fix: 修复gewechat无法at人和发语音失败的问题 #447 #438

.gitignore

@@ -21,4 +21,5 @@ node_modules/
 .DS_Store
 package-lock.json
 package.json
-venv/*
+venv/*
+packages/python_interpreter/workplace

astrbot/core/platform/sources/gewechat/client.py

@@ -312,12 +312,14 @@ class SimpleGewechatClient():
             self.appid = appid
             logger.info(f"已保存 APPID: {appid}")
     
-    async def post_text(self, to_wxid, content: str):
+    async def post_text(self, to_wxid, content: str, ats: str = ""):
         payload = {
             "appId": self.appid,
             "toWxid": to_wxid,
             "content": content,
         }
+        if ats:
+            payload['ats'] = ats
         
         async with aiohttp.ClientSession() as session:
             async with session.post(

astrbot/core/platform/sources/gewechat/gewechat_event.py

@@ -6,7 +6,7 @@ from astrbot.core.utils.tencent_record_helper import wav_to_tencent_silk
 from astrbot.api import logger
 from astrbot.api.event import AstrMessageEvent, MessageChain
 from astrbot.api.platform import AstrBotMessage, PlatformMetadata
-from astrbot.api.message_components import Plain, Image, Record
+from astrbot.api.message_components import Plain, Image, Record, At, File, Reply
 from .client import SimpleGewechatClient
 
 def get_wav_duration(file_path):
@@ -15,6 +15,8 @@ def get_wav_duration(file_path):
         n_channels, sampwidth, framerate, n_frames = wav_file.getparams()[:4]
         if n_frames == 2147483647:
             duration = (file_size - 44) / (n_channels * sampwidth * framerate)
+        elif n_frames == 0:
+            duration = (file_size - 44) / (n_channels * sampwidth * framerate)
         else:
             duration = n_frames / float(framerate)
         return duration
@@ -43,9 +45,31 @@ class GewechatPlatformEvent(AstrMessageEvent):
             logger.error("无法获取到 to_wxid。")
             return
         
+        #  检查@
+        ats = []
+        ats_names = []
+        for comp in message.chain:
+            if isinstance(comp, At):
+                ats.append(comp.qq)
+                ats_names.append(comp.name)
+        has_at = False
+        
         for comp in message.chain:
             if isinstance(comp, Plain):
-                await self.client.post_text(to_wxid, comp.text)
+                text = comp.text
+                payload = {
+                    "to_wxid": to_wxid,
+                    "content": text,
+                }
+                if not has_at and ats:
+                    ats = f"{','.join(ats)}"
+                    ats_names = f"@{' @'.join(ats_names)}"
+                    text = f"{ats_names} {text}"
+                    payload["content"] = text
+                    payload["ats"] = ats
+                    has_at = True
+                await self.client.post_text(**payload)
+                
             elif isinstance(comp, Image):
                 img_url = comp.file
                 img_path = ""
@@ -80,6 +104,7 @@ class GewechatPlatformEvent(AstrMessageEvent):
                     record_path = record_url
                     
                 silk_path = f"data/temp/{uuid.uuid4()}.silk"
+                logger.info("开始转换语音文件: " + record_path)
                 duration = await wav_to_tencent_silk(record_path, silk_path)
                 
                 print(f"duration: {duration}, {silk_path}")

astrbot/core/provider/entites.py

@@ -52,7 +52,7 @@ class ProviderRequest():
 @dataclass
 class LLMResponse:
     role: str
-    '''角色'''
+    '''角色, assistant, tool, err'''
     completion_text: str = ""
     '''LLM 返回的文本'''
     tools_call_args: List[Dict[str, any]] = field(default_factory=list)

astrbot/core/utils/tencent_record_helper.py

@@ -22,21 +22,27 @@ async def tencent_silk_to_wav(silk_path: str, output_path: str) -> str:
 
 async def wav_to_tencent_silk(wav_path: str, output_path: str) -> int:
     '''返回 duration'''
-    import pysilk
-
-    with wave.open(wav_path, 'rb') as wav:
-        wav_data = wav.readframes(wav.getnframes())
-        wav_data = BytesIO(wav_data)
-        output_io = BytesIO()
-        pysilk.encode(wav_data, output_io, 24000, 24000)
-        output_io.seek(0)
+    try:
+        import pilk
+    except (ImportError, ModuleNotFoundError) as _:
+        raise Exception("pilk 模块未安装，请前往管理面板->控制台->安装pip库 安装 pilk 这个库")
+    # with wave.open(wav_path, 'rb') as wav:
+    #     wav_data = wav.readframes(wav.getnframes())
+    #     wav_data = BytesIO(wav_data)
+    #     output_io = BytesIO()
+    #     pysilk.encode(wav_data, output_io, 24000, 24000)
+    #     output_io.seek(0)
         
-        # 在首字节添加 \x02,去除结尾的\xff\xff
-        silk_data = output_io.read()
-        silk_data_with_prefix = b'\x02' + silk_data[:-2]
+    #     # 在首字节添加 \x02,去除结尾的\xff\xff
+    #     silk_data = output_io.read()
+    #     silk_data_with_prefix = b'\x02' + silk_data[:-2]
         
-        # return BytesIO(silk_data_with_prefix)
-        with open(output_path, "wb") as f:
-            f.write(silk_data_with_prefix)
+    #     # return BytesIO(silk_data_with_prefix)
+    #     with open(output_path, "wb") as f:
+    #         f.write(silk_data_with_prefix)
             
-        return 0
+    #     return 0
+    with wave.open(wav_path, 'rb') as wav:
+        rate = wav.getframerate()
+        duration = pilk.encode(wav_path, output_path, pcm_rate=rate, tencent=True)
+        return duration