audio-cpp/Nemotron-3-Diarization-GGUF

Nemotron-3-Diarization-GGUF

Use With audio.cpp

Update 2026-09-26: Since the initial release, Nemotron 3 Diarization has received improved NeMo parity for ragged batches and streaming tails, selectable flash/eager attention, self-describing speaker-probability output, ASR-compatible latency profiles, and integration with Nemotron’s speaker-tagged transcription pipeline. Existing published GGUFs remain compatible.

GGUF weights for native inference with audio.cpp. Tested on CUDA, Vulkan, and Metal(Mac) backends. Nemotron 3 Diarization identifies who spoke when, with up to eight speakers. It supports offline, streaming, and server batch diarization. It returns speaker turns and timestamps, not transcribed words.

PRs are welcome!

Weights

This package was converted from the original .nemo checkpoint, not from an existing GGUF. Each file contains 362 tensors, frontend configuration, and an embedded model spec, with no separate weight or configuration files required. The BF16 file retains the original model weight precision, with the frontend mel filter stored in F32. BF16 is parity safe with Nemo with TF32 control.

Q8_0 quantizes 130 tensors; 229 remain BF16, two use F16, and the mel filter remains F32. See the Q8 caveats below before choosing it.

Upstream: nvidia/Nemotron-3-Diarization, pinned to revision 723e19c601d99b7e58fba6a14e32153e0afe48d9. This is a format conversion for audio.cpp, not an original NVIDIA release.

Run

Use a current audio.cpp build with nemotron_3_diar support and a 16 kHz WAV:

audiocpp_cli --task diar --family nemotron_3_diar \
  --model /path/to/nemotron-3-diarization-bf16.gguf \
  --backend cuda --threads 8 --audio meeting.wav \
  --turns-out turns.json --log

For streaming, add --mode streaming and, for example, --session-option nemotron_3_diar.latency_profile=low. See the model documentation for profiles and options.

Server Batching

Create server.json with an absolute path to the GGUF:

{
  "host": "127.0.0.1",
  "port": 8080,
  "backend": "cuda",
  "device": 0,
  "threads": 8,
  "lazy_load": false,
  "models": [
    {
      "id": "nemotron-3-diar",
      "family": "nemotron_3_diar",
      "path": "/path/to/nemotron-3-diarization-bf16.gguf",
      "task": "diar",
      "mode": "offline"
    }
  ]
}

audiocpp_server --config server.json --log

Send different-length recordings in one request:

curl -N http://127.0.0.1:8080/v1/batches/transcriptions \
  -F model=nemotron-3-diar \
  -F file=@meeting-a.wav \
  -F file=@meeting-b.wav \
  -F file=@meeting-c.wav

The default response is SSE. Each completed file receives a batch.transcription.result event containing speaker_turns; use index to associate it with the uploaded file. Files do not need to have equal durations.

Performance Compared With Python

Measured on an RTX 5090 using CUDA and eight CPU threads, with matching input order and chunk settings:

• —audio.cpp: debug server, BF16 or Q80 GGUF, `--log`, and the default `veryhigh` latency profile, using the server configuration above.
• —Python: official NeMo model with PyTorch 2.11.0+cu128, eval() and torch.inference_mode(), torch.set_num_threads(8), and torch.manual_seed(663). Called _check_streaming_parameters() after setting the shared parameters below. async_streaming=False.
• —Python inference used diarize(audio=waveforms, sample_rate=16000, batch_size=len(waveforms), verbose=False) with preloaded mono waveforms. Single requests used batch_size=1.
• —Both C++ and Python used identical parameters: chunk_len=340, chunk_right_context=40, fifo_len=40, spkcache_update_period=300, and spkcache_len=264. These are C++'s default very_high profile values.
• —No precision or TF32 overrides were applied.

The batch test varied batch size from one to six files and shuffled their order across repeated requests. Recordings were 37-311 seconds long. The 24 mixed batches contained 12,482 seconds of audio in total; the ten non-batch requests contained 1,531 seconds. Models stayed loaded between requests.

Python was faster on the mixed batches; audio.cpp was faster on the single requests with either GGUF. Q8 took about 10% less time than BF16 for batches, but 19% more time for single requests, while reducing peak VRAM in both. The C++ rows are fresh sequential BF16/Q8 reruns; the Python rows retain the earlier same-day measurements on the identical request plans. Each row is one workload sequence, not a median across repeated full benchmark runs. Results depend on hardware, input lengths, and batch composition.

RTF is total inference time divided by total audio duration; lower is better. Timing includes frontend and postprocessing, but excludes model loading, input file I/O, and HTTP transfer. VRAM is per-process memory sampled every 20 ms, including weights and CUDA context. Batch VRAM covers the full 33-request sequence, including nine control/reference requests omitted from the timing table; single-request VRAM includes warmup. Brief peaks may be missed.

Q8 Results And Caveats

Q8_0 is not exact-parity-safe. Use at your own risk: it can change speaker boundaries and turn counts. Performance and VRAM for matched BF16/Q8 workloads are reported above. The following output comparisons come from the earlier separate CUDA validation against C++ BF16, not Python or human-annotated ground truth. It used an RTX 5090, debug server, eight threads, logging enabled, and the default very_high latency profile, without a TF32 override.

• —Five short recordings retained their turn counts, with mostly 10 ms boundary shifts.
• —The 97.6-second recording retained 31 turns; speaker activity differed over 0.174% of the timeline.
• —The 311-second recording changed from 161 to 152 turns; speaker activity differed over 1.174% of the timeline.
• —Repeated requests within each precision produced identical turns and confidence.

The percentages compare same-label speaker activity on a 10 ms grid; they are not diarization error rates (DER). Use BF16 when preserving the original model's output matters; Q8 trades output consistency for smaller weights and lower VRAM usage. The Q8 results above do not validate streaming or other backends.

License

The weights are governed by OpenMDW-1.1, not audio.cpp's code license. A downloaded copy is included in `LICENSE.html`. Retain the license and applicable notices of origin when redistributing this package. Original model by NVIDIA; GGUF conversion for audio.cpp.