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text-to-speech

Practical mastering steps for TTS audio: cleanup, loudness normalization, alignment, and delivery specs.

Skill by benchflow-ai

IndexedApache-2.0★ 1,870
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Works with Claude Code, Claude apps, Codex, Cursor, Gemini CLI, GitHub Copilot, VS Code, OpenCode, Goose, Windsurf. Read the repository before you give any tool access to your data.

npx agentmarket install skill/benchflow-ai-text-to-speech

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About

Practical mastering steps for TTS audio: cleanup, loudness normalization, alignment, and delivery specs.

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What this skill can do and what it asks for, from its SKILL.md. Checked Oct 25, 2026.

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  • codeload.github.com · found in install command

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  • ~/.claude/skills/text-to-speech

What's inside

Skill name
text-to-speech
Description
Practical mastering steps for TTS audio: cleanup, loudness normalization, alignment, and delivery specs.
License
Apache-2.0
Instructions
58 lines in SKILL.md · view on GitHub

Files (1)

  • SKILL.md

SKILL.md (first lines)

# SKILL: TTS Audio Mastering

This skill focuses on producing clean, consistent, and delivery-ready TTS audio for video tasks. It covers speech cleanup, loudness normalization, segment boundaries, and export specs.

## 1. TTS Engine & Output Basics

Choose a TTS engine based on deployment constraints and quality needs:

* **Neural offline** (e.g., Kokoro): stable, high quality, no network dependency.
* **Cloud TTS** (e.g., Edge-TTS / OpenAI TTS): convenient, higher naturalness but network-dependent.
* **Formant TTS** (e.g., espeak-ng): for prototyping only; often less natural.

**Key rule:** Always confirm the **native sample rate** of the generated audio before resampling for video delivery.

---

## 2. Speech Cleanup (Per Segment)

Apply lightweight processing to avoid common artifacts:

* **Rumble/DC removal:** high-pass filter around **20 Hz**
* **Harshness control:** optional low-pass around **16 kHz** (helps remove digital fizz)
* **Click/pop prevention:** short fades at boundaries (e.g., **50 ms** fade-in and fade-out)

Recommended FFmpeg pattern (example):

* Add filters in a single chain, and keep them consistent across segments.

---

## 3. Loudness Normalization

Target loudness depends on the benchmark/task spec. A common target is ITU-R BS.1770 loudness measurement:

* **Integrated loudness:** **-23 LUFS**
* **True peak:** around **-1.5 dBTP**
* **LRA:** around **11** (optional)

Recommended workflow:

1. **Measure loudness** using FFmpeg `ebur128` (or equivalent meter).
2. **Apply normalization** (e.g., `loudnorm`) as the final step after cleanup and timing edits.
3. If you adjust tempo/duration after normalization, re-normalize again.

---

## 4. Timing & Segment Boundary Handling

When stitching segment-level TTS into a full track:

* Match each segment to its target window as closely as possible.
* If a segment is shorter than its window, pad with silence.
* If a segment is longer, use gentle duration control (small speed change) or truncate carefully.
* Always apply boundary fades after padding/trimming to avoid clicks.

**Sync guideline:** keep end-to-end drift small (e.g., **<= 0.2s**) unless the task states otherwise.

From benchflow-ai/skillsbench, Apache-2.0.

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Forks
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Last commit (this folder)
Jun 5, 2026 (4 months ago)
Last release
Jun 14, 2026 (4 months ago)

From GitHub (fetched Oct 24, 2026).

Details

License
Apache-2.0
First seen
Oct 24, 2026
Last checked
Oct 24, 2026

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text-to-speech skill. Agent Mart. https://agentmart.dev/skills/benchflow-ai-text-to-speech (last updated 2026-10-24)