Signed-off-by: Yuekai Zhang <zhangyuekai@foxmail.com> Co-authored-by: Claude Fable 5 <noreply@anthropic.com>
114 lines
5.3 KiB
Markdown
114 lines
5.3 KiB
Markdown
# NemotronLabs VoiceChat 11B
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> Offline speech-to-speech: a 16 kHz user utterance in, the agent's spoken reply
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> (text + 22.05 kHz WAV) out, on a 3-stage vLLM-Omni pipeline.
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## Summary
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- Vendor: NVIDIA
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- Model: [`nvidia/NVIDIA-NemotronLabs-VoiceChat-11B`](https://huggingface.co/nvidia/NVIDIA-NemotronLabs-VoiceChat-11B)
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- Task: Speech-to-speech voice chat. The model runs a frame-locked 12.5 Hz
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timeline: a Conformer + NemotronH hybrid-Mamba thinker emits one text token
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per acoustic frame, a Gemma3-1B EAR-TTS talker turns the text timeline into
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31-quantizer RVQ code stacks, and an RVQ-VAE codec decodes them to audio.
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- Mode: Offline single-turn inference (batch=1). Online/duplex serving,
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function calling, and batch>1 are documented follow-ups.
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- Maintainer: [`@yuekaizhang`](https://github.com/yuekaizhang)
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## When to use this recipe
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Use this recipe to run a single-turn voice-chat exchange with
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`NVIDIA-NemotronLabs-VoiceChat-11B` on one GPU: you provide a user utterance as
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a WAV file (any sample rate; it is resampled to 16 kHz mono) plus an optional
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spoken-style system prompt, and get back the agent's reply as text and a
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22.05 kHz WAV. The integration is NeMo-free at runtime — the perception
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Conformer, EAR-TTS talker, and RVQ-VAE codec are vendored, dependency-stripped
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NeMo modules (`nemo_vendored/`), so no `nemo_toolkit` install is needed.
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## References
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- Offline example:
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[`examples/offline_inference/nemotron_voicechat/end2end.py`](../../examples/offline_inference/nemotron_voicechat/end2end.py)
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- Model modules (thinker / talker / code2wav / vendored NeMo):
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[`vllm_omni/model_executor/models/nemotron_voicechat/`](../../vllm_omni/model_executor/models/nemotron_voicechat/)
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- Staged pipeline config:
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[`vllm_omni/deploy/nemotron_labs_voicechat.yaml`](../../vllm_omni/deploy/nemotron_labs_voicechat.yaml)
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- Upstream model card:
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[`nvidia/NVIDIA-NemotronLabs-VoiceChat-11B`](https://huggingface.co/nvidia/NVIDIA-NemotronLabs-VoiceChat-11B)
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- Reference implementation: NVIDIA-NeMo/Speech, branch `nemotron-labs-voicechat`
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## Pipeline
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| stage | arch | dtype | role |
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| 0 thinker | `NemotronVoiceChatThinkerForConditionalGeneration` (LLM_AR) | fp32 | WAV + system prompt -> frame-locked text-token timeline (+ function channel) |
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| 1 talker | `NemotronVoiceChatTalker` (LLM_AR) | fp32 | text timeline -> 31-quantizer RVQ code stacks (one per 80 ms frame) |
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| 2 code2wav | `NemotronVoiceChatCode2Wav` (LLM_GENERATION) | fp32 | RVQ-VAE decode -> 22.05 kHz PCM |
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Stages 0/1 default to fp32 for exact parity with the NeMo reference
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implementation (greedy decoding matches it token for token on the acceptance
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fixture). The deploy yaml documents a ~2x-faster bf16 thinker option whose
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output stayed within one word of the reference in testing.
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## Hardware Support
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## GPU
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### 1x H100 80GB
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#### Environment
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- OS: Linux
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- Python: 3.12
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- vLLM version: 0.26.0
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- vLLM-Omni version or commit: this PR / current `main`
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#### Command
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```bash
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# Tokenizer: the checkpoint ships no HF tokenizer; it resolves from the
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# nvidia/NVIDIA-Nemotron-Nano-9B-v2 HF id automatically. For air-gapped runs,
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# point NEMOTRON_VOICECHAT_LLM_PATH at a local snapshot of that repo instead.
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python examples/offline_inference/nemotron_voicechat/end2end.py \
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--checkpoint /path/to/NVIDIA-NemotronLabs-VoiceChat-11B \
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--wav /path/to/user_question.wav \
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--output-dir results/nemotron_voicechat
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```
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#### Verification
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```bash
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ls results/nemotron_voicechat
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# <stem>_output.txt the agent reply as text
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# <stem>_output.wav the agent reply as 22.05 kHz audio
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# <stem>_text_tokens.json the frame-locked text-token timeline
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```
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The reply text should read as a coherent spoken-style answer to the question in
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the input WAV, and the WAV should transcribe to (approximately) the same text
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with any ASR model.
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#### Notes
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- Memory usage: the shipped yaml runs all three stages on one GPU
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(`gpu_memory_utilization` 0.62 / 0.12 / 0.06); peak usage is dominated by the
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fp32 thinker. The fp32 default has a hard floor of roughly 43 GB of thinker
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weights alone (9B backbone + 587M `embed_tokens` + 587M `function_head` +
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0.6B Conformer), so 48 GB cards cannot run it — use the bf16 thinker option
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documented in the deploy yaml on anything smaller than an 80 GB part.
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- Input sizing: the timeline is frame-locked, so the reply budget IS the input
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duration. The acoustic channel trails the text channel; if the WAV does not
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carry enough trailing silence for the reply to finish, the spoken answer is
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truncated silently. Leave generous trailing silence (a question ending at
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~4.5 s truncated in an 8 s WAV but completed cleanly in 16 s); the offline
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example warns when the text channel is still speaking near the last frame.
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- Key flags: sampling is greedy end to end. The thinker is frame-locked —
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`max_tokens` equals the acoustic frame count with `ignore_eos=True`. Do NOT
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set `min_tokens` on the thinker: the tokenizer's EOS token is also the
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frame-locked PAD/silence token, so masking it forces the model to babble
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instead of pausing.
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- The talker's `max_tokens` is 16383 (its stage prompt is one placeholder
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token, and the stage context is 16384).
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- Known limitations: batch=1 only; offline single turn (no online/duplex
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serving yet); the function-call channel is decoded but not yet acted on.
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