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[](https://huggingface.co/devsy0117/ultra_diar_streaming_sortformer_5spk_v1)
[](https://huggingface.co/devsy0117/ultra_diar_streaming_sortformer_8spk_v1)
This repository records the **transition from a fixed 4-speaker cap to a configurable N-speaker Streaming Sortformer** (**N > 4**). The pattern is always the same: grow the speaker head with **SVD-based orthogonal initialization**, keep **base vs. new** weights in separate modules, and **fine-tune with split learning rates** so behavior on 2–4 speaker audio stays stable while the extra dimensions learn. You can target any **N** supported by your data and VRAM; we publish checkpoints for **N = 5** and **N = 8** as reference points.
**Released models (Hugging Face)**
- [ultra_diar_streaming_sortformer_5spk_v1](https://huggingface.co/devsy0117/ultra_diar_streaming_sortformer_5spk_v1) — **N = 5**
- [ultra_diar_streaming_sortformer_8spk_v1](https://huggingface.co/devsy0117/ultra_diar_streaming_sortformer_8spk_v1) — **N = 8**
---
## Table of Contents
1. [Background](#background)
2. [Architecture Overview](#architecture-overview)
3. [Extension Journey](#extension-journey)
- [Step 1: Output Layer Extension (4 → N)](#step-1-output-layer-extension-4--n)
- [Step 2: Split Learning Rate Training](#step-2-split-learning-rate-training)
- [Step 3: Scaling to Larger N (Example: 8 Speakers)](#step-3-scaling-to-larger-n-example-8-speakers)
4. [Benchmark](#benchmark)
5. [Synthetic Training Data](#synthetic-training-data)
- [Prerequisites](#synthesis-prerequisites)
- [How it differs from stock NeMo](#how-it-differs-from-stock-nemo)
- [Configuration](#synthesis-configuration)
- [CLI](#synthesis-cli)
- [Session length and speaker enforcement](#session-length-and-speaker-enforcement)
- [Outputs](#synthesis-outputs)
- [Example command](#synthesis-example)
6. [Training](#training)
7. [Requirements](#requirements)
---
## Background
NVIDIA's [diar_streaming_sortformer_4spk-v2.1](https://huggingface.co/nvidia/diar_streaming_sortformer_4spk-v2.1) is a streaming speaker diarization model based on the Sortformer architecture. It uses a **FastConformer encoder** (17 layers) followed by a **Transformer encoder** (18 layers) to produce per-frame speaker activity predictions. The final output layer is a single linear mapping from hidden states to **four** speaker probabilities.
The model supports real-time streaming diarization with a chunk-based speaker cache.
**Problem**: The public checkpoint is **hard-limited to four simultaneous speakers**. Scenes with more talkers are handled poorly once you only relabel data without widening the head.
**Goal**: Make **`max_num_of_spks = N` with N > 4** a first-class training target, without sacrificing the 2–4 speaker regime. This README walks through the mechanics; concrete runs in the repo use **N = 5** (smallest jump) and **N = 8** (larger jump from the same base).
---
## Architecture Overview
```
Audio Input
│
▼
┌─────────────────────────────┐
│ Preprocessor │
└─────────────────────────────┘
│
▼
┌──────────────────────────────┐
│ FastConformer Encoder │ ← 17 layers, d_model=512
│ (NEST Encoder) │ Subsampling factor: 8
└──────────────────────────────┘
│
▼
┌──────────────────────────────┐
│ Transformer Encoder │ ← 18 layers, d_model=192
└──────────────────────────────┘
│
▼
┌──────────────────────────────┐
│ SortformerModules │ ← Speaker Cache + Attention
│ (Streaming Speaker Cache) │
└──────────────────────────────┘
│
▼
┌──────────────────────────────┐
│ single_hidden_to_spks │ ← Linear(192, N_spk) ← KEY LAYER
└──────────────────────────────┘
│
▼
Per-frame speaker activity predictions [batch, time, N_spk]
```
**Stock:** one `single_hidden_to_spks` (`Linear(192, N_spk)`). **This repo (extended):** `single_hidden_to_spks_base` + `single_hidden_to_spks_new` for split LR (Step 1–2). **4 → N** speakers means **N − 4** new rows (or **N_new** from any **N_base**).
---
## Extension Journey
### Step 1: Output Layer Extension (4 → N)
**Script**: `scripts/extend_output_layer.py`
We treat the baseline as **N_base = 4** speakers and grow the matrix to **N = N_base + N_new**. The first shipped milestone is **N = 5** (**N_new = 1**); the **N = 8** model adds **N_new = 4** rows in one shot with the same procedure.
Random initialization for the new rows **destroys** accuracy on existing speakers. We instead use **SVD-based orthogonal initialization** so new logits start orthogonal to the subspace spanned by the original weights.
#### How it works
Let the existing weight matrix be `W ∈ ℝ^{N_base×H}` (H = 192):
```python
# SVD decomposition of existing weights
U, S, Vh = torch.linalg.svd(W, full_matrices=True)
# New speaker rows = right singular directions beyond the first N_base
# Vh[N_base], Vh[N_base+1], ... are orthogonal to the row space of W
new_row = Vh[N_base] # first new speaker; repeat indexing for additional speakers
# Normalize to match typical row norms
avg_norm = W.norm(dim=1).mean()
new_row = new_row * (avg_norm / new_row.norm())
```
Repeat for each new speaker index until the head reaches the target **N**.
The extended checkpoint is saved in **split** form so optimizers can treat base and new rows differently:
```
single_hidden_to_spks_base (N_base speakers) ← frozen / low LR
single_hidden_to_spks_new (N_new speakers) ← higher LR
```
---
### Step 2: Split Learning Rate Training
**Key insight**: A **single** learning rate on the whole expanded head tends to **erase** the old 2–4 speaker solution while the new dimensions still fit.
#### What we saw early on
- Synthetic **val_2spk–val_4spk** quality dropped under uniform LR.
- The network **over-used** the new capacity (e.g., predicting the full trained **N** on 3–4 speaker clips).
#### Mitigation: differential learning rates
| Component | Learning rate | Role |
|-----------|---------------|------|
| `single_hidden_to_spks_base` (speakers 1…N_base) | `1e-5` | Preserve the pretrained head |
| `single_hidden_to_spks_new` (added speakers) | `1e-4` | Faster adaptation on new dimensions |
| Rest of the model | `1e-5` | Standard fine-tuning |
Implemented via `setup_optimizer_param_groups` in `sortformer_diar_models.py`:
```python
def setup_optimizer_param_groups(self):
sm = self.sortformer_modules
n_base = getattr(sm, 'n_base_spks', 0)
new_lr = self._cfg.get('optim_new_lr', None)
if n_base > 0 and new_lr is not None and hasattr(sm, 'single_hidden_to_spks_new'):
new_params = list(sm.single_hidden_to_spks_new.parameters())
new_param_ids = {id(p) for p in new_params}
base_params = [p for p in self.parameters() if id(p) not in new_param_ids]
self._optimizer_param_groups = [
{"params": base_params},
{"params": new_params, "lr": new_lr},
]
```
---
### Step 3: Scaling to Larger N (Example: 8 Speakers)
The **N = 8** release uses the **same pipeline**: start from NVIDIA **4-spk**, extend the Sortformer head with orthogonal / split weights (**N_base = 4**, **N_new = 4**), then fine-tune with **~1e-5** on the bulk of parameters and **~1e-4** on `single_hidden_to_spks_new` on mixed synthetic + real meeting data.
- **Hugging Face**: [devsy0117/ultra_diar_streaming_sortformer_8spk_v1](https://huggingface.co/devsy0117/ultra_diar_streaming_sortformer_8spk_v1)
- **Weights**: `ultra_diar_streaming_sortformer_8spk_v1.nemo` (same artifact as the Hub upload)
Other **N**: keep `max_num_of_spks`, manifests, and `n_base_spks` consistent with your checkpoint; NeMo patches are listed under [Training](#training).
---
## Synthetic Training Data
**`scripts/sentence_level_multispeaker_simulator.py`** subclasses NeMo’s [`MultiSpeakerSimulator`](https://github.com/NVIDIA/NeMo/blob/main/nemo/collections/asr/data/data_simulation.py) (same idea as [`multispeaker_simulator.py`](https://github.com/NVIDIA/NeMo/blob/main/tools/speech_data_simulator/multispeaker_simulator.py)) and keeps NeMo’s `data_simulator.yaml` session pipeline. **`_build_sentence`** only is overridden: turns use **whole manifest utterances**, not word-aligned slices.
### Source Data
Single-speaker utterances come from the **[다화자 음성합성 데이터 (Multi-speaker Speech Synthesis Dataset)](https://www.aihub.or.kr/aihubdata/data/view.do?aihubDataSe=data&dataSetSn=542)** on [AI-Hub](https://www.aihub.or.kr) (NIA). It spans 3,400+ Korean speakers (10s–60s), ~10k hours.
| Split | Approx. #Utterances | Language |
|--------|---------------------|----------|
| `multispeaker_speech_synthesis_data/Training` | 8,666,803 | Korean |
| `multispeaker_speech_synthesis_data/Validation` | 1,225,244 | Korean |
Build a **NeMo-style JSON manifest** listing `audio_filepath`, `speaker` (or compatible id), and optionally `text`, `words`, `alignments` for labels. The simulator groups rows by speaker id to sample per turn.
### Synthesis prerequisites
1. **Install NeMo** (see [Requirements](#requirements)). The simulator imports `nemo` from your environment; it looks for `NeMo/tools/speech_data_simulator/conf/data_simulator.yaml` **only if** a sibling `NeMo/` directory exists next to this repo. If you use a **pip-only** install, pass `--config_file` pointing to that YAML (e.g. from a checkout or a copied file).
2. **System audio libraries**: `libsndfile1` and `ffmpeg` (listed under Requirements) are required for decoding/writing audio in practice.
3. Generation forces **CPU** (`CUDA_VISIBLE_DEVICES=""`) for stable runs without a working GPU stack.
### How it differs from stock NeMo
| Aspect | Stock `MultiSpeakerSimulator` | `SentenceLevelMultiSpeakerSimulator` |
|--------|------------------------------|--------------------------------------|
| Turn content | Word-aligned slices; reads audio in chunks up to `max_audio_read_sec` | One or more **entire** utterances per turn (mono, resampled to `sr`) |
| Turn length cap | Word-count target from `sentence_length_params` | **`max_sentences_per_turn`**: uniform **1…N** utterances per turn (CLI default **N = 3**). If unset in YAML and not overridden, falls back to negative binomial on **utterance count** (often too long—prefer explicit **N**) |
| Optional YAML | Same | `session_params.max_turn_duration_sec` caps samples per turn when set |
### Synthesis configuration
- **Base config**: `NeMo/tools/speech_data_simulator/conf/data_simulator.yaml` (or pass `--config_file`).
- Important YAML knobs (not all exposed on CLI):
- `session_config.{num_speakers,num_sessions,session_length}` — target speakers per session, session count, **nominal** duration in **seconds**.
- `session_params.{mean_silence,mean_overlap,...}` — global silence/overlap **means** (per-session values vary).
- `speaker_enforcement.enforce_num_speakers` — if `true`, NeMo may **continue past `session_length`** and **pad** the waveform until every speaker has spoken; real duration can exceed `session_length`. Set `enforce_num_speakers: false` in YAML if you need a hard cap at the cost of possibly missing speakers in a session.
- `sr`, `outputs.output_filename`, augmentors, background noise, etc. — unchanged from NeMo.
### Synthesis CLI
| Argument | Role |
|----------|------|
| `--manifest_filepath` | Input NeMo JSON manifest (single-speaker rows with speaker id). |
| `--output_dir` | Output directory for `.wav`, `.rttm`, `.json`, `params.yaml`, etc. |
| `--config_file` | Optional override YAML (defaults to NeMo’s `data_simulator.yaml`). |
| `--num_speakers` | Override `session_config.num_speakers`. |
| `--num_sessions` | Override `session_config.num_sessions`. |
| `--session_length` | Override nominal session length (**seconds**). |
| `--mean_silence` | Session mean silence ratio in **[0, 1)**. |
| `--mean_overlap` | Session mean overlap ratio in **[0, 1)**; invalid `mean_overlap_var` is clamped for stability. |
| `--max_sentences_per_turn` / `--max_sent` | Max utterances concatenated in one speaker turn; each run draws **uniformly from 1…N** (default **N = 3**). |
### Session length and speaker enforcement
`session_length` is a **target** timeline length in samples (`session_length × sr`). With **`enforce_num_speakers: true`** (NeMo default), the generator can **extend** the buffer so late speakers still get turns. For utterance-level simulation, combine **`--max_sent`** (small **N**) with YAML tuning (`enforce_num_speakers`, optional `max_turn_duration_sec`) if you need durations close to the nominal cap.
### Synthesis outputs
Per session index `i`: `multispeaker_session_i.wav`, `multispeaker_session_i.rttm`, `multispeaker_session_i.json` (and CTM if enabled), plus a copied **`params.yaml`** under `--output_dir`. Merge session manifests into NeMo diarization train/val JSON with your own tooling. Under `scripts/`, this repo includes `sentence_level_multispeaker_simulator.py` (synthesis) and `inference.py` (minimal HF Sortformer diarize example).
### Synthesis example
From the repository root (with `NeMo` installed and discoverable as above):
```bash
python scripts/sentence_level_multispeaker_simulator.py \
--manifest_filepath /path/to/manifest.json \
--output_dir /path/to/synthetic_run \
--num_speakers 8 \
--num_sessions 1000 \
--session_length 180 \
--mean_silence 0.10 \
--mean_overlap 0.05 \
--max_sent 3
```
Adjust paths, speaker count, session count, and overlap/silence means to match your experiment grid.
### Generated datasets (this project)
Synthetic grids for **2–8 speakers** used two mean-overlap settings: **`ov0.05`** (~5%) and **`ov0.15`** (~15%, harder). Both used comparable mean silence (~10%); overlap and silence are **session means**, so per-session values vary.
---
## Benchmark
Tables and evaluation protocol: **[`results/benchmark.md`](results/benchmark.md)**.
> **Note**: More speakers can shift speaker-count behavior on short or low-speaker clips; read `Spk_Count_Acc` next to DER. Model cards on Hugging Face have more context.
---
## Training
Upstream NeMo does **not** ship the split head / split LR below—patch your checkout (or use a fork that includes the same edits).
### NeMo Modifications
**`nemo/collections/asr/models/sortformer_diar_models.py`**
- Added `setup_optimizer_param_groups()` override for differential learning rates
**`nemo/collections/asr/modules/sortformer_modules.py`**
- Added `n_base_spks` parameter to enable split output layers (`single_hidden_to_spks_base` + `single_hidden_to_spks_new`)
### What we froze (ablation)
We compared three setups using NeMo’s `freeze_encoder` / `freeze_transformer_encoder` flags on the **Fast Conformer encoder** (`encoder.*`) and **Transformer encoder** (`transformer_encoder.*`):
| Setup | Conformer encoder | Transformer encoder |
|-------|-------------------|---------------------|
| Encoder frozen | frozen | trainable |
| Encoder + Transformer frozen | frozen | frozen |
| Full fine-tuning | trainable | trainable |
**Full fine-tuning** gave the smoothest, most reliable **training loss decrease**, so the released Ultra-Sortformer runs use **no freezing** (both stacks trainable), together with the split speaker head and split LR above.
### Training Configuration
Example keys (see also NeMo `examples/speaker_tasks/diarization/conf/neural_diarizer/streaming_sortformer_diarizer_4spk-v2.yaml`):
```yaml
model:
max_num_of_spks: 6 # Set to your target N
lr: 1e-5 # Base learning rate
# optim_new_lr: 1e-4 # Higher LR for single_hidden_to_spks_new (split-head fine-tuning)
sortformer_modules:
num_spks: ${model.max_num_of_spks}
# n_base_spks: 4 # Base speaker count when using split output layers
```
---
## Requirements
System packages (Debian/Ubuntu; use `sudo` if you are not root):
```bash
sudo apt-get update && sudo apt-get install -y libsndfile1 ffmpeg
```
Python (recommended: a fresh virtual environment). `nemo_toolkit[asr]` pulls in PyTorch and most ASR dependencies declared by NeMo; install a **CUDA-enabled PyTorch wheel first** from [pytorch.org](https://pytorch.org) if you train or run on GPU.
```bash
pip install Cython packaging
pip install "git+https://github.com/NVIDIA/NeMo.git@main#egg=nemo_toolkit[asr]"
```
**Note:** `pip` NeMo from NVIDIA does not include the [Training](#training) patches (`n_base_spks`, split head, optimizer groups).
**Optional extras:**
- **`pyannote.metrics`** — used for DER / benchmark-style evaluation in this project’s docs and scripts.
- **`librosa`** — only if your own preprocessing or tooling uses it.
- **Sibling `NeMo/` clone** — handy for `NeMo/examples/...` and default `data_simulator.yaml`; otherwise pass `--config_file` to the simulator.
---
## License
Apache License 2.0
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