Desktop STT transcription timeout too short

NousResearch/hermes-agent · cluster 39448 · apps/desktop

Data as of

Nothing open on GitHub. Fix merged in #68056. Check it covers your case.

The short answer

0open issues
0open PRs
6items grouped
Known?
Yes. 6 related items grouped here, 2 issues already closed.
Fix event
A fix was merged to main in PR #68056. Check it covers your case. No release data is tracked.
Tracker
The maintainers' tracker marks this unresolved. That is separate from the fix event.
Still open
Nothing open on GitHub.

Multiple issues and fixes address the desktop app's audio transcription timeout being too short (defaulting to ~15s), causing local speech-to-text to fail on longer audio. Proposed solutions include overriding, extending, or making the timeout configurable.

Open issues0

No open issues.

Open PRs0

No open pull requests.

Fixes on record

Merged to main. A stamp counts the issues a fix is linked to.

  • #68056 fix(desktop): extend read aloud + transcription timeouts (salvage of #39286)

    merged to main

    files: apps/desktop/src/hermes.test.ts, apps/desktop/src/hermes.ts

    Show the diff
    • apps/desktop/src/hermes.test.ts
      @@ -1,6 +1,12 @@
       import { afterEach, beforeEach, describe, expect, it, vi } from 'vitest'
       
       import {
      +  AUDIO_SPEAK_MAX_REQUEST_TIMEOUT_MS,
      +  AUDIO_SPEAK_MIN_REQUEST_TIMEOUT_MS,
      +  AUDIO_TRANSCRIBE_MAX_REQUEST_TIMEOUT_MS,
      +  AUDIO_TRANSCRIBE_MIN_REQUEST_TIMEOUT_MS,
      +  audioSpeakRequestTimeoutMs,
      +  audioTranscribeRequestTimeoutMs,
         getCronJobs,
         getGlobalModelInfo,
         getGlobalModelOptions,
      @@ -12,7 +18,9 @@ import {
         listAllProfileSessions,
         listSessions,
         listSidebarSessions,
      -  resetSidebarBatchCapability
      +  resetSidebarBatchCapability,
      +  speakText,
      +  transcribeAudio
       } from './hermes'
       import { refreshActiveProfile } from './store/profile'
       
      @@ -23,7 +31,7 @@ const emptySessionsResponse = {
         total: 0
       }
       
      -describe('Hermes REST session helpers', () => {
      +describe('Hermes REST helpers', () => {
         let api: ReturnType<typeof vi.fn>
       
         beforeEach(() => {
      @@ -322,6 +330,62 @@ describe('Hermes REST session helpers', () => {
           })
         })
       
      +  it('bounds blocking TTS synthesis timeouts by text length', () => {
      +    expect(audioSpeakRequestTimeoutMs('short message')).toBe(AUDIO_SPEAK_MIN_REQUEST_TIMEOUT_MS)
      +    expect(audioSpeakRequestTimeoutMs('x'.repeat(8_000))).toBe(280_000)
      +    expect(audioSpeakRequestTimeoutMs('x'.repeat(100_000))).toBe(AUDIO_SPEAK_MAX_REQUEST_TIMEOUT_MS)
      +  })
      +
      +  it('uses an extended timeout for blocking TTS synthesis', async () => {
      +    api.mockResolvedValueOnce({
      +      data_url: 'data:audio/mpeg;base64,AA==',
      +      mime_type: 'audio/mpeg',
      +      ok: true,
      +      provider: 'openai'
      +    })
      +
      +    await expect(speakText('Read this aloud')).resolves.toEqual({
      +      data_url: 'data:audio/mpeg;base64,AA==',
      +      mime_type: 'audio/mpeg',
      +      ok: true,
      +      provider: 'openai'
      +    })
      +
      +    expect(api).toHaveBeenCalledWith({
      +      body: { text: 'Read this aloud' },
      +      method: 'POST',
      +      path: '/api/audio/speak',
      +      timeoutMs: AUDIO_SPEAK_MIN_REQUEST_TIMEOUT_MS
      +    })
      +  })
      +
      +  it('bounds blocking transcription timeouts by payload length', () => {
      +    expect(audioTranscribeRequestTimeoutMs('data:audio/webm;base64,AA==')).toBe(AUDIO_TRANSCRIBE_MIN_REQUEST_TIMEOUT_MS)
      +    expect(audioTranscribeRequestTimeoutMs('x'.repeat(3_000_000))).toBe(300_000)
      +    expect(audioTranscribeRequestTimeoutMs('x'.repeat(9_000_000))).toBe(AUDIO_TRANSCRIBE_MAX_REQUEST_TIMEOUT_MS)
      +  })
      +
      +  it('uses an extended timeout for blocking transcription', async () => {
      +    api.mockResolvedValueOnce({
      +      ok: true,
      +      provider: 'openai',
      +      text: 'transcribed text'
      +    })
      +
      +    await expect(transcribeAudio('data:audio/webm;base64,AA==', 'audio/webm')).resolves.toEqual({
      +      ok: true,
      +      provider: 'openai',
      +      text: 'transcribed text'
      +    })
      +
      +    expect(api).toHaveBeenCalledWith({
      +      body: { data_url: 'data:audio/webm;base64,AA==', mime_type: 'audio/webm' },
      +      method: 'POST',
      +      path: '/api/audio/transcribe',
      +      timeoutMs: AUDIO_TRANSCRIBE_MIN_REQUEST_TIMEOUT_MS
      +    })
      +  })
      +
         it('defaults model options to configured providers only', async () => {
           await getGlobalModelOptions()
       
    • apps/desktop/src/hermes.ts
      @@ -84,6 +84,36 @@ const SESSION_LIST_REQUEST_TIMEOUT_MS = 60_000
       // agent-turn ceiling (agent.gateway_timeout = 1800s) so the ack timeout only
       // ever fires when the turn itself would have been abandoned server-side.
       export const PROMPT_SUBMIT_REQUEST_TIMEOUT_MS = 1_800_000
      +export const AUDIO_SPEAK_MIN_REQUEST_TIMEOUT_MS = 180_000
      +export const AUDIO_SPEAK_MAX_REQUEST_TIMEOUT_MS = 600_000
      +const AUDIO_SPEAK_TIMEOUT_MS_PER_CHAR = 35
      +
      +export function audioSpeakRequestTimeoutMs(text: string): number {
      +  const estimated = Math.max(
      +    AUDIO_SPEAK_MIN_REQUEST_TIMEOUT_MS,
      +    Math.ceil(String(text || '').length * AUDIO_SPEAK_TIMEOUT_MS_PER_CHAR)
      +  )
      +
      +  return Math.min(AUDIO_SPEAK_MAX_REQUEST_TIMEOUT_MS, estimated)
      +}
      +
      +export const AUDIO_TRANSCRIBE_MIN_REQUEST_TIMEOUT_MS = 180_000
      +export const AUDIO_TRANSCRIBE_MAX_REQUEST_TIMEOUT_MS = 600_000
      +// The transcribe payload is the base64 audio data URL itself, so its string
      +// length tracks clip size. ~0.1ms/char keeps short clips at the floor while
      +// letting multi-minute recordings scale toward the cap (a base64 char is
      +// ~0.75 bytes, so at 128kbps ≈ 21k chars/s of audio this budgets ~2s of
      +// timeout per 1s of audio before the cap clamps it).
      +const AUDIO_TRANSCRIBE_TIMEOUT_MS_PER_CHAR = 0.1
      +
      +export function audioTranscribeRequestTimeoutMs(dataUrl: string): number {
      +  const estimated = Math.max(
      +    AUDIO_TRANSCRIBE_MIN_REQUEST_TIMEOUT_MS,
      +    Math.ceil(String(dataUrl || '').length * AUDIO_TRANSCRIBE_TIMEOUT_MS_PER_CHAR)
      +  )
      +
      +  return Math.min(AUDIO_TRANSCRIBE_MAX_REQUEST_TIMEOUT_MS, estimated)
      +}
       
       export type {
         ActionResponse,
      @@ -1346,7 +1376,11 @@ export function transcribeAudio(dataUrl: string, mimeType?: string): Promise<Aud
           body: {
             data_url: dataUrl,
             mime_type: mimeType
      -    }
      +    },
      +    // Transcription blocks until provider STT, file handling, and response
      +    // encoding finish. Remote providers and long clips regularly exceed the
      +    // default 15s Electron backend timeout.
      +    timeoutMs: audioTranscribeRequestTimeoutMs(dataUrl)
         })
       }
       
      @@ -1354,7 +1388,11 @@ export function speakText(text: string): Promise<AudioSpeakResponse> {
         return window.hermesDesktop.api<AudioSpeakResponse>({
           path: '/api/audio/speak',
           method: 'POST',
      -    body: { text }
      +    body: { text },
      +    // TTS blocks until provider synthesis, file read, and base64 encoding
      +    // finish. Remote providers and large messages regularly exceed the
      +    // default 15s Electron backend timeout.
      +    timeoutMs: audioSpeakRequestTimeoutMs(text)
         })
       }
       

3 PRs closed without merging

From the close comments: done elsewhere 1 · duplicate 1 · superseded by another PR 1

  • #46579 fix(desktop): override request timeout for transcribeAudio to prevent local STT timeout · superseded by another PR → #46530
  • #46740 fix(desktop): add timeoutMs to transcribeAudio for local STT models · done elsewhere → 2bb531b, #68056
  • #51862 fix(desktop): extend audio transcription timeout · duplicate → #46530, #46636
Closed issues · 2
  • Closed#39448[Feature Request] Configurable timeout for STT transcription (currently hardcoded 15s in Desktop)
  • Closed#46573Desktop: transcribeAudio missing timeoutMs — local STT fails for audio >15s
How this cluster was grouped
  • Link evidence (partly structural links): some members are connected by structural evidence (shared changed files, or a closing/superseded reference); the rest are joined by an automated relevance judgement over their text.
  • Members outside that structural core may belong to a different underlying problem.
  • Pairwise link density 0.29: the share of item pairs that carry a link, counting a structural link in full and a relevance-only link at 0.36. 1.00 means every pair is linked.

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At a glance

Subsystem
apps/desktop
Merged PRs
#68056
Closed issues
2
Attempts
3 PRs closed without merging
Fix link checked
20 Jul 2026

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