Get Your API Key
An API key is required to use BizCrush APIs. You can create up to 5 keys from your account settings.
Speech-to-Text API
FeaturedSpeech-to-Text API for audio transcription with speaker diarization. Supports file-based (batch) and real-time (streaming) transcription.
Build Your Own STT Service
Use the BizCrush STT API to build your own speech-to-text service. Integrate real-time transcription into your apps, process audio files at scale, or create custom voice-powered workflows — all with speaker diarization and multi-language support.
Automatic language detection across 60 languages — no configuration required. Just send audio and get results.
Overview
Authentication
All STT requests require an API key passed as a query parameter:
?api_key=YOUR_API_KEY
You can issue your API key at bizcru.sh/settings.
File STT
Send an audio file URL and receive the full transcript with speaker diarization.
/stt
Auth: ?api_key
Transcribe an audio file URL and return the full transcript with speaker diarization.
Request Body (JSON)
| Name | Type | Required | Description |
|---|---|---|---|
| audio_url | string | required | Public audio file URL (signed URL, CDN URL, etc.) |
| session_id | string | optional | Session identifier. Auto-generated if omitted |
| enable_diarization | boolean | optional | Enable speaker diarization (default: true) |
| language_hints | string[] | optional | Language codes, e.g. ["ko","en"] or "ko,en". Up to 3; extras dropped. Omit for auto-detection. Legacy `language_hint` (single string) also accepted. |
| context_keywords | string[] | optional | Domain keywords/terms to bias recognition toward names, jargon, etc., e.g. ["Albert Gee","BizCrush"] or "Albert Gee,BizCrush". Up to 2000 characters total; extras dropped. |
Response Fields
| Field | Type | Description |
|---|---|---|
| text | string | Full transcript text |
| detected_language | string | Detected language code |
| confidence | float | Average confidence (0~1) |
| utterances | array | Speaker-diarized utterance segments |
| utterances[].speaker | string | Speaker number ("0", "1", "2", ...) |
| utterances[].text | string | Utterance text |
| utterances[].start_ms | integer | Start time (milliseconds) |
| utterances[].end_ms | integer | End time (milliseconds) |
| utterances[].confidence | float | Utterance confidence |
| utterances[].language | string | Utterance language |
Response Example
{
"text": "Could you introduce yourself and tell us about your role?",
"detected_language": "en",
"confidence": 0.961,
"utterances": [
{
"speaker": "1",
"text": "Could you introduce yourself and tell us about your role?",
"start_ms": 960,
"end_ms": 5400,
"confidence": 0.94,
"language": "en"
},
{
"speaker": "2",
"text": "Sure. My name is Ethan Kim, and I'm CTO of BizCrush.",
"start_ms": 5820,
"end_ms": 9600,
"confidence": 0.96,
"language": "en"
}
]
}
cURL Example
curl -X POST \
"https://extapi.bizcrush.ai/v1/stt?api_key=YOUR_API_KEY" \
-H "Content-Type: application/json" \
-d '{"audio_url": "https://example.com/audio.oga", "enable_diarization": true}'
Note: Response time depends on audio length — may take up to several minutes. Set timeout to 600+ seconds.
Live STT (WebSocket)
Real-time audio streaming transcription via WebSocket. Send audio chunks and receive interim/final results in real time.
wss://extapi.bizcrush.ai/v1/stt/stream
Query Parameters
| Name | Type | Required | Description |
|---|---|---|---|
| api_key | string | required | API key |
| format | string | optional | "json" for JSON text frames, omit for protobuf binary |
| reduce_noise | string | optional | "true" to enable real-time noise reduction |
| language_hints | string[] | optional | Language codes as comma-separated (?language_hints=ko,en) or repeated (?language_hints=ko&language_hints=en). Up to 3; extras dropped. Omit for auto-detection. |
| enable_diarization | string | optional | Speaker diarization toggle. "false" to disable; enabled by default. When enabled, each chunk carries a speaker label. |
Protocol
Send a JSON text frame with encoding and optional session_id
| Field | Type | Description |
|---|---|---|
| encoding | string | "pcm16" or "opus" (default: "pcm16") |
| session_id | string | Session identifier (optional, auto-generated if omitted) |
| language_hints | string[] | Optional language codes, e.g. ["ko","en"] or "ko,en". Up to 3 entries; extras dropped. Overrides query param. Omit for auto-detection. |
| enable_diarization | boolean | Speaker diarization toggle (default: true). Overrides the query param. |
| context_keywords | string[] | Domain keywords/terms to bias recognition toward names, jargon, etc., e.g. ["Albert Gee","BizCrush"] or "Albert Gee,BizCrush". Up to 2000 characters total; extras dropped. |
{"encoding": "pcm16", "session_id": "optional-session-id"}
Server responds with connection status
{"connected": true}
Send raw audio as binary WebSocket frames
| Encoding | Format | Chunk Size |
|---|---|---|
| pcm16 | 16kHz, mono, 16-bit little-endian | 640 bytes (20ms) recommended |
| opus | 16kHz, mono. Raw Opus packets or OGG/Opus container | Opus frame unit |
Server sends interim and final results as JSON text frames
| Field | Type | Description |
|---|---|---|
| chunk.id | string | Chunk ID — same across interim/final for one utterance, new ID after final |
| chunk.session_id | string | Session ID |
| chunk.text | string | Transcribed text (cumulative — each interim contains full text so far) |
| chunk.is_final | boolean | false: interim result, true: final confirmed result |
| chunk.speaker | string | Speaker label ("0", "1", "2", ...) when diarization is enabled. Omitted when disabled or the speaker is unresolved. |
{"chunk": {"id": "01KMCHP57H31XYZABC", "session_id": "my-session", "text": "Hello, how are you?", "is_final": true, "speaker": "1"}}
When no more audio will be sent, send an empty binary frame (0 bytes) and KEEP THE WEBSOCKET OPEN. The server finalizes any pending utterances (including ones still in interim state), streams the trailing is_final results back, then closes the WebSocket. Closing the WebSocket without this marker drops the trailing finals — WebSocket has no half-close, so once the client closes the server cannot deliver the remaining results. Finalize completes within ~5 seconds after the EOS marker. After the trailing finals the server sends a terminal frame ({"finished": true} for JSON, disconnect_reason=SESSION_ENDED for protobuf) and then closes.
ws.send(b'') # Python
ws.send(new ArrayBuffer(0)); // JavaScript
Examples
Python
import asyncio
import json
import websockets
async def live_stt(api_key: str):
url = f"wss://extapi.bizcrush.ai/v1/stt/stream?api_key={api_key}&format=json"
async with websockets.connect(url) as ws:
await ws.send(json.dumps({"encoding": "pcm16"}))
resp = json.loads(await ws.recv())
assert resp["connected"], f"Connection failed: {resp}"
async def send_audio():
with open("audio.pcm", "rb") as f:
while chunk := f.read(640):
await ws.send(chunk)
await asyncio.sleep(0.02)
# End-of-audio marker. DO NOT close the WebSocket here — the
# server needs the connection open to deliver trailing finals.
# The server will close after finalize completes (~5 seconds).
await ws.send(b'')
async def receive_results():
try:
async for msg in ws:
data = json.loads(msg)
if "chunk" in data:
chunk = data["chunk"]
status = "FINAL" if chunk["is_final"] else "interim"
speaker = chunk.get("speaker", "?")
print(f"[{status}] (S{speaker}) {chunk['text']}")
except websockets.exceptions.ConnectionClosed:
pass
await asyncio.gather(send_audio(), receive_results())
asyncio.run(live_stt("YOUR_API_KEY"))
JavaScript (Browser)
const ws = new WebSocket(
"wss://extapi.bizcrush.ai/v1/stt/stream?api_key=YOUR_API_KEY&format=json"
);
ws.onopen = () => {
ws.send(JSON.stringify({ encoding: "pcm16" }));
};
ws.onmessage = (event) => {
const data = JSON.parse(event.data);
if (data.connected) {
console.log("Connected! Start sending audio...");
}
if (data.chunk) {
const { id, text, is_final, speaker } = data.chunk;
console.log(`[${is_final ? "FINAL" : "interim"}] (S${speaker ?? "?"}) ${text}`);
}
};
// Server closes the WebSocket after delivering trailing finals.
ws.onclose = () => {
console.log("Stream finalized by server");
};
function sendAudioChunk(pcmData) {
if (ws.readyState === WebSocket.OPEN) {
ws.send(pcmData);
}
}
// Call this when you have no more audio to send.
// DO NOT call ws.close() — the server needs the socket open to deliver
// trailing finals, then closes it itself (~5 seconds).
function endStream() {
if (ws.readyState === WebSocket.OPEN) {
ws.send(new ArrayBuffer(0)); // end-of-audio marker
}
}
Text-to-Speech (TTS)
Convert text to natural-sounding speech and receive a signed download URL (audio uploaded to Firebase Storage).
/tts
Auth: ?api_key
Generate speech audio from text and return a signed download URL (audio uploaded to Firebase Storage at tts/{session_id}/{message_id}.ogg).
Request Body (Protobuf)
application/x-protobuf (request and response are protobuf-encoded SessionTTSRequest / SessionTTSResponse)
| Name | Type | Required | Description |
|---|---|---|---|
| session_id | string | required | Session identifier |
| message_id | string | required | Message identifier (used in storage path) |
| text | string | required | Text to convert to speech |
Response Fields
| Field | Type | Description |
|---|---|---|
| audio_url | string | Signed download URL (Opus / OGG) — valid for several days |
| error | string | Set on failure; audio_url empty |
TTS Supported Languages
Text-to-speech generates natural-sounding audio across 57 languages. The language is detected automatically from the input text — no configuration required.
| Code | Language |
|---|---|
| en | English |
| zh | 中文 |
| hi | हिन्दी |
| es | Español |
| ar | العربية |
| fr | Français |
| pt | Português |
| ru | Русский |
| id | Bahasa Indonesia |
| de | Deutsch |
| ja | 日本語 |
| vi | Tiếng Việt |
| it | Italiano |
| ko | 한국어 |
| th | ภาษาไทย |
| af | Afrikaans |
| hy | Հայերեն |
| az | Azərbaycan |
| be | Беларуская |
| bs | Bosanski |
| bg | Български |
| ca | Català |
| hr | Hrvatski |
| cs | Čeština |
| da | Dansk |
| nl | Nederlands |
| et | Eesti |
| fi | Suomi |
| gl | Galego |
| el | Ελληνικά |
| he | עברית |
| hu | Magyar |
| is | Íslenska |
| kn | ಕನ್ನಡ |
| kk | Қазақ |
| lv | Latviešu |
| lt | Lietuvių |
| mk | Македонски |
| ms | Bahasa Melayu |
| mi | Te Reo Māori |
| mr | मराठी |
| ne | नेपाली |
| no | Norsk |
| fa | فارسی |
| pl | Polski |
| ro | Română |
| sr | Српски |
| sk | Slovenčina |
| sl | Slovenščina |
| sw | Kiswahili |
| sv | Svenska |
| tl | Tagalog |
| ta | தமிழ் |
| tr | Türkçe |
| uk | Українська |
| ur | اردو |
| cy | Cymraeg |
Audio Noise Reduction
Remove background noise from audio files using BizCrush AI. Provide a download URL and receive a denoised M4A file.
/v1/denoise
X-API-Key
Denoise an audio file using BizCrush AI. Provide a download URL; the server applies noise reduction, re-encodes to M4A (AAC), and returns a signed download URL valid for 1 hour.
Request Body
| Name | Type | Required | Description |
|---|---|---|---|
| audio_url | string | required | Input audio file download URL (any format supported by ffmpeg) |
| atten_limit_db | number | optional | Noise attenuation limit in dB. 0 uses the server default; higher values are stronger (0-100) |
| disable_adaptive_mix | boolean | optional | Use fixed attenuation for a stronger effect; may damage speech (default: false) |
Response Fields
| Field | Type | Description |
|---|---|---|
| denoised_audio_url | string | Signed download URL for the denoised M4A file (valid ~1 hour) |
| applied_atten_db | number | Average attenuation limit actually applied |
| frame_lsnr_db | number[] | Frame-level local SNR measurements |
Supported Languages
Automatic language detection is built in — no configuration needed. The STT engine detects and transcribes across 60 languages seamlessly, even when speakers switch languages mid-conversation. Optionally use language_hint for faster detection.
| Code | Language |
|---|---|
| en | English |
| zh | 中文 |
| hi | हिन्दी |
| es | Español |
| ar | العربية |
| fr | Français |
| pt | Português |
| ru | Русский |
| id | Bahasa Indonesia |
| de | Deutsch |
| ja | 日本語 |
| vi | Tiếng Việt |
| it | Italiano |
| ko | 한국어 |
| th | ภาษาไทย |
| af | Afrikaans |
| sq | Shqip |
| az | Azərbaycan |
| eu | Euskara |
| be | Беларуская |
| bn | বাংলা |
| bs | Bosanski |
| bg | Български |
| ca | Català |
| hr | Hrvatski |
| cs | Čeština |
| da | Dansk |
| nl | Nederlands |
| et | Eesti |
| fi | Suomi |
| gl | Galego |
| el | Ελληνικά |
| gu | ગુજરાતી |
| he | עברית |
| hu | Magyar |
| kn | ಕನ್ನಡ |
| kk | Қазақ |
| lv | Latviešu |
| lt | Lietuvių |
| mk | Македонски |
| ms | Bahasa Melayu |
| ml | മലയാളം |
| mr | मराठी |
| no | Norsk |
| fa | فارسی |
| pl | Polski |
| pa | ਪੰਜਾਬੀ |
| ro | Română |
| sr | Српски |
| sk | Slovenčina |
| sl | Slovenščina |
| sw | Kiswahili |
| sv | Svenska |
| tl | Tagalog |
| ta | தமிழ் |
| te | తెలుగు |
| tr | Türkçe |
| uk | Українська |
| ur | اردو |
| cy | Cymraeg |
STT Error Codes
| Status | Error | Description |
|---|---|---|
| 401 | Missing authorization | API key not provided |
| 401 | Invalid API key | Invalid API key |
| 400 | Empty request body | Request body is empty |
| 400 | Missing audio_url | audio_url field missing (File STT) |
| 400 | Missing session_id, message_id, or text | Required TTS field missing |
| 400 | Invalid JSON | JSON parsing failed |
| 500 | TTS generation failed | OpenAI TTS call failed |
| 500 | Storage upload failed | Firebase Storage upload failed |
| 500 | (varies) | Internal server error (transcription failure, etc.) |
| WS 1008 | Auth failure | WebSocket closed with code 1008 on authentication failure |
Text Translation
Translate plain text into multiple target languages with BizCrush AI.
/v1/translate-text
Translate plain-text input while preserving document formatting.
Parameters
| Name | Type | Required | Description |
|---|---|---|---|
| text | str | required | Plain text to translate (up to 100,000 characters by default) |
| target_languages | list[str] | required | Target language codes or names, such as ["ko", "en", "Japanese"] (up to 10) |
| source_language | str | optional | Source language code or name (default: auto) |