Commit Graph
8 Commits
Author SHA1 Message Date
Ben LenartsandXubin Ren 4cfc99f4b3 fix(transcription): resolve ${VAR} env refs in transcription api_key/api_base
config.loader.load_config() intentionally returns the raw config with ${VAR}
references intact — env interpolation is a separate, explicit step
(resolve_config_env_vars) so that settings read/edit/save paths never
materialize secrets to disk or to the UI.

The transcription config path does not apply that step: both
channels/base.py (channel voice notes) and webui/transcription_ws.py (WebUI
recording) build their effective config via
resolve_transcription_config(load_config()). As a result a configured
api_key of "${GROQ_API_KEY}" (the documented way to reference secrets) is
passed to the provider verbatim, which fails with 401 Invalid API Key. No
amount of rotating the real key helps, because the literal placeholder
string is what gets sent.

Resolve the reference at the single choke point both callers share —
_resolve_transcription_api_key / _resolve_transcription_api_base — using a
new lenient loader.resolve_env_refs() helper (unset var -> empty string, so
a missing variable degrades to "not configured" rather than raising or
leaking). This fixes both entry points at once and cannot drift the way a
per-call-site fix does. Resolving inside load_config() was rejected: the
~20 settings-UI callers depend on it returning raw ${VAR} placeholders.

Literal keys are unaffected; the settings API only reads the derived
`configured` flag (never the key), which now reflects the resolved value.

Claude-Session: https://claude.ai/code/session_01Q3HuVaJAAQJA3kgVQVJ2Zt
2026-07-21 17:35:16 +08:00
Xubin Ren b8a4ceb30c test(webui): cover siliconflow transcription settings 2026-06-10 23:05:12 +08:00
moranandXubin Ren 9ed638ad70 feat(transcription): add SiliconFlow as transcription provider
- Register SiliconFlow in transcription registry with default model
  FunAudioLLM/SenseVoiceSmall and alias 'silicon'
- Reuse existing OpenAITranscriptionProvider adapter (Whisper-compatible)
- Add generic key/base resolution: fallback to registry env_key and
  default_api_base when provider config is absent
- Add tests for registry entry, alias, adapter, default model, and
  config resolution with env var fallback
2026-06-10 23:05:12 +08:00
moranandXubin Ren 7930058348 feat(asr): add StepFun ASR SSE transcription provider
- Add StepFunTranscriptionProvider class in nanobot/providers/transcription.py
- New _post_stepfun_asr_with_retry() function handling SSE stream parsing
  (transcript.text.delta → transcript.text.done event sequence)
- Register 'stepfun' in transcription_registry.py with default model stepaudio-2.5-asr
- Reuse existing stepfun provider config (apiBase can point to Plan endpoint)
- Add 17 tests covering SSE parsing, retry contract, empty-text edge case, and registry integration
- Update docs/configuration.md with stepfun ASR documentation

StepFun ASR uses a dedicated SSE endpoint (/v1/audio/asr/sse) rather
than the chat-completions or Whisper multipart formats used by other
providers. Users on Step Plan can set apiBase to the Plan endpoint.
2026-06-10 15:50:38 +08:00
comadrejaandXubin Ren f3eb2aa08b feat(transcription): add AssemblyAI as transcription provider
Add AssemblyAI as a third transcription provider option alongside
OpenAI and Groq. AssemblyAI offers better accuracy for certain
audio types (distant voices, noisy environments) and serves as a
reliable fallback when other providers struggle.

Changes:
- Add AssemblyAITranscriptionProvider class in providers/transcription.py
- Add 'assemblyai' option in base channel's transcribe_audio()
- Per-channel configuration via transcriptionProvider in config

Usage:
  Set transcriptionProvider: 'assemblyai' and provide an AssemblyAI
  API key via transcriptionApiKey in the channel config.
2026-06-09 05:33:18 +08:00
c20ecc52d7 feat(transcription): add Xiaomi MiMo ASR provider (mimo-v2.5-asr)
Add support for Xiaomi MiMo ASR as a third transcription backend alongside
Groq and OpenAI Whisper. Xiaomi ASR uses the /v1/chat/completions endpoint
with base64-encoded audio input, rather than the standard Whisper multipart
upload format.

Co-Authored-By:连 <lian@tangping.homes>
2026-06-09 04:29:09 +08:00
0eb3010e40 feat(transcription): configurable STT model + OpenRouter provider
Add a `transcriptionModel` channel setting and an OpenRouter transcription
backend so voice messages can be transcribed through OpenRouter's
speech-to-text endpoint (e.g. nvidia/parakeet-tdt-0.6b-v3, openai/whisper-1),
alongside the existing Groq/OpenAI Whisper providers.

- schema: add channels.transcriptionModel (None = provider default)
- providers/transcription: extract a shared POST/retry skeleton; add a
  JSON+base64 OpenRouterTranscriptionProvider; make the STT model a
  constructor param on all providers instead of hardcoding it
- channels: route transcriptionProvider="openrouter" and thread the model
  through the manager to each channel
- docs + tests

Only dedicated STT models work on OpenRouter's transcription endpoint;
chat LLMs (e.g. google/gemini-3.5-flash) are rejected there.

Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
2026-06-09 04:01:37 +08:00
Xubin RenandGitHub 9c81280300 feat(transcription): add shared voice input support (#4232)
* feat(webui): add voice transcription input

* feat(webui): render ANSI output in code blocks

* refactor(webui): isolate voice recorder logic

* refactor(transcription): keep websocket ingress thin

* refactor(transcription): resolve channel audio settings on demand

* style(webui): neutralize voice waveform color

* feat(webui): add voice input tooltip

* feat(webui): add voice input keyboard shortcut

* fix(webui): distinguish voice shortcut platforms

* fix(webui): place voice button after model selector

* refactor(webui): share voice hold recording helpers

* fix(desktop): allow microphone voice input

* fix(webui): stabilize token usage month labels

* feat(webui): show voice input on settings overview

* fix(webui): label voice capability as recognition

* fix(webui): align capability overview status

* refactor(webui): isolate transcription socket handling

* fix(webui): soften silent voice waveform

* refactor(audio): clarify transcription service location

* docs(transcription): clarify audio and provider boundaries

* fix(exec): reduce session output polling flake
2026-06-09 01:08:49 +08:00