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27 min readBrassTranscripts Team

Best Speaker Diarization Models: Complete Comparison [2025]

You need speaker diarization for your project. You've heard of Pyannote, NEMO, Kaldi, and WhisperX—but which one should you actually use? They all claim state-of-the-art performance, but accuracy, speed, and ease of use vary wildly.

Choosing the wrong speaker diarization model wastes days of setup time, produces poor results, or costs more than necessary. This guide compares the top models with real benchmarks, implementation difficulty, and production recommendations so you can make the right choice for your specific needs.

What you'll learn:

  • Comprehensive comparison of 6 speaker diarization models
  • Real accuracy benchmarks from published research
  • Speed and resource requirements for each model
  • Ease of implementation (with code examples)
  • When to use open-source vs. commercial services
  • Decision framework for choosing the right model

Models covered:

  1. Pyannote 3.1 (most popular open-source)
  2. NVIDIA NeMo (enterprise-grade performance)
  3. WhisperX (Whisper integration)
  4. Kaldi (traditional academic standard)
  5. SpeechBrain (research-focused toolkit)
  6. Commercial services (BrassTranscripts, AssemblyAI, Deepgram)

Who this is for: Developers, ML engineers, researchers, and technical decision-makers evaluating speaker diarization solutions.

Quick Navigation

Speaker Diarization Models: Quick Comparison

Here's a high-level comparison to help you quickly understand the landscape. Detailed analysis and benchmarks follow in later sections.

Model Accuracy Speed Ease of Use Best For
Pyannote 3.1 ⭐⭐⭐⭐⭐ ⭐⭐⭐ ⭐⭐⭐ Most developers
NVIDIA NeMo ⭐⭐⭐⭐⭐ ⭐⭐⭐⭐ ⭐⭐ Enterprise scale
WhisperX ⭐⭐⭐⭐ ⭐⭐⭐⭐ ⭐⭐⭐⭐ Whisper users
Kaldi ⭐⭐⭐ ⭐⭐ Academic research
SpeechBrain ⭐⭐⭐⭐ ⭐⭐⭐ ⭐⭐ Researchers
BrassTranscripts ⭐⭐⭐⭐⭐ ⭐⭐⭐⭐⭐ ⭐⭐⭐⭐⭐ Production use

Key takeaway: Pyannote 3.1 offers the best balance of accuracy and accessibility for most open-source needs. NVIDIA NeMo excels at production scale with NVIDIA GPUs. WhisperX is ideal if you need both transcription and speaker diarization. Commercial services like BrassTranscripts provide the highest accuracy with zero setup time.

Detailed comparison with specific metrics, code examples, and use cases follows.

How Speaker Diarization Models Work

Understanding the technical approaches helps you evaluate tradeoffs between models.

Traditional Approach (Kaldi-Style)

The traditional speaker diarization pipeline uses established signal processing techniques:

How it works:

  1. Feature extraction: Extract i-vectors or x-vectors (speaker embeddings) from audio segments
  2. Clustering: Group similar embeddings using algorithms like Probabilistic Linear Discriminant Analysis (PLDA) or Agglomerative Hierarchical Clustering (AHC)
  3. Segmentation: Determine speaker change points through statistical analysis

Advantages:

  • Interpretable pipeline (understand each step)
  • Established research foundation
  • Works on CPU without GPUs
  • Lower memory requirements

Limitations:

  • Manual parameter tuning required
  • Lower accuracy than deep learning approaches
  • Struggles with overlapping speech
  • Time-consuming setup

Representative model: Kaldi

Deep Learning Approach (Pyannote, NeMo)

Modern speaker diarization uses end-to-end neural networks trained on large datasets:

How it works:

  1. Neural embeddings: Deep learning models extract speaker representations
  2. Segmentation network: Neural network detects speaker changes and overlaps
  3. End-to-end training: All components trained together for optimal performance

Advantages:

  • Higher accuracy on challenging audio
  • Handles overlapping speech
  • Learns patterns automatically from data
  • Continuous improvement through model updates

Limitations:

  • Requires GPU for practical speed
  • Black box (less interpretable)
  • Higher memory requirements
  • Needs HuggingFace or model authentication

Representative models: Pyannote 3.1, NVIDIA NeMo

Hybrid Approach (WhisperX)

WhisperX combines separate transcription and diarization models for an integrated solution:

How it works:

  1. Whisper transcription: Converts speech to text with word-level timestamps
  2. Pyannote diarization: Identifies speaker segments
  3. Alignment: Assigns speaker labels to transcribed words

Advantages:

  • Single solution for transcription + diarization
  • Leverages best-in-class models for each task
  • Word-level speaker attribution

Limitations:

  • Dependent on both models working well
  • More resource-intensive (runs two models)
  • Complexity of integration

Representative model: WhisperX

Understanding Key Metrics

When evaluating speaker diarization models, these metrics matter:

Diarization Error Rate (DER):

  • Primary accuracy metric
  • Measures percentage of time speakers are incorrectly identified
  • Lower is better (e.g., 10% DER = 90% accuracy)
  • Includes missed speech, false alarms, and speaker confusion

Jaccard Error Rate (JER):

  • Alternative accuracy metric
  • Focuses on speaker assignment accuracy
  • Used in some academic benchmarks

Real-Time Factor (RTF):

  • Speed metric comparing processing time to audio length
  • RTF of 0.5 = 30 minutes to process 1 hour of audio
  • Lower is faster (e.g., 0.2 RTF is very fast)

Resource Requirements:

  • GPU memory (VRAM)
  • System RAM
  • CPU vs GPU capability

Understanding these fundamentals helps you interpret the model comparisons that follow.

Pyannote.audio is the most widely adopted open-source speaker diarization solution, developed by Hervé Bredin at CNRS (French National Centre for Scientific Research).

Architecture & Technology

Pyannote 3.1 uses state-of-the-art deep learning architecture:

Core components:

  • PyanNet segmentation model: Neural network for voice activity detection and speaker change detection
  • WeSpeaker embeddings: Deep speaker embeddings for speaker identification
  • Powerset multi-class segmentation: Handles overlapping speech where multiple people talk simultaneously
  • Overlapped speech detection: Dedicated model for detecting simultaneous speakers

Training data: Trained on diverse multi-speaker conversational datasets for robust performance across domains.

Accuracy Performance

Pyannote 3.1 achieves competitive accuracy on standard benchmarks:

Published benchmark results (from Pyannote research papers):

  • AMI corpus (meeting data): DER ~19% on test set
  • DIHARD III (diverse difficult audio): DER ~27% on full evaluation set
  • VoxConverse (YouTube videos): DER ~11% on test set

Real-world performance (based on typical use cases):

  • Clear audio, 2-3 speakers: Professional-grade accuracy
  • Moderate audio quality, 4-6 speakers: Good performance with occasional errors
  • Challenging audio, 7+ speakers: Accuracy degrades, manual review recommended

Accuracy varies significantly based on:

  • Audio quality (background noise, microphone quality)
  • Number of speakers
  • Amount of overlapping speech
  • Speaker similarity (similar voices are harder to distinguish)

Speed & Resource Requirements

Processing speed (approximate, varies by hardware):

  • GPU (NVIDIA RTX 3090): Faster than real-time (1 hour audio in 18-30 minutes)
  • CPU (modern multi-core): Slower than real-time (1 hour audio in 2-3 hours)

Resource requirements:

  • RAM: 4-8GB minimum
  • GPU VRAM: 2-4GB (if using GPU)
  • Storage: ~1GB for model weights

GPU dramatically improves processing speed—recommended for production use.

Ease of Implementation

Pyannote 3.1 is relatively straightforward to implement but requires HuggingFace authentication.

Basic implementation:

from pyannote.audio import Pipeline

# Load pre-trained pipeline (requires HuggingFace token)
pipeline = Pipeline.from_pretrained(
    "pyannote/speaker-diarization-3.1",
    use_auth_token="YOUR_HUGGINGFACE_TOKEN"
)

# Apply diarization to audio file
diarization = pipeline("audio.wav")

# Print results
for turn, _, speaker in diarization.itertracks(yield_label=True):
    print(f"{speaker}: {turn.start:.1f}s - {turn.end:.1f}s")

Setup steps:

  1. Create free HuggingFace account
  2. Accept model license agreements (speaker-diarization-3.1 and segmentation-3.0)
  3. Generate access token
  4. Install pyannote.audio: pip install pyannote.audio

Implementation difficulty: Medium (requires Python knowledge, HuggingFace setup)

Pros and Cons

Advantages:

  • ✅ State-of-the-art accuracy among open-source models
  • ✅ Actively maintained with regular updates
  • ✅ Excellent documentation and community support
  • ✅ Free and open-source (MIT license)
  • ✅ Large user base (most popular solution)
  • ✅ Handles overlapping speech reasonably well

Limitations:

  • ❌ Requires HuggingFace authentication (extra setup step)
  • ❌ Slow on CPU (GPU strongly recommended)
  • ❌ Speaker labels may switch mid-recording (known issue)
  • ❌ Memory intensive for very long files (>2 hours)
  • ❌ No real-time streaming support

Best Use Cases

Pyannote 3.1 excels for:

Research projects: Well-documented, reproducible, widely cited in academic papers

Prototype development: Quick to test speaker diarization capabilities

Medium-scale production: Works well with GPU infrastructure

Learning speaker diarization: Best starting point for understanding the technology

Resources

NVIDIA NeMo: Enterprise-Grade Speaker Diarization

NVIDIA NeMo is an enterprise-focused toolkit for conversational AI, including production-grade speaker diarization.

Architecture & Technology

NeMo uses NVIDIA's research advances optimized for their hardware:

Core components:

  • TitaNet speaker embeddings: NVIDIA's proprietary speaker embedding model
  • Multi-scale Diarization Decoder (MSDD): Advanced neural clustering for speaker assignment
  • Neural clustering: Learns optimal speaker grouping from data
  • GPU optimization: Specifically optimized for NVIDIA CUDA hardware

Training data: Trained on large-scale internal datasets at NVIDIA for robust commercial performance.

Accuracy Performance

NeMo achieves competitive accuracy with optimization for production scenarios:

Performance characteristics:

  • Competitive with Pyannote 3.1 on standard benchmarks
  • Often performs better on long-form audio (>1 hour)
  • Strong overlapping speech handling
  • Consistent performance across audio qualities

Accuracy varies by: Audio quality, speaker count, and deployment configuration. Specific DER numbers depend on benchmark dataset and evaluation protocol.

Speed & Resource Requirements

NeMo's main advantage is processing speed on NVIDIA hardware:

Processing speed:

  • NVIDIA GPU (A100, RTX 4090): Very fast, well under real-time
  • CPU: Not recommended (much slower, loses main advantage)

Resource requirements:

  • Requires NVIDIA GPU with CUDA support (not optional for practical use)
  • RAM: 8-16GB recommended
  • GPU VRAM: 4-8GB depending on model configuration
  • Storage: Several GB for model weights

Ease of Implementation

NeMo has a steeper learning curve than Pyannote:

Basic implementation:

import nemo.collections.asr as nemo_asr

# Load pre-trained model
msdd_model = nemo_asr.models.ClusteringDiarizer.from_pretrained(
    "diar_msdd_telephonic"
)

# Apply diarization
diarization = msdd_model.diarize("audio.wav")

Setup steps:

  1. Install NVIDIA CUDA toolkit
  2. Install PyTorch with CUDA support
  3. Install NeMo: pip install nemo_toolkit[asr]
  4. Configure GPU environment

Implementation difficulty: Hard (complex dependencies, NVIDIA-specific requirements)

Pros and Cons

Advantages:

  • ✅ Excellent accuracy on production workloads
  • ✅ Very fast on NVIDIA GPUs
  • ✅ Production-ready with enterprise support available
  • ✅ Actively maintained by NVIDIA
  • ✅ Scales well for high-volume processing

Limitations:

  • Requires NVIDIA GPU (not usable without compatible hardware)
  • ❌ Complex installation and environment setup
  • ❌ Steeper learning curve than alternatives
  • ❌ Smaller community compared to Pyannote
  • ❌ More components to configure correctly

Best Use Cases

NVIDIA NeMo excels for:

Large-scale production deployments: Optimized for processing thousands of hours

Real-time or near-real-time processing: Speed advantage critical

NVIDIA GPU infrastructure: Already using NVIDIA hardware

Enterprise projects: Professional support available from NVIDIA

Resources

WhisperX: Whisper + Pyannote Integration

WhisperX combines OpenAI Whisper (transcription) with Pyannote (speaker diarization) for an all-in-one solution.

Architecture & Technology

WhisperX integrates two separate models into a cohesive pipeline:

Components:

  • OpenAI Whisper: Speech-to-text transcription (large-v2 or large-v3 models)
  • Pyannote 3.1: Speaker diarization (same model as standalone)
  • Word-level alignment: Improved timestamp accuracy for speaker assignment
  • Speaker-word mapping: Assigns speaker labels to individual words

Accuracy Performance

WhisperX accuracy depends on both underlying models:

Transcription accuracy: Professional-grade (inherits Whisper's transcription quality)

Diarization accuracy: Good (uses Pyannote 3.1, slightly lower than standalone due to integration complexity)

Combined performance: Provides both transcription and speaker labels, though integration introduces some accuracy tradeoffs compared to using Pyannote alone for diarization.

Speed & Resource Requirements

WhisperX runs both models sequentially, affecting speed:

Processing speed:

  • Comparable to running Whisper and Pyannote separately
  • GPU: Faster than real-time for combined transcription + diarization
  • CPU: Slower than real-time (1 hour audio takes 2-4 hours)

Resource requirements:

  • Higher than standalone diarization (runs two models)
  • RAM: 8-16GB recommended
  • GPU VRAM: 4-8GB for optimal performance

Ease of Implementation

WhisperX is easier than manually integrating Whisper and Pyannote:

Basic implementation:

import whisperx

# Load audio
audio = whisperx.load_audio("audio.wav")

# Transcribe with Whisper
model = whisperx.load_model("large-v2", device="cuda")
result = model.transcribe(audio)

# Align timestamps
model_a, metadata = whisperx.load_align_model(
    language_code=result["language"],
    device="cuda"
)
result = whisperx.align(result["segments"], model_a, metadata, audio, device="cuda")

# Speaker diarization
diarize_model = whisperx.DiarizationPipeline(
    use_auth_token="YOUR_HUGGINGFACE_TOKEN",
    device="cuda"
)
diarize_segments = diarize_model(audio)

# Assign speakers to words
result = whisperx.assign_word_speakers(diarize_segments, result)

# Output
for segment in result["segments"]:
    speaker = segment.get("speaker", "UNKNOWN")
    text = segment["text"]
    print(f"{speaker}: {text}")

Setup steps:

  1. Install WhisperX: pip install whisperx
  2. Install ffmpeg separately
  3. HuggingFace authentication (same as Pyannote)

Implementation difficulty: Medium (simpler than separate integration, requires HuggingFace setup)

Pros and Cons

Advantages:

  • ✅ Combined transcription + speaker diarization in one pipeline
  • ✅ Word-level speaker assignment (not just segment-level)
  • ✅ Good documentation and active community
  • ✅ Improved timestamp alignment compared to basic Whisper
  • ✅ All-in-one solution for complete workflow

Limitations:

  • ❌ Diarization accuracy slightly lower than standalone Pyannote
  • ❌ Runs two heavy models (more resource-intensive)
  • ❌ Complexity of managing combined system
  • ❌ Debugging issues requires understanding both Whisper and Pyannote

Best Use Cases

WhisperX excels when you need:

Transcription + diarization together: One workflow instead of two separate steps

Whisper users wanting speaker labels: Natural extension if already using Whisper

Word-level speaker information: Needed for detailed analysis or editing

Complete pipeline: Prefer integrated solution over managing separate tools

Resources

Other Speaker Diarization Models Worth Considering

Beyond the top three, several other models serve specific use cases.

Kaldi Speaker Diarization

Kaldi represents the traditional academic approach to speaker diarization.

Technology:

  • i-vector and x-vector speaker embeddings
  • Traditional clustering algorithms (AHC, spectral clustering)
  • Established signal processing pipeline

Accuracy: Lower than modern deep learning approaches

Speed: Slow (CPU-focused, older algorithms)

Ease of use: Very difficult (complex installation, extensive configuration)

Best for:

  • Academic research requiring traditional baseline
  • Reproducibility with published papers
  • Understanding classical approaches

Verdict: Unless you specifically need Kaldi for academic reasons, use Pyannote or NeMo instead. Modern deep learning models outperform Kaldi significantly.

SpeechBrain Speaker Diarization

SpeechBrain is a PyTorch-based toolkit for speech processing including speaker diarization.

Technology:

  • ECAPA-TDNN speaker embeddings (modern approach)
  • PyTorch-based (easier customization than Kaldi)
  • Research-oriented toolkit

Accuracy: Competitive with Pyannote on some benchmarks

Speed: Moderate (similar to Pyannote)

Ease of use: Medium (requires understanding toolkit structure)

Best for:

  • Research projects needing customization
  • PyTorch users wanting full control
  • Building custom speaker diarization models

Verdict: Good choice for researchers, but Pyannote is more polished for production use. Consider SpeechBrain if you need to modify the speaker diarization pipeline significantly.

Comparison of Alternative Models

Model Accuracy Speed Setup Difficulty Best Use Case
Kaldi ⭐⭐⭐ ⭐⭐ Very Hard Academic baselines
SpeechBrain ⭐⭐⭐⭐ ⭐⭐⭐ Hard Research customization
Pyannote 3.1 ⭐⭐⭐⭐⭐ ⭐⭐⭐ Medium General production
NVIDIA NeMo ⭐⭐⭐⭐⭐ ⭐⭐⭐⭐ Hard Enterprise scale

For most developers and projects, Pyannote 3.1 or NVIDIA NeMo remain the recommended choices unless you have specific research needs requiring Kaldi or SpeechBrain.

Professional Speaker Diarization Services

Commercial services eliminate setup time and provide optimized accuracy, trading cost for convenience.

When to Use Commercial vs. Open-Source

Choose commercial services when:

  • You need results immediately (no 2-4 hour setup)
  • Highest accuracy is critical for your use case
  • You don't have technical team to manage open-source
  • Production reliability and SLA matter
  • Your time is more valuable than per-minute cost

Choose open-source when:

  • Budget is primary constraint (free software)
  • You have GPU infrastructure available
  • You enjoy technical implementation
  • You need full customization of the pipeline
  • Processing large volumes (setup cost amortized)

BrassTranscripts

BrassTranscripts uses optimized speaker diarization with WhisperX large-v3 and proprietary improvements.

Accuracy: Professional-grade speaker diarization with optimized models

Speed: Typically faster than real-time (5-10 minutes for 1 hour audio)

Ease of use: Zero setup—web interface, drag-and-drop upload

Pricing: $0.15/minute for audio over 15 minutes, $2.25 flat for 0-15 minutes

Best for:

  • Production use cases requiring reliability
  • Teams without ML expertise
  • When time-to-result matters most
  • Users wanting guaranteed accuracy

Try it: Free trial available

AssemblyAI

AssemblyAI provides developer-focused API for transcription and speaker diarization.

Accuracy: Good speaker diarization performance

Speed: Fast cloud processing

Ease of use: API-first (requires developer integration)

Pricing: Varies by volume and features (check current pricing)

Best for:

  • Developers building integrations
  • Applications needing API access
  • Real-time or batch processing

Deepgram

Deepgram specializes in real-time speech recognition with speaker diarization.

Accuracy: Competitive with proprietary Nova-2 model

Speed: Real-time and streaming capabilities

Ease of use: API-based (developer-focused)

Pricing: Varies by volume and features (check current pricing)

Best for:

  • Real-time or streaming audio
  • Live transcription applications
  • Low-latency requirements

Commercial vs. Open-Source Comparison

Aspect Open-Source Commercial Services
Setup time 2-4 hours 0 minutes
Accuracy Professional-grade Professional-grade with optimization
Speed Varies by hardware Consistently fast
Cost Free (compute costs) Per-minute pricing
Support Community forums Professional support
Reliability Self-managed SLA guarantees
Customization Full control Limited
Offline use Yes No (cloud-based)

Recommendation: For production use cases, commercial services typically provide better ROI when accounting for engineer time, infrastructure costs, and reliability requirements. Open-source excels for research, learning, and high-volume processing where setup time is amortized.

Speaker Diarization Model Benchmarks

Understanding real-world performance helps you set expectations and choose appropriately.

Benchmark Methodology Note

Speaker diarization accuracy varies significantly based on:

  • Audio quality (background noise, microphone type)
  • Number of speakers
  • Amount of overlapping speech
  • Speaker voice similarity
  • Recording duration
  • Acoustic environment

Important: Benchmark results from controlled datasets don't always reflect real-world performance. Use published benchmarks as general guidance, not absolute predictions.

Published Benchmark Results

AMI Meeting Corpus (common academic benchmark):

  • Pyannote 3.1: DER ~19% (published in Pyannote papers)
  • NVIDIA NeMo: Competitive performance (specific numbers vary by configuration)
  • Kaldi baseline: DER ~25-30% (traditional approaches)

DIHARD III (difficult evaluation set):

  • Pyannote 3.1: DER ~27% (published results)
  • Top systems: DER 20-30% range
  • Baseline systems: DER 35-45%

VoxConverse (YouTube videos):

  • Pyannote 3.1: DER ~11% (published results)
  • Easier dataset with clearer audio

Interpretation: Lower DER indicates better accuracy. DER of 10-15% is excellent, 15-25% is good, above 25% indicates challenging audio or limitations.

Real-World Performance Expectations

Based on typical use cases with clear to moderate audio quality:

2-3 speakers, clear audio:

  • Modern models (Pyannote, NeMo): Professional-grade accuracy
  • Occasional errors on speaker overlap or very similar voices
  • Generally suitable for production use

4-6 speakers, moderate audio:

  • Modern models: Good performance with some errors
  • Manual review recommended for critical applications
  • Accuracy decreases with speaker count

7+ speakers, challenging audio:

  • All models struggle significantly
  • Accuracy degrades substantially
  • Extensive manual correction likely needed
  • Consider professional human transcription

Speed Benchmarks

Processing speed for 1 hour of audio (approximate, varies by hardware):

GPU Processing (NVIDIA RTX 3090 or equivalent):

  • NVIDIA NeMo: ~15-20 minutes (very fast)
  • Pyannote 3.1: ~20-30 minutes (fast)
  • WhisperX: ~25-35 minutes (moderate, runs two models)
  • Kaldi: ~40-60 minutes (slower traditional approach)

CPU Processing (modern multi-core processor):

  • NVIDIA NeMo: Not recommended (designed for GPU)
  • Pyannote 3.1: ~2-3 hours (slow but usable)
  • WhisperX: ~3-4 hours (very slow)
  • Kaldi: ~2-4 hours (CPU-focused but still slow)

Commercial services:

  • BrassTranscripts: ~5-10 minutes (cloud infrastructure)
  • AssemblyAI / Deepgram: Similar cloud speeds

Key insight: GPU dramatically improves open-source model speed. If you're processing audio regularly, GPU is essentially required for practical turnaround times.

Resource Requirements Comparison

Memory usage for 1 hour audio file:

Model RAM (CPU) VRAM (GPU) Storage (models)
Pyannote 3.1 4-8GB 2-4GB ~1GB
NVIDIA NeMo 8-16GB 4-8GB ~3GB
WhisperX 8-16GB 4-8GB ~4GB
Kaldi 4-6GB N/A ~2GB

Recommendation: For open-source models, plan for at least 16GB system RAM and 8GB GPU VRAM for comfortable processing of typical audio files.

Which Speaker Diarization Model Should You Use?

Decision framework based on your specific requirements and constraints.

Choose Pyannote 3.1 If:

You match these criteria:

  • ✅ Need proven open-source solution
  • ✅ Have GPU available (or willing to tolerate CPU slowness)
  • ✅ Can spend 1-2 hours on initial setup
  • ✅ Community support is sufficient (no enterprise SLA needed)
  • ✅ Budget is primary constraint (free software)
  • ✅ Want most popular solution (large user base)

Example scenarios:

  • Research project requiring reproducibility
  • Startup building initial prototype
  • Developer learning speaker diarization
  • Medium-scale production with GPU infrastructure

Get started: Pyannote GitHub

Choose NVIDIA NeMo If:

You match these criteria:

  • ✅ Have NVIDIA GPU infrastructure (A100, RTX 4090, etc.)
  • ✅ Need production-grade performance at scale
  • ✅ Speed is critical (processing thousands of hours)
  • ✅ Want enterprise support option from NVIDIA
  • ✅ Have technical team comfortable with complex setup
  • ✅ Processing large volumes regularly

Example scenarios:

  • Large-scale production deployment
  • Call center analytics (thousands of calls daily)
  • Enterprise with existing NVIDIA infrastructure
  • Projects requiring fastest possible processing

Get started: NVIDIA NeMo GitHub

Choose WhisperX If:

You match these criteria:

  • ✅ Need transcription AND speaker diarization together
  • ✅ Already using Whisper for transcription
  • ✅ Want word-level speaker attribution
  • ✅ Prefer integrated solution over separate tools
  • ✅ Willing to trade some accuracy for convenience

Example scenarios:

  • Podcast transcription with speaker labels
  • Interview transcripts with speaker names
  • Content creation workflows
  • Applications requiring word-level speaker data

Get started: WhisperX GitHub or our Whisper Speaker Diarization Guide

Choose BrassTranscripts If:

You match these criteria:

  • ✅ Need results immediately (no setup time)
  • ✅ Highest accuracy is critical
  • ✅ No technical team or GPU infrastructure
  • ✅ Production reliability and SLA matter
  • ✅ Time is more valuable than per-minute cost
  • ✅ Want professional support when needed

Example scenarios:

  • Business meetings transcription
  • Legal proceedings documentation
  • Medical consultations
  • Content teams without developers
  • When accuracy matters more than cost

Get started: Try BrassTranscripts free

Choose Kaldi/SpeechBrain If:

You match these criteria:

  • ✅ Academic research project
  • ✅ Need to reproduce published baselines
  • ✅ Building custom speaker diarization model
  • ✅ Require full pipeline customization
  • ✅ Have expertise in speech processing

Example scenarios:

  • PhD research in speaker diarization
  • Publishing academic papers
  • Building novel diarization approaches
  • Teaching/learning speech processing fundamentals

Get started: Kaldi or SpeechBrain

Decision Tree

Simple flowchart for quick decisions:

1. Do you need production reliability with SLA?

  • Yes → BrassTranscripts or commercial service
  • No → Continue to #2

2. Do you have technical team?

  • No → BrassTranscripts (easiest option)
  • Yes → Continue to #3

3. Do you have NVIDIA GPU infrastructure?

  • Yes → NVIDIA NeMo (fastest open-source)
  • No → Continue to #4

4. Do you need transcription + diarization?

  • Yes → WhisperX (integrated solution)
  • No → Pyannote 3.1 (best general-purpose open-source)

5. Academic research only?

  • Yes → Consider Kaldi or SpeechBrain (established baselines)
  • No → Pyannote 3.1

Getting Started with Speaker Diarization Models

Quick start instructions for each major model.

Pyannote 3.1 Quick Start

Prerequisites: Python 3.8-3.11, HuggingFace account

# Install
pip install pyannote.audio

# Create HuggingFace account and generate token at:
# https://huggingface.co/settings/tokens

# Accept license agreements at:
# https://huggingface.co/pyannote/speaker-diarization-3.1
# https://huggingface.co/pyannote/segmentation-3.0

Basic usage:

from pyannote.audio import Pipeline

pipeline = Pipeline.from_pretrained(
    "pyannote/speaker-diarization-3.1",
    use_auth_token="YOUR_TOKEN"
)

diarization = pipeline("audio.wav")

for turn, _, speaker in diarization.itertracks(yield_label=True):
    print(f"{speaker}: {turn.start:.1f}s - {turn.end:.1f}s")

Complete tutorial: Pyannote documentation

NVIDIA NeMo Quick Start

Prerequisites: NVIDIA GPU with CUDA, Python 3.8-3.10

# Install NVIDIA CUDA toolkit first
# Download from: https://developer.nvidia.com/cuda-downloads

# Install PyTorch with CUDA
pip install torch torchvision torchaudio --index-url https://download.pytorch.org/whl/cu118

# Install NeMo
pip install nemo_toolkit[asr]

Basic usage:

import nemo.collections.asr as nemo_asr

msdd_model = nemo_asr.models.ClusteringDiarizer.from_pretrained(
    "diar_msdd_telephonic"
)

diarization = msdd_model.diarize("audio.wav")

Complete tutorial: NeMo documentation

WhisperX Quick Start

Prerequisites: Python 3.8-3.11, ffmpeg, HuggingFace account

# Install ffmpeg
# Mac: brew install ffmpeg
# Ubuntu: sudo apt install ffmpeg
# Windows: https://ffmpeg.org/download.html

# Install WhisperX
pip install whisperx

Basic usage: See complete code example in Whisper Speaker Diarization Guide

BrassTranscripts Quick Start

No installation required:

  1. Visit BrassTranscripts
  2. Upload audio/video file (drag-and-drop)
  3. Download transcript with speaker labels
  4. Free trial available—no credit card needed

Processing time: 5-10 minutes for typical recordings

Speaker Diarization Models FAQ

What is the best speaker diarization model?

For open-source: Pyannote 3.1 offers the best balance of accuracy and ease of use for most developers. NVIDIA NeMo is faster on NVIDIA GPUs for production scale.

For production: BrassTranscripts provides professional-grade accuracy with zero setup time and guaranteed reliability.

Best depends on your specific needs: accuracy requirements, technical expertise, infrastructure, and budget.

Is Pyannote better than NeMo?

Both are excellent with different strengths:

Pyannote 3.1:

  • Easier to set up and use
  • Larger community and more documentation
  • Works on any GPU (not NVIDIA-specific)
  • Best for general development and research

NVIDIA NeMo:

  • Faster processing on NVIDIA GPUs
  • Better optimized for production at scale
  • Enterprise support available
  • Requires NVIDIA hardware

Choose based on your infrastructure and scale requirements.

Can I use speaker diarization models for free?

Yes. Pyannote, NVIDIA NeMo, WhisperX, Kaldi, and SpeechBrain are all open-source and free to use. You only pay for:

  • Compute resources (your GPU/CPU, electricity, or cloud costs)
  • Time spent on setup and maintenance

Commercial services (BrassTranscripts, AssemblyAI, Deepgram) charge per minute of audio but require no setup or infrastructure.

How accurate are speaker diarization models?

Accuracy varies significantly based on audio quality and speaker count:

With clear audio, 2-3 speakers:

  • Modern models: Professional-grade accuracy (DER 10-15%)
  • Suitable for production use
  • Occasional errors on overlaps or similar voices

With moderate audio, 4-6 speakers:

  • Modern models: Good accuracy (DER 15-25%)
  • Some manual correction may be needed
  • Performance degrades with speaker count

With challenging audio, 7+ speakers:

  • All models struggle (DER 25%+)
  • Extensive manual review required
  • Consider professional human transcription

Which speaker diarization model is fastest?

Fastest open-source: NVIDIA NeMo on NVIDIA GPUs (processes 1 hour in ~15 minutes)

Fastest overall: Commercial services like BrassTranscripts (5-10 minutes total including upload)

CPU processing: All open-source models are slow on CPU (2-4 hours for 1 hour audio). GPU is essentially required for practical speed.

Do I need a GPU for speaker diarization?

Recommended but not required:

  • With GPU: Processing is practical (20-30 min for 1 hour audio)
  • Without GPU: Processing is very slow (2-4 hours for 1 hour audio)

If you don't have GPU:

  • Consider commercial services (BrassTranscripts) for faster results
  • Use CPU for occasional processing where speed isn't critical
  • Cloud GPU rental (Google Colab, AWS) for batch processing

Can speaker diarization models run offline?

Yes, all open-source models (Pyannote, NeMo, WhisperX, Kaldi, SpeechBrain) can run completely offline once installed:

  1. Download model weights once (requires internet)
  2. Process audio files offline (no internet needed)
  3. All processing happens on your local machine

Commercial services (BrassTranscripts, AssemblyAI, Deepgram) require internet connection as they're cloud-based.

What is DER in speaker diarization?

Diarization Error Rate (DER) measures speaker diarization accuracy:

What it measures:

  • Percentage of time speakers are incorrectly identified
  • Includes missed speech, false alarms, and speaker confusion
  • Lower is better (10% DER = 90% accurate speaker labels)

Interpretation:

  • DER < 15%: Excellent performance
  • DER 15-25%: Good performance
  • DER > 25%: Needs improvement or challenging audio

DER is the standard metric for comparing speaker diarization systems in research and benchmarks.

Are newer speaker diarization models always better?

Generally yes, with caveats:

Newer models improve:

  • Pyannote 3.1 (2023) significantly outperforms Pyannote 2.1 (2021)
  • Active development incorporates latest research advances
  • Trained on larger, more diverse datasets

But context matters:

  • Newer isn't better if it requires unavailable hardware (NVIDIA GPU)
  • Complexity increases with newer models (setup may be harder)
  • Well-tuned older models can outperform poorly configured newer ones

Recommendation: Use latest stable versions (Pyannote 3.1, latest NeMo) for best results, but ensure your infrastructure supports them.

Can I combine multiple speaker diarization models?

Yes, ensemble approaches can improve accuracy:

Fusion techniques:

  • Run multiple models on same audio
  • Combine results through voting or confidence weighting
  • Can improve accuracy by 2-5% in some cases

Practical considerations:

  • Increases processing time (run multiple models)
  • More complex implementation
  • Diminishing returns (small accuracy improvement for significant complexity)

Recommendation: Single well-configured model (Pyannote 3.1 or NeMo) is usually sufficient. Only consider ensemble for critical applications where accuracy improvement justifies complexity.

Choosing Your Speaker Diarization Model

Speaker diarization technology has matured significantly, with multiple excellent options available.

Summary of Top Models

Best open-source general-purpose: Pyannote 3.1

  • Proven accuracy, active development, large community
  • Easiest open-source option to implement
  • Recommended starting point for most developers

Best for production scale: NVIDIA NeMo

  • Fastest processing on NVIDIA GPUs
  • Enterprise-grade performance and support
  • Ideal when you have NVIDIA infrastructure

Best for Whisper integration: WhisperX

  • Combined transcription + speaker diarization
  • Word-level speaker attribution
  • Natural choice if already using Whisper

Best for ease and accuracy: BrassTranscripts

  • Zero setup, professional-grade accuracy
  • Fastest time to results (5 minutes vs. 3+ hours)
  • Recommended for production use cases

Our Recommendations by Use Case

Research and development:

  • Start with Pyannote 3.1 (most documented, widely used)
  • Experiment with WhisperX if you need transcription too
  • Consider SpeechBrain for customization needs

Production deployment:

  • Use BrassTranscripts for fastest deployment and reliability
  • Consider NVIDIA NeMo if you have NVIDIA GPU infrastructure at scale
  • Pyannote 3.1 works for medium-scale production with GPU

Learning speaker diarization:

  • Begin with Pyannote 3.1 (excellent documentation)
  • Try multiple models to understand differences
  • Read research papers for each approach

Ready to Get Started?

Want the easiest path to speaker-labeled transcripts?

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Want to implement open-source?

Learn more about speaker diarization:


Questions about choosing the right speaker diarization model? The models discussed here are actively maintained with strong communities. For production use without technical overhead, try BrassTranscripts free - we handle the complexity so you can focus on your content.

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