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Local meeting recorder with transcription and speaker diarization for Obsidian

Project description

tapeback

Local meeting recorder for Linux. Records system audio + microphone via PipeWire/PulseAudio, transcribes with Whisper, identifies speakers, saves Markdown to your Obsidian vault. Everything runs on your machine — no cloud, no bots, no API calls for transcription.

Works with any video call platform: Google Meet, Zoom, Teams, Telegram, Discord, Slack huddles.

Features

  • Platform-agnostic — captures OS-level audio, works with any app
  • Local transcription — faster-whisper, runs on CPU or CUDA GPU
  • Speaker diarization — pyannote identifies who said what
  • Stereo channel separation — your mic (left) vs. others (right) for accurate "You" attribution
  • Obsidian-native output — Markdown with YAML frontmatter, wikilinks to audio files
  • LLM summarization — optional summaries via Anthropic, OpenAI, Groq, Gemini, DeepSeek, OpenRouter, Qwen (with automatic provider fallback)
  • CLI-firsttapeback start, Ctrl+C to stop, done

Requirements

  • Linux (PipeWire or PulseAudio)
  • Python 3.13+
  • ffmpeg
  • parecord (usually comes with pulseaudio-utils or pipewire-pulse)
  • NVIDIA GPU (optional, for faster transcription and diarization)

Installation

1. System dependencies

# Arch / Manjaro
sudo pacman -S python uv ffmpeg pipewire-pulse

# Ubuntu / Debian
sudo apt install python3 pipx ffmpeg pulseaudio-utils
pipx ensurepath  # adds ~/.local/bin to PATH

# Fedora
sudo dnf install python3 pipx ffmpeg pipewire-pulseaudio
pipx ensurepath

2. Install tapeback

The base package records audio and transcribes locally. Optional extras add speaker diarization and LLM summaries:

Extra What it adds Size
(none) Recording + transcription ~150 MB
[llm] LLM summarization (Anthropic, OpenAI, Gemini, etc.) +50 MB
[diarize] Speaker diarization (pyannote + PyTorch) +2 GB
[llm,diarize] Everything +2 GB

With uv (recommended)

uv tool install tapeback                  # basic
uv tool install "tapeback[llm]"           # + summaries
uv tool install "tapeback[diarize]"       # + speaker diarization
uv tool install "tapeback[llm,diarize]"   # everything

With pipx

pipx install tapeback                     # basic
pipx install "tapeback[llm]"              # + summaries
pipx install "tapeback[diarize]"          # + speaker diarization
pipx install "tapeback[llm,diarize]"      # everything

Arch Linux (AUR)

yay -S tapeback                  # basic
yay -S tapeback-llm              # + summaries
yay -S tapeback-diarize          # + speaker diarization (~2 GB PyTorch)

Nix

nix run github:yastcher/tapeback              # basic
nix run github:yastcher/tapeback#llm          # + summaries
nix run github:yastcher/tapeback#diarize      # + speaker diarization
nix run github:yastcher/tapeback#full         # everything

From source (development)

git clone https://github.com/yastcher/tapeback
cd tapeback
uv sync --group dev    # all dependencies + dev tools

3. Uninstall

# Remove tapeback
uv tool uninstall tapeback    # if installed with uv
pipx uninstall tapeback       # if installed with pipx

# Remove cached ML models (~2-5 GB)
# ⚠ Skip if you have other HuggingFace projects
rm -rf ~/.cache/huggingface/

# Arch Linux
yay -R tapeback tapeback-diarize tapeback-llm

Quick start

1. Configure

# Required: set your Obsidian vault path
export TAPEBACK_VAULT_PATH=~/Documents/obsidian/vault

# Or create a .env file in the project root:
echo 'TAPEBACK_VAULT_PATH=~/Documents/obsidian/vault' > .env

2. Record a meeting

# Start recording (blocks, Ctrl+C to stop and transcribe)
tapeback start

# Optionally give the session a name
tapeback start "weekly-standup"

3. Check your vault

The recording is saved as a Markdown note with audio attachment:

vault/
  meetings/2026-03-23_14-30-00.md
  attachments/audio/2026-03-23_14-30-00.wav

Process an existing recording

# Transcribe any audio file (mp3, m4a, ogg, wav)
tapeback process meeting.mp3

# With options
tapeback process call.wav --name "client-call" --no-diarize

Add summary to existing transcript

tapeback summarize vault/meetings/2026-03-23.md
tapeback summarize transcript.md --provider gemini

Configuration

All settings via environment variables (prefix TAPEBACK_) or .env file. Copy .env.example to .env and adjust:

cp .env.example .env

Core

Variable Default Description
TAPEBACK_VAULT_PATH (required) Path to Obsidian vault
TAPEBACK_MEETINGS_DIR meetings Subdirectory for meeting notes
TAPEBACK_ATTACHMENTS_DIR attachments/audio Subdirectory for audio files

Transcription

Variable Default Description
TAPEBACK_WHISPER_MODEL large-v3-turbo Whisper model (tiny, base, small, medium, large-v3-turbo)
TAPEBACK_LANGUAGE en Transcription language code
TAPEBACK_DEVICE cuda cuda or cpu
TAPEBACK_COMPUTE_TYPE float16 float16, int8, or float32
TAPEBACK_BEAM_SIZE 5 Whisper beam search width
TAPEBACK_PAUSE_THRESHOLD 1.0 Seconds — split segments on silence gaps >= this

Audio

Variable Default Description
TAPEBACK_MONITOR_SOURCE auto PulseAudio monitor source name
TAPEBACK_MIC_SOURCE auto PulseAudio mic source name
TAPEBACK_SAMPLE_RATE 48000 Recording sample rate

Speaker diarization

Variable Default Description
TAPEBACK_DIARIZE true Enable speaker diarization (requires tapeback[diarize])
TAPEBACK_HF_TOKEN (empty) HuggingFace token for pyannote models
TAPEBACK_MAX_SPEAKERS (auto) Maximum number of speakers

Speaker diarization requires the diarize extra (uv tool install "tapeback[diarize]") and a HuggingFace token with access to pyannote models:

  1. Create account at huggingface.co
  2. Accept license at pyannote/speaker-diarization-3.1
  3. Accept license at pyannote/segmentation-3.0
  4. Accept license at pyannote/speaker-diarization-community-1
  5. Create token at huggingface.co/settings/tokens
  6. Set TAPEBACK_HF_TOKEN=hf_your_token_here

Without a token, tapeback still works — it just skips diarization.

LLM summarization

Requires the llm extra: uv tool install "tapeback[llm]"

Variable Default Description
TAPEBACK_SUMMARIZE true Enable LLM summarization
TAPEBACK_LLM_PROVIDER anthropic Primary LLM provider
TAPEBACK_LLM_API_KEY (empty) API key (or use provider-specific env var)
TAPEBACK_LLM_MODEL (provider default) Override model name

Supported providers and their env vars:

Provider Env var Default model
anthropic ANTHROPIC_API_KEY claude-sonnet-4-20250514
openai OPENAI_API_KEY gpt-4o
groq GROQ_API_KEY llama-3.3-70b-versatile
gemini GEMINI_API_KEY gemini-2.5-flash
openrouter OPENROUTER_API_KEY google/gemini-2.5-flash:free
deepseek DEEPSEEK_API_KEY deepseek-chat
qwen DASHSCOPE_API_KEY qwen-turbo

If the primary provider fails, tapeback automatically tries the next available provider (any provider with an API key set).

CLI reference

tapeback --help                    Show help and quick start guide
tapeback start [NAME]              Start recording (Ctrl+C to stop)
tapeback stop                      Stop recording from another terminal
tapeback process <FILE> [--name N] Transcribe an existing audio file
tapeback summarize <FILE>          Add LLM summary to transcript
tapeback status                    Show recording status and settings

Common options

tapeback start --no-diarize        # Skip speaker identification
tapeback start --no-summarize      # Skip LLM summary
tapeback process file.mp3 --name "weekly-standup"
tapeback summarize file.md --provider gemini --model gemini-2.5-pro

Output format

---
date: 2026-03-23
time: "14:30"
duration: "01:23:45"
language: en
tags:
  - meeting
  - transcript
---

## Summary

Brief overview of the meeting.

### Action Items

- [ ] **You:** Send the report by Friday
- [ ] **Speaker 1:** Review the PR

### Key Decisions

- Use PostgreSQL instead of MongoDB

---

# Meeting 2026-03-23 14:30

![[attachments/audio/2026-03-23_14-30-00.wav]]

[00:00:01] **You:** Hello, let's start with the backend changes.

[00:01:23] **Speaker 1:** Sure, I have the slides ready.

[00:02:45] **Speaker 2:** Can we start with the backend changes?

Architecture

src/tapeback/
  cli.py          Click CLI — start, stop, process, summarize, status
  recorder.py     PulseAudio recording via parecord
  audio.py        ffmpeg audio processing (split channels, normalize, convert)
  transcriber.py  faster-whisper transcription
  diarizer.py     pyannote speaker diarization + spectral speaker merging
  formatter.py    Markdown generation (pure formatting, no I/O)
  vault.py        Obsidian vault file I/O
  summarizer.py   LLM summarization with multi-provider fallback
  models.py       Domain objects (Segment, Word, DiarizationSegment, Summary)
  settings.py     pydantic-settings configuration

Development

git clone https://github.com/yastcher/tapeback
cd tapeback
uv sync --group dev

uv run ruff check       # lint
uv run ruff format      # format
uv run ty check         # type check
uv run pytest           # test (coverage >= 85%)

Roadmap

Future development directions:

  • Custom diarization model — train a speaker embedding model optimized for meeting audio, replacing generic pyannote for better accuracy
  • Windows client — native Windows support via WASAPI loopback capture
  • Real-time transcription — live streaming transcription with partial results
  • Web dashboard — browser UI for reviewing and searching meeting history
  • Speaker profiles — learn and remember recurring speakers across meetings
  • Multi-language meetings — detect and handle language switches mid-meeting

License

Apache-2.0. See LICENSE.

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