podcast-edit

name	podcast-edit
description	Edit podcast audio — trim pre/post-show chat, remove filler words, cut silences, and enhance audio quality. Use when the user asks to edit a podcast, clean up audio, remove fillers, trim a recording, or improve voice quality.
user_invocable	true

Podcast Edit Skill

Process raw podcast/meeting recordings into polished podcast episodes.

Capabilities

1. Smart trimming — Find where the actual podcast starts/ends by transcribing and detecting intros/outros
2. Filler word removal — Remove verbal tics: 嗯, 呃, 啊, 哦, 对对对, um, uh, etc.
3. Silence trimming — Cut long dead air (>2s) down to natural pauses (~0.6s)
4. Audio enhancement — Noise reduction, EQ, multi-speaker volume balancing, loudness normalization to podcast standard (−16 LUFS)
5. Re-cutting a finished episode — Surgically remove flagged sections from an already-rendered episode without re-running the whole pipeline
6. Highlight clips & reel — Cut shareable soundbites and stitch a ~1-minute reel with music

Prerequisites

• ffmpeg and ffprobe installed
• OPENAI_API_KEY in environment (for Whisper API transcription)
• Python 3 with stdlib only (no extra deps for the helper script)
• Optional: resemblyzer (pip install resemblyzer) — only for speaker diarization when building highlight reels

Workflow

Step 1: Inspect the audio file

ffprobe -v quiet -print_format json -show_format -show_streams "INPUT_FILE"

Note: duration, sample rate, channels, codec, bitrate.

Step 2: Find podcast start/end (if user says to trim front/back)

Split into 5-minute chunks and transcribe via OpenAI Whisper API with segment-level timestamps:

# Extract chunk
ffmpeg -y -i "INPUT_FILE" -ss OFFSET -t 300 -ar 16000 -ac 1 /tmp/chunk_OFFSET.mp3

# Transcribe
curl -s https://api.openai.com/v1/audio/transcriptions \
  -H "Authorization: Bearer $OPENAI_API_KEY" \
  -F file="@/tmp/chunk_OFFSET.mp3" \
  -F model="whisper-1" \
  -F response_format="verbose_json" \
  -F language="LANG" \
  -F 'timestamp_granularities[]=segment' > /tmp/transcript_OFFSET.json

Scan transcriptions for:

• Start markers: "welcome", "hello everyone", "大家好", "欢迎", intro music, first substantive topic sentence
• End markers: "see you next time", "bye", "下期见", "感谢收听", followed by post-show chat

Do an initial trim with -ss START -to END and -c copy (no re-encode) to create a working file.

Step 3: Remove filler words

Split the trimmed file into 5-minute chunks and transcribe each with word-level timestamps:

# Extract chunks
for i in $(seq 0 300 DURATION); do
  ffmpeg -y -i "TRIMMED_FILE" -ss $i -t 300 -ar 16000 -ac 1 /tmp/wchunk_${i}.mp3
done

# Transcribe each chunk (can run in parallel)
for i in $(seq 0 300 DURATION); do
  curl -s https://api.openai.com/v1/audio/transcriptions \
    -H "Authorization: Bearer $OPENAI_API_KEY" \
    -F file="@/tmp/wchunk_${i}.mp3" \
    -F model="whisper-1" \
    -F response_format="verbose_json" \
    -F language="LANG" \
    -F 'timestamp_granularities[]=word' \
    -F 'timestamp_granularities[]=segment' > /tmp/wtranscript_${i}.json &
done
wait

Then run the filler removal script that ships with this skill:

python3 ./filler_removal.py \
  --total-duration DURATION \
  --end-at END_TIMESTAMP \
  --cut START1:END1 --cut START2:END2 \
  --chunk-offsets 0,300,600,900,...

Arguments:

• --total-duration: Duration of the trimmed input file in seconds (required)
• --end-at: Cut everything after this timestamp (e.g., post-show chat start)
• --cut START:END: Cut a specific range. Can be repeated.
• --chunk-offsets: Comma-separated chunk offsets (default: auto 0,300,600,…)

The script outputs /tmp/ffmpeg_filter.txt with an atrim+concat filter.

Apply the filter in two passes:

# Step A: Cut fillers → intermediate WAV (avoids re-encoding artifacts)
ffmpeg -y -i "TRIMMED_FILE" \
  -filter_complex_script /tmp/ffmpeg_filter.txt \
  -map '[out]' -c:a pcm_s16le -ar 44100 /tmp/podcast_cut.wav

# Step B: Enhance audio → final MP3
ffmpeg -y -i /tmp/podcast_cut.wav \
  -af "ENHANCEMENT_CHAIN" \
  -c:a libmp3lame -b:a 192k "OUTPUT_FILE"

Limitations: Whisper word-level timestamps for Chinese can miss fillers that are blended into adjacent speech. The script catches standalone fillers reliably but may miss ~10–20% of embedded ones.

Step 4: Audio enhancement filter chain

Default chain (guest-friendly — handles multi-speaker volume imbalance). The biggest mistake in past runs is using a noise gate (agate) that silences the quieter guest entirely. Never add agate back to the default chain.

highpass=f=80,                                    # Remove room rumble
lowpass=f=12000,                                  # Remove hiss (use 7500 for 16kHz sources)
afftdn=nf=-25:nr=8:nt=w,                         # Gentle FFT noise reduction
equalizer=f=180:t=q:w=1.5:g=-2,                  # Cut mud
equalizer=f=2500:t=q:w=1.2:g=3,                  # Boost presence
equalizer=f=4500:t=q:w=1.5:g=1.5,                # Boost clarity
dynaudnorm=f=200:g=5:p=0.95:m=5:s=0,             # Rolling-window normalization — lifts the quieter speaker independently
acompressor=threshold=-20dB:ratio=2:attack=5:release=200:makeup=1,  # Gentle glue
loudnorm=I=-16:TP=-1.5:LRA=13                    # Podcast standard loudness

Why dynaudnorm is the star: it normalizes in 200 ms rolling windows, so when the guest is speaking, that window gets lifted independently of the host's louder windows. Order matters — run dynaudnorm BEFORE acompressor so the compressor sees a balanced signal.

Never add these to the default chain:

• agate (noise gate) — cuts off any speaker quieter than the threshold; kills the guest.
• Heavy compression (ratio >3:1, makeup >2 dB) — flattens dynamics and makes the guest sound pumped.
• Narrow LRA (<12) in loudnorm — crushes natural speech dynamics.

Adjust lowpass based on source sample rate:

• 16kHz source → lowpass=7500
• 44.1kHz+ source → lowpass=12000 (or skip)

Verify guest audibility after rendering: run ffmpeg -i OUTPUT -af "ebur128=peak=true" -f null - and check I: is near −16 LUFS and LRA: is 4–6 LU (tighter LRA is fine because dynaudnorm did per-window balancing first). If the output sounds like the guest was cut, suspect a gate or aggressive compressor crept back in.

Step 5: Verify output

ls -lh "OUTPUT_FILE"
ffprobe -v quiet -show_entries format=duration -of csv=p=0 "OUTPUT_FILE"

Report: duration, file size, what was removed (filler count, silence count, time saved).

Output conventions

• Format: MP3, 192 kbps, mono (unless source is stereo with separate speakers per channel)
• Loudness: −16 LUFS (podcast standard)
• Always two-pass: cut to WAV first, then enhance to MP3

Re-cutting an already-finished episode

When the user flags specific problems in a published episode ("a few sentences were left in", "cut the part about X", "the ending was re-recorded", "drop the duplicate intro"), do NOT re-run the whole pipeline from raw. Surgically cut the offending ranges out of the finished MP3 and re-encode once:

1. Re-transcribe the final file in 5-min chunks (segment granularity) → one unified transcript with absolute timestamps. This is your map.
2. For each flagged item, get word-level timestamps for just the chunk containing it (these phrases are usually embedded mid-sentence, not on segment boundaries). Pin tight in/out points.
3. Build one atrim+concat filter that keeps everything except the cut ranges; render to WAV (pcm_s16le) then a single MP3 pass. Do not re-apply the enhancement chain — the final is already enhanced/loudnorm'd; re-running it double-processes.
4. Verify every splice: extract an ~11s window centred on each new join (in the new timeline) and re-transcribe it. Confirm the bad audio is gone and the rejoin reads cleanly.
5. Recompute any show-notes timeline: every timestamp after a cut shifts earlier by the summed length of all cuts before it. If the published file prepends a highlight reel, also account for the reel's duration as an offset (a re-cut reel changes length, so the whole episode shifts).

Re-cut checklist — what the first pass commonly leaves in

These survive filler/silence removal because they're blended into real sentences. Scan the transcript for them explicitly:

• Meeting/tech chatter: "you're muted / unmute", "你声音太小 / 听得到吗", "can you see my screen", "等一下我看一下". Cut the phrase, keep the surrounding content.
• Dead-end topics: the panel raises a product/case nobody researched, then says "我们就跳过吧 / let's skip it" — cut the whole detour, not just the skip line.
• Re-recorded segments (esp. endings): a messy first take, then a marker like "重新开始 / 从头来 / let's restart / feels redundant", then a clean re-take. Cut the false-start take, keep the re-record. Always inspect the last ~3 min for this.
• Pre-show / duplicate intros: a self-introduction recorded before the official start that gets repeated later — cut the pre-show copy.

Whisper API flakiness

Parallel word-level calls sometimes return empty (0 bytes). Retry the empties sequentially with a sleep 1 between calls.

Highlight clips & 1-min reel

Cut short, shareable soundbites from a finished episode (controversial / insightful moments), and optionally stitch them into a ~1-minute reel with music.

Picking soundbites

• Each clip ≈ 2 sentences, ~8–20s. Favor lines that are controversial, surprising, or quotable.
• COVER EVERY SPEAKER — including the host(s). On a multi-guest panel, do not let the reel collapse onto the 1–2 most talkative voices — pick at least one strong soundbite per participant and check attribution before building. (This is the #1 mistake: a "highlight" that's secretly one person.) The facilitator/host is the easiest to omit because they mostly ask questions — find a quotable host line (a reaction, a joke, a sharp framing) and include it too.
• You cannot attribute speakers from transcript text — do not try, and do not ask the user to do it for you. Whisper gives you words, not who said them; guessing the speaker from phrasing fails (a line that reads like the host is often actually a guest). When the reel needs a specific person, diarize the audio (recipe below) and attribution becomes a measurement, not a guess.
• Order the reel for flow (hook → … → strong closer) and never put the same speaker back-to-back.

Identify WHO is speaking — speaker diarization (resemblyzer)

When you need a specific person's clip (or the user says "X isn't in the reel"), resolve it by voice, not by reading the transcript. pip install resemblyzer (bundles its own encoder — no HF token). Recipe that worked on a 5-speaker panel:

1. Build clean reference voices for everyone you can identify, from segments where they say their own name (intros) or pitch their own product (closing plugs) — these are guaranteed single-speaker. Average 2–3 windows per person for stability: enc.embed_utterance(preprocess_wav(slice, source_sr=16000)), mean, L2-normalize.
2. Don't blind-cluster the whole episode — Whisper segments contain crosstalk, so AgglomerativeClustering collapses into one giant blob + singletons. Instead, score every segment's embedding (cosine) against your clean references. Segments that match nobody well (best sim ≲ 0.78 when real matches land 0.88–0.94) are the unidentified Nth speaker.
3. Confirm it's one consistent voice: take those low-match segments, check they're mutually similar (≳0.87) and collectively distinct from each known reference. Build a reference from them, re-score all segments — the high-confidence hits should all be that person.
4. Sanity-check against a known: a confirmed clip (e.g. someone's product pitch) must score highest to its own reference (≳0.9). If your references don't separate, lengthen the windows and average more.

Load the whole episode once as 16 kHz mono float via an ffmpeg pipe (-f f32le -) and slice in memory — far faster than one ffmpeg call per segment.

Verify before you cut (mandatory)

For each candidate, extract the window and re-transcribe it to (a) confirm it's the right content/speaker and (b) find clean sentence boundaries. Whisper mangles names — never trust the first transcript's spelling. Use word-level granularity to pin a start that doesn't clip the first word and an end that drops stutters/repeats.

Two gotchas that waste time

• -ss / -to must be INPUT options (before -i). As output options they produce silence or wrong ranges. ffmpeg -ss START -to END -i in.mp3 ...
• zsh arrays are 1-indexed. for s in "${SEG[@]}" (iterate values) — never ${SEG[$i]} from i=0.
• If the episode was edited (intro/low-quality cut) after the master was made, source clips from the edited file with shifted timestamps (subtract the seconds removed before each cut point).

Build one clip (pauses removed + tune in/out)

# 1) extract + remove pauses (collapse gaps >0.2s; gaps sit near -25dB after dynaudnorm, so threshold ~-23dB)
ffmpeg -y -ss START -to END -i FINAL.mp3 -ar 44100 -ac 1 \
  -af "silenceremove=start_periods=1:start_silence=0.04:start_threshold=-30dB:stop_periods=-1:stop_duration=0.20:stop_threshold=-23dB:detection=peak" sr.wav
# 2) speech fades (compute fade-out start from sr.wav duration)
ffmpeg -y -i sr.wav -af "afade=t=in:st=0:d=0.12,afade=t=out:st=${DUR-0.3}:d=0.3" f.wav
# 3) tune in -> speech -> tune out
ffmpeg -y -i sting_in.wav -i f.wav -i sting_out.wav \
  -filter_complex "[0][1]acrossfade=d=0.18:c1=tri:c2=tri[a];[a][2]acrossfade=d=0.18:c1=tri:c2=tri[out]" \
  -map "[out]" -c:a libmp3lame -b:a 192k clip.mp3

Synthesize the stings (no audio assets needed) — a soft bell chord, low volume:

# tune-in: bright C-E-G bell
ffmpeg -y -f lavfi -i "sine=f=523.25:d=0.85" -f lavfi -i "sine=f=659.25:d=0.85" -f lavfi -i "sine=f=783.99:d=0.85" \
  -filter_complex "[0][1][2]amix=inputs=3:normalize=1,afade=t=in:st=0:d=0.02,afade=t=out:st=0.2:d=0.65,lowpass=f=3800,volume=0.30[s]" -map "[s]" -ar 44100 -ac 1 sting_in.wav
# tune-out: lower, gentler G-C bell
ffmpeg -y -f lavfi -i "sine=f=392:d=0.75" -f lavfi -i "sine=f=523.25:d=0.75" \
  -filter_complex "[0][1]amix=inputs=2:normalize=1,afade=t=in:st=0:d=0.06,afade=t=out:st=0.15:d=0.6,lowpass=f=3200,volume=0.22[s]" -map "[s]" -ar 44100 -ac 1 sting_out.wav

1-min reel (concat + music bed + fade in/out)

Concat the pause-trimmed speech segments (1.5s silent lead/tail, ~0.35s gaps between) via the concat demuxer, then mix a soft synth pad underneath and fade the whole piece in/out. Don't reuse the per-clip stings inside the reel — one master fade is cleaner.

# music bed: warm 4-chord pad (C-G-Am-F), each chord 4s, lowpass+tremolo, concat -> pad16.wav, then loop
# (mkchord mixes 3 sines, normalize=1, lowpass=f=750, tremolo=f=4.5:d=0.25, afade in/out)
ffmpeg -y -i speech_reel.wav -stream_loop 6 -i pad16.wav \
  -filter_complex "[1]atrim=0:${TOT},volume=0.075[m];[0][m]amix=inputs=2:normalize=0:duration=first[mix];[mix]afade=t=in:st=0:d=1.3,afade=t=out:st=${TOT-1.6}:d=1.6[out]" \
  -map "[out]" -c:a libmp3lame -b:a 192k highlight.mp3

• Pad volume ~0.075 keeps speech fully dominant (verify by re-transcribing the final mix — every line should still read cleanly). amix … normalize=0 so the voice isn't ducked.
• Target the same −16 LUFS; the music shouldn't push it past ~−16.5.

Output layout

• Individual clips: episodes/ep{NNN}/highlights/ep{NNN}-clip{N}-{who}.mp3
• Single reel: episodes/ep{NNN}/ep{NNN}-highlight.mp3

Cover art generation (optional)

Generate episode cover art with the OpenAI GPT Image API (gpt-image-1), matching your show's house style. Supply a style reference image of your own (a previous cover, your wordmark, your palette) — the model imitates it.

import openai, base64

client = openai.OpenAI()  # uses OPENAI_API_KEY from env
style_img = open("YOUR_STYLE_REFERENCE.png", "rb")

result = client.images.edit(
    model="gpt-image-1",
    image=[style_img],                       # add a content reference as a 2nd image if you have one
    prompt="""Create an illustration in the EXACT same art style as this image
(match the line work, color palette, background, and decorative elements).
Depict: [DESCRIBE THE SCENE]. Keep [YOUR SHOW NAME / wordmark] in the same
style and position as the reference.""",
    size="1024x1024",
)
with open("cover.png", "wb") as f:
    f.write(base64.b64decode(result.data[0].b64_json))

Notes: load OPENAI_API_KEY from the environment, output a 1024×1024 PNG, and keep your show's wordmark/branding consistent across episodes.

Show notes — bilingual writing (if applicable)

If the host is producing bilingual Chinese/English show notes, the Chinese section must be written in actual Chinese — not Chinese grammar with English verbs and nouns sprinkled in. Code-switching like "close 了一个 deal", "build 出来的 agent", or "PR 不是 buy 来的" reads like a draft and is the #1 mistake to avoid.

Translation rules

Translate these common startup/tech English loanwords into Chinese:

• close deal → 拿下订单 / 成交 / 签下
• build (a product) → 搭建 / 做出 / 打造
• integration → 集成
• view (video/page views) → 播放 / 浏览
• stack (tech stack) → 体系 / 技术栈
• category leader → 品类领导者
• front-end / front end (product sense) → 外壳 / 前端
• success story → 客户案例 / 成功故事
• SMB → 中小企业
• Enterprise (segment) → 大型企业 / 企业级
• aha moment → 顿悟时刻
• onboarding → 上手 / 入门
• retention → 留存
• churn → 流失
• pipeline → 销售漏斗 / 业务线

What to KEEP in English inside Chinese text

• Brand and product names — company / product / person names stay as-is
• Very common startup acronyms — CEO, CTO, CMO, PMF, ARR, MRR, PR, AI, AI Agent, SaaS, API
• Currency with numeric prefix — $20K, $200K, or 200 美金 (either form is fine when paired with a number)

Before finalizing

Re-read the Chinese section as a Chinese reader. If any sentence feels like it was half-translated — e.g., contains "build", "close", "deal", "view", "stack", "leader" as standalone English words — rewrite those words in Chinese. The only English that should survive a re-read is brand names and the acronyms above.

Name verification (CRITICAL)

Whisper frequently mangles company names, product names, and personal names. Before generating show notes or any output that includes names and links:

1. After transcription, extract all proper nouns — company names, product names, personal names, URLs mentioned.
2. Ask the user to confirm/correct them — Whisper hears similar-sounding but wrong tokens for brand names.
3. Never guess URLs from transcribed names — a name that sounds like "Acme" could be acme.com, acmehq.com, or something else entirely. Always ask.
4. Use confirmed names consistently in show notes, titles, episode metadata, and all outputs.

This is especially important when generating backlinks or social posts — a misspelled domain is a wasted link.

Show notes structure (recommended)

Two separate sections — Chinese first, then English (or whichever languages the show targets). Do NOT interleave or put them side-by-side.

Heading rule: keep headings shallow and consistent — pick one level (e.g. H2) and flatten all sub-sections to it. Some publishing platforms only render a single heading level plus bold; if yours does, match it.

Timestamp format: always MM:SS with leading zeros (e.g., 08:25, 00:00, 42:10). Never 0:00 or 1:05.

EP{NNN}: {Episode title}

---

## 中文

**嘉宾：** {中文姓名 English Name}, {中文职位} {公司} (URL)

## 简介
{完整中文段落}

## 时间轴
- 00:00 — {中文描述}
- 08:25 — {中文描述}

## 核心要点
- {中文要点}

## 相关链接
- {品牌名}：{URL}

---

## English

**Guest:** {English Name}, {Title} at {Company} (URL)

## Summary
{Full English paragraph}

## Timestamps
- 00:00 — {English description}
- 08:25 — {English description}

## Key Takeaways
- {English takeaway}

## Links
- {Brand}: {URL}

Why two sections instead of bilingual bullets: Chinese readers want clean Chinese prose, English readers want clean English prose. Alternating "中文 / English" on every bullet makes both halves harder to read. Write each section as if it were the only one.

Archiving & naming conventions (recommended)

Keep each episode self-contained in its own folder. A simple, zero-padded layout scales cleanly:

episodes/
├── ep001/
│   ├── ep001-final.mp3      # the finished episode
│   ├── ep001-highlight.mp3  # optional 1-min reel
│   ├── cover.png
│   └── shownotes.md
└── ep002/
    └── ...

• Directory: ep{NNN} (zero-padded 3 digits)
• Audio: ep{NNN}-final.mp3; highlight clips under ep{NNN}/highlights/ep{NNN}-clip{N}-{who}.mp3
• Episode title: lead with a catchy, descriptive title (hook the reader, don't just state the topic).

Quick trim (no filler removal)

If the user just wants a simple trim (e.g., "cut the first 3s"):

ffmpeg -y -i "INPUT" -ss 3 -c copy "OUTPUT"

Use -c copy for instant lossless trim when no audio processing is needed.