| name | faster-whisper |
| description | Local speech-to-text using faster-whisper. 4-6x faster than OpenAI Whisper with identical accuracy; GPU acceleration enables ~20x realtime transcription. SRT/VTT/TTML/CSV subtitles, speaker diarization, URL/YouTube input, batch processing with ETA, transcript search, chapter detection, per-file language map. |
| version | 1.5.1 |
| author | ThePlasmak |
| homepage | https://github.com/ThePlasmak/faster-whisper |
| tags | ["audio","transcription","whisper","speech-to-text","ml","cuda","gpu","subtitles","diarization","podcast","chapters","search","csv","ttml","batch"] |
| platforms | ["linux","macos","wsl2"] |
| metadata | {"openclaw":{"emoji":"🗣️","requires":{"bins":["python3"],"optionalBins":["ffmpeg","yt-dlp"],"optionalPaths":["~/.cache/huggingface/token"]}}} |
Faster Whisper
Local speech-to-text using faster-whisper — a CTranslate2 reimplementation of OpenAI's Whisper that runs 4-6x faster with identical accuracy. With GPU acceleration, expect ~20x realtime transcription (a 10-minute audio file in ~30 seconds).
When to Use
Use this skill when you need to:
- Transcribe audio/video files — meetings, interviews, podcasts, lectures, YouTube videos
- Generate subtitles — SRT, VTT, ASS, LRC, or TTML broadcast-standard subtitles
- Identify speakers — diarization labels who said what (
--diarize)
- Transcribe from URLs — YouTube links and direct audio URLs (auto-downloads via yt-dlp)
- Transcribe podcast feeds —
--rss <feed-url> fetches and transcribes episodes
- Batch process files — glob patterns, directories, skip-existing support; ETA shown automatically
- Convert speech to text locally — no API costs, works offline (after model download)
- Translate to English — translate any language to English with
--translate
- Do multilingual transcription — supports 99+ languages with auto-detection
- Transcribe a batch of files in different languages —
--language-map assigns a different language per file
- Transcribe multilingual audio —
--multilingual for mixed-language audio
- Transcribe audio with specific terms — use
--initial-prompt for jargon-heavy content or any other terms to look out for
- Preprocess noisy audio (before transcription) —
--normalize and --denoise before transcription
- Stream output —
--stream shows segments as they're transcribed
- Clip time ranges —
--clip-timestamps to transcribe specific sections
- Search the transcript —
--search "term" finds all timestamps where a word/phrase appears
- Detect chapters —
--detect-chapters finds section breaks from silence gaps
- Export speaker audio —
--export-speakers DIR saves each speaker's turns as separate WAV files
- Spreadsheet output —
--format csv produces a properly-quoted CSV with timestamps
Trigger phrases:
"transcribe this audio", "convert speech to text", "what did they say", "make a transcript",
"audio to text", "subtitle this video", "who's speaking", "translate this audio", "translate to English",
"find where X is mentioned", "search transcript for", "when did they say", "at what timestamp",
"add chapters", "detect chapters", "find breaks in the audio", "table of contents for this recording",
"TTML subtitles", "DFXP subtitles", "broadcast format subtitles", "Netflix format",
"ASS subtitles", "aegisub format", "advanced substation alpha", "mpv subtitles",
"LRC subtitles", "timed lyrics", "karaoke subtitles", "music player lyrics",
"HTML transcript", "confidence-colored transcript", "color-coded transcript",
"separate audio per speaker", "export speaker audio", "split by speaker",
"transcript as CSV", "spreadsheet output", "transcribe podcast", "podcast RSS feed",
"different languages in batch", "per-file language",
"transcribe in multiple formats", "srt and txt at the same time", "output both srt and text",
"remove filler words", "clean up ums and uhs", "strip hesitation sounds", "remove you know and I mean",
"transcribe left channel", "transcribe right channel", "stereo channel", "left track only",
"wrap subtitle lines", "character limit per line", "max chars per subtitle",
"detect paragraphs", "paragraph breaks", "group into paragraphs", "add paragraph spacing"
⚠️ Agent guidance — keep invocations minimal:
CORE RULE: default command (./scripts/transcribe audio.mp3) is the fastest path — add flags only when the user explicitly asks for that capability.
Transcription:
- Only add
--diarize if the user asks "who said what" / "identify speakers" / "label speakers"
- Only add
--format srt/vtt/ass/lrc/ttml if the user asks for subtitles/captions in that format
- Only add
--format csv if the user asks for CSV or spreadsheet output
- Only add
--word-timestamps if the user needs word-level timing
- Only add
--initial-prompt if there's domain-specific jargon to prime
- Only add
--translate if the user wants non-English audio translated to English
- Only add
--normalize/--denoise if the user mentions bad audio quality or noise
- Only add
--stream if the user wants live/progressive output for long files
- Only add
--clip-timestamps if the user wants a specific time range
- Only add
--temperature 0.0 if the model is hallucinating on music/silence
- Only add
--vad-threshold if VAD is aggressively cutting speech or including noise
- Only add
--min-speakers/--max-speakers when you know the speaker count
- Only add
--hf-token if the token is not cached at ~/.cache/huggingface/token
- Only add
--max-words-per-line for subtitle readability on long segments
- Only add
--filter-hallucinations if the transcript contains obvious artifacts (music markers, duplicates)
- Only add
--merge-sentences if the user asks for sentence-level subtitle cues
- Only add
--clean-filler if the user asks to remove filler words (um, uh, you know, I mean, hesitation sounds)
- Only add
--channel left|right if the user mentions stereo tracks, dual-channel recordings, or asks for a specific channel
- Only add
--max-chars-per-line N when the user specifies a character limit per subtitle line (e.g., "Netflix format", "42 chars per line"); takes priority over --max-words-per-line
- Only add
--detect-paragraphs if the user asks for paragraph breaks or structured text output; --paragraph-gap (default 3.0s) only if they want a custom gap
- Only add
--speaker-names "Alice,Bob" when the user provides real names to replace SPEAKER_1/2 — always requires --diarize
- Only add
--hotwords WORDS when the user names specific rare terms not well served by --initial-prompt; prefer --initial-prompt for general domain jargon
- Only add
--prefix TEXT when the user knows the exact words the audio starts with
- Only add
--detect-language-only when the user only wants to identify the language, not transcribe
- Only add
--stats-file PATH if the user asks for performance stats, RTF, or benchmark info
- Only add
--parallel N for large CPU batch jobs; GPU handles one file efficiently on its own — don't add for single files or small batches
- Only add
--retries N for unreliable inputs (URLs, network files) where transient failures are expected
- Only add
--burn-in OUTPUT only when user explicitly asks to embed/burn subtitles into the video; requires ffmpeg and a video file input
- Only add
--keep-temp when the user may re-process the same URL to avoid re-downloading
- Only add
--output-template when user specifies a custom naming pattern in batch mode
- Multi-format output (
--format srt,text): only when user explicitly wants multiple formats in one pass; always pair with -o <dir>
- Any word-level feature auto-runs wav2vec2 alignment (~5-10s overhead)
--diarize adds ~20-30s on top of that
Search:
- Only add
--search "term" when the user asks to find/locate/search for a specific word or phrase in audio
--search replaces the normal transcript output — it prints only matching segments with timestamps- Add
--search-fuzzy only when the user mentions approximate/partial matching or typos
- To save search results to a file, use
-o results.txt
Chapter detection:
- Only add
--detect-chapters when the user asks for chapters, sections, a table of contents, or "where does the topic change"
- Default
--chapter-gap 8 (8-second silence = new chapter) works for most podcasts/lectures; tune down for dense content
--chapter-format youtube (default) outputs YouTube-ready timestamps; use json for programmatic use- Always use
--chapters-file PATH when combining chapters with a transcript output — avoids mixing chapter markers into the transcript text
- If the user only wants chapters (not the transcript), pipe stdout to a file with
-o /dev/null and use --chapters-file
- Batch mode limitation:
--chapters-file takes a single path — in batch mode, each file's chapters overwrite the previous. For batch chapter detection, omit --chapters-file (chapters print to stdout under === CHAPTERS (N) ===) or use a separate run per file
Speaker audio export:
- Only add
--export-speakers DIR when the user explicitly asks to save each speaker's audio separately
- Always pair with
--diarize — it silently skips if no speaker labels are present
- Requires ffmpeg; outputs
SPEAKER_1.wav, SPEAKER_2.wav, etc. (or real names if --speaker-names is set)
Language map:
- Only add
--language-map in batch mode when the user has confirmed different languages across files
- Inline format:
"interview*.mp3=en,lecture*.mp3=fr" — fnmatch globs on filename
- JSON file format:
@/path/to/map.json where the file is {"pattern": "lang_code"}
RSS / Podcast:
- Only add
--rss URL when the user provides a podcast RSS feed URL
- Default fetches 5 newest episodes;
--rss-latest 0 for all; --skip-existing to resume safely
- Always use
-o <dir> with --rss — without it, all episode transcripts print to stdout concatenated, which is hard to use; each episode gets its own file when -o <dir> is set
Output format for agent relay:
- Search results (
--search) → print directly to user; output is human-readable
- Chapter output → if no
--chapters-file, chapters appear in stdout under === CHAPTERS (N) === header after the transcript; with --format json, chapters are also embedded in the JSON under "chapters" key
- Subtitle formats (SRT, VTT, ASS, LRC, TTML) → always write to
-o file; tell the user the output path, never paste raw subtitle content
- Data formats (CSV, HTML, TTML, JSON) → always write to
-o file; tell the user the output path, don't paste raw XML/CSV/HTML
- ASS format → for Aegisub, VLC, mpv; write to file and tell user they can open it in Aegisub or play it in VLC/mpv
- LRC format → timed lyrics for music players (Foobar2000, AIMP, VLC); write to file
- Multi-format (
--format srt,text) → requires -o <dir>; each format goes to a separate file; tell user all paths written
- JSON format → useful for programmatic post-processing; not ideal to paste in full to user
- Text/transcript → safe to show directly to user for short files; summarise for long ones
- Stats output (
--stats-file) → summarise key fields (duration, processing time, RTF) for the user rather than pasting raw JSON
- Language detection (
--detect-language-only) → print the result directly; it's a single line
- ETA is printed automatically to stderr for batch jobs; no action needed
When NOT to use:
- Cloud-only environments without local compute
- Files <10 seconds where API call latency doesn't matter
faster-whisper vs whisperx:
This skill covers everything whisperx does — diarization (--diarize), word-level timestamps (--word-timestamps), SRT/VTT subtitles — so whisperx is not needed. Use whisperx only if you specifically need its pyannote pipeline or batch-GPU features not covered here.
Quick Reference
| Task | Command | Notes |
|---|
| Basic transcription | ./scripts/transcribe audio.mp3 | Batched inference, VAD on, distil-large-v3.5 |
| SRT subtitles | ./scripts/transcribe audio.mp3 --format srt -o subs.srt | Word timestamps auto-enabled |
| VTT subtitles | ./scripts/transcribe audio.mp3 --format vtt -o subs.vtt | WebVTT format |
| Word timestamps | ./scripts/transcribe audio.mp3 --word-timestamps --format srt | wav2vec2 aligned (~10ms) |
| Speaker diarization | ./scripts/transcribe audio.mp3 --diarize | Requires pyannote.audio |
| Translate → English | ./scripts/transcribe audio.mp3 --translate | Any language → English |
| Stream output | ./scripts/transcribe audio.mp3 --stream | Live segments as transcribed |
| Clip time range | ./scripts/transcribe audio.mp3 --clip-timestamps "30,60" | Only 30s–60s |
| Denoise + normalize | ./scripts/transcribe audio.mp3 --denoise --normalize | Clean up noisy audio first |
| Reduce hallucination | ./scripts/transcribe audio.mp3 --hallucination-silence-threshold 1.0 | Skip hallucinated silence |
| YouTube/URL | ./scripts/transcribe https://youtube.com/watch?v=... | Auto-downloads via yt-dlp |
| Batch process | ./scripts/transcribe *.mp3 -o ./transcripts/ | Output to directory |
| Batch with skip | ./scripts/transcribe *.mp3 --skip-existing -o ./out/ | Resume interrupted batches |
| Domain terms | ./scripts/transcribe audio.mp3 --initial-prompt 'Kubernetes gRPC' | Boost rare terminology |
| Hotwords boost | ./scripts/transcribe audio.mp3 --hotwords 'JIRA Kubernetes' | Bias decoder toward specific words |
| Prefix conditioning | ./scripts/transcribe audio.mp3 --prefix 'Good morning,' | Seed the first segment with known opening words |
| Pin model version | ./scripts/transcribe audio.mp3 --revision v1.2.0 | Reproducible transcription with a pinned revision |
| Debug library logs | ./scripts/transcribe audio.mp3 --log-level debug | Show faster_whisper internal logs |
| Turbo model | ./scripts/transcribe audio.mp3 -m turbo | Alias for large-v3-turbo |
| Faster English | ./scripts/transcribe audio.mp3 --model distil-medium.en -l en | English-only, 6.8x faster |
| Maximum accuracy | ./scripts/transcribe audio.mp3 --model large-v3 --beam-size 10 | Full model |
| JSON output | ./scripts/transcribe audio.mp3 --format json -o out.json | Programmatic access with stats |
| Filter noise | ./scripts/transcribe audio.mp3 --min-confidence 0.6 | Drop low-confidence segments |
| Hybrid quantization | ./scripts/transcribe audio.mp3 --compute-type int8_float16 | Save VRAM, minimal quality loss |
| Reduce batch size | ./scripts/transcribe audio.mp3 --batch-size 4 | If OOM on GPU |
| TSV output | ./scripts/transcribe audio.mp3 --format tsv -o out.tsv | OpenAI Whisper–compatible TSV |
| Fix hallucinations | ./scripts/transcribe audio.mp3 --temperature 0.0 --no-speech-threshold 0.8 | Lock temperature + skip silence |
| Tune VAD sensitivity | ./scripts/transcribe audio.mp3 --vad-threshold 0.6 --min-silence-duration 500 | Tighter speech detection |
| Known speaker count | ./scripts/transcribe meeting.wav --diarize --min-speakers 2 --max-speakers 3 | Constrain diarization |
| Subtitle word wrapping | ./scripts/transcribe audio.mp3 --format srt --word-timestamps --max-words-per-line 8 | Split long cues |
| Private/gated model | ./scripts/transcribe audio.mp3 --hf-token hf_xxx | Pass token directly |
| Show version | ./scripts/transcribe --version | Print faster-whisper version |
| Upgrade in-place | ./setup.sh --update | Upgrade without full reinstall |
| System check | ./setup.sh --check | Verify GPU, Python, ffmpeg, venv, yt-dlp, pyannote |
| Detect language only | ./scripts/transcribe audio.mp3 --detect-language-only | Fast language ID, no transcription |
| Detect language JSON | ./scripts/transcribe audio.mp3 --detect-language-only --format json | Machine-readable language detection |
| LRC subtitles | ./scripts/transcribe audio.mp3 --format lrc -o lyrics.lrc | Timed lyrics format for music players |
| ASS subtitles | ./scripts/transcribe audio.mp3 --format ass -o subtitles.ass | Advanced SubStation Alpha (Aegisub, mpv, VLC) |
| Merge sentences | ./scripts/transcribe audio.mp3 --format srt --merge-sentences | Join fragments into sentence chunks |
| Stats sidecar | ./scripts/transcribe audio.mp3 --stats-file stats.json | Write perf stats JSON after transcription |
| Batch stats | ./scripts/transcribe *.mp3 --stats-file ./stats/ | One stats file per input in dir |
| Template naming | ./scripts/transcribe audio.mp3 -o ./out/ --output-template "{stem}_{lang}.{ext}" | Custom batch output filenames |
| Stdin input | ffmpeg -i input.mp4 -f wav - | ./scripts/transcribe - | Pipe audio directly from stdin |
| Custom model dir | ./scripts/transcribe audio.mp3 --model-dir ~/my-models | Custom HuggingFace cache dir |
| Local model | ./scripts/transcribe audio.mp3 -m ./my-model-ct2 | CTranslate2 model dir |
| HTML transcript | ./scripts/transcribe audio.mp3 --format html -o out.html | Confidence-colored |
| Burn subtitles | ./scripts/transcribe video.mp4 --burn-in output.mp4 | Requires ffmpeg + video input |
| Name speakers | ./scripts/transcribe audio.mp3 --diarize --speaker-names "Alice,Bob" | Replaces SPEAKER_1/2 |
| Filter hallucinations | ./scripts/transcribe audio.mp3 --filter-hallucinations | Removes artifacts |
| Keep temp files | ./scripts/transcribe https://... --keep-temp | For URL re-processing |
| Parallel batch | ./scripts/transcribe *.mp3 --parallel 4 -o ./out/ | CPU multi-file |
| RTX 3070 recommended | ./scripts/transcribe audio.mp3 --compute-type int8_float16 | Saves ~1GB VRAM, minimal quality loss |
| CPU thread count | ./scripts/transcribe audio.mp3 --threads 8 | Force CPU thread count (default: auto) |
| Podcast RSS (latest 5) | ./scripts/transcribe --rss https://feeds.example.com/podcast.xml | Downloads & transcribes newest 5 episodes |
| Podcast RSS (all episodes) | ./scripts/transcribe --rss https://... --rss-latest 0 -o ./episodes/ | All episodes, one file each |
| Podcast + SRT subtitles | ./scripts/transcribe --rss https://... --format srt -o ./subs/ | Subtitle all episodes |
| Retry on failure | ./scripts/transcribe *.mp3 --retries 3 -o ./out/ | Retry up to 3× with backoff on error |
| CSV output | ./scripts/transcribe audio.mp3 --format csv -o out.csv | Spreadsheet-ready with header row; properly quoted |
| CSV with speakers | ./scripts/transcribe audio.mp3 --diarize --format csv -o out.csv | Adds speaker column |
| Language map (inline) | ./scripts/transcribe *.mp3 --language-map "interview*.mp3=en,lecture.wav=fr" | Per-file language in batch |
| Language map (JSON) | ./scripts/transcribe *.mp3 --language-map @langs.json | JSON file: {"pattern": "lang"} |
| Batch with ETA | ./scripts/transcribe *.mp3 -o ./out/ | Automatic ETA shown for each file in batch |
| TTML subtitles | ./scripts/transcribe audio.mp3 --format ttml -o subtitles.ttml | Broadcast-standard DFXP/TTML (Netflix, BBC, Amazon) |
| TTML with speaker labels | ./scripts/transcribe audio.mp3 --diarize --format ttml -o subtitles.ttml | Speaker-labeled TTML |
| Search transcript | ./scripts/transcribe audio.mp3 --search "keyword" | Find timestamps where keyword appears |
| Search to file | ./scripts/transcribe audio.mp3 --search "keyword" -o results.txt | Save search results |
| Fuzzy search | ./scripts/transcribe audio.mp3 --search "aproximate" --search-fuzzy | Approximate/partial matching |
| Detect chapters | ./scripts/transcribe audio.mp3 --detect-chapters | Auto-detect chapters from silence gaps |
| Chapter gap tuning | ./scripts/transcribe audio.mp3 --detect-chapters --chapter-gap 5 | Chapters on gaps ≥5s (default: 8s) |
| Chapters to file | ./scripts/transcribe audio.mp3 --detect-chapters --chapters-file ch.txt | Save YouTube-format chapter list |
| Chapters JSON | ./scripts/transcribe audio.mp3 --detect-chapters --chapter-format json | Machine-readable chapter list |
| Export speaker audio | ./scripts/transcribe audio.mp3 --diarize --export-speakers ./speakers/ | Save each speaker's audio to separate WAV files |
| Multi-format output | ./scripts/transcribe audio.mp3 --format srt,text -o ./out/ | Write SRT + TXT in one pass |
| Remove filler words | ./scripts/transcribe audio.mp3 --clean-filler | Strip um/uh/er/ah/hmm and discourse markers |
| Left channel only | ./scripts/transcribe audio.mp3 --channel left | Extract left stereo channel before transcribing |
| Right channel only | ./scripts/transcribe audio.mp3 --channel right | Extract right stereo channel |
| Max chars per line | ./scripts/transcribe audio.mp3 --format srt --max-chars-per-line 42 | Character-based subtitle wrapping |
| Detect paragraphs | ./scripts/transcribe audio.mp3 --detect-paragraphs | Insert paragraph breaks in text output |
| Paragraph gap tuning | ./scripts/transcribe audio.mp3 --detect-paragraphs --paragraph-gap 5.0 | Tune gap threshold (default 3.0s) |
Model Selection
Choose the right model for your needs:
digraph model_selection {
rankdir=LR;
node [shape=box, style=rounded];
start [label="Start", shape=doublecircle];
need_accuracy [label="Need maximum\naccuracy?", shape=diamond];
multilingual [label="Multilingual\ncontent?", shape=diamond];
resource_constrained [label="Resource\nconstraints?", shape=diamond];
large_v3 [label="large-v3\nor\nlarge-v3-turbo", style="rounded,filled", fillcolor=lightblue];
large_turbo [label="large-v3-turbo", style="rounded,filled", fillcolor=lightblue];
distil_large [label="distil-large-v3.5\n(default)", style="rounded,filled", fillcolor=lightgreen];
distil_medium [label="distil-medium.en", style="rounded,filled", fillcolor=lightyellow];
distil_small [label="distil-small.en", style="rounded,filled", fillcolor=lightyellow];
start -> need_accuracy;
need_accuracy -> large_v3 [label="yes"];
need_accuracy -> multilingual [label="no"];
multilingual -> large_turbo [label="yes"];
multilingual -> resource_constrained [label="no (English)"];
resource_constrained -> distil_small [label="mobile/edge"];
resource_constrained -> distil_medium [label="some limits"];
resource_constrained -> distil_large [label="no"];
}
Model Table
Standard Models (Full Whisper)
| Model | Size | Speed | Accuracy | Use Case |
|---|
tiny / tiny.en | 39M | Fastest | Basic | Quick drafts |
base / base.en | 74M | Very fast | Good | General use |
small / small.en | 244M | Fast | Better | Most tasks |
medium / medium.en | 769M | Moderate | High | Quality transcription |
large-v1/v2/v3 | 1.5GB | Slower | Best | Maximum accuracy |
large-v3-turbo | 809M | Fast | Excellent | High accuracy (slower than distil) |
Distilled Models (~6x Faster, ~1% WER difference)
| Model | Size | Speed vs Standard | Accuracy | Use Case |
|---|
distil-large-v3.5 | 756M | ~6.3x faster | 7.08% WER | Default, best balance |
distil-large-v3 | 756M | ~6.3x faster | 7.53% WER | Previous default |
distil-large-v2 | 756M | ~5.8x faster | 10.1% WER | Fallback |
distil-medium.en | 394M | ~6.8x faster | 11.1% WER | English-only, resource-constrained |
distil-small.en | 166M | ~5.6x faster | 12.1% WER | Mobile/edge devices |
.en models are English-only and slightly faster/better for English content.
Note for distil models: HuggingFace recommends disabling condition_on_previous_text for all distil models to prevent repetition loops. The script auto-applies --no-condition-on-previous-text whenever a distil-* model is detected. Pass --condition-on-previous-text to override if needed.
Custom & Fine-tuned Models
WhisperModel accepts local CTranslate2 model directories and HuggingFace repo names — no code changes needed.
Load a local CTranslate2 model
./scripts/transcribe audio.mp3 --model /path/to/my-model-ct2
Convert a HuggingFace model to CTranslate2
pip install ctranslate2
ct2-transformers-converter \
--model openai/whisper-large-v3 \
--output_dir whisper-large-v3-ct2 \
--copy_files tokenizer.json preprocessor_config.json \
--quantization float16
./scripts/transcribe audio.mp3 --model ./whisper-large-v3-ct2
Load a model by HuggingFace repo name (auto-downloads)
./scripts/transcribe audio.mp3 --model username/whisper-large-v3-ct2
Custom model cache directory
By default, models are cached in ~/.cache/huggingface/. Use --model-dir to override:
./scripts/transcribe audio.mp3 --model-dir ~/my-models
Setup
Linux / macOS / WSL2
./setup.sh
./setup.sh --diarize
Requirements:
- Python 3.10+
- ffmpeg is not required for basic transcription — PyAV (bundled with faster-whisper) handles audio decoding. ffmpeg is only needed for
--burn-in, --normalize, and --denoise.
- Optional: yt-dlp (for URL/YouTube input)
- Optional: pyannote.audio (for
--diarize, installed via setup.sh --diarize)
Platform Support
| Platform | Acceleration | Speed |
|---|
| Linux + NVIDIA GPU | CUDA | ~20x realtime 🚀 |
| WSL2 + NVIDIA GPU | CUDA | ~20x realtime 🚀 |
| macOS Apple Silicon | CPU* | ~3-5x realtime |
| macOS Intel | CPU | ~1-2x realtime |
| Linux (no GPU) | CPU | ~1x realtime |
*faster-whisper uses CTranslate2 which is CPU-only on macOS, but Apple Silicon is fast enough for practical use.
GPU Support (IMPORTANT!)
The setup script auto-detects your GPU and installs PyTorch with CUDA. Always use GPU if available — CPU transcription is extremely slow.
| Hardware | Speed | 9-min video |
|---|
| RTX 3070 (GPU) | ~20x realtime | ~27 sec |
| CPU (int8) | ~0.3x realtime | ~30 min |
RTX 3070 tip: Use --compute-type int8_float16 for hybrid quantization — saves ~1GB VRAM with minimal quality loss. Ideal for running diarization alongside transcription.
If setup didn't detect your GPU, manually install PyTorch with CUDA:
uv pip install --python .venv/bin/python torch --index-url https://download.pytorch.org/whl/cu121
uv pip install --python .venv/bin/python torch --index-url https://download.pytorch.org/whl/cu118
Usage
./scripts/transcribe audio.mp3
./scripts/transcribe audio.mp3 --format srt -o subtitles.srt
./scripts/transcribe audio.mp3 --format vtt -o subtitles.vtt
./scripts/transcribe https://youtube.com/watch?v=dQw4w9WgXcQ --language en
./scripts/transcribe meeting.wav --diarize
./scripts/transcribe meeting.wav --diarize --format vtt -o meeting.vtt
./scripts/transcribe lecture.mp3 --initial-prompt "Kubernetes, gRPC, PostgreSQL, NGINX"
./scripts/transcribe ./recordings/ -o ./transcripts/
./scripts/transcribe *.mp3 --skip-existing -o ./transcripts/
./scripts/transcribe noisy-audio.mp3 --min-confidence 0.6
./scripts/transcribe audio.mp3 --format json -o result.json
./scripts/transcribe audio.mp3 --language en
Options
Input:
AUDIO Audio file(s), directory, glob pattern, or URL
Accepts: mp3, wav, m4a, flac, ogg, webm, mp4, mkv, avi, wma, aac
URLs auto-download via yt-dlp (YouTube, direct links, etc.)
Model & Language:
-m, --model NAME Whisper model (default: distil-large-v3.5; "turbo" = large-v3-turbo)
--revision REV Model revision (git branch/tag/commit) to pin a specific version
-l, --language CODE Language code, e.g. en, es, fr (auto-detects if omitted)
--initial-prompt TEXT Prompt to condition the model (terminology, formatting style)
--prefix TEXT Prefix to condition the first segment (e.g. known starting words)
--hotwords WORDS Space-separated hotwords to boost recognition
--translate Translate any language to English (instead of transcribing)
--multilingual Enable multilingual/code-switching mode (helps smaller models)
--hf-token TOKEN HuggingFace token for private/gated models and diarization
--model-dir PATH Custom model cache directory (default: ~/.cache/huggingface/)
Output Format:
-f, --format FMT text | json | srt | vtt | tsv | lrc | html | ass | ttml (default: text)
Accepts comma-separated list: --format srt,text writes both in one pass
Multi-format requires -o <dir> when saving to files
--word-timestamps Include word-level timestamps (wav2vec2 aligned automatically)
--stream Output segments as they are transcribed (disables diarize/alignment)
--max-words-per-line N For SRT/VTT, split segments into sub-cues of at most N words
--max-chars-per-line N For SRT/VTT/ASS/TTML, split lines so each fits within N characters
Takes priority over --max-words-per-line when both are set
--clean-filler Remove hesitation fillers (um, uh, er, ah, hmm, hm) and discourse markers
(you know, I mean, you see) from transcript text. Off by default.
--detect-paragraphs Insert paragraph breaks (blank lines) in text output at natural boundaries.
A new paragraph starts when: silence gap ≥ --paragraph-gap, OR the previous
segment ends a sentence AND the gap ≥ 1.5s.
--paragraph-gap SEC Minimum silence gap in seconds to start a new paragraph (default: 3.0).
Used with --detect-paragraphs.
--channel {left,right,mix}
Stereo channel to transcribe: left (c0), right (c1), or mix (default: mix).
Extracts the channel via ffmpeg before transcription. Requires ffmpeg.
--merge-sentences Merge consecutive segments into sentence-level chunks
(improves SRT/VTT readability; groups by terminal punctuation or >2s gap)
-o, --output PATH Output file or directory (directory for batch mode)
--output-template TEMPLATE
Batch output filename template. Variables: {stem}, {lang}, {ext}, {model}
Example: "{stem}_{lang}.{ext}" → "interview_en.srt"
Inference Tuning:
--beam-size N Beam search size; higher = more accurate but slower (default: 5)
--temperature T Sampling temperature or comma-separated fallback list, e.g.
'0.0' or '0.0,0.2,0.4' (default: faster-whisper's schedule)
--no-speech-threshold PROB
Probability threshold to mark segments as silence (default: 0.6)
--batch-size N Batched inference batch size (default: 8; reduce if OOM)
--no-vad Disable voice activity detection (on by default)
--vad-threshold T VAD speech probability threshold (default: 0.5)
--vad-neg-threshold T VAD negative threshold for ending speech (default: auto)
--vad-onset T Alias for --vad-threshold (legacy)
--vad-offset T Alias for --vad-neg-threshold (legacy)
--min-speech-duration MS Minimum speech segment duration in ms (default: 0)
--max-speech-duration SEC Maximum speech segment duration in seconds (default: unlimited)
--min-silence-duration MS Minimum silence before splitting a segment in ms (default: 2000)
--speech-pad MS Padding around speech segments in ms (default: 400)
--no-batch Disable batched inference (use standard WhisperModel)
--hallucination-silence-threshold SEC
Skip silent sections where model hallucinates (e.g. 1.0)
--no-condition-on-previous-text
Don't condition on previous text (reduces repetition/hallucination loops;
auto-enabled for distil models per HuggingFace recommendation)
--condition-on-previous-text
Force-enable conditioning on previous text (overrides auto-disable for distil models)
--compression-ratio-threshold RATIO
Filter segments above this compression ratio (default: 2.4)
--log-prob-threshold PROB
Filter segments below this avg log probability (default: -1.0)
--max-new-tokens N Maximum tokens per segment (prevents runaway generation)
--clip-timestamps RANGE
Transcribe specific time ranges: '30,60' or '0,30;60,90' (seconds)
--progress Show transcription progress bar
--best-of N Candidates when sampling with non-zero temperature (default: 5)
--patience F Beam search patience factor (default: 1.0)
--repetition-penalty F Penalty for repeated tokens (default: 1.0)
--no-repeat-ngram-size N Prevent n-gram repetitions of this size (default: 0 = off)
Advanced Inference:
--no-timestamps Output text without timing info (faster; incompatible with
--word-timestamps, --format srt/vtt/tsv, --diarize)
--chunk-length N Audio chunk length in seconds for batched inference (default: auto)
--language-detection-threshold T
Confidence threshold for language auto-detection (default: 0.5)
--language-detection-segments N
Audio segments to sample for language detection (default: 1)
--length-penalty F Beam search length penalty; >1 favors longer, <1 favors shorter (default: 1.0)
--prompt-reset-on-temperature T
Reset initial prompt when temperature fallback hits threshold (default: 0.5)
--no-suppress-blank Disable blank token suppression (may help soft/quiet speech)
--suppress-tokens IDS Comma-separated token IDs to suppress in addition to default -1
--max-initial-timestamp T
Maximum timestamp for the first segment in seconds (default: 1.0)
--prepend-punctuations CHARS
Punctuation characters merged into preceding word (default: "'¿([{-)
--append-punctuations CHARS
Punctuation characters merged into following word (default: "'.。,，!！?？:：")]}、")
Preprocessing:
--normalize Normalize audio volume (EBU R128 loudnorm) before transcription
--denoise Apply noise reduction (high-pass + FFT denoise) before transcription
Advanced:
--diarize Speaker diarization (requires pyannote.audio)
--min-speakers N Minimum number of speakers hint for diarization
--max-speakers N Maximum number of speakers hint for diarization
--speaker-names NAMES Comma-separated names to replace SPEAKER_1, SPEAKER_2 (e.g. 'Alice,Bob')
Requires --diarize
--min-confidence PROB Filter segments below this avg word confidence (0.0–1.0)
--skip-existing Skip files whose output already exists (batch mode)
--detect-language-only
Detect language and exit (no transcription). Output: "Language: en (probability: 0.984)"
With --format json: {"language": "en", "language_probability": 0.984}
--stats-file PATH Write JSON stats sidecar after transcription (processing time, RTF, word count, etc.)
Directory path → writes {stem}.stats.json inside; file path → exact path
--burn-in OUTPUT Burn subtitles into the original video (single-file mode only; requires ffmpeg)
--filter-hallucinations
Filter common Whisper hallucinations: music/applause markers, duplicate segments,
'Thank you for watching', lone punctuation, etc.
--keep-temp Keep temp files from URL downloads (useful for re-processing without re-downloading)
--parallel N Number of parallel workers for batch processing (default: sequential)
--retries N Retry failed files up to N times with exponential backoff (default: 0;
incompatible with --parallel)
Batch ETA:
Automatically shown for sequential batch jobs (no flag needed). After each file completes,
the next file's progress line includes: [current/total] filename | ETA: Xm Ys
ETA is calculated from average time per file × remaining files.
Shown to stderr (surfaced to users via OpenClaw/Clawdbot output).
Language Map (per-file language override):
--language-map MAP Per-file language override for batch mode. Two forms:
Inline: "interview*.mp3=en,lecture.wav=fr,keynote.wav=de"
JSON file: "@/path/to/map.json" (must be {pattern: lang} dict)
Patterns support fnmatch globs on filename or stem.
Priority: exact filename > exact stem > glob on filename > glob on stem > fallback.
Files not matched fall back to --language (or auto-detect if not set).
Transcript Search:
--search TERM Search the transcript for TERM and print matching segments with timestamps.
Replaces normal transcript output (use -o to save results to a file).
Case-insensitive exact substring match by default.
--search-fuzzy Enable fuzzy/approximate matching with --search (useful for typos, phonetic
near-misses, or partial words; uses SequenceMatcher ratio ≥ 0.6)
Chapter Detection:
--detect-chapters Auto-detect chapter/section breaks from silence gaps and print chapter markers.
Output is printed after the transcript (or to --chapters-file).
--chapter-gap SEC Minimum silence gap in seconds between consecutive segments to start a new
chapter (default: 8.0). Tune down for dense speech, up for sparse content.
--chapters-file PATH Write chapter markers to this file (default: stdout after transcript)
--chapter-format FMT youtube | text | json — chapter output format:
youtube: "0:00 Chapter 1" (YouTube description ready)
text: "Chapter 1: 00:00:00"
json: JSON array with chapter, start, title fields
(default: youtube)
Speaker Audio Export:
--export-speakers DIR After diarization, export each speaker's audio turns concatenated into
separate WAV files saved in DIR. Requires --diarize and ffmpeg.
Output: SPEAKER_1.wav, SPEAKER_2.wav, … (or real names if --speaker-names set)
RSS / Podcast:
--rss URL Podcast RSS feed URL — extracts audio enclosures and transcribes them.
AUDIO positional is optional when --rss is used.
--rss-latest N Number of most-recent episodes to process (default: 5; 0 = all episodes)
Device:
--device DEV auto | cpu | cuda (default: auto)
--compute-type TYPE auto | int8 | int8_float16 | float16 | float32 (default: auto)
int8_float16 = hybrid mode for GPU (saves VRAM, minimal quality loss)
--threads N CPU thread count for CTranslate2 (default: auto)
-q, --quiet Suppress progress and status messages
--log-level LEVEL Set faster_whisper library logging level: debug | info | warning | error
(default: warning; use debug to see CTranslate2/VAD internals)
Utility:
--version Print installed faster-whisper version and exit
--update Upgrade faster-whisper in the skill venv and exit
Output Formats
Text (default)
Plain transcript text. With --diarize, speaker labels are inserted:
[SPEAKER_1]
Hello, welcome to the meeting.
[SPEAKER_2]
Thanks for having me.
JSON (--format json)
Full metadata including segments, timestamps, language detection, and performance stats:
{
"file": "audio.mp3",
"text": "Hello, welcome...",
"language": "en",
"language_probability": 0.98,
"duration": 600.5,
"segments": [...],
"speakers": ["SPEAKER_1", "SPEAKER_2"],
"stats": {
"processing_time": 28.3,
"realtime_factor": 21.2
}
}
SRT (--format srt)
Standard subtitle format for video players:
1
00:00:00,000 --> 00:00:02,500
[SPEAKER_1] Hello, welcome to the meeting.
2
00:00:02,800 --> 00:00:04,200
[SPEAKER_2] Thanks for having me.
VTT (--format vtt)
WebVTT format for web video players:
WEBVTT
1
00:00:00.000 --> 00:00:02.500
[SPEAKER_1] Hello, welcome to the meeting.
2
00:00:02.800 --> 00:00:04.200
[SPEAKER_2] Thanks for having me.
TSV (--format tsv)
Tab-separated values, OpenAI Whisper–compatible. Columns: start_ms, end_ms, text:
0 2500 Hello, welcome to the meeting.
2800 4200 Thanks for having me.
Useful for piping into other tools or spreadsheets. No header row.
ASS/SSA (--format ass)
Advanced SubStation Alpha format — supported by Aegisub, VLC, mpv, MPC-HC, and most video editors. Offers richer styling than SRT (font, size, color, position) via the [V4+ Styles] section:
[Script Info]
ScriptType: v4.00+
...
[V4+ Styles]
Style: Default,Arial,20,&H00FFFFFF,...
[Events]
Format: Layer, Start, End, Style, Name, ..., Text
Dialogue: 0,0:00:00.00,0:00:02.50,Default,,[SPEAKER_1] Hello, welcome.
Dialogue: 0,0:00:02.80,0:00:04.20,Default,,[SPEAKER_2] Thanks for having me.
Timestamps use H:MM:SS.cc (centiseconds). Edit the [V4+ Styles] block in Aegisub to customise font, color, and position without re-transcribing.
LRC (--format lrc)
Timed lyrics format used by music players (e.g., Foobar2000, VLC, AIMP). Timestamps use [mm:ss.xx] where xx = centiseconds:
[00:00.50]Hello, welcome to the meeting.
[00:02.80]Thanks for having me.
With diarization, speaker labels are included:
[00:00.50][SPEAKER_1] Hello, welcome to the meeting.
[00:02.80][SPEAKER_2] Thanks for having me.
Default file extension: .lrc. Useful for music transcription, karaoke, and any workflow requiring timed text with music-player compatibility.
Speaker Diarization
Identifies who spoke when using pyannote.audio.
Setup:
./setup.sh --diarize
Requirements:
- HuggingFace token at
~/.cache/huggingface/token (huggingface-cli login)
- Accepted model agreements:
Usage:
./scripts/transcribe meeting.wav --diarize
./scripts/transcribe meeting.wav --diarize --format srt -o meeting.srt
./scripts/transcribe meeting.wav --diarize --format json
Speakers are labeled SPEAKER_1, SPEAKER_2, etc. in order of first appearance. Diarization runs on GPU automatically if CUDA is available.
Precise Word Timestamps
Whenever word-level timestamps are computed (--word-timestamps, --diarize, or --min-confidence), a wav2vec2 forced alignment pass automatically refines them from Whisper's ~100-200ms accuracy to ~10ms. No extra flag needed.
./scripts/transcribe audio.mp3 --word-timestamps --format json
./scripts/transcribe meeting.wav --diarize
./scripts/transcribe audio.mp3 --word-timestamps --format srt -o subtitles.srt
Uses the MMS (Massively Multilingual Speech) model from torchaudio — supports 1000+ languages. The model is cached after first load, so batch processing stays fast.
URL & YouTube Input
Pass any URL as input — audio is downloaded automatically via yt-dlp:
./scripts/transcribe https://youtube.com/watch?v=dQw4w9WgXcQ
./scripts/transcribe https://example.com/podcast.mp3
./scripts/transcribe https://youtube.com/watch?v=... --language en --format srt -o subs.srt
Requires yt-dlp (checks PATH and ~/.local/share/pipx/venvs/yt-dlp/bin/yt-dlp).
Batch Processing
Process multiple files at once with glob patterns, directories, or multiple paths:
./scripts/transcribe *.mp3
./scripts/transcribe ./recordings/
./scripts/transcribe *.mp3 -o ./transcripts/
./scripts/transcribe *.mp3 --skip-existing -o ./transcripts/
./scripts/transcribe file1.mp3 file2.wav ./more-recordings/
./scripts/transcribe *.mp3 --format srt -o ./subtitles/
When outputting to a directory, files are named {input-stem}.{ext} (e.g., audio.mp3 → audio.srt).
Batch mode prints a summary after all files complete:
📊 Done: 12 files, 3h24m audio in 10m15s (19.9× realtime)
Workflows
End-to-end pipelines for common use cases.
Podcast Transcription Pipeline
Fetch and transcribe the latest 5 episodes from any podcast RSS feed:
./scripts/transcribe --rss https://feeds.megaphone.fm/mypodcast -o ./transcripts/
./scripts/transcribe --rss https://... --rss-latest 0 --format srt -o ./subtitles/
./scripts/transcribe --rss https://... --skip-existing -o ./transcripts/
./scripts/transcribe --rss https://... --diarize --retries 2 -o ./transcripts/
Meeting Notes Pipeline
Transcribe a meeting recording with speaker labels, then output clean text:
./scripts/transcribe meeting.wav --diarize --speaker-names "Alice,Bob" -o meeting.txt
./scripts/transcribe meeting.wav --diarize --format json -o meeting.json
./scripts/transcribe meeting.wav --stream
Video Subtitle Pipeline
Generate ready-to-use subtitles for a video file:
./scripts/transcribe video.mp4 --format srt --merge-sentences -o subtitles.srt
./scripts/transcribe video.mp4 --format srt --burn-in video_subtitled.mp4
./scripts/transcribe video.mp4 --format srt --word-timestamps --max-words-per-line 8 -o subs.srt
YouTube Batch Pipeline
Transcribe multiple YouTube videos at once:
./scripts/transcribe "https://youtube.com/watch?v=abc123" --format srt -o subs.srt
cat urls.txt | xargs ./scripts/transcribe -o ./transcripts/
./scripts/transcribe https://youtube.com/watch?v=abc123 --keep-temp
Noisy Audio Pipeline
Clean up poor-quality recordings before transcribing:
./scripts/transcribe interview.mp3 --denoise --normalize -o interview.txt
./scripts/transcribe *.mp3 --denoise --filter-hallucinations -o ./out/
Batch Recovery Pipeline
Process a large folder with retries — safe to re-run after failures:
./scripts/transcribe ./recordings/ --skip-existing --retries 3 -o ./transcripts/
Server Mode (OpenAI-Compatible API)
speaches runs faster-whisper as an OpenAI-compatible /v1/audio/transcriptions endpoint — drop-in replacement for OpenAI Whisper API with streaming, Docker support, and live transcription.
Quick start (Docker)
docker run --gpus all -p 8000:8000 ghcr.io/speaches-ai/speaches:latest-cuda
Test it
curl http://localhost:8000/v1/audio/transcriptions \
-F file=@audio.mp3 \
-F model=Systran/faster-whisper-large-v3
Use with any OpenAI SDK
from openai import OpenAI
client = OpenAI(base_url="http://localhost:8000", api_key="none")
with open("audio.mp3", "rb") as f:
result = client.audio.transcriptions.create(model="Systran/faster-whisper-large-v3", file=f)
print(result.text)
Useful when you want to expose transcription as a local API for other tools (Home Assistant, n8n, custom apps).
Common Mistakes
| Mistake | Problem | Solution |
|---|
| Using CPU when GPU available | 10-20x slower transcription | Check nvidia-smi; verify CUDA installation |
| Not specifying language | Wastes time auto-detecting on known content | Use --language en when you know the language |
| Using wrong model | Unnecessary slowness or poor accuracy | Default distil-large-v3.5 is excellent; only use large-v3 if accuracy issues |
| Ignoring distilled models | Missing 6x speedup with <1% accuracy loss | Try distil-large-v3.5 before reaching for standard models |
| Forgetting ffmpeg | Setup fails or audio can't be processed | Setup script handles this; manual installs need ffmpeg separately |
| Out of memory errors | Model too large for available VRAM/RAM | Use smaller model, --compute-type int8, or --batch-size 4 |
| Over-engineering beam size | Diminishing returns past beam-size 5-7 | Default 5 is fine; try 10 for critical transcripts |
| --diarize without pyannote | Import error at runtime | Run setup.sh --diarize first |
| --diarize without HuggingFace token | Model download fails | Run huggingface-cli login and accept model agreements |
| URL input without yt-dlp | Download fails | Install: pipx install yt-dlp |
| --min-confidence too high | Drops good segments with natural pauses | Start at 0.5, adjust up; check JSON output for probabilities |
| Using --word-timestamps for basic transcription | Adds ~5-10s overhead for negligible benefit | Only use when word-level precision matters |
| Batch without -o directory | All output mixed in stdout | Use -o ./transcripts/ to write one file per input |
Performance Notes
- First run: Downloads model to
~/.cache/huggingface/ (one-time)
- Batched inference: Enabled by default via
BatchedInferencePipeline — ~3x faster than standard mode; VAD on by default
- GPU: Automatically uses CUDA if available
- Quantization: INT8 used on CPU for ~4x speedup with minimal accuracy loss
- Performance stats: Every transcription shows audio duration, processing time, and realtime factor
- Benchmark (RTX 3070, 21-min file): ~24s with batched inference (both distil-large-v3 and v3.5) vs ~69s without
- --precise overhead: Adds ~5-10s for wav2vec2 model load + alignment (model cached for batch)
- Diarization overhead: Adds ~10-30s depending on audio length (runs on GPU if available)
- Memory:
distil-large-v3: ~2GB RAM / ~1GB VRAMlarge-v3-turbo: ~4GB RAM / ~2GB VRAMtiny/base: <1GB RAM- Diarization: additional ~1-2GB VRAM
- OOM: Lower
--batch-size (try 4) if you hit out-of-memory errors
- Pre-convert to WAV (optional):
ffmpeg -i input.mp3 -ar 16000 -ac 1 input.wav converts to 16kHz mono WAV before transcription. Benefit is minimal (~5%) for one-off use since PyAV decodes efficiently — most useful when re-processing the same file multiple times (research/experiments) or when a format causes PyAV decode issues. Note: --normalize and --denoise already perform this conversion automatically.
- Silero VAD V6: faster-whisper 1.2.1 upgraded to Silero VAD V6 (improved speech detection). Run
./setup.sh --update to get it.
- Batched silence removal: faster-whisper 1.2.0+ automatically removes silence in
BatchedInferencePipeline (used by default). Upgrade with ./setup.sh --update to get this if you installed before August 2024.
Why faster-whisper?
- Speed: ~4-6x faster than OpenAI's original Whisper
- Accuracy: Identical (uses same model weights)
- Efficiency: Lower memory usage via quantization
- Production-ready: Stable C++ backend (CTranslate2)
- Distilled models: ~6x faster with <1% accuracy loss
- Subtitles: Native SRT/VTT/HTML output
- Precise alignment: Automatic wav2vec2 refinement (~10ms word boundaries)
- Diarization: Optional speaker identification via pyannote;
--speaker-names maps to real names
- URLs: Direct YouTube/URL input;
--keep-temp preserves downloads for re-use
- Custom models: Load local CTranslate2 dirs or HuggingFace repos;
--model-dir controls cache
- Quality control:
--filter-hallucinations strips music/applause markers and duplicates
- Parallel batch:
--parallel N for multi-threaded batch processing
- Subtitle burn-in:
--burn-in overlays subtitles directly into video via ffmpeg
v1.5.0 New Features
Multi-format output:
--format srt,text — write multiple formats in one pass (e.g. SRT + plain text simultaneously)- Comma-separated list accepted:
srt,vtt,json, srt,text, etc.
- Requires
-o <dir> when writing multiple formats; single format unchanged
Filler word removal:
--clean-filler — strip hesitation sounds (um, uh, er, ah, hmm, hm) and discourse markers
(you know, I mean, you see) from transcript text; off by default- Conservative regex matching at word boundaries to avoid false positives
- Segments that become empty after cleaning are dropped automatically
Stereo channel selection:
--channel left|right|mix — extract a specific stereo channel before transcribing (default: mix)- Useful for dual-track recordings (interviewer on left, interviewee on right)
- Uses ffmpeg pan filter; falls back gracefully to full mix if ffmpeg not found
Character-based subtitle wrapping:
--max-chars-per-line N — split subtitle cues so each line fits within N characters- Works for SRT, VTT, ASS, and TTML formats; takes priority over
--max-words-per-line
- Requires word-level timestamps; falls back to full segment if no word data
Paragraph detection:
--detect-paragraphs — insert \n\n paragraph breaks in text output at natural boundaries--paragraph-gap SEC — minimum silence gap for a paragraph (default: 3.0s)- Also detects paragraph breaks when the previous segment ends a sentence and gap ≥ 1.5s
Subtitle formats:
--format ass — Advanced SubStation Alpha (Aegisub, VLC, mpv, MPC-HC)--format lrc — Timed lyrics format for music players--format html — Confidence-colored HTML transcript (green/yellow/red per word)--format ttml — W3C TTML 1.0 (DFXP) broadcast standard (Netflix, Amazon Prime, BBC)--format csv — Spreadsheet-ready CSV with header row; RFC 4180 quoting; speaker column when diarized
Transcript tools:
--search TERM — Find all timestamps where a word/phrase appears; replaces normal output; -o to save--search-fuzzy — Approximate/partial matching with --search--detect-chapters — Auto-detect chapter breaks from silence gaps; --chapter-gap SEC (default 8s)--chapters-file PATH — Write chapters to file instead of stdout; --chapter-format youtube|text|json--export-speakers DIR — After --diarize, save each speaker's turns as separate WAV files via ffmpeg
Batch improvements:
- ETA —
[N/total] filename | ETA: Xm Ys shown before each file in sequential batch; no flag needed
--language-map "pat=lang,..." — Per-file language override; fnmatch glob patterns; @file.json form--retries N — Retry failed files with exponential backoff; failed-file summary at end--rss URL — Transcribe podcast RSS feeds; --rss-latest N for episode count--skip-existing / --parallel N / --output-template / --stats-file / --merge-sentences
Model & inference:
distil-large-v3.5 default (replaced distil-large-v3)- Auto-disables
condition_on_previous_text for distil models (prevents repetition loops)
--condition-on-previous-text to override; --log-level for library debug output--model-dir PATH — Custom HuggingFace cache dir; local CTranslate2 model support--no-timestamps, --chunk-length, --length-penalty, --repetition-penalty, --no-repeat-ngram-size--clip-timestamps, --stream, --progress, --best-of, --patience, --max-new-tokens--hotwords, --prefix, --revision, --suppress-tokens, --max-initial-timestamp
Speaker & quality:
--speaker-names "Alice,Bob" — Replace SPEAKER_1/2 with real names (requires --diarize)--filter-hallucinations — Remove music/applause markers, duplicates, "Thank you for watching"--burn-in OUTPUT — Burn subtitles into video via ffmpeg--keep-temp — Preserve URL-downloaded audio for re-processing
Setup:
setup.sh --check — System diagnostic: GPU, CUDA, Python, ffmpeg, pyannote, HuggingFace token (completes in ~12s)- ffmpeg no longer required for basic transcription (PyAV handles decoding);
skill.json updated to reflect this (ffmpeg is now optionalBins)
Troubleshooting
"CUDA not available — using CPU": Install PyTorch with CUDA (see GPU Support above)
Setup fails: Make sure Python 3.10+ is installed
Out of memory: Use smaller model, --compute-type int8, or --batch-size 4
Slow on CPU: Expected — use GPU for practical transcription
Model download fails: Check ~/.cache/huggingface/ permissions
Diarization model fails: Ensure HuggingFace token exists and model agreements accepted;
or pass token directly with --hf-token hf_xxx
URL download fails: Check yt-dlp is installed (pipx install yt-dlp)
No audio files in batch: Check file extensions match supported formats
Check installed version: Run ./scripts/transcribe --version
Upgrade faster-whisper: Run ./setup.sh --update (upgrades in-place, no full reinstall)
Hallucinations on silence/music: Try --temperature 0.0 --no-speech-threshold 0.8
VAD splits speech incorrectly: Tune with --vad-threshold 0.3 (lower) or --min-silence-duration 300
Improve speech detection: Run ./setup.sh --update to upgrade faster-whisper to the latest version (includes Silero VAD V6).
References
