This bounty is for bringing up the LLVC (Low-Latency Low-Resource Voice Conversion) model using TTNN APIs on Tenstorrent hardware (Wormhole or Blackhole).
LLVC is a real-time voice conversion model from Koe AI specifically optimized for low latency and CPU efficiency. Published in 2023, key features include:
Ultra-low latency: Designed for real-time voice conversion with minimal delay
CPU-optimized: Efficient enough to run on CPU in real-time
Streaming support: True streaming inference with chunked processing
High quality: Natural voice conversion while maintaining low latency
Compatible architecture: Based on RVC/QuickVC with optimizations
Small model size: Low resource requirements for edge deployment
No F0 dependency: Optional pitch-independent conversion mode
MIT License: Free for commercial use
LLVC achieves real-time factor (RTF) < 0.1 on CPU, making it suitable for live applications like real-time dubbing, streaming, gaming voice chat, and interactive voice applications.
The goal is to enable this model to run on TT hardware for ultra-high-throughput, ultra-low-latency voice conversion optimized for real-time applications.
What Success Looks Like
A successful submission will fulfill all requirements in the following stages. Payout is made after all three stages are completed.
Stage 1 β Bring-Up
Implement LLVC using TTNN APIs (Python)
Implements the full generation pipeline:
Lightweight encoder (optimized for low latency)
Streaming-capable architecture
Efficient decoder
Optional pitch extraction (F0-based and F0-free modes)
Vocoder integration
Model runs on either N150 or N300 Tenstorrent hardware with no errors
Supports both modes:
Streaming mode: Real-time conversion with chunked processing
Non-streaming mode: Full-context conversion
Produces valid converted audio output
Output is verifiable (audio quality assessment, compare with PyTorch reference)
Achieves baseline throughput target:
At least 50 tokens/second for decoder generation
Real-time factor (RTF) < 0.3 for streaming mode
Latency < 100ms for streaming chunks
Accuracy evaluation:
Speaker similarity > 70% (cosine similarity)
Content preservation: WER < 3.0
Token-level accuracy > 95% against PyTorch reference
Audio quality: Natural prosody with minimal artifacts despite low latency
Clear instructions for setup and running the model
Stage 2 β Basic Optimizations
Use optimal sharded/interleaved memory configs for encoder-decoder layers
Implement efficient sharding strategy for:
Lightweight encoder layers
Decoder layers optimized for streaming
Convolutional layers with causal padding
Cached convolution states for streaming
Optional pitch extraction module
Fuse simple ops where possible (e.g., layer normalization, activation functions)
Store intermediate activations in L1 where beneficial
Use recommended TTNN/tt-metal flows for streaming audio models
Leverage TT library of fused ops for convolution blocks
Optimize chunk-based processing for streaming
Efficient state management for causal convolutions
Streaming inference pipeline with state management
Validate each submoduleβs output against PyTorch reference before integration
Experiment with different sharding strategies:
Channel sharding for convolutions
Sequence length sharding for chunks
Minimize cross-chip communication
Use TTNN profiling tools to identify bottlenecks in:
Per-chunk processing time (critical metric)
Encoder latency
Decoder latency
State management overhead
Vocoder latency
Test diverse use cases:
Streaming mode with various chunk sizes (50ms, 100ms, 200ms)
Non-streaming mode for comparison
F0-based vs F0-free modes
Multiple concurrent streams
Real-time conversion scenarios
Open a draft PR early to get feedback on your approach
Result Submission Guidelines
Beyond the model implementation itself, contributors must submit the following material as proof of work.
However, feel free to open a PR at any time if you want us checking that you are on the right track.
Just understand that payout is only made after all 3 stages are completed.