[Core] Generalized turn-based /v1/realtime implementation (#8339)
Signed-off-by: Nick Cao <ncao@redhat.com> Co-authored-by: Codex <noreply@openai.com>
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@@ -277,61 +277,6 @@ cd examples/online_serving/qwen3_omni
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python examples/online_serving/openai_chat_completion_client_for_multimodal_generation.py --model Qwen/Qwen3-Omni-30B-A3B-Instruct --query-type use_image --port 8091 --host "localhost"
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```
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#### Realtime WebSocket client (`openai_realtime_client.py`)
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[`openai_realtime_client.py`](./openai_realtime_client.py) connects to **`ws://<host>:<port>/v1/realtime`**, streams a local WAV as **PCM16 mono @ 16 kHz** in fixed-size chunks, and receives **`response.output_audio.*`**, **`response.output_text.*`**, and transcript events. By default it uses explicit `input_audio_buffer.commit`; pass **`--server-vad`** to let the server detect speech endpoints and commit the turn automatically. The client concatenates audio deltas and writes **`--output-wav`** (model output is typically **24 kHz**). Optional **`--delta-dump-dir`** saves each delta as `delta_000001.wav`, … for debugging.
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Streaming input works well for translation-style use cases; if the Thinker runs while input is still incomplete, consider limiting **`max_tokens`** in your session / server defaults to avoid over-generation.
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**Dependencies:**
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```bash
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pip install websockets
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```
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**From this directory** (`examples/online_serving/qwen3_omni`):
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```bash
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python openai_realtime_client.py \
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--url ws://localhost:8091/v1/realtime \
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--model Qwen/Qwen3-Omni-30B-A3B-Instruct \
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--input-wav /path/to/input_16k_mono.wav \
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--output-wav realtime_output.wav \
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--server-vad \
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--delta-dump-dir ./rt_delta_wavs
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```
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**Arguments:**
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| Flag | Default | Description |
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| ------ | --------- | ------------- |
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| `--url` | `ws://localhost:8091/v1/realtime` | Full WebSocket URL including path |
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| `--model` | `Qwen/Qwen3-Omni-30B-A3B-Instruct` | Must match the served model (sent in `session.update`) |
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| `--input-wav` | *(required)* | Input WAV: mono, 16-bit PCM, **16 kHz** |
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| `--output-wav` | `realtime_output.wav` | Output path for concatenated reply audio |
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| `--output-text` | *(optional)* | If set, write final transcription text to this path |
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| `--chunk-ms` | `200` | Size of each uploaded audio chunk (milliseconds of audio) |
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| `--send-delay-ms` | `0` | Delay between chunk sends (simulate realtime upload) |
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| `--delta-dump-dir` | *(optional)* | Directory to write per-`response.output_audio.delta` WAV files |
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| `--server-vad` | disabled | Enable server-side Silero VAD instead of sending explicit commits |
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| `--num-requests` | `1` | Number of sequential sessions (see `--concurrency`) |
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| `--concurrency` | `1` | Max concurrent WebSocket sessions when `--num-requests` > 1 |
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Server VAD uses the pinned Silero v6.2 ONNX artifact from `istupakov/silero-vad-onnx` and never downloads model files while processing a session. Make the artifact available in the Hugging Face cache before startup, or configure a local artifact in the deployment YAML:
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```yaml
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base_config: /path/to/vllm-omni/vllm_omni/deploy/qwen3_omni_duplex.yaml
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duplex_session:
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server_vad_model_path: /models/silero_vad.onnx
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```
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For the engine-owned plugin, use an overlay based on
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`vllm_omni/deploy/qwen3_omni_duplex.yaml`, with `duplex_session.server_vad_model_path`
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pointing to your local artifact. See the [complete duplex setup](../realtime_web/README.md#qwen3-with-server-vad-automatic-turns).
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The default `qwen3_omni_moe.yaml` remains a turn deployment. Duplex mode serves
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`/v1/realtime?duplex=1` and `/v1/chat/completions`; the legacy STT handler requires
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a turn deployment.
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The Python client supports the following command-line arguments:
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- `--query-type` (or `-q`): Query type (default: `use_video`). Options: `text`, `use_audio`, `use_image`, `use_video`
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@@ -358,6 +303,13 @@ python examples/online_serving/openai_chat_completion_client_for_multimodal_gene
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bash run_curl_multimodal_generation.sh use_image
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```
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### Realtime WebSocket
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The [shared Realtime UI](../realtime_web/README.md#qwen3-explicit-turn-realtime)
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supports explicit turns through `/v1/realtime?duplex=0`. The
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[Server VAD setup](../realtime_web/README.md#qwen3-with-server-vad-automatic-turns)
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uses the duplex endpoint.
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### FAQ
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## Modality control
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@@ -582,26 +534,6 @@ The gradio script supports the following arguments:
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Run these commands from the repository root with the vLLM-Omni environment activated.
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### Without VAD: manually submit each turn
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Start the backend:
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```bash
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vllm serve Qwen/Qwen3-Omni-30B-A3B-Instruct --omni --port 8091
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```
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In another terminal, check backend readiness and start the UI:
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```bash
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curl --fail http://127.0.0.1:8091/health
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python -m examples.online_serving.qwen3_omni.realtime_web \
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--backend ws://127.0.0.1:8091 --stt --port 7863
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```
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Open `http://localhost:7863`, start a session, speak, and press **Send turn**.
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`--stt` is the default. Each turn uses a separate connection without shared model
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history; the conversation shown in the browser is a local log.
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### With VAD: automatically submit after silence
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Start the backend with a Server VAD deployment overlay and a compatible Silero
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@@ -610,7 +542,7 @@ ONNX artifact, following the
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After its health check succeeds, start the UI:
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```bash
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python -m examples.online_serving.qwen3_omni.realtime_web \
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python -m examples.online_serving.realtime_web --profile qwen3-turn \
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--backend ws://127.0.0.1:8091 --vad --port 7863
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```
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@@ -621,9 +553,7 @@ VAD mode keeps uploading audio during replies and supports speech interruption.
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The engine-owned Qwen duplex plugin maintains conversation history and tracks
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played replies through playback acknowledgements. Each committed utterance starts
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a new generation request; the model does not use native streaming-input KV decoding.
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STT mode pauses microphone upload while the reply is generated and played.
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In VAD mode, click **Camera** to upload sampled frames with your spoken question;
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see [camera setup and limits](../realtime_web/README.md#qwen-vad-camera-input).
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STT remains audio-only. To switch back to manual turns, use the default turn
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deployment, restart the UI with `--stt`, refresh the page, and reconnect. Stop the existing UI before reusing port 7863.
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See the [shared UI guide](../realtime_web/README.md) for HTTPS access and testing.
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For explicit turns, use `--manual` with the shared UI as described in the
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[shared UI guide](../realtime_web/README.md#qwen3-explicit-turn-realtime).
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@@ -7,14 +7,14 @@ This implements [RFC #7222](https://github.com/vllm-project/vllm-omni/issues/722
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| Profile | Turn control | Camera | Playback acknowledgement |
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| --- | --- | --- | --- |
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| `minicpm-native` | Model-controlled listen/speak, continuous audio input | Frames accompany audio | Yes |
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| `qwen3-turn --stt` (default) | User presses **Send turn** | No | No |
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| `qwen3-turn --manual` | Press **Send turn**; mic pauses while the reply plays | No | No |
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| `qwen3-turn --vad` | Server detects trailing silence; speech interrupts replies | Sampled frames with each spoken turn | Yes |
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| `aura-ptt` | Hold **Hold to talk**; release commits. Hold also stops local playback | Sticky frames with speech; vision-follow every 2 frames only while unlocked (locked from PTT release until text-final / listen) | Yes |
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Qwen3 VAD uses an engine-owned duplex plugin: microphone upload continues
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while replies stream, and speech can interrupt generation and playback. The
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checkpoint still generates committed turns; this is not native streaming-input
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KV decoding. STT remains explicit-turn and audio-only.
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KV decoding.
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## MiniCPM: existing command stays valid
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@@ -62,45 +62,36 @@ just-committed speech turn on Stage0/1 (`max_num_seqs=1`). Audio still playing
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does not hold the next vision turn once text is done.
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MiniCPM / Qwen profiles do not set `pushToTalk`, so the PTT control stays hidden.
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## Qwen3: explicit-turn STT adapter
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## Qwen3: explicit-turn Realtime
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Start the backend, then the UI in a second terminal:
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The shared UI sends OpenAI Realtime events to `/v1/realtime?duplex=0` and keeps
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conversation history on the same connection:
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Start the turn-based backend:
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```bash
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vllm serve Qwen/Qwen3-Omni-30B-A3B-Instruct --omni --port 8091
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python -m examples.online_serving.qwen3_omni.realtime_web \
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--backend ws://127.0.0.1:8091 --stt --port 7863
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```
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Open `http://localhost:7863`, start a session, speak, and press **Send turn**.
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Wait for the answer to finish playing before speaking again. The UI opens a
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fresh STT connection for each turn; the displayed conversation is a local log,
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and **previous turns are not provided as model history**. The STT endpoint
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accepts only model selection, so system-prompt controls are hidden in this mode.
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In another terminal, check readiness and start the UI:
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The adapter follows the shipped `qwen3_omni/openai_realtime_client.py` and
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`vllm_omni/entrypoints/openai/realtime_connection.py`:
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```bash
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curl --fail http://127.0.0.1:8091/health
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python -m examples.online_serving.realtime_web --profile qwen3-turn \
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--backend ws://127.0.0.1:8091 --manual --port 7863
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```
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1. Explicit `duplex=0` selects the legacy STT handler on a turn deployment.
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It does not enable STT on a duplex deployment.
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2. Send `{type: "session.update", model: ...}` and `commit(final=false)`.
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3. Stream mono PCM16 at 16 kHz; **Send turn** flushes buffered audio, then sends `commit(final=true)`.
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4. `response.output_audio.delta.audio` contains PCM; `response.output_audio.done`
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terminates this adapter, including a response with no audio.
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5. The local implementation emits **model output** on `transcription.*`, so the
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UI displays it as assistant text. This differs from the early RFC's suggested
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user-transcript mapping. The VAD adapter also supports `response.output_text.*`
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and audio-transcript events without duplicating the answer.
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Speak, then press **Send turn**. The client commits the audio and requests a
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response; recording pauses while the reply plays.
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## Qwen3 with Server VAD (automatic turns)
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Run commands from the repository root with the vLLM-Omni environment activated.
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Prepare the pinned Silero v6.2 ONNX artifact as described in the
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[Qwen3 Server VAD instructions](../qwen3_omni/README.md#realtime-websocket-client-openai_realtime_clientpy).
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The backend requires ONNX Runtime and a compatible local artifact; setting
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`--vad` on the UI alone does not enable VAD on the backend.
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The plugin uses the pinned Silero v6.2 ONNX artifact from
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`istupakov/silero-vad-onnx` and does not download it while serving. Make it
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available in the Hugging Face cache or set `server_vad_model_path` to a local
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copy in the overlay below. The backend requires ONNX Runtime; setting `--vad`
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on the UI alone does not enable VAD.
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Create an overlay named `qwen3_vad.yaml`, replacing both paths with absolute paths
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on the backend host:
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@@ -124,7 +115,7 @@ In a second terminal, wait until the health check succeeds, then start the UI:
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```bash
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curl --fail http://127.0.0.1:8091/health
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python -m examples.online_serving.qwen3_omni.realtime_web \
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python -m examples.online_serving.realtime_web --profile qwen3-turn \
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--backend ws://127.0.0.1:8091 --vad --port 7863
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```
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@@ -148,21 +139,17 @@ The bundled duplex deploy allows four audio inputs and eight images per prompt.
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Use headphones to avoid speaker audio triggering VAD interruption. Reference
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voices and tool calls are not supported by this plugin.
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### Switching between VAD and manual turns
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### Using the Qwen3 web UI
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| Mode | Backend | UI flag | Submit a turn |
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| --- | --- | --- | --- |
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| Without VAD | Default Qwen turn deployment | `--stt` (default) | Press **Send turn** |
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| With VAD | Server VAD configuration and Silero artifact | `--vad` | Pause after speaking |
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The shared UI supports both explicit turns and Server VAD. Use the manual UI
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command above with the turn-based deployment, or start the VAD deployment above
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and launch the UI with `--vad`.
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Stop the existing UI process before starting another on the same port, then
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refresh the browser and reconnect. To use STT, start a backend with the default
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turn deployment and restart the UI with `--stt`. The duplex backend exposes
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Realtime duplex and ChatCompletion; it does not expose the legacy STT handler.
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To enable VAD on a default backend, first restart that backend with the overlay.
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Make `--backend` match the actual backend port; the UI and backend use separate
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ports. `Connection refused` means the target backend is unavailable: check its
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startup log and `/health` before connecting.
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refresh the browser and reconnect. To enable VAD on a default backend, restart
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that backend with the overlay. Make `--backend` match the actual backend port;
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the UI and backend use separate ports. `Connection refused` means the target
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backend is unavailable: check its startup log and `/health` before connecting.
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VAD connects with `duplex=1` and sends nested `session.audio.input` configuration
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with `create_response: true` and `interrupt_response: true`. It sends playback
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@@ -210,8 +197,7 @@ native camera protocol, where Stage 0 interleaves frames at unit boundaries.
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This is a supported subset, not full OpenAI Realtime API compatibility.
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`duplex=1`, `overlap_policy`, `playback.ack`, and `session.close` remain vLLM-Omni
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extensions. The legacy `--stt` mode is unchanged and is not the standard Realtime
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conversation API. This demo does not add WebRTC, semantic VAD, or tool calling.
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extensions. This demo does not add WebRTC, semantic VAD, or tool calling.
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## Shared host options
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@@ -219,11 +205,14 @@ The shared entry point can select a profile explicitly:
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```bash
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python -m examples.online_serving.realtime_web --profile qwen3-turn \
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--backend ws://127.0.0.1:8091 --stt --port 7863
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--backend ws://127.0.0.1:8091 --vad --port 7863
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```
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This Qwen3 UI profile requires the server-VAD deployment above.
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- `--backend` / `--ws-backend`: backend WebSocket origin (not the full `/v1/realtime` URL).
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- `--model`: override the wrapper's model name.
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- `--manual` / `--vad`: Qwen turn control; manual turns are the default.
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- `--public-realtime-url`: optional browser-visible WebSocket URL; otherwise the
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static host proxies `/v1/realtime` on the same origin.
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- `--host`, `--port`: UI bind address and port (default port 7862).
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@@ -245,7 +234,7 @@ pytest -m cpu tests/model_executor/models/qwen3_omni/test_duplex_plugin.py \
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tests/engine/test_duplex_orchestrator.py
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```
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These CPU tests cover profile messages, two STT turns, VAD terminal/drain ordering,
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These CPU tests cover profile message serialization, VAD terminal/drain ordering,
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backpressure, native capture during playback, server config injection, and shared
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worklets. Backend regressions cover current audio inclusion in history-based
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prompts, delayed playback ACK ordering, and no extra generation during silence
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@@ -2,24 +2,24 @@
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'use strict';
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const profiles = root.OmniRealtimeProfiles;
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profiles['qwen3-turn'] = (config) => {
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const vad = config.adapter === 'vad';
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const vad = config.turnMode === 'vad';
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let interruptOnSpeech = vad;
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let imageSeq = 0;
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let imageItems = [];
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// The engine refuses a ninth image (or 4 MiB of them) for the whole
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// session, so the camera track retires its own oldest frame instead.
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const MAX_IMAGE_ITEMS = 8;
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const hint = 'This Qwen deploy needs session_mode: duplex and Server VAD support; use --stt or enable Server VAD.';
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const hint = 'This Qwen deployment needs session_mode: duplex with Server VAD enabled.';
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return {
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title: 'Qwen3-Omni Voice', eyebrow: vad ? 'Engine-owned duplex call' : 'Turn-based realtime call',
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title: 'Qwen3-Omni Voice', eyebrow: vad ? 'Engine-owned duplex call' : 'Turn-based Realtime call',
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policy: vad ? 'Server VAD • interruptible responses' : 'Send turn • one response at a time',
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description: vad
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? 'Pause to send your turn; speak again to interrupt. Microphone upload continues during playback. Requires the Qwen duplex plugin and Silero. Optional camera frames accompany each spoken turn.'
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: 'Speak, then press Send turn. Microphone upload pauses while the model answers. Each turn uses a fresh connection; conversation history is not carried between turns. No barge-in or camera input.',
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: 'Speak, then press Send turn. Microphone upload pauses while the model answers. Conversation history stays on the Realtime session. No barge-in or camera input.',
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waiting: 'Waiting for you', camera: vad, cameraMaxDimension: 448, playbackAck: vad, clientCommit: !vad,
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waitForResponseDone: true, halfDuplex: !vad, deduplicateTranscript: true, closeSession: vad, reconnectEachTurn: !vad,
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inputSampleRate: vad ? 24000 : 16000,
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readyEvent: vad ? 'session.updated' : 'session.created', instructions: vad, sendIntervalMs: 200,
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waitForResponseDone: true, halfDuplex: !vad, deduplicateTranscript: true, closeSession: vad,
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inputSampleRate: 24000,
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readyEvent: 'session.updated', instructions: true, sendIntervalMs: 200,
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presets: { assistant: 'You are a helpful assistant. Answer clearly and concisely.' },
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url(config, location) {
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const url = profiles.url(config, location);
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@@ -33,22 +33,17 @@
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initialMessages(config, instructions) {
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imageItems = [];
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interruptOnSpeech = vad;
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if (!vad) return [
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{ type: 'session.update', model: config.model },
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{ type: 'input_audio_buffer.commit', final: false },
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];
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const session = { model: config.model, overlap_policy: 'barge_in_on_speech', audio: { input: {
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const session = { model: config.model, audio: { input: {
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format: { type: 'audio/pcm', rate: 24000 },
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turn_detection: { type: 'server_vad', threshold: 0.5, prefix_padding_ms: 300,
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silence_duration_ms: 500, create_response: true, interrupt_response: true },
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turn_detection: vad ? { type: 'server_vad', threshold: 0.5, prefix_padding_ms: 300,
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silence_duration_ms: 500, create_response: true, interrupt_response: true } : null,
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} } };
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if (vad) session.overlap_policy = 'barge_in_on_speech';
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else session.type = 'realtime';
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if (instructions) session.instructions = instructions;
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return [{ type: 'session.update', session }];
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},
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append(audio) {
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return vad ? { type: 'input_audio_buffer.append', audio }
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: { type: 'input_audio_buffer.append', audio, format: 'pcm16', sample_rate_hz: 16000 };
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},
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append: (audio) => ({ type: 'input_audio_buffer.append', audio }),
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// Qwen is a turn model: a frame is a picture added to the conversation,
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// not a track interleaved into the audio. That is exactly the OpenAI
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// Realtime image interface, so the camera sends conversation items
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@@ -68,7 +63,10 @@
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} });
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return messages;
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},
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commitMessages: () => vad ? [] : [{ type: 'input_audio_buffer.commit', final: true }],
|
||||
commitMessages: () => vad ? [] : [
|
||||
{ type: 'input_audio_buffer.commit' },
|
||||
{ type: 'response.create' },
|
||||
],
|
||||
ack: (responseId, playedMs) => vad ? { type: 'playback.ack', response_id: responseId,
|
||||
item_id: `item_${responseId}`, played_ms: playedMs, committed_ms: playedMs } : null,
|
||||
mapEvent(event) {
|
||||
@@ -87,30 +85,28 @@
|
||||
return { kind: 'interrupt' };
|
||||
}
|
||||
if (event.type === 'playback.acknowledged') return { kind: 'ack', committedMs: (event.event || event).committed_ms || 0 };
|
||||
if (event.type === 'response.output_text.delta' || event.type === 'transcription.delta') {
|
||||
// On the shipped STT route transcription.* is model-generated text,
|
||||
// not a separate ASR transcript of the user. See realtime_connection.py.
|
||||
if (event.type === 'response.output_text.delta') {
|
||||
return { kind: 'text', role: 'assistant', channel: 'text', text: event.delta || '' };
|
||||
}
|
||||
if (event.type === 'response.output_text.done' || event.type === 'transcription.done') {
|
||||
if (event.type === 'response.output_text.done') {
|
||||
return { kind: 'text-final', role: 'assistant', channel: 'text', text: event.text || '' };
|
||||
}
|
||||
if (vad && (event.type === 'output_audio_buffer.cleared' ||
|
||||
(event.type === 'response.done' && event.response?.status === 'cancelled'))) {
|
||||
return { kind: 'interrupt', responseId: event.response_id || event.response?.id || null };
|
||||
}
|
||||
if (event.type === 'input_audio_buffer.committed') return { kind: 'begin' };
|
||||
if (vad && event.type === 'input_audio_buffer.committed') return { kind: 'begin' };
|
||||
if (!vad && event.type === 'input_audio_buffer.cleared') return { kind: 'ignore' };
|
||||
if (event.type === 'input_audio_buffer.cleared') return { kind: 'backpressure', message: 'Input cleared. Model is answering; please wait.' };
|
||||
if (!vad && event.type === 'response.output_audio.done') return { kind: 'done' };
|
||||
const action = profiles.event(event);
|
||||
if (action.kind === 'error') {
|
||||
if (action.code === 'input_backpressure') return { ...action, kind: 'backpressure', message: 'Input buffer is full; please repeat dropped speech after the response.' };
|
||||
if (action.code === 'server_vad_initialization_failed') action.message = 'Server VAD could not load Silero. Configure the server artifact or restart this UI with --stt.';
|
||||
if (action.code === 'server_vad_initialization_failed') action.message = 'Server VAD could not load Silero. Configure the server artifact and restart the UI.';
|
||||
else if (vad && action.code === 'unsupported') action.message += ` ${hint}`;
|
||||
}
|
||||
return action;
|
||||
},
|
||||
connectionHint: vad ? hint : 'Check the Qwen /v1/realtime STT endpoint and model name.',
|
||||
connectionHint: vad ? hint : 'Check the Qwen turn-based /v1/realtime deployment and model name.',
|
||||
};
|
||||
};
|
||||
})(globalThis);
|
||||
@@ -74,12 +74,12 @@ def build_app(
|
||||
public_realtime_url: str | None = None,
|
||||
ref_audio: str | None = None,
|
||||
profile: str = "minicpm-native",
|
||||
adapter: str = "stt",
|
||||
turn_mode: str = "manual",
|
||||
) -> FastAPI:
|
||||
if profile not in {"minicpm-native", "qwen3-turn", "aura-ptt"}:
|
||||
raise ValueError(f"Unknown profile: {profile}")
|
||||
if adapter not in {"stt", "vad"}:
|
||||
raise ValueError(f"Unknown adapter: {adapter}")
|
||||
if turn_mode not in {"manual", "vad"}:
|
||||
raise ValueError(f"Unknown turn mode: {turn_mode}")
|
||||
if profile == "qwen3-turn" and ref_audio:
|
||||
raise ValueError("Qwen3 profile does not accept --ref-audio")
|
||||
if profile == "aura-ptt" and ref_audio:
|
||||
@@ -116,7 +116,7 @@ def build_app(
|
||||
{
|
||||
"model": model,
|
||||
"profile": profile,
|
||||
"adapter": adapter,
|
||||
"turnMode": turn_mode,
|
||||
"realtimePath": public_realtime_url or "v1/realtime",
|
||||
"refAudio": ref_audio_uri,
|
||||
"appVersion": app_version,
|
||||
@@ -190,9 +190,9 @@ def main(default_profile: str = "minicpm-native") -> None:
|
||||
parser.add_argument("--profile", choices=("minicpm-native", "qwen3-turn", "aura-ptt"), default=default_profile)
|
||||
parser.add_argument("--ws-backend", "--backend", dest="ws_backend")
|
||||
mode = parser.add_mutually_exclusive_group()
|
||||
mode.add_argument("--stt", dest="adapter", action="store_const", const="stt")
|
||||
mode.add_argument("--vad", dest="adapter", action="store_const", const="vad")
|
||||
parser.set_defaults(adapter="stt")
|
||||
mode.add_argument("--manual", dest="turn_mode", action="store_const", const="manual")
|
||||
mode.add_argument("--vad", dest="turn_mode", action="store_const", const="vad")
|
||||
parser.set_defaults(turn_mode="manual")
|
||||
parser.add_argument(
|
||||
"--public-realtime-url",
|
||||
help="Browser-visible ws:// or wss:// Realtime URL; defaults to the same-origin proxy.",
|
||||
@@ -225,7 +225,7 @@ def main(default_profile: str = "minicpm-native") -> None:
|
||||
public_realtime_url=args.public_realtime_url,
|
||||
ref_audio=args.ref_audio,
|
||||
profile=args.profile,
|
||||
adapter=args.adapter,
|
||||
turn_mode=args.turn_mode,
|
||||
),
|
||||
host=args.host,
|
||||
port=args.port,
|
||||
|
||||
Reference in new issue
Block a user