Audio Transcription Pipeline

Audio Transcription Pipeline

Build production speech-to-text pipelines with Whisper, Deepgram, and AssemblyAI for batch and real-time transcription with speaker diarization.

Decision Points

Engine Selection Decision Tree

If requirements include:
├─ Real-time streaming required?
│  ├─ Yes: Use Deepgram Nova-3 WebSocket (fastest)
│  └─ No: Continue to accuracy requirements
├─ Highest accuracy needed + have GPU?
│  ├─ Yes: Use Whisper large-v3 with faster-whisper
│  └─ No: Continue to cost analysis
├─ Budget < $0.0059/min AND have compute?
│  ├─ Yes: Use local Whisper
│  └─ No: Use AssemblyAI Universal-2 (best API diarization)

Batch vs Stream Processing

If audio characteristics:
├─ Duration > 30 minutes?
│  ├─ Yes: Use batch with chunking (split at silence)
│  └─ No: Continue to latency check
├─ Need results in < 10 seconds?
│  ├─ Yes: Use streaming (Deepgram/Whisper.cpp)
│  └─ No: Use batch for better accuracy

VAD (Voice Activity Detection) Usage

If content type:
├─ Live conversation/meeting?
│  ├─ Yes: Enable VAD (saves 40-60% compute on silence)
│  └─ No: Continue to content check
├─ Lecture/presentation with pauses?
│  ├─ Yes: Use conservative VAD (min_silence_duration_ms: 1000)
│  └─ No: Skip VAD for dense speech (audiobooks, etc.)

Model Selection by Domain

If language/accent:
├─ Non-English or heavy accent?
│  ├─ Yes: Use Whisper large-v3 (best multilingual)
│  └─ No: Continue to speed check
├─ Need < 1 second latency?
│  ├─ Yes: Use Deepgram Nova-3 streaming
│  └─ No: Use AssemblyAI for business/medical terminology

Failure Modes

Hallucination on Silence

Detection: Transcripts show repeated phrases during quiet sections or phantom music descriptions Diagnosis: VAD disabled or threshold too low, model generating text from background noise Fix: Enable VAD with min_silence_duration_ms: 500, use --no_speech_threshold 0.6 in Whisper

Diarization Collapse at >10 Speakers

Detection: All speakers labeled as "Speaker 1" after 10-15 minutes, or random speaker switching mid-sentence Diagnosis: Speaker embedding model saturated, overlap confusion in crowded audio Fix: Pre-segment by silence, use AssemblyAI dual_channel if stereo, limit to 8 active speakers max

Timestamp Sync Drift with Video

Detection: Subtitles appear 2-5 seconds before/after corresponding video frames Diagnosis: Audio preprocessing changed duration, or VAD removed segments without timestamp adjustment Fix: Use --preserve_timing in preprocessing, sync with original audio timecode, validate against known speech events

Language Auto-Detection Failure

Detection: English words transcribed as gibberish when speaker has accent, or code-switching ignored Diagnosis: Model locked to wrong language in first 30 seconds, or multilingual content confused classifier Fix: Force language with language="en" for accented English, use task="translate" for non-English to English

Memory Overflow on Long Files

Detection: OOM crashes on files >45 minutes, or sudden quality drops after 30 minutes Diagnosis: Model keeping full context window, GPU VRAM exhausted Fix: Chunk at 25-minute boundaries with 30-second overlap, use --without_timestamps for RAM efficiency

Worked Examples

Complete Meeting Transcription Walkthrough

Scenario: 90-minute board meeting, 6 speakers, need accurate speaker identification and timestamps for minutes

Input Analysis:

• 90 minutes → requires chunking
• 6 speakers → within diarization limits
• Business context → use AssemblyAI for terminology

Decision Path:

1. Duration > 30 min → batch processing required
2. Speaker count = 6 → diarization feasible
3. Business domain → AssemblyAI Universal-2 chosen
4. High accuracy needed → enable speaker labels + smart formatting

Implementation:

# Step 1: Preprocess (expert catches: normalize audio levels)
ffmpeg -i meeting.mp4 -ar 16000 -ac 1 -filter:a "volume=0.8" meeting_clean.wav

# Step 2: Chunk with overlap (novice would process whole file)
from pydub import AudioSegment
audio = AudioSegment.from_wav("meeting_clean.wav")
chunks = []
for i in range(0, len(audio), 25*60*1000):  # 25min chunks
    chunk = audio[i:i+27*60*1000]  # 27min with 2min overlap
    chunks.append(chunk)

# Step 3: Transcribe with diarization
results = []
for chunk in chunks:
    response = assemblyai_client.transcribe(
        chunk, speaker_labels=True, auto_punctuation=True,
        dual_channel=False, speaker_labels_max=8
    )
    results.append(response.get_paragraphs())

# Step 4: Merge overlapping segments (expert step novice misses)
merged = merge_overlapping_transcripts(results, overlap_seconds=120)

Expert vs Novice:

• Novice: Processes 90-min file directly → OOM crash or poor accuracy
• Expert: Chunks with overlap, validates speaker consistency across chunks, normalizes audio first

Quality Gates

• WER (Word Error Rate) < 5% on test sample with known ground truth
• Speaker accuracy > 90% (correct speaker ID for each utterance when ground truth available)
• Real-time latency < 2 seconds end-to-end for streaming implementations
• SRT/VTT files pass validation (proper timestamps, no overlapping segments)
• Audio preprocessing completed: 16kHz mono WAV/FLAC format confirmed
• VAD parameters tuned: < 10% false positive silence detection on sample
• Memory usage < 8GB for files up to 2 hours (chunking working properly)
• API error handling tested: handles timeouts, retries, and quota exceeded
• Diarization speaker count matches expected (±1 speaker tolerance)
• Output timestamps align with original audio within 100ms accuracy

NOT-FOR Boundaries

This skill should NOT be used for:

• Text-to-speech synthesis → Use voice-audio-engineer instead
• Music transcription or lyric extraction → Use ai-engineer for music AI models
• Audio classification without transcription → Use ai-engineer for audio classification
• Real-time translation (transcribe + translate) → Combine with language-translator skill
• Voice biometrics or speaker identification → Use ai-engineer for speaker recognition models
• Audio quality enhancement or noise reduction → Use voice-audio-engineer for audio processing

Delegate when:

• Need custom wake word detection → voice-audio-engineer
• Require audio fingerprinting → data-pipeline-engineer
• Building voice assistants → voice-audio-engineer + conversation-designer
• Processing >10,000 hours → data-pipeline-engineer for orchestration