in progress.tqdm(list_slicer, total=len(list_slicer), desc="slicer files"):
+    data=""
+    for file_audio in progress.tqdm(file_audios, desc="transcribe files",total=len((file_audios))):
+        
+        audio, _ = librosa.load(file_audio, sr=24000, mono=True) 
+
+        list_slicer=slicer.slice(audio)
+        for chunk, start, end in progress.tqdm(list_slicer,total=len(list_slicer), desc="slicer files"): 
+            
             name_segment = os.path.join(f"segment_{num}")
-            file_segment = os.path.join(path_project_wavs, f"{name_segment}.wav")
-
+            file_segment = os.path.join(path_project_wavs, f"{name_segment}.wav")    
+            
             tmp_max = np.abs(chunk).max()
-            if tmp_max > 1:
-                chunk /= tmp_max
+            if(tmp_max>1):chunk/=tmp_max
             chunk = (chunk / tmp_max * (_max * alpha)) + (1 - alpha) * chunk
-            wavfile.write(file_segment, 24000, (chunk * 32767).astype(np.int16))
-
+            wavfile.write(file_segment,24000, (chunk * 32767).astype(np.int16))
+            
             try:
-                text = transcribe(file_segment, language)
-                text = text.lower().strip().replace('"', "")
+               text=transcribe(file_segment,language)
+               text = text.lower().strip().replace('"',"")
 
-                data += f"{name_segment}|{text}\n"
+               data+= f"{name_segment}|{text}\n"
 
-                num += 1
-            except:  # noqa: E722
-                error_num += 1
+               num+=1
+            except:
+               error_num +=1
 
-    with open(file_metadata, "w", encoding="utf-8") as f:
+    with open(file_metadata,"w",encoding="utf-8") as f:
         f.write(data)
-
-    if error_num != []:
-        error_text = f"\nerror files : {error_num}"
+    
+    if error_num!=[]:
+       error_text=f"\nerror files : {error_num}"    
     else:
-        error_text = ""
-
+       error_text=""
+       
     return f"transcribe complete samples : {num}\npath : {path_project_wavs}{error_text}"
 
-
 def format_seconds_to_hms(seconds):
     hours = int(seconds / 3600)
     minutes = int((seconds % 3600) / 60)
     seconds = seconds % 60
     return "{:02d}:{:02d}:{:02d}".format(hours, minutes, int(seconds))
 
-
-def create_metadata(name_project, progress=gr.Progress()):
-    name_project += "_pinyin"
-    path_project = os.path.join(path_data, name_project)
-    path_project_wavs = os.path.join(path_project, "wavs")
-    file_metadata = os.path.join(path_project, "metadata.csv")
-    file_raw = os.path.join(path_project, "raw.arrow")
-    file_duration = os.path.join(path_project, "duration.json")
-    file_vocab = os.path.join(path_project, "vocab.txt")
-
-    if not os.path.isfile(file_metadata):
-        return "The file was not found in " + file_metadata
-
-    with open(file_metadata, "r", encoding="utf-8") as f:
-        data = f.read()
-
-    audio_path_list = []
-    text_list = []
-    duration_list = []
-
-    count = data.split("\n")
-    lenght = 0
-    result = []
-    error_files = []
-    for line in progress.tqdm(data.split("\n"), total=count):
-        sp_line = line.split("|")
-        if len(sp_line) != 2:
-            continue
-        name_audio, text = sp_line[:2]
+def create_metadata(name_project,progress=gr.Progress()):
+    name_project+="_pinyin"
+    path_project= os.path.join(path_data,name_project)
+    path_project_wavs = os.path.join(path_project,"wavs")
+    file_metadata = os.path.join(path_project,"metadata.csv")
+    file_raw = os.path.join(path_project,"raw.arrow")
+    file_duration = os.path.join(path_project,"duration.json")
+    file_vocab = os.path.join(path_project,"vocab.txt")
+
+    if os.path.isfile(file_metadata)==False: return "The file was not found in " + file_metadata 
+    
+    with open(file_metadata,"r",encoding="utf-8") as f:
+        data=f.read()
+    
+    audio_path_list=[]
+    text_list=[]
+    duration_list=[]
+    
+    count=data.split("\n")
+    lenght=0
+    result=[]
+    error_files=[]
+    for line in progress.tqdm(data.split("\n"),total=count):
+        sp_line=line.split("|")
+        if len(sp_line)!=2:continue
+        name_audio,text = sp_line[:2]
 
         file_audio = os.path.join(path_project_wavs, name_audio + ".wav")
 
-        if not os.path.isfile(file_audio):
+        if os.path.isfile(file_audio)==False:
             error_files.append(file_audio)
             continue
 
         duraction = get_audio_duration(file_audio)
-        if duraction < 2 and duraction > 15:
-            continue
-        if len(text) < 4:
-            continue
+        if duraction<2 and duraction>15:continue
+        if len(text)<4:continue
 
         text = clear_text(text)
-        text = convert_char_to_pinyin([text], polyphone=True)[0]
+        text = convert_char_to_pinyin([text], polyphone = True)[0]
 
         audio_path_list.append(file_audio)
         duration_list.append(duraction)
         text_list.append(text)
-
+      
         result.append({"audio_path": file_audio, "text": text, "duration": duraction})
 
-        lenght += duraction
+        lenght+=duraction
 
-    if duration_list == []:
-        error_files_text = "\n".join(error_files)
+    if duration_list==[]:
+        error_files_text="\n".join(error_files)
         return f"Error: No audio files found in the specified path : \n{error_files_text}"
-
-    min_second = round(min(duration_list), 2)
-    max_second = round(max(duration_list), 2)
+    
+    min_second = round(min(duration_list),2)   
+    max_second = round(max(duration_list),2)
 
     with ArrowWriter(path=file_raw, writer_batch_size=1) as writer:
-        for line in progress.tqdm(result, total=len(result), desc="prepare data"):
+        for line in progress.tqdm(result,total=len(result), desc=f"prepare data"):
             writer.write(line)
 
-    with open(file_duration, "w", encoding="utf-8") as f:
+    with open(file_duration, 'w', encoding='utf-8') as f:
         json.dump({"duration": duration_list}, f, ensure_ascii=False)
-
-    file_vocab_finetune = "data/Emilia_ZH_EN_pinyin/vocab.txt"
-    if not os.path.isfile(file_vocab_finetune):
-        return "Error: Vocabulary file 'Emilia_ZH_EN_pinyin' not found!"
+ 
+    file_vocab_finetune = "data/Emilia_ZH_EN_pinyin/vocab.txt"    
+    if os.path.isfile(file_vocab_finetune==False):return "Error: Vocabulary file 'Emilia_ZH_EN_pinyin' not found!"
     shutil.copy2(file_vocab_finetune, file_vocab)
-
-    if error_files != []:
-        error_text = "error files\n" + "\n".join(error_files)
+    
+    if error_files!=[]:
+       error_text="error files\n" + "\n".join(error_files)
     else:
-        error_text = ""
-
+       error_text=""
+        
     return f"prepare complete \nsamples : {len(text_list)}\ntime data : {format_seconds_to_hms(lenght)}\nmin sec : {min_second}\nmax sec : {max_second}\nfile_arrow : {file_raw}\n{error_text}"
 
-
 def check_user(value):
-    return gr.update(visible=not value), gr.update(visible=value)
-
-
-def calculate_train(
-    name_project,
-    batch_size_type,
-    max_samples,
-    learning_rate,
-    num_warmup_updates,
-    save_per_updates,
-    last_per_steps,
-    finetune,
-):
-    name_project += "_pinyin"
-    path_project = os.path.join(path_data, name_project)
-    file_duraction = os.path.join(path_project, "duration.json")
-
-    if not os.path.isfile(file_duraction):
-        return (
-            1000,
-            max_samples,
-            num_warmup_updates,
-            save_per_updates,
-            last_per_steps,
-            "project not found !",
-            learning_rate,
-        )
+    return gr.update(visible=not value),gr.update(visible=value)
 
-    with open(file_duraction, "r") as file:
-        data = json.load(file)
+def calculate_train(name_project,batch_size_type,max_samples,learning_rate,num_warmup_updates,save_per_updates,last_per_steps,finetune):
+    name_project+="_pinyin"
+    path_project= os.path.join(path_data,name_project)
+    file_duraction = os.path.join(path_project,"duration.json")
 
-    duration_list = data["duration"]
+    with open(file_duraction, 'r') as file:
+        data = json.load(file)  
+    
+    duration_list = data['duration']
 
     samples = len(duration_list)
 
     if torch.cuda.is_available():
         gpu_properties = torch.cuda.get_device_properties(0)
-        total_memory = gpu_properties.total_memory / (1024**3)
+        total_memory = gpu_properties.total_memory / (1024 ** 3)  
     elif torch.backends.mps.is_available():
-        total_memory = psutil.virtual_memory().available / (1024**3)
-
-    if batch_size_type == "frame":
-        batch = int(total_memory * 0.5)
-        batch = (lambda num: num + 1 if num % 2 != 0 else num)(batch)
-        batch_size_per_gpu = int(38400 / batch)
-    else:
-        batch_size_per_gpu = int(total_memory / 8)
-        batch_size_per_gpu = (lambda num: num + 1 if num % 2 != 0 else num)(batch_size_per_gpu)
-        batch = batch_size_per_gpu
-
-    if batch_size_per_gpu <= 0:
-        batch_size_per_gpu = 1
-
-    if samples < 64:
-        max_samples = int(samples * 0.25)
+        total_memory = psutil.virtual_memory().available / (1024 ** 3)
+    
+    if batch_size_type=="frame":
+       batch = int(total_memory * 0.5)
+       batch = (lambda num: num + 1 if num % 2 != 0 else num)(batch)
+       batch_size_per_gpu = int(38400 / batch )
     else:
-        max_samples = 64
-
-    num_warmup_updates = int(samples * 0.05)
-    save_per_updates = int(samples * 0.10)
-    last_per_steps = int(save_per_updates * 5)
-
+       batch_size_per_gpu = int(total_memory / 8)   
+       batch_size_per_gpu = (lambda num: num + 1 if num % 2 != 0 else num)(batch_size_per_gpu)
+       batch = batch_size_per_gpu
+
+    if batch_size_per_gpu<=0:batch_size_per_gpu=1
+
+    if samples<64:
+       max_samples = int(samples * 0.25) 
+    
+    num_warmup_updates = int(samples * 0.10) 
+    save_per_updates = int(samples * 0.25) 
+    last_per_steps =int(save_per_updates * 5)
+    
     max_samples = (lambda num: num + 1 if num % 2 != 0 else num)(max_samples)
     num_warmup_updates = (lambda num: num + 1 if num % 2 != 0 else num)(num_warmup_updates)
     save_per_updates = (lambda num: num + 1 if num % 2 != 0 else num)(save_per_updates)
     last_per_steps = (lambda num: num + 1 if num % 2 != 0 else num)(last_per_steps)
 
-    if finetune:
-        learning_rate = 1e-5
-    else:
-        learning_rate = 7.5e-5
-
-    return batch_size_per_gpu, max_samples, num_warmup_updates, save_per_updates, last_per_steps, samples, learning_rate
+    if finetune:learning_rate=1e-4
+    else:learning_rate=7.5e-5
 
+    return batch_size_per_gpu,max_samples,num_warmup_updates,save_per_updates,last_per_steps,samples,learning_rate
 
 def extract_and_save_ema_model(checkpoint_path: str, new_checkpoint_path: str) -> None:
     try:
         checkpoint = torch.load(checkpoint_path)
         print("Original Checkpoint Keys:", checkpoint.keys())
-
-        ema_model_state_dict = checkpoint.get("ema_model_state_dict", None)
+        
+        ema_model_state_dict = checkpoint.get('ema_model_state_dict', None)
 
         if ema_model_state_dict is not None:
-            new_checkpoint = {"ema_model_state_dict": ema_model_state_dict}
+            new_checkpoint = {'ema_model_state_dict': ema_model_state_dict}
             torch.save(new_checkpoint, new_checkpoint_path)
             return f"New checkpoint saved at: {new_checkpoint_path}"
         else:
@@ -608,136 +557,65 @@ def extract_and_save_ema_model(checkpoint_path: str, new_checkpoint_path: str) -
     except Exception as e:
         return f"An error occurred: {e}"
 
-
 def vocab_check(project_name):
     name_project = project_name + "_pinyin"
     path_project = os.path.join(path_data, name_project)
 
     file_metadata = os.path.join(path_project, "metadata.csv")
-
-    file_vocab = "data/Emilia_ZH_EN_pinyin/vocab.txt"
-    if not os.path.isfile(file_vocab):
+   
+    file_vocab="data/Emilia_ZH_EN_pinyin/vocab.txt"
+    if os.path.isfile(file_vocab)==False:
         return f"the file {file_vocab} not found !"
-
-    with open(file_vocab, "r", encoding="utf-8") as f:
-        data = f.read()
+    
+    with open(file_vocab,"r",encoding="utf-8") as f:
+         data=f.read()
 
     vocab = data.split("\n")
 
-    if not os.path.isfile(file_metadata):
+    if os.path.isfile(file_metadata)==False:
         return f"the file {file_metadata} not found !"
 
-    with open(file_metadata, "r", encoding="utf-8") as f:
-        data = f.read()
+    with open(file_metadata,"r",encoding="utf-8") as f:
+         data=f.read()
 
-    miss_symbols = []
-    miss_symbols_keep = {}
+    miss_symbols=[]
+    miss_symbols_keep={}
     for item in data.split("\n"):
-        sp = item.split("|")
-        if len(sp) != 2:
-            continue
+         sp=item.split("|")
+         if len(sp)!=2:continue
+         text=sp[1].lower().strip()
 
-        text = sp[1].lower().strip()
+         for t in text:
+             if (t in vocab)==False and (t in miss_symbols_keep)==False:
+                miss_symbols.append(t) 
+                miss_symbols_keep[t]=t
 
-        for t in text:
-            if t not in vocab and t not in miss_symbols_keep:
-                miss_symbols.append(t)
-                miss_symbols_keep[t] = t
-    if miss_symbols == []:
-        info = "You can train using your language !"
-    else:
-        info = f"The following symbols are missing in your language : {len(miss_symbols)}\n\n" + "\n".join(miss_symbols)
+            
+    if miss_symbols==[]:info ="You can train using your language !"     
+    else:info = f"The following symbols are missing in your language : {len(miss_symbols)}\n\n" + "\n".join(miss_symbols)   
 
     return info
 
 
-def get_random_sample_prepare(project_name):
-    name_project = project_name + "_pinyin"
-    path_project = os.path.join(path_data, name_project)
-    file_arrow = os.path.join(path_project, "raw.arrow")
-    if not os.path.isfile(file_arrow):
-        return "", None
-    dataset = Dataset_.from_file(file_arrow)
-    random_sample = dataset.shuffle(seed=random.randint(0, 1000)).select([0])
-    text = "[" + " , ".join(["' " + t + " '" for t in random_sample["text"][0]]) + "]"
-    audio_path = random_sample["audio_path"][0]
-    return text, audio_path
-
-
-def get_random_sample_transcribe(project_name):
-    name_project = project_name + "_pinyin"
-    path_project = os.path.join(path_data, name_project)
-    file_metadata = os.path.join(path_project, "metadata.csv")
-    if not os.path.isfile(file_metadata):
-        return "", None
-
-    data = ""
-    with open(file_metadata, "r", encoding="utf-8") as f:
-        data = f.read()
-
-    list_data = []
-    for item in data.split("\n"):
-        sp = item.split("|")
-        if len(sp) != 2:
-            continue
-        list_data.append([os.path.join(path_project, "wavs", sp[0] + ".wav"), sp[1]])
 
-    if list_data == []:
-        return "", None
-
-    random_item = random.choice(list_data)
-
-    return random_item[1], random_item[0]
-
-
-def get_random_sample_infer(project_name):
-    text, audio = get_random_sample_transcribe(project_name)
-    return (
-        text,
-        text,
-        audio,
-    )
-
-
-def infer(file_checkpoint, exp_name, ref_text, ref_audio, gen_text, nfe_step):
-    global last_checkpoint, last_device, tts_api
-
-    if not os.path.isfile(file_checkpoint):
-        return None
-
-    if training_process is not None:
-        device_test = "cpu"
-    else:
-        device_test = None
-
-    if last_checkpoint != file_checkpoint or last_device != device_test:
-        if last_checkpoint != file_checkpoint:
-            last_checkpoint = file_checkpoint
-        if last_device != device_test:
-            last_device = device_test
-
-        tts_api = F5TTS(model_type=exp_name, ckpt_file=file_checkpoint, device=device_test)
-
-        print("update", device_test, file_checkpoint)
+with gr.Blocks() as app:
 
-    with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
-        tts_api.infer(gen_text=gen_text, ref_text=ref_text, ref_file=ref_audio, nfe_step=nfe_step, file_wave=f.name)
-        return f.name
+    with gr.Row():
+         project_name=gr.Textbox(label="project name",value="my_speak")
+         bt_create=gr.Button("create new project")
+    
+    bt_create.click(fn=create_data_project,inputs=[project_name])
 
+    with gr.Tabs():
+         
 
-with gr.Blocks() as app:
-    with gr.Row():
-        project_name = gr.Textbox(label="project name", value="my_speak")
-        bt_create = gr.Button("create new project")
+         with gr.TabItem("transcribe Data"):
 
-    bt_create.click(fn=create_data_project, inputs=[project_name])
 
-    with gr.Tabs():
-        with gr.TabItem("transcribe Data"):
-            ch_manual = gr.Checkbox(label="user", value=False)
+              ch_manual = gr.Checkbox(label="user",value=False)
 
-            mark_info_transcribe = gr.Markdown(
-                """```plaintext    
+              mark_info_transcribe=gr.Markdown(
+                         """```plaintext    
      Place your 'wavs' folder and 'metadata.csv' file in the {your_project_name}' directory. 
                  
      my_speak/
@@ -746,36 +624,18 @@ with gr.Blocks() as app:
          ├── audio1.wav
          └── audio2.wav
          ...
-     ```""",
-                visible=False,
-            )
-
-            audio_speaker = gr.File(label="voice", type="filepath", file_count="multiple")
-            txt_lang = gr.Text(label="Language", value="english")
-            bt_transcribe = bt_create = gr.Button("transcribe")
-            txt_info_transcribe = gr.Text(label="info", value="")
-            bt_transcribe.click(
-                fn=transcribe_all,
-                inputs=[project_name, audio_speaker, txt_lang, ch_manual],
-                outputs=[txt_info_transcribe],
-            )
-            ch_manual.change(fn=check_user, inputs=[ch_manual], outputs=[audio_speaker, mark_info_transcribe])
-
-            random_sample_transcribe = gr.Button("random sample")
-
-            with gr.Row():
-                random_text_transcribe = gr.Text(label="Text")
-                random_audio_transcribe = gr.Audio(label="Audio", type="filepath")
-
-            random_sample_transcribe.click(
-                fn=get_random_sample_transcribe,
-                inputs=[project_name],
-                outputs=[random_text_transcribe, random_audio_transcribe],
-            )
-
-        with gr.TabItem("prepare Data"):
-            gr.Markdown(
-                """```plaintext    
+     ```""",visible=False)
+
+              audio_speaker = gr.File(label="voice",type="filepath",file_count="multiple")
+              txt_lang = gr.Text(label="Language",value="english")
+              bt_transcribe=bt_create=gr.Button("transcribe")
+              txt_info_transcribe=gr.Text(label="info",value="")
+              bt_transcribe.click(fn=transcribe_all,inputs=[project_name,audio_speaker,txt_lang,ch_manual],outputs=[txt_info_transcribe])
+              ch_manual.change(fn=check_user,inputs=[ch_manual],outputs=[audio_speaker,mark_info_transcribe])
+         
+         with gr.TabItem("prepare Data"):
+              gr.Markdown(
+                         """```plaintext    
      place all your wavs folder and your metadata.csv file in {your name project}                                 
      my_speak/
      │
@@ -792,136 +652,61 @@ with gr.Blocks() as app:
      audio2|text1
      ...
 
-     ```"""
-            )
-
-            bt_prepare = bt_create = gr.Button("prepare")
-            txt_info_prepare = gr.Text(label="info", value="")
-            bt_prepare.click(fn=create_metadata, inputs=[project_name], outputs=[txt_info_prepare])
-
-            random_sample_prepare = gr.Button("random sample")
-
-            with gr.Row():
-                random_text_prepare = gr.Text(label="Pinyin")
-                random_audio_prepare = gr.Audio(label="Audio", type="filepath")
-
-            random_sample_prepare.click(
-                fn=get_random_sample_prepare, inputs=[project_name], outputs=[random_text_prepare, random_audio_prepare]
-            )
-
-        with gr.TabItem("train Data"):
-            with gr.Row():
-                bt_calculate = bt_create = gr.Button("Auto Settings")
-                ch_finetune = bt_create = gr.Checkbox(label="finetune", value=True)
-                lb_samples = gr.Label(label="samples")
-                batch_size_type = gr.Radio(label="Batch Size Type", choices=["frame", "sample"], value="frame")
-
-            with gr.Row():
-                exp_name = gr.Radio(label="Model", choices=["F5TTS_Base", "E2TTS_Base"], value="F5TTS_Base")
-                learning_rate = gr.Number(label="Learning Rate", value=1e-5, step=1e-5)
-
-            with gr.Row():
-                batch_size_per_gpu = gr.Number(label="Batch Size per GPU", value=1000)
-                max_samples = gr.Number(label="Max Samples", value=64)
-
-            with gr.Row():
-                grad_accumulation_steps = gr.Number(label="Gradient Accumulation Steps", value=1)
-                max_grad_norm = gr.Number(label="Max Gradient Norm", value=1.0)
-
-            with gr.Row():
-                epochs = gr.Number(label="Epochs", value=10)
-                num_warmup_updates = gr.Number(label="Warmup Updates", value=5)
-
-            with gr.Row():
-                save_per_updates = gr.Number(label="Save per Updates", value=10)
-                last_per_steps = gr.Number(label="Last per Steps", value=50)
-
-            with gr.Row():
-                start_button = gr.Button("Start Training")
-                stop_button = gr.Button("Stop Training", interactive=False)
-
-            txt_info_train = gr.Text(label="info", value="")
-            start_button.click(
-                fn=start_training,
-                inputs=[
-                    project_name,
-                    exp_name,
-                    learning_rate,
-                    batch_size_per_gpu,
-                    batch_size_type,
-                    max_samples,
-                    grad_accumulation_steps,
-                    max_grad_norm,
-                    epochs,
-                    num_warmup_updates,
-                    save_per_updates,
-                    last_per_steps,
-                    ch_finetune,
-                ],
-                outputs=[txt_info_train, start_button, stop_button],
-            )
-            stop_button.click(fn=stop_training, outputs=[txt_info_train, start_button, stop_button])
-            bt_calculate.click(
-                fn=calculate_train,
-                inputs=[
-                    project_name,
-                    batch_size_type,
-                    max_samples,
-                    learning_rate,
-                    num_warmup_updates,
-                    save_per_updates,
-                    last_per_steps,
-                    ch_finetune,
-                ],
-                outputs=[
-                    batch_size_per_gpu,
-                    max_samples,
-                    num_warmup_updates,
-                    save_per_updates,
-                    last_per_steps,
-                    lb_samples,
-                    learning_rate,
-                ],
-            )
-
-        with gr.TabItem("reduse checkpoint"):
-            txt_path_checkpoint = gr.Text(label="path checkpoint :")
-            txt_path_checkpoint_small = gr.Text(label="path output :")
-            txt_info_reduse = gr.Text(label="info", value="")
-            reduse_button = gr.Button("reduse")
-            reduse_button.click(
-                fn=extract_and_save_ema_model,
-                inputs=[txt_path_checkpoint, txt_path_checkpoint_small],
-                outputs=[txt_info_reduse],
-            )
-
-        with gr.TabItem("vocab check experiment"):
-            check_button = gr.Button("check vocab")
-            txt_info_check = gr.Text(label="info", value="")
-            check_button.click(fn=vocab_check, inputs=[project_name], outputs=[txt_info_check])
-
-        with gr.TabItem("test model"):
-            exp_name = gr.Radio(label="Model", choices=["F5-TTS", "E2-TTS"], value="F5-TTS")
-            nfe_step = gr.Number(label="n_step", value=32)
-            file_checkpoint_pt = gr.Textbox(label="Checkpoint", value="")
-
-            random_sample_infer = gr.Button("random sample")
-
-            ref_text = gr.Textbox(label="ref text")
-            ref_audio = gr.Audio(label="audio ref", type="filepath")
-            gen_text = gr.Textbox(label="gen text")
-            random_sample_infer.click(
-                fn=get_random_sample_infer, inputs=[project_name], outputs=[ref_text, gen_text, ref_audio]
-            )
-            check_button_infer = gr.Button("infer")
-            gen_audio = gr.Audio(label="audio gen", type="filepath")
-
-            check_button_infer.click(
-                fn=infer,
-                inputs=[file_checkpoint_pt, exp_name, ref_text, ref_audio, gen_text, nfe_step],
-                outputs=[gen_audio],
-            )
-
+     ```""")
+
+              bt_prepare=bt_create=gr.Button("prepare")
+              txt_info_prepare=gr.Text(label="info",value="")
+              bt_prepare.click(fn=create_metadata,inputs=[project_name],outputs=[txt_info_prepare])
+         
+         with gr.TabItem("train Data"):
+              
+              with gr.Row():    
+                   bt_calculate=bt_create=gr.Button("Auto Settings") 
+                   ch_finetune=bt_create=gr.Checkbox(label="finetune",value=True)   
+                   lb_samples = gr.Label(label="samples")  
+                   batch_size_type = gr.Radio(label="Batch Size Type", choices=["frame", "sample"], value="frame")  
+                
+              with gr.Row():     
+                   exp_name = gr.Radio(label="Model", choices=["F5TTS_Base", "E2TTS_Base"], value="F5TTS_Base")
+                   learning_rate = gr.Number(label="Learning Rate", value=1e-4, step=1e-4)
+                   
+              with gr.Row():
+                   batch_size_per_gpu = gr.Number(label="Batch Size per GPU", value=1000)
+                   max_samples = gr.Number(label="Max Samples", value=16)
+                   
+              with gr.Row():
+                   grad_accumulation_steps = gr.Number(label="Gradient Accumulation Steps", value=1)
+                   max_grad_norm = gr.Number(label="Max Gradient Norm", value=1.0)
+                   
+              with gr.Row():
+                   epochs = gr.Number(label="Epochs", value=10)
+                   num_warmup_updates = gr.Number(label="Warmup Updates", value=5)
+
+              with gr.Row():     
+                   save_per_updates = gr.Number(label="Save per Updates", value=10)
+                   last_per_steps = gr.Number(label="Last per Steps", value=50)
+       
+              with gr.Row():
+                   start_button = gr.Button("Start Training")
+                   stop_button = gr.Button("Stop Training",interactive=False)
+         
+              txt_info_train=gr.Text(label="info",value="")
+              start_button.click(fn=start_training,inputs=[project_name,exp_name,learning_rate,batch_size_per_gpu,batch_size_type,max_samples,grad_accumulation_steps,max_grad_norm,epochs,num_warmup_updates,save_per_updates,last_per_steps,ch_finetune],outputs=[txt_info_train,start_button,stop_button])
+              stop_button.click(fn=stop_training,outputs=[txt_info_train,start_button,stop_button])
+              bt_calculate.click(fn=calculate_train,inputs=[project_name,batch_size_type,max_samples,learning_rate,num_warmup_updates,save_per_updates,last_per_steps,ch_finetune],outputs=[batch_size_per_gpu,max_samples,num_warmup_updates,save_per_updates,last_per_steps,lb_samples,learning_rate]) 
+         
+         with gr.TabItem("reduse checkpoint"):
+              txt_path_checkpoint = gr.Text(label="path checkpoint :") 
+              txt_path_checkpoint_small = gr.Text(label="path output :") 
+              txt_info_reduse = gr.Text(label="info",value="")
+              reduse_button = gr.Button("reduse") 
+              reduse_button.click(fn=extract_and_save_ema_model,inputs=[txt_path_checkpoint,txt_path_checkpoint_small],outputs=[txt_info_reduse])
+
+         with gr.TabItem("vocab check experiment"):
+              check_button = gr.Button("check vocab") 
+              txt_info_check=gr.Text(label="info",value="")
+              check_button.click(fn=vocab_check,inputs=[project_name],outputs=[txt_info_check])
+       
 
 @click.command()
 @click.option("--port", "-p", default=None, type=int, help="Port to run the app on")
@@ -936,9 +721,10 @@ with gr.Blocks() as app:
 @click.option("--api", "-a", default=True, is_flag=True, help="Allow API access")
 def main(port, host, share, api):
     global app
-    print("Starting app...")
-    app.queue(api_open=api).launch(server_name=host, server_port=port, share=share, show_api=api)
-
+    print(f"Starting app...")
+    app.queue(api_open=api).launch(
+        server_name=host, server_port=port, share=share, show_api=api
+    )
 
 if __name__ == "__main__":
     main()

gradio_app.py

@@ -1,824 +0,0 @@
-import os
-import re
-import torch
-import torchaudio
-import gradio as gr
-import numpy as np
-import tempfile
-from einops import rearrange
-from vocos import Vocos
-from pydub import AudioSegment, silence
-from model import CFM, UNetT, DiT, MMDiT
-from cached_path import cached_path
-from model.utils import (
-    load_checkpoint,
-    get_tokenizer,
-    convert_char_to_pinyin,
-    save_spectrogram,
-)
-from transformers import pipeline
-import librosa
-import click
-import soundfile as sf
-
-try:
-    import spaces
-    USING_SPACES = True
-except ImportError:
-    USING_SPACES = False
-
-def gpu_decorator(func):
-    if USING_SPACES:
-        return spaces.GPU(func)
-    else:
-        return func
-
-
-
-SPLIT_WORDS = [
-    "but", "however", "nevertheless", "yet", "still",
-    "therefore", "thus", "hence", "consequently",
-    "moreover", "furthermore", "additionally",
-    "meanwhile", "alternatively", "otherwise",
-    "namely", "specifically", "for example", "such as",
-    "in fact", "indeed", "notably",
-    "in contrast", "on the other hand", "conversely",
-    "in conclusion", "to summarize", "finally"
-]
-
-device = (
-    "cuda"
-    if torch.cuda.is_available()
-    else "mps" if torch.backends.mps.is_available() else "cpu"
-)
-
-print(f"Using {device} device")
-
-pipe = pipeline(
-    "automatic-speech-recognition",
-    model="openai/whisper-large-v3-turbo",
-    torch_dtype=torch.float16,
-    device=device,
-)
-vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
-
-# --------------------- Settings -------------------- #
-
-target_sample_rate = 24000
-n_mel_channels = 100
-hop_length = 256
-target_rms = 0.1
-nfe_step = 32  # 16, 32
-cfg_strength = 2.0
-ode_method = "euler"
-sway_sampling_coef = -1.0
-speed = 1.0
-# fix_duration = 27  # None or float (duration in seconds)
-fix_duration = None
-
-
-def load_model(repo_name, exp_name, model_cls, model_cfg, ckpt_step):
-    ckpt_path = str(cached_path(f"hf://SWivid/{repo_name}/{exp_name}/model_{ckpt_step}.safetensors"))
-    # ckpt_path = f"ckpts/{exp_name}/model_{ckpt_step}.pt"  # .pt | .safetensors
-    vocab_char_map, vocab_size = get_tokenizer("Emilia_ZH_EN", "pinyin")
-    model = CFM(
-        transformer=model_cls(
-            **model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels
-        ),
-        mel_spec_kwargs=dict(
-            target_sample_rate=target_sample_rate,
-            n_mel_channels=n_mel_channels,
-            hop_length=hop_length,
-        ),
-        odeint_kwargs=dict(
-            method=ode_method,
-        ),
-        vocab_char_map=vocab_char_map,
-    ).to(device)
-
-    model = load_checkpoint(model, ckpt_path, device, use_ema = True)
-
-    return model
-
-
-# load models
-F5TTS_model_cfg = dict(
-    dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4
-)
-E2TTS_model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
-
-F5TTS_ema_model = load_model(
-    "F5-TTS", "F5TTS_Base", DiT, F5TTS_model_cfg, 1200000
-)
-E2TTS_ema_model = load_model(
-    "E2-TTS", "E2TTS_Base", UNetT, E2TTS_model_cfg, 1200000
-)
-
-def split_text_into_batches(text, max_chars=200, split_words=SPLIT_WORDS):
-    if len(text.encode('utf-8')) <= max_chars:
-        return [text]
-    if text[-1] not in ['。', '.', '!', '！', '?', '？']:
-        text += '.'
-        
-    sentences = re.split('([。.!?！？])', text)
-    sentences = [''.join(i) for i in zip(sentences[0::2], sentences[1::2])]
-    
-    batches = []
-    current_batch = ""
-    
-    def split_by_words(text):
-        words = text.split()
-        current_word_part = ""
-        word_batches = []
-        for word in words:
-            if len(current_word_part.encode('utf-8')) + len(word.encode('utf-8')) + 1 <= max_chars:
-                current_word_part += word + ' '
-            else:
-                if current_word_part:
-                    # Try to find a suitable split word
-                    for split_word in split_words:
-                        split_index = current_word_part.rfind(' ' + split_word + ' ')
-                        if split_index != -1:
-                            word_batches.append(current_word_part[:split_index].strip())
-                            current_word_part = current_word_part[split_index:].strip() + ' '
-                            break
-                    else:
-                        # If no suitable split word found, just append the current part
-                        word_batches.append(current_word_part.strip())
-                        current_word_part = ""
-                current_word_part += word + ' '
-        if current_word_part:
-            word_batches.append(current_word_part.strip())
-        return word_batches
-
-    for sentence in sentences:
-        if len(current_batch.encode('utf-8')) + len(sentence.encode('utf-8')) <= max_chars:
-            current_batch += sentence
-        else:
-            # If adding this sentence would exceed the limit
-            if current_batch:
-                batches.append(current_batch)
-                current_batch = ""
-            
-            # If the sentence itself is longer than max_chars, split it
-            if len(sentence.encode('utf-8')) > max_chars:
-                # First, try to split by colon
-                colon_parts = sentence.split(':')
-                if len(colon_parts) > 1:
-                    for part in colon_parts:
-                        if len(part.encode('utf-8')) <= max_chars:
-                            batches.append(part)
-                        else:
-                            # If colon part is still too long, split by comma
-                            comma_parts = re.split('[,，]', part)
-                            if len(comma_parts) > 1:
-                                current_comma_part = ""
-                                for comma_part in comma_parts:
-                                    if len(current_comma_part.encode('utf-8')) + len(comma_part.encode('utf-8')) <= max_chars:
-                                        current_comma_part += comma_part + ','
-                                    else:
-                                        if current_comma_part:
-                                            batches.append(current_comma_part.rstrip(','))
-                                        current_comma_part = comma_part + ','
-                                if current_comma_part:
-                                    batches.append(current_comma_part.rstrip(','))
-                            else:
-                                # If no comma, split by words
-                                batches.extend(split_by_words(part))
-                else:
-                    # If no colon, split by comma
-                    comma_parts = re.split('[,，]', sentence)
-                    if len(comma_parts) > 1:
-                        current_comma_part = ""
-                        for comma_part in comma_parts:
-                            if len(current_comma_part.encode('utf-8')) + len(comma_part.encode('utf-8')) <= max_chars:
-                                current_comma_part += comma_part + ','
-                            else:
-                                if current_comma_part:
-                                    batches.append(current_comma_part.rstrip(','))
-                                current_comma_part = comma_part + ','
-                        if current_comma_part:
-                            batches.append(current_comma_part.rstrip(','))
-                    else:
-                        # If no comma, split by words
-                        batches.extend(split_by_words(sentence))
-            else:
-                current_batch = sentence
-    
-    if current_batch:
-        batches.append(current_batch)
-    
-    return batches
-
-def infer_batch(ref_audio, ref_text, gen_text_batches, exp_name, remove_silence, progress=gr.Progress()):
-    if exp_name == "F5-TTS":
-        ema_model = F5TTS_ema_model
-    elif exp_name == "E2-TTS":
-        ema_model = E2TTS_ema_model
-
-    audio, sr = ref_audio
-    if audio.shape[0] > 1:
-        audio = torch.mean(audio, dim=0, keepdim=True)
-
-    rms = torch.sqrt(torch.mean(torch.square(audio)))
-    if rms < target_rms:
-        audio = audio * target_rms / rms
-    if sr != target_sample_rate:
-        resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
-        audio = resampler(audio)
-    audio = audio.to(device)
-
-    generated_waves = []
-    spectrograms = []
-
-    for i, gen_text in enumerate(progress.tqdm(gen_text_batches)):
-        # Prepare the text
-        if len(ref_text[-1].encode('utf-8')) == 1:
-            ref_text = ref_text + " "
-        text_list = [ref_text + gen_text]
-        final_text_list = convert_char_to_pinyin(text_list)
-
-        # Calculate duration
-        ref_audio_len = audio.shape[-1] // hop_length
-        zh_pause_punc = r"。，、；：？！"
-        ref_text_len = len(ref_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, ref_text))
-        gen_text_len = len(gen_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, gen_text))
-        duration = ref_audio_len + int(ref_audio_len / ref_text_len * gen_text_len / speed)
-
-        # inference
-        with torch.inference_mode():
-            generated, _ = ema_model.sample(
-                cond=audio,
-                text=final_text_list,
-                duration=duration,
-                steps=nfe_step,
-                cfg_strength=cfg_strength,
-                sway_sampling_coef=sway_sampling_coef,
-            )
-
-        generated = generated[:, ref_audio_len:, :]
-        generated_mel_spec = rearrange(generated, "1 n d -> 1 d n")
-        generated_wave = vocos.decode(generated_mel_spec.cpu())
-        if rms < target_rms:
-            generated_wave = generated_wave * rms / target_rms
-
-        # wav -> numpy
-        generated_wave = generated_wave.squeeze().cpu().numpy()
-        
-        generated_waves.append(generated_wave)
-        spectrograms.append(generated_mel_spec[0].cpu().numpy())
-
-    # Combine all generated waves
-    final_wave = np.concatenate(generated_waves)
-
-    # Remove silence
-    if remove_silence:
-        with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
-            sf.write(f.name, final_wave, target_sample_rate)
-            aseg = AudioSegment.from_file(f.name)
-            non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
-            non_silent_wave = AudioSegment.silent(duration=0)
-            for non_silent_seg in non_silent_segs:
-                non_silent_wave += non_silent_seg
-            aseg = non_silent_wave
-            aseg.export(f.name, format="wav")
-            final_wave, _ = torchaudio.load(f.name)
-        final_wave = final_wave.squeeze().cpu().numpy()
-
-    # Create a combined spectrogram
-    combined_spectrogram = np.concatenate(spectrograms, axis=1)
-    
-    with tempfile.NamedTemporaryFile(suffix=".png", delete=False) as tmp_spectrogram:
-        spectrogram_path = tmp_spectrogram.name
-        save_spectrogram(combined_spectrogram, spectrogram_path)
-
-    return (target_sample_rate, final_wave), spectrogram_path
-
-def infer(ref_audio_orig, ref_text, gen_text, exp_name, remove_silence, custom_split_words=''):
-    if not custom_split_words.strip():
-        custom_words = [word.strip() for word in custom_split_words.split(',')]
-        global SPLIT_WORDS
-        SPLIT_WORDS = custom_words
-
-    print(gen_text)
-
-    gr.Info("Converting audio...")
-    with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
-        aseg = AudioSegment.from_file(ref_audio_orig)
-
-        non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
-        non_silent_wave = AudioSegment.silent(duration=0)
-        for non_silent_seg in non_silent_segs:
-            non_silent_wave += non_silent_seg
-        aseg = non_silent_wave
-
-        audio_duration = len(aseg)
-        if audio_duration > 15000:
-            gr.Warning("Audio is over 15s, clipping to only first 15s.")
-            aseg = aseg[:15000]
-        aseg.export(f.name, format="wav")
-        ref_audio = f.name
-
-    if not ref_text.strip():
-        gr.Info("No reference text provided, transcribing reference audio...")
-        ref_text = pipe(
-            ref_audio,
-            chunk_length_s=30,
-            batch_size=128,
-            generate_kwargs={"task": "transcribe"},
-            return_timestamps=False,
-        )["text"].strip()
-        gr.Info("Finished transcription")
-    else:
-        gr.Info("Using custom reference text...")
-
-    # Split the input text into batches
-    audio, sr = torchaudio.load(ref_audio)
-    max_chars = int(len(ref_text.encode('utf-8')) / (audio.shape[-1] / sr) * (30 - audio.shape[-1] / sr))
-    gen_text_batches = split_text_into_batches(gen_text, max_chars=max_chars)
-    print('ref_text', ref_text)
-    for i, gen_text in enumerate(gen_text_batches):
-        print(f'gen_text {i}', gen_text)
-    
-    gr.Info(f"Generating audio using {exp_name} in {len(gen_text_batches)} batches")
-    return infer_batch((audio, sr), ref_text, gen_text_batches, exp_name, remove_silence)
-    
-def generate_podcast(script, speaker1_name, ref_audio1, ref_text1, speaker2_name, ref_audio2, ref_text2, exp_name, remove_silence):
-    # Split the script into speaker blocks
-    speaker_pattern = re.compile(f"^({re.escape(speaker1_name)}|{re.escape(speaker2_name)}):", re.MULTILINE)
-    speaker_blocks = speaker_pattern.split(script)[1:]  # Skip the first empty element
-    
-    generated_audio_segments = []
-    
-    for i in range(0, len(speaker_blocks), 2):
-        speaker = speaker_blocks[i]
-        text = speaker_blocks[i+1].strip()
-        
-        # Determine which speaker is talking
-        if speaker == speaker1_name:
-            ref_audio = ref_audio1
-            ref_text = ref_text1
-        elif speaker == speaker2_name:
-            ref_audio = ref_audio2
-            ref_text = ref_text2
-        else:
-            continue  # Skip if the speaker is neither speaker1 nor speaker2
-        
-        # Generate audio for this block
-        audio, _ = infer(ref_audio, ref_text, text, exp_name, remove_silence)
-        
-        # Convert the generated audio to a numpy array
-        sr, audio_data = audio
-        
-        # Save the audio data as a WAV file
-        with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file:
-            sf.write(temp_file.name, audio_data, sr)
-            audio_segment = AudioSegment.from_wav(temp_file.name)
-        
-        generated_audio_segments.append(audio_segment)
-        
-        # Add a short pause between speakers
-        pause = AudioSegment.silent(duration=500)  # 500ms pause
-        generated_audio_segments.append(pause)
-    
-    # Concatenate all audio segments
-    final_podcast = sum(generated_audio_segments)
-    
-    # Export the final podcast
-    with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file:
-        podcast_path = temp_file.name
-        final_podcast.export(podcast_path, format="wav")
-    
-    return podcast_path
-
-def parse_speechtypes_text(gen_text):
-    # Pattern to find (Emotion)
-    pattern = r'\((.*?)\)'
-
-    # Split the text by the pattern
-    tokens = re.split(pattern, gen_text)
-
-    segments = []
-
-    current_emotion = 'Regular'
-
-    for i in range(len(tokens)):
-        if i % 2 == 0:
-            # This is text
-            text = tokens[i].strip()
-            if text:
-                segments.append({'emotion': current_emotion, 'text': text})
-        else:
-            # This is emotion
-            emotion = tokens[i].strip()
-            current_emotion = emotion
-
-    return segments
-
-def update_speed(new_speed):
-    global speed
-    speed = new_speed
-    return f"Speed set to: {speed}"
-
-with gr.Blocks() as app_credits:
-    gr.Markdown("""
-# Credits
-
-* [mrfakename](https://github.com/fakerybakery) for the original [online demo](https://huggingface.co/spaces/mrfakename/E2-F5-TTS)
-* [RootingInLoad](https://github.com/RootingInLoad) for the podcast generation
-""")
-with gr.Blocks() as app_tts:
-    gr.Markdown("# Batched TTS")
-    ref_audio_input = gr.Audio(label="Reference Audio", type="filepath")
-    gen_text_input = gr.Textbox(label="Text to Generate", lines=10)
-    model_choice = gr.Radio(
-        choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS"
-    )
-    generate_btn = gr.Button("Synthesize", variant="primary")
-    with gr.Accordion("Advanced Settings", open=False):
-        ref_text_input = gr.Textbox(
-            label="Reference Text",
-            info="Leave blank to automatically transcribe the reference audio. If you enter text it will override automatic transcription.",
-            lines=2,
-        )
-        remove_silence = gr.Checkbox(
-            label="Remove Silences",
-            info="The model tends to produce silences, especially on longer audio. We can manually remove silences if needed. Note that this is an experimental feature and may produce strange results. This will also increase generation time.",
-            value=True,
-        )
-        split_words_input = gr.Textbox(
-            label="Custom Split Words",
-            info="Enter custom words to split on, separated by commas. Leave blank to use default list.",
-            lines=2,
-        )
-        speed_slider = gr.Slider(
-            label="Speed",
-            minimum=0.3,
-            maximum=2.0,
-            value=speed,
-            step=0.1,
-            info="Adjust the speed of the audio.",
-        )
-    speed_slider.change(update_speed, inputs=speed_slider)
-
-    audio_output = gr.Audio(label="Synthesized Audio")
-    spectrogram_output = gr.Image(label="Spectrogram")
-
-    generate_btn.click(
-        infer,
-        inputs=[
-            ref_audio_input,
-            ref_text_input,
-            gen_text_input,
-            model_choice,
-            remove_silence,
-            split_words_input,
-        ],
-        outputs=[audio_output, spectrogram_output],
-    )
-    
-with gr.Blocks() as app_podcast:
-    gr.Markdown("# Podcast Generation")
-    speaker1_name = gr.Textbox(label="Speaker 1 Name")
-    ref_audio_input1 = gr.Audio(label="Reference Audio (Speaker 1)", type="filepath")
-    ref_text_input1 = gr.Textbox(label="Reference Text (Speaker 1)", lines=2)
-    
-    speaker2_name = gr.Textbox(label="Speaker 2 Name")
-    ref_audio_input2 = gr.Audio(label="Reference Audio (Speaker 2)", type="filepath")
-    ref_text_input2 = gr.Textbox(label="Reference Text (Speaker 2)", lines=2)
-    
-    script_input = gr.Textbox(label="Podcast Script", lines=10, 
-                                placeholder="Enter the script with speaker names at the start of each block, e.g.:\nSean: How did you start studying...\n\nMeghan: I came to my interest in technology...\nIt was a long journey...\n\nSean: That's fascinating. Can you elaborate...")
-    
-    podcast_model_choice = gr.Radio(
-        choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS"
-    )
-    podcast_remove_silence = gr.Checkbox(
-        label="Remove Silences",
-        value=True,
-    )
-    generate_podcast_btn = gr.Button("Generate Podcast", variant="primary")
-    podcast_output = gr.Audio(label="Generated Podcast")
-
-    def podcast_generation(script, speaker1, ref_audio1, ref_text1, speaker2, ref_audio2, ref_text2, model, remove_silence):
-        return generate_podcast(script, speaker1, ref_audio1, ref_text1, speaker2, ref_audio2, ref_text2, model, remove_silence)
-
-    generate_podcast_btn.click(
-        podcast_generation,
-        inputs=[
-            script_input,
-            speaker1_name,
-            ref_audio_input1,
-            ref_text_input1,
-            speaker2_name,
-            ref_audio_input2,
-            ref_text_input2,
-            podcast_model_choice,
-            podcast_remove_silence,
-        ],
-        outputs=podcast_output,
-    )
-
-def parse_emotional_text(gen_text):
-    # Pattern to find (Emotion)
-    pattern = r'\((.*?)\)'
-
-    # Split the text by the pattern
-    tokens = re.split(pattern, gen_text)
-
-    segments = []
-
-    current_emotion = 'Regular'
-
-    for i in range(len(tokens)):
-        if i % 2 == 0:
-            # This is text
-            text = tokens[i].strip()
-            if text:
-                segments.append({'emotion': current_emotion, 'text': text})
-        else:
-            # This is emotion
-            emotion = tokens[i].strip()
-            current_emotion = emotion
-
-    return segments
-
-with gr.Blocks() as app_emotional:
-    # New section for emotional generation
-    gr.Markdown(
-        """
-    # Multiple Speech-Type Generation
-
-    This section allows you to upload different audio clips for each speech type. 'Regular' emotion is mandatory. You can add additional speech types by clicking the "Add Speech Type" button. Enter your text in the format shown below, and the system will generate speech using the appropriate emotions. If unspecified, the model will use the regular speech type. The current speech type will be used until the next speech type is specified.
-
-    **Example Input:**
-
-    (Regular) Hello, I'd like to order a sandwich please. (Surprised) What do you mean you're out of bread? (Sad) I really wanted a sandwich though... (Angry) You know what, darn you and your little shop, you suck! (Whisper) I'll just go back home and cry now. (Shouting) Why me?!
-    """
-    )
-
-    gr.Markdown("Upload different audio clips for each speech type. 'Regular' emotion is mandatory. You can add additional speech types by clicking the 'Add Speech Type' button.")
-
-    # Regular speech type (mandatory)
-    with gr.Row():
-        regular_name = gr.Textbox(value='Regular', label='Speech Type Name', interactive=False)
-        regular_audio = gr.Audio(label='Regular Reference Audio', type='filepath')
-        regular_ref_text = gr.Textbox(label='Reference Text (Regular)', lines=2)
-
-    # Additional speech types (up to 9 more)
-    max_speech_types = 10
-    speech_type_names = []
-    speech_type_audios = []
-    speech_type_ref_texts = []
-    speech_type_delete_btns = []
-
-    for i in range(max_speech_types - 1):
-        with gr.Row():
-            name_input = gr.Textbox(label='Speech Type Name', visible=False)
-            audio_input = gr.Audio(label='Reference Audio', type='filepath', visible=False)
-            ref_text_input = gr.Textbox(label='Reference Text', lines=2, visible=False)
-            delete_btn = gr.Button("Delete", variant="secondary", visible=False)
-        speech_type_names.append(name_input)
-        speech_type_audios.append(audio_input)
-        speech_type_ref_texts.append(ref_text_input)
-        speech_type_delete_btns.append(delete_btn)
-
-    # Button to add speech type
-    add_speech_type_btn = gr.Button("Add Speech Type")
-
-    # Keep track of current number of speech types
-    speech_type_count = gr.State(value=0)
-
-    # Function to add a speech type
-    def add_speech_type_fn(speech_type_count):
-        if speech_type_count < max_speech_types - 1:
-            speech_type_count += 1
-            # Prepare updates for the components
-            name_updates = []
-            audio_updates = []
-            ref_text_updates = []
-            delete_btn_updates = []
-            for i in range(max_speech_types - 1):
-                if i < speech_type_count:
-                    name_updates.append(gr.update(visible=True))
-                    audio_updates.append(gr.update(visible=True))
-                    ref_text_updates.append(gr.update(visible=True))
-                    delete_btn_updates.append(gr.update(visible=True))
-                else:
-                    name_updates.append(gr.update())
-                    audio_updates.append(gr.update())
-                    ref_text_updates.append(gr.update())
-                    delete_btn_updates.append(gr.update())
-        else:
-            # Optionally, show a warning
-            # gr.Warning("Maximum number of speech types reached.")
-            name_updates = [gr.update() for _ in range(max_speech_types - 1)]
-            audio_updates = [gr.update() for _ in range(max_speech_types - 1)]
-            ref_text_updates = [gr.update() for _ in range(max_speech_types - 1)]
-            delete_btn_updates = [gr.update() for _ in range(max_speech_types - 1)]
-        return [speech_type_count] + name_updates + audio_updates + ref_text_updates + delete_btn_updates
-
-    add_speech_type_btn.click(
-        add_speech_type_fn,
-        inputs=speech_type_count,
-        outputs=[speech_type_count] + speech_type_names + speech_type_audios + speech_type_ref_texts + speech_type_delete_btns
-    )
-
-    # Function to delete a speech type
-    def make_delete_speech_type_fn(index):
-        def delete_speech_type_fn(speech_type_count):
-            # Prepare updates
-            name_updates = []
-            audio_updates = []
-            ref_text_updates = []
-            delete_btn_updates = []
-
-            for i in range(max_speech_types - 1):
-                if i == index:
-                    name_updates.append(gr.update(visible=False, value=''))
-                    audio_updates.append(gr.update(visible=False, value=None))
-                    ref_text_updates.append(gr.update(visible=False, value=''))
-                    delete_btn_updates.append(gr.update(visible=False))
-                else:
-                    name_updates.append(gr.update())
-                    audio_updates.append(gr.update())
-                    ref_text_updates.append(gr.update())
-                    delete_btn_updates.append(gr.update())
-
-            speech_type_count = max(0, speech_type_count - 1)
-
-            return [speech_type_count] + name_updates + audio_updates + ref_text_updates + delete_btn_updates
-
-        return delete_speech_type_fn
-
-    for i, delete_btn in enumerate(speech_type_delete_btns):
-        delete_fn = make_delete_speech_type_fn(i)
-        delete_btn.click(
-            delete_fn,
-            inputs=speech_type_count,
-            outputs=[speech_type_count] + speech_type_names + speech_type_audios + speech_type_ref_texts + speech_type_delete_btns
-        )
-
-    # Text input for the prompt
-    gen_text_input_emotional = gr.Textbox(label="Text to Generate", lines=10)
-
-    # Model choice
-    model_choice_emotional = gr.Radio(
-        choices=["F5-TTS", "E2-TTS"], label="Choose TTS Model", value="F5-TTS"
-    )
-
-    with gr.Accordion("Advanced Settings", open=False):
-        remove_silence_emotional = gr.Checkbox(
-            label="Remove Silences",
-            value=True,
-        )
-
-    # Generate button
-    generate_emotional_btn = gr.Button("Generate Emotional Speech", variant="primary")
-
-    # Output audio
-    audio_output_emotional = gr.Audio(label="Synthesized Audio")
-
-    def generate_emotional_speech(
-        regular_audio,
-        regular_ref_text,
-        gen_text,
-        *args,
-    ):
-        num_additional_speech_types = max_speech_types - 1
-        speech_type_names_list = args[:num_additional_speech_types]
-        speech_type_audios_list = args[num_additional_speech_types:2 * num_additional_speech_types]
-        speech_type_ref_texts_list = args[2 * num_additional_speech_types:3 * num_additional_speech_types]
-        model_choice = args[3 * num_additional_speech_types]
-        remove_silence = args[3 * num_additional_speech_types + 1]
-
-        # Collect the speech types and their audios into a dict
-        speech_types = {'Regular': {'audio': regular_audio, 'ref_text': regular_ref_text}}
-
-        for name_input, audio_input, ref_text_input in zip(speech_type_names_list, speech_type_audios_list, speech_type_ref_texts_list):
-            if name_input and audio_input:
-                speech_types[name_input] = {'audio': audio_input, 'ref_text': ref_text_input}
-
-        # Parse the gen_text into segments
-        segments = parse_speechtypes_text(gen_text)
-
-        # For each segment, generate speech
-        generated_audio_segments = []
-        current_emotion = 'Regular'
-
-        for segment in segments:
-            emotion = segment['emotion']
-            text = segment['text']
-
-            if emotion in speech_types:
-                current_emotion = emotion
-            else:
-                # If emotion not available, default to Regular
-                current_emotion = 'Regular'
-
-            ref_audio = speech_types[current_emotion]['audio']
-            ref_text = speech_types[current_emotion].get('ref_text', '')
-
-            # Generate speech for this segment
-            audio, _ = infer(ref_audio, ref_text, text, model_choice, remove_silence, "")
-            sr, audio_data = audio
-
-            generated_audio_segments.append(audio_data)
-
-        # Concatenate all audio segments
-        if generated_audio_segments:
-            final_audio_data = np.concatenate(generated_audio_segments)
-            return (sr, final_audio_data)
-        else:
-            gr.Warning("No audio generated.")
-            return None
-
-    generate_emotional_btn.click(
-        generate_emotional_speech,
-        inputs=[
-            regular_audio,
-            regular_ref_text,
-            gen_text_input_emotional,
-        ] + speech_type_names + speech_type_audios + speech_type_ref_texts + [
-            model_choice_emotional,
-            remove_silence_emotional,
-        ],
-        outputs=audio_output_emotional,
-    )
-
-    # Validation function to disable Generate button if speech types are missing
-    def validate_speech_types(
-        gen_text,
-        regular_name,
-        *args
-    ):
-        num_additional_speech_types = max_speech_types - 1
-        speech_type_names_list = args[:num_additional_speech_types]
-
-        # Collect the speech types names
-        speech_types_available = set()
-        if regular_name:
-            speech_types_available.add(regular_name)
-        for name_input in speech_type_names_list:
-            if name_input:
-                speech_types_available.add(name_input)
-
-        # Parse the gen_text to get the speech types used
-        segments = parse_emotional_text(gen_text)
-        speech_types_in_text = set(segment['emotion'] for segment in segments)
-
-        # Check if all speech types in text are available
-        missing_speech_types = speech_types_in_text - speech_types_available
-
-        if missing_speech_types:
-            # Disable the generate button
-            return gr.update(interactive=False)
-        else:
-            # Enable the generate button
-            return gr.update(interactive=True)
-
-    gen_text_input_emotional.change(
-        validate_speech_types,
-        inputs=[gen_text_input_emotional, regular_name] + speech_type_names,
-        outputs=generate_emotional_btn
-    )
-with gr.Blocks() as app:
-    gr.Markdown(
-        """
-# E2/F5 TTS
-
-This is a local web UI for F5 TTS with advanced batch processing support. This app supports the following TTS models:
-
-* [F5-TTS](https://arxiv.org/abs/2410.06885) (A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching)
-* [E2 TTS](https://arxiv.org/abs/2406.18009) (Embarrassingly Easy Fully Non-Autoregressive Zero-Shot TTS)
-
-The checkpoints support English and Chinese.
-
-If you're having issues, try converting your reference audio to WAV or MP3, clipping it to 15s, and shortening your prompt.
-
-**NOTE: Reference text will be automatically transcribed with Whisper if not provided. For best results, keep your reference clips short (<15s). Ensure the audio is fully uploaded before generating.**
-"""
-    )
-    gr.TabbedInterface([app_tts, app_podcast, app_emotional, app_credits], ["TTS", "Podcast", "Multi-Style", "Credits"])
-
-@click.command()
-@click.option("--port", "-p", default=None, type=int, help="Port to run the app on")
-@click.option("--host", "-H", default=None, help="Host to run the app on")
-@click.option(
-    "--share",
-    "-s",
-    default=False,
-    is_flag=True,
-    help="Share the app via Gradio share link",
-)
-@click.option("--api", "-a", default=True, is_flag=True, help="Allow API access")
-def main(port, host, share, api):
-    global app
-    print(f"Starting app...")
-    app.queue(api_open=api).launch(
-        server_name=host, server_port=port, share=share, show_api=api
-    )
-
-
-if __name__ == "__main__":
-    main()

inference-cli.py

@@ -1,22 +1,24 @@
 import argparse
 import codecs
 import re
+import tempfile
 from pathlib import Path
 
 import numpy as np
 import soundfile as sf
 import tomli
+import torch
+import torchaudio
+import tqdm
 from cached_path import cached_path
+from einops import rearrange
+from pydub import AudioSegment, silence
+from transformers import pipeline
+from vocos import Vocos
 
-from model import DiT, UNetT
-from model.utils_infer import (
-    load_vocoder,
-    load_model,
-    preprocess_ref_audio_text,
-    infer_process,
-    remove_silence_for_generated_wav,
-)
-
+from model import CFM, DiT, MMDiT, UNetT
+from model.utils import (convert_char_to_pinyin, get_tokenizer,
+                         load_checkpoint, save_spectrogram)
 
 parser = argparse.ArgumentParser(
     prog="python3 inference-cli.py",
@@ -35,17 +37,18 @@ parser.add_argument(
     help="F5-TTS | E2-TTS",
 )
 parser.add_argument(
-    "-p",
-    "--ckpt_file",
-    help="The Checkpoint .pt",
+    "-r",
+    "--ref_audio",
+    type=str,
+    help="Reference audio file < 15 seconds."
 )
 parser.add_argument(
-    "-v",
-    "--vocab_file",
-    help="The vocab .txt",
+    "-s",
+    "--ref_text",
+    type=str,
+    default="666",
+    help="Subtitle for the reference audio."
 )
-parser.add_argument("-r", "--ref_audio", type=str, help="Reference audio file < 15 seconds.")
-parser.add_argument("-s", "--ref_text", type=str, default="666", help="Subtitle for the reference audio.")
 parser.add_argument(
     "-t",
     "--gen_text",
@@ -85,86 +88,305 @@ if gen_file:
     gen_text = codecs.open(gen_file, "r", "utf-8").read()
 output_dir = args.output_dir if args.output_dir else config["output_dir"]
 model = args.model if args.model else config["model"]
-ckpt_file = args.ckpt_file if args.ckpt_file else ""
-vocab_file = args.vocab_file if args.vocab_file else ""
 remove_silence = args.remove_silence if args.remove_silence else config["remove_silence"]
-wave_path = Path(output_dir) / "out.wav"
-spectrogram_path = Path(output_dir) / "out.png"
+wave_path = Path(output_dir)/"out.wav"
+spectrogram_path = Path(output_dir)/"out.png"
 vocos_local_path = "../checkpoints/charactr/vocos-mel-24khz"
 
-vocos = load_vocoder(is_local=args.load_vocoder_from_local, local_path=vocos_local_path)
+device = (
+    "cuda"
+    if torch.cuda.is_available()
+    else "mps" if torch.backends.mps.is_available() else "cpu"
+)
+
+if args.load_vocoder_from_local:
+    print(f"Load vocos from local path {vocos_local_path}")
+    vocos = Vocos.from_hparams(f"{vocos_local_path}/config.yaml")
+    state_dict = torch.load(f"{vocos_local_path}/pytorch_model.bin", map_location=device)
+    vocos.load_state_dict(state_dict)
+    vocos.eval()
+else:
+    print("Donwload Vocos from huggingface charactr/vocos-mel-24khz")
+    vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
+
+print(f"Using {device} device")
+
+# --------------------- Settings -------------------- #
+
+target_sample_rate = 24000
+n_mel_channels = 100
+hop_length = 256
+target_rms = 0.1
+nfe_step = 32  # 16, 32
+cfg_strength = 2.0
+ode_method = "euler"
+sway_sampling_coef = -1.0
+speed = 1.0
+# fix_duration = 27  # None or float (duration in seconds)
+fix_duration = None
+
+def load_model(repo_name, exp_name, model_cls, model_cfg, ckpt_step):
+    ckpt_path = f"ckpts/{exp_name}/model_{ckpt_step}.pt" # .pt | .safetensors
+    if not Path(ckpt_path).exists():
+        ckpt_path = str(cached_path(f"hf://SWivid/{repo_name}/{exp_name}/model_{ckpt_step}.safetensors"))
+    vocab_char_map, vocab_size = get_tokenizer("Emilia_ZH_EN", "pinyin")
+    model = CFM(
+        transformer=model_cls(
+            **model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels
+        ),
+        mel_spec_kwargs=dict(
+            target_sample_rate=target_sample_rate,
+            n_mel_channels=n_mel_channels,
+            hop_length=hop_length,
+        ),
+        odeint_kwargs=dict(
+            method=ode_method,
+        ),
+        vocab_char_map=vocab_char_map,
+    ).to(device)
+
+    model = load_checkpoint(model, ckpt_path, device, use_ema = True)
+
+    return model
 
 
 # load models
-if model == "F5-TTS":
-    model_cls = DiT
-    model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
-    if ckpt_file == "":
-        repo_name = "F5-TTS"
-        exp_name = "F5TTS_Base"
-        ckpt_step = 1200000
-        ckpt_file = str(cached_path(f"hf://SWivid/{repo_name}/{exp_name}/model_{ckpt_step}.safetensors"))
-        # ckpt_file = f"ckpts/{exp_name}/model_{ckpt_step}.pt"  # .pt | .safetensors; local path
-
-elif model == "E2-TTS":
-    model_cls = UNetT
-    model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
-    if ckpt_file == "":
-        repo_name = "E2-TTS"
-        exp_name = "E2TTS_Base"
-        ckpt_step = 1200000
-        ckpt_file = str(cached_path(f"hf://SWivid/{repo_name}/{exp_name}/model_{ckpt_step}.safetensors"))
-        # ckpt_file = f"ckpts/{exp_name}/model_{ckpt_step}.pt"  # .pt | .safetensors; local path
-
-print(f"Using {model}...")
-ema_model = load_model(model_cls, model_cfg, ckpt_file, vocab_file)
-
-
-def main_process(ref_audio, ref_text, text_gen, model_obj, remove_silence):
-    main_voice = {"ref_audio": ref_audio, "ref_text": ref_text}
+F5TTS_model_cfg = dict(
+    dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4
+)
+E2TTS_model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
+
+
+def chunk_text(text, max_chars=135):
+    """
+    Splits the input text into chunks, each with a maximum number of characters.
+    Args:
+        text (str): The text to be split.
+        max_chars (int): The maximum number of characters per chunk.
+    Returns:
+        List[str]: A list of text chunks.
+    """
+    chunks = []
+    current_chunk = ""
+    # Split the text into sentences based on punctuation followed by whitespace
+    sentences = re.split(r'(?<=[;:,.!?])\s+|(?<=[；：，。！？])', text)
+
+    for sentence in sentences:
+        if len(current_chunk.encode('utf-8')) + len(sentence.encode('utf-8')) <= max_chars:
+            current_chunk += sentence + " " if sentence and len(sentence[-1].encode('utf-8')) == 1 else sentence
+        else:
+            if current_chunk:
+                chunks.append(current_chunk.strip())
+            current_chunk = sentence + " " if sentence and len(sentence[-1].encode('utf-8')) == 1 else sentence
+
+    if current_chunk:
+        chunks.append(current_chunk.strip())
+
+    return chunks
+
+
+def infer_batch(ref_audio, ref_text, gen_text_batches, model, remove_silence, cross_fade_duration=0.15):
+    if model == "F5-TTS":
+        ema_model = load_model(model, "F5TTS_Base", DiT, F5TTS_model_cfg, 1200000)
+    elif model == "E2-TTS":
+        ema_model = load_model(model, "E2TTS_Base", UNetT, E2TTS_model_cfg, 1200000)
+
+    audio, sr = ref_audio
+    if audio.shape[0] > 1:
+        audio = torch.mean(audio, dim=0, keepdim=True)
+
+    rms = torch.sqrt(torch.mean(torch.square(audio)))
+    if rms < target_rms:
+        audio = audio * target_rms / rms
+    if sr != target_sample_rate:
+        resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
+        audio = resampler(audio)
+    audio = audio.to(device)
+
+    generated_waves = []
+    spectrograms = []
+
+    for i, gen_text in enumerate(tqdm.tqdm(gen_text_batches)):
+        # Prepare the text
+        if len(ref_text[-1].encode('utf-8')) == 1:
+            ref_text = ref_text + " "
+        text_list = [ref_text + gen_text]
+        final_text_list = convert_char_to_pinyin(text_list)
+
+        # Calculate duration
+        ref_audio_len = audio.shape[-1] // hop_length
+        zh_pause_punc = r"。，、；：？！"
+        ref_text_len = len(ref_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, ref_text))
+        gen_text_len = len(gen_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, gen_text))
+        duration = ref_audio_len + int(ref_audio_len / ref_text_len * gen_text_len / speed)
+
+        # inference
+        with torch.inference_mode():
+            generated, _ = ema_model.sample(
+                cond=audio,
+                text=final_text_list,
+                duration=duration,
+                steps=nfe_step,
+                cfg_strength=cfg_strength,
+                sway_sampling_coef=sway_sampling_coef,
+            )
+
+        generated = generated[:, ref_audio_len:, :]
+        generated_mel_spec = rearrange(generated, "1 n d -> 1 d n")
+        generated_wave = vocos.decode(generated_mel_spec.cpu())
+        if rms < target_rms:
+            generated_wave = generated_wave * rms / target_rms
+
+        # wav -> numpy
+        generated_wave = generated_wave.squeeze().cpu().numpy()
+        
+        generated_waves.append(generated_wave)
+        spectrograms.append(generated_mel_spec[0].cpu().numpy())
+
+    # Combine all generated waves with cross-fading
+    if cross_fade_duration <= 0:
+        # Simply concatenate
+        final_wave = np.concatenate(generated_waves)
+    else:
+        final_wave = generated_waves[0]
+        for i in range(1, len(generated_waves)):
+            prev_wave = final_wave
+            next_wave = generated_waves[i]
+
+            # Calculate cross-fade samples, ensuring it does not exceed wave lengths
+            cross_fade_samples = int(cross_fade_duration * target_sample_rate)
+            cross_fade_samples = min(cross_fade_samples, len(prev_wave), len(next_wave))
+
+            if cross_fade_samples <= 0:
+                # No overlap possible, concatenate
+                final_wave = np.concatenate([prev_wave, next_wave])
+                continue
+
+            # Overlapping parts
+            prev_overlap = prev_wave[-cross_fade_samples:]
+            next_overlap = next_wave[:cross_fade_samples]
+
+            # Fade out and fade in
+            fade_out = np.linspace(1, 0, cross_fade_samples)
+            fade_in = np.linspace(0, 1, cross_fade_samples)
+
+            # Cross-faded overlap
+            cross_faded_overlap = prev_overlap * fade_out + next_overlap * fade_in
+
+            # Combine
+            new_wave = np.concatenate([
+                prev_wave[:-cross_fade_samples],
+                cross_faded_overlap,
+                next_wave[cross_fade_samples:]
+            ])
+
+            final_wave = new_wave
+
+    # Create a combined spectrogram
+    combined_spectrogram = np.concatenate(spectrograms, axis=1)
+
+    return final_wave, combined_spectrogram
+
+def process_voice(ref_audio_orig, ref_text):
+    print("Converting audio...")
+    with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
+        aseg = AudioSegment.from_file(ref_audio_orig)
+
+        non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=1000)
+        non_silent_wave = AudioSegment.silent(duration=0)
+        for non_silent_seg in non_silent_segs:
+            non_silent_wave += non_silent_seg
+        aseg = non_silent_wave
+
+        audio_duration = len(aseg)
+        if audio_duration > 15000:
+            print("Audio is over 15s, clipping to only first 15s.")
+            aseg = aseg[:15000]
+        aseg.export(f.name, format="wav")
+        ref_audio = f.name
+
+    if not ref_text.strip():
+        print("No reference text provided, transcribing reference audio...")
+        pipe = pipeline(
+            "automatic-speech-recognition",
+            model="openai/whisper-large-v3-turbo",
+            torch_dtype=torch.float16,
+            device=device,
+        )
+        ref_text = pipe(
+            ref_audio,
+            chunk_length_s=30,
+            batch_size=128,
+            generate_kwargs={"task": "transcribe"},
+            return_timestamps=False,
+        )["text"].strip()
+        print("Finished transcription")
+    else:
+        print("Using custom reference text...")
+    return ref_audio, ref_text    
+
+def infer(ref_audio, ref_text, gen_text, model, remove_silence, cross_fade_duration=0.15):
+    print(gen_text)
+    # Add the functionality to ensure it ends with ". "
+    if not ref_text.endswith(". ") and not ref_text.endswith("。"):
+        if ref_text.endswith("."):
+            ref_text += " "
+        else:
+            ref_text += ". "
+
+    # Split the input text into batches
+    audio, sr = torchaudio.load(ref_audio)
+    max_chars = int(len(ref_text.encode('utf-8')) / (audio.shape[-1] / sr) * (25 - audio.shape[-1] / sr))
+    gen_text_batches = chunk_text(gen_text, max_chars=max_chars)
+    print('ref_text', ref_text)
+    for i, gen_text in enumerate(gen_text_batches):
+        print(f'gen_text {i}', gen_text)
+    
+    print(f"Generating audio using {model} in {len(gen_text_batches)} batches, loading models...")
+    return infer_batch((audio, sr), ref_text, gen_text_batches, model, remove_silence, cross_fade_duration)
+    
+
+def process(ref_audio, ref_text, text_gen, model, remove_silence):
+    main_voice = {"ref_audio":ref_audio, "ref_text":ref_text}
     if "voices" not in config:
         voices = {"main": main_voice}
     else:
         voices = config["voices"]
         voices["main"] = main_voice
     for voice in voices:
-        voices[voice]["ref_audio"], voices[voice]["ref_text"] = preprocess_ref_audio_text(
-            voices[voice]["ref_audio"], voices[voice]["ref_text"]
-        )
-        print("Voice:", voice)
-        print("Ref_audio:", voices[voice]["ref_audio"])
-        print("Ref_text:", voices[voice]["ref_text"])
+        voices[voice]['ref_audio'], voices[voice]['ref_text'] = process_voice(voices[voice]['ref_audio'], voices[voice]['ref_text'])
 
     generated_audio_segments = []
-    reg1 = r"(?=\[\w+\])"
+    reg1 = r'(?=\[\w+\])'
     chunks = re.split(reg1, text_gen)
-    reg2 = r"\[(\w+)\]"
+    reg2 = r'\[(\w+)\]'
     for text in chunks:
         match = re.match(reg2, text)
-        if match:
-            voice = match[1]
-        else:
-            print("No voice tag found, using main.")
-            voice = "main"
-        if voice not in voices:
-            print(f"Voice {voice} not found, using main.")
+        if not match or voice not in voices:
             voice = "main"
+        else:
+            voice = match[1]
         text = re.sub(reg2, "", text)
         gen_text = text.strip()
-        ref_audio = voices[voice]["ref_audio"]
-        ref_text = voices[voice]["ref_text"]
+        ref_audio = voices[voice]['ref_audio']
+        ref_text = voices[voice]['ref_text']
         print(f"Voice: {voice}")
-        audio, final_sample_rate, spectragram = infer_process(ref_audio, ref_text, gen_text, model_obj)
+        audio, spectragram = infer(ref_audio, ref_text, gen_text, model, remove_silence)
         generated_audio_segments.append(audio)
 
     if generated_audio_segments:
         final_wave = np.concatenate(generated_audio_segments)
         with open(wave_path, "wb") as f:
-            sf.write(f.name, final_wave, final_sample_rate)
+            sf.write(f.name, final_wave, target_sample_rate)
             # Remove silence
             if remove_silence:
-                remove_silence_for_generated_wav(f.name)
+                aseg = AudioSegment.from_file(f.name)
+                non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
+                non_silent_wave = AudioSegment.silent(duration=0)
+                for non_silent_seg in non_silent_segs:
+                    non_silent_wave += non_silent_seg
+                aseg = non_silent_wave
+                aseg.export(f.name, format="wav")
             print(f.name)
 
-
-main_process(ref_audio, ref_text, gen_text, ema_model, remove_silence)
+process(ref_audio, ref_text, gen_text, model, remove_silence)

model/__init__.py

@@ -5,6 +5,3 @@ from model.backbones.dit import DiT
 from model.backbones.mmdit import MMDiT
 
 from model.trainer import Trainer
-
-
-__all__ = ["CFM", "UNetT", "DiT", "MMDiT", "Trainer"]

model/backbones/dit.py

@@ -13,6 +13,8 @@ import torch
 from torch import nn
 import torch.nn.functional as F
 
+from einops import repeat
+
 from x_transformers.x_transformers import RotaryEmbedding
 
 from model.modules import (
@@ -21,16 +23,14 @@ from model.modules import (
     ConvPositionEmbedding,
     DiTBlock,
     AdaLayerNormZero_Final,
-    precompute_freqs_cis,
-    get_pos_embed_indices,
+    precompute_freqs_cis, get_pos_embed_indices,
 )
 
 
 # Text embedding
 
-
 class TextEmbedding(nn.Module):
-    def __init__(self, text_num_embeds, text_dim, conv_layers=0, conv_mult=2):
+    def __init__(self, text_num_embeds, text_dim, conv_layers = 0, conv_mult = 2):
         super().__init__()
         self.text_embed = nn.Embedding(text_num_embeds + 1, text_dim)  # use 0 as filler token
 
@@ -38,22 +38,20 @@ class TextEmbedding(nn.Module):
             self.extra_modeling = True
             self.precompute_max_pos = 4096  # ~44s of 24khz audio
             self.register_buffer("freqs_cis", precompute_freqs_cis(text_dim, self.precompute_max_pos), persistent=False)
-            self.text_blocks = nn.Sequential(
-                *[ConvNeXtV2Block(text_dim, text_dim * conv_mult) for _ in range(conv_layers)]
-            )
+            self.text_blocks = nn.Sequential(*[ConvNeXtV2Block(text_dim, text_dim * conv_mult) for _ in range(conv_layers)])
         else:
             self.extra_modeling = False
 
-    def forward(self, text: int["b nt"], seq_len, drop_text=False):  # noqa: F722
+    def forward(self, text: int['b nt'], seq_len, drop_text = False):
+        batch, text_len = text.shape[0], text.shape[1]
         text = text + 1  # use 0 as filler token. preprocess of batch pad -1, see list_str_to_idx()
         text = text[:, :seq_len]  # curtail if character tokens are more than the mel spec tokens
-        batch, text_len = text.shape[0], text.shape[1]
-        text = F.pad(text, (0, seq_len - text_len), value=0)
+        text = F.pad(text, (0, seq_len - text_len), value = 0)
 
         if drop_text:  # cfg for text
             text = torch.zeros_like(text)
 
-        text = self.text_embed(text)  # b n -> b n d
+        text = self.text_embed(text) # b n -> b n d
 
         # possible extra modeling
         if self.extra_modeling:
@@ -71,91 +69,88 @@ class TextEmbedding(nn.Module):
 
 # noised input audio and context mixing embedding
 
-
 class InputEmbedding(nn.Module):
     def __init__(self, mel_dim, text_dim, out_dim):
         super().__init__()
         self.proj = nn.Linear(mel_dim * 2 + text_dim, out_dim)
-        self.conv_pos_embed = ConvPositionEmbedding(dim=out_dim)
+        self.conv_pos_embed = ConvPositionEmbedding(dim = out_dim)
 
-    def forward(self, x: float["b n d"], cond: float["b n d"], text_embed: float["b n d"], drop_audio_cond=False):  # noqa: F722
+    def forward(self, x: float['b n d'], cond: float['b n d'], text_embed: float['b n d'], drop_audio_cond = False):
         if drop_audio_cond:  # cfg for cond audio
             cond = torch.zeros_like(cond)
 
-        x = self.proj(torch.cat((x, cond, text_embed), dim=-1))
+        x = self.proj(torch.cat((x, cond, text_embed), dim = -1))
         x = self.conv_pos_embed(x) + x
         return x
-
+    
 
 # Transformer backbone using DiT blocks
 
-
 class DiT(nn.Module):
-    def __init__(
-        self,
-        *,
-        dim,
-        depth=8,
-        heads=8,
-        dim_head=64,
-        dropout=0.1,
-        ff_mult=4,
-        mel_dim=100,
-        text_num_embeds=256,
-        text_dim=None,
-        conv_layers=0,
-        long_skip_connection=False,
+    def __init__(self, *, 
+                 dim, depth = 8, heads = 8, dim_head = 64, dropout = 0.1, ff_mult = 4,
+                 mel_dim = 100, text_num_embeds = 256, text_dim = None, conv_layers = 0,
+                 long_skip_connection = False,
     ):
         super().__init__()
 
         self.time_embed = TimestepEmbedding(dim)
         if text_dim is None:
             text_dim = mel_dim
-        self.text_embed = TextEmbedding(text_num_embeds, text_dim, conv_layers=conv_layers)
+        self.text_embed = TextEmbedding(text_num_embeds, text_dim, conv_layers = conv_layers)
         self.input_embed = InputEmbedding(mel_dim, text_dim, dim)
 
         self.rotary_embed = RotaryEmbedding(dim_head)
 
         self.dim = dim
         self.depth = depth
-
+        
         self.transformer_blocks = nn.ModuleList(
-            [DiTBlock(dim=dim, heads=heads, dim_head=dim_head, ff_mult=ff_mult, dropout=dropout) for _ in range(depth)]
+            [
+                DiTBlock(
+                    dim = dim,
+                    heads = heads,
+                    dim_head = dim_head,
+                    ff_mult = ff_mult,
+                    dropout = dropout
+                )
+                for _ in range(depth)
+            ]
         )
-        self.long_skip_connection = nn.Linear(dim * 2, dim, bias=False) if long_skip_connection else None
-
+        self.long_skip_connection = nn.Linear(dim * 2, dim, bias = False) if long_skip_connection else None
+        
         self.norm_out = AdaLayerNormZero_Final(dim)  # final modulation
         self.proj_out = nn.Linear(dim, mel_dim)
 
     def forward(
         self,
-        x: float["b n d"],  # nosied input audio  # noqa: F722
-        cond: float["b n d"],  # masked cond audio  # noqa: F722
-        text: int["b nt"],  # text  # noqa: F722
-        time: float["b"] | float[""],  # time step  # noqa: F821 F722
+        x: float['b n d'],     # nosied input audio
+        cond: float['b n d'],  # masked cond audio
+        text: int['b nt'],     # text
+        time: float['b'] | float[''],  # time step
         drop_audio_cond,  # cfg for cond audio
-        drop_text,  # cfg for text
-        mask: bool["b n"] | None = None,  # noqa: F722
+        drop_text,        # cfg for text
+        mask: bool['b n'] | None = None,
     ):
         batch, seq_len = x.shape[0], x.shape[1]
         if time.ndim == 0:
-            time = time.repeat(batch)
-
+            time = repeat(time, ' -> b', b = batch)
+        
         # t: conditioning time, c: context (text + masked cond audio), x: noised input audio
         t = self.time_embed(time)
-        text_embed = self.text_embed(text, seq_len, drop_text=drop_text)
-        x = self.input_embed(x, cond, text_embed, drop_audio_cond=drop_audio_cond)
-
+        text_embed = self.text_embed(text, seq_len, drop_text = drop_text)
+        x = self.input_embed(x, cond, text_embed, drop_audio_cond = drop_audio_cond)
+        
         rope = self.rotary_embed.forward_from_seq_len(seq_len)
 
         if self.long_skip_connection is not None:
             residual = x
 
         for block in self.transformer_blocks:
-            x = block(x, t, mask=mask, rope=rope)
+            x = block(x, t, mask = mask, rope = rope)
 
         if self.long_skip_connection is not None:
-            x = self.long_skip_connection(torch.cat((x, residual), dim=-1))
+            x = self.long_skip_connection(torch.cat((x, residual), dim = -1))
 
         x = self.norm_out(x, t)
         output = self.proj_out(x)

model/backbones/mmdit.py

@@ -12,6 +12,8 @@ from __future__ import annotations
 import torch
 from torch import nn
 
+from einops import repeat
+
 from x_transformers.x_transformers import RotaryEmbedding
 
 from model.modules import (
@@ -19,14 +21,12 @@ from model.modules import (
     ConvPositionEmbedding,
     MMDiTBlock,
     AdaLayerNormZero_Final,
-    precompute_freqs_cis,
-    get_pos_embed_indices,
+    precompute_freqs_cis, get_pos_embed_indices,
 )
 
 
 # text embedding
 
-
 class TextEmbedding(nn.Module):
     def __init__(self, out_dim, text_num_embeds):
         super().__init__()
@@ -35,7 +35,7 @@ class TextEmbedding(nn.Module):
         self.precompute_max_pos = 1024
         self.register_buffer("freqs_cis", precompute_freqs_cis(out_dim, self.precompute_max_pos), persistent=False)
 
-    def forward(self, text: int["b nt"], drop_text=False) -> int["b nt d"]:  # noqa: F722
+    def forward(self, text: int['b nt'], drop_text = False) -> int['b nt d']:
         text = text + 1
         if drop_text:
             text = torch.zeros_like(text)
@@ -54,37 +54,27 @@ class TextEmbedding(nn.Module):
 
 # noised input & masked cond audio embedding
 
-
 class AudioEmbedding(nn.Module):
     def __init__(self, in_dim, out_dim):
         super().__init__()
         self.linear = nn.Linear(2 * in_dim, out_dim)
         self.conv_pos_embed = ConvPositionEmbedding(out_dim)
 
-    def forward(self, x: float["b n d"], cond: float["b n d"], drop_audio_cond=False):  # noqa: F722
+    def forward(self, x: float['b n d'], cond: float['b n d'], drop_audio_cond = False):
         if drop_audio_cond:
             cond = torch.zeros_like(cond)
-        x = torch.cat((x, cond), dim=-1)
+        x = torch.cat((x, cond), dim = -1)
         x = self.linear(x)
         x = self.conv_pos_embed(x) + x
         return x
-
+    
 
 # Transformer backbone using MM-DiT blocks
 
-
 class MMDiT(nn.Module):
-    def __init__(
-        self,
-        *,
-        dim,
-        depth=8,
-        heads=8,
-        dim_head=64,
-        dropout=0.1,
-        ff_mult=4,
-        text_num_embeds=256,
-        mel_dim=100,
+    def __init__(self, *, 
+                 dim, depth = 8, heads = 8, dim_head = 64, dropout = 0.1, ff_mult = 4,
+                 text_num_embeds = 256, mel_dim = 100,
     ):
         super().__init__()
 
@@ -96,16 +86,16 @@ class MMDiT(nn.Module):
 
         self.dim = dim
         self.depth = depth
-
+        
         self.transformer_blocks = nn.ModuleList(
             [
                 MMDiTBlock(
-                    dim=dim,
-                    heads=heads,
-                    dim_head=dim_head,
-                    dropout=dropout,
-                    ff_mult=ff_mult,
-                    context_pre_only=i == depth - 1,
+                    dim = dim,
+                    heads = heads,
+                    dim_head = dim_head,
+                    dropout = dropout,
+                    ff_mult = ff_mult,
+                    context_pre_only = i == depth - 1,
                 )
                 for i in range(depth)
             ]
@@ -115,30 +105,30 @@ class MMDiT(nn.Module):
 
     def forward(
         self,
-        x: float["b n d"],  # nosied input audio  # noqa: F722
-        cond: float["b n d"],  # masked cond audio  # noqa: F722
-        text: int["b nt"],  # text  # noqa: F722
-        time: float["b"] | float[""],  # time step  # noqa: F821 F722
+        x: float['b n d'],     # nosied input audio
+        cond: float['b n d'],  # masked cond audio
+        text: int['b nt'],     # text
+        time: float['b'] | float[''],  # time step
         drop_audio_cond,  # cfg for cond audio
-        drop_text,  # cfg for text
-        mask: bool["b n"] | None = None,  # noqa: F722
+        drop_text,        # cfg for text
+        mask: bool['b n'] | None = None,
     ):
         batch = x.shape[0]
         if time.ndim == 0:
-            time = time.repeat(batch)
+            time = repeat(time, ' -> b', b = batch)
 
         # t: conditioning (time), c: context (text + masked cond audio), x: noised input audio
         t = self.time_embed(time)
-        c = self.text_embed(text, drop_text=drop_text)
-        x = self.audio_embed(x, cond, drop_audio_cond=drop_audio_cond)
+        c = self.text_embed(text, drop_text = drop_text)
+        x = self.audio_embed(x, cond, drop_audio_cond = drop_audio_cond)
 
         seq_len = x.shape[1]
         text_len = text.shape[1]
         rope_audio = self.rotary_embed.forward_from_seq_len(seq_len)
         rope_text = self.rotary_embed.forward_from_seq_len(text_len)
-
+        
         for block in self.transformer_blocks:
-            c, x = block(x, c, t, mask=mask, rope=rope_audio, c_rope=rope_text)
+            c, x = block(x, c, t, mask = mask, rope = rope_audio, c_rope = rope_text)
 
         x = self.norm_out(x, t)
         output = self.proj_out(x)

model/backbones/unett.py

@@ -14,6 +14,8 @@ import torch
 from torch import nn
 import torch.nn.functional as F
 
+from einops import repeat, pack, unpack
+
 from x_transformers import RMSNorm
 from x_transformers.x_transformers import RotaryEmbedding
 
@@ -24,16 +26,14 @@ from model.modules import (
     Attention,
     AttnProcessor,
     FeedForward,
-    precompute_freqs_cis,
-    get_pos_embed_indices,
+    precompute_freqs_cis, get_pos_embed_indices,
 )
 
 
 # Text embedding
 
-
 class TextEmbedding(nn.Module):
-    def __init__(self, text_num_embeds, text_dim, conv_layers=0, conv_mult=2):
+    def __init__(self, text_num_embeds, text_dim, conv_layers = 0, conv_mult = 2):
         super().__init__()
         self.text_embed = nn.Embedding(text_num_embeds + 1, text_dim)  # use 0 as filler token
 
@@ -41,22 +41,20 @@ class TextEmbedding(nn.Module):
             self.extra_modeling = True
             self.precompute_max_pos = 4096  # ~44s of 24khz audio
             self.register_buffer("freqs_cis", precompute_freqs_cis(text_dim, self.precompute_max_pos), persistent=False)
-            self.text_blocks = nn.Sequential(
-                *[ConvNeXtV2Block(text_dim, text_dim * conv_mult) for _ in range(conv_layers)]
-            )
+            self.text_blocks = nn.Sequential(*[ConvNeXtV2Block(text_dim, text_dim * conv_mult) for _ in range(conv_layers)])
         else:
             self.extra_modeling = False
 
-    def forward(self, text: int["b nt"], seq_len, drop_text=False):  # noqa: F722
+    def forward(self, text: int['b nt'], seq_len, drop_text = False):
+        batch, text_len = text.shape[0], text.shape[1]
         text = text + 1  # use 0 as filler token. preprocess of batch pad -1, see list_str_to_idx()
         text = text[:, :seq_len]  # curtail if character tokens are more than the mel spec tokens
-        batch, text_len = text.shape[0], text.shape[1]
-        text = F.pad(text, (0, seq_len - text_len), value=0)
+        text = F.pad(text, (0, seq_len - text_len), value = 0)
 
         if drop_text:  # cfg for text
             text = torch.zeros_like(text)
 
-        text = self.text_embed(text)  # b n -> b n d
+        text = self.text_embed(text) # b n -> b n d
 
         # possible extra modeling
         if self.extra_modeling:
@@ -74,40 +72,28 @@ class TextEmbedding(nn.Module):
 
 # noised input audio and context mixing embedding
 
-
 class InputEmbedding(nn.Module):
     def __init__(self, mel_dim, text_dim, out_dim):
         super().__init__()
         self.proj = nn.Linear(mel_dim * 2 + text_dim, out_dim)
-        self.conv_pos_embed = ConvPositionEmbedding(dim=out_dim)
+        self.conv_pos_embed = ConvPositionEmbedding(dim = out_dim)
 
-    def forward(self, x: float["b n d"], cond: float["b n d"], text_embed: float["b n d"], drop_audio_cond=False):  # noqa: F722
+    def forward(self, x: float['b n d'], cond: float['b n d'], text_embed: float['b n d'], drop_audio_cond = False):
         if drop_audio_cond:  # cfg for cond audio
             cond = torch.zeros_like(cond)
 
-        x = self.proj(torch.cat((x, cond, text_embed), dim=-1))
+        x = self.proj(torch.cat((x, cond, text_embed), dim = -1))
         x = self.conv_pos_embed(x) + x
         return x
 
 
 # Flat UNet Transformer backbone
 
-
 class UNetT(nn.Module):
-    def __init__(
-        self,
-        *,
-        dim,
-        depth=8,
-        heads=8,
-        dim_head=64,
-        dropout=0.1,
-        ff_mult=4,
-        mel_dim=100,
-        text_num_embeds=256,
-        text_dim=None,
-        conv_layers=0,
-        skip_connect_type: Literal["add", "concat", "none"] = "concat",
+    def __init__(self, *,
+                 dim, depth = 8, heads = 8, dim_head = 64, dropout = 0.1, ff_mult = 4,
+                 mel_dim = 100, text_num_embeds = 256, text_dim = None, conv_layers = 0,
+                 skip_connect_type: Literal['add', 'concat', 'none'] = 'concat',
     ):
         super().__init__()
         assert depth % 2 == 0, "UNet-Transformer's depth should be even."
@@ -115,7 +101,7 @@ class UNetT(nn.Module):
         self.time_embed = TimestepEmbedding(dim)
         if text_dim is None:
             text_dim = mel_dim
-        self.text_embed = TextEmbedding(text_num_embeds, text_dim, conv_layers=conv_layers)
+        self.text_embed = TextEmbedding(text_num_embeds, text_dim, conv_layers = conv_layers)
         self.input_embed = InputEmbedding(mel_dim, text_dim, dim)
 
         self.rotary_embed = RotaryEmbedding(dim_head)
@@ -124,7 +110,7 @@ class UNetT(nn.Module):
 
         self.dim = dim
         self.skip_connect_type = skip_connect_type
-        needs_skip_proj = skip_connect_type == "concat"
+        needs_skip_proj = skip_connect_type == 'concat'
 
         self.depth = depth
         self.layers = nn.ModuleList([])
@@ -134,57 +120,53 @@ class UNetT(nn.Module):
 
             attn_norm = RMSNorm(dim)
             attn = Attention(
-                processor=AttnProcessor(),
-                dim=dim,
-                heads=heads,
-                dim_head=dim_head,
-                dropout=dropout,
-            )
+                processor = AttnProcessor(),
+                dim = dim,
+                heads = heads,
+                dim_head = dim_head,
+                dropout = dropout,
+                )
 
             ff_norm = RMSNorm(dim)
-            ff = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
-
-            skip_proj = nn.Linear(dim * 2, dim, bias=False) if needs_skip_proj and is_later_half else None
-
-            self.layers.append(
-                nn.ModuleList(
-                    [
-                        skip_proj,
-                        attn_norm,
-                        attn,
-                        ff_norm,
-                        ff,
-                    ]
-                )
-            )
+            ff = FeedForward(dim = dim, mult = ff_mult, dropout = dropout, approximate = "tanh")
+
+            skip_proj = nn.Linear(dim * 2, dim, bias = False) if needs_skip_proj and is_later_half else None
+
+            self.layers.append(nn.ModuleList([
+                skip_proj,
+                attn_norm,
+                attn,
+                ff_norm,
+                ff,
+            ]))
 
         self.norm_out = RMSNorm(dim)
         self.proj_out = nn.Linear(dim, mel_dim)
 
     def forward(
         self,
-        x: float["b n d"],  # nosied input audio  # noqa: F722
-        cond: float["b n d"],  # masked cond audio  # noqa: F722
-        text: int["b nt"],  # text  # noqa: F722
-        time: float["b"] | float[""],  # time step  # noqa: F821 F722
+        x: float['b n d'],     # nosied input audio
+        cond: float['b n d'],  # masked cond audio
+        text: int['b nt'],     # text
+        time: float['b'] | float[''],  # time step
         drop_audio_cond,  # cfg for cond audio
-        drop_text,  # cfg for text
-        mask: bool["b n"] | None = None,  # noqa: F722
+        drop_text,        # cfg for text
+        mask: bool['b n'] | None = None,
     ):
         batch, seq_len = x.shape[0], x.shape[1]
         if time.ndim == 0:
-            time = time.repeat(batch)
-
+            time = repeat(time, ' -> b', b = batch)
+        
         # t: conditioning time, c: context (text + masked cond audio), x: noised input audio
         t = self.time_embed(time)
-        text_embed = self.text_embed(text, seq_len, drop_text=drop_text)
-        x = self.input_embed(x, cond, text_embed, drop_audio_cond=drop_audio_cond)
+        text_embed = self.text_embed(text, seq_len, drop_text = drop_text)
+        x = self.input_embed(x, cond, text_embed, drop_audio_cond = drop_audio_cond)
 
         # postfix time t to input x, [b n d] -> [b n+1 d]
-        x = torch.cat([t.unsqueeze(1), x], dim=1)  # pack t to x
+        x, ps = pack((t, x), 'b * d')
         if mask is not None:
             mask = F.pad(mask, (1, 0), value=1)
-
+        
         rope = self.rotary_embed.forward_from_seq_len(seq_len + 1)
 
         # flat unet transformer
@@ -202,18 +184,18 @@ class UNetT(nn.Module):
 
             if is_later_half:
                 skip = skips.pop()
-                if skip_connect_type == "concat":
-                    x = torch.cat((x, skip), dim=-1)
+                if skip_connect_type == 'concat':
+                    x = torch.cat((x, skip), dim = -1)
                     x = maybe_skip_proj(x)
-                elif skip_connect_type == "add":
+                elif skip_connect_type == 'add':
                     x = x + skip
 
             # attention and feedforward blocks
-            x = attn(attn_norm(x), rope=rope, mask=mask) + x
+            x = attn(attn_norm(x), rope = rope, mask = mask) + x
             x = ff(ff_norm(x)) + x
 
         assert len(skips) == 0
 
-        x = self.norm_out(x)[:, 1:, :]  # unpack t from x
+        _, x = unpack(self.norm_out(x), ps, 'b * d')
 
         return self.proj_out(x)

model/cfm.py

@@ -18,34 +18,34 @@ from torch.nn.utils.rnn import pad_sequence
 
 from torchdiffeq import odeint
 
+from einops import rearrange
+
 from model.modules import MelSpec
+
 from model.utils import (
-    default,
-    exists,
-    list_str_to_idx,
-    list_str_to_tensor,
-    lens_to_mask,
-    mask_from_frac_lengths,
-)
+    default, exists, 
+    list_str_to_idx, list_str_to_tensor, 
+    lens_to_mask, mask_from_frac_lengths,
+) 
 
 
 class CFM(nn.Module):
     def __init__(
         self,
         transformer: nn.Module,
-        sigma=0.0,
+        sigma = 0.,
         odeint_kwargs: dict = dict(
             # atol = 1e-5,
             # rtol = 1e-5,
-            method="euler"  # 'midpoint'
+            method = 'euler'  # 'midpoint'
         ),
-        audio_drop_prob=0.3,
-        cond_drop_prob=0.2,
-        num_channels=None,
+        audio_drop_prob = 0.3,
+        cond_drop_prob = 0.2,
+        num_channels = None,
         mel_spec_module: nn.Module | None = None,
         mel_spec_kwargs: dict = dict(),
-        frac_lengths_mask: tuple[float, float] = (0.7, 1.0),
-        vocab_char_map: dict[str:int] | None = None,
+        frac_lengths_mask: tuple[float, float] = (0.7, 1.),
+        vocab_char_map: dict[str: int] | None = None
     ):
         super().__init__()
 
@@ -81,37 +81,34 @@ class CFM(nn.Module):
     @torch.no_grad()
     def sample(
         self,
-        cond: float["b n d"] | float["b nw"],  # noqa: F722
-        text: int["b nt"] | list[str],  # noqa: F722
-        duration: int | int["b"],  # noqa: F821
+        cond: float['b n d'] | float['b nw'],
+        text: int['b nt'] | list[str],
+        duration: int | int['b'],
         *,
-        lens: int["b"] | None = None,  # noqa: F821
-        steps=32,
-        cfg_strength=1.0,
-        sway_sampling_coef=None,
+        lens: int['b'] | None = None,
+        steps = 32,
+        cfg_strength = 1., 
+        sway_sampling_coef = None,
         seed: int | None = None,
-        max_duration=4096,
-        vocoder: Callable[[float["b d n"]], float["b nw"]] | None = None,  # noqa: F722
-        no_ref_audio=False,
-        duplicate_test=False,
-        t_inter=0.1,
-        edit_mask=None,
+        max_duration = 4096, 
+        vocoder: Callable[[float['b d n']], float['b nw']] | None = None,
+        no_ref_audio = False,
+        duplicate_test = False,
+        t_inter = 0.1,
+        edit_mask = None,
     ):
         self.eval()
 
-        if next(self.parameters()).dtype == torch.float16:
-            cond = cond.half()
-
         # raw wave
 
         if cond.ndim == 2:
             cond = self.mel_spec(cond)
-            cond = cond.permute(0, 2, 1)
+            cond = rearrange(cond, 'b d n -> b n d')
             assert cond.shape[-1] == self.num_channels
 
         batch, cond_seq_len, device = *cond.shape[:2], cond.device
         if not exists(lens):
-            lens = torch.full((batch,), cond_seq_len, device=device, dtype=torch.long)
+            lens = torch.full((batch,), cond_seq_len, device = device, dtype = torch.long)
 
         # text
 
@@ -123,8 +120,8 @@ class CFM(nn.Module):
             assert text.shape[0] == batch
 
         if exists(text):
-            text_lens = (text != -1).sum(dim=-1)
-            lens = torch.maximum(text_lens, lens)  # make sure lengths are at least those of the text characters
+            text_lens = (text != -1).sum(dim = -1)
+            lens = torch.maximum(text_lens, lens) # make sure lengths are at least those of the text characters
 
         # duration
 
@@ -133,22 +130,20 @@ class CFM(nn.Module):
             cond_mask = cond_mask & edit_mask
 
         if isinstance(duration, int):
-            duration = torch.full((batch,), duration, device=device, dtype=torch.long)
+            duration = torch.full((batch,), duration, device = device, dtype = torch.long)
 
-        duration = torch.maximum(lens + 1, duration)  # just add one token so something is generated
-        duration = duration.clamp(max=max_duration)
+        duration = torch.maximum(lens + 1, duration) # just add one token so something is generated
+        duration = duration.clamp(max = max_duration)
         max_duration = duration.amax()
-
+        
         # duplicate test corner for inner time step oberservation
         if duplicate_test:
-            test_cond = F.pad(cond, (0, 0, cond_seq_len, max_duration - 2 * cond_seq_len), value=0.0)
-
-        cond = F.pad(cond, (0, 0, 0, max_duration - cond_seq_len), value=0.0)
-        cond_mask = F.pad(cond_mask, (0, max_duration - cond_mask.shape[-1]), value=False)
-        cond_mask = cond_mask.unsqueeze(-1)
-        step_cond = torch.where(
-            cond_mask, cond, torch.zeros_like(cond)
-        )  # allow direct control (cut cond audio) with lens passed in
+            test_cond = F.pad(cond, (0, 0, cond_seq_len, max_duration - 2*cond_seq_len), value = 0.)
+            
+        cond = F.pad(cond, (0, 0, 0, max_duration - cond_seq_len), value = 0.)
+        cond_mask = F.pad(cond_mask, (0, max_duration - cond_mask.shape[-1]), value = False)
+        cond_mask = rearrange(cond_mask, '... -> ... 1')
+        step_cond = torch.where(cond_mask, cond, torch.zeros_like(cond))  # allow direct control (cut cond audio) with lens passed in
 
         if batch > 1:
             mask = lens_to_mask(duration)
@@ -166,15 +161,11 @@ class CFM(nn.Module):
             # step_cond = torch.where(cond_mask, cond, torch.zeros_like(cond))
 
             # predict flow
-            pred = self.transformer(
-                x=x, cond=step_cond, text=text, time=t, mask=mask, drop_audio_cond=False, drop_text=False
-            )
+            pred = self.transformer(x = x, cond = step_cond, text = text, time = t, mask = mask, drop_audio_cond = False, drop_text = False)
             if cfg_strength < 1e-5:
                 return pred
-
-            null_pred = self.transformer(
-                x=x, cond=step_cond, text=text, time=t, mask=mask, drop_audio_cond=True, drop_text=True
-            )
+            
+            null_pred = self.transformer(x = x, cond = step_cond, text = text, time = t, mask = mask, drop_audio_cond = True, drop_text = True)
             return pred + (pred - null_pred) * cfg_strength
 
         # noise input
@@ -184,8 +175,8 @@ class CFM(nn.Module):
         for dur in duration:
             if exists(seed):
                 torch.manual_seed(seed)
-            y0.append(torch.randn(dur, self.num_channels, device=self.device, dtype=step_cond.dtype))
-        y0 = pad_sequence(y0, padding_value=0, batch_first=True)
+            y0.append(torch.randn(dur, self.num_channels, device = self.device))
+        y0 = pad_sequence(y0, padding_value = 0, batch_first = True)
 
         t_start = 0
 
@@ -195,37 +186,37 @@ class CFM(nn.Module):
             y0 = (1 - t_start) * y0 + t_start * test_cond
             steps = int(steps * (1 - t_start))
 
-        t = torch.linspace(t_start, 1, steps, device=self.device, dtype=step_cond.dtype)
+        t = torch.linspace(t_start, 1, steps, device = self.device)
         if sway_sampling_coef is not None:
             t = t + sway_sampling_coef * (torch.cos(torch.pi / 2 * t) - 1 + t)
 
         trajectory = odeint(fn, y0, t, **self.odeint_kwargs)
-
+        
         sampled = trajectory[-1]
         out = sampled
         out = torch.where(cond_mask, cond, out)
 
         if exists(vocoder):
-            out = out.permute(0, 2, 1)
+            out = rearrange(out, 'b n d -> b d n')
             out = vocoder(out)
 
         return out, trajectory
 
     def forward(
         self,
-        inp: float["b n d"] | float["b nw"],  # mel or raw wave  # noqa: F722
-        text: int["b nt"] | list[str],  # noqa: F722
+        inp: float['b n d'] | float['b nw'], # mel or raw wave
+        text: int['b nt'] | list[str],
         *,
-        lens: int["b"] | None = None,  # noqa: F821
+        lens: int['b'] | None = None,
         noise_scheduler: str | None = None,
     ):
         # handle raw wave
         if inp.ndim == 2:
             inp = self.mel_spec(inp)
-            inp = inp.permute(0, 2, 1)
+            inp = rearrange(inp, 'b d n -> b n d')
             assert inp.shape[-1] == self.num_channels
 
-        batch, seq_len, dtype, device, _σ1 = *inp.shape[:2], inp.dtype, self.device, self.sigma
+        batch, seq_len, dtype, device, σ1 = *inp.shape[:2], inp.dtype, self.device, self.sigma
 
         # handle text as string
         if isinstance(text, list):
@@ -237,12 +228,12 @@ class CFM(nn.Module):
 
         # lens and mask
         if not exists(lens):
-            lens = torch.full((batch,), seq_len, device=device)
-
-        mask = lens_to_mask(lens, length=seq_len)  # useless here, as collate_fn will pad to max length in batch
+            lens = torch.full((batch,), seq_len, device = device)
+        
+        mask = lens_to_mask(lens, length = seq_len)  # useless here, as collate_fn will pad to max length in batch
 
         # get a random span to mask out for training conditionally
-        frac_lengths = torch.zeros((batch,), device=self.device).float().uniform_(*self.frac_lengths_mask)
+        frac_lengths = torch.zeros((batch,), device = self.device).float().uniform_(*self.frac_lengths_mask)
         rand_span_mask = mask_from_frac_lengths(lens, frac_lengths)
 
         if exists(mask):
@@ -255,16 +246,19 @@ class CFM(nn.Module):
         x0 = torch.randn_like(x1)
 
         # time step
-        time = torch.rand((batch,), dtype=dtype, device=self.device)
+        time = torch.rand((batch,), dtype = dtype, device = self.device)
         # TODO. noise_scheduler
 
         # sample xt (φ_t(x) in the paper)
-        t = time.unsqueeze(-1).unsqueeze(-1)
+        t = rearrange(time, 'b -> b 1 1')
         φ = (1 - t) * x0 + t * x1
         flow = x1 - x0
 
         # only predict what is within the random mask span for infilling
-        cond = torch.where(rand_span_mask[..., None], torch.zeros_like(x1), x1)
+        cond = torch.where(
+            rand_span_mask[..., None],
+            torch.zeros_like(x1), x1
+        )
 
         # transformer and cfg training with a drop rate
         drop_audio_cond = random() < self.audio_drop_prob  # p_drop in voicebox paper
@@ -273,15 +267,13 @@ class CFM(nn.Module):
             drop_text = True
         else:
             drop_text = False
-
+            
         # if want rigourously mask out padding, record in collate_fn in dataset.py, and pass in here
         # adding mask will use more memory, thus also need to adjust batchsampler with scaled down threshold for long sequences
-        pred = self.transformer(
-            x=φ, cond=cond, text=text, time=time, drop_audio_cond=drop_audio_cond, drop_text=drop_text
-        )
+        pred = self.transformer(x = φ, cond = cond, text = text, time = time, drop_audio_cond = drop_audio_cond, drop_text = drop_text)
 
         # flow matching loss
-        loss = F.mse_loss(pred, flow, reduction="none")
+        loss = F.mse_loss(pred, flow, reduction = 'none')
         loss = loss[rand_span_mask]
 
         return loss.mean(), cond, pred

model/dataset.py

@@ -6,67 +6,65 @@ import torch
 import torch.nn.functional as F
 from torch.utils.data import Dataset, Sampler
 import torchaudio
-from datasets import load_from_disk
+from datasets import load_dataset, load_from_disk
 from datasets import Dataset as Dataset_
-from torch import nn
+
+from einops import rearrange
 
 from model.modules import MelSpec
-from model.utils import default
 
 
 class HFDataset(Dataset):
     def __init__(
         self,
         hf_dataset: Dataset,
-        target_sample_rate=24_000,
-        n_mel_channels=100,
-        hop_length=256,
+        target_sample_rate = 24_000,
+        n_mel_channels = 100,
+        hop_length = 256,
     ):
         self.data = hf_dataset
         self.target_sample_rate = target_sample_rate
         self.hop_length = hop_length
-        self.mel_spectrogram = MelSpec(
-            target_sample_rate=target_sample_rate, n_mel_channels=n_mel_channels, hop_length=hop_length
-        )
-
+        self.mel_spectrogram = MelSpec(target_sample_rate=target_sample_rate, n_mel_channels=n_mel_channels, hop_length=hop_length)
+        
     def get_frame_len(self, index):
         row = self.data[index]
-        audio = row["audio"]["array"]
-        sample_rate = row["audio"]["sampling_rate"]
+        audio = row['audio']['array']
+        sample_rate = row['audio']['sampling_rate']
         return audio.shape[-1] / sample_rate * self.target_sample_rate / self.hop_length
 
     def __len__(self):
         return len(self.data)
-
+    
     def __getitem__(self, index):
         row = self.data[index]
-        audio = row["audio"]["array"]
+        audio = row['audio']['array']
 
         # logger.info(f"Audio shape: {audio.shape}")
 
-        sample_rate = row["audio"]["sampling_rate"]
+        sample_rate = row['audio']['sampling_rate']
         duration = audio.shape[-1] / sample_rate
 
         if duration > 30 or duration < 0.3:
             return self.__getitem__((index + 1) % len(self.data))
-
+        
         audio_tensor = torch.from_numpy(audio).float()
-
+        
         if sample_rate != self.target_sample_rate:
             resampler = torchaudio.transforms.Resample(sample_rate, self.target_sample_rate)
             audio_tensor = resampler(audio_tensor)
-
-        audio_tensor = audio_tensor.unsqueeze(0)  # 't -> 1 t')
-
+        
+        audio_tensor = rearrange(audio_tensor, 't -> 1 t')
+        
         mel_spec = self.mel_spectrogram(audio_tensor)
-
-        mel_spec = mel_spec.squeeze(0)  # '1 d t -> d t'
-
-        text = row["text"]
-
+        
+        mel_spec = rearrange(mel_spec, '1 d t -> d t')
+        
+        text = row['text']
+        
         return dict(
-            mel_spec=mel_spec,
-            text=text,
+            mel_spec = mel_spec,
+            text = text,
         )
 
 
@@ -74,39 +72,28 @@ class CustomDataset(Dataset):
     def __init__(
         self,
         custom_dataset: Dataset,
-        durations=None,
-        target_sample_rate=24_000,
-        hop_length=256,
-        n_mel_channels=100,
-        preprocessed_mel=False,
-        mel_spec_module: nn.Module | None = None,
+        durations = None,
+        target_sample_rate = 24_000,
+        hop_length = 256,
+        n_mel_channels = 100,
+        preprocessed_mel = False,
     ):
         self.data = custom_dataset
         self.durations = durations
         self.target_sample_rate = target_sample_rate
         self.hop_length = hop_length
         self.preprocessed_mel = preprocessed_mel
-
         if not preprocessed_mel:
-            self.mel_spectrogram = default(
-                mel_spec_module,
-                MelSpec(
-                    target_sample_rate=target_sample_rate,
-                    hop_length=hop_length,
-                    n_mel_channels=n_mel_channels,
-                ),
-            )
+            self.mel_spectrogram = MelSpec(target_sample_rate=target_sample_rate, hop_length=hop_length, n_mel_channels=n_mel_channels)
 
     def get_frame_len(self, index):
-        if (
-            self.durations is not None
-        ):  # Please make sure the separately provided durations are correct, otherwise 99.99% OOM
+        if self.durations is not None:  # Please make sure the separately provided durations are correct, otherwise 99.99% OOM
             return self.durations[index] * self.target_sample_rate / self.hop_length
         return self.data[index]["duration"] * self.target_sample_rate / self.hop_length
-
+    
     def __len__(self):
         return len(self.data)
-
+    
     def __getitem__(self, index):
         row = self.data[index]
         audio_path = row["audio_path"]
@@ -118,57 +105,48 @@ class CustomDataset(Dataset):
 
         else:
             audio, source_sample_rate = torchaudio.load(audio_path)
-            if audio.shape[0] > 1:
-                audio = torch.mean(audio, dim=0, keepdim=True)
 
             if duration > 30 or duration < 0.3:
                 return self.__getitem__((index + 1) % len(self.data))
-
+            
             if source_sample_rate != self.target_sample_rate:
                 resampler = torchaudio.transforms.Resample(source_sample_rate, self.target_sample_rate)
                 audio = resampler(audio)
-
+            
             mel_spec = self.mel_spectrogram(audio)
-            mel_spec = mel_spec.squeeze(0)  # '1 d t -> d t')
-
+            mel_spec = rearrange(mel_spec, '1 d t -> d t')
+        
         return dict(
-            mel_spec=mel_spec,
-            text=text,
+            mel_spec = mel_spec,
+            text = text,
         )
-
+    
 
 # Dynamic Batch Sampler
 
-
 class DynamicBatchSampler(Sampler[list[int]]):
-    """Extension of Sampler that will do the following:
-    1.  Change the batch size (essentially number of sequences)
-        in a batch to ensure that the total number of frames are less
-        than a certain threshold.
-    2.  Make sure the padding efficiency in the batch is high.
+    """ Extension of Sampler that will do the following:
+        1.  Change the batch size (essentially number of sequences)
+            in a batch to ensure that the total number of frames are less
+            than a certain threshold.
+        2.  Make sure the padding efficiency in the batch is high.
     """
 
-    def __init__(
-        self, sampler: Sampler[int], frames_threshold: int, max_samples=0, random_seed=None, drop_last: bool = False
-    ):
+    def __init__(self, sampler: Sampler[int], frames_threshold: int, max_samples=0, random_seed=None, drop_last: bool = False):
         self.sampler = sampler
         self.frames_threshold = frames_threshold
         self.max_samples = max_samples
 
         indices, batches = [], []
         data_source = self.sampler.data_source
-
-        for idx in tqdm(
-            self.sampler, desc="Sorting with sampler... if slow, check whether dataset is provided with duration"
-        ):
+        
+        for idx in tqdm(self.sampler, desc=f"Sorting with sampler... if slow, check whether dataset is provided with duration"):
             indices.append((idx, data_source.get_frame_len(idx)))
-        indices.sort(key=lambda elem: elem[1])
+        indices.sort(key=lambda elem : elem[1])
 
         batch = []
         batch_frames = 0
-        for idx, frame_len in tqdm(
-            indices, desc=f"Creating dynamic batches with {frames_threshold} audio frames per gpu"
-        ):
+        for idx, frame_len in tqdm(indices, desc=f"Creating dynamic batches with {frames_threshold} audio frames per gpu"):
             if batch_frames + frame_len <= self.frames_threshold and (max_samples == 0 or len(batch) < max_samples):
                 batch.append(idx)
                 batch_frames += frame_len
@@ -204,91 +182,76 @@ class DynamicBatchSampler(Sampler[list[int]]):
 
 # Load dataset
 
-
 def load_dataset(
-    dataset_name: str,
-    tokenizer: str = "pinyin",
-    dataset_type: str = "CustomDataset",
-    audio_type: str = "raw",
-    mel_spec_module: nn.Module | None = None,
-    mel_spec_kwargs: dict = dict(),
-) -> CustomDataset | HFDataset:
-    """
+        dataset_name: str,
+        tokenizer: str = "pinyin",
+        dataset_type: str = "CustomDataset", 
+        audio_type: str = "raw", 
+        mel_spec_kwargs: dict = dict()
+        ) -> CustomDataset | HFDataset:
+    '''
     dataset_type    - "CustomDataset" if you want to use tokenizer name and default data path to load for train_dataset
                     - "CustomDatasetPath" if you just want to pass the full path to a preprocessed dataset without relying on tokenizer
-    """
-
+    '''
+    
     print("Loading dataset ...")
 
     if dataset_type == "CustomDataset":
         if audio_type == "raw":
             try:
                 train_dataset = load_from_disk(f"data/{dataset_name}_{tokenizer}/raw")
-            except:  # noqa: E722
+            except:
                 train_dataset = Dataset_.from_file(f"data/{dataset_name}_{tokenizer}/raw.arrow")
             preprocessed_mel = False
         elif audio_type == "mel":
             train_dataset = Dataset_.from_file(f"data/{dataset_name}_{tokenizer}/mel.arrow")
             preprocessed_mel = True
-        with open(f"data/{dataset_name}_{tokenizer}/duration.json", "r", encoding="utf-8") as f:
+        with open(f"data/{dataset_name}_{tokenizer}/duration.json", 'r', encoding='utf-8') as f:
             data_dict = json.load(f)
         durations = data_dict["duration"]
-        train_dataset = CustomDataset(
-            train_dataset,
-            durations=durations,
-            preprocessed_mel=preprocessed_mel,
-            mel_spec_module=mel_spec_module,
-            **mel_spec_kwargs,
-        )
-
+        train_dataset = CustomDataset(train_dataset, durations=durations, preprocessed_mel=preprocessed_mel, **mel_spec_kwargs)
+        
     elif dataset_type == "CustomDatasetPath":
         try:
             train_dataset = load_from_disk(f"{dataset_name}/raw")
-        except:  # noqa: E722
+        except:
             train_dataset = Dataset_.from_file(f"{dataset_name}/raw.arrow")
-
-        with open(f"{dataset_name}/duration.json", "r", encoding="utf-8") as f:
+            
+        with open(f"{dataset_name}/duration.json", 'r', encoding='utf-8') as f:
             data_dict = json.load(f)
         durations = data_dict["duration"]
-        train_dataset = CustomDataset(
-            train_dataset, durations=durations, preprocessed_mel=preprocessed_mel, **mel_spec_kwargs
-        )
-
+        train_dataset = CustomDataset(train_dataset, durations=durations, preprocessed_mel=preprocessed_mel, **mel_spec_kwargs)
+            
     elif dataset_type == "HFDataset":
-        print(
-            "Should manually modify the path of huggingface dataset to your need.\n"
-            + "May also the corresponding script cuz different dataset may have different format."
-        )
+        print("Should manually modify the path of huggingface dataset to your need.\n" +
+              "May also the corresponding script cuz different dataset may have different format.")
         pre, post = dataset_name.split("_")
-        train_dataset = HFDataset(
-            load_dataset(f"{pre}/{pre}", split=f"train.{post}", cache_dir="./data"),
-        )
+        train_dataset = HFDataset(load_dataset(f"{pre}/{pre}", split=f"train.{post}", cache_dir="./data"),)
 
     return train_dataset
 
 
 # collation
 
-
 def collate_fn(batch):
-    mel_specs = [item["mel_spec"].squeeze(0) for item in batch]
+    mel_specs = [item['mel_spec'].squeeze(0) for item in batch]
     mel_lengths = torch.LongTensor([spec.shape[-1] for spec in mel_specs])
     max_mel_length = mel_lengths.amax()
 
     padded_mel_specs = []
     for spec in mel_specs:  # TODO. maybe records mask for attention here
         padding = (0, max_mel_length - spec.size(-1))
-        padded_spec = F.pad(spec, padding, value=0)
+        padded_spec = F.pad(spec, padding, value = 0)
         padded_mel_specs.append(padded_spec)
-
+    
     mel_specs = torch.stack(padded_mel_specs)
 
-    text = [item["text"] for item in batch]
+    text = [item['text'] for item in batch]
     text_lengths = torch.LongTensor([len(item) for item in text])
 
     return dict(
-        mel=mel_specs,
-        mel_lengths=mel_lengths,
-        text=text,
-        text_lengths=text_lengths,
+        mel = mel_specs,
+        mel_lengths = mel_lengths,
+        text = text,
+        text_lengths = text_lengths,
     )

model/ecapa_tdnn.py

@@ -9,14 +9,13 @@ import torch.nn as nn
 import torch.nn.functional as F
 
 
-""" Res2Conv1d + BatchNorm1d + ReLU
-"""
-
+''' Res2Conv1d + BatchNorm1d + ReLU
+'''
 
 class Res2Conv1dReluBn(nn.Module):
-    """
+    '''
     in_channels == out_channels == channels
-    """
+    '''
 
     def __init__(self, channels, kernel_size=1, stride=1, padding=0, dilation=1, bias=True, scale=4):
         super().__init__()
@@ -52,9 +51,8 @@ class Res2Conv1dReluBn(nn.Module):
         return out
 
 
-""" Conv1d + BatchNorm1d + ReLU
-"""
-
+''' Conv1d + BatchNorm1d + ReLU
+'''
 
 class Conv1dReluBn(nn.Module):
     def __init__(self, in_channels, out_channels, kernel_size=1, stride=1, padding=0, dilation=1, bias=True):
@@ -66,9 +64,8 @@ class Conv1dReluBn(nn.Module):
         return self.bn(F.relu(self.conv(x)))
 
 
-""" The SE connection of 1D case.
-"""
-
+''' The SE connection of 1D case.
+'''
 
 class SE_Connect(nn.Module):
     def __init__(self, channels, se_bottleneck_dim=128):
@@ -85,8 +82,8 @@ class SE_Connect(nn.Module):
         return out
 
 
-""" SE-Res2Block of the ECAPA-TDNN architecture.
-"""
+''' SE-Res2Block of the ECAPA-TDNN architecture.
+'''
 
 # def SE_Res2Block(channels, kernel_size, stride, padding, dilation, scale):
 #     return nn.Sequential(
@@ -96,7 +93,6 @@ class SE_Connect(nn.Module):
 #         SE_Connect(channels)
 #     )
 
-
 class SE_Res2Block(nn.Module):
     def __init__(self, in_channels, out_channels, kernel_size, stride, padding, dilation, scale, se_bottleneck_dim):
         super().__init__()
@@ -126,9 +122,8 @@ class SE_Res2Block(nn.Module):
         return x + residual
 
 
-""" Attentive weighted mean and standard deviation pooling.
-"""
-
+''' Attentive weighted mean and standard deviation pooling.
+'''
 
 class AttentiveStatsPool(nn.Module):
     def __init__(self, in_dim, attention_channels=128, global_context_att=False):
@@ -143,6 +138,7 @@ class AttentiveStatsPool(nn.Module):
         self.linear2 = nn.Conv1d(attention_channels, in_dim, kernel_size=1)  # equals V and k in the paper
 
     def forward(self, x):
+
         if self.global_context_att:
             context_mean = torch.mean(x, dim=-1, keepdim=True).expand_as(x)
             context_std = torch.sqrt(torch.var(x, dim=-1, keepdim=True) + 1e-10).expand_as(x)
@@ -155,52 +151,38 @@ class AttentiveStatsPool(nn.Module):
         # alpha = F.relu(self.linear1(x_in))
         alpha = torch.softmax(self.linear2(alpha), dim=2)
         mean = torch.sum(alpha * x, dim=2)
-        residuals = torch.sum(alpha * (x**2), dim=2) - mean**2
+        residuals = torch.sum(alpha * (x ** 2), dim=2) - mean ** 2
         std = torch.sqrt(residuals.clamp(min=1e-9))
         return torch.cat([mean, std], dim=1)
 
 
 class ECAPA_TDNN(nn.Module):
-    def __init__(
-        self,
-        feat_dim=80,
-        channels=512,
-        emb_dim=192,
-        global_context_att=False,
-        feat_type="wavlm_large",
-        sr=16000,
-        feature_selection="hidden_states",
-        update_extract=False,
-        config_path=None,
-    ):
+    def __init__(self, feat_dim=80, channels=512, emb_dim=192, global_context_att=False,
+                 feat_type='wavlm_large', sr=16000, feature_selection="hidden_states", update_extract=False, config_path=None):
         super().__init__()
 
         self.feat_type = feat_type
         self.feature_selection = feature_selection
         self.update_extract = update_extract
         self.sr = sr
-
-        torch.hub._validate_not_a_forked_repo = lambda a, b, c: True
+        
+        torch.hub._validate_not_a_forked_repo=lambda a,b,c: True
         try:
             local_s3prl_path = os.path.expanduser("~/.cache/torch/hub/s3prl_s3prl_main")
-            self.feature_extract = torch.hub.load(local_s3prl_path, feat_type, source="local", config_path=config_path)
-        except:  # noqa: E722
-            self.feature_extract = torch.hub.load("s3prl/s3prl", feat_type)
+            self.feature_extract = torch.hub.load(local_s3prl_path, feat_type, source='local', config_path=config_path)
+        except:
+            self.feature_extract = torch.hub.load('s3prl/s3prl', feat_type)
 
-        if len(self.feature_extract.model.encoder.layers) == 24 and hasattr(
-            self.feature_extract.model.encoder.layers[23].self_attn, "fp32_attention"
-        ):
+        if len(self.feature_extract.model.encoder.layers) == 24 and hasattr(self.feature_extract.model.encoder.layers[23].self_attn, "fp32_attention"):
             self.feature_extract.model.encoder.layers[23].self_attn.fp32_attention = False
-        if len(self.feature_extract.model.encoder.layers) == 24 and hasattr(
-            self.feature_extract.model.encoder.layers[11].self_attn, "fp32_attention"
-        ):
+        if len(self.feature_extract.model.encoder.layers) == 24 and hasattr(self.feature_extract.model.encoder.layers[11].self_attn, "fp32_attention"):
             self.feature_extract.model.encoder.layers[11].self_attn.fp32_attention = False
 
         self.feat_num = self.get_feat_num()
         self.feature_weight = nn.Parameter(torch.zeros(self.feat_num))
 
-        if feat_type != "fbank" and feat_type != "mfcc":
-            freeze_list = ["final_proj", "label_embs_concat", "mask_emb", "project_q", "quantizer"]
+        if feat_type != 'fbank' and feat_type != 'mfcc':
+            freeze_list = ['final_proj', 'label_embs_concat', 'mask_emb', 'project_q', 'quantizer']
             for name, param in self.feature_extract.named_parameters():
                 for freeze_val in freeze_list:
                     if freeze_val in name:
@@ -216,46 +198,18 @@ class ECAPA_TDNN(nn.Module):
         self.channels = [channels] * 4 + [1536]
 
         self.layer1 = Conv1dReluBn(feat_dim, self.channels[0], kernel_size=5, padding=2)
-        self.layer2 = SE_Res2Block(
-            self.channels[0],
-            self.channels[1],
-            kernel_size=3,
-            stride=1,
-            padding=2,
-            dilation=2,
-            scale=8,
-            se_bottleneck_dim=128,
-        )
-        self.layer3 = SE_Res2Block(
-            self.channels[1],
-            self.channels[2],
-            kernel_size=3,
-            stride=1,
-            padding=3,
-            dilation=3,
-            scale=8,
-            se_bottleneck_dim=128,
-        )
-        self.layer4 = SE_Res2Block(
-            self.channels[2],
-            self.channels[3],
-            kernel_size=3,
-            stride=1,
-            padding=4,
-            dilation=4,
-            scale=8,
-            se_bottleneck_dim=128,
-        )
+        self.layer2 = SE_Res2Block(self.channels[0], self.channels[1], kernel_size=3, stride=1, padding=2, dilation=2, scale=8, se_bottleneck_dim=128)
+        self.layer3 = SE_Res2Block(self.channels[1], self.channels[2], kernel_size=3, stride=1, padding=3, dilation=3, scale=8, se_bottleneck_dim=128)
+        self.layer4 = SE_Res2Block(self.channels[2], self.channels[3], kernel_size=3, stride=1, padding=4, dilation=4, scale=8, se_bottleneck_dim=128)
 
         # self.conv = nn.Conv1d(self.channels[-1], self.channels[-1], kernel_size=1)
         cat_channels = channels * 3
         self.conv = nn.Conv1d(cat_channels, self.channels[-1], kernel_size=1)
-        self.pooling = AttentiveStatsPool(
-            self.channels[-1], attention_channels=128, global_context_att=global_context_att
-        )
+        self.pooling = AttentiveStatsPool(self.channels[-1], attention_channels=128, global_context_att=global_context_att)
         self.bn = nn.BatchNorm1d(self.channels[-1] * 2)
         self.linear = nn.Linear(self.channels[-1] * 2, emb_dim)
 
+
     def get_feat_num(self):
         self.feature_extract.eval()
         wav = [torch.randn(self.sr).to(next(self.feature_extract.parameters()).device)]
@@ -272,12 +226,12 @@ class ECAPA_TDNN(nn.Module):
             x = self.feature_extract([sample for sample in x])
         else:
             with torch.no_grad():
-                if self.feat_type == "fbank" or self.feat_type == "mfcc":
+                if self.feat_type == 'fbank' or self.feat_type == 'mfcc':
                     x = self.feature_extract(x) + 1e-6  # B x feat_dim x time_len
                 else:
                     x = self.feature_extract([sample for sample in x])
 
-        if self.feat_type == "fbank":
+        if self.feat_type == 'fbank':
             x = x.log()
 
         if self.feat_type != "fbank" and self.feat_type != "mfcc":
@@ -309,22 +263,6 @@ class ECAPA_TDNN(nn.Module):
         return out
 
 
-def ECAPA_TDNN_SMALL(
-    feat_dim,
-    emb_dim=256,
-    feat_type="wavlm_large",
-    sr=16000,
-    feature_selection="hidden_states",
-    update_extract=False,
-    config_path=None,
-):
-    return ECAPA_TDNN(
-        feat_dim=feat_dim,
-        channels=512,
-        emb_dim=emb_dim,
-        feat_type=feat_type,
-        sr=sr,
-        feature_selection=feature_selection,
-        update_extract=update_extract,
-        config_path=config_path,
-    )
+def ECAPA_TDNN_SMALL(feat_dim, emb_dim=256, feat_type='wavlm_large', sr=16000, feature_selection="hidden_states", update_extract=False, config_path=None):
+    return ECAPA_TDNN(feat_dim=feat_dim, channels=512, emb_dim=emb_dim,
+                      feat_type=feat_type, sr=sr, feature_selection=feature_selection, update_extract=update_extract, config_path=config_path)

model/modules.py

@@ -16,45 +16,45 @@ from torch import nn
 import torch.nn.functional as F
 import torchaudio
 
+from einops import rearrange
 from x_transformers.x_transformers import apply_rotary_pos_emb
 
 
 # raw wav to mel spec
 
-
 class MelSpec(nn.Module):
     def __init__(
         self,
-        filter_length=1024,
-        hop_length=256,
-        win_length=1024,
-        n_mel_channels=100,
-        target_sample_rate=24_000,
-        normalize=False,
-        power=1,
-        norm=None,
-        center=True,
+        filter_length = 1024,
+        hop_length = 256,
+        win_length = 1024,
+        n_mel_channels = 100,
+        target_sample_rate = 24_000,
+        normalize = False,
+        power = 1,
+        norm = None,
+        center = True,
     ):
         super().__init__()
         self.n_mel_channels = n_mel_channels
 
         self.mel_stft = torchaudio.transforms.MelSpectrogram(
-            sample_rate=target_sample_rate,
-            n_fft=filter_length,
-            win_length=win_length,
-            hop_length=hop_length,
-            n_mels=n_mel_channels,
-            power=power,
-            center=center,
-            normalized=normalize,
-            norm=norm,
+            sample_rate = target_sample_rate,
+            n_fft = filter_length,
+            win_length = win_length,
+            hop_length = hop_length,
+            n_mels = n_mel_channels,
+            power = power,
+            center = center,
+            normalized = normalize,
+            norm = norm,
         )
 
-        self.register_buffer("dummy", torch.tensor(0), persistent=False)
+        self.register_buffer('dummy', torch.tensor(0), persistent = False)
 
     def forward(self, inp):
         if len(inp.shape) == 3:
-            inp = inp.squeeze(1)  # 'b 1 nw -> b nw'
+            inp = rearrange(inp, 'b 1 nw -> b nw')
 
         assert len(inp.shape) == 2
 
@@ -62,13 +62,12 @@ class MelSpec(nn.Module):
             self.to(inp.device)
 
         mel = self.mel_stft(inp)
-        mel = mel.clamp(min=1e-5).log()
+        mel = mel.clamp(min = 1e-5).log()
         return mel
-
+    
 
 # sinusoidal position embedding
 
-
 class SinusPositionEmbedding(nn.Module):
     def __init__(self, dim):
         super().__init__()
@@ -86,37 +85,35 @@ class SinusPositionEmbedding(nn.Module):
 
 # convolutional position embedding
 
-
 class ConvPositionEmbedding(nn.Module):
-    def __init__(self, dim, kernel_size=31, groups=16):
+    def __init__(self, dim, kernel_size = 31, groups = 16):
         super().__init__()
         assert kernel_size % 2 != 0
         self.conv1d = nn.Sequential(
-            nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=kernel_size // 2),
+            nn.Conv1d(dim, dim, kernel_size, groups = groups, padding = kernel_size // 2),
             nn.Mish(),
-            nn.Conv1d(dim, dim, kernel_size, groups=groups, padding=kernel_size // 2),
+            nn.Conv1d(dim, dim, kernel_size, groups = groups, padding = kernel_size // 2),
             nn.Mish(),
         )
 
-    def forward(self, x: float["b n d"], mask: bool["b n"] | None = None):  # noqa: F722
+    def forward(self, x: float['b n d'], mask: bool['b n'] | None  = None):
         if mask is not None:
             mask = mask[..., None]
-            x = x.masked_fill(~mask, 0.0)
+            x = x.masked_fill(~mask, 0.)
 
-        x = x.permute(0, 2, 1)
+        x = rearrange(x, 'b n d -> b d n')
         x = self.conv1d(x)
-        out = x.permute(0, 2, 1)
+        out = rearrange(x, 'b d n -> b n d')
 
         if mask is not None:
-            out = out.masked_fill(~mask, 0.0)
+            out = out.masked_fill(~mask, 0.)
 
         return out
 
 
 # rotary positional embedding related
 
-
-def precompute_freqs_cis(dim: int, end: int, theta: float = 10000.0, theta_rescale_factor=1.0):
+def precompute_freqs_cis(dim: int, end: int, theta: float = 10000.0, theta_rescale_factor=1.):
     # proposed by reddit user bloc97, to rescale rotary embeddings to longer sequence length without fine-tuning
     # has some connection to NTK literature
     # https://www.reddit.com/r/LocalLLaMA/comments/14lz7j5/ntkaware_scaled_rope_allows_llama_models_to_have/
@@ -129,14 +126,12 @@ def precompute_freqs_cis(dim: int, end: int, theta: float = 10000.0, theta_resca
     freqs_sin = torch.sin(freqs)  # imaginary part
     return torch.cat([freqs_cos, freqs_sin], dim=-1)
 
-
-def get_pos_embed_indices(start, length, max_pos, scale=1.0):
+def get_pos_embed_indices(start, length, max_pos, scale=1.):
     # length = length if isinstance(length, int) else length.max()
     scale = scale * torch.ones_like(start, dtype=torch.float32)  # in case scale is a scalar
-    pos = (
-        start.unsqueeze(1)
-        + (torch.arange(length, device=start.device, dtype=torch.float32).unsqueeze(0) * scale.unsqueeze(1)).long()
-    )
+    pos = start.unsqueeze(1) + (
+            torch.arange(length, device=start.device, dtype=torch.float32).unsqueeze(0) *
+            scale.unsqueeze(1)).long()
     # avoid extra long error.
     pos = torch.where(pos < max_pos, pos, max_pos - 1)
     return pos
@@ -144,7 +139,6 @@ def get_pos_embed_indices(start, length, max_pos, scale=1.0):
 
 # Global Response Normalization layer (Instance Normalization ?)
 
-
 class GRN(nn.Module):
     def __init__(self, dim):
         super().__init__()
@@ -160,7 +154,6 @@ class GRN(nn.Module):
 # ConvNeXt-V2 Block https://github.com/facebookresearch/ConvNeXt-V2/blob/main/models/convnextv2.py
 # ref: https://github.com/bfs18/e2_tts/blob/main/rfwave/modules.py#L108
 
-
 class ConvNeXtV2Block(nn.Module):
     def __init__(
         self,
@@ -170,9 +163,7 @@ class ConvNeXtV2Block(nn.Module):
     ):
         super().__init__()
         padding = (dilation * (7 - 1)) // 2
-        self.dwconv = nn.Conv1d(
-            dim, dim, kernel_size=7, padding=padding, groups=dim, dilation=dilation
-        )  # depthwise conv
+        self.dwconv = nn.Conv1d(dim, dim, kernel_size=7, padding=padding, groups=dim, dilation=dilation)  # depthwise conv
         self.norm = nn.LayerNorm(dim, eps=1e-6)
         self.pwconv1 = nn.Linear(dim, intermediate_dim)  # pointwise/1x1 convs, implemented with linear layers
         self.act = nn.GELU()
@@ -195,7 +186,6 @@ class ConvNeXtV2Block(nn.Module):
 # AdaLayerNormZero
 # return with modulated x for attn input, and params for later mlp modulation
 
-
 class AdaLayerNormZero(nn.Module):
     def __init__(self, dim):
         super().__init__()
@@ -205,7 +195,7 @@ class AdaLayerNormZero(nn.Module):
 
         self.norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
 
-    def forward(self, x, emb=None):
+    def forward(self, x, emb = None):
         emb = self.linear(self.silu(emb))
         shift_msa, scale_msa, gate_msa, shift_mlp, scale_mlp, gate_mlp = torch.chunk(emb, 6, dim=1)
 
@@ -216,7 +206,6 @@ class AdaLayerNormZero(nn.Module):
 # AdaLayerNormZero for final layer
 # return only with modulated x for attn input, cuz no more mlp modulation
 
-
 class AdaLayerNormZero_Final(nn.Module):
     def __init__(self, dim):
         super().__init__()
@@ -236,16 +225,22 @@ class AdaLayerNormZero_Final(nn.Module):
 
 # FeedForward
 
-
 class FeedForward(nn.Module):
-    def __init__(self, dim, dim_out=None, mult=4, dropout=0.0, approximate: str = "none"):
+    def __init__(self, dim, dim_out = None, mult = 4, dropout = 0., approximate: str = 'none'):
         super().__init__()
         inner_dim = int(dim * mult)
         dim_out = dim_out if dim_out is not None else dim
 
         activation = nn.GELU(approximate=approximate)
-        project_in = nn.Sequential(nn.Linear(dim, inner_dim), activation)
-        self.ff = nn.Sequential(project_in, nn.Dropout(dropout), nn.Linear(inner_dim, dim_out))
+        project_in = nn.Sequential(
+            nn.Linear(dim, inner_dim),
+            activation
+        )
+        self.ff = nn.Sequential(
+            project_in,
+            nn.Dropout(dropout),
+            nn.Linear(inner_dim, dim_out)
+        )
 
     def forward(self, x):
         return self.ff(x)
@@ -254,7 +249,6 @@ class FeedForward(nn.Module):
 # Attention with possible joint part
 # modified from diffusers/src/diffusers/models/attention_processor.py
 
-
 class Attention(nn.Module):
     def __init__(
         self,
@@ -263,8 +257,8 @@ class Attention(nn.Module):
         heads: int = 8,
         dim_head: int = 64,
         dropout: float = 0.0,
-        context_dim: Optional[int] = None,  # if not None -> joint attention
-        context_pre_only=None,
+        context_dim: Optional[int] = None, # if not None -> joint attention
+        context_pre_only = None,
     ):
         super().__init__()
 
@@ -300,21 +294,20 @@ class Attention(nn.Module):
 
     def forward(
         self,
-        x: float["b n d"],  # noised input x  # noqa: F722
-        c: float["b n d"] = None,  # context c  # noqa: F722
-        mask: bool["b n"] | None = None,  # noqa: F722
-        rope=None,  # rotary position embedding for x
-        c_rope=None,  # rotary position embedding for c
+        x: float['b n d'], # noised input x
+        c: float['b n d'] = None,  # context c
+        mask: bool['b n'] | None = None,
+        rope = None,  # rotary position embedding for x
+        c_rope = None,  # rotary position embedding for c
     ) -> torch.Tensor:
         if c is not None:
-            return self.processor(self, x, c=c, mask=mask, rope=rope, c_rope=c_rope)
+            return self.processor(self, x, c = c, mask = mask, rope = rope, c_rope = c_rope)
         else:
-            return self.processor(self, x, mask=mask, rope=rope)
+            return self.processor(self, x, mask = mask, rope = rope)
 
 
 # Attention processor
 
-
 class AttnProcessor:
     def __init__(self):
         pass
@@ -322,10 +315,11 @@ class AttnProcessor:
     def __call__(
         self,
         attn: Attention,
-        x: float["b n d"],  # noised input x  # noqa: F722
-        mask: bool["b n"] | None = None,  # noqa: F722
-        rope=None,  # rotary position embedding
+        x: float['b n d'], # noised input x
+        mask: bool['b n'] | None = None,
+        rope = None,  # rotary position embedding
     ) -> torch.FloatTensor:
+
         batch_size = x.shape[0]
 
         # `sample` projections.
@@ -336,7 +330,7 @@ class AttnProcessor:
         # apply rotary position embedding
         if rope is not None:
             freqs, xpos_scale = rope
-            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
+            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale ** -1.) if xpos_scale is not None else (1., 1.)
 
             query = apply_rotary_pos_emb(query, freqs, q_xpos_scale)
             key = apply_rotary_pos_emb(key, freqs, k_xpos_scale)
@@ -351,7 +345,7 @@ class AttnProcessor:
         # mask. e.g. inference got a batch with different target durations, mask out the padding
         if mask is not None:
             attn_mask = mask
-            attn_mask = attn_mask.unsqueeze(1).unsqueeze(1)  # 'b n -> b 1 1 n'
+            attn_mask = rearrange(attn_mask, 'b n -> b 1 1 n')
             attn_mask = attn_mask.expand(batch_size, attn.heads, query.shape[-2], key.shape[-2])
         else:
             attn_mask = None
@@ -366,16 +360,15 @@ class AttnProcessor:
         x = attn.to_out[1](x)
 
         if mask is not None:
-            mask = mask.unsqueeze(-1)
-            x = x.masked_fill(~mask, 0.0)
+            mask = rearrange(mask, 'b n -> b n 1')
+            x = x.masked_fill(~mask, 0.)
 
         return x
-
+    
 
 # Joint Attention processor for MM-DiT
 # modified from diffusers/src/diffusers/models/attention_processor.py
 
-
 class JointAttnProcessor:
     def __init__(self):
         pass
@@ -383,11 +376,11 @@ class JointAttnProcessor:
     def __call__(
         self,
         attn: Attention,
-        x: float["b n d"],  # noised input x  # noqa: F722
-        c: float["b nt d"] = None,  # context c, here text # noqa: F722
-        mask: bool["b n"] | None = None,  # noqa: F722
-        rope=None,  # rotary position embedding for x
-        c_rope=None,  # rotary position embedding for c
+        x: float['b n d'], # noised input x
+        c: float['b nt d'] = None,  # context c, here text
+        mask: bool['b n'] | None = None,
+        rope = None,  # rotary position embedding for x
+        c_rope = None,  # rotary position embedding for c
     ) -> torch.FloatTensor:
         residual = x
 
@@ -406,12 +399,12 @@ class JointAttnProcessor:
         # apply rope for context and noised input independently
         if rope is not None:
             freqs, xpos_scale = rope
-            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
+            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale ** -1.) if xpos_scale is not None else (1., 1.)
             query = apply_rotary_pos_emb(query, freqs, q_xpos_scale)
             key = apply_rotary_pos_emb(key, freqs, k_xpos_scale)
         if c_rope is not None:
             freqs, xpos_scale = c_rope
-            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale**-1.0) if xpos_scale is not None else (1.0, 1.0)
+            q_xpos_scale, k_xpos_scale = (xpos_scale, xpos_scale ** -1.) if xpos_scale is not None else (1., 1.)
             c_query = apply_rotary_pos_emb(c_query, freqs, q_xpos_scale)
             c_key = apply_rotary_pos_emb(c_key, freqs, k_xpos_scale)
 
@@ -428,8 +421,8 @@ class JointAttnProcessor:
 
         # mask. e.g. inference got a batch with different target durations, mask out the padding
         if mask is not None:
-            attn_mask = F.pad(mask, (0, c.shape[1]), value=True)  # no mask for c (text)
-            attn_mask = attn_mask.unsqueeze(1).unsqueeze(1)  # 'b n -> b 1 1 n'
+            attn_mask = F.pad(mask, (0, c.shape[1]), value = True)  # no mask for c (text)
+            attn_mask = rearrange(attn_mask, 'b n -> b 1 1 n')
             attn_mask = attn_mask.expand(batch_size, attn.heads, query.shape[-2], key.shape[-2])
         else:
             attn_mask = None
@@ -440,8 +433,8 @@ class JointAttnProcessor:
 
         # Split the attention outputs.
         x, c = (
-            x[:, : residual.shape[1]],
-            x[:, residual.shape[1] :],
+            x[:, :residual.shape[1]],
+            x[:, residual.shape[1]:],
         )
 
         # linear proj
@@ -452,8 +445,8 @@ class JointAttnProcessor:
             c = attn.to_out_c(c)
 
         if mask is not None:
-            mask = mask.unsqueeze(-1)
-            x = x.masked_fill(~mask, 0.0)
+            mask = rearrange(mask, 'b n -> b n 1')
+            x = x.masked_fill(~mask, 0.)
             # c = c.masked_fill(~mask, 0.)  # no mask for c (text)
 
         return x, c
@@ -461,24 +454,24 @@ class JointAttnProcessor:
 
 # DiT Block
 
-
 class DiTBlock(nn.Module):
-    def __init__(self, dim, heads, dim_head, ff_mult=4, dropout=0.1):
-        super().__init__()
 
+    def __init__(self, dim, heads, dim_head, ff_mult = 4, dropout = 0.1):
+        super().__init__()
+        
         self.attn_norm = AdaLayerNormZero(dim)
         self.attn = Attention(
-            processor=AttnProcessor(),
-            dim=dim,
-            heads=heads,
-            dim_head=dim_head,
-            dropout=dropout,
-        )
-
+            processor = AttnProcessor(),
+            dim = dim,
+            heads = heads,
+            dim_head = dim_head,
+            dropout = dropout,
+            )
+        
         self.ff_norm = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
-        self.ff = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
+        self.ff = FeedForward(dim = dim, mult = ff_mult, dropout = dropout, approximate = "tanh")
 
-    def forward(self, x, t, mask=None, rope=None):  # x: noised input, t: time embedding
+    def forward(self, x, t, mask = None, rope = None): # x: noised input, t: time embedding
         # pre-norm & modulation for attention input
         norm, gate_msa, shift_mlp, scale_mlp, gate_mlp = self.attn_norm(x, emb=t)
 
@@ -487,7 +480,7 @@ class DiTBlock(nn.Module):
 
         # process attention output for input x
         x = x + gate_msa.unsqueeze(1) * attn_output
-
+        
         norm = self.ff_norm(x) * (1 + scale_mlp[:, None]) + shift_mlp[:, None]
         ff_output = self.ff(norm)
         x = x + gate_mlp.unsqueeze(1) * ff_output
@@ -497,9 +490,8 @@ class DiTBlock(nn.Module):
 
 # MMDiT Block https://arxiv.org/abs/2403.03206
 
-
 class MMDiTBlock(nn.Module):
-    r"""
+    r""" 
     modified from diffusers/src/diffusers/models/attention.py
 
     notes.
@@ -508,33 +500,33 @@ class MMDiTBlock(nn.Module):
     context_pre_only: last layer only do prenorm + modulation cuz no more ffn
     """
 
-    def __init__(self, dim, heads, dim_head, ff_mult=4, dropout=0.1, context_pre_only=False):
+    def __init__(self, dim, heads, dim_head, ff_mult = 4, dropout = 0.1, context_pre_only = False):
         super().__init__()
 
         self.context_pre_only = context_pre_only
-
+        
         self.attn_norm_c = AdaLayerNormZero_Final(dim) if context_pre_only else AdaLayerNormZero(dim)
         self.attn_norm_x = AdaLayerNormZero(dim)
         self.attn = Attention(
-            processor=JointAttnProcessor(),
-            dim=dim,
-            heads=heads,
-            dim_head=dim_head,
-            dropout=dropout,
-            context_dim=dim,
-            context_pre_only=context_pre_only,
-        )
+            processor = JointAttnProcessor(),
+            dim = dim,
+            heads = heads,
+            dim_head = dim_head,
+            dropout = dropout,
+            context_dim = dim,
+            context_pre_only = context_pre_only,
+            )
 
         if not context_pre_only:
             self.ff_norm_c = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
-            self.ff_c = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
+            self.ff_c = FeedForward(dim = dim, mult = ff_mult, dropout = dropout, approximate = "tanh")
         else:
             self.ff_norm_c = None
             self.ff_c = None
         self.ff_norm_x = nn.LayerNorm(dim, elementwise_affine=False, eps=1e-6)
-        self.ff_x = FeedForward(dim=dim, mult=ff_mult, dropout=dropout, approximate="tanh")
+        self.ff_x = FeedForward(dim = dim, mult = ff_mult, dropout = dropout, approximate = "tanh")
 
-    def forward(self, x, c, t, mask=None, rope=None, c_rope=None):  # x: noised input, c: context, t: time embedding
+    def forward(self, x, c, t, mask = None, rope = None, c_rope = None): # x: noised input, c: context, t: time embedding
         # pre-norm & modulation for attention input
         if self.context_pre_only:
             norm_c = self.attn_norm_c(c, t)
@@ -548,7 +540,7 @@ class MMDiTBlock(nn.Module):
         # process attention output for context c
         if self.context_pre_only:
             c = None
-        else:  # if not last layer
+        else: # if not last layer
             c = c + c_gate_msa.unsqueeze(1) * c_attn_output
 
             norm_c = self.ff_norm_c(c) * (1 + c_scale_mlp[:, None]) + c_shift_mlp[:, None]
@@ -557,7 +549,7 @@ class MMDiTBlock(nn.Module):
 
         # process attention output for input x
         x = x + x_gate_msa.unsqueeze(1) * x_attn_output
-
+        
         norm_x = self.ff_norm_x(x) * (1 + x_scale_mlp[:, None]) + x_shift_mlp[:, None]
         x_ff_output = self.ff_x(norm_x)
         x = x + x_gate_mlp.unsqueeze(1) * x_ff_output
@@ -567,15 +559,17 @@ class MMDiTBlock(nn.Module):
 
 # time step conditioning embedding
 
-
 class TimestepEmbedding(nn.Module):
     def __init__(self, dim, freq_embed_dim=256):
         super().__init__()
         self.time_embed = SinusPositionEmbedding(freq_embed_dim)
-        self.time_mlp = nn.Sequential(nn.Linear(freq_embed_dim, dim), nn.SiLU(), nn.Linear(dim, dim))
+        self.time_mlp = nn.Sequential(
+            nn.Linear(freq_embed_dim, dim),
+            nn.SiLU(),
+            nn.Linear(dim, dim)
+        )
 
-    def forward(self, timestep: float["b"]):  # noqa: F821
+    def forward(self, timestep: float['b']):
         time_hidden = self.time_embed(timestep)
-        time_hidden = time_hidden.to(timestep.dtype)
         time = self.time_mlp(time_hidden)  # b d
         return time

model/trainer.py

@@ -10,6 +10,8 @@ from torch.optim import AdamW
 from torch.utils.data import DataLoader, Dataset, SequentialSampler
 from torch.optim.lr_scheduler import LinearLR, SequentialLR
 
+from einops import rearrange
+
 from accelerate import Accelerator
 from accelerate.utils import DistributedDataParallelKwargs
 
@@ -22,69 +24,66 @@ from model.dataset import DynamicBatchSampler, collate_fn
 
 # trainer
 
-
 class Trainer:
     def __init__(
         self,
         model: CFM,
         epochs,
         learning_rate,
-        num_warmup_updates=20000,
-        save_per_updates=1000,
-        checkpoint_path=None,
-        batch_size=32,
+        num_warmup_updates = 20000,
+        save_per_updates = 1000, 
+        checkpoint_path = None,
+        batch_size = 32, 
         batch_size_type: str = "sample",
-        max_samples=32,
-        grad_accumulation_steps=1,
-        max_grad_norm=1.0,
+        max_samples = 32,
+        grad_accumulation_steps = 1,
+        max_grad_norm = 1.0,
         noise_scheduler: str | None = None,
         duration_predictor: torch.nn.Module | None = None,
-        wandb_project="test_e2-tts",
-        wandb_run_name="test_run",
+        wandb_project = "test_e2-tts",
+        wandb_run_name = "test_run",
         wandb_resume_id: str = None,
-        last_per_steps=None,
+        last_per_steps = None,
         accelerate_kwargs: dict = dict(),
-        ema_kwargs: dict = dict(),
-        bnb_optimizer: bool = False,
+        ema_kwargs: dict = dict()
     ):
-        ddp_kwargs = DistributedDataParallelKwargs(find_unused_parameters=True)
-
-        logger = "wandb" if wandb.api.api_key else None
-        print(f"Using logger: {logger}")
+        
+        ddp_kwargs = DistributedDataParallelKwargs(find_unused_parameters = True)
 
         self.accelerator = Accelerator(
-            log_with=logger,
-            kwargs_handlers=[ddp_kwargs],
-            gradient_accumulation_steps=grad_accumulation_steps,
-            **accelerate_kwargs,
+            log_with = "wandb",
+            kwargs_handlers = [ddp_kwargs],
+            gradient_accumulation_steps = grad_accumulation_steps,
+            **accelerate_kwargs
         )
-
-        if logger == "wandb":
-            if exists(wandb_resume_id):
-                init_kwargs = {"wandb": {"resume": "allow", "name": wandb_run_name, "id": wandb_resume_id}}
-            else:
-                init_kwargs = {"wandb": {"resume": "allow", "name": wandb_run_name}}
-            self.accelerator.init_trackers(
-                project_name=wandb_project,
-                init_kwargs=init_kwargs,
-                config={
-                    "epochs": epochs,
+        
+        if exists(wandb_resume_id):
+            init_kwargs={"wandb": {"resume": "allow", "name": wandb_run_name, 'id': wandb_resume_id}}
+        else:
+            init_kwargs={"wandb": {"resume": "allow", "name": wandb_run_name}}
+        self.accelerator.init_trackers(
+            project_name = wandb_project, 
+            init_kwargs=init_kwargs,
+            config={"epochs": epochs,
                     "learning_rate": learning_rate,
-                    "num_warmup_updates": num_warmup_updates,
+                    "num_warmup_updates": num_warmup_updates, 
                     "batch_size": batch_size,
                     "batch_size_type": batch_size_type,
                     "max_samples": max_samples,
                     "grad_accumulation_steps": grad_accumulation_steps,
                     "max_grad_norm": max_grad_norm,
                     "gpus": self.accelerator.num_processes,
-                    "noise_scheduler": noise_scheduler,
-                },
+                    "noise_scheduler": noise_scheduler}
             )
 
         self.model = model
 
         if self.is_main:
-            self.ema_model = EMA(model, include_online_model=False, **ema_kwargs)
+            self.ema_model = EMA(
+                model,
+                include_online_model = False,
+                **ema_kwargs
+            )
 
             self.ema_model.to(self.accelerator.device)
 
@@ -92,7 +91,7 @@ class Trainer:
         self.num_warmup_updates = num_warmup_updates
         self.save_per_updates = save_per_updates
         self.last_per_steps = default(last_per_steps, save_per_updates * grad_accumulation_steps)
-        self.checkpoint_path = default(checkpoint_path, "ckpts/test_e2-tts")
+        self.checkpoint_path = default(checkpoint_path, 'ckpts/test_e2-tts')
 
         self.batch_size = batch_size
         self.batch_size_type = batch_size_type
@@ -104,13 +103,10 @@ class Trainer:
 
         self.duration_predictor = duration_predictor
 
-        if bnb_optimizer:
-            import bitsandbytes as bnb
-
-            self.optimizer = bnb.optim.AdamW8bit(model.parameters(), lr=learning_rate)
-        else:
-            self.optimizer = AdamW(model.parameters(), lr=learning_rate)
-        self.model, self.optimizer = self.accelerator.prepare(self.model, self.optimizer)
+        self.optimizer = AdamW(model.parameters(), lr=learning_rate)
+        self.model, self.optimizer = self.accelerator.prepare(
+            self.model, self.optimizer
+        )
 
     @property
     def is_main(self):
@@ -120,112 +116,81 @@ class Trainer:
         self.accelerator.wait_for_everyone()
         if self.is_main:
             checkpoint = dict(
-                model_state_dict=self.accelerator.unwrap_model(self.model).state_dict(),
-                optimizer_state_dict=self.accelerator.unwrap_model(self.optimizer).state_dict(),
-                ema_model_state_dict=self.ema_model.state_dict(),
-                scheduler_state_dict=self.scheduler.state_dict(),
-                step=step,
+                model_state_dict = self.accelerator.unwrap_model(self.model).state_dict(),
+                optimizer_state_dict = self.accelerator.unwrap_model(self.optimizer).state_dict(),
+                ema_model_state_dict = self.ema_model.state_dict(),
+                scheduler_state_dict = self.scheduler.state_dict(),
+                step = step
             )
             if not os.path.exists(self.checkpoint_path):
                 os.makedirs(self.checkpoint_path)
-            if last:
+            if last == True:
                 self.accelerator.save(checkpoint, f"{self.checkpoint_path}/model_last.pt")
                 print(f"Saved last checkpoint at step {step}")
             else:
                 self.accelerator.save(checkpoint, f"{self.checkpoint_path}/model_{step}.pt")
 
     def load_checkpoint(self):
-        if (
-            not exists(self.checkpoint_path)
-            or not os.path.exists(self.checkpoint_path)
-            or not os.listdir(self.checkpoint_path)
-        ):
+        if not exists(self.checkpoint_path) or not os.path.exists(self.checkpoint_path) or not os.listdir(self.checkpoint_path):
             return 0
-
+        
         self.accelerator.wait_for_everyone()
         if "model_last.pt" in os.listdir(self.checkpoint_path):
             latest_checkpoint = "model_last.pt"
         else:
-            latest_checkpoint = sorted(
-                [f for f in os.listdir(self.checkpoint_path) if f.endswith(".pt")],
-                key=lambda x: int("".join(filter(str.isdigit, x))),
-            )[-1]
+            latest_checkpoint = sorted([f for f in os.listdir(self.checkpoint_path) if f.endswith('.pt')], key=lambda x: int(''.join(filter(str.isdigit, x))))[-1]
         # checkpoint = torch.load(f"{self.checkpoint_path}/{latest_checkpoint}", map_location=self.accelerator.device)  # rather use accelerator.load_state ಥ_ಥ
         checkpoint = torch.load(f"{self.checkpoint_path}/{latest_checkpoint}", weights_only=True, map_location="cpu")
 
         if self.is_main:
-            self.ema_model.load_state_dict(checkpoint["ema_model_state_dict"])
+            self.ema_model.load_state_dict(checkpoint['ema_model_state_dict'])
 
-        if "step" in checkpoint:
-            self.accelerator.unwrap_model(self.model).load_state_dict(checkpoint["model_state_dict"])
-            self.accelerator.unwrap_model(self.optimizer).load_state_dict(checkpoint["optimizer_state_dict"])
+        if 'step' in checkpoint:
+            self.accelerator.unwrap_model(self.model).load_state_dict(checkpoint['model_state_dict'])
+            self.accelerator.unwrap_model(self.optimizer).load_state_dict(checkpoint['optimizer_state_dict'])
             if self.scheduler:
-                self.scheduler.load_state_dict(checkpoint["scheduler_state_dict"])
-            step = checkpoint["step"]
+                self.scheduler.load_state_dict(checkpoint['scheduler_state_dict'])
+            step = checkpoint['step']
         else:
-            checkpoint["model_state_dict"] = {
-                k.replace("ema_model.", ""): v
-                for k, v in checkpoint["ema_model_state_dict"].items()
-                if k not in ["initted", "step"]
-            }
-            self.accelerator.unwrap_model(self.model).load_state_dict(checkpoint["model_state_dict"])
+            checkpoint['model_state_dict'] = {k.replace("ema_model.", ""): v for k, v in checkpoint['ema_model_state_dict'].items() if k not in ["initted", "step"]}
+            self.accelerator.unwrap_model(self.model).load_state_dict(checkpoint['model_state_dict'])
             step = 0
 
-        del checkpoint
-        gc.collect()
+        del checkpoint; gc.collect()
         return step
 
     def train(self, train_dataset: Dataset, num_workers=16, resumable_with_seed: int = None):
+        
         if exists(resumable_with_seed):
             generator = torch.Generator()
             generator.manual_seed(resumable_with_seed)
-        else:
+        else: 
             generator = None
 
         if self.batch_size_type == "sample":
-            train_dataloader = DataLoader(
-                train_dataset,
-                collate_fn=collate_fn,
-                num_workers=num_workers,
-                pin_memory=True,
-                persistent_workers=True,
-                batch_size=self.batch_size,
-                shuffle=True,
-                generator=generator,
-            )
+            train_dataloader = DataLoader(train_dataset, collate_fn=collate_fn, num_workers=num_workers, pin_memory=True, persistent_workers=True,
+                                          batch_size=self.batch_size, shuffle=True, generator=generator)
         elif self.batch_size_type == "frame":
             self.accelerator.even_batches = False
             sampler = SequentialSampler(train_dataset)
-            batch_sampler = DynamicBatchSampler(
-                sampler, self.batch_size, max_samples=self.max_samples, random_seed=resumable_with_seed, drop_last=False
-            )
-            train_dataloader = DataLoader(
-                train_dataset,
-                collate_fn=collate_fn,
-                num_workers=num_workers,
-                pin_memory=True,
-                persistent_workers=True,
-                batch_sampler=batch_sampler,
-            )
+            batch_sampler = DynamicBatchSampler(sampler, self.batch_size, max_samples=self.max_samples, random_seed=resumable_with_seed, drop_last=False)
+            train_dataloader = DataLoader(train_dataset, collate_fn=collate_fn, num_workers=num_workers, pin_memory=True, persistent_workers=True,
+                                          batch_sampler=batch_sampler)
         else:
             raise ValueError(f"batch_size_type must be either 'sample' or 'frame', but received {self.batch_size_type}")
-
+        
         #  accelerator.prepare() dispatches batches to devices;
         #  which means the length of dataloader calculated before, should consider the number of devices
-        warmup_steps = (
-            self.num_warmup_updates * self.accelerator.num_processes
-        )  # consider a fixed warmup steps while using accelerate multi-gpu ddp
-        # otherwise by default with split_batches=False, warmup steps change with num_processes
+        warmup_steps = self.num_warmup_updates * self.accelerator.num_processes  # consider a fixed warmup steps while using accelerate multi-gpu ddp
+                                                                                 # otherwise by default with split_batches=False, warmup steps change with num_processes
         total_steps = len(train_dataloader) * self.epochs / self.grad_accumulation_steps
         decay_steps = total_steps - warmup_steps
         warmup_scheduler = LinearLR(self.optimizer, start_factor=1e-8, end_factor=1.0, total_iters=warmup_steps)
         decay_scheduler = LinearLR(self.optimizer, start_factor=1.0, end_factor=1e-8, total_iters=decay_steps)
-        self.scheduler = SequentialLR(
-            self.optimizer, schedulers=[warmup_scheduler, decay_scheduler], milestones=[warmup_steps]
-        )
-        train_dataloader, self.scheduler = self.accelerator.prepare(
-            train_dataloader, self.scheduler
-        )  # actual steps = 1 gpu steps / gpus
+        self.scheduler = SequentialLR(self.optimizer, 
+                                      schedulers=[warmup_scheduler, decay_scheduler],
+                                      milestones=[warmup_steps])
+        train_dataloader, self.scheduler = self.accelerator.prepare(train_dataloader, self.scheduler)  # actual steps = 1 gpu steps / gpus
         start_step = self.load_checkpoint()
         global_step = start_step
 
@@ -240,36 +205,23 @@ class Trainer:
         for epoch in range(skipped_epoch, self.epochs):
             self.model.train()
             if exists(resumable_with_seed) and epoch == skipped_epoch:
-                progress_bar = tqdm(
-                    skipped_dataloader,
-                    desc=f"Epoch {epoch+1}/{self.epochs}",
-                    unit="step",
-                    disable=not self.accelerator.is_local_main_process,
-                    initial=skipped_batch,
-                    total=orig_epoch_step,
-                )
+                progress_bar = tqdm(skipped_dataloader, desc=f"Epoch {epoch+1}/{self.epochs}", unit="step", disable=not self.accelerator.is_local_main_process, 
+                                    initial=skipped_batch, total=orig_epoch_step)
             else:
-                progress_bar = tqdm(
-                    train_dataloader,
-                    desc=f"Epoch {epoch+1}/{self.epochs}",
-                    unit="step",
-                    disable=not self.accelerator.is_local_main_process,
-                )
+                progress_bar = tqdm(train_dataloader, desc=f"Epoch {epoch+1}/{self.epochs}", unit="step", disable=not self.accelerator.is_local_main_process)
 
             for batch in progress_bar:
                 with self.accelerator.accumulate(self.model):
-                    text_inputs = batch["text"]
-                    mel_spec = batch["mel"].permute(0, 2, 1)
+                    text_inputs = batch['text']
+                    mel_spec = rearrange(batch['mel'], 'b d n -> b n d')
                     mel_lengths = batch["mel_lengths"]
 
                     # TODO. add duration predictor training
                     if self.duration_predictor is not None and self.accelerator.is_local_main_process:
-                        dur_loss = self.duration_predictor(mel_spec, lens=batch.get("durations"))
+                        dur_loss = self.duration_predictor(mel_spec, lens=batch.get('durations'))
                         self.accelerator.log({"duration loss": dur_loss.item()}, step=global_step)
 
-                    loss, cond, pred = self.model(
-                        mel_spec, text=text_inputs, lens=mel_lengths, noise_scheduler=self.noise_scheduler
-                    )
+                    loss, cond, pred = self.model(mel_spec, text=text_inputs, lens=mel_lengths, noise_scheduler=self.noise_scheduler)
                     self.accelerator.backward(loss)
 
                     if self.max_grad_norm > 0 and self.accelerator.sync_gradients:
@@ -286,15 +238,13 @@ class Trainer:
 
                 if self.accelerator.is_local_main_process:
                     self.accelerator.log({"loss": loss.item(), "lr": self.scheduler.get_last_lr()[0]}, step=global_step)
-
+                
                 progress_bar.set_postfix(step=str(global_step), loss=loss.item())
-
+                
                 if global_step % (self.save_per_updates * self.grad_accumulation_steps) == 0:
                     self.save_checkpoint(global_step)
-
+                
                 if global_step % self.last_per_steps == 0:
                     self.save_checkpoint(global_step, last=True)
-
-        self.save_checkpoint(global_step, last=True)
-
+        
         self.accelerator.end_training()

model/utils.py

@@ -1,6 +1,7 @@
 from __future__ import annotations
 
 import os
+import re
 import math
 import random
 import string
@@ -8,7 +9,6 @@ from tqdm import tqdm
 from collections import defaultdict
 
 import matplotlib
-
 matplotlib.use("Agg")
 import matplotlib.pylab as plt
 
@@ -17,6 +17,9 @@ import torch.nn.functional as F
 from torch.nn.utils.rnn import pad_sequence
 import torchaudio
 
+import einx
+from einops import rearrange, reduce
+
 import jieba
 from pypinyin import lazy_pinyin, Style
 
@@ -26,102 +29,107 @@ from model.modules import MelSpec
 
 # seed everything
 
-
-def seed_everything(seed=0):
+def seed_everything(seed = 0):
     random.seed(seed)
-    os.environ["PYTHONHASHSEED"] = str(seed)
+    os.environ['PYTHONHASHSEED'] = str(seed)
     torch.manual_seed(seed)
     torch.cuda.manual_seed(seed)
     torch.cuda.manual_seed_all(seed)
     torch.backends.cudnn.deterministic = True
     torch.backends.cudnn.benchmark = False
 
-
 # helpers
 
-
 def exists(v):
     return v is not None
 
-
 def default(v, d):
     return v if exists(v) else d
 
-
 # tensor helpers
 
+def lens_to_mask(
+    t: int['b'],
+    length: int | None = None
+) -> bool['b n']:
 
-def lens_to_mask(t: int["b"], length: int | None = None) -> bool["b n"]:  # noqa: F722 F821
     if not exists(length):
         length = t.amax()
 
-    seq = torch.arange(length, device=t.device)
-    return seq[None, :] < t[:, None]
-
-
-def mask_from_start_end_indices(seq_len: int["b"], start: int["b"], end: int["b"]):  # noqa: F722 F821
-    max_seq_len = seq_len.max().item()
-    seq = torch.arange(max_seq_len, device=start.device).long()
-    start_mask = seq[None, :] >= start[:, None]
-    end_mask = seq[None, :] < end[:, None]
-    return start_mask & end_mask
+    seq = torch.arange(length, device = t.device)
+    return einx.less('n, b -> b n', seq, t)
 
+def mask_from_start_end_indices(
+    seq_len: int['b'],
+    start: int['b'],
+    end: int['b']
+):
+    max_seq_len = seq_len.max().item()  
+    seq = torch.arange(max_seq_len, device = start.device).long()
+    return einx.greater_equal('n, b -> b n', seq, start) & einx.less('n, b -> b n', seq, end)
 
-def mask_from_frac_lengths(seq_len: int["b"], frac_lengths: float["b"]):  # noqa: F722 F821
+def mask_from_frac_lengths(
+    seq_len: int['b'],
+    frac_lengths: float['b']
+):
     lengths = (frac_lengths * seq_len).long()
     max_start = seq_len - lengths
 
     rand = torch.rand_like(frac_lengths)
-    start = (max_start * rand).long().clamp(min=0)
+    start = (max_start * rand).long().clamp(min = 0)
     end = start + lengths
 
     return mask_from_start_end_indices(seq_len, start, end)
 
+def maybe_masked_mean(
+    t: float['b n d'],
+    mask: bool['b n'] = None
+) -> float['b d']:
 
-def maybe_masked_mean(t: float["b n d"], mask: bool["b n"] = None) -> float["b d"]:  # noqa: F722
     if not exists(mask):
-        return t.mean(dim=1)
+        return t.mean(dim = 1)
 
-    t = torch.where(mask[:, :, None], t, torch.tensor(0.0, device=t.device))
-    num = t.sum(dim=1)
-    den = mask.float().sum(dim=1)
+    t = einx.where('b n, b n d, -> b n d', mask, t, 0.)
+    num = reduce(t, 'b n d -> b d', 'sum')
+    den = reduce(mask.float(), 'b n -> b', 'sum')
 
-    return num / den.clamp(min=1.0)
+    return einx.divide('b d, b -> b d', num, den.clamp(min = 1.))
 
 
 # simple utf-8 tokenizer, since paper went character based
-def list_str_to_tensor(text: list[str], padding_value=-1) -> int["b nt"]:  # noqa: F722
-    list_tensors = [torch.tensor([*bytes(t, "UTF-8")]) for t in text]  # ByT5 style
-    text = pad_sequence(list_tensors, padding_value=padding_value, batch_first=True)
+def list_str_to_tensor(
+    text: list[str],
+    padding_value = -1
+) -> int['b nt']:
+    list_tensors = [torch.tensor([*bytes(t, 'UTF-8')]) for t in text]  # ByT5 style
+    text = pad_sequence(list_tensors, padding_value = padding_value, batch_first = True)
     return text
 
-
 # char tokenizer, based on custom dataset's extracted .txt file
 def list_str_to_idx(
     text: list[str] | list[list[str]],
     vocab_char_map: dict[str, int],  # {char: idx}
-    padding_value=-1,
-) -> int["b nt"]:  # noqa: F722
+    padding_value = -1
+) -> int['b nt']:
     list_idx_tensors = [torch.tensor([vocab_char_map.get(c, 0) for c in t]) for t in text]  # pinyin or char style
-    text = pad_sequence(list_idx_tensors, padding_value=padding_value, batch_first=True)
+    text = pad_sequence(list_idx_tensors, padding_value = padding_value, batch_first = True)
     return text
 
 
 # Get tokenizer
 
-
 def get_tokenizer(dataset_name, tokenizer: str = "pinyin"):
-    """
+    ''' 
     tokenizer   - "pinyin" do g2p for only chinese characters, need .txt vocab_file
                 - "char" for char-wise tokenizer, need .txt vocab_file
                 - "byte" for utf-8 tokenizer
                 - "custom" if you're directly passing in a path to the vocab.txt you want to use
     vocab_size  - if use "pinyin", all available pinyin types, common alphabets (also those with accent) and symbols
                 - if use "char", derived from unfiltered character & symbol counts of custom dataset
-                - if use "byte", set to 256 (unicode byte range)
-    """
+                - if use "byte", set to 256 (unicode byte range) 
+    ''' 
     if tokenizer in ["pinyin", "char"]:
-        with open(f"data/{dataset_name}_{tokenizer}/vocab.txt", "r", encoding="utf-8") as f:
+        with open (f"data/{dataset_name}_{tokenizer}/vocab.txt", "r", encoding="utf-8") as f:
             vocab_char_map = {}
             for i, char in enumerate(f):
                 vocab_char_map[char[:-1]] = i
@@ -132,7 +140,7 @@ def get_tokenizer(dataset_name, tokenizer: str = "pinyin"):
         vocab_char_map = None
         vocab_size = 256
     elif tokenizer == "custom":
-        with open(dataset_name, "r", encoding="utf-8") as f:
+        with open (dataset_name, "r", encoding="utf-8") as f:
             vocab_char_map = {}
             for i, char in enumerate(f):
                 vocab_char_map[char[:-1]] = i
@@ -143,19 +151,16 @@ def get_tokenizer(dataset_name, tokenizer: str = "pinyin"):
 
 # convert char to pinyin
 
-
-def convert_char_to_pinyin(text_list, polyphone=True):
+def convert_char_to_pinyin(text_list, polyphone = True):
     final_text_list = []
-    god_knows_why_en_testset_contains_zh_quote = str.maketrans(
-        {"“": '"', "”": '"', "‘": "'", "’": "'"}
-    )  # in case librispeech (orig no-pc) test-clean
-    custom_trans = str.maketrans({";": ","})  # add custom trans here, to address oov
+    god_knows_why_en_testset_contains_zh_quote = str.maketrans({'“': '"', '”': '"', '‘': "'", '’': "'"})  # in case librispeech (orig no-pc) test-clean
+    custom_trans = str.maketrans({';': ','})  # add custom trans here, to address oov
     for text in text_list:
         char_list = []
         text = text.translate(god_knows_why_en_testset_contains_zh_quote)
         text = text.translate(custom_trans)
         for seg in jieba.cut(text):
-            seg_byte_len = len(bytes(seg, "UTF-8"))
+            seg_byte_len = len(bytes(seg, 'UTF-8'))
             if seg_byte_len == len(seg):  # if pure alphabets and symbols
                 if char_list and seg_byte_len > 1 and char_list[-1] not in " :'\"":
                     char_list.append(" ")
@@ -184,7 +189,7 @@ def convert_char_to_pinyin(text_list, polyphone=True):
 # save spectrogram
 def save_spectrogram(spectrogram, path):
     plt.figure(figsize=(12, 4))
-    plt.imshow(spectrogram, origin="lower", aspect="auto")
+    plt.imshow(spectrogram, origin='lower', aspect='auto')
     plt.colorbar()
     plt.savefig(path)
     plt.close()
@@ -192,15 +197,13 @@ def save_spectrogram(spectrogram, path):
 
 # seedtts testset metainfo: utt, prompt_text, prompt_wav, gt_text, gt_wav
 def get_seedtts_testset_metainfo(metalst):
-    f = open(metalst)
-    lines = f.readlines()
-    f.close()
+    f = open(metalst); lines = f.readlines(); f.close()
     metainfo = []
     for line in lines:
-        if len(line.strip().split("|")) == 5:
-            utt, prompt_text, prompt_wav, gt_text, gt_wav = line.strip().split("|")
-        elif len(line.strip().split("|")) == 4:
-            utt, prompt_text, prompt_wav, gt_text = line.strip().split("|")
+        if len(line.strip().split('|')) == 5:
+            utt, prompt_text, prompt_wav, gt_text, gt_wav = line.strip().split('|')
+        elif len(line.strip().split('|')) == 4:
+            utt, prompt_text, prompt_wav, gt_text = line.strip().split('|')
             gt_wav = os.path.join(os.path.dirname(metalst), "wavs", utt + ".wav")
         if not os.path.isabs(prompt_wav):
             prompt_wav = os.path.join(os.path.dirname(metalst), prompt_wav)
@@ -210,20 +213,18 @@ def get_seedtts_testset_metainfo(metalst):
 
 # librispeech test-clean metainfo: gen_utt, ref_txt, ref_wav, gen_txt, gen_wav
 def get_librispeech_test_clean_metainfo(metalst, librispeech_test_clean_path):
-    f = open(metalst)
-    lines = f.readlines()
-    f.close()
+    f = open(metalst); lines = f.readlines(); f.close()
     metainfo = []
     for line in lines:
-        ref_utt, ref_dur, ref_txt, gen_utt, gen_dur, gen_txt = line.strip().split("\t")
+        ref_utt, ref_dur, ref_txt, gen_utt, gen_dur, gen_txt = line.strip().split('\t')
 
         # ref_txt = ref_txt[0] + ref_txt[1:].lower() + '.'  # if use librispeech test-clean (no-pc)
-        ref_spk_id, ref_chaptr_id, _ = ref_utt.split("-")
-        ref_wav = os.path.join(librispeech_test_clean_path, ref_spk_id, ref_chaptr_id, ref_utt + ".flac")
+        ref_spk_id, ref_chaptr_id, _ =  ref_utt.split('-')
+        ref_wav = os.path.join(librispeech_test_clean_path, ref_spk_id, ref_chaptr_id, ref_utt + '.flac')
 
         # gen_txt = gen_txt[0] + gen_txt[1:].lower() + '.'  # if use librispeech test-clean (no-pc)
-        gen_spk_id, gen_chaptr_id, _ = gen_utt.split("-")
-        gen_wav = os.path.join(librispeech_test_clean_path, gen_spk_id, gen_chaptr_id, gen_utt + ".flac")
+        gen_spk_id, gen_chaptr_id, _ =  gen_utt.split('-')
+        gen_wav = os.path.join(librispeech_test_clean_path, gen_spk_id, gen_chaptr_id, gen_utt + '.flac')
 
         metainfo.append((gen_utt, ref_txt, ref_wav, " " + gen_txt, gen_wav))
 
@@ -235,30 +236,21 @@ def padded_mel_batch(ref_mels):
     max_mel_length = torch.LongTensor([mel.shape[-1] for mel in ref_mels]).amax()
     padded_ref_mels = []
     for mel in ref_mels:
-        padded_ref_mel = F.pad(mel, (0, max_mel_length - mel.shape[-1]), value=0)
+        padded_ref_mel = F.pad(mel, (0, max_mel_length - mel.shape[-1]), value = 0)
         padded_ref_mels.append(padded_ref_mel)
     padded_ref_mels = torch.stack(padded_ref_mels)
-    padded_ref_mels = padded_ref_mels.permute(0, 2, 1)
+    padded_ref_mels = rearrange(padded_ref_mels, 'b d n -> b n d')
     return padded_ref_mels
 
 
 # get prompts from metainfo containing: utt, prompt_text, prompt_wav, gt_text, gt_wav
 
-
 def get_inference_prompt(
-    metainfo,
-    speed=1.0,
-    tokenizer="pinyin",
-    polyphone=True,
-    target_sample_rate=24000,
-    n_mel_channels=100,
-    hop_length=256,
-    target_rms=0.1,
-    use_truth_duration=False,
-    infer_batch_size=1,
-    num_buckets=200,
-    min_secs=3,
-    max_secs=40,
+    metainfo, 
+    speed = 1., tokenizer = "pinyin", polyphone = True, 
+    target_sample_rate = 24000, n_mel_channels = 100, hop_length = 256, target_rms = 0.1,
+    use_truth_duration = False,
+    infer_batch_size = 1, num_buckets = 200, min_secs = 3, max_secs = 40,
 ):
     prompts_all = []
 
@@ -266,15 +258,13 @@ def get_inference_prompt(
     max_tokens = max_secs * target_sample_rate // hop_length
 
     batch_accum = [0] * num_buckets
-    utts, ref_rms_list, ref_mels, ref_mel_lens, total_mel_lens, final_text_list = (
-        [[] for _ in range(num_buckets)] for _ in range(6)
-    )
+    utts, ref_rms_list, ref_mels, ref_mel_lens, total_mel_lens, final_text_list = \
+        ([[] for _ in range(num_buckets)] for _ in range(6))
 
-    mel_spectrogram = MelSpec(
-        target_sample_rate=target_sample_rate, n_mel_channels=n_mel_channels, hop_length=hop_length
-    )
+    mel_spectrogram = MelSpec(target_sample_rate=target_sample_rate, n_mel_channels=n_mel_channels, hop_length=hop_length)
 
     for utt, prompt_text, prompt_wav, gt_text, gt_wav in tqdm(metainfo, desc="Processing prompts..."):
+
         # Audio
         ref_audio, ref_sr = torchaudio.load(prompt_wav)
         ref_rms = torch.sqrt(torch.mean(torch.square(ref_audio)))
@@ -286,11 +276,11 @@ def get_inference_prompt(
             ref_audio = resampler(ref_audio)
 
         # Text
-        if len(prompt_text[-1].encode("utf-8")) == 1:
+        if len(prompt_text[-1].encode('utf-8')) == 1:
             prompt_text = prompt_text + " "
         text = [prompt_text + gt_text]
         if tokenizer == "pinyin":
-            text_list = convert_char_to_pinyin(text, polyphone=polyphone)
+            text_list = convert_char_to_pinyin(text, polyphone = polyphone)
         else:
             text_list = text
 
@@ -306,19 +296,19 @@ def get_inference_prompt(
             # # test vocoder resynthesis
             # ref_audio = gt_audio
         else:
-            ref_text_len = len(prompt_text.encode("utf-8"))
-            gen_text_len = len(gt_text.encode("utf-8"))
+            zh_pause_punc = r"。，、；：？！"
+            ref_text_len = len(prompt_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, prompt_text))
+            gen_text_len = len(gt_text.encode('utf-8')) + 3 * len(re.findall(zh_pause_punc, gt_text))
             total_mel_len = ref_mel_len + int(ref_mel_len / ref_text_len * gen_text_len / speed)
 
         # to mel spectrogram
         ref_mel = mel_spectrogram(ref_audio)
-        ref_mel = ref_mel.squeeze(0)
+        ref_mel = rearrange(ref_mel, '1 d n -> d n')
 
         # deal with batch
         assert infer_batch_size > 0, "infer_batch_size should be greater than 0."
-        assert (
-            min_tokens <= total_mel_len <= max_tokens
-        ), f"Audio {utt} has duration {total_mel_len*hop_length//target_sample_rate}s out of range [{min_secs}, {max_secs}]."
+        assert min_tokens <= total_mel_len <= max_tokens, \
+            f"Audio {utt} has duration {total_mel_len*hop_length//target_sample_rate}s out of range [{min_secs}, {max_secs}]."
         bucket_i = math.floor((total_mel_len - min_tokens) / (max_tokens - min_tokens + 1) * num_buckets)
 
         utts[bucket_i].append(utt)
@@ -332,39 +322,28 @@ def get_inference_prompt(
 
         if batch_accum[bucket_i] >= infer_batch_size:
             # print(f"\n{len(ref_mels[bucket_i][0][0])}\n{ref_mel_lens[bucket_i]}\n{total_mel_lens[bucket_i]}")
-            prompts_all.append(
-                (
-                    utts[bucket_i],
-                    ref_rms_list[bucket_i],
-                    padded_mel_batch(ref_mels[bucket_i]),
-                    ref_mel_lens[bucket_i],
-                    total_mel_lens[bucket_i],
-                    final_text_list[bucket_i],
-                )
-            )
+            prompts_all.append((
+                utts[bucket_i], 
+                ref_rms_list[bucket_i], 
+                padded_mel_batch(ref_mels[bucket_i]), 
+                ref_mel_lens[bucket_i], 
+                total_mel_lens[bucket_i], 
+                final_text_list[bucket_i]
+            ))
             batch_accum[bucket_i] = 0
-            (
-                utts[bucket_i],
-                ref_rms_list[bucket_i],
-                ref_mels[bucket_i],
-                ref_mel_lens[bucket_i],
-                total_mel_lens[bucket_i],
-                final_text_list[bucket_i],
-            ) = [], [], [], [], [], []
+            utts[bucket_i], ref_rms_list[bucket_i], ref_mels[bucket_i], ref_mel_lens[bucket_i], total_mel_lens[bucket_i], final_text_list[bucket_i] = [], [], [], [], [], []
 
     # add residual
     for bucket_i, bucket_frames in enumerate(batch_accum):
         if bucket_frames > 0:
-            prompts_all.append(
-                (
-                    utts[bucket_i],
-                    ref_rms_list[bucket_i],
-                    padded_mel_batch(ref_mels[bucket_i]),
-                    ref_mel_lens[bucket_i],
-                    total_mel_lens[bucket_i],
-                    final_text_list[bucket_i],
-                )
-            )
+            prompts_all.append((
+                utts[bucket_i], 
+                ref_rms_list[bucket_i], 
+                padded_mel_batch(ref_mels[bucket_i]), 
+                ref_mel_lens[bucket_i], 
+                total_mel_lens[bucket_i], 
+                final_text_list[bucket_i]
+            ))
     # not only leave easy work for last workers
     random.seed(666)
     random.shuffle(prompts_all)
@@ -375,7 +354,6 @@ def get_inference_prompt(
 # get wav_res_ref_text of seed-tts test metalst
 # https://github.com/BytedanceSpeech/seed-tts-eval
 
-
 def get_seed_tts_test(metalst, gen_wav_dir, gpus):
     f = open(metalst)
     lines = f.readlines()
@@ -383,14 +361,14 @@ def get_seed_tts_test(metalst, gen_wav_dir, gpus):
 
     test_set_ = []
     for line in tqdm(lines):
-        if len(line.strip().split("|")) == 5:
-            utt, prompt_text, prompt_wav, gt_text, gt_wav = line.strip().split("|")
-        elif len(line.strip().split("|")) == 4:
-            utt, prompt_text, prompt_wav, gt_text = line.strip().split("|")
+        if len(line.strip().split('|')) == 5:
+            utt, prompt_text, prompt_wav, gt_text, gt_wav = line.strip().split('|')
+        elif len(line.strip().split('|')) == 4:
+            utt, prompt_text, prompt_wav, gt_text = line.strip().split('|')
 
-        if not os.path.exists(os.path.join(gen_wav_dir, utt + ".wav")):
+        if not os.path.exists(os.path.join(gen_wav_dir, utt + '.wav')):
             continue
-        gen_wav = os.path.join(gen_wav_dir, utt + ".wav")
+        gen_wav = os.path.join(gen_wav_dir, utt + '.wav')
         if not os.path.isabs(prompt_wav):
             prompt_wav = os.path.join(os.path.dirname(metalst), prompt_wav)
 
@@ -399,69 +377,65 @@ def get_seed_tts_test(metalst, gen_wav_dir, gpus):
     num_jobs = len(gpus)
     if num_jobs == 1:
         return [(gpus[0], test_set_)]
-
+    
     wav_per_job = len(test_set_) // num_jobs + 1
     test_set = []
     for i in range(num_jobs):
-        test_set.append((gpus[i], test_set_[i * wav_per_job : (i + 1) * wav_per_job]))
+        test_set.append((gpus[i], test_set_[i*wav_per_job:(i+1)*wav_per_job]))
 
     return test_set
 
 
 # get librispeech test-clean cross sentence test
 
-
-def get_librispeech_test(metalst, gen_wav_dir, gpus, librispeech_test_clean_path, eval_ground_truth=False):
+def get_librispeech_test(metalst, gen_wav_dir, gpus, librispeech_test_clean_path, eval_ground_truth = False):
     f = open(metalst)
     lines = f.readlines()
     f.close()
 
     test_set_ = []
     for line in tqdm(lines):
-        ref_utt, ref_dur, ref_txt, gen_utt, gen_dur, gen_txt = line.strip().split("\t")
+        ref_utt, ref_dur, ref_txt, gen_utt, gen_dur, gen_txt = line.strip().split('\t')
 
         if eval_ground_truth:
-            gen_spk_id, gen_chaptr_id, _ = gen_utt.split("-")
-            gen_wav = os.path.join(librispeech_test_clean_path, gen_spk_id, gen_chaptr_id, gen_utt + ".flac")
+            gen_spk_id, gen_chaptr_id, _ =  gen_utt.split('-')
+            gen_wav = os.path.join(librispeech_test_clean_path, gen_spk_id, gen_chaptr_id, gen_utt + '.flac')
         else:
-            if not os.path.exists(os.path.join(gen_wav_dir, gen_utt + ".wav")):
+            if not os.path.exists(os.path.join(gen_wav_dir, gen_utt + '.wav')):
                 raise FileNotFoundError(f"Generated wav not found: {gen_utt}")
-            gen_wav = os.path.join(gen_wav_dir, gen_utt + ".wav")
+            gen_wav = os.path.join(gen_wav_dir, gen_utt + '.wav')
 
-        ref_spk_id, ref_chaptr_id, _ = ref_utt.split("-")
-        ref_wav = os.path.join(librispeech_test_clean_path, ref_spk_id, ref_chaptr_id, ref_utt + ".flac")
+        ref_spk_id, ref_chaptr_id, _ =  ref_utt.split('-')
+        ref_wav = os.path.join(librispeech_test_clean_path, ref_spk_id, ref_chaptr_id, ref_utt + '.flac')
 
         test_set_.append((gen_wav, ref_wav, gen_txt))
 
     num_jobs = len(gpus)
     if num_jobs == 1:
         return [(gpus[0], test_set_)]
-
+    
     wav_per_job = len(test_set_) // num_jobs + 1
     test_set = []
     for i in range(num_jobs):
-        test_set.append((gpus[i], test_set_[i * wav_per_job : (i + 1) * wav_per_job]))
+        test_set.append((gpus[i], test_set_[i*wav_per_job:(i+1)*wav_per_job]))
 
     return test_set
 
 
 # load asr model
 
-
-def load_asr_model(lang, ckpt_dir=""):
+def load_asr_model(lang, ckpt_dir = ""):
     if lang == "zh":
         from funasr import AutoModel
-
         model = AutoModel(
-            model=os.path.join(ckpt_dir, "paraformer-zh"),
-            # vad_model = os.path.join(ckpt_dir, "fsmn-vad"),
+            model = os.path.join(ckpt_dir, "paraformer-zh"), 
+            # vad_model = os.path.join(ckpt_dir, "fsmn-vad"), 
             # punc_model = os.path.join(ckpt_dir, "ct-punc"),
-            # spk_model = os.path.join(ckpt_dir, "cam++"),
+            # spk_model = os.path.join(ckpt_dir, "cam++"), 
             disable_update=True,
-        )  # following seed-tts setting
+            )  # following seed-tts setting
     elif lang == "en":
         from faster_whisper import WhisperModel
-
         model_size = "large-v3" if ckpt_dir == "" else ckpt_dir
         model = WhisperModel(model_size, device="cuda", compute_type="float16")
     return model
@@ -469,50 +443,44 @@ def load_asr_model(lang, ckpt_dir=""):
 
 # WER Evaluation, the way Seed-TTS does
 
-
 def run_asr_wer(args):
     rank, lang, test_set, ckpt_dir = args
 
     if lang == "zh":
         import zhconv
-
         torch.cuda.set_device(rank)
     elif lang == "en":
         os.environ["CUDA_VISIBLE_DEVICES"] = str(rank)
     else:
-        raise NotImplementedError(
-            "lang support only 'zh' (funasr paraformer-zh), 'en' (faster-whisper-large-v3), for now."
-        )
-
-    asr_model = load_asr_model(lang, ckpt_dir=ckpt_dir)
+        raise NotImplementedError("lang support only 'zh' (funasr paraformer-zh), 'en' (faster-whisper-large-v3), for now.")
 
+    asr_model = load_asr_model(lang, ckpt_dir = ckpt_dir)
+    
     from zhon.hanzi import punctuation
-
     punctuation_all = punctuation + string.punctuation
     wers = []
 
     from jiwer import compute_measures
-
     for gen_wav, prompt_wav, truth in tqdm(test_set):
         if lang == "zh":
             res = asr_model.generate(input=gen_wav, batch_size_s=300, disable_pbar=True)
             hypo = res[0]["text"]
-            hypo = zhconv.convert(hypo, "zh-cn")
+            hypo = zhconv.convert(hypo, 'zh-cn')
         elif lang == "en":
             segments, _ = asr_model.transcribe(gen_wav, beam_size=5, language="en")
-            hypo = ""
+            hypo = ''
             for segment in segments:
-                hypo = hypo + " " + segment.text
+                hypo = hypo + ' ' + segment.text
 
         # raw_truth = truth
         # raw_hypo = hypo
 
         for x in punctuation_all:
-            truth = truth.replace(x, "")
-            hypo = hypo.replace(x, "")
+            truth = truth.replace(x, '')
+            hypo = hypo.replace(x, '')
 
-        truth = truth.replace("  ", " ")
-        hypo = hypo.replace("  ", " ")
+        truth = truth.replace('  ', ' ')
+        hypo = hypo.replace('  ', ' ')
 
         if lang == "zh":
             truth = " ".join([x for x in truth])
@@ -536,22 +504,22 @@ def run_asr_wer(args):
 
 # SIM Evaluation
 
-
 def run_sim(args):
     rank, test_set, ckpt_dir = args
     device = f"cuda:{rank}"
 
-    model = ECAPA_TDNN_SMALL(feat_dim=1024, feat_type="wavlm_large", config_path=None)
+    model = ECAPA_TDNN_SMALL(feat_dim=1024, feat_type='wavlm_large', config_path=None)
     state_dict = torch.load(ckpt_dir, weights_only=True, map_location=lambda storage, loc: storage)
-    model.load_state_dict(state_dict["model"], strict=False)
+    model.load_state_dict(state_dict['model'], strict=False)
 
-    use_gpu = True if torch.cuda.is_available() else False
+    use_gpu=True if torch.cuda.is_available() else False
     if use_gpu:
         model = model.cuda(device)
     model.eval()
 
     sim_list = []
     for wav1, wav2, truth in tqdm(test_set):
+
         wav1, sr1 = torchaudio.load(wav1)
         wav2, sr2 = torchaudio.load(wav2)
 
@@ -566,21 +534,20 @@ def run_sim(args):
         with torch.no_grad():
             emb1 = model(wav1)
             emb2 = model(wav2)
-
+        
         sim = F.cosine_similarity(emb1, emb2)[0].item()
         # print(f"VSim score between two audios: {sim:.4f} (-1.0, 1.0).")
         sim_list.append(sim)
-
+    
     return sim_list
 
 
 # filter func for dirty data with many repetitions
 
-
-def repetition_found(text, length=2, tolerance=10):
+def repetition_found(text, length = 2, tolerance = 10):
     pattern_count = defaultdict(int)
     for i in range(len(text) - length + 1):
-        pattern = text[i : i + length]
+        pattern = text[i:i + length]
         pattern_count[pattern] += 1
     for pattern, count in pattern_count.items():
         if count > tolerance:
@@ -590,31 +557,24 @@ def repetition_found(text, length=2, tolerance=10):
 
 # load model checkpoint for inference
 
-
-def load_checkpoint(model, ckpt_path, device, use_ema=True):
-    if device == "cuda":
-        model = model.half()
+def load_checkpoint(model, ckpt_path, device, use_ema = True):
+    from ema_pytorch import EMA
 
     ckpt_type = ckpt_path.split(".")[-1]
     if ckpt_type == "safetensors":
         from safetensors.torch import load_file
-
-        checkpoint = load_file(ckpt_path)
+        checkpoint = load_file(ckpt_path, device=device)
     else:
-        checkpoint = torch.load(ckpt_path, weights_only=True)
+        checkpoint = torch.load(ckpt_path, weights_only=True, map_location=device)
 
-    if use_ema:
+    if use_ema == True:
+        ema_model = EMA(model, include_online_model = False).to(device)
         if ckpt_type == "safetensors":
-            checkpoint = {"ema_model_state_dict": checkpoint}
-        checkpoint["model_state_dict"] = {
-            k.replace("ema_model.", ""): v
-            for k, v in checkpoint["ema_model_state_dict"].items()
-            if k not in ["initted", "step"]
-        }
-        model.load_state_dict(checkpoint["model_state_dict"])
+            ema_model.load_state_dict(checkpoint)
+        else:
+            ema_model.load_state_dict(checkpoint['ema_model_state_dict'])
+        ema_model.copy_params_from_ema_to_model()
     else:
-        if ckpt_type == "safetensors":
-            checkpoint = {"model_state_dict": checkpoint}
-        model.load_state_dict(checkpoint["model_state_dict"])
-
-    return model.to(device)
+        model.load_state_dict(checkpoint['model_state_dict'])
+        
+    return model

model/utils_infer.py

@@ -1,357 +0,0 @@
-# A unified script for inference process
-# Make adjustments inside functions, and consider both gradio and cli scripts if need to change func output format
-
-import re
-import tempfile
-
-import numpy as np
-import torch
-import torchaudio
-import tqdm
-from pydub import AudioSegment, silence
-from transformers import pipeline
-from vocos import Vocos
-
-from model import CFM
-from model.utils import (
-    load_checkpoint,
-    get_tokenizer,
-    convert_char_to_pinyin,
-)
-
-
-device = "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu"
-
-vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
-
-
-# -----------------------------------------
-
-target_sample_rate = 24000
-n_mel_channels = 100
-hop_length = 256
-target_rms = 0.1
-cross_fade_duration = 0.15
-ode_method = "euler"
-nfe_step = 32  # 16, 32
-cfg_strength = 2.0
-sway_sampling_coef = -1.0
-speed = 1.0
-fix_duration = None
-
-# -----------------------------------------
-
-
-# chunk text into smaller pieces
-
-
-def chunk_text(text, max_chars=135):
-    """
-    Splits the input text into chunks, each with a maximum number of characters.
-
-    Args:
-        text (str): The text to be split.
-        max_chars (int): The maximum number of characters per chunk.
-
-    Returns:
-        List[str]: A list of text chunks.
-    """
-    chunks = []
-    current_chunk = ""
-    # Split the text into sentences based on punctuation followed by whitespace
-    sentences = re.split(r"(?<=[;:,.!?])\s+|(?<=[；：，。！？])", text)
-
-    for sentence in sentences:
-        if len(current_chunk.encode("utf-8")) + len(sentence.encode("utf-8")) <= max_chars:
-            current_chunk += sentence + " " if sentence and len(sentence[-1].encode("utf-8")) == 1 else sentence
-        else:
-            if current_chunk:
-                chunks.append(current_chunk.strip())
-            current_chunk = sentence + " " if sentence and len(sentence[-1].encode("utf-8")) == 1 else sentence
-
-    if current_chunk:
-        chunks.append(current_chunk.strip())
-
-    return chunks
-
-
-# load vocoder
-def load_vocoder(is_local=False, local_path="", device=device):
-    if is_local:
-        print(f"Load vocos from local path {local_path}")
-        vocos = Vocos.from_hparams(f"{local_path}/config.yaml")
-        state_dict = torch.load(f"{local_path}/pytorch_model.bin", map_location=device)
-        vocos.load_state_dict(state_dict)
-        vocos.eval()
-    else:
-        print("Download Vocos from huggingface charactr/vocos-mel-24khz")
-        vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
-    return vocos
-
-
-# load asr pipeline
-
-asr_pipe = None
-
-
-def initialize_asr_pipeline(device=device):
-    global asr_pipe
-    asr_pipe = pipeline(
-        "automatic-speech-recognition",
-        model="openai/whisper-large-v3-turbo",
-        torch_dtype=torch.float16,
-        device=device,
-    )
-
-
-# load model for inference
-
-
-def load_model(model_cls, model_cfg, ckpt_path, vocab_file="", ode_method=ode_method, use_ema=True, device=device):
-    if vocab_file == "":
-        vocab_file = "Emilia_ZH_EN"
-        tokenizer = "pinyin"
-    else:
-        tokenizer = "custom"
-
-    print("\nvocab : ", vocab_file)
-    print("tokenizer : ", tokenizer)
-    print("model : ", ckpt_path, "\n")
-
-    vocab_char_map, vocab_size = get_tokenizer(vocab_file, tokenizer)
-    model = CFM(
-        transformer=model_cls(**model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels),
-        mel_spec_kwargs=dict(
-            target_sample_rate=target_sample_rate,
-            n_mel_channels=n_mel_channels,
-            hop_length=hop_length,
-        ),
-        odeint_kwargs=dict(
-            method=ode_method,
-        ),
-        vocab_char_map=vocab_char_map,
-    ).to(device)
-
-    model = load_checkpoint(model, ckpt_path, device, use_ema=use_ema)
-
-    return model
-
-
-# preprocess reference audio and text
-
-
-def preprocess_ref_audio_text(ref_audio_orig, ref_text, show_info=print, device=device):
-    show_info("Converting audio...")
-    with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
-        aseg = AudioSegment.from_file(ref_audio_orig)
-
-        non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=1000)
-        non_silent_wave = AudioSegment.silent(duration=0)
-        for non_silent_seg in non_silent_segs:
-            non_silent_wave += non_silent_seg
-        aseg = non_silent_wave
-
-        audio_duration = len(aseg)
-        if audio_duration > 15000:
-            show_info("Audio is over 15s, clipping to only first 15s.")
-            aseg = aseg[:15000]
-        aseg.export(f.name, format="wav")
-        ref_audio = f.name
-
-    if not ref_text.strip():
-        global asr_pipe
-        if asr_pipe is None:
-            initialize_asr_pipeline(device=device)
-        show_info("No reference text provided, transcribing reference audio...")
-        ref_text = asr_pipe(
-            ref_audio,
-            chunk_length_s=30,
-            batch_size=128,
-            generate_kwargs={"task": "transcribe"},
-            return_timestamps=False,
-        )["text"].strip()
-        show_info("Finished transcription")
-    else:
-        show_info("Using custom reference text...")
-
-    # Add the functionality to ensure it ends with ". "
-    if not ref_text.endswith(". ") and not ref_text.endswith("。"):
-        if ref_text.endswith("."):
-            ref_text += " "
-        else:
-            ref_text += ". "
-
-    return ref_audio, ref_text
-
-
-# infer process: chunk text -> infer batches [i.e. infer_batch_process()]
-
-
-def infer_process(
-    ref_audio,
-    ref_text,
-    gen_text,
-    model_obj,
-    show_info=print,
-    progress=tqdm,
-    target_rms=target_rms,
-    cross_fade_duration=cross_fade_duration,
-    nfe_step=nfe_step,
-    cfg_strength=cfg_strength,
-    sway_sampling_coef=sway_sampling_coef,
-    speed=speed,
-    fix_duration=fix_duration,
-    device=device,
-):
-    # Split the input text into batches
-    audio, sr = torchaudio.load(ref_audio)
-    max_chars = int(len(ref_text.encode("utf-8")) / (audio.shape[-1] / sr) * (25 - audio.shape[-1] / sr))
-    gen_text_batches = chunk_text(gen_text, max_chars=max_chars)
-    for i, gen_text in enumerate(gen_text_batches):
-        print(f"gen_text {i}", gen_text)
-
-    show_info(f"Generating audio in {len(gen_text_batches)} batches...")
-    return infer_batch_process(
-        (audio, sr),
-        ref_text,
-        gen_text_batches,
-        model_obj,
-        progress=progress,
-        target_rms=target_rms,
-        cross_fade_duration=cross_fade_duration,
-        nfe_step=nfe_step,
-        cfg_strength=cfg_strength,
-        sway_sampling_coef=sway_sampling_coef,
-        speed=speed,
-        fix_duration=fix_duration,
-        device=device,
-    )
-
-
-# infer batches
-
-
-def infer_batch_process(
-    ref_audio,
-    ref_text,
-    gen_text_batches,
-    model_obj,
-    progress=tqdm,
-    target_rms=0.1,
-    cross_fade_duration=0.15,
-    nfe_step=32,
-    cfg_strength=2.0,
-    sway_sampling_coef=-1,
-    speed=1,
-    fix_duration=None,
-    device=None,
-):
-    audio, sr = ref_audio
-    if audio.shape[0] > 1:
-        audio = torch.mean(audio, dim=0, keepdim=True)
-
-    rms = torch.sqrt(torch.mean(torch.square(audio)))
-    if rms < target_rms:
-        audio = audio * target_rms / rms
-    if sr != target_sample_rate:
-        resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
-        audio = resampler(audio)
-    audio = audio.to(device)
-
-    generated_waves = []
-    spectrograms = []
-
-    if len(ref_text[-1].encode("utf-8")) == 1:
-        ref_text = ref_text + " "
-    for i, gen_text in enumerate(progress.tqdm(gen_text_batches)):
-        # Prepare the text
-        text_list = [ref_text + gen_text]
-        final_text_list = convert_char_to_pinyin(text_list)
-
-        ref_audio_len = audio.shape[-1] // hop_length
-        if fix_duration is not None:
-            duration = int(fix_duration * target_sample_rate / hop_length)
-        else:
-            # Calculate duration
-            ref_text_len = len(ref_text.encode("utf-8"))
-            gen_text_len = len(gen_text.encode("utf-8"))
-            duration = ref_audio_len + int(ref_audio_len / ref_text_len * gen_text_len / speed)
-
-        # inference
-        with torch.inference_mode():
-            generated, _ = model_obj.sample(
-                cond=audio,
-                text=final_text_list,
-                duration=duration,
-                steps=nfe_step,
-                cfg_strength=cfg_strength,
-                sway_sampling_coef=sway_sampling_coef,
-            )
-
-        generated = generated.to(torch.float32)
-        generated = generated[:, ref_audio_len:, :]
-        generated_mel_spec = generated.permute(0, 2, 1)
-        generated_wave = vocos.decode(generated_mel_spec.cpu())
-        if rms < target_rms:
-            generated_wave = generated_wave * rms / target_rms
-
-        # wav -> numpy
-        generated_wave = generated_wave.squeeze().cpu().numpy()
-
-        generated_waves.append(generated_wave)
-        spectrograms.append(generated_mel_spec[0].cpu().numpy())
-
-    # Combine all generated waves with cross-fading
-    if cross_fade_duration <= 0:
-        # Simply concatenate
-        final_wave = np.concatenate(generated_waves)
-    else:
-        final_wave = generated_waves[0]
-        for i in range(1, len(generated_waves)):
-            prev_wave = final_wave
-            next_wave = generated_waves[i]
-
-            # Calculate cross-fade samples, ensuring it does not exceed wave lengths
-            cross_fade_samples = int(cross_fade_duration * target_sample_rate)
-            cross_fade_samples = min(cross_fade_samples, len(prev_wave), len(next_wave))
-
-            if cross_fade_samples <= 0:
-                # No overlap possible, concatenate
-                final_wave = np.concatenate([prev_wave, next_wave])
-                continue
-
-            # Overlapping parts
-            prev_overlap = prev_wave[-cross_fade_samples:]
-            next_overlap = next_wave[:cross_fade_samples]
-
-            # Fade out and fade in
-            fade_out = np.linspace(1, 0, cross_fade_samples)
-            fade_in = np.linspace(0, 1, cross_fade_samples)
-
-            # Cross-faded overlap
-            cross_faded_overlap = prev_overlap * fade_out + next_overlap * fade_in
-
-            # Combine
-            new_wave = np.concatenate(
-                [prev_wave[:-cross_fade_samples], cross_faded_overlap, next_wave[cross_fade_samples:]]
-            )
-
-            final_wave = new_wave
-
-    # Create a combined spectrogram
-    combined_spectrogram = np.concatenate(spectrograms, axis=1)
-
-    return final_wave, target_sample_rate, combined_spectrogram
-
-
-# remove silence from generated wav
-
-
-def remove_silence_for_generated_wav(filename):
-    aseg = AudioSegment.from_file(filename)
-    non_silent_segs = silence.split_on_silence(aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500)
-    non_silent_wave = AudioSegment.silent(duration=0)
-    for non_silent_seg in non_silent_segs:
-        non_silent_wave += non_silent_seg
-    aseg = non_silent_wave
-    aseg.export(filename, format="wav")

packages.txt

@@ -0,0 +1 @@
+ffmpeg

pyproject.toml

@@ -1,62 +0,0 @@
-[build-system]
-requires = ["setuptools >= 61.0", "setuptools-scm>=8.0"]
-build-backend = "setuptools.build_meta"
-
-[project]
-name = "f5-tts"
-version = "0.3.1"
-description = "F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching"
-readme = "README.md"
-license = {text = "MIT License"}
-classifiers = [
-    "License :: OSI Approved :: MIT License",
-    "Operating System :: OS Independent",
-    "Programming Language :: Python :: 3",
-]
-dependencies = [
-    "accelerate>=0.33.0",
-    "bitsandbytes>0.37.0; platform_machine != 'arm64' and platform_system != 'Darwin'",
-    "cached_path",
-    "click",
-    "datasets",
-    "ema_pytorch>=0.5.2",
-    "gradio>=3.45.2",
-    "hydra-core>=1.3.0",
-    "jieba",
-    "librosa",
-    "matplotlib",
-    "numpy<=1.26.4",
-    "pydub",
-    "pypinyin",
-    "safetensors",
-    "soundfile",
-    "tomli",
-    "torch>=2.0.0",
-    "torchaudio>=2.0.0",
-    "torchdiffeq",
-    "tqdm>=4.65.0",
-    "transformers",
-    "transformers_stream_generator",
-    "vocos",
-    "wandb",
-    "x_transformers>=1.31.14",
-]
-
-[project.optional-dependencies]
-eval = [
-    "faster_whisper==0.10.1",
-    "funasr",
-    "jiwer",
-    "modelscope",
-    "zhconv",
-    "zhon",
-]
-
-[project.urls]
-Homepage = "https://github.com/SWivid/F5-TTS"
-
-[project.scripts]
-"f5-tts_infer-cli" = "f5_tts.infer.infer_cli:main"
-"f5-tts_infer-gradio" = "f5_tts.infer.infer_gradio:main"
-"f5-tts_finetune-cli" = "f5_tts.train.finetune_cli:main"
-"f5-tts_finetune-gradio" = "f5_tts.train.finetune_gradio:main"

requirements.txt

@@ -1,10 +1,9 @@
-torch
-torchaudio
 accelerate>=0.33.0
-bitsandbytes>0.37.0
 cached_path
 click
 datasets
+einops>=0.8.0
+einx>=0.3.0
 ema_pytorch>=0.5.2
 gradio
 jieba
@@ -22,5 +21,3 @@ transformers
 vocos
 wandb
 x_transformers>=1.31.14
-f5_tts @ git+https://huggingface.co/spaces/mrfakename/E2-F5-TTS
-detoxify @ git+https://github.com/unitaryai/detoxify

ruff.toml

@@ -1,10 +0,0 @@
-line-length = 120
-target-version = "py310"
-
-[lint]
-# Only ignore variables with names starting with "_".
-dummy-variable-rgx = "^_.*$"
-
-[lint.isort]
-force-single-line = true
-lines-after-imports = 2

scripts/count_max_epoch.py

@@ -1,7 +1,6 @@
-"""ADAPTIVE BATCH SIZE"""
-
-print("Adaptive batch size: using grouping batch sampler, frames_per_gpu fixed fed in")
-print("  -> least padding, gather wavs with accumulated frames in a batch\n")
+'''ADAPTIVE BATCH SIZE'''
+print('Adaptive batch size: using grouping batch sampler, frames_per_gpu fixed fed in')
+print('  -> least padding, gather wavs with accumulated frames in a batch\n')
 
 # data
 total_hours = 95282

scripts/count_params_gflops.py

@@ -1,15 +1,13 @@
-import sys
-import os
-
+import sys, os
 sys.path.append(os.getcwd())
 
-from model import M2_TTS, DiT
+from model import M2_TTS, UNetT, DiT, MMDiT
 
 import torch
 import thop
 
 
-""" ~155M """
+''' ~155M '''
 # transformer =     UNetT(dim = 768, depth = 20, heads = 12, ff_mult = 4)
 # transformer =     UNetT(dim = 768, depth = 20, heads = 12, ff_mult = 4, text_dim = 512, conv_layers = 4)
 # transformer =       DiT(dim = 768, depth = 18, heads = 12, ff_mult = 2)
@@ -17,11 +15,11 @@ import thop
 # transformer =       DiT(dim = 768, depth = 18, heads = 12, ff_mult = 2, text_dim = 512, conv_layers = 4, long_skip_connection = True)
 # transformer =     MMDiT(dim = 512, depth = 16, heads = 16, ff_mult = 2)
 
-""" ~335M """
+''' ~335M '''
 # FLOPs: 622.1 G, Params: 333.2 M
 # transformer =     UNetT(dim = 1024, depth = 24, heads = 16, ff_mult = 4)
 # FLOPs: 363.4 G, Params: 335.8 M
-transformer = DiT(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
+transformer =       DiT(dim = 1024, depth = 22, heads = 16, ff_mult = 2, text_dim = 512, conv_layers = 4)
 
 
 model = M2_TTS(transformer=transformer)
@@ -32,8 +30,6 @@ duration = 20
 frame_length = int(duration * target_sample_rate / hop_length)
 text_length = 150
 
-flops, params = thop.profile(
-    model, inputs=(torch.randn(1, frame_length, n_mel_channels), torch.zeros(1, text_length, dtype=torch.long))
-)
+flops, params = thop.profile(model, inputs=(torch.randn(1, frame_length, n_mel_channels), torch.zeros(1, text_length, dtype=torch.long)))
 print(f"FLOPs: {flops / 1e9} G")
 print(f"Params: {params / 1e6} M")

scripts/eval_infer_batch.py

@@ -1,6 +1,4 @@
-import sys
-import os
-
+import sys, os
 sys.path.append(os.getcwd())
 
 import time
@@ -11,14 +9,15 @@ import argparse
 import torch
 import torchaudio
 from accelerate import Accelerator
+from einops import rearrange
 from vocos import Vocos
 
 from model import CFM, UNetT, DiT
 from model.utils import (
     load_checkpoint,
-    get_tokenizer,
-    get_seedtts_testset_metainfo,
-    get_librispeech_test_clean_metainfo,
+    get_tokenizer, 
+    get_seedtts_testset_metainfo, 
+    get_librispeech_test_clean_metainfo, 
     get_inference_prompt,
 )
 
@@ -40,16 +39,16 @@ tokenizer = "pinyin"
 
 parser = argparse.ArgumentParser(description="batch inference")
 
-parser.add_argument("-s", "--seed", default=None, type=int)
-parser.add_argument("-d", "--dataset", default="Emilia_ZH_EN")
-parser.add_argument("-n", "--expname", required=True)
-parser.add_argument("-c", "--ckptstep", default=1200000, type=int)
+parser.add_argument('-s', '--seed', default=None, type=int)
+parser.add_argument('-d', '--dataset', default="Emilia_ZH_EN")
+parser.add_argument('-n', '--expname', required=True)
+parser.add_argument('-c', '--ckptstep', default=1200000, type=int)
 
-parser.add_argument("-nfe", "--nfestep", default=32, type=int)
-parser.add_argument("-o", "--odemethod", default="euler")
-parser.add_argument("-ss", "--swaysampling", default=-1, type=float)
+parser.add_argument('-nfe', '--nfestep', default=32, type=int)
+parser.add_argument('-o', '--odemethod', default="euler")
+parser.add_argument('-ss', '--swaysampling', default=-1, type=float)
 
-parser.add_argument("-t", "--testset", required=True)
+parser.add_argument('-t', '--testset', required=True)
 
 args = parser.parse_args()
 
@@ -68,26 +67,26 @@ testset = args.testset
 
 
 infer_batch_size = 1  # max frames. 1 for ddp single inference (recommended)
-cfg_strength = 2.0
-speed = 1.0
+cfg_strength = 2.
+speed = 1.
 use_truth_duration = False
 no_ref_audio = False
 
 
 if exp_name == "F5TTS_Base":
     model_cls = DiT
-    model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
+    model_cfg = dict(dim = 1024, depth = 22, heads = 16, ff_mult = 2, text_dim = 512, conv_layers = 4)
 
 elif exp_name == "E2TTS_Base":
     model_cls = UNetT
-    model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
+    model_cfg = dict(dim = 1024, depth = 24, heads = 16, ff_mult = 4)
 
 
 if testset == "ls_pc_test_clean":
     metalst = "data/librispeech_pc_test_clean_cross_sentence.lst"
     librispeech_test_clean_path = "<SOME_PATH>/LibriSpeech/test-clean"  # test-clean path
     metainfo = get_librispeech_test_clean_metainfo(metalst, librispeech_test_clean_path)
-
+    
 elif testset == "seedtts_test_zh":
     metalst = "data/seedtts_testset/zh/meta.lst"
     metainfo = get_seedtts_testset_metainfo(metalst)
@@ -98,16 +97,13 @@ elif testset == "seedtts_test_en":
 
 
 # path to save genereted wavs
-if seed is None:
-    seed = random.randint(-10000, 10000)
-output_dir = (
-    f"results/{exp_name}_{ckpt_step}/{testset}/"
-    f"seed{seed}_{ode_method}_nfe{nfe_step}"
-    f"{f'_ss{sway_sampling_coef}' if sway_sampling_coef else ''}"
-    f"_cfg{cfg_strength}_speed{speed}"
-    f"{'_gt-dur' if use_truth_duration else ''}"
+if seed is None: seed = random.randint(-10000, 10000)
+output_dir = f"results/{exp_name}_{ckpt_step}/{testset}/" \
+    f"seed{seed}_{ode_method}_nfe{nfe_step}" \
+    f"{f'_ss{sway_sampling_coef}' if sway_sampling_coef else ''}" \
+    f"_cfg{cfg_strength}_speed{speed}" \
+    f"{'_gt-dur' if use_truth_duration else ''}" \
     f"{'_no-ref-audio' if no_ref_audio else ''}"
-)
 
 
 # -------------------------------------------------#
@@ -115,15 +111,15 @@ output_dir = (
 use_ema = True
 
 prompts_all = get_inference_prompt(
-    metainfo,
-    speed=speed,
-    tokenizer=tokenizer,
-    target_sample_rate=target_sample_rate,
-    n_mel_channels=n_mel_channels,
-    hop_length=hop_length,
-    target_rms=target_rms,
-    use_truth_duration=use_truth_duration,
-    infer_batch_size=infer_batch_size,
+    metainfo, 
+    speed = speed, 
+    tokenizer = tokenizer, 
+    target_sample_rate = target_sample_rate, 
+    n_mel_channels = n_mel_channels,
+    hop_length = hop_length,
+    target_rms = target_rms,
+    use_truth_duration = use_truth_duration,
+    infer_batch_size = infer_batch_size,
 )
 
 # Vocoder model
@@ -142,19 +138,23 @@ vocab_char_map, vocab_size = get_tokenizer(dataset_name, tokenizer)
 
 # Model
 model = CFM(
-    transformer=model_cls(**model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels),
-    mel_spec_kwargs=dict(
-        target_sample_rate=target_sample_rate,
-        n_mel_channels=n_mel_channels,
-        hop_length=hop_length,
+    transformer = model_cls(
+        **model_cfg,
+        text_num_embeds = vocab_size, 
+        mel_dim = n_mel_channels
     ),
-    odeint_kwargs=dict(
-        method=ode_method,
+    mel_spec_kwargs = dict(
+        target_sample_rate = target_sample_rate, 
+        n_mel_channels = n_mel_channels,
+        hop_length = hop_length,
     ),
-    vocab_char_map=vocab_char_map,
+    odeint_kwargs = dict(
+        method = ode_method,
+    ),
+    vocab_char_map = vocab_char_map,
 ).to(device)
 
-model = load_checkpoint(model, ckpt_path, device, use_ema=use_ema)
+model = load_checkpoint(model, ckpt_path, device, use_ema = use_ema)
 
 if not os.path.exists(output_dir) and accelerator.is_main_process:
     os.makedirs(output_dir)
@@ -164,29 +164,30 @@ accelerator.wait_for_everyone()
 start = time.time()
 
 with accelerator.split_between_processes(prompts_all) as prompts:
+
     for prompt in tqdm(prompts, disable=not accelerator.is_local_main_process):
         utts, ref_rms_list, ref_mels, ref_mel_lens, total_mel_lens, final_text_list = prompt
         ref_mels = ref_mels.to(device)
-        ref_mel_lens = torch.tensor(ref_mel_lens, dtype=torch.long).to(device)
-        total_mel_lens = torch.tensor(total_mel_lens, dtype=torch.long).to(device)
-
+        ref_mel_lens = torch.tensor(ref_mel_lens, dtype = torch.long).to(device)
+        total_mel_lens = torch.tensor(total_mel_lens, dtype = torch.long).to(device)
+        
         # Inference
         with torch.inference_mode():
             generated, _ = model.sample(
-                cond=ref_mels,
-                text=final_text_list,
-                duration=total_mel_lens,
-                lens=ref_mel_lens,
-                steps=nfe_step,
-                cfg_strength=cfg_strength,
-                sway_sampling_coef=sway_sampling_coef,
-                no_ref_audio=no_ref_audio,
-                seed=seed,
+                cond = ref_mels,
+                text = final_text_list,
+                duration = total_mel_lens,
+                lens = ref_mel_lens,
+                steps = nfe_step,
+                cfg_strength = cfg_strength,
+                sway_sampling_coef = sway_sampling_coef,
+                no_ref_audio = no_ref_audio,
+                seed = seed,
             )
         # Final result
         for i, gen in enumerate(generated):
-            gen = gen[ref_mel_lens[i] : total_mel_lens[i], :].unsqueeze(0)
-            gen_mel_spec = gen.permute(0, 2, 1)
+            gen = gen[ref_mel_lens[i]:total_mel_lens[i], :].unsqueeze(0)
+            gen_mel_spec = rearrange(gen, '1 n d -> 1 d n')
             generated_wave = vocos.decode(gen_mel_spec.cpu())
             if ref_rms_list[i] < target_rms:
                 generated_wave = generated_wave * ref_rms_list[i] / target_rms

scripts/eval_librispeech_test_clean.py

@@ -1,8 +1,6 @@
 # Evaluate with Librispeech test-clean, ~3s prompt to generate 4-10s audio (the way of valle/voicebox evaluation)
 
-import sys
-import os
-
+import sys, os
 sys.path.append(os.getcwd())
 
 import multiprocessing as mp
@@ -21,7 +19,7 @@ metalst = "data/librispeech_pc_test_clean_cross_sentence.lst"
 librispeech_test_clean_path = "<SOME_PATH>/LibriSpeech/test-clean"  # test-clean path
 gen_wav_dir = "PATH_TO_GENERATED"  # generated wavs
 
-gpus = [0, 1, 2, 3, 4, 5, 6, 7]
+gpus = [0,1,2,3,4,5,6,7]
 test_set = get_librispeech_test(metalst, gen_wav_dir, gpus, librispeech_test_clean_path)
 
 ## In LibriSpeech, some speakers utilized varying voice characteristics for different characters in the book,
@@ -48,7 +46,7 @@ if eval_task == "wer":
         for wers_ in results:
             wers.extend(wers_)
 
-    wer = round(np.mean(wers) * 100, 3)
+    wer = round(np.mean(wers)*100, 3)
     print(f"\nTotal {len(wers)} samples")
     print(f"WER      : {wer}%")
 
@@ -64,6 +62,6 @@ if eval_task == "sim":
         for sim_ in results:
             sim_list.extend(sim_)
 
-    sim = round(sum(sim_list) / len(sim_list), 3)
+    sim = round(sum(sim_list)/len(sim_list), 3)
     print(f"\nTotal {len(sim_list)} samples")
     print(f"SIM      : {sim}")

scripts/eval_seedtts_testset.py

@@ -1,8 +1,6 @@
 # Evaluate with Seed-TTS testset
 
-import sys
-import os
-
+import sys, os
 sys.path.append(os.getcwd())
 
 import multiprocessing as mp
@@ -16,21 +14,21 @@ from model.utils import (
 
 
 eval_task = "wer"  # sim | wer
-lang = "zh"  # zh | en
+lang = "zh"        # zh | en
 metalst = f"data/seedtts_testset/{lang}/meta.lst"  # seed-tts testset
 # gen_wav_dir = f"data/seedtts_testset/{lang}/wavs"  # ground truth wavs
-gen_wav_dir = "PATH_TO_GENERATED"  # generated wavs
+gen_wav_dir = f"PATH_TO_GENERATED"  # generated wavs
 
 
 # NOTE. paraformer-zh result will be slightly different according to the number of gpus, cuz batchsize is different
-#       zh 1.254 seems a result of 4 workers wer_seed_tts
-gpus = [0, 1, 2, 3, 4, 5, 6, 7]
+#       zh 1.254 seems a result of 4 workers wer_seed_tts 
+gpus = [0,1,2,3,4,5,6,7]
 test_set = get_seed_tts_test(metalst, gen_wav_dir, gpus)
 
 local = False
 if local:  # use local custom checkpoint dir
     if lang == "zh":
-        asr_ckpt_dir = "../checkpoints/funasr"  # paraformer-zh dir under funasr
+        asr_ckpt_dir = "../checkpoints/funasr" # paraformer-zh dir under funasr
     elif lang == "en":
         asr_ckpt_dir = "../checkpoints/Systran/faster-whisper-large-v3"
 else:
@@ -50,7 +48,7 @@ if eval_task == "wer":
         for wers_ in results:
             wers.extend(wers_)
 
-    wer = round(np.mean(wers) * 100, 3)
+    wer = round(np.mean(wers)*100, 3)
     print(f"\nTotal {len(wers)} samples")
     print(f"WER      : {wer}%")
 
@@ -66,6 +64,6 @@ if eval_task == "sim":
         for sim_ in results:
             sim_list.extend(sim_)
 
-    sim = round(sum(sim_list) / len(sim_list), 3)
+    sim = round(sum(sim_list)/len(sim_list), 3)
     print(f"\nTotal {len(sim_list)} samples")
     print(f"SIM      : {sim}")

scripts/prepare_csv_wavs.py

@@ -1,6 +1,4 @@
-import sys
-import os
-
+import sys, os
 sys.path.append(os.getcwd())
 
 from pathlib import Path
@@ -19,11 +17,10 @@ from model.utils import (
 
 PRETRAINED_VOCAB_PATH = Path(__file__).parent.parent / "data/Emilia_ZH_EN_pinyin/vocab.txt"
 
-
 def is_csv_wavs_format(input_dataset_dir):
     fpath = Path(input_dataset_dir)
     metadata = fpath / "metadata.csv"
-    wavs = fpath / "wavs"
+    wavs = fpath / 'wavs'
     return metadata.exists() and metadata.is_file() and wavs.exists() and wavs.is_dir()
 
 
@@ -49,24 +46,22 @@ def prepare_csv_wavs_dir(input_dir):
 
     return sub_result, durations, vocab_set
 
-
 def get_audio_duration(audio_path):
     audio, sample_rate = torchaudio.load(audio_path)
     num_channels = audio.shape[0]
     return audio.shape[1] / (sample_rate * num_channels)
 
-
 def read_audio_text_pairs(csv_file_path):
     audio_text_pairs = []
 
     parent = Path(csv_file_path).parent
-    with open(csv_file_path, mode="r", newline="", encoding="utf-8") as csvfile:
-        reader = csv.reader(csvfile, delimiter="|")
+    with open(csv_file_path, mode='r', newline='', encoding='utf-8') as csvfile:
+        reader = csv.reader(csvfile, delimiter='|')
         next(reader)  # Skip the header row
         for row in reader:
             if len(row) >= 2:
                 audio_file = row[0].strip()  # First column: audio file path
-                text = row[1].strip()  # Second column: text
+                text = row[1].strip()          # Second column: text
                 audio_file_path = parent / audio_file
                 audio_text_pairs.append((audio_file_path.as_posix(), text))
 
@@ -83,12 +78,12 @@ def save_prepped_dataset(out_dir, result, duration_list, text_vocab_set, is_fine
     # dataset.save_to_disk(f"data/{dataset_name}/raw", max_shard_size="2GB")
     raw_arrow_path = out_dir / "raw.arrow"
     with ArrowWriter(path=raw_arrow_path.as_posix(), writer_batch_size=1) as writer:
-        for line in tqdm(result, desc="Writing to raw.arrow ..."):
+        for line in tqdm(result, desc=f"Writing to raw.arrow ..."):
             writer.write(line)
 
     # dup a json separately saving duration in case for DynamicBatchSampler ease
     dur_json_path = out_dir / "duration.json"
-    with open(dur_json_path.as_posix(), "w", encoding="utf-8") as f:
+    with open(dur_json_path.as_posix(), 'w', encoding='utf-8') as f:
         json.dump({"duration": duration_list}, f, ensure_ascii=False)
 
     # vocab map, i.e. tokenizer
@@ -125,14 +120,13 @@ def cli():
     # finetune: python scripts/prepare_csv_wavs.py /path/to/input_dir /path/to/output_dir_pinyin
     # pretrain: python scripts/prepare_csv_wavs.py /path/to/output_dir_pinyin --pretrain
     parser = argparse.ArgumentParser(description="Prepare and save dataset.")
-    parser.add_argument("inp_dir", type=str, help="Input directory containing the data.")
-    parser.add_argument("out_dir", type=str, help="Output directory to save the prepared data.")
-    parser.add_argument("--pretrain", action="store_true", help="Enable for new pretrain, otherwise is a fine-tune")
+    parser.add_argument('inp_dir', type=str, help="Input directory containing the data.")
+    parser.add_argument('out_dir', type=str, help="Output directory to save the prepared data.")
+    parser.add_argument('--pretrain', action='store_true', help="Enable for new pretrain, otherwise is a fine-tune")
 
     args = parser.parse_args()
 
     prepare_and_save_set(args.inp_dir, args.out_dir, is_finetune=not args.pretrain)
 
-
 if __name__ == "__main__":
     cli()

scripts/prepare_emilia.py

@@ -4,9 +4,7 @@
 # generate audio text map for Emilia ZH & EN
 # evaluate for vocab size
 
-import sys
-import os
-
+import sys, os
 sys.path.append(os.getcwd())
 
 from pathlib import Path
@@ -14,6 +12,7 @@ import json
 from tqdm import tqdm
 from concurrent.futures import ProcessPoolExecutor
 
+from datasets import Dataset
 from datasets.arrow_writer import ArrowWriter
 
 from model.utils import (
@@ -22,89 +21,13 @@ from model.utils import (
 )
 
 
-out_zh = {
-    "ZH_B00041_S06226",
-    "ZH_B00042_S09204",
-    "ZH_B00065_S09430",
-    "ZH_B00065_S09431",
-    "ZH_B00066_S09327",
-    "ZH_B00066_S09328",
-}
+out_zh = {"ZH_B00041_S06226", "ZH_B00042_S09204", "ZH_B00065_S09430", "ZH_B00065_S09431", "ZH_B00066_S09327", "ZH_B00066_S09328"}
 zh_filters = ["い", "て"]
 # seems synthesized audios, or heavily code-switched
 out_en = {
-    "EN_B00013_S00913",
-    "EN_B00042_S00120",
-    "EN_B00055_S04111",
-    "EN_B00061_S00693",
-    "EN_B00061_S01494",
-    "EN_B00061_S03375",
-    "EN_B00059_S00092",
-    "EN_B00111_S04300",
-    "EN_B00100_S03759",
-    "EN_B00087_S03811",
-    "EN_B00059_S00950",
-    "EN_B00089_S00946",
-    "EN_B00078_S05127",
-    "EN_B00070_S04089",
-    "EN_B00074_S09659",
-    "EN_B00061_S06983",
-    "EN_B00061_S07060",
-    "EN_B00059_S08397",
-    "EN_B00082_S06192",
-    "EN_B00091_S01238",
-    "EN_B00089_S07349",
-    "EN_B00070_S04343",
-    "EN_B00061_S02400",
-    "EN_B00076_S01262",
-    "EN_B00068_S06467",
-    "EN_B00076_S02943",
-    "EN_B00064_S05954",
-    "EN_B00061_S05386",
-    "EN_B00066_S06544",
-    "EN_B00076_S06944",
-    "EN_B00072_S08620",
-    "EN_B00076_S07135",
-    "EN_B00076_S09127",
-    "EN_B00065_S00497",
-    "EN_B00059_S06227",
-    "EN_B00063_S02859",
-    "EN_B00075_S01547",
-    "EN_B00061_S08286",
-    "EN_B00079_S02901",
-    "EN_B00092_S03643",
-    "EN_B00096_S08653",
-    "EN_B00063_S04297",
-    "EN_B00063_S04614",
-    "EN_B00079_S04698",
-    "EN_B00104_S01666",
-    "EN_B00061_S09504",
-    "EN_B00061_S09694",
-    "EN_B00065_S05444",
-    "EN_B00063_S06860",
-    "EN_B00065_S05725",
-    "EN_B00069_S07628",
-    "EN_B00083_S03875",
-    "EN_B00071_S07665",
-    "EN_B00071_S07665",
-    "EN_B00062_S04187",
-    "EN_B00065_S09873",
-    "EN_B00065_S09922",
-    "EN_B00084_S02463",
-    "EN_B00067_S05066",
-    "EN_B00106_S08060",
-    "EN_B00073_S06399",
-    "EN_B00073_S09236",
-    "EN_B00087_S00432",
-    "EN_B00085_S05618",
-    "EN_B00064_S01262",
-    "EN_B00072_S01739",
-    "EN_B00059_S03913",
-    "EN_B00069_S04036",
-    "EN_B00067_S05623",
-    "EN_B00060_S05389",
-    "EN_B00060_S07290",
-    "EN_B00062_S08995",
+    "EN_B00013_S00913", "EN_B00042_S00120", "EN_B00055_S04111", "EN_B00061_S00693", "EN_B00061_S01494", "EN_B00061_S03375",
+    
+    "EN_B00059_S00092", "EN_B00111_S04300", "EN_B00100_S03759", "EN_B00087_S03811", "EN_B00059_S00950", "EN_B00089_S00946", "EN_B00078_S05127", "EN_B00070_S04089", "EN_B00074_S09659", "EN_B00061_S06983", "EN_B00061_S07060", "EN_B00059_S08397", "EN_B00082_S06192", "EN_B00091_S01238", "EN_B00089_S07349", "EN_B00070_S04343", "EN_B00061_S02400", "EN_B00076_S01262", "EN_B00068_S06467", "EN_B00076_S02943", "EN_B00064_S05954", "EN_B00061_S05386", "EN_B00066_S06544", "EN_B00076_S06944", "EN_B00072_S08620", "EN_B00076_S07135", "EN_B00076_S09127", "EN_B00065_S00497", "EN_B00059_S06227", "EN_B00063_S02859", "EN_B00075_S01547", "EN_B00061_S08286", "EN_B00079_S02901", "EN_B00092_S03643", "EN_B00096_S08653", "EN_B00063_S04297", "EN_B00063_S04614", "EN_B00079_S04698", "EN_B00104_S01666", "EN_B00061_S09504", "EN_B00061_S09694", "EN_B00065_S05444", "EN_B00063_S06860", "EN_B00065_S05725", "EN_B00069_S07628", "EN_B00083_S03875", "EN_B00071_S07665", "EN_B00071_S07665", "EN_B00062_S04187", "EN_B00065_S09873", "EN_B00065_S09922", "EN_B00084_S02463", "EN_B00067_S05066", "EN_B00106_S08060", "EN_B00073_S06399", "EN_B00073_S09236", "EN_B00087_S00432", "EN_B00085_S05618", "EN_B00064_S01262", "EN_B00072_S01739", "EN_B00059_S03913", "EN_B00069_S04036", "EN_B00067_S05623", "EN_B00060_S05389", "EN_B00060_S07290", "EN_B00062_S08995",
 }
 en_filters = ["ا", "い", "て"]
 
@@ -120,24 +43,18 @@ def deal_with_audio_dir(audio_dir):
         for line in tqdm(lines, desc=f"{audio_jsonl.stem}"):
             obj = json.loads(line)
             text = obj["text"]
-            if obj["language"] == "zh":
+            if obj['language'] == "zh":
                 if obj["wav"].split("/")[1] in out_zh or any(f in text for f in zh_filters) or repetition_found(text):
                     bad_case_zh += 1
                     continue
                 else:
-                    text = text.translate(
-                        str.maketrans({",": "，", "!": "！", "?": "？"})
-                    )  # not "。" cuz much code-switched
-            if obj["language"] == "en":
-                if (
-                    obj["wav"].split("/")[1] in out_en
-                    or any(f in text for f in en_filters)
-                    or repetition_found(text, length=4)
-                ):
+                    text = text.translate(str.maketrans({',': '，', '!': '！', '?': '？'}))  # not "。" cuz much code-switched
+            if obj['language'] == "en":
+                if obj["wav"].split("/")[1] in out_en or any(f in text for f in en_filters) or repetition_found(text, length=4):
                     bad_case_en += 1
                     continue
             if tokenizer == "pinyin":
-                text = convert_char_to_pinyin([text], polyphone=polyphone)[0]
+                text = convert_char_to_pinyin([text], polyphone = polyphone)[0]
             duration = obj["duration"]
             sub_result.append({"audio_path": str(audio_dir.parent / obj["wav"]), "text": text, "duration": duration})
             durations.append(duration)
@@ -179,11 +96,11 @@ def main():
     # dataset = Dataset.from_dict({"audio_path": audio_path_list, "text": text_list, "duration": duration_list})  # oom
     # dataset.save_to_disk(f"data/{dataset_name}/raw", max_shard_size="2GB")
     with ArrowWriter(path=f"data/{dataset_name}/raw.arrow") as writer:
-        for line in tqdm(result, desc="Writing to raw.arrow ..."):
+        for line in tqdm(result, desc=f"Writing to raw.arrow ..."):
             writer.write(line)
 
     # dup a json separately saving duration in case for DynamicBatchSampler ease
-    with open(f"data/{dataset_name}/duration.json", "w", encoding="utf-8") as f:
+    with open(f"data/{dataset_name}/duration.json", 'w', encoding='utf-8') as f:
         json.dump({"duration": duration_list}, f, ensure_ascii=False)
 
     # vocab map, i.e. tokenizer
@@ -197,13 +114,12 @@ def main():
     print(f"\nFor {dataset_name}, sample count: {len(result)}")
     print(f"For {dataset_name}, vocab size is: {len(text_vocab_set)}")
     print(f"For {dataset_name}, total {sum(duration_list)/3600:.2f} hours")
-    if "ZH" in langs:
-        print(f"Bad zh transcription case: {total_bad_case_zh}")
-    if "EN" in langs:
-        print(f"Bad en transcription case: {total_bad_case_en}\n")
+    if "ZH" in langs: print(f"Bad zh transcription case: {total_bad_case_zh}")
+    if "EN" in langs: print(f"Bad en transcription case: {total_bad_case_en}\n")
 
 
 if __name__ == "__main__":
+
     max_workers = 32
 
     tokenizer = "pinyin"  # "pinyin" | "char"

scripts/prepare_wenetspeech4tts.py

@@ -1,9 +1,7 @@
 # generate audio text map for WenetSpeech4TTS
 # evaluate for vocab size
 
-import sys
-import os
-
+import sys, os
 sys.path.append(os.getcwd())
 
 import json
@@ -25,7 +23,7 @@ def deal_with_sub_path_files(dataset_path, sub_path):
 
     audio_paths, texts, durations = [], [], []
     for text_file in tqdm(text_files):
-        with open(os.path.join(text_dir, text_file), "r", encoding="utf-8") as file:
+        with open(os.path.join(text_dir, text_file), 'r', encoding='utf-8') as file:
             first_line = file.readline().split("\t")
         audio_nm = first_line[0]
         audio_path = os.path.join(audio_dir, audio_nm + ".wav")
@@ -34,7 +32,7 @@ def deal_with_sub_path_files(dataset_path, sub_path):
         audio_paths.append(audio_path)
 
         if tokenizer == "pinyin":
-            texts.extend(convert_char_to_pinyin([text], polyphone=polyphone))
+            texts.extend(convert_char_to_pinyin([text], polyphone = polyphone))
         elif tokenizer == "char":
             texts.append(text)
 
@@ -48,7 +46,7 @@ def main():
     assert tokenizer in ["pinyin", "char"]
 
     audio_path_list, text_list, duration_list = [], [], []
-
+    
     executor = ProcessPoolExecutor(max_workers=max_workers)
     futures = []
     for dataset_path in dataset_paths:
@@ -70,10 +68,8 @@ def main():
     dataset = Dataset.from_dict({"audio_path": audio_path_list, "text": text_list, "duration": duration_list})
     dataset.save_to_disk(f"data/{dataset_name}_{tokenizer}/raw", max_shard_size="2GB")  # arrow format
 
-    with open(f"data/{dataset_name}_{tokenizer}/duration.json", "w", encoding="utf-8") as f:
-        json.dump(
-            {"duration": duration_list}, f, ensure_ascii=False
-        )  # dup a json separately saving duration in case for DynamicBatchSampler ease
+    with open(f"data/{dataset_name}_{tokenizer}/duration.json", 'w', encoding='utf-8') as f:
+        json.dump({"duration": duration_list}, f, ensure_ascii=False)  # dup a json separately saving duration in case for DynamicBatchSampler ease
 
     print("\nEvaluating vocab size (all characters and symbols / all phonemes) ...")
     text_vocab_set = set()
@@ -89,21 +85,22 @@ def main():
             f.write(vocab + "\n")
     print(f"\nFor {dataset_name}, sample count: {len(text_list)}")
     print(f"For {dataset_name}, vocab size is: {len(text_vocab_set)}\n")
-
+    
 
 if __name__ == "__main__":
+
     max_workers = 32
 
     tokenizer = "pinyin"  # "pinyin" | "char"
     polyphone = True
     dataset_choice = 1  # 1: Premium, 2: Standard, 3: Basic
 
-    dataset_name = ["WenetSpeech4TTS_Premium", "WenetSpeech4TTS_Standard", "WenetSpeech4TTS_Basic"][dataset_choice - 1]
+    dataset_name = ["WenetSpeech4TTS_Premium", "WenetSpeech4TTS_Standard", "WenetSpeech4TTS_Basic"][dataset_choice-1]
     dataset_paths = [
         "<SOME_PATH>/WenetSpeech4TTS/Basic",
         "<SOME_PATH>/WenetSpeech4TTS/Standard",
         "<SOME_PATH>/WenetSpeech4TTS/Premium",
-    ][-dataset_choice:]
+        ][-dataset_choice:]
     print(f"\nChoose Dataset: {dataset_name}\n")
 
     main()
@@ -112,8 +109,8 @@ if __name__ == "__main__":
     # WenetSpeech4TTS       Basic     Standard     Premium
     # samples count       3932473      1941220      407494
     # pinyin vocab size      1349         1348        1344   (no polyphone)
-    #                           -            -        1459   (polyphone)
+    #                           -            -        1459   (polyphone) 
     # char   vocab size      5264         5219        5042
-
+    
     # vocab size may be slightly different due to jieba tokenizer and pypinyin (e.g. way of polyphoneme)
     # please be careful if using pretrained model, make sure the vocab.txt is same

speech_edit.py

@@ -3,13 +3,14 @@ import os
 import torch
 import torch.nn.functional as F
 import torchaudio
+from einops import rearrange
 from vocos import Vocos
 
-from model import CFM, UNetT, DiT
+from model import CFM, UNetT, DiT, MMDiT
 from model.utils import (
     load_checkpoint,
-    get_tokenizer,
-    convert_char_to_pinyin,
+    get_tokenizer, 
+    convert_char_to_pinyin, 
     save_spectrogram,
 )
 
@@ -35,20 +36,20 @@ exp_name = "F5TTS_Base"  # F5TTS_Base | E2TTS_Base
 ckpt_step = 1200000
 
 nfe_step = 32  # 16, 32
-cfg_strength = 2.0
-ode_method = "euler"  # euler | midpoint
-sway_sampling_coef = -1.0
-speed = 1.0
+cfg_strength = 2.
+ode_method = 'euler'  # euler | midpoint
+sway_sampling_coef = -1.
+speed = 1.
 
 if exp_name == "F5TTS_Base":
     model_cls = DiT
-    model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
+    model_cfg = dict(dim = 1024, depth = 22, heads = 16, ff_mult = 2, text_dim = 512, conv_layers = 4)
 
 elif exp_name == "E2TTS_Base":
     model_cls = UNetT
-    model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
+    model_cfg = dict(dim = 1024, depth = 24, heads = 16, ff_mult = 4)
 
-ckpt_path = f"ckpts/{exp_name}/model_{ckpt_step}.safetensors"
+ckpt_path = f"ckpts/{exp_name}/model_{ckpt_step}.pt"
 output_dir = "tests"
 
 # [leverage https://github.com/MahmoudAshraf97/ctc-forced-aligner to get char level alignment]
@@ -62,14 +63,8 @@ output_dir = "tests"
 audio_to_edit = "tests/ref_audio/test_en_1_ref_short.wav"
 origin_text = "Some call me nature, others call me mother nature."
 target_text = "Some call me optimist, others call me realist."
-parts_to_edit = [
-    [1.42, 2.44],
-    [4.04, 4.9],
-]  # stard_ends of "nature" & "mother nature", in seconds
-fix_duration = [
-    1.2,
-    1,
-]  # fix duration for "optimist" & "realist", in seconds
+parts_to_edit = [[1.42, 2.44], [4.04, 4.9], ]  # stard_ends of "nature" & "mother nature", in seconds
+fix_duration = [1.2, 1, ]  # fix duration for "optimist" & "realist", in seconds
 
 # audio_to_edit = "tests/ref_audio/test_zh_1_ref_short.wav"
 # origin_text = "对，这就是我，万人敬仰的太乙真人。"
@@ -92,7 +87,7 @@ if local:
     vocos = Vocos.from_hparams(f"{vocos_local_path}/config.yaml")
     state_dict = torch.load(f"{vocos_local_path}/pytorch_model.bin", weights_only=True, map_location=device)
     vocos.load_state_dict(state_dict)
-
+    
     vocos.eval()
 else:
     vocos = Vocos.from_pretrained("charactr/vocos-mel-24khz")
@@ -102,19 +97,23 @@ vocab_char_map, vocab_size = get_tokenizer(dataset_name, tokenizer)
 
 # Model
 model = CFM(
-    transformer=model_cls(**model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels),
-    mel_spec_kwargs=dict(
-        target_sample_rate=target_sample_rate,
-        n_mel_channels=n_mel_channels,
-        hop_length=hop_length,
+    transformer = model_cls(
+        **model_cfg,
+        text_num_embeds = vocab_size, 
+        mel_dim = n_mel_channels
+    ),
+    mel_spec_kwargs = dict(
+        target_sample_rate = target_sample_rate, 
+        n_mel_channels = n_mel_channels,
+        hop_length = hop_length,
     ),
-    odeint_kwargs=dict(
-        method=ode_method,
+    odeint_kwargs = dict(
+        method = ode_method,
     ),
-    vocab_char_map=vocab_char_map,
+    vocab_char_map = vocab_char_map,
 ).to(device)
 
-model = load_checkpoint(model, ckpt_path, device, use_ema=use_ema)
+model = load_checkpoint(model, ckpt_path, device, use_ema = use_ema)
 
 # Audio
 audio, sr = torchaudio.load(audio_to_edit)
@@ -134,18 +133,14 @@ for part in parts_to_edit:
     part_dur = end - start if fix_duration is None else fix_duration.pop(0)
     part_dur = part_dur * target_sample_rate
     start = start * target_sample_rate
-    audio_ = torch.cat((audio_, audio[:, round(offset) : round(start)], torch.zeros(1, round(part_dur))), dim=-1)
-    edit_mask = torch.cat(
-        (
-            edit_mask,
-            torch.ones(1, round((start - offset) / hop_length), dtype=torch.bool),
-            torch.zeros(1, round(part_dur / hop_length), dtype=torch.bool),
-        ),
-        dim=-1,
-    )
+    audio_ = torch.cat((audio_, audio[:, round(offset):round(start)], torch.zeros(1, round(part_dur))), dim = -1)
+    edit_mask = torch.cat((edit_mask, 
+                           torch.ones(1, round((start - offset) / hop_length), dtype = torch.bool), 
+                           torch.zeros(1, round(part_dur / hop_length), dtype = torch.bool)
+                           ), dim = -1)
     offset = end * target_sample_rate
 # audio = torch.cat((audio_, audio[:, round(offset):]), dim = -1)
-edit_mask = F.pad(edit_mask, (0, audio.shape[-1] // hop_length - edit_mask.shape[-1] + 1), value=True)
+edit_mask = F.pad(edit_mask, (0, audio.shape[-1] // hop_length - edit_mask.shape[-1] + 1), value = True)
 audio = audio.to(device)
 edit_mask = edit_mask.to(device)
 
@@ -165,25 +160,24 @@ duration = audio.shape[-1] // hop_length
 # Inference
 with torch.inference_mode():
     generated, trajectory = model.sample(
-        cond=audio,
-        text=final_text_list,
-        duration=duration,
-        steps=nfe_step,
-        cfg_strength=cfg_strength,
-        sway_sampling_coef=sway_sampling_coef,
-        seed=seed,
-        edit_mask=edit_mask,
+        cond = audio,
+        text = final_text_list,
+        duration = duration,
+        steps = nfe_step,
+        cfg_strength = cfg_strength,
+        sway_sampling_coef = sway_sampling_coef,
+        seed = seed,
+        edit_mask = edit_mask,
     )
 print(f"Generated mel: {generated.shape}")
 
 # Final result
-generated = generated.to(torch.float32)
 generated = generated[:, ref_audio_len:, :]
-generated_mel_spec = generated.permute(0, 2, 1)
+generated_mel_spec = rearrange(generated, '1 n d -> 1 d n')
 generated_wave = vocos.decode(generated_mel_spec.cpu())
 if rms < target_rms:
     generated_wave = generated_wave * rms / target_rms
 
-save_spectrogram(generated_mel_spec[0].cpu().numpy(), f"{output_dir}/speech_edit_out.png")
-torchaudio.save(f"{output_dir}/speech_edit_out.wav", generated_wave, target_sample_rate)
+save_spectrogram(generated_mel_spec[0].cpu().numpy(), f"{output_dir}/test_single_edit.png")
+torchaudio.save(f"{output_dir}/test_single_edit.wav", generated_wave, target_sample_rate)
 print(f"Generated wav: {generated_wave.shape}")

src/f5_tts/api.py

@@ -1,166 +0,0 @@
-import random
-import sys
-from importlib.resources import files
-
-import soundfile as sf
-import tqdm
-from cached_path import cached_path
-
-from f5_tts.infer.utils_infer import (
-    hop_length,
-    infer_process,
-    load_model,
-    load_vocoder,
-    preprocess_ref_audio_text,
-    remove_silence_for_generated_wav,
-    save_spectrogram,
-    transcribe,
-    target_sample_rate,
-)
-from f5_tts.model import DiT, UNetT
-from f5_tts.model.utils import seed_everything
-
-
-class F5TTS:
-    def __init__(
-        self,
-        model_type="F5-TTS",
-        ckpt_file="",
-        vocab_file="",
-        ode_method="euler",
-        use_ema=True,
-        vocoder_name="vocos",
-        local_path=None,
-        device=None,
-        hf_cache_dir=None,
-    ):
-        # Initialize parameters
-        self.final_wave = None
-        self.target_sample_rate = target_sample_rate
-        self.hop_length = hop_length
-        self.seed = -1
-        self.mel_spec_type = vocoder_name
-
-        # Set device
-        if device is not None:
-            self.device = device
-        else:
-            import torch
-
-            self.device = "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu"
-
-        # Load models
-        self.load_vocoder_model(vocoder_name, local_path=local_path, hf_cache_dir=hf_cache_dir)
-        self.load_ema_model(
-            model_type, ckpt_file, vocoder_name, vocab_file, ode_method, use_ema, hf_cache_dir=hf_cache_dir
-        )
-
-    def load_vocoder_model(self, vocoder_name, local_path=None, hf_cache_dir=None):
-        self.vocoder = load_vocoder(vocoder_name, local_path is not None, local_path, self.device, hf_cache_dir)
-
-    def load_ema_model(self, model_type, ckpt_file, mel_spec_type, vocab_file, ode_method, use_ema, hf_cache_dir=None):
-        if model_type == "F5-TTS":
-            if not ckpt_file:
-                if mel_spec_type == "vocos":
-                    ckpt_file = str(
-                        cached_path("hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.safetensors", cache_dir=hf_cache_dir)
-                    )
-                elif mel_spec_type == "bigvgan":
-                    ckpt_file = str(
-                        cached_path("hf://SWivid/F5-TTS/F5TTS_Base_bigvgan/model_1250000.pt", cache_dir=hf_cache_dir)
-                    )
-            model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
-            model_cls = DiT
-        elif model_type == "E2-TTS":
-            if not ckpt_file:
-                ckpt_file = str(
-                    cached_path("hf://SWivid/E2-TTS/E2TTS_Base/model_1200000.safetensors", cache_dir=hf_cache_dir)
-                )
-            model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
-            model_cls = UNetT
-        else:
-            raise ValueError(f"Unknown model type: {model_type}")
-
-        self.ema_model = load_model(
-            model_cls, model_cfg, ckpt_file, mel_spec_type, vocab_file, ode_method, use_ema, self.device
-        )
-
-    def transcribe(self, ref_audio, language=None):
-        return transcribe(ref_audio, language)
-
-    def export_wav(self, wav, file_wave, remove_silence=False):
-        sf.write(file_wave, wav, self.target_sample_rate)
-
-        if remove_silence:
-            remove_silence_for_generated_wav(file_wave)
-
-    def export_spectrogram(self, spect, file_spect):
-        save_spectrogram(spect, file_spect)
-
-    def infer(
-        self,
-        ref_file,
-        ref_text,
-        gen_text,
-        show_info=print,
-        progress=tqdm,
-        target_rms=0.1,
-        cross_fade_duration=0.15,
-        sway_sampling_coef=-1,
-        cfg_strength=2,
-        nfe_step=32,
-        speed=1.0,
-        fix_duration=None,
-        remove_silence=False,
-        file_wave=None,
-        file_spect=None,
-        seed=-1,
-    ):
-        if seed == -1:
-            seed = random.randint(0, sys.maxsize)
-        seed_everything(seed)
-        self.seed = seed
-
-        ref_file, ref_text = preprocess_ref_audio_text(ref_file, ref_text, device=self.device)
-
-        wav, sr, spect = infer_process(
-            ref_file,
-            ref_text,
-            gen_text,
-            self.ema_model,
-            self.vocoder,
-            self.mel_spec_type,
-            show_info=show_info,
-            progress=progress,
-            target_rms=target_rms,
-            cross_fade_duration=cross_fade_duration,
-            nfe_step=nfe_step,
-            cfg_strength=cfg_strength,
-            sway_sampling_coef=sway_sampling_coef,
-            speed=speed,
-            fix_duration=fix_duration,
-            device=self.device,
-        )
-
-        if file_wave is not None:
-            self.export_wav(wav, file_wave, remove_silence)
-
-        if file_spect is not None:
-            self.export_spectrogram(spect, file_spect)
-
-        return wav, sr, spect
-
-
-if __name__ == "__main__":
-    f5tts = F5TTS()
-
-    wav, sr, spect = f5tts.infer(
-        ref_file=str(files("f5_tts").joinpath("infer/examples/basic/basic_ref_en.wav")),
-        ref_text="some call me nature, others call me mother nature.",
-        gen_text="""I don't really care what you call me. I've been a silent spectator, watching species evolve, empires rise and fall. But always remember, I am mighty and enduring. Respect me and I'll nurture you; ignore me and you shall face the consequences.""",
-        file_wave=str(files("f5_tts").joinpath("../../tests/api_out.wav")),
-        file_spect=str(files("f5_tts").joinpath("../../tests/api_out.png")),
-        seed=-1,  # random seed = -1
-    )
-
-    print("seed :", f5tts.seed)

src/f5_tts/configs/E2TTS_Base_train.yaml

@@ -1,44 +0,0 @@
-hydra:
-  run:
-    dir: ckpts/${model.name}_${model.mel_spec.mel_spec_type}_${model.tokenizer}_${datasets.name}/${now:%Y-%m-%d}/${now:%H-%M-%S}
-    
-datasets:
-  name: Emilia_ZH_EN  # dataset name
-  batch_size_per_gpu: 38400  # 8 GPUs, 8 * 38400 = 307200
-  batch_size_type: frame  # "frame" or "sample"
-  max_samples: 64  # max sequences per batch if use frame-wise batch_size. we set 32 for small models, 64 for base models
-  num_workers: 16
-
-optim:
-  epochs: 15
-  learning_rate: 7.5e-5
-  num_warmup_updates: 20000  # warmup steps
-  grad_accumulation_steps: 1  # note: updates = steps / grad_accumulation_steps
-  max_grad_norm: 1.0  # gradient clipping
-  bnb_optimizer: False  # use bnb 8bit AdamW optimizer or not
-
-model:
-  name: E2TTS_Base
-  tokenizer: pinyin
-  tokenizer_path: None  # if tokenizer = 'custom', define the path to the tokenizer you want to use (should be vocab.txt)
-  arch:
-    dim: 1024
-    depth: 24
-    heads: 16
-    ff_mult: 4
-  mel_spec:
-    target_sample_rate: 24000
-    n_mel_channels: 100
-    hop_length: 256
-    win_length: 1024
-    n_fft: 1024
-    mel_spec_type: vocos  # 'vocos' or 'bigvgan'
-  vocoder:
-    is_local: False  # use local offline ckpt or not
-    local_path: None  # local vocoder path
-
-ckpts:
-  logger: wandb  # wandb | tensorboard | None
-  save_per_updates: 50000  # save checkpoint per steps
-  last_per_steps: 5000  # save last checkpoint per steps
-  save_dir: ckpts/${model.name}_${model.mel_spec.mel_spec_type}_${model.tokenizer}_${datasets.name}