docs(compat): say that elevenlabs language_code now maps to the fish language field

developer-guide/compat/migrate-from-elevenlabs.mdx

@@ -1,7 +1,7 @@
 ---
 title: "Migrate from ElevenLabs"
 sidebarTitle: "From ElevenLabs"
 description: "Point the ElevenLabs SDK at Fish Audio: TTS, timestamps, realtime streaming input, speech-to-text, and your voice library."
 icon: "waveform-lines"
 ---
 
@@ -48,11 +48,11 @@
 | Feature                                 | Supported |
 | --------------------------------------- | --------- |
 | Text to speech (+ character timestamps) | ✓         |
 | Realtime input streaming (WebSocket)    | ✓         |
 | Speech to text + word timestamps        | ✓         |
 | Realtime STT (manual commit)            | ✓         |
 | Voice library & model catalog           | ✓         |
 | Diarization / redaction / webhooks      | –         |
 | μ-law / A-law output                    | –         |
 
 ## Text to speech
@@ -96,23 +96,25 @@
 `voice_settings` maps onto Fish's synthesis controls: `speed` sets the speech
 rate, and `stability` becomes Fish `temperature = 1 − stability`. The
 vendor-specific tuning knobs (`similarity_boost`, `style`,
-`use_speaker_boost`, `language_code`, `seed`) are accepted and ignored, never
-an error.
+`use_speaker_boost`, `seed`) are accepted and ignored, never an error.
+`language_code` is passed on as Fish's [`language`](/features/text-to-speech#language)
+field: it picks the text-normalization rules (how numbers, dates and symbols
+are spelled out) and the word-timestamp language, and does not change the voice.
 
 ### Output formats
 
 `mp3_*`, `pcm_*`, `opus_*`, and `wav_*` (a Fish Audio extension) work at the
 supported sample rates: 32000 or 44100 for `mp3_*`, 48000 for `opus_*`.
 The 22.05 kHz family (`mp3_22050_32`, `pcm_22050`, …) is refused. The
 sample-rate segment of the format name is honored; the `mp3_*` bitrate segment
 snaps **up** to the nearest Fish tier
 ([details](/developer-guide/compat/capabilities#bitrate-snapping)), while Opus
 bitrate is automatic. `ulaw_*` and `alaw_*` return a 400.
 
 ### Timestamps
 
 The `with-timestamps` variants return the official shapes: REST uses snake_case
 keys in **seconds**; the WebSocket uses camelCase keys in **milliseconds**
 (`charStartTimesMs`). That asymmetry matches ElevenLabs' own API. Character
 timings are interpolated from Fish segment timestamps at millisecond precision.
 
@@ -127,7 +129,7 @@
   a client concatenate every frame's audio and alignment and have the captions
   line up. Don't add your own per-turn offset; you would double-count it.
 
 ## Realtime input streaming (WebSocket)
 
 `wss://api.fish.audio/compat/elevenlabs/v1/text-to-speech/{voice_id}/stream-input`
 
@@ -193,7 +195,7 @@
   Both are quirks of the SDK itself.
 
   The same SDK force-upgrades `http://` base URLs to `wss://`, so it cannot be
   pointed at a plaintext endpoint, including a local test proxy.
 </Note>
 
 ## Speech to text
@@ -238,7 +240,7 @@
 
 - `language_probability` is always `1.0`.
 - `words[].logprob` is always `0`.
 - `words[].speaker_id` is always `null`; requesting diarization returns an
   explicit error instead.
 
 And `language_code` in the response **echoes what you sent**, never a
@@ -249,7 +251,8 @@
 Code branching on `language_probability > 0.9` will act on that empty string.
 
 (On the *synthesis* side `language_code` is a different story: there it is
-accepted and dropped, along with the other vendor tuning knobs.)
+passed on as Fish's `language` field, which selects the
+[normalization rules](/features/text-to-speech#supported-normalization-languages).)
 
 Options that would change the response contract are **rejected with 400**
 rather than silently ignored:
@@ -259,7 +262,7 @@
 | `diarize`                          | speaker attribution is not available here   |
 | `use_multi_channel`                | per-channel transcription is not available  |
 | `entity_detection`                 | entities would be missing from the response |
 | `entity_redaction`                 | text would come back unredacted             |
 | `entity_redaction_mode`            | same                                        |
 | `no_verbatim`                      | the transcript would still be verbatim      |
 | `tag_audio_events`                 | event tags are not available here           |
@@ -280,12 +283,12 @@
   parameter; the transcript itself is unaffected.
 </Warning>
 
 Booleans count as requested only when true: `False`, `"False"`, `0`, and `off`
 are treated as unset, so default SDK serialization never trips any of these.
 
 ### Realtime STT
 
 The realtime WebSocket (`/v1/speech-to-text/realtime`) supports
 `commit_strategy=manual` only: stream `input_audio_chunk` messages and set
 `commit: true` to receive the committed transcript. Your application decides
 the segment boundaries, so `commit_strategy=vad` is refused at the handshake
@@ -301,7 +304,7 @@
   a fluent, confident, completely wrong transcript**, word timestamps and all:
   the samples are read at the wrong rate, so the audio is effectively played at
   the wrong speed before it ever reaches the recognizer. There is no error to
   catch. Pass `?audio_format=pcm_44100` (or resample to 16 kHz before sending).
 
   Only `pcm_<rate>` values are accepted; `ulaw_8000` and other containerized
   names are refused at the handshake rather than opening a session that would