first commit

.github/workflows/validate.yml

@@ -0,0 +1,29 @@
+name: "Validation And Formatting"
+on:
+  push:
+  pull_request:
+  workflow_dispatch:
+  schedule:
+    - cron: '0 0 * * *'
+jobs:
+  hacs:
+    runs-on: "ubuntu-latest"
+    name: HACS
+    steps:
+      - name: Check out the repository
+        uses: "actions/checkout@v3"
+
+      - name: HACS validation
+        uses: "hacs/action@main"
+        with:
+          category: "integration"
+
+  hassfest:
+    runs-on: "ubuntu-latest"
+    name: Hassfest
+    steps:
+      - name: Check out the repository
+        uses: "actions/checkout@v3"
+
+      - name: Hassfest validation
+        uses: "home-assistant/actions/hassfest@master"

.gitignore

@@ -0,0 +1,3 @@
+.idea/
+*.mp3
+*.pyc

LICENSE

@@ -0,0 +1,201 @@
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README.md

@@ -0,0 +1,60 @@
+# Home Assistant 阿里云通义千问 STT 语音识别插件
+
+本插件集成了阿里云通义千问 (Qwen) 实时语音识别服务，为 Home Assistant 提供高精度的语音转文字功能。
+
+## 功能特性
+
+- **模型支持**：支持 `qwen3-asr-flash-realtime`（通义千问实时识别）。
+- **图形化配置**：支持通过 Home Assistant 前端界面直接添加集成。
+- **多语言支持**：支持中文、英文、粤语等多种语言识别。
+- **实时识别**：基于 WebSocket 协议，低延迟实时转写。
+- **多区域支持**：支持北京（国内）和新加坡（国际）节点。
+- **音频格式**：WAV（PCM 16-bit / 16kHz / 单声道）。
+
+## 安装说明
+
+### 方法一：手动安装
+
+1. 下载本项目代码。
+2. 将 `hass_stt_qwen` 文件夹复制到您的 Home Assistant 配置目录下的 `custom_components` 文件夹中。
+   - 最终路径应为：`/config/custom_components/hass_stt_qwen/`
+3. 重启 Home Assistant。
+
+## 配置指南
+
+1. 登录 Home Assistant。
+2. 点击左侧菜单的 **配置** -> **设备与服务**。
+3. 点击右下角的 **添加集成** 按钮。
+4. 搜索 **阿里通义千问语音识别** 并点击。
+5. 在弹出的配置窗口中输入以下信息：
+   - **API 密钥**: 您的阿里云 DashScope API 密钥。
+     - 获取方式：[阿里云百炼控制台](https://bailian.console.aliyun.com/)
+   - **模型**: 选择您需要使用的模型（推荐 `qwen3-asr-flash-realtime`）。
+   - **区域**: 根据您的 API Key 所在区域选择（北京或新加坡）。
+6. 点击 **提交** 完成配置。
+
+## 使用方法
+
+配置完成后，您可以在 Home Assistant 的语音助手设置中选择 **阿里通义千问语音识别** 作为语音转文字 (STT) 引擎。
+
+1. 进入 **配置** -> **语音助手**。
+2. 选择或创建一个语音助手。
+3. 在 **语音转文字** 选项中选择 **阿里通义千问语音识别**。
+
+## 常见问题
+
+**Q: 如何获取 API Key？**
+A: 请访问 [阿里云百炼控制台](https://bailian.console.aliyun.com/)，开通服务并创建 API Key。
+
+**Q: 插件支持哪些 Home Assistant 版本？**
+A: 需要 Home Assistant 支持 SpeechToTextEntity（STT 实体）的版本。
+
+**Q: 支持哪些语言？**
+A: 通义千问 Qwen ASR 支持中文、英文、粤语、日语等多种语言。
+
+**Q: 遇到连接失败怎么办？**
+A: 请检查 API Key 是否正确，以及网络连接是否正常。如果您在海外，请尝试切换到新加坡区域。
+
+**Q: 如何调整 VAD/超时/接入点？**
+A: 在集成的“选项”里可以配置自定义 WebSocket 接入点、是否启用 Server VAD、VAD 参数与超时。
+

custom_components/hass_stt_qwen/__init__.py

@@ -0,0 +1,25 @@
+"""Custom integration for Qwen ASR Speech-to-Text."""
+from __future__ import annotations
+
+from homeassistant.config_entries import ConfigEntry
+from homeassistant.core import HomeAssistant
+
+from .const import DOMAIN
+
+PLATFORMS = ["stt"]
+
+async def async_setup_entry(hass: HomeAssistant, entry: ConfigEntry) -> bool:
+    """Set up Qwen ASR from a config entry."""
+    hass.data.setdefault(DOMAIN, {})
+    hass.data[DOMAIN][entry.entry_id] = entry.data
+
+    await hass.config_entries.async_forward_entry_setups(entry, PLATFORMS)
+    return True
+
+async def async_unload_entry(hass: HomeAssistant, entry: ConfigEntry) -> bool:
+    """Unload a config entry."""
+    unload_ok = await hass.config_entries.async_unload_platforms(entry, PLATFORMS)
+    if unload_ok:
+        hass.data[DOMAIN].pop(entry.entry_id)
+
+    return unload_ok

custom_components/hass_stt_qwen/config_flow.py

@@ -0,0 +1,176 @@
+"""Config flow for Qwen ASR Speech-to-Text integration."""
+from __future__ import annotations
+
+import asyncio
+import logging
+from typing import Any
+
+from aiohttp import ClientError, WSMsgType, WSServerHandshakeError
+import voluptuous as vol
+
+from homeassistant import config_entries
+from homeassistant.data_entry_flow import FlowResult
+from homeassistant.helpers.aiohttp_client import async_get_clientsession
+
+from .const import (
+    CONF_API_KEY,
+    CONF_API_URL,
+    CONF_ENABLE_SERVER_VAD,
+    CONF_MODEL,
+    CONF_REGION,
+    CONF_TIMEOUT,
+    CONF_VAD_SILENCE_DURATION_MS,
+    CONF_VAD_THRESHOLD,
+    DEFAULT_API_URL_BEIJING,
+    DEFAULT_API_URL_SINGAPORE,
+    DEFAULT_MODEL,
+    DEFAULT_REGION,
+    DOMAIN,
+    SUPPORTED_MODELS,
+    SUPPORTED_REGIONS,
+)
+
+_LOGGER = logging.getLogger(__name__)
+
+STEP_USER_DATA_SCHEMA = vol.Schema(
+    {
+        vol.Required(CONF_API_KEY): str,
+        vol.Optional(CONF_MODEL, default=DEFAULT_MODEL): vol.In(SUPPORTED_MODELS),
+        vol.Optional(CONF_REGION, default=DEFAULT_REGION): vol.In(SUPPORTED_REGIONS),
+    }
+)
+
+STEP_OPTIONS_DATA_SCHEMA = vol.Schema(
+    {
+        vol.Optional(CONF_API_URL): str,
+        vol.Optional(CONF_ENABLE_SERVER_VAD, default=False): bool,
+        vol.Optional(CONF_VAD_THRESHOLD, default=0.0): vol.Coerce(float),
+        vol.Optional(CONF_VAD_SILENCE_DURATION_MS, default=400): vol.Coerce(int),
+        vol.Optional(CONF_TIMEOUT, default=30): vol.Coerce(int),
+    }
+)
+
+class QwenSTTConfigFlow(config_entries.ConfigFlow, domain=DOMAIN):
+    """Handle a config flow for Qwen ASR Speech-to-Text."""
+
+    VERSION = 1
+
+
+    async def _async_validate(self, user_input: dict[str, Any]) -> None:
+        api_key = user_input[CONF_API_KEY]
+        model = user_input.get(CONF_MODEL, DEFAULT_MODEL)
+        region = user_input.get(CONF_REGION, DEFAULT_REGION)
+        api_url = (
+            DEFAULT_API_URL_SINGAPORE if region == "singapore" else DEFAULT_API_URL_BEIJING
+        )
+        uri = f"{api_url}/realtime?model={model}"
+        headers = {
+            "Authorization": f"Bearer {api_key}",
+            "OpenAI-Beta": "realtime=v1",
+        }
+
+        session = async_get_clientsession(self.hass)
+        async with session.ws_connect(uri, headers=headers, heartbeat=30) as ws:
+            await ws.send_json(
+                {
+                    "event_id": "event_validate",
+                    "type": "session.update",
+                    "session": {
+                        "modalities": ["text"],
+                        "input_audio_format": "pcm",
+                        "sample_rate": 16000,
+                        "input_audio_transcription": {"language": "zh"},
+                        "turn_detection": None,
+                    },
+                }
+            )
+            async with asyncio.timeout(5):
+                async for msg in ws:
+                    if msg.type != WSMsgType.TEXT:
+                        continue
+                    try:
+                        data = msg.json()
+                    except Exception:
+                        continue
+                    msg_type = data.get("type")
+                    if msg_type in ("session.created", "session.updated"):
+                        return
+                    if msg_type == "error":
+                        raise ClientError(str(data.get("error", {})))
+
+    async def async_step_user(
+        self, user_input: dict[str, Any] | None = None
+    ) -> FlowResult:
+        """Handle the initial step."""
+        if user_input is None:
+            return self.async_show_form(
+                step_id="user", data_schema=STEP_USER_DATA_SCHEMA
+            )
+
+        errors: dict[str, str] = {}
+
+        # Check if already configured
+        if self._async_current_entries():
+            return self.async_abort(reason="already_configured")
+
+        try:
+            await self._async_validate(user_input)
+        except WSServerHandshakeError as err:
+            if err.status in (401, 403):
+                errors["base"] = "invalid_auth"
+            else:
+                errors["base"] = "cannot_connect"
+        except TimeoutError:
+            errors["base"] = "cannot_connect"
+        except ClientError:
+            errors["base"] = "cannot_connect"
+        except Exception:
+            _LOGGER.exception("Unexpected error validating Qwen ASR configuration")
+            errors["base"] = "cannot_connect"
+
+        if errors:
+            return self.async_show_form(
+                step_id="user",
+                data_schema=STEP_USER_DATA_SCHEMA,
+                errors=errors,
+            )
+
+        return self.async_create_entry(title="Qwen ASR", data=user_input)
+
+    @staticmethod
+    def async_get_options_flow(config_entry: config_entries.ConfigEntry) -> config_entries.OptionsFlow:
+        return QwenSTTOptionsFlow(config_entry)
+
+
+class QwenSTTOptionsFlow(config_entries.OptionsFlow):
+    def __init__(self, config_entry: config_entries.ConfigEntry) -> None:
+        super().__init__(config_entry)
+
+    async def async_step_init(self, user_input: dict[str, Any] | None = None) -> FlowResult:
+        if user_input is not None:
+            return self.async_create_entry(title="", data=user_input)
+
+        opts = self.config_entry.options
+        schema = vol.Schema(
+            {
+                vol.Optional(CONF_API_URL, default=opts.get(CONF_API_URL, "")): str,
+                vol.Optional(
+                    CONF_ENABLE_SERVER_VAD,
+                    default=opts.get(CONF_ENABLE_SERVER_VAD, False),
+                ): bool,
+                vol.Optional(
+                    CONF_VAD_THRESHOLD,
+                    default=opts.get(CONF_VAD_THRESHOLD, 0.0),
+                ): vol.Coerce(float),
+                vol.Optional(
+                    CONF_VAD_SILENCE_DURATION_MS,
+                    default=opts.get(CONF_VAD_SILENCE_DURATION_MS, 400),
+                ): vol.Coerce(int),
+                vol.Optional(
+                    CONF_TIMEOUT,
+                    default=opts.get(CONF_TIMEOUT, 30),
+                ): vol.Coerce(int),
+            }
+        )
+
+        return self.async_show_form(step_id="init", data_schema=schema)

custom_components/hass_stt_qwen/const.py

@@ -0,0 +1,48 @@
+"""Constants for the Qwen ASR Speech-to-Text integration."""
+
+DOMAIN = "hass_stt_qwen"
+
+CONF_API_KEY = "api_key"
+CONF_API_URL = "api_url"
+CONF_MODEL = "model"
+CONF_REGION = "region"
+CONF_ENABLE_SERVER_VAD = "enable_server_vad"
+CONF_VAD_THRESHOLD = "vad_threshold"
+CONF_VAD_SILENCE_DURATION_MS = "vad_silence_duration_ms"
+CONF_TIMEOUT = "timeout"
+
+DEFAULT_API_URL_BEIJING = "wss://dashscope.aliyuncs.com/api-ws/v1"
+DEFAULT_API_URL_SINGAPORE = "wss://dashscope-intl.aliyuncs.com/api-ws/v1"
+DEFAULT_MODEL = "qwen3-asr-flash-realtime"
+DEFAULT_REGION = "beijing"
+
+SUPPORTED_MODELS = [
+    "qwen3-asr-flash-realtime",
+    "qwen3-asr-flash-realtime-2025-10-27"
+]
+
+SUPPORTED_REGIONS = [
+    "beijing",
+    "singapore",
+]
+
+# Qwen ASR supports multiple languages
+SUPPORTED_LANGUAGES = [
+    "zh",  # Chinese
+    "en",  # English
+    "yue",  # Cantonese
+    "ja",  # Japanese
+    "ko",  # Korean
+    "es",  # Spanish
+    "fr",  # French
+    "de",  # German
+    "it",  # Italian
+    "pt",  # Portuguese
+    "ru",  # Russian
+    "ar",  # Arabic
+    "tr",  # Turkish
+    "vi",  # Vietnamese
+    "th",  # Thai
+    "id",  # Indonesian
+    "ms",  # Malay
+]

custom_components/hass_stt_qwen/manifest.json

@@ -0,0 +1,10 @@
+{
+  "domain": "hass_stt_qwen",
+  "name": "阿里通义千问语音识别",
+  "codeowners": ["@CircleLiu"],
+  "config_flow": true,
+  "documentation": "https://bbs.nextrt.com/d/2-home-assistant-a-li-yun-tong-yi-qian-wen-stt-yu-yin-shi-bie-cha-jian",
+  "iot_class": "cloud_push",
+  "issue_tracker": "https://bbs.nextrt.com/d/2-home-assistant-a-li-yun-tong-yi-qian-wen-stt-yu-yin-shi-bie-cha-jian",
+  "version": "0.1.0"
+}

custom_components/hass_stt_qwen/qwen3_asr_client.py

@@ -0,0 +1,372 @@
+"""WebSocket client for Qwen3 ASR."""
+
+from __future__ import annotations
+
+import asyncio
+import base64
+from collections.abc import AsyncIterable
+import json
+import logging
+import time
+from typing import Final
+
+from aiohttp import ClientError, WSCloseCode, WSMsgType
+
+from homeassistant.components.stt import SpeechMetadata, SpeechResult, SpeechResultState
+
+_LOGGER = logging.getLogger(__name__)
+
+# Maximum time to wait for a response (in seconds)
+WEBSOCKET_TIMEOUT: Final = 30
+
+
+class Qwen3AsrClientError(Exception):
+    pass
+
+
+class Qwen3AsrClientTimeout(Qwen3AsrClientError):
+    pass
+
+
+class Qwen3AsrClient:
+    """WebSocket client for Qwen3 ASR API."""
+
+    def __init__(
+        self,
+        client,
+        api_key: str,
+        api_url: str,
+        model: str,
+    ) -> None:
+        """Initialize the WebSocket client."""
+        self.client = client
+        self.api_key = api_key
+        self.api_url = api_url
+        self.model = model
+        self.start_time = 0.0
+        self.ws = None
+        self.timeout = WEBSOCKET_TIMEOUT
+        self.enable_server_vad = False
+        self.vad_threshold = 0.0
+        self.vad_silence_duration_ms = 400
+
+    def _new_event_id(self) -> str:
+        return f"event_{int(time.time() * 1000)}"
+
+    def _normalize_language(self, language: str | None) -> str:
+        if not language:
+            return "zh"
+        lower = language.lower()
+        if lower.startswith("zh"):
+            return "zh"
+        return lower
+
+    async def _iter_pcm16(self, stream: AsyncIterable[bytes]) -> AsyncIterable[bytes]:
+        buf = bytearray()
+        state = "probe"
+        pos = 0
+        data_remaining: int | None = None
+
+        async for chunk in stream:
+            if state == "raw":
+                if chunk:
+                    yield chunk
+                continue
+
+            if chunk:
+                buf.extend(chunk)
+
+            if state == "probe":
+                if len(buf) < 12:
+                    continue
+                if buf[:4] != b"RIFF" or buf[8:12] != b"WAVE":
+                    yield bytes(buf)
+                    buf.clear()
+                    state = "raw"
+                    continue
+                state = "wav_chunks"
+                pos = 12
+
+            if state == "wav_chunks":
+                while True:
+                    if len(buf) < pos + 8:
+                        break
+                    chunk_id = bytes(buf[pos : pos + 4])
+                    chunk_size = int.from_bytes(buf[pos + 4 : pos + 8], "little")
+                    end = pos + 8 + chunk_size + (chunk_size % 2)
+                    if len(buf) < end:
+                        break
+                    if chunk_id == b"data":
+                        del buf[: pos + 8]
+                        data_remaining = chunk_size
+                        state = "wav_data"
+                        break
+                    pos = end
+                    if pos > 65536:
+                        yield bytes(buf)
+                        buf.clear()
+                        state = "raw"
+                        break
+
+            if state == "wav_data" and data_remaining is not None:
+                while buf and data_remaining > 0:
+                    take = min(len(buf), data_remaining)
+                    yield bytes(buf[:take])
+                    del buf[:take]
+                    data_remaining -= take
+                if data_remaining == 0:
+                    if buf:
+                        yield bytes(buf)
+                        buf.clear()
+                    state = "raw"
+
+    async def _send_audio_stream(self, stream: AsyncIterable[bytes]) -> None:
+        """Send audio chunks to WebSocket server."""
+        try:
+            chunk_count = 0
+            async for chunk in self._iter_pcm16(stream):
+                if not chunk or self.ws.closed:
+                    _LOGGER.debug("Stopping audio send: empty chunk or closed connection")
+                    break
+                # Audio data must be base64 encoded
+                b64 = base64.b64encode(chunk).decode("utf-8")
+                await self.ws.send_json(
+                    {
+                        "event_id": self._new_event_id(),
+                        "type": "input_audio_buffer.append",
+                        "audio": b64,
+                    }
+                )
+                chunk_count += 1
+                _LOGGER.debug("Audio chunk #%d sent (%d bytes)", chunk_count, len(chunk))
+
+            if not self.ws.closed:
+                # Signal the end of the audio stream to the server
+                _LOGGER.info("Sent %d audio chunks, sending finish", chunk_count)
+                if not self.enable_server_vad:
+                    await self.ws.send_json(
+                        {
+                            "event_id": self._new_event_id(),
+                            "type": "input_audio_buffer.commit",
+                        }
+                    )
+                await self.ws.send_json(
+                    {
+                        "event_id": self._new_event_id(),
+                        "type": "session.finish",
+                    }
+                )
+
+                # Set start time after sending all audio data
+                self.start_time = time.perf_counter()
+
+        except asyncio.CancelledError:
+            _LOGGER.debug("send_audio() was cancelled")
+            raise
+        except Exception:
+            _LOGGER.exception("Error sending audio")
+            if not self.ws.closed:
+                await self.ws.close(
+                    code=WSCloseCode.INTERNAL_ERROR,
+                    message=b"Error sending audio",
+                )
+            raise
+
+    async def _receive_transcription(self, send_task: asyncio.Task) -> str:
+        """Receive transcription results from WebSocket server."""
+        final_text: str | None = None
+        try:
+            async with asyncio.timeout(self.timeout):
+                async for msg in self.ws:
+                    if msg.type == WSMsgType.TEXT:
+                        data = json.loads(msg.data)
+                        msg_type = data.get("type")
+                        _LOGGER.debug("Received message type: %s", msg_type)
+
+                        if msg_type == "session.created":
+                            _LOGGER.info("Session created: %s", data.get("session", {}).get("id"))
+                        elif msg_type == "session.updated":
+                            _LOGGER.info("Session updated")
+                        elif msg_type == "conversation.item.input_audio_transcription.text":
+                            text = data.get("text") or data.get("stash") or ""
+                            if text:
+                                _LOGGER.debug('Intermediate transcription: "%s"', text)
+                        elif msg_type == "conversation.item.input_audio_transcription.completed":
+                            # Get final transcription
+                            final_text = (data.get("transcript") or data.get("text") or "").strip()
+                            if self.start_time > 0:
+                                duration = time.perf_counter() - self.start_time
+                                _LOGGER.info(
+                                    "Transcription processing duration: %.2f seconds",
+                                    duration,
+                                )
+                            _LOGGER.info('Final transcription received: "%s"', final_text)
+                            return final_text or ""
+                        elif msg_type == "input_audio_buffer.speech_started":
+                            _LOGGER.info("Speech started")
+                        elif msg_type == "input_audio_buffer.speech_stopped":
+                            _LOGGER.info("Speech stopped")
+                        elif msg_type == "input_audio_buffer.committed":
+                            _LOGGER.info("Audio buffer committed")
+                        elif msg_type == "session.finished":
+                            final_text = (data.get("transcript") or final_text or "").strip()
+                            _LOGGER.info('Session finished: "%s"', final_text)
+                            return final_text or ""
+                        elif msg_type == "error":
+                            error_msg = data.get("error", {})
+                            raise Qwen3AsrClientError(str(error_msg))
+                        else:
+                            _LOGGER.debug("Unhandled message type: %s, data: %s", msg_type, data)
+                    elif msg.type == WSMsgType.BINARY:
+                        _LOGGER.warning("Received unexpected binary message (%d bytes)", len(msg.data))
+                    elif msg.type == WSMsgType.ERROR:
+                        raise Qwen3AsrClientError(str(self.ws.exception()))
+                    elif msg.type == WSMsgType.CLOSED:
+                        _LOGGER.info("WebSocket closed by server")
+                        break
+                    else:
+                        _LOGGER.debug("Received message of type: %s", msg.type)
+        except TimeoutError:
+            raise Qwen3AsrClientTimeout("Timeout waiting for transcription response")
+        except asyncio.CancelledError:
+            _LOGGER.debug("receive_transcription() was cancelled")
+            raise
+        except Qwen3AsrClientError:
+            raise
+        except Exception as err:
+            raise Qwen3AsrClientError(str(err)) from err
+        finally:
+            if not send_task.done():
+                send_task.cancel()
+
+        if final_text is not None:
+            return final_text
+        raise Qwen3AsrClientError("WebSocket closed before transcription completed")
+
+    def _create_session_config(self, metadata: SpeechMetadata) -> dict:
+        """Create configuration for the transcription session."""
+        language = self._normalize_language(metadata.language)
+        if self.enable_server_vad:
+            turn_detection = {
+                "type": "server_vad",
+                "threshold": self.vad_threshold,
+                "silence_duration_ms": self.vad_silence_duration_ms,
+            }
+        else:
+            turn_detection = None
+        config = {
+            "event_id": self._new_event_id(),
+            "type": "session.update",
+            "session": {
+                "modalities": ["text"],
+                "input_audio_format": "pcm",
+                "sample_rate": 16000,
+                "input_audio_transcription": {
+                    "language": language,
+                },
+                "turn_detection": turn_detection,
+            },
+        }
+        return config
+
+    async def _handle_tasks(
+        self, send_task: asyncio.Task, recv_task: asyncio.Task
+    ) -> None:
+        """Handle task completion and cancellation logic."""
+        try:
+            done, pending = await asyncio.wait(
+                [send_task, recv_task],
+                return_when=asyncio.FIRST_COMPLETED,
+            )
+
+            # Handle task completion and cancellation
+            if recv_task in done:
+                _LOGGER.debug("Transcription finished - cancelling audio task")
+                if not send_task.done():
+                    send_task.cancel()
+                await asyncio.gather(send_task, return_exceptions=True)
+            elif send_task in done:
+                _LOGGER.debug("Audio finished - waiting for final transcription")
+                await recv_task
+            else:
+                _LOGGER.warning(
+                    "Unexpected state in task completion, ensuring tasks are awaited/cancelled"
+                )
+                for task in pending:
+                    if not task.done():
+                        task.cancel()
+                await asyncio.gather(*pending, return_exceptions=True)
+
+            for task in done:
+                if task.exception():
+                    _LOGGER.error("Task completed with exception: %s", task.exception())
+        finally:
+            if not self.ws.closed:
+                try:
+                    await self.ws.close()
+                    _LOGGER.debug("WebSocket closed cleanly")
+                except Exception:
+                    _LOGGER.exception("Error closing WebSocket connection")
+
+    async def async_process_audio_stream(
+        self, metadata: SpeechMetadata, stream: AsyncIterable[bytes]
+    ) -> SpeechResult:
+        """Process audio stream via WebSocket to Qwen3 ASR Realtime API."""
+
+        # Construct WebSocket URL for realtime API
+        uri = f"{self.api_url}/realtime?model={self.model}"
+        headers = {
+            "Authorization": f"Bearer {self.api_key}",
+            "OpenAI-Beta": "realtime=v1",
+        }
+
+        _LOGGER.info(
+            "Starting Qwen3 ASR transcription: language=%s, format=%s, sample_rate=%s, channels=%s",
+            metadata.language,
+            metadata.format,
+            metadata.sample_rate,
+            metadata.channel,
+        )
+
+        try:
+            _LOGGER.debug("Opening WebSocket connection to %s", uri)
+            async with self.client.ws_connect(uri, headers=headers, heartbeat=30) as ws:
+                self.ws = ws
+                self.start_time = 0  # Reset start_time
+
+                config = self._create_session_config(metadata)
+                _LOGGER.debug("Sending session configuration: %s", config)
+                await ws.send_json(config)
+
+                # Create and manage concurrent tasks
+                send_task = asyncio.create_task(self._send_audio_stream(stream))
+                recv_task = asyncio.create_task(self._receive_transcription(send_task))
+
+                # Handle tasks completion
+                await self._handle_tasks(send_task, recv_task)
+
+                # Process final result
+                if not recv_task.done() or recv_task.cancelled():
+                    return SpeechResult("", SpeechResultState.ERROR)
+
+                exc = recv_task.exception()
+                if exc:
+                    _LOGGER.error("Transcription failed: %s", exc)
+                    return SpeechResult("", SpeechResultState.ERROR)
+
+                final_text = recv_task.result().strip()
+
+                _LOGGER.info('Transcription completed successfully: "%s"', final_text)
+
+                if not final_text:
+                    _LOGGER.warning("Qwen3 ASR transcription resulted in empty text")
+                    return SpeechResult("", SpeechResultState.SUCCESS)
+
+                return SpeechResult(final_text, SpeechResultState.SUCCESS)
+
+        except ClientError as err:
+            _LOGGER.error("WebSocket connection error: %s", err)
+            return SpeechResult("", SpeechResultState.ERROR)
+        except Exception:
+            _LOGGER.exception("Unexpected error in WebSocket communication")
+            return SpeechResult("", SpeechResultState.ERROR)

custom_components/hass_stt_qwen/stt.py

@@ -0,0 +1,181 @@
+"""Setting up QwenSTTProvider."""
+from __future__ import annotations
+
+from collections.abc import AsyncIterable
+import logging
+
+from homeassistant.components.stt import (
+    AudioBitRates,
+    AudioChannels,
+    AudioCodecs,
+    AudioFormats,
+    AudioSampleRates,
+    SpeechMetadata,
+    SpeechResult,
+    SpeechResultState,
+)
+from homeassistant.components.stt import SpeechToTextEntity
+
+from homeassistant.config_entries import ConfigEntry
+from homeassistant.core import HomeAssistant
+from homeassistant.helpers.aiohttp_client import async_get_clientsession
+from homeassistant.helpers.entity_platform import AddEntitiesCallback
+
+from .const import (
+    CONF_API_KEY,
+    CONF_API_URL,
+    CONF_ENABLE_SERVER_VAD,
+    CONF_MODEL,
+    CONF_REGION,
+    CONF_TIMEOUT,
+    CONF_VAD_SILENCE_DURATION_MS,
+    CONF_VAD_THRESHOLD,
+    DEFAULT_API_URL_BEIJING,
+    DEFAULT_API_URL_SINGAPORE,
+    DEFAULT_MODEL,
+    DEFAULT_REGION,
+    DOMAIN,
+    SUPPORTED_LANGUAGES,
+)
+from .qwen3_asr_client import Qwen3AsrClient
+
+_LOGGER = logging.getLogger(__name__)
+
+async def async_setup_entry(
+    hass: HomeAssistant,
+    config_entry: ConfigEntry,
+    async_add_entities: AddEntitiesCallback,
+) -> None:
+    """Set up Qwen STT from a config entry."""
+    data = config_entry.data
+    options = config_entry.options
+
+    api_key = data[CONF_API_KEY]
+    model = data.get(CONF_MODEL, DEFAULT_MODEL)
+    region = data.get(CONF_REGION, DEFAULT_REGION)
+
+    api_url = options.get(CONF_API_URL) or DEFAULT_API_URL_SINGAPORE if region == "singapore" else DEFAULT_API_URL_BEIJING
+    enable_server_vad = options.get(CONF_ENABLE_SERVER_VAD, False)
+    vad_threshold = options.get(CONF_VAD_THRESHOLD, 0.0)
+    vad_silence_duration_ms = options.get(CONF_VAD_SILENCE_DURATION_MS, 400)
+    timeout = options.get(CONF_TIMEOUT, 30)
+
+    async_add_entities(
+        [
+            QwenSTTEntity(
+                hass,
+                config_entry,
+                api_key,
+                api_url,
+                model,
+                enable_server_vad=enable_server_vad,
+                vad_threshold=vad_threshold,
+                vad_silence_duration_ms=vad_silence_duration_ms,
+                timeout=timeout,
+            )
+        ]
+    )
+
+
+class QwenSTTEntity(SpeechToTextEntity):
+    """The Qwen STT entity."""
+
+    def __init__(
+        self,
+        hass: HomeAssistant,
+        config_entry: ConfigEntry,
+        api_key: str,
+        api_url: str,
+        model: str,
+        *,
+        enable_server_vad: bool,
+        vad_threshold: float,
+        vad_silence_duration_ms: int,
+        timeout: int,
+    ) -> None:
+        """Init Qwen STT service."""
+        self.hass = hass
+        self._config_entry = config_entry
+        self._api_key = api_key
+        self._api_url = api_url
+        self._model = model
+        self._enable_server_vad = enable_server_vad
+        self._vad_threshold = vad_threshold
+        self._vad_silence_duration_ms = vad_silence_duration_ms
+        self._timeout = timeout
+        self._client = self._create_client()
+        
+        self._attr_name = f"Qwen ASR ({model})"
+        self._attr_unique_id = f"{config_entry.entry_id}_{model}"
+
+    @property
+    def supported_languages(self) -> list[str]:
+        """Return a list of supported languages."""
+        return SUPPORTED_LANGUAGES
+
+    @property
+    def supported_formats(self) -> list[AudioFormats]:
+        """Return a list of supported formats."""
+        return [AudioFormats.WAV]
+
+    @property
+    def supported_codecs(self) -> list[AudioCodecs]:
+        """Return a list of supported codecs."""
+        return [AudioCodecs.PCM]
+
+    @property
+    def supported_bit_rates(self) -> list[AudioBitRates]:
+        """Return a list of supported bitrates."""
+        return [AudioBitRates.BITRATE_16]
+
+    @property
+    def supported_sample_rates(self) -> list[AudioSampleRates]:
+        """Return a list of supported samplerates."""
+        return [AudioSampleRates.SAMPLERATE_16000]
+
+    @property
+    def supported_channels(self) -> list[AudioChannels]:
+        """Return a list of supported channels."""
+        return [AudioChannels.CHANNEL_MONO]
+
+    def _create_client(self):
+        """Create and return the appropriate client based on model."""
+        session = async_get_clientsession(self.hass)
+
+        if self._model.startswith("qwen3-asr"):
+            client = Qwen3AsrClient(
+                session,
+                self._api_key,
+                self._api_url,
+                self._model,
+            )
+            client.enable_server_vad = self._enable_server_vad
+            client.vad_threshold = self._vad_threshold
+            client.vad_silence_duration_ms = self._vad_silence_duration_ms
+            client.timeout = self._timeout
+            return client
+        else:
+            _LOGGER.error("Unsupported model: %s", self._model)
+            raise ValueError(f"Unsupported model: {self._model}")
+
+    async def async_process_audio_stream(
+        self, metadata: SpeechMetadata, stream: AsyncIterable[bytes]
+    ) -> SpeechResult:
+        """Process audio stream using the configured client."""
+        if (
+            metadata.format not in self.supported_formats
+            or metadata.codec not in self.supported_codecs
+            or metadata.bit_rate not in self.supported_bit_rates
+            or metadata.sample_rate not in self.supported_sample_rates
+            or metadata.channel not in self.supported_channels
+        ):
+            _LOGGER.error(
+                "Unsupported audio metadata: format=%s codec=%s bit_rate=%s sample_rate=%s channel=%s",
+                metadata.format,
+                metadata.codec,
+                metadata.bit_rate,
+                metadata.sample_rate,
+                metadata.channel,
+            )
+            return SpeechResult("", state=SpeechResultState.ERROR)
+        return await self._client.async_process_audio_stream(metadata, stream)

custom_components/hass_stt_qwen/translations/en.json

@@ -0,0 +1,36 @@
+{
+  "config": {
+    "step": {
+      "user": {
+        "title": "Connect to Qwen ASR",
+        "description": "Enter your Alibaba Cloud DashScope API details.",
+        "data": {
+          "api_key": "API Key",
+          "model": "Model",
+          "region": "Region"
+        }
+      }
+    },
+    "error": {
+      "cannot_connect": "Failed to connect",
+      "invalid_auth": "Invalid authentication"
+    },
+    "abort": {
+      "already_configured": "Service is already configured"
+    }
+  },
+  "options": {
+    "step": {
+      "init": {
+        "title": "Qwen ASR options",
+        "data": {
+          "api_url": "WebSocket endpoint (optional)",
+          "enable_server_vad": "Enable Server VAD",
+          "vad_threshold": "VAD threshold",
+          "vad_silence_duration_ms": "VAD silence duration (ms)",
+          "timeout": "Timeout (seconds)"
+        }
+      }
+    }
+  }
+}

custom_components/hass_stt_qwen/translations/zh-Hans.json

@@ -0,0 +1,36 @@
+{
+  "config": {
+    "step": {
+      "user": {
+        "title": "连接到阿里云 Qwen ASR",
+        "description": "请输入您的阿里云 DashScope API 信息。",
+        "data": {
+          "api_key": "API 密钥",
+          "model": "模型",
+          "region": "区域"
+        }
+      }
+    },
+    "error": {
+      "cannot_connect": "连接失败",
+      "invalid_auth": "无效的身份验证"
+    },
+    "abort": {
+      "already_configured": "服务已配置"
+    }
+  },
+  "options": {
+    "step": {
+      "init": {
+        "title": "Qwen ASR 选项",
+        "data": {
+          "api_url": "WebSocket 接入点（可选）",
+          "enable_server_vad": "启用 Server VAD",
+          "vad_threshold": "VAD 阈值",
+          "vad_silence_duration_ms": "VAD 静默时长（毫秒）",
+          "timeout": "超时（秒）"
+        }
+      }
+    }
+  }
+}

hacs.json

@@ -0,0 +1,5 @@
+{
+    "name": "阿里语音识别",
+    "render_readme": true,
+    "country": ["CN"]
+}