Update app.py

app.py

@@ -22,8 +22,9 @@ from utils import (
 from prompts import SYSTEM_PROMPT
 
 # The new Q&A with mic
-from qa import AudioBufferProcessor, handle_qa_exchange
-from streamlit_webrtc import webrtc_streamer, WebRtcMode
+from qa import AudioBufferProcessor, handle_qa_exchange, transcribe_audio_deepgram
+
+from streamlit_webrtc import webrtc_streamer, WebRtcMode, RTCConfiguration
 
 MAX_QA_QUESTIONS = 5
 
@@ -145,6 +146,7 @@ def generate_podcast(
         except Exception as e:
             return None, f"Error researching topic: {str(e)}"
 
+    from utils import truncate_text
     text = truncate_text(text)
 
     extra_instructions = []
@@ -332,7 +334,6 @@ def main():
         progress_bar.progress(75)
         time.sleep(1.0)
 
-        from utils import truncate_text
         audio_bytes, transcript = generate_podcast(
             file,
             url,
@@ -458,15 +459,21 @@ def main():
             st.write(f"You can ask up to {remaining} more question(s).")
 
             st.write("### Record Your Follow-Up Question:")
-            from streamlit_webrtc import webrtc_streamer, WebRtcMode, RTCConfiguration
-            from qa import AudioBufferProcessor
 
-            RTC_CONFIGURATION = {"iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]}
+            # EXPLICIT STUN SERVER
+            # So we can confirm ICE candidates are gathered
+            new_rtc_config = RTCConfiguration(
+                {
+                    "iceServers": [
+                        {"urls": ["stun:stun.l.google.com:19302"]}
+                    ]
+                }
+            )
 
             webrtc_ctx = webrtc_streamer(
                 key="qna-audio-stream",
                 mode=WebRtcMode.SENDONLY,
-                rtc_configuration=RTC_CONFIGURATION,
+                rtc_configuration=new_rtc_config,  # <--- STUN server explicitly set
                 media_stream_constraints={"audio": True, "video": False},
                 audio_processor_factory=AudioBufferProcessor
             )
@@ -478,11 +485,12 @@ def main():
             if webrtc_ctx.state.playing and webrtc_ctx.audio_processor:
                 st.session_state["audio-processor"] = webrtc_ctx.audio_processor
 
-            # Instead of checking .state.status, we check .state.playing
+                # Debug print: how many frames have arrived?
+                st.write("Frames so far:", webrtc_ctx.audio_processor.frame_count)
+
             if not webrtc_ctx.state.playing:
                 st.write("Recording Stopped. You may now submit your question.")
 
-            # Submit button
             if st.button("Submit Q&A"):
                 if used_questions >= MAX_QA_QUESTIONS:
                     st.warning("You have reached the Q&A limit.")
@@ -495,7 +503,6 @@ def main():
                         if not local_wav_path:
                             st.warning("No audio frames found. Please record again.")
                         else:
-                            from qa import transcribe_audio_deepgram
                             st.write("Transcribing your voice question via Deepgram...")
                             question_text = transcribe_audio_deepgram(local_wav_path)
                             if not question_text.strip():