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fix(tts): encode normalized audio at VBR q2 instead of 64k to reduce fatigue

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The wav->mp3 normalization path re-encoded pristine high-fidelity TTS (e.g.
Supertonic's 44.1 kHz wav) at 64k CBR, which lowpasses hard and introduces
high-frequency artifacts that cause listening fatigue. Switch the live/segment
transcode to libmp3lame VBR -q:a 2 (~170-210 kbps for speech), effectively
transparent. The audiobook export still re-encodes to 64k for file size.
This commit is contained in:
Richard R 2026-06-09 09:52:37 -06:00
parent 5a0b40658a
commit f09d14549a
1 changed files with 7 additions and 2 deletions
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9
src/lib/server/tts/audio-format.ts
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@ -109,7 +109,12 @@ function spawnFfmpegToBuffer(args: string[], signal?: AbortSignal): Promise<Buff
/**
* Transcode an arbitrary audio buffer to mp3 using the bundled ffmpeg. The input
* is written to a temp file (ffmpeg needs seekable input for some containers) and
* the mp3 is captured from stdout. The 64k bitrate matches the audiobook pipeline.
* the mp3 is captured from stdout.
*
* Uses VBR quality 2 (~170-210 kbps) rather than a low CBR bitrate: this audio is
* what the user listens to live, and aggressive low-bitrate encoding of pristine
* high-fidelity TTS (e.g. 44.1 kHz wav) introduces high-frequency artifacts that
* cause listening fatigue. The audiobook export still re-encodes to 64k for size.
*/
export async function transcodeToMp3(buffer: Buffer, signal?: AbortSignal): Promise<Buffer> {
let workDir: string | null = null;
@ -125,7 +130,7 @@ export async function transcodeToMp3(buffer: Buffer, signal?: AbortSignal): Prom
'-i', inputPath,
'-vn',
'-c:a', 'libmp3lame',
'-b:a', '64k',
'-q:a', '2',
'-f', 'mp3',
'pipe:1',
],