Add gain slider in webapp mode

src/auracast/server/multicast_frontend.py

@@ -5,6 +5,10 @@ import requests
 from auracast import auracast_config
 import logging as log
 
+# Track whether WebRTC stream is active across Streamlit reruns
+if 'stream_started' not in st.session_state:
+    st.session_state['stream_started'] = False
+
 # Global: desired packetization time in ms for Opus (should match backend)
 PTIME = 40
 BACKEND_URL = "http://localhost:5000"
@@ -47,6 +51,11 @@ if audio_mode in ["Webapp", "USB"]:
     default_input = saved_settings.get('input_device') or 'default'
     stream_name = st.text_input("Channel Name", value=default_name)
     language = st.text_input("Language (ISO 639-3)", value=default_lang)
+    # Gain slider for Webapp mode
+    if audio_mode == "Webapp":
+        mic_gain = st.slider("Microphone Gain", 0.0, 4.0, 1.0, 0.1, help="Adjust microphone volume sent to Auracast")
+    else:
+        mic_gain = 1.0
 
     # Input device selection for USB mode
     if audio_mode == "USB":
@@ -83,7 +92,17 @@ if audio_mode in ["Webapp", "USB"]:
     start_stream = st.button("Start Auracast")
     stop_stream = st.button("Stop Auracast")
 
+    # If gain slider moved while streaming, send update to JS without restarting
+    if audio_mode == "Webapp" and st.session_state.get('stream_started'):
+        update_js = f"""
+        <script>
+            if (window.gainNode) {{ window.gainNode.gain.value = {mic_gain}; }}
+        </script>
+        """
+        st.components.v1.html(update_js, height=0)
+
     if stop_stream:
+        st.session_state['stream_started'] = False
         try:
             r = requests.post(f"{BACKEND_URL}/stop_audio").json()
             if r['was_running']:
@@ -133,14 +152,28 @@ if audio_mode in ["Webapp", "USB"]:
         except Exception as e:
             st.error(f"Error: {e}")
 
-    if audio_mode == "Webapp" and start_stream:
+    # Render / maintain WebRTC component
+    if audio_mode == "Webapp" and (start_stream or st.session_state.get('stream_started')):
         st.markdown("Starting microphone; allow access if prompted and speak.")
         component = f"""
         <script>
             (async () => {{
+                if (window.webrtc_started) return; // Prevent re-init on rerun
+                window.webrtc_started = true;
+                const GAIN_VALUE = {mic_gain};
                 const pc = new RTCPeerConnection(); // No STUN needed for localhost
-                const stream = await navigator.mediaDevices.getUserMedia({{audio:true}});
-                stream.getTracks().forEach(t => pc.addTrack(t, stream));
+                const micStream = await navigator.mediaDevices.getUserMedia({{audio:true}});
+                // Create Web Audio gain processing
+                const audioCtx = new (window.AudioContext || window.webkitAudioContext)();
+                const source = audioCtx.createMediaStreamSource(micStream);
+                const gainNode = audioCtx.createGain();
+                gainNode.gain.value = GAIN_VALUE;
+                // Expose for later adjustments
+                window.gainNode = gainNode;
+                const dest = audioCtx.createMediaStreamDestination();
+                source.connect(gainNode).connect(dest);
+                // Add processed tracks to WebRTC
+                dest.stream.getTracks().forEach(t => pc.addTrack(t, dest.stream));
                 // --- WebRTC offer/answer exchange ---
                 const offer = await pc.createOffer();
                 // Patch SDP offer to include a=ptime using global PTIME
@@ -169,6 +202,7 @@ if audio_mode in ["Webapp", "USB"]:
         </script>
         """
         st.components.v1.html(component, height=0)
+        st.session_state['stream_started'] = True
 else:
     st.header("Advertised Streams (Cloud Announcements)")
     st.info("This feature requires backend support to list advertised streams.")