/interview-transcription
Transcription workflows, recording management, and quote extraction for journalists. Use when processing audio/video recordings, generating transcripts with timestamps, extracting quotes for fact-checking, or building source-and-recording databases. For interview question design
$ npx -y skills add jamditis/claude-skills-journalism --skill interview-transcription --agent claude-codeHow it fires
How this skill gets triggered: by you, by Claude, or both.
- Fires itselfAuto-invocation. Claude auto-loads it when your prompt matches the work.Auto-invocation is when the right skill fires by itself at the right moment, driven by a FLOW.md router and a hook, instead of you invoking it by name. It is the difference between a skill being installed and a skill actually getting used.Read the full definition →
- You can call itInvoke it directly when you want it.
- Slash command
/interview-transcription
Context preview
The summary Claude sees to decide when to auto-load this skill.
Transcription workflows, recording management, and quote extraction for journalists. Use when processing audio/video recordings, generating transcripts with timestamps, extracting quotes for fact-checking, or building source-and-recording databases. For interview question design
SKILL.md
interview-transcription.SKILL.mdname: interview-transcription
description: Transcription workflows, recording management, and quote extraction for journalists. Use when processing audio/video recordings, generating transcripts with timestamps, extracting quotes for fact-checking, or building source-and-recording databases. For interview question design and pre-interview preparation, see the interview-prep skill.
Interview transcription and management
Practical workflows for journalists managing interviews from preparation through publication.
When to activate
- Preparing questions for an interview
- Processing audio/video recordings
- Creating or managing transcripts
- Organizing notes from multiple sources
- Building a source relationship database
- Generating timestamped quotes for fact-checking
- Converting recordings to publishable quotes
Recording setup for transcription
For pre-interview research, question design, attribution agreements, and consent scripts, use the **interview-prep** skill. The notes here cover only the recording configuration that affects transcription quality.
# Standard recording configuration for clean transcription
RECORDING_SETTINGS = {
'format': 'wav', # Lossless for transcription
'sample_rate': 16000, # Whisper resamples to 16k anyway; 16k saves disk
'channels': 1, # Mono is fine for speech; stereo only if mics are positionally distinct
'backup': True, # Always run a backup recorder
}
# File naming convention
# YYYY-MM-DD_source-lastname_topic.wav
# Example: 2026-05-08_smith_budget-hearing.wav**Two-device rule.** Always record on two devices. Phone as backup minimum. If using a wireless lav mic, the recorder built into the lav unit is one device; the phone running a backup app is the second.
**Mono is preferred** unless each speaker has their own dedicated microphone routed to a distinct channel. Stereo with both speakers bleeding into both channels is worse for diarization than clean mono.
Transcription workflows
Automated transcription pipeline
Vanilla OpenAI Whisper transcribes audio to text but does **not** assign speaker labels. To get diarized output ("Speaker 1:" / "Speaker 2:" / etc.) you need a tool that combines Whisper with a diarization model — typically **WhisperX** (`m-bain/whisperX`), which wraps faster-whisper transcription with pyannote.audio diarization and produces word-level timestamps with speaker IDs in one pass.
from pathlib import Path
import subprocess
import json
def transcribe_interview(
audio_path: str,
output_dir: str = "./transcripts",
diarize: bool = True,
hf_token: str | None = None,
min_speakers: int = 2,
max_speakers: int = 2,
) -> dict:
"""
Transcribe an interview using WhisperX (Whisper + pyannote diarization).
Returns a transcript with word-level timestamps and speaker labels.
Diarization needs a Hugging Face token with access to the pyannote
speaker-diarization-3.1 model. Accept the model EULA at
huggingface.co/pyannote/speaker-diarization-3.1 once, then pass the token.
"""
Path(output_dir).mkdir(exist_ok=True)
cmd = [
'whisperx', audio_path,
'--model', 'large-v3',
'--output_format', 'json',
'--output_dir', output_dir,
'--language', 'en',
'--compute_type', 'int8', # CPU-friendly; use 'float16' on GPU
'--min_speakers', str(min_speakers),
'--max_speakers', str(max_speakers),
]
if diarize:
cmd.append('--diarize')
if hf_token:
cmd += ['--hf_token', hf_token]
subprocess.run(cmd, check=True, capture_output=True)
json_path = Path(output_dir) / f"{Path(audio_path).stem}.json"
with open(json_path) as f:
return json.load(f)
def format_for_editing(transcript: dict) -> str:
"""Convert to journalist-friendly format with timestamps."""
lines = []
for segment in transcript.get('segments', []):
timestamp = format_timestamp(segment['start'])
text = segment['text'].strip()
lines.append(f"[{timestamp}] {text}")
return '\n\n'.join(lines)
def format_timestamp(seconds: float) -> str:
"""Convert seconds to HH:MM:SS format."""
h = int(seconds // 3600)
m = int((seconds % 3600) // 60)
s = int(seconds % 60)
return f"{h:02d}:{m:02d}:{s:02d}"**Falling back to plain Whisper.** If diarization is overkill or you can't get a Hugging Face token, drop the `--diarize` flag — the model still produces accurate timestamped transcription and you label speakers manually based on context. `faster-whisper` (CTranslate2 backend) is the speed-optimized variant and works the same way at the CLI. `whisper.cpp` is the C++ port for resource-constrained machines (Raspberry Pi, older laptops); it doesn't include diarization but runs the small/medium models on CPU comfortably.
Manual transcription template
For sensitive interviews or when AI transcription fails:
## Transcript: [Source] - [Date]
**Recording file**: [filename]
**Duration**: [XX:XX]
**Transcribed by**: [name]
**Verified against recording**: [ ] Yes / [ ] No
---
[00:00:15] **Q**: [Your question]
[00:00:45] **A**: [Source response - verbatim, including ums, pauses noted as (...)]
[00:01:30] **Q**: [Follow-up]
[00:01:42] **A**: [Response]
---
## Notes
- [Anything not captured in audio: gestures, documents shown, etc.]
## Potential quotes
- [00:01:42] "Quote that stands out" - context: [why it matters]
Quote extraction and verification
Pull quotes workflow
from dataclasses import dataclass
from typing import Optional
import re
@dataclass
class Quote:
text: str
timestamp: str
speaker: str
context: str
verified: bool = False
used_in: Optional[str] = None
class QuoteBank:
"""Manage quotes from interview transcripts."""
def __init__(self):
self.quotes = []
def extract_quote(self,Read more
name: interview-transcription description: Transcription workflows, recording management, and quote extraction for journalists. Use when processing audio/video recordings, generating transcripts with timestamps, extracting quotes for fact-checking, or building source-and-recording databases. For interview question design and pre-interview preparation, see the interview-prep skill.
Interview transcription and management
Practical workflows for journalists managing interviews from preparation through publication.
When to activate
- Preparing questions for an interview
- Processing audio/video recordings
- Creating or managing transcripts
- Organizing notes from multiple sources
- Building a source relationship database
- Generating timestamped quotes for fact-checking
- Converting recordings to publishable quotes
Recording setup for transcription
For pre-interview research, question design, attribution agreements, and consent scripts, use the **interview-prep** skill. The notes here cover only the recording configuration that affects transcription quality.
# Standard recording configuration for clean transcription
RECORDING_SETTINGS = {
'format': 'wav', # Lossless for transcription
'sample_rate': 16000, # Whisper resamples to 16k anyway; 16k saves disk
'channels': 1, # Mono is fine for speech; stereo only if mics are positionally distinct
'backup': True, # Always run a backup recorder
}
# File naming convention
# YYYY-MM-DD_source-lastname_topic.wav
# Example: 2026-05-08_smith_budget-hearing.wav**Two-device rule.** Always record on two devices. Phone as backup minimum. If using a wireless lav mic, the recorder built into the lav unit is one device; the phone running a backup app is the second.
**Mono is preferred** unless each speaker has their own dedicated microphone routed to a distinct channel. Stereo with both speakers bleeding into both channels is worse for diarization than clean mono.
Transcription workflows
Automated transcription pipeline
Vanilla OpenAI Whisper transcribes audio to text but does **not** assign speaker labels. To get diarized output ("Speaker 1:" / "Speaker 2:" / etc.) you need a tool that combines Whisper with a diarization model — typically **WhisperX** (`m-bain/whisperX`), which wraps faster-whisper transcription with pyannote.audio diarization and produces word-level timestamps with speaker IDs in one pass.
from pathlib import Path
import subprocess
import json
def transcribe_interview(
audio_path: str,
output_dir: str = "./transcripts",
diarize: bool = True,
hf_token: str | None = None,
min_speakers: int = 2,
max_speakers: int = 2,
) -> dict:
"""
Transcribe an interview using WhisperX (Whisper + pyannote diarization).
Returns a transcript with word-level timestamps and speaker labels.
Diarization needs a Hugging Face token with access to the pyannote
speaker-diarization-3.1 model. Accept the model EULA at
huggingface.co/pyannote/speaker-diarization-3.1 once, then pass the token.
"""
Path(output_dir).mkdir(exist_ok=True)
cmd = [
'whisperx', audio_path,
'--model', 'large-v3',
'--output_format', 'json',
'--output_dir', output_dir,
'--language', 'en',
'--compute_type', 'int8', # CPU-friendly; use 'float16' on GPU
'--min_speakers', str(min_speakers),
'--max_speakers', str(max_speakers),
]
if diarize:
cmd.append('--diarize')
if hf_token:
cmd += ['--hf_token', hf_token]
subprocess.run(cmd, check=True, capture_output=True)
json_path = Path(output_dir) / f"{Path(audio_path).stem}.json"
with open(json_path) as f:
return json.load(f)
def format_for_editing(transcript: dict) -> str:
"""Convert to journalist-friendly format with timestamps."""
lines = []
for segment in transcript.get('segments', []):
timestamp = format_timestamp(segment['start'])
text = segment['text'].strip()
lines.append(f"[{timestamp}] {text}")
return '\n\n'.join(lines)
def format_timestamp(seconds: float) -> str:
"""Convert seconds to HH:MM:SS format."""
h = int(seconds // 3600)
m = int((seconds % 3600) // 60)
s = int(seconds % 60)
return f"{h:02d}:{m:02d}:{s:02d}"**Falling back to plain Whisper.** If diarization is overkill or you can't get a Hugging Face token, drop the `--diarize` flag — the model still produces accurate timestamped transcription and you label speakers manually based on context. `faster-whisper` (CTranslate2 backend) is the speed-optimized variant and works the same way at the CLI. `whisper.cpp` is the C++ port for resource-constrained machines (Raspberry Pi, older laptops); it doesn't include diarization but runs the small/medium models on CPU comfortably.
Manual transcription template
For sensitive interviews or when AI transcription fails:
## Transcript: [Source] - [Date] **Recording file**: [filename] **Duration**: [XX:XX] **Transcribed by**: [name] **Verified against recording**: [ ] Yes / [ ] No --- [00:00:15] **Q**: [Your question] [00:00:45] **A**: [Source response - verbatim, including ums, pauses noted as (...)] [00:01:30] **Q**: [Follow-up] [00:01:42] **A**: [Response] --- ## Notes - [Anything not captured in audio: gestures, documents shown, etc.] ## Potential quotes - [00:01:42] "Quote that stands out" - context: [why it matters]
Quote extraction and verification
Pull quotes workflow
from dataclasses import dataclass
from typing import Optional
import re
@dataclass
class Quote:
text: str
timestamp: str
speaker: str
context: str
verified: bool = False
used_in: Optional[str] = None
class QuoteBank:
"""Manage quotes from interview transcripts."""
def __init__(self):
self.quotes = []
def extract_quote(self,A collection of Agent Skills for journalists, researchers, academics, media professionals, and communications practitioners. The same repository serves Claude Code and Codex while keeping Claude-only commands, agents, and hooks clearly labeled.
Repo: jamditis/claude-skills-journalism
Other skills on claude-skills-journalism.
- /accessibility-compliance
Web accessibility patterns for news sites, journalism tools, and academic platforms. Use when building accessible interfaces, auditing existing sites for WCAG compliance, writing alt text for news images, creating accessible data visualizations, or ensuring content reaches all
Open skill - /claude-md-updater
Use this skill when the user asks to update CLAUDE.md, save a lesson, or persist something from the current session: phrases like "update claude.md", "what should we remember", "save this lesson", or "add to context". Scans the conversation for hard-won lessons, new file paths,
Open skill - /electron-dev
Electron desktop application development with React, TypeScript, and Vite. Use when building desktop apps, implementing IPC communication, managing windows/tray, handling PTY terminals, integrating WebRTC/audio, or packaging with electron-builder. Covers patterns from AudioBash,
Open skill - /mobile-debugging
Remote JavaScript console access and debugging on mobile devices. Use when debugging web pages on phones/tablets, accessing console errors without desktop DevTools, testing responsive designs on real devices, or diagnosing mobile-specific issues. Covers locally hosted Eruda and
Open skill - /one-way-door
Use this skill when creating new files that represent architectural decisions — data models, infrastructure configs, auth boundaries, API contracts, CI/CD pipelines, or event systems. Flags irreversible decisions and forces a discussion about trade-offs before committing.
Open skill - /python-pipeline
Python data processing pipelines with modular architecture. Use when building content processing workflows, implementing dispatcher patterns, integrating Google Sheets/Drive APIs, or creating batch processing systems. Covers patterns from rosen-scraper, image-analyzer, and
Open skill

