> ## Documentation Index
> Fetch the complete documentation index at: https://omi-codex-mobile-prompt-metadata.mintlify.site/llms.txt
> Use this file to discover all available pages before exploring further.
# Real-time Transcription
> A comprehensive guide to Omi's real-time audio transcription system, covering WebSocket connections, STT providers, speaker diarization, message formats, and building external custom STT services.
## Overview
Omi's transcription system provides **real-time speech-to-text** conversion with speaker identification, multiple language support, and seamless integration with the conversation processing pipeline.
```mermaid theme={null}
flowchart LR
subgraph Client["📱 Omi App"]
Audio[Audio Capture]
end
subgraph Backend["🖥️ Backend"]
WS["/v4/listen
WebSocket"]
Decode[Audio Decoder]
end
subgraph STT["🎧 STT Providers"]
Parakeet[Parakeet]
Modulate[Modulate Velma-2]
end
Audio -->|Binary stream| WS
WS --> Decode
Decode --> Parakeet
Decode --> Modulate
Parakeet -->|Transcript| WS
Modulate -->|Transcript| WS
WS -->|JSON segments| Audio
```
Connect to `/v4/listen` WebSocket with your user token and start streaming audio. Transcripts arrive in real-time as JSON.
Read through for complete endpoint details, configuration options, and message formats.
* Multiple STT providers with automatic fallback
* Speech profile for user identification
* Dual-socket architecture for speaker training
* [External Custom STT](#external-custom-stt-service) for your own transcription service
## WebSocket Endpoint
WebSocket connections require Firebase authentication. The `uid` parameter must be a valid user ID obtained through Firebase Auth.
### Endpoint URL
```
wss://api.omi.me/v4/listen?uid={uid}&language={lang}&sample_rate={rate}&codec={codec}
```
### Query Parameters
**Type:** `string`
User ID obtained from Firebase authentication. Required for all connections.
**Type:** `string` | **Default:** `'en'`
Language code for transcription. Supports:
* Standard codes: `'en'`, `'es'`, `'fr'`, `'de'`, `'ja'`, `'zh'`, etc.
* Multi-language: `'multi'` for automatic language detection
**Type:** `integer` | **Default:** `8000`
Audio sample rate in Hz. Common values: `8000`, `16000`, `44100`, `48000`
**Type:** `string` | **Default:** `'pcm8'`
Audio codec. Supported options:
* `pcm8` - 8-bit PCM (default)
* `pcm16` - 16-bit PCM
* `opus` - Opus codec (16kHz)
* `opus_fs320` - Opus with 320 frame size
* `aac` - AAC codec
* `lc3` - LC3 codec
* `lc3_fs1030` - LC3 with 1030 frame size
**Type:** `integer` | **Default:** `1`
Number of audio channels. Use `1` for mono, `2` for stereo.
**Type:** `boolean` | **Default:** `true`
Enable speaker identification using the user's stored speech profile. When enabled, the system extracts a speaker embedding from the user's speech profile and uses it to identify the user's voice via biometric matching.
**Type:** `integer` | **Default:** `120` | **Range:** `2-14400`
Seconds of silence before the conversation is automatically processed. After this timeout, the conversation is saved and LLM processing begins.
**Type:** `string` | **Optional**
Serving provider selection is controlled by the deployment policy. The supported providers are `parakeet` and `modulate`; clients cannot opt into retired providers.
**Type:** `string` | **Default:** `'disabled'`
Enable custom STT mode. When set to `'enabled'`, the backend accepts app-provided transcripts instead of using STT services. Useful for apps with their own transcription.
**Type:** `string` | **Optional**
Conversation source identifier. Examples: `'omi'`, `'openglass'`, `'phone'`
## Audio Codecs
The system supports multiple audio codecs with automatic decoding:
| Codec | Sample Rate | Description | Use Case |
| ------------ | ----------- | -------------- | ---------------------- |
| `pcm8` | 8kHz | 8-bit PCM | Default, low bandwidth |
| `pcm16` | 16kHz | 16-bit PCM | Better quality |
| `opus` | 16kHz | Opus encoded | Efficient compression |
| `opus_fs320` | 16kHz | Opus 320 frame | Alternative frame size |
| `aac` | Variable | AAC encoded | iOS compatibility |
| `lc3` | Variable | LC3 codec | Bluetooth audio |
| `lc3_fs1030` | Variable | LC3 1030 frame | Alternative LC3 |
All audio is internally converted to 16-bit linear PCM before being sent to STT providers.
## STT Service Selection
The canonical provider/surface matrix and default model order live in
`backend/config/stt_provider_policy.py`. The deployment validator requires
`STT_SERVICE_MODELS` and `STT_PRERECORDED_MODEL` to match that code-owned
policy; an environment edit cannot re-enable hosted Deepgram. The normal serving
defaults are `modulate-velma-2,parakeet`. Self-hosted Deepgram is a distinct,
streaming-only provider that requires `DEEPGRAM_SELF_HOSTED_ENABLED=true` and a
non-cloud `DEEPGRAM_SELF_HOSTED_URL`; it is never a fallback to
`api.deepgram.com`. If no supported provider can serve a language, the request
fails closed.
```mermaid theme={null}
flowchart TD
Start[Incoming Audio] --> Lang{Language?}
Lang -->|Supported by first configured model| Selected[Parakeet / Modulate]
Lang -->|Unsupported| Unavailable[Fail closed]
```
### Provider Capabilities
| Provider | Languages | Model | Best For |
| ------------ | -------------------- | ---------- | ----------------------------------- |
| **Modulate** | Provider-dependent | `velma-2` | Supported configured languages |
| **Parakeet** | Deployment-dependent | `parakeet` | Self-hosted streaming and batch STT |
## Transcription outcome contract
Transport success is not transcription success. The backend uses one bounded
semantic vocabulary across voice upload, voice-message SSE, offline sync, and
live provider failures:
| Outcome | Meaning | Retryable |
| ------------------ | ---------------------------------------------------------- | --------- |
| `success` | Speech-eligible audio produced non-empty normalized text | No |
| `expected_silence` | An explicit speech gate found no eligible speech | No |
| `empty_unexpected` | Eligible audio reached STT but produced no usable text | Yes |
| `timeout` | The selected provider timed out | Yes |
| `upstream_error` | The provider or response parser failed | Yes |
| `config_error` | The selected provider is not deployable on this runtime | No |
| `invalid_input` | The audio cannot be decoded or violates the route contract | No |
`POST /v2/voice-message/transcribe` returns `outcome` alongside `transcript`,
`stt_provider`, and `stt_model`. True silence remains HTTP 200 with an empty
transcript and `outcome=expected_silence`. All failure outcomes use a non-2xx
status and a fixed response body containing only `error`, `outcome`,
`provider`, `retryable`, and a public-safe message; provider response bodies,
audio identifiers, and exception text are never exposed.
The `/v2/voice-messages` stream emits the same safe failure object in a terminal
`error:` SSE frame. A normal empty stream is reserved for explicit
`expected_silence`.
For `/v4/listen`, an unusable initial or mid-session STT socket emits this event
before the client connection closes with WebSocket code 1011:
```json theme={null}
{
"type": "service_status",
"status": "stt_failed",
"outcome": "upstream_error",
"provider": "parakeet",
"retryable": true,
"reason": "connection_lost"
}
```
The backend retains any buffered audio it could not hand to the provider. It
does not keep a green client WebSocket while discarding later audio; the close
activates the client's existing reconnect or local-recovery path.
## Production transcription serving smoke
Development deployments retain the no-traffic, uniquely tagged `backend`
candidate gate. Production does not use a tagged URL, Cloud Run Job, service
account, or temporary IAM binding. Instead it validates exact no-traffic Cloud
Run revisions, snapshots traffic, promotes them, verifies the exact serving
release vector, and only then proves the real `/v2/voice-message/transcribe`
route through `https://api.omi.me`. This is post-promotion serving evidence,
not candidate evidence: any serving-vector, route-presence, or known-audio
failure restores the saved Cloud Run traffic snapshot.
`backend/testing/release_fixtures/transcription-release-probe.wav` and its
versioned JSON manifest provide the known audio, language, expected transcript,
SHA-256 digest, and CC-BY-4.0 LibriSpeech provenance. The gate requires HTTP
200, `outcome=success`, and the exact normalized transcript. It intentionally
does not assert provider or model identity: this is a semantic capability gate,
not an unreviewed routing-policy setting.
The shared `transcription-release-candidate-probe` action uses the existing
authenticated deploy identity to read the existing `FIREBASE_API_KEY` Secret
Manager secret, sign a five-minute Firebase custom token for the dedicated
non-human `omi-release-probe` UID, and exchange it immediately for an ID token.
It writes that token only to an owner-only temporary runner file, never exposes
it through workflow outputs or evidence, and deletes it at step exit. The
workflow makes no IAM changes: the existing deploy identity must be reviewed to
have only the Secret Manager read and `iam.serviceAccounts.signJwt` access this
action needs. Missing access fails closed before promotion. Reports contain only
redacted booleans and status codes.
The production smoke first makes a deliberately unauthorized, malformed-safe
reservation-route request and requires its expected validation response; it
does not create a reservation. It then runs the known-audio probe with the same
runner-local token file. The token is never a workflow output, command-line
argument, artifact, or resource metadata.
## Automatic development candidate acceptance
`gcp_backend_auto_dev.yml` keeps the development backend mutation lock while
it deploys the four Cloud Run services (`backend`, `backend-sync`,
`backend-sync-backfill`, and `backend-integration`) with the service-scoped
`candidate` tag. It resolves each exact tag URL from Cloud Run status, runs the
source-owned `backend/deploy/dev_candidate_acceptance.json` manifest, and only
then permits the one traffic-promotion step. The backend uses the authenticated
What Matters Now contract; the worker services use bounded `/v1/health` checks.
The evidence artifact records only service, bounded contract category, and
outcome — never URLs, tokens, user data, or response bodies.
Candidate requests carry an OIDC token in `X-Serverless-Authorization` while
the application keeps its own `Authorization` header. The token audience is
the canonical Cloud Run service URL as required by Cloud Run; the HTTP target
remains the exact no-traffic tagged candidate URL.
Development pusher is deliberately **GKE-only** (`pusher.omiapi.com`). A
legacy Cloud Run `pusher` service is not a deploy, candidate, or health-report
surface; do not publish an image merely to make that retired surface appear
ready.
During the direct-URL retirement window, the public dev Cloud Run `backend`
and legacy `backend-listen` services must use `http://pusher.omiapi.com`.
Other dev Cloud Run services and jobs must not define `HOSTED_PUSHER_API_URL`.
Retire those public endpoints only after they show no `/v4/listen` traffic.
## Serving STT Configuration
Serving revisions use `HOSTED_PARAKEET_API_URL` for Parakeet and
`MODULATE_API_KEY` for Modulate. Both are required because the selected
provider depends on language capability. The retained self-hosted Deepgram
deployment uses `DEEPGRAM_API_KEY`, `DEEPGRAM_SELF_HOSTED_ENABLED`, and
`DEEPGRAM_SELF_HOSTED_URL` only in its explicitly configured GKE workload;
hosted Deepgram is disabled.
## External Custom STT Service
Build your own transcription/diarization WebSocket service that integrates with Omi.
```mermaid theme={null}
flowchart LR
subgraph App["📱 Omi App"]
Capture[Audio Capture]
end
subgraph Custom["🎧 Your STT Service"]
WS[WebSocket Server]
end
subgraph Backend["🖥️ Omi Backend"]
API["/v4/listen"]
end
Capture -->|Binary audio| WS
WS -->|JSON transcripts| Capture
Capture -->|suggested_transcript| API
```
### Your Service Receives
| Message | Format | Description |
| ------------------------- | ------ | ----------------------------------------------------------------- |
| Audio frames | Binary | Raw audio bytes (codec configured by app, typically `opus` 16kHz) |
| `{"type": "CloseStream"}` | JSON | End of audio stream |
### Your Service Sends
**Format:** JSON object with `segments` array
```json theme={null}
{
"segments": [
{
"text": "Hello, how are you?",
"speaker": "SPEAKER_00",
"start": 0.0,
"end": 1.5
},
{
"text": "I'm doing great, thanks!",
"speaker": "SPEAKER_01",
"start": 1.6,
"end": 3.2
}
]
}
```
### Segment Fields
| Field | Type | Required | Description |
| --------- | -------- | -------- | ------------------------------------------------ |
| `text` | `string` | Yes | Transcribed text |
| `speaker` | `string` | No | Speaker label (`SPEAKER_00`, `SPEAKER_01`, etc.) |
| `start` | `float` | No | Start time in seconds |
| `end` | `float` | No | End time in seconds |
### Requirements
* Response **must be an object** with `segments` key. Raw arrays `[{...}]` will fail.
* Do **not** include a `type` field, or set it to `"Results"`. Other values are ignored.
* Connection closes after **90 seconds** of inactivity.
## Speech Profile & Speaker Embedding
When a user has a speech profile, the system uses speaker embedding comparison to identify the user's voice in real-time.
### How It Works
```mermaid theme={null}
sequenceDiagram
participant App as 📱 Omi App
participant Backend as 🖥️ Backend
participant STT as 🎧 Selected STT provider
participant Embed as 🧠 Embedding API
Note over Backend: User has speech profile
App->>Backend: Connect WebSocket
Backend->>STT: Create single socket
Backend->>Embed: Extract user embedding from profile WAV
loop Audio streaming
App->>Backend: Audio chunk
Backend->>STT: Forward decoded audio
STT-->>Backend: Transcript with speaker IDs
end
Note over Backend: New speaker detected (2s+ audio)
Backend->>Embed: Extract speaker embedding from audio
Embed-->>Backend: Compare with user embedding
Backend-->>App: Segments (is_user: true/false)
```
### Speech Profile Benefits
1. **User Identification**: Speaker embedding comparison identifies the device owner by voice biometrics
2. **No Startup Delay**: Transcription begins immediately (no profile audio prepending)
3. **Single Socket**: One selected-provider connection per session
## Transcription Flow
WebSocket connection accepted, user validated, STT provider selected based on language.
App sends binary audio chunks. Backend decodes based on codec parameter.
Decoded audio is sent to the selected Parakeet or Modulate provider. The provider returns word-level transcripts with speaker IDs.
Words grouped into segments. Same-speaker consecutive words merged. Timing adjusted for speech profile offset.
JSON segments streamed back to app immediately. UI updates as user speaks.
Background task monitors silence. After `conversation_timeout`, conversation is processed and saved.
## Message Formats
### Incoming Messages (App → Backend)
**Format:** Binary
Raw audio bytes encoded according to the `codec` parameter. Sent continuously during recording.
```
[Binary audio chunk - varies by codec]
```
**Keep-alive:** Messages of 2 bytes or less are treated as heartbeat pings.
**Format:** JSON
Assign a known person to detected speakers:
```json theme={null}
{
"type": "speaker_assigned",
"speaker_id": 1,
"person_id": "person-uuid-here",
"person_name": "John",
"segment_ids": ["seg-uuid-1", "seg-uuid-2"]
}
```
**Format:** JSON
When `custom_stt=enabled`, apps can provide their own transcripts:
```json theme={null}
{
"type": "suggested_transcript",
"segments": [
{
"text": "Hello there",
"speaker": "SPEAKER_00",
"speaker_id": 0,
"start": 0.0,
"end": 1.5,
"is_user": true,
"person_id": "known-person-uuid-or-null"
}
],
"stt_provider": "custom-provider-name"
}
```
See [External Custom STT Service](#external-custom-stt-service) for building your own transcription service.
**Format:** JSON
For OpenGlass and visual captures:
```json theme={null}
{
"type": "image_chunk",
"id": "temp-image-id",
"index": 0,
"total": 3,
"data": "base64-encoded-chunk"
}
```
### Outgoing Messages (Backend → App)
**Format:** JSON Array
Real-time transcript segments as they're detected:
```json theme={null}
[
{
"id": "uuid-string",
"text": "Hello there",
"speaker": "SPEAKER_00",
"speaker_id": 0,
"is_user": true,
"person_id": null,
"start": 0.0,
"end": 1.5,
"speech_profile_processed": true,
"stt_provider": "parakeet"
}
]
```
**Format:** JSON
Connection and service status updates:
```json theme={null}
{
"type": "service_status",
"status": "ready",
"status_text": "Service Ready"
}
```
**Format:** JSON
System suggests a known person for a detected speaker:
```json theme={null}
{
"type": "speaker_label_suggestion",
"speaker_id": 1,
"person_id": "person-uuid",
"person_name": "John",
"segment_id": "segment-uuid"
}
```
**Format:** JSON
Sent when conversation timeout triggers processing:
```json theme={null}
{
"type": "memory_created",
"memory": {
"id": "conversation-uuid",
"structured": {
"title": "Meeting Discussion",
"overview": "..."
}
},
"messages": []
}
```
**Format:** JSON
When translation is enabled:
```json theme={null}
{
"type": "translation",
"segments": [
{
"id": "segment-uuid",
"translations": [
{"lang": "es", "text": "Hola ahí"}
]
}
]
}
```
## Transcript Segment Model
Each transcript segment contains:
| Field | Type | Description |
| -------------------------- | --------- | ---------------------------------------------------- |
| `id` | `string` | Unique UUID for the segment |
| `text` | `string` | Transcribed text content |
| `speaker` | `string` | Speaker label (`"SPEAKER_00"`, `"SPEAKER_01"`, etc.) |
| `speaker_id` | `integer` | Numeric speaker ID (0, 1, 2...) |
| `is_user` | `boolean` | `true` if spoken by device owner |
| `person_id` | `string?` | UUID of identified person (if matched) |
| `start` | `float` | Start time in seconds |
| `end` | `float` | End time in seconds |
| `speech_profile_processed` | `boolean` | Whether speech profile was used for identification |
| `stt_provider` | `string?` | Name of STT provider used |
## Connection Lifecycle
```mermaid theme={null}
stateDiagram-v2
[*] --> Connecting: WebSocket request
Connecting --> Authenticating: Connection accepted
Authenticating --> Ready: User validated
Authenticating --> Closed: Auth failed
Ready --> Streaming: Audio received
Streaming --> Streaming: More audio
Streaming --> Processing: Silence timeout
Processing --> Streaming: New audio
Processing --> Closed: Session complete
Ready --> Closed: Client disconnect
Streaming --> Closed: Client disconnect
note right of Processing
Conversation saved
LLM extracts structure
Memories extracted
end note
```
### Lifecycle Events
1. WebSocket accepted
2. User authentication verified
3. Language/STT service selected
4. STT connections initialized (with retry logic)
5. Speech profile loaded in background
6. Heartbeat task started (10s interval)
1. Audio received and decoded
2. Sent to STT provider(s)
3. Results collected in buffers
4. Processed every 600ms
5. Segments sent to client
6. Speaker suggestions generated
1. Usage statistics recorded
2. All STT sockets closed
3. Client WebSocket closed (code 1000/1001)
4. Buffers and collections cleared
## Error Handling & Retry Logic
The system includes robust error handling:
| Error Type | Handling |
| ------------------------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| **STT Connection Failed** | Before live audio is accepted, bounded connection retry may select a configured provider. After a live provider becomes unusable, emit `service_status(stt_failed)` and close the client WebSocket with `1011`; do not discard audio as successful. |
| **Provider Error** | Pre-recorded/sync work classifies the provider failure and remains retryable. Live `/v4/listen` fails the session visibly; the desktop/mobile client retains local retry material and reconnects. |
| **Decode Error** | Log and skip corrupted audio chunk |
| **WebSocket Error** | Clean close with appropriate code |
Live STT never silently downgrades a terminal provider failure to a successful
transcription. The app receives the bounded failure status before the `1011`
close and must retain unconfirmed local audio until a later successful sync.
## Key File Locations
| Component | Path |
| ------------------------- | -------------------------------------- |
| WebSocket Handler | `backend/routers/transcribe.py` |
| Streaming STT Integration | `backend/utils/stt/streaming.py` |
| Audio Decoding | `backend/routers/transcribe.py` |
| Speech Profile | `backend/utils/stt/speech_profile.py` |
| VAD (Voice Activity) | `backend/utils/stt/vad.py` |
| Transcript Model | `backend/models/transcript_segment.py` |
## Related Documentation
Complete backend architecture overview
How conversations and memories are stored
How the AI chat system uses transcriptions
Environment setup and configuration