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| license: mit | |
| language: | |
| - zh | |
| - en | |
| base_model: inclusionAI/Ming-3.0-audio-tiny | |
| tags: | |
| - audio.cpp | |
| - gguf | |
| - automatic-speech-recognition | |
| - text-to-speech | |
| # Ming 3.0 Audio Tiny GGUF | |
| Follow or star [audio.cpp](https://github.com/0xShug0/audio.cpp) for the inference | |
| code and integration updates. | |
| ### Update 2026-10-23: The model now runs 2x faster with 20% lower peak VRAM on CUDA in fully resident mode! For upstream music example (152.84 s) and meeting example (1,244.64 s), Q4_K resident mode RTF drops from 0.050 to 0.024, while peak VRAM decreases from 14,600 to 11,596 MiB. | |
| GGUF conversion of | |
| [inclusionAI/Ming-3.0-audio-tiny](https://huggingface.co/inclusionAI/Ming-3.0-audio-tiny/tree/25b96e30be3739849bf76c0f48ac11db4a262fb3), | |
| pinned to revision `25b96e30be3739849bf76c0f48ac11db4a262fb3`. | |
| Upstream code: [Ming-3.0-audio-flash](https://github.com/inclusionAI/Ming-3.0-audio-flash). | |
| The single GGUF contains the audio encoder, KDA/MLA MoE Thinker, Talker, audio VAE, | |
| CAMPPlus speaker encoder, tokenizers, and configuration. It supports offline | |
| transcription, audio understanding, named voices, voice cloning, and spoken | |
| responses. Speech output is mono 44.1 kHz. | |
| ## Package | |
| | File | Precision | | |
| | --- | --- | | |
| | `ming-3.0-audio-tiny-bf16.gguf` | Original BF16 tensors; expert routers and CAMPPlus in F32 | | |
| | `ming-3.0-audio-tiny-q8_0.gguf` | Q8_0 Thinker expert banks; other tensors retain source precision | | |
| | `ming-3.0-audio-tiny-q4_k.gguf` | Q4_K Thinker expert banks; Q8_0 text output head | | |
| Both quantized packages preserve the audio encoder, Thinker attention and MLP projections, | |
| Talker, VAE and speaker encoder. TTS-only memory usage therefore does not shrink | |
| with Thinker expert quantization. | |
| ## Usage | |
| ```bash | |
| audiocpp_cli --family ming3_audio --task asr --backend cuda \ | |
| --model ming-3.0-audio-tiny-bf16.gguf --audio recording.wav --log | |
| audiocpp_cli --family ming3_audio --task tts --backend cuda \ | |
| --model ming-3.0-audio-tiny-bf16.gguf \ | |
| --text "The sun warms the earth, and a soft breeze moves through the garden." \ | |
| --request-option voice_id=US_James --request-option language=en \ | |
| --out speech.wav --log | |
| audiocpp_cli --family ming3_audio --task clon --backend cuda \ | |
| --model ming-3.0-audio-tiny-bf16.gguf --voice-ref reference.wav \ | |
| --text "Please leave the lamp beside the window." \ | |
| --request-option language=en --out clone.wav --log | |
| ``` | |
| Reference audio conditions speaker identity. Add `reference_text` through the | |
| request-option map to also use its acoustic prefix. Use `s2s` with input audio | |
| for a Thinker answer synthesized by the Talker. | |
| Prefer a reference in the target language for more reliable pronunciation. | |
| For lower GPU memory use, add | |
| `--session-option ming3_audio.disk_experts=true`. Expert weights remain in host | |
| memory and are staged into a bounded GPU cache; neural computation stays on the | |
| selected GPU. The default is eight expert slots per layer. Long speech output | |
| uses framework text chunking, controlled by `text_chunk_size` (default 100). | |
| In offload mode, Thinker KV storage grows with actual context use instead of | |
| reserving the entire generation allowance upfront; `max_tokens` is unchanged. | |
| See the [model guide](https://github.com/0xShug0/audio.cpp/blob/main/docs/models/ming3_audio.md) | |
| for controls and conversion instructions. The native frontend uses framework | |
| text chunking, not Python's word/character buffering, and does not run Python's | |
| optional FST text normalizer. Spell out numbers when pronunciation matters. | |
| ## Performance | |
| Fresh measurements collected on October 21, 2026, on an RTX 5090 with a Debug | |
| C++17 build. Each Python/audio.cpp pair uses the same input audio or text, | |
| prompt, and reference conditioning. Python uses the official BF16 checkpoint, | |
| PyTorch 2.11.0+cu128, Transformers 5.12.1, and SDPA attention. | |
| RTF is inference time divided by input audio duration for ASR/understanding, | |
| or generated audio duration for TTS; loading and file writing are excluded. | |
| Values average two measured requests after a separate warmup. Peak VRAM is sampled | |
| process usage across each corresponding request sequence, not retained memory. | |
|  | |
| ASR uses greedy decoding with a 256-token allowance. The complete meeting uses | |
| the same transcription/summary prompt and a 32,768-token allowance on both | |
| implementations, ending naturally. Q4 meeting answers differ from BF16; | |
| these are matched-input performance measurements, not an output-parity claim. | |
| Both TTS cases use the same 1,518-character English text, seed 42, guidance scale 2, | |
| sigma 0.25, and temperature 0. Named-voice TTS uses `voice_id=US_James` without | |
| reference audio; cloning uses the US_James recording and reference transcript. | |
| Both implementations keep their normal chunkers: audio.cpp uses a 100-character | |
| budget; Python uses its own word/character buffering. Timing includes reference | |
| preparation. Long outputs | |
| are 65.4 s for Python and 63.9 s for audio.cpp in the clone case, and 78.1 s | |
| versus 79.7 s for named-voice TTS. | |
| All short-ASR transcripts match Python exactly. For long cloning, Whisper turbo | |
| reports 0% WER for Python versus 0.40% for audio.cpp with the 100-character budget. | |
| The two audio.cpp clone WAVs are byte-identical after an intervening short request. | |
| For named-voice TTS, Whisper medium reports 1.98% WER for Python versus 0% for | |
| audio.cpp; the two audio.cpp named-voice WAVs are also byte-identical. | |
| CPU inference also passes short ASR and TTS functional checks; it is not included | |
| in this CUDA performance comparison. | |
| ## License | |
| MIT, following the upstream checkpoint. | |