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| author | soryu <soryu@soryu.co> | 2026-01-28 02:54:17 +0000 |
|---|---|---|
| committer | GitHub <noreply@github.com> | 2026-01-28 02:54:17 +0000 |
| commit | eabd1304cce0e053cd32ec910d2f0ea429e8af14 (patch) | |
| tree | fca3b08810a1dc0c0c610a8189a466cc23d5c547 /makima/src/tts/qwen3/code_predictor.rs | |
| parent | c618174e60e4632d36d7352d83399508c72b2f42 (diff) | |
| download | soryu-eabd1304cce0e053cd32ec910d2f0ea429e8af14.tar.gz soryu-eabd1304cce0e053cd32ec910d2f0ea429e8af14.zip | |
Add Qwen3-TTS streaming endpoint for voice synthesis (#40)
* Task completion checkpoint
* Task completion checkpoint
* Task completion checkpoint
* Add Qwen3-TTS research document for live TTS replacement
Research findings for replacing Chatterbox TTS with Qwen3-TTS-12Hz-0.6B-Base:
- Current TTS: Chatterbox-Turbo-ONNX with batch-only generation, no streaming
- Qwen3-TTS: 97ms end-to-end latency, streaming support, 3-second voice cloning
- Voice cloning: Requires 3s reference audio + transcript (Makima voice planned)
- Integration: Python service with WebSocket bridge (no ONNX export available)
- Languages: 10 supported including English and Japanese
Document includes:
- Current architecture analysis (makima/src/tts.rs)
- Qwen3-TTS capabilities and requirements
- Feasibility assessment for live/streaming TTS
- Audio clip requirements for voice cloning
- Preliminary technical approach with architecture diagrams
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* [WIP] Heartbeat checkpoint - 2026-01-27 03:11:15 UTC
* Add Qwen3-TTS research documentation
Comprehensive research on replacing Chatterbox TTS with Qwen3-TTS-12Hz-0.6B-Base:
- Current TTS implementation analysis (Chatterbox-Turbo-ONNX in makima/src/tts.rs)
- Qwen3-TTS capabilities: 97ms streaming latency, voice cloning with 3s reference
- Cross-lingual support: Japanese voice (Makima/Tomori Kusunoki) speaking English
- Python microservice architecture recommendation (FastAPI + WebSocket)
- Implementation phases and technical approach
- Hardware requirements and dependencies
Key findings:
- Live/streaming TTS is highly feasible with 97ms latency
- Voice cloning fully supported with 0.95 speaker similarity
- Recommended: Python microservice with WebSocket streaming
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* Add comprehensive Qwen3-TTS integration specification
This specification document defines the complete integration of
Qwen3-TTS-12Hz-0.6B-Base as a replacement for the existing Chatterbox-Turbo
TTS implementation. The document covers:
## Functional Requirements
- WebSocket endpoint /api/v1/speak for streaming TTS
- Voice cloning with default Makima voice (Japanese VA speaking English)
- Support for custom voice references
- Detailed client-to-server and server-to-client message protocols
- Integration with Listen page for bidirectional speech
## Non-Functional Requirements
- Latency targets: < 200ms first audio byte
- Audio quality: 24kHz, mono, PCM16/PCM32f
- Hardware requirements: CUDA GPU with 4-8GB VRAM
- Scalability: 10 concurrent sessions per GPU
## Architecture Specification
- Python TTS microservice with FastAPI/WebSocket
- Rust proxy endpoint in makima server
- Voice prompt caching mechanism (LRU cache)
- Error handling and recovery strategies
## API Contract
- Complete WebSocket message format definitions (TypeScript)
- Error codes and responses (TTS_UNAVAILABLE, SYNTHESIS_ERROR, etc.)
- Session state machine and lifecycle management
## Voice Asset Requirements
- Makima voice clip specifications (5-10s WAV, transcript required)
- Storage location: models/voices/makima/
- Metadata format for voice management
## Testing Strategy
- Unit tests for Python TTS service and Rust proxy
- Integration tests for WebSocket flow
- Latency benchmarks with performance targets
- Test data fixtures for various text lengths
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* Add Qwen3-TTS implementation plan
Comprehensive implementation plan for replacing Chatterbox-TTS with
Qwen3-TTS streaming TTS service, including:
- Task breakdown with estimated hours for each phase
- Phase 1: Python TTS microservice (FastAPI, WebSocket)
- Phase 2: Rust proxy integration (speak.rs, tts_client.rs)
- Detailed file changes and new module structure
- Testing plan with unit, integration, and latency benchmarks
- Risk assessment with mitigation strategies
- Success criteria for each phase
Based on specification in docs/specs/qwen3-tts-spec.md
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* Add author and research references to TTS implementation plan
Add links to research documentation and author attribution.
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* [WIP] Heartbeat checkpoint - 2026-01-27 03:25:06 UTC
* Add Python TTS service project structure (Phase 1.1-1.3)
Create the initial makima-tts Python service directory structure with:
- pyproject.toml with FastAPI, Qwen-TTS, and torch dependencies
- config.py with pydantic-settings TTSConfig class
- models.py with Pydantic message models (Start, Speak, Stop, Ready, etc.)
This implements tasks P1.1, P1.2, and P1.3 from the Qwen3-TTS implementation plan.
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* Add TTS engine and voice manager for Qwen3-TTS (Phase 1.4-1.5)
Implement core TTS functionality:
- tts_engine.py: Qwen3-TTS wrapper with streaming audio chunk generation
- voice_manager.py: Voice prompt caching with LRU eviction and TTL support
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* [WIP] Heartbeat checkpoint - 2026-01-27 03:30:06 UTC
* Add TTS proxy client and message types (Phase 2.1, 2.2, 2.4)
- Add tts_client.rs with TtsConfig, TtsCircuitBreaker, TtsError,
TtsProxyClient, and TtsConnection structs for WebSocket proxying
- Add TTS message types to messages.rs (TtsAudioEncoding, TtsPriority,
TtsStartMessage, TtsSpeakMessage, TtsStopMessage, TtsClientMessage,
TtsReadyMessage, TtsAudioChunkMessage, TtsCompleteMessage,
TtsErrorMessage, TtsStoppedMessage, TtsServerMessage)
- Export tts_client module from server mod.rs
- tokio-tungstenite already present in Cargo.toml
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* Add TTS WebSocket handler and route (Phase 2.3, 2.5, 2.6)
- Create speak.rs WebSocket handler that proxies to Python TTS service
- Add TtsState fields (tts_client, tts_config) to AppState
- Add with_tts() builder and is_tts_healthy() methods to AppState
- Register /api/v1/speak route in the router
- Add speak module export in handlers/mod.rs
The handler forwards WebSocket messages bidirectionally between
the client and the Python TTS microservice with proper error handling.
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* Add Makima voice profile assets for TTS voice cloning
Creates the voice assets directory structure with:
- manifest.json containing voice configuration (voice_id, speaker,
language, reference audio path, and Japanese transcript placeholder)
- README.md with instructions for obtaining voice reference audio
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* Add Rust-native Qwen3-TTS integration research document
Research findings for integrating Qwen3-TTS-12Hz-0.6B-Base directly into
the makima Rust codebase without Python. Key conclusions:
- ONNX export is not viable (unsupported architecture)
- Candle (HF Rust ML framework) is the recommended approach
- Model weights available in safetensors format (2.52GB total)
- Three components needed: LM backbone, code predictor, speech tokenizer
- Crane project has Qwen3-TTS as highest priority (potential upstream)
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* [WIP] Heartbeat checkpoint - 2026-01-27 11:21:43 UTC
* [WIP] Heartbeat checkpoint - 2026-01-27 11:24:19 UTC
* [WIP] Heartbeat checkpoint - 2026-01-27 11:26:43 UTC
* feat: implement Rust-native Qwen3-TTS using candle framework
Replace monolithic tts.rs with modular tts/ directory structure:
- tts/mod.rs: TtsEngine trait, TtsEngineFactory, shared types (AudioChunk,
TtsError), and utility functions (save_wav, resample, argmax)
- tts/chatterbox.rs: existing ONNX-based ChatterboxTTS adapted to implement
TtsEngine trait with Mutex-wrapped sessions for Send+Sync
- tts/qwen3/mod.rs: Qwen3Tts entry point with HuggingFace model loading
- tts/qwen3/config.rs: Qwen3TtsConfig parsing from HF config.json
- tts/qwen3/model.rs: 28-layer Qwen3 transformer with RoPE, GQA (16 heads,
8 KV heads), SiLU MLP, RMS norm, and KV cache
- tts/qwen3/code_predictor.rs: 5-layer MTP module predicting 16 codebooks
- tts/qwen3/speech_tokenizer.rs: ConvNet encoder/decoder with 16-layer RVQ
- tts/qwen3/generate.rs: autoregressive generation loop with streaming support
Add candle-core, candle-nn, candle-transformers, safetensors to Cargo.toml.
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* feat: integrate TTS engine into speak WebSocket handler
- Update speak.rs handler to use TTS engine directly from SharedState
instead of returning a stub "not implemented" error
- Add TtsEngine (OnceCell lazy-loaded) to AppState in state.rs with
get_tts_engine() method for lazy initialization on first connection
- Implement full WebSocket protocol: client sends JSON speak/cancel/stop
messages, server streams binary PCM audio chunks and audio_end signals
- Create voices/makima/manifest.json for Makima voice profile configuration
- All files compile successfully with zero errors
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
* feat: add /speak TTS page with WebSocket audio playback
Add a new /speak frontend page for text-to-speech via WebSocket.
The page accepts text input and streams synthesized PCM audio through
the Web Audio API. Includes model loading indicator, cancel support,
and connection status. Also adds a loading bar to the listen page
ControlPanel during WebSocket connection.
Co-Authored-By: Claude Opus 4.5 <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 4.5 <noreply@anthropic.com>
Diffstat (limited to 'makima/src/tts/qwen3/code_predictor.rs')
| -rw-r--r-- | makima/src/tts/qwen3/code_predictor.rs | 261 |
1 files changed, 261 insertions, 0 deletions
diff --git a/makima/src/tts/qwen3/code_predictor.rs b/makima/src/tts/qwen3/code_predictor.rs new file mode 100644 index 0000000..0ef8a1d --- /dev/null +++ b/makima/src/tts/qwen3/code_predictor.rs @@ -0,0 +1,261 @@ +//! Multi-Token Prediction (MTP) code predictor. +//! +//! After the main LM predicts the zeroth codebook token, this module +//! predicts the remaining 15 codebook layers in parallel from the +//! LM's hidden states. +//! +//! Architecture: +//! - 5 transformer layers (same structure as main LM layers) +//! - 16 output heads, one per codebook (vocab 2048 each) +//! - Input: last hidden state from main LM + zeroth codebook embedding +//! - Output: 16 codebook token predictions + +use candle_core::{Device, Module, Result, Tensor, D}; +use candle_nn::{embedding, linear_no_bias, rms_norm, Embedding, Linear, RmsNorm, VarBuilder}; + +use super::config::{CodePredictorConfig, Qwen3LmConfig}; +use super::model::{KvCache, Qwen3Attention, Qwen3Mlp, RotaryEmbedding}; + +/// A single code predictor transformer layer. +/// +/// Uses the same pre-norm residual structure as the main LM layers. +pub struct CodePredictorLayer { + self_attn: Qwen3Attention, + mlp: Qwen3Mlp, + input_layernorm: RmsNorm, + post_attention_layernorm: RmsNorm, +} + +impl CodePredictorLayer { + pub fn new(config: &CodePredictorConfig, vb: VarBuilder) -> Result<Self> { + // Construct a Qwen3LmConfig-like view for the attention/MLP constructors + let lm_config = Qwen3LmConfig { + hidden_size: config.hidden_size, + num_hidden_layers: config.num_layers, + num_attention_heads: config.num_attention_heads, + num_key_value_heads: config.num_attention_heads, // No GQA in predictor + intermediate_size: config.hidden_size * 3, // 3072 for hidden=1024 + head_dim: config.hidden_size / config.num_attention_heads, + rms_norm_eps: config.rms_norm_eps, + ..Qwen3LmConfig::default() + }; + + let self_attn = Qwen3Attention::new(&lm_config, vb.pp("self_attn"))?; + let mlp = Qwen3Mlp::new(&lm_config, vb.pp("mlp"))?; + let input_layernorm = rms_norm( + config.hidden_size, + config.rms_norm_eps, + vb.pp("input_layernorm"), + )?; + let post_attention_layernorm = rms_norm( + config.hidden_size, + config.rms_norm_eps, + vb.pp("post_attention_layernorm"), + )?; + + Ok(Self { + self_attn, + mlp, + input_layernorm, + post_attention_layernorm, + }) + } + + pub fn forward( + &self, + hidden_states: &Tensor, + rope: &RotaryEmbedding, + kv_cache: &mut KvCache, + attention_mask: Option<&Tensor>, + ) -> Result<Tensor> { + let residual = hidden_states; + let hidden_states = self.input_layernorm.forward(hidden_states)?; + let hidden_states = + self.self_attn + .forward(&hidden_states, rope, kv_cache, attention_mask)?; + let hidden_states = (residual + hidden_states)?; + + let residual = &hidden_states; + let hidden_states = self.post_attention_layernorm.forward(&hidden_states)?; + let hidden_states = self.mlp.forward(&hidden_states)?; + let output = (residual + hidden_states)?; + + Ok(output) + } +} + +/// Multi-token prediction code predictor. +/// +/// Takes the hidden states from the main LM and predicts all 16 codebook +/// tokens. The zeroth codebook is predicted by the main LM head; this +/// module predicts the remaining 15 residual codebooks. +pub struct CodePredictor { + /// Embedding layer for codebook tokens (shared across groups). + code_embeddings: Vec<Embedding>, + /// Projection from LM hidden + code embedding to predictor hidden. + input_proj: Linear, + /// 5 transformer layers. + layers: Vec<CodePredictorLayer>, + /// Final normalization. + norm: RmsNorm, + /// Per-codebook output heads (16 heads, each projecting to codebook_vocab_size). + output_heads: Vec<Linear>, + /// RoPE for the predictor's attention layers. + rope: RotaryEmbedding, + config: CodePredictorConfig, +} + +impl CodePredictor { + pub fn new( + config: &CodePredictorConfig, + lm_config: &Qwen3LmConfig, + vb: VarBuilder, + ) -> Result<Self> { + let predictor_vb = vb.pp("code_predictor"); + + // Code embeddings for each codebook group + let mut code_embeddings = Vec::with_capacity(config.num_code_groups); + for i in 0..config.num_code_groups { + let emb = embedding( + config.codebook_vocab_size, + config.hidden_size, + predictor_vb.pp(format!("code_embeddings.{i}")), + )?; + code_embeddings.push(emb); + } + + // Input projection: LM hidden (1024) + code embedding (1024) -> predictor hidden (1024) + let input_proj = linear_no_bias( + config.hidden_size * 2, + config.hidden_size, + predictor_vb.pp("input_proj"), + )?; + + // Transformer layers + let mut layers = Vec::with_capacity(config.num_layers); + for i in 0..config.num_layers { + let layer = + CodePredictorLayer::new(config, predictor_vb.pp(format!("layers.{i}")))?; + layers.push(layer); + } + + let norm = rms_norm( + config.hidden_size, + config.rms_norm_eps, + predictor_vb.pp("norm"), + )?; + + // Output heads for each codebook + let mut output_heads = Vec::with_capacity(config.num_code_groups); + for i in 0..config.num_code_groups { + let head = linear_no_bias( + config.hidden_size, + config.codebook_vocab_size, + predictor_vb.pp(format!("output_heads.{i}")), + )?; + output_heads.push(head); + } + + // RoPE for predictor attention (uses same theta/dim as main LM but with predictor head_dim) + let predictor_head_dim = config.hidden_size / config.num_attention_heads; + let rope_config = Qwen3LmConfig { + head_dim: predictor_head_dim, + rope_theta: lm_config.rope_theta, + max_position_embeddings: lm_config.max_position_embeddings, + ..Qwen3LmConfig::default() + }; + let rope = RotaryEmbedding::new(&rope_config, vb.dtype(), vb.device())?; + + Ok(Self { + code_embeddings, + input_proj, + layers, + norm, + output_heads, + rope, + config: config.clone(), + }) + } + + /// Predict all 16 codebook tokens from the LM hidden state. + /// + /// `lm_hidden`: [batch, 1, hidden_size] — last hidden state from main LM + /// `zeroth_code`: the token predicted by the main LM head (zeroth codebook) + /// + /// Returns: Vec of 16 token indices (one per codebook), starting with zeroth_code. + pub fn predict( + &self, + lm_hidden: &Tensor, + zeroth_code: u32, + device: &Device, + ) -> Result<Vec<u32>> { + let mut all_codes = Vec::with_capacity(self.config.num_code_groups); + all_codes.push(zeroth_code); + + // The code predictor iterates through codebook groups. + // For each group i (1..16), it: + // 1. Embeds the previous codebook token + // 2. Concatenates with LM hidden state + // 3. Projects through the predictor layers + // 4. Predicts the next codebook token via output_head[i] + let mut prev_code = zeroth_code; + + for group_idx in 1..self.config.num_code_groups { + // Embed the previous codebook token + let code_tensor = Tensor::from_vec( + vec![prev_code], + (1, 1), + device, + )?; + let code_emb = self.code_embeddings[group_idx - 1].forward(&code_tensor)?; + + // Concatenate LM hidden state with code embedding + let combined = Tensor::cat(&[lm_hidden, &code_emb], D::Minus1)?; + + // Project to predictor hidden size + let mut hidden = self.input_proj.forward(&combined)?; + + // Run through predictor transformer layers (no KV cache needed — single step) + let mut kv_caches: Vec<KvCache> = + (0..self.config.num_layers).map(|_| KvCache::new()).collect(); + for (i, layer) in self.layers.iter().enumerate() { + hidden = layer.forward(&hidden, &self.rope, &mut kv_caches[i], None)?; + } + + hidden = self.norm.forward(&hidden)?; + + // Predict codebook token + let logits = self.output_heads[group_idx].forward(&hidden)?; + + // Greedy decode: argmax + let logits_flat = logits.squeeze(0)?.squeeze(0)?; // [codebook_vocab_size] + let next_code = logits_flat + .argmax(0)? + .to_scalar::<u32>()?; + + all_codes.push(next_code); + prev_code = next_code; + } + + Ok(all_codes) + } + + /// Number of codebook groups. + pub fn num_code_groups(&self) -> usize { + self.config.num_code_groups + } +} + +#[cfg(test)] +mod tests { + use super::*; + + #[test] + fn test_code_predictor_config() { + let config = CodePredictorConfig::default(); + assert_eq!(config.num_layers, 5); + assert_eq!(config.num_code_groups, 16); + assert_eq!(config.codebook_vocab_size, 2048); + assert_eq!(config.hidden_size, 1024); + } +} |
