OpenVox TTS Windows v2.3.2 is out, and this is probably our biggest Windows update so far. (AMD GPU Support!)
The biggest change is under the hood: we’ve introduced an all-new native C++ TTS audio engine. By removing a lot of the Python overhead, we're seeing roughly 2x–5x faster generation depending on the model and hardware, along with lower memory usage and faster loading.
We’ve also spent quite a bit of time improving support for people who don’t have huge amounts of VRAM.
⚡ Native C++ TTS Engine
- Around 2x–5x faster generation
- Lower memory/VRAM overhead
- Around 30% faster app startup
- Native acceleration for supported models
🔴 AMD GPU Support
AMD users can now use native ROCm/HIP acceleration.
This has been one of the most requested Windows features for a while, so I'm particularly interested in hearing how it performs across different Radeon cards.
The video attached to this post shows it running on an RX 9060 XT.
🧠 Quantized Models
We've added Large and Medium quantized versions of larger TTS models including:
- Qwen3 TTS
- Chatterbox
- OmniVoice
- Vibevoice
This should make these models much more practical on GPUs with limited VRAM.
🎬 New AI Dubbing Studio
This is the other big part of the update.
You can now:
- Drop in a video or audio file
- Automatically transcribe it
- Detect up to 4 different speakers
- Translate the dialogue
- Rename/reassign speakers
- Assign a different AI voice to each speaker
- Generate synchronized dubbed speech
- Export the resulting dub
There’s also HTDemucs integration, so OpenVox can separate vocals from the original background audio and preserve things like music while replacing the dialogue.
The video demonstrates the full workflow by taking a video, identifying its speakers, separating the background track, translating the dialogue to Spanish, assigning voices and generating the final dub.
📝 Multi-Speaker Transcription
Drop & Transcribe can now perform speaker diarization as well.
You can rename speakers directly, reassign incorrectly detected segments and export a speaker-labelled, time-coded SRT.
We also fixed the Sortformer speaker diarization issue introduced in v2.3.0.
🔊 Select & Read Improvements
Select & Read has also received some attention in this release:
- Better reliability
- Smoother reading experience
- New draggable Select & Read overlay
So you can move the controls somewhere convenient instead of having them stuck in one position.
🔌 API
Pause tags are now supported through the API as well.
And there are a bunch of smaller UI optimizations and bug fixes throughout the app.
Would especially love feedback from AMD GPU users and anyone trying the new Dubbing Studio / multi-speaker transcription workflows.
If you run into something weird, let me know your GPU, VRAM and the model you were using. It helps a lot when tracking down hardware-specific issues.
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