Models & Comparisons

Speaker Diarization and Analysis

Last updated 2026-07-28

What's new

2026-07-28
  • OpenAI's new ChatGPT voice feature lets you control AI agents (AI programs that can do tasks for you) with your voice, making it easier to use AI anywhere, anytime.
  • Unlike old voice features that just type what you say (called dictation), this new tool lets you command multiple AI agents across all your devices (like your phone, tablet, or computer) to get things done instantly.
  • ChatGPT voice can check the status of your projects, give you updates, and even create new AI agents to handle tasks for you, like fixing errors or building websites.
  • This tool can greatly increase your productivity (getting more done in less time) by removing barriers between having an idea and seeing it become reality.
2026-07-25
  • You can now use a smart AI tool (Local AI) that runs completely free, offline, and privately on your own computer, with no data leaving your device.
  • Local AI uses "open weights" (free, downloadable AI models) that you can own and use without paying for access or worrying about data privacy.
  • The free AI models have improved significantly, with some like GLM 5.2 (a large, capable AI model) performing close to top paid models on many tasks.
  • Using Local AI gives you control and privacy, as you're not renting a service from a company that can change or restrict access, and your data stays on your machine.
2026-07-13
  • Open Claw (a tool for managing secure access to apps) now supports "trusted proxy auth mode," a feature that removes the need for tokens and device pairing, improving both security and user experience.
  • This mode uses an identity-aware proxy (a security approach that combines identity provider, policy engine, and reverse proxy) to control access, making it easier to secure internal apps.
  • The update allows configuration through both onboarding and TUI (text-based user interface), providing flexibility in setup.
  • The speaker highlighted the active Open Claw community, noting how contributors quickly addressed a bug they missed, demonstrating the power of open-source software (OSS).
2026-06-10
  • DeepMind has released Gemma 4, an open model with audio understanding capabilities that can run on edge devices (small, portable computing devices).
  • Gemini 3.1 flash life is a real-time, full duplex (two-way) sound-to-sound conversational model that also supports text and vision inputs.
  • Echo Script, built with Google AI Studio (a platform for creating AI-powered apps), uses Gemini 3 to analyze audio recordings, extracting details like speaker names, timestamps, languages, emotions, and summaries.
  • Gemini 3 can handle complex audio tasks, such as transcribing overlapping speech, switching between languages, and identifying speaker emotions.

Key points

What it is

  • Speaker diarization is a process that identifies who spoke when in an audio recording, assigning each speech turn to a specific speaker.
  • It goes beyond basic transcription (converting speech to text) by labeling who said what, providing context about the conversation.
  • This is important for richer conversation analysis, as knowing who said what can be as crucial as the words themselves.

How to use it

  • First, use voice activity detection (VAD) to identify when anyone is speaking, then segment those speech regions into turns.
  • Next, assign each turn to a specific speaker using a diarization pipeline, then merge that with your transcription from a speech-to-text model.
  • To build a voice profile, gather your past content, transcribe those files, and extract patterns across dimensions like word choice, pacing, and filler words.

Watch out for

  • Common errors include confusion (assigning a word to the wrong speaker) and segmentation mistakes.
  • Diarization accuracy can be affected by factors like distant microphones, speaker changes, cross talk, interruptions, and code switching (changing languages mid-conversation).
  • Performance varies completely depending on the use case, so always benchmark your diarization pipeline on data similar to your target use case.

Tools named

  • Pyannote (an open-source tool for speaker diarization), Whisper (a speech-to-text model), Claude (a tool for analyzing writing or speech examples).

Lesson 1: What is Speaker Diarization and Analysis and why it matters

Speaker diarization is the process of answering the question "who spoke when" in an audio recording. It automatically detects how many speakers are present and assigns each speech turn to a specific identity (for example, labeling one voice as "green" and another as "black" in a two-person conversation). This is different from plain transcription, which only captures what was said without tracking who said it. As one expert noted, "who said what is usually just as important as what was said."

The importance for AI development is twofold. First, it enables richer conversation analysis. Without speaker diarization, a transcript is just a block of text—you lose context about which speaker asked a question, which speaker responded, and how each person's state of mind might differ. Second, speaker diarization remains a difficult unsolved problem. Performance varies completely depending on the use case: a clean telephone conversation might achieve low error rates, but background noise (like traffic or horns) or overlapping speech can dramatically degrade accuracy. This means AI developers must carefully benchmark models for their specific environment rather than assuming a one-size-fits-all solution.

In practice, speaker diarization sits alongside related tasks like speaker identification (assigning a known name to each voice) and is often used as a preprocessing step for building voice agents or analyzing call center conversations. Because the problem is still active in research, state-of-the-art performance depends heavily on the quality of the input and the number of speakers involved.

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Lesson 2: How to use Speaker Diarization and Analysis: step-by-step

Speaker diarization answers "who spoke when" in a recorded conversation. It goes beyond basic transcription (converting speech to text) because raw transcription gives you words but no speaker labels. To get a speaker-attributed transcription, you first run voice activity detection (VAD), which tells you when anyone is speaking. Next, you segment those speech regions into turns, then assign each turn to a specific speaker using a diarization pipeline.

For a concrete example, take an audio file of two people chatting. Use an open-source tool like pyannote, which runs a diarization algorithm. It outputs something like "Speaker A from 0:00–0:12, Speaker B from 0:13–0:25." You then merge that with your transcription from a speech-to-text model (e.g., whisper) to label each word or phrase, producing a full speaker-attributed transcript. The process isn't perfect—common errors include confusion (assigning a word to the wrong speaker) and segmentation mistakes—so visualize the output against reference annotations to spot and fix mistakes.

To build a voice profile (a detailed document capturing your unique speaking style), start by gathering your past content—podcasts, videos, or chats you've recorded. Transcribe those files, then extract patterns across six dimensions, such as word choice, pacing, and filler words. Use a tool like Claude to analyze multiple writing or speech examples and generate a voice profile markdown file. This profile then becomes a reference for any AI tool you use, ensuring your outputs sound like you.

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Lesson 3: Best practices and pitfalls

Speaker diarization (identifying who spoke when) often fails because speech-to-text models don’t generalize well to multi-speaker recordings. Common pitfalls include distant microphones, speaker changes, cross talk, interruptions, and code switching (changing languages mid-conversation). Even when diarization and transcription work separately, assigning each word to the right speaker is surprisingly difficult. Most speech-to-text models aren’t designed for that alignment task.

To evaluate diarization accuracy, practitioners use diarization error rate (a metric that measures mistakes like confusing speakers). In a benchmark, a system showed just 3% diarization error on telephone speech, but performance varies completely by use case. The task remains unsolved because acoustic conditions and overlapping speech create persistent challenges.

Building a voice profile for an AI system requires extracting your personal speaking patterns—word choices, pacing, filler words. The best approach is to transcribe recordings of yourself, such as podcast appearances or videos, then analyze the tone across multiple dimensions. You can paste real writing samples (LinkedIn posts, emails, newsletters) into a tool that extracts voice patterns from those examples. Skipping this extraction yields less accurate outputs.

Best practices: start with clean, varied recordings of natural speech. Always benchmark your diarization pipeline on data similar to your target use case. For voice profiles, prioritize authentic self-recordings over generic answers. Avoid assuming any single tool works well across all environments—test specifically for your recording conditions.

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