Update app.py

app.py

@@ -7,7 +7,7 @@ from gtts import gTTS
 import google.generativeai as genai
 from io import BytesIO
 import PyPDF2
-from streamlit_webrtc import webrtc_streamer, WebRtcMode, RTCConfiguration
+from audio_recorder_streamlit import audio_recorder
 
 # Set your API key
 api_key = "AIzaSyAHD0FwX-Ds6Y3eI-i5Oz7IdbJqR6rN7pg"  # Replace with your actual API key
@@ -108,7 +108,7 @@ def send_message(audio_data=None):
                 {"role": "user", "parts": [get_file_base64(file_content, uploaded_file.type)]}
             )
 
-    # Handle audio data from WebRTC
+    # Handle audio data
     if audio_data:
         prompt_parts.append(get_file_base64(audio_data, 'audio/wav'))
         st.session_state['chat_history'].append(
@@ -145,6 +145,7 @@ def send_message(audio_data=None):
     # Update the chat history display
     display_chat_history()
 
+
 # --- User Input Area ---
 col1, col2 = st.columns([3, 1])
 with col1:
@@ -168,25 +169,12 @@ uploaded_files = st.file_uploader(
     key=st.session_state.file_uploader_key
 )
 
-# --- WebRTC Audio Recording ---
-RTC_CONFIGURATION = RTCConfiguration({"iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]})
-
-webrtc_ctx = webrtc_streamer(
-    key="audio-recorder",
-    mode=WebRtcMode.SENDONLY,
-    rtc_configuration=RTC_CONFIGURATION,
-    audio_receiver_size=256,
-    media_stream_constraints={"video": False, "audio": True},
-)
-
-if webrtc_ctx.audio_receiver:
-    st.write("Recording audio...")
-    audio_frames = webrtc_ctx.audio_receiver.get_frames(timeout=None)
-    audio_data = b"".join([frame for frame in audio_frames])
-
-    # Send the recorded audio when the "Send" button is clicked
+# --- Audio Recorder ---
+audio_bytes = audio_recorder()
+if audio_bytes:
+    st.audio(audio_bytes, format="audio/wav")
     if st.button("Send Recording"):
-        send_message(audio_data=audio_data)
+        send_message(audio_data=audio_bytes)
 
 # --- Other Buttons ---
 st.button("Clear Conversation", on_click=clear_conversation)