Vocal2Midi
Vocal2Midi is a Windows-first desktop tool and inference pipeline for turning vocal audio into lyric-aligned MIDI, USTX, and supporting editing artifacts.
The current runtime is ONNX-first:
llama.cpp is used for the Qwen decoder and runs on Vulkan or CPU.
- ONNX models default to DirectML and fall back to CPU when DirectML is unavailable.
- The main user-facing entrypoint is the Fluent GUI in
app_fluent.py.
Highlights
- End-to-end vocal-to-MIDI workflow in one project
- Chinese and Japanese lyric handling
- GUI workflow for interactive use
- Batch slice + ASR CLI for folder processing
- Portable-folder packaging flow for Windows distribution
- ONNX-based inference stack for ASR, alignment, note extraction, and RMVPE
What Vocal2Midi Does
At a high level, the hybrid pipeline looks like this:
audio
-> optional RMVPE pitch curve
-> slicing
-> ASR
-> lyric matching / .lab generation
-> HubertFA forced alignment
-> GAME note extraction
-> quantization
-> export
There is also a no-lyrics path:
audio
-> optional RMVPE pitch curve
-> slicing
-> GAME pitch-only extraction
-> export
Runtime Stack
| Component |
Current backend |
Location |
| Qwen3-ASR |
ONNX Runtime + llama.cpp |
inference/qwen3asr_dml/ |
| RomajiASR |
ONNX Runtime |
inference/romaji_asr/ |
| HubertFA |
ONNX Runtime |
inference/HubertFA/ |
| GAME |
ONNX Runtime |
inference/game/ |
| RMVPE |
ONNX Runtime |
inference/API/rmvpe_api.py |
| Device normalization |
DirectML / CPU helpers |
inference/device_utils.py |
Some public function names still contain _torch for compatibility, but the active backend is ONNX-based.
Repository Layout
application/ application-layer orchestration and config objects
docs/ architecture notes and supporting docs
experiments/ local model directories
gui/ PyQt5 + qfluentwidgets desktop UI
inference/ ASR, alignment, pitch extraction, slicing, quantization, export
scripts/ batch CLI and portable build helpers
tests/ automated tests
Quick Start
1. Install dependencies
Use your preferred Python environment, then install:
pip install -r requirements.txt
The main runtime dependencies are:
onnxruntime-directml
PyQt5
PyQt-Fluent-Widgets
librosa
soundfile
mido
pyopenjtalk
An environment.yml file is also included as a reference environment snapshot.
2. Prepare model folders
By default, the GUI expects models in these locations:
| Component |
Default path |
| GAME |
experiments/GAME-1.0.3-medium-onnx |
| HubertFA |
experiments/1218_hfa_model_new_dict |
| Qwen3-ASR |
experiments/Qwen3-ASR-1.7B-dml |
| Japanese mora ASR (RomajiASR) |
experiments/romajiASR |
| RMVPE |
experiments/RMVPE/rmvpe.onnx |
You can change these paths in the GUI settings panel.
3. Launch the GUI
For a normal developer environment:
python app_fluent.py
GUI Workflow
The GUI is the main way to use Vocal2Midi interactively. It lets you:
- choose model paths
- pick the runtime device
- set slicing mode and slice length bounds
- choose language and lyric output mode
- provide optional reference lyrics
- export MIDI, USTX, text, CSV, chunk audio, and alignment artifacts
The application-layer job entrypoint is run_auto_lyric_job() in application/pipeline.py, which dispatches into the hybrid inference pipeline in inference/pipeline/auto_lyric_hybrid.py.
Language Behavior
Chinese
- Qwen3-ASR provides text transcription.
- The lyric matcher and G2P path prepare
.lab content for HubertFA.
- Lyrics can be exported in Hanzi or pinyin-oriented forms depending on mode.
Japanese
In the main hybrid lyric pipeline:
romaji and kana lyric modes use the dedicated mora ASR path
- if the output mode is
romaji, the pipeline uses mora ASR output directly
- if the output mode is
kana, the pipeline converts matched mora output to kana for display
- if reference lyrics are provided, the reference text is processed through
pyopenjtalk, converted to kana mora tokens, then converted again to romaji mora tokens for matching
This keeps Japanese lyric matching consistent with the mora-based ASR path instead of routing through the old phoneme-ASR forced-alignment branch.
The current Japanese mora / romaji ASR integration in this repository is based on
Xiantaidu/RomajiASR, the separate
Japanese singing ASR project used for the experiments/romajiASR model path
and the inference/romaji_asr/ runtime integration.
Runtime Device Rules
Visible device options in the current UI are:
Notes:
dml is the default ONNX device
- if DirectML is unavailable, ONNX Runtime falls back to CPU
- legacy
cuda values are still accepted by some public interfaces, but they are normalized to dml
llama.cpp remains CPU-based in the current design
Slicing
The user-facing slice duration settings currently support:
- minimum slice length:
0 to 60 seconds
- maximum slice length:
0 to 60 seconds
Current defaults:
- minimum:
8.0 seconds
- maximum:
22.0 seconds
Validation rules:
slice_max_sec must be greater than 0
slice_min_sec must be less than or equal to slice_max_sec
Batch Slice + ASR CLI
For folder-based batch ASR processing:
python scripts/slice_asr_cli.py <input_dir> <output_dir> \
--asr-model experiments/Qwen3-ASR-1.7B-dml \
--device dml \
--language zh
This CLI is focused on:
- scanning input audio files
- slicing audio or bypassing slicing
- running local Qwen3-ASR
- saving chunk audio and
.lab outputs
- optionally saving JSON timing / ASR metadata
Supported input extensions currently include:
Useful options:
--no-slice bypass slicing and send the whole file to ASR
--asr-batch-size ASR batch size
--file-batch-size number of audio files per batch
--rmvpe-model enable RMVPE-assisted smart slicing
--rmvpe-batch-size RMVPE batch size
--keep-model keep the ASR runtime alive across the batch
--keep-rmvpe keep the RMVPE runtime alive across the batch
--save-json save slice timing and ASR outputs as JSON
--no-recursive scan only the top level
--no-skip-existing force reprocessing of existing outputs
Japanese whole-file example:
python scripts/slice_asr_cli.py input output \
--asr-model experiments/Qwen3-ASR-1.7B-dml \
--device dml \
--language ja \
--no-slice
Windows Setup Scripts
The repository also includes Windows helper scripts:
These are useful for a smaller distribution model where the user downloads or initializes the runtime on first setup rather than receiving a fully bundled python/ folder.
Depending on the selected workflow, Vocal2Midi can export:
.mid
.ustx
.txt
.csv
TextGrid
- chunk
.wav files
.lab
- ASR matching logs
Project Notes
- The repository has already migrated away from the earlier Torch-heavy runtime design for the main inference path.
- Some historical function names remain for compatibility.
- Model assets are expected to exist locally under
experiments/ or another user-provided path.
- The codebase is still being cleaned up in places, so you may still see a few legacy names or UI strings from earlier iterations.
License
The overall Vocal2Midi repository is distributed under the Apache License 2.0. See LICENSE.
Third-party components, vendored code, model assets, dictionaries, and other embedded materials may also carry their own original licenses, notices, or attribution requirements. Those original notices remain applicable to the corresponding materials. See ACKNOWLEDGEMENTS.md and any embedded license files for details.
Development and Testing
The repo includes a focused automated test suite under tests/.
Examples:
python -m pytest tests/test_auto_lyric_hybrid_pipeline.py
python -m pytest tests/test_asr_api.py tests/test_game_api.py tests/test_rmvpe_api.py
python -m pytest tests/test_device_selection.py tests/test_hubertfa_decoder.py
For architecture details, see docs/architecture.md.
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