Merge branch 'develop' into jms_20250106_prod

src/core/callcenter/test.py

@@ -1,3 +1,6 @@
-import json
-from src.core.callcenter.api import CallInfo, DeviceInfo, NextCommand
-
+#
+# import jieba
+#
+# TestStr = "能帮我查一下，我家水费欠多少"
+# seg_list = jieba.cut(TestStr, cut_all=False, HMM=True)
+# print ("Default Mode:", "/ ".join(seg_list))

src/core/voip/asr.py

@@ -114,6 +114,7 @@ class TestSt:
             )
             self.sr.start(
                 aformat="pcm",
+                sample_rate=8000,
                 enable_intermediate_result=True,
                 enable_punctuation_prediction=True,
                 enable_inverse_text_normalization=True

src/core/voip/bot.py

@@ -599,6 +599,7 @@ class BotAgent:
             media_cfg.jbMinPre = 4  # Minimum pre-fetch frames
             media_cfg.jbMaxPre = 16  # Maximum pre-fetch frames
             media_cfg.noVad = True  # Disable Voice Activity Detection if needed
+            media_cfg.clockRate = 8000
             self.ep.medConfig = media_cfg  # Apply media config to endpoint
 
             # Create SIP transport. Error handling sample is shown

src/core/voip/constant.py

@@ -19,10 +19,12 @@ def build_audio_format():
     fmt = pj.MediaFormatAudio()
     fmt.type = pj.PJMEDIA_TYPE_AUDIO
     fmt.id = pj.PJMEDIA_FORMAT_PCM
-    fmt.clockRate = 16000  # 采样率
     fmt.channelCount = 1  # 通道数
-    fmt.frameTimeUsec = 20000  # 每帧的时间（20 毫秒）
     fmt.bitsPerSample = 16  # 每个采样的位数
+    fmt.clockRate = 8000  # 采样率
+    fmt.frameTimeUsec = 12500  # 每帧的时间（20 毫秒）
+    # fmt.clockRate = 16000  # 采样率
+    # fmt.frameTimeUsec = 20000  # 每帧的时间（20 毫秒）
     return fmt