NeverSight/learn-skills.dev

When this skill is activated, always start your first response with the :speaker: emoji.

Video Audio Design

Video audio design is the practice of layering narration, sound effects, and background music into programmatic video compositions. Great audio transforms a slide-deck video into a polished production - narration guides the viewer, music sets the emotional tone, and SFX punctuate key moments. This skill covers generating speech with ElevenLabs and alternative TTS providers, creating synthetic sound effects with FFmpeg, sourcing royalty-free background music, implementing audio ducking so speech stays intelligible, and mixing all layers together in Remotion compositions with frame-accurate timing.

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When to use this skill

Trigger this skill when the user:

• Wants to add narration or voiceover to a programmatic video
• Needs to generate speech with ElevenLabs, OpenAI TTS, or Edge TTS
• Asks about voice selection, voice settings, or voice cloning
• Wants to add background music or needs royalty-free music sources
• Asks about creating sound effects programmatically
• Wants to implement audio ducking (lowering music during speech)
• Needs to mix multiple audio layers in Remotion
• Asks about audio timing, volume levels, or frame-based audio sync

Do NOT trigger this skill for:

• Video scripting or storyboarding - use the video-scriptwriting skill
• Remotion component architecture or rendering - use the remotion-video skill
• Professional audio production in a DAW (Ableton, Logic, Pro Tools)
• Music composition or MIDI programming

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Key principles

1. Layered audio architecture - Every video has three audio layers: narration on top (loudest), SFX in the middle (accent volume), and background music at the base (lowest).

2. Narration drives timing - Generate narration first, measure its duration, then set scene timing to match. Never fit narration into arbitrary scene lengths.

3. Duck music during speech - Background music must drop 50-60% when narration plays. Use smooth ramps (10-15 frames) to avoid jarring jumps.

4. SFX as accents, not distractions - Keep SFX short (under 0.5s), subtle in volume, and relevant to on-screen action.

5. Test audio in context - Always preview the full mix with all layers together. Listen for muddy speech, volume spikes, or dead silence.

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Core concepts

3-layer audio architecture

Layer	Role	Base Volume	During Narration
Narration	Conveys information, drives pacing	0.8-1.0	N/A (top layer)
SFX	Accents transitions and actions	0.3-0.5	0.3-0.5 (unchanged)
Background Music	Sets emotional tone, fills silence	0.3-0.5	0.15-0.25 (ducked)

ElevenLabs API model

ElevenLabs provides neural TTS via a REST API. The core flow:

1. Pick a voice (pre-made or cloned) - each has a voice_id
2. Send text + voice settings to /v1/text-to-speech/{voice_id}
3. Receive raw audio bytes (mp3 by default)
4. Write to file and measure duration for scene timing

Voice settings:

Setting	Range	Low	High	Recommended
stability	0-1	More expressive, variable	More consistent, monotone	0.4-0.6
similarity_boost	0-1	More creative	Closer to original voice	0.6-0.8
style	0-1	Neutral delivery	Exaggerated style	0.3-0.6

Audio ducking concept

Audio ducking reduces background music volume when narration starts and restores it when narration ends. In Remotion, use interpolate():

Music volume:  0.4 ---\              /--- 0.4
                       \            /
               0.15     \__________/
                     narration start → end

Ramps should take 10-15 frames (~0.3-0.5s at 30fps).

Frame-based audio sync in Remotion

• useCurrentFrame() returns the current frame number
• interpolate() maps frame ranges to value ranges (e.g., volume)
• <Sequence from={frame}> places audio at a specific frame
• <Audio volume={fn}> accepts a static number or a per-frame function

Convert seconds to frames: frames = seconds * fps.

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Common tasks

1. Set up ElevenLabs API key and generate narration

import fs from 'fs';

const ELEVENLABS_API_URL = 'https://api.elevenlabs.io/v1';

async function generateNarration(
  text: string,
  voiceId: string,
  outputPath: string
): Promise<void> {
  const response = await fetch(
    `${ELEVENLABS_API_URL}/text-to-speech/${voiceId}`,
    {
      method: 'POST',
      headers: {
        'Content-Type': 'application/json',
        'xi-api-key': process.env.ELEVEN_LABS_API_KEY!,
      },
      body: JSON.stringify({
        text,
        model_id: 'eleven_multilingual_v2',
        voice_settings: {
          stability: 0.5,
          similarity_boost: 0.75,
          style: 0.5,
          use_speaker_boost: true,
        },
      }),
    }
  );

  if (!response.ok) {
    const error = await response.text();
    throw new Error(`ElevenLabs API error ${response.status}: ${error}`);
  }

  const buffer = Buffer.from(await response.arrayBuffer());
  fs.writeFileSync(outputPath, buffer);
}

2. Select and configure voice settings

Voice selection questions: gender, age range, accent, energy level, warmth.

interface VoiceSettings {
  stability: number;
  similarity_boost: number;
  style: number;
  use_speaker_boost: boolean;
}

const presets: Record<string, VoiceSettings> = {
  explainer: { stability: 0.6, similarity_boost: 0.75, style: 0.4, use_speaker_boost: true },
  promo: { stability: 0.3, similarity_boost: 0.7, style: 0.7, use_speaker_boost: true },
  tutorial: { stability: 0.7, similarity_boost: 0.8, style: 0.2, use_speaker_boost: false },
};

3. Generate narration per scene from a script

import { execSync } from 'child_process';
import path from 'path';

interface Scene { id: string; narrationText: string; }
interface SceneWithAudio extends Scene {
  audioPath: string;
  durationMs: number;
  durationFrames: number;
}

function getAudioDurationMs(filePath: string): number {
  const output = execSync(
    `ffprobe -v error -show_entries format=duration -of csv=p=0 "${filePath}"`
  ).toString().trim();
  return Math.round(parseFloat(output) * 1000);
}

async function generateSceneNarrations(
  scenes: Scene[], voiceId: string, outputDir: string, fps: number
): Promise<SceneWithAudio[]> {
  const results: SceneWithAudio[] = [];
  for (const scene of scenes) {
    const audioPath = path.join(outputDir, `${scene.id}.mp3`);
    await generateNarration(scene.narrationText, voiceId, audioPath);
    const durationMs = getAudioDurationMs(audioPath);
    results.push({
      ...scene, audioPath, durationMs,
      durationFrames: Math.ceil((durationMs / 1000) * fps),
    });
  }
  return results;
}

4. Source background music

Royalty-free music sources:

• Pixabay Audio: https://pixabay.com/music/ (free, no attribution)
• Freesound: https://freesound.org/ (CC0/CC-BY)
• YouTube Audio Library: download from YouTube Studio
• Local files: place in public/audio/ for Remotion's staticFile()

5. Generate SFX with FFmpeg

# Click sound - short sine burst
ffmpeg -f lavfi -i "sine=frequency=800:duration=0.05" \
  -af "afade=t=out:st=0.02:d=0.03" click.wav

# Keyboard typing - filtered noise burst
ffmpeg -f lavfi -i "anoisesrc=d=0.08:c=white:a=0.3" \
  -af "highpass=f=2000,lowpass=f=8000,afade=t=out:st=0.04:d=0.04" type.wav

# Whoosh - frequency sweep
ffmpeg -f lavfi -i "sine=frequency=200:duration=0.4" \
  -af "vibrato=f=8:d=0.5,afade=t=in:d=0.1,afade=t=out:st=0.2:d=0.2,lowpass=f=1000" \
  whoosh.wav

# Ding/chime - bell synthesis
ffmpeg -f lavfi -i "sine=frequency=1200:duration=0.6" \
  -af "afade=t=out:st=0.1:d=0.5,aecho=0.8:0.88:40:0.4" ding.wav

# Pop - impulse
ffmpeg -f lavfi -i "sine=frequency=400:duration=0.08" \
  -af "afade=t=out:st=0.02:d=0.06,lowpass=f=600" pop.wav

# Transition swoosh
ffmpeg -f lavfi -i "sine=frequency=300:duration=0.3" \
  -af "vibrato=f=12:d=0.8,afade=t=in:d=0.05,afade=t=out:st=0.15:d=0.15,bandpass=f=500:w=400" \
  swoosh.wav

6. Implement audio ducking in Remotion

import React from 'react';
import { Audio, useCurrentFrame, interpolate, Sequence } from 'remotion';

const AudioMixer: React.FC<{
  narrationSrc: string;
  musicSrc: string;
  narrationStart: number;
  narrationDuration: number;
}> = ({ narrationSrc, musicSrc, narrationStart, narrationDuration }) => {
  const frame = useCurrentFrame();

  const duckRampFrames = 10;
  const musicVolume = interpolate(
    frame,
    [
      narrationStart - duckRampFrames,
      narrationStart,
      narrationStart + narrationDuration,
      narrationStart + narrationDuration + duckRampFrames,
    ],
    [0.4, 0.15, 0.15, 0.4],
    { extrapolateLeft: 'clamp', extrapolateRight: 'clamp' }
  );

  return (
    <>
      <Audio src={musicSrc} volume={musicVolume} />
      <Sequence from={narrationStart} durationInFrames={narrationDuration}>
        <Audio src={narrationSrc} volume={0.9} />
      </Sequence>
    </>
  );
};

export default AudioMixer;

7. Mix 3 audio layers in a Remotion composition

import React from 'react';
import { Audio, Sequence, useCurrentFrame, interpolate } from 'remotion';

interface NarrationSegment { src: string; startFrame: number; durationFrames: number; }
interface SfxEvent { src: string; frame: number; }

const FullAudioMix: React.FC<{
  narrations: NarrationSegment[];
  sfxEvents: SfxEvent[];
  musicSrc: string;
}> = ({ narrations, sfxEvents, musicSrc }) => {
  const frame = useCurrentFrame();
  const duckRamp = 10;

  let musicVolume = 0.4;
  for (const seg of narrations) {
    const duck = interpolate(
      frame,
      [seg.startFrame - duckRamp, seg.startFrame,
       seg.startFrame + seg.durationFrames, seg.startFrame + seg.durationFrames + duckRamp],
      [1, 0.375, 0.375, 1],
      { extrapolateLeft: 'clamp', extrapolateRight: 'clamp' }
    );
    musicVolume = musicVolume * duck;
  }

  return (
    <>
      <Audio src={musicSrc} volume={musicVolume} loop />
      {sfxEvents.map((sfx, i) => (
        <Sequence key={i} from={sfx.frame} durationInFrames={30}>
          <Audio src={sfx.src} volume={0.4} />
        </Sequence>
      ))}
      {narrations.map((seg, i) => (
        <Sequence key={i} from={seg.startFrame} durationInFrames={seg.durationFrames}>
          <Audio src={seg.src} volume={0.9} />
        </Sequence>
      ))}
    </>
  );
};

export default FullAudioMix;

8. Use alternative TTS providers

OpenAI TTS - good quality, simple API, six built-in voices:

import OpenAI from 'openai';
import fs from 'fs';

const openai = new OpenAI();

async function generateWithOpenAI(
  text: string,
  outputPath: string,
  voice: 'alloy' | 'echo' | 'fable' | 'onyx' | 'nova' | 'shimmer' = 'alloy'
): Promise<void> {
  const mp3 = await openai.audio.speech.create({
    model: 'tts-1-hd',
    voice,
    input: text,
  });
  const buffer = Buffer.from(await mp3.arrayBuffer());
  fs.writeFileSync(outputPath, buffer);
}

Edge TTS - free, many voices, uses Microsoft Edge's TTS service:

pip install edge-tts
edge-tts --voice en-US-AriaNeural --text "Hello world" --write-media output.mp3
edge-tts --list-voices

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Anti-patterns / common mistakes

Mistake	Why it is wrong	What to do instead
Music same volume during narration	Speech becomes unintelligible	Implement audio ducking - drop music 50-60% during speech
Hardcoding ElevenLabs API key	Key leaks into version control	Use environment variables: process.env.ELEVEN_LABS_API_KEY
Using TTS without measuring duration	Scene timing wrong, narration cut off	Measure audio duration with ffprobe after generation
SFX louder than narration	Distracts from content	SFX at 0.3-0.5, narration at 0.8-1.0
No fade on music start/end	Abrupt start/stop sounds like a bug	Add 0.5-1s fade-in at start and fade-out at end
Using low-quality TTS model	Robotic voice undermines quality	Use eleven_multilingual_v2 or tts-1-hd
Ignoring audio file format	Some formats add silence padding	Use MP3 for narration, WAV for SFX

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Gotchas

1. ElevenLabs rate limits and character quotas - The free tier has a monthly character limit. Cache generated audio aggressively and only regenerate when text changes. Use a hash of the text as the cache key.

2. MP3 encoder padding adds silence - MP3 files often have 20-50ms of silence at the start. Trim with ffmpeg -af silenceremove=1:0:-50dB or account for the offset in frame timing.

3. Remotion Audio volume is per-component, not global - Two <Audio> components at volume 1.0 can clip. Keep total volume across simultaneous layers under 1.0.

4. FFmpeg SFX sound different across systems - Always specify -ar 44100 -sample_fmt s16 for consistent output across machines.

5. Voice consistency across scenes - ElevenLabs can produce different tones for the same settings with varying text. Use stability >= 0.5 for multi-scene narration.

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References

For detailed patterns on specific audio sub-domains, read the relevant file from the references/ folder:

• references/elevenlabs-api.md - advanced ElevenLabs API patterns including voice cloning, streaming TTS, websocket API, pronunciation dictionaries, and quota management
• references/audio-mixing-patterns.md - advanced mixing patterns including multi-segment ducking, crossfades between scenes, volume automation curves, and mastering the final mix
• references/sfx-generation.md - comprehensive SFX generation with FFmpeg including complex synthesis, layering multiple generators, and building a reusable SFX library

Only load a references file if the current task requires it - they are long and will consume context.

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Companion check

On first activation of this skill in a conversation: check which companion skills are installed by running ls ~/.claude/skills/ ~/.agent/skills/ ~/.agents/skills/ .claude/skills/ .agent/skills/ .agents/skills/ 2>/dev/null. Compare the results against the recommended_skills field in this file's frontmatter. For any that are missing, mention them once and offer to install:

npx skills add AbsolutelySkilled/AbsolutelySkilled --skill <name>

Skip entirely if recommended_skills is empty or all companions are already installed.