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Update src/streamlit_app.py
Browse files- src/streamlit_app.py +76 -27
src/streamlit_app.py
CHANGED
@@ -6,7 +6,7 @@ import base64
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import io
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import threading
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import traceback
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-
import atexit
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import time
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import logging
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from dotenv import load_dotenv
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@@ -24,21 +24,20 @@ from streamlit_webrtc import (
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WebRtcMode,
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AudioProcessorBase,
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VideoProcessorBase,
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-
# ClientSettings # Removed as it's not used in this version
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)
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# from aiortc import RTCIceServer, RTCConfiguration # RTCConfiguration object not needed directly
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# --- Configuration ---
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load_dotenv()
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logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(threadName)s - %(levelname)s - %(message)s')
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# Audio configuration
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PYAUDIO_FORMAT = pyaudio.paInt16
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PYAUDIO_CHANNELS = 1
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WEBRTC_REQUESTED_AUDIO_CHANNELS = 1
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WEBRTC_REQUESTED_SEND_SAMPLE_RATE = 16000
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GEMINI_AUDIO_RECEIVE_SAMPLE_RATE = 24000
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PYAUDIO_PLAYBACK_CHUNK_SIZE = 1024
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AUDIO_PLAYBACK_QUEUE_MAXSIZE = 50
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MEDIA_TO_GEMINI_QUEUE_MAXSIZE = 30
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@@ -72,10 +71,10 @@ def cleanup_pyaudio():
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pya.terminate()
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atexit.register(cleanup_pyaudio)
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# --- Global Queues
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video_frames_to_gemini_q
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audio_chunks_to_gemini_q
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audio_from_gemini_playback_q
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# --- Gemini Client Setup ---
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GEMINI_API_KEY = os.environ.get("GEMINI_API_KEY")
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@@ -106,6 +105,8 @@ class GeminiInteractionLoop:
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self.async_event_loop = None
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self.is_running = True
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self.playback_stream = None
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async def send_text_input_to_gemini(self, user_text):
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if not user_text or not self.gemini_session or not self.is_running:
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@@ -120,6 +121,10 @@ class GeminiInteractionLoop:
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async def stream_media_to_gemini(self):
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logging.info("Task started: Stream media from WebRTC queues to Gemini.")
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async def get_media_from_queues():
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try:
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video_frame = await asyncio.wait_for(video_frames_to_gemini_q.get(), timeout=0.02)
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video_frames_to_gemini_q.task_done()
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@@ -154,6 +159,8 @@ class GeminiInteractionLoop:
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while self.is_running:
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if not self.gemini_session:
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await asyncio.sleep(0.1); continue
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try:
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turn_response = self.gemini_session.receive()
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async for chunk in turn_response:
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@@ -176,6 +183,11 @@ class GeminiInteractionLoop:
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async def play_gemini_audio(self):
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logging.info("Task started: Play Gemini audio responses.")
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try:
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self.playback_stream = await asyncio.to_thread(
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pya.open, format=PYAUDIO_FORMAT, channels=PYAUDIO_CHANNELS, rate=GEMINI_AUDIO_RECEIVE_SAMPLE_RATE, output=True, frames_per_buffer=PYAUDIO_PLAYBACK_CHUNK_SIZE
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)
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@@ -202,15 +214,26 @@ class GeminiInteractionLoop:
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def signal_stop(self):
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logging.info("Signal to stop GeminiInteractionLoop received.")
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self.is_running = False
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for q in [video_frames_to_gemini_q, audio_chunks_to_gemini_q, audio_from_gemini_playback_q]:
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-
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-
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-
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async def run_main_loop(self):
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self.async_event_loop = asyncio.get_running_loop()
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self.is_running = True
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logging.info("GeminiInteractionLoop run_main_loop starting...")
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if client is None:
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logging.critical("Gemini client is None in run_main_loop. Aborting.")
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return
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@@ -245,7 +268,11 @@ class GeminiInteractionLoop:
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logging.info("GeminiInteractionLoop.run_main_loop() finishing...")
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self.is_running = False
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self.gemini_session = None
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-
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# --- WebRTC Media Processors ---
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class VideoProcessor(VideoProcessorBase):
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@@ -254,6 +281,10 @@ class VideoProcessor(VideoProcessorBase):
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self.last_gemini_send_time = time.monotonic()
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async def _process_and_queue_frame_async(self, frame_ndarray):
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self.frame_counter += 1
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current_time = time.monotonic()
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if (current_time - self.last_gemini_send_time) < (1.0 / VIDEO_FPS_TO_GEMINI):
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@@ -277,11 +308,22 @@ class VideoProcessor(VideoProcessorBase):
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async def recv(self, frame):
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img_bgr = frame.to_ndarray(format="bgr24")
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-
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return frame
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class AudioProcessor(AudioProcessorBase):
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async def _process_and_queue_audio_async(self, audio_frames):
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for frame in audio_frames:
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audio_data = frame.planes[0].to_bytes()
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mime_type = f"audio/L16;rate={frame.sample_rate};channels={frame.layout.channels}"
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@@ -295,7 +337,12 @@ class AudioProcessor(AudioProcessorBase):
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except Exception as e: logging.error(f"Error queueing audio chunk: {e}", exc_info=True)
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async def recv(self, frames):
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-
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return frames
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# --- Streamlit UI and Application Logic ---
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@@ -324,10 +371,7 @@ def run_streamlit_app():
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st.session_state.gemini_session_active = True
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st.session_state.chat_messages = [{"role": "system", "content": "Assistant activating. Please allow camera/microphone access in your browser if prompted."}]
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-
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while not q.empty():
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try: q.get_nowait()
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except asyncio.QueueEmpty: break
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gemini_loop = GeminiInteractionLoop()
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st.session_state.gemini_loop_instance = gemini_loop
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@@ -358,15 +402,20 @@ def run_streamlit_app():
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}
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}
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webrtc_ctx = webrtc_streamer(
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key=st.session_state.webrtc_component_key,
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mode=WebRtcMode.SENDONLY,
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rtc_configuration={
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"iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]
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},
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media_stream_constraints=MEDIA_STREAM_CONSTRAINTS,
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video_processor_factory=VideoProcessor,
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audio_processor_factory=AudioProcessor,
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async_processing=True,
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)
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@@ -374,7 +423,7 @@ def run_streamlit_app():
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st.caption("WebRTC connected. Streaming your camera and microphone.")
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elif st.session_state.gemini_session_active:
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st.caption("WebRTC attempting to connect. Ensure camera/microphone permissions are granted in your browser.")
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if hasattr(webrtc_ctx.state, 'error') and webrtc_ctx.state.error:
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st.error(f"WebRTC Connection Error: {webrtc_ctx.state.error}")
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else:
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st.info("Click 'Start Session' in the sidebar to enable the live feed and assistant.")
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import io
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import threading
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import traceback
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import atexit
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import time
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import logging
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from dotenv import load_dotenv
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WebRtcMode,
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AudioProcessorBase,
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VideoProcessorBase,
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)
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# from aiortc import RTCIceServer, RTCConfiguration # RTCConfiguration object not needed directly
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# --- Configuration ---
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load_dotenv()
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logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(threadName)s - %(levelname)s - %(message)s')
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# Audio configuration
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PYAUDIO_FORMAT = pyaudio.paInt16
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PYAUDIO_CHANNELS = 1
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WEBRTC_REQUESTED_AUDIO_CHANNELS = 1
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WEBRTC_REQUESTED_SEND_SAMPLE_RATE = 16000
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GEMINI_AUDIO_RECEIVE_SAMPLE_RATE = 24000
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PYAUDIO_PLAYBACK_CHUNK_SIZE = 1024
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AUDIO_PLAYBACK_QUEUE_MAXSIZE = 50
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MEDIA_TO_GEMINI_QUEUE_MAXSIZE = 30
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pya.terminate()
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atexit.register(cleanup_pyaudio)
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# --- Global Queues - Declare as None, initialize later ---
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video_frames_to_gemini_q: asyncio.Queue = None
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audio_chunks_to_gemini_q: asyncio.Queue = None
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audio_from_gemini_playback_q: asyncio.Queue = None
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# --- Gemini Client Setup ---
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GEMINI_API_KEY = os.environ.get("GEMINI_API_KEY")
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self.async_event_loop = None
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self.is_running = True
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self.playback_stream = None
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# Queues will be initialized in run_main_loop and assigned to global vars
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# This class will use the global queue variables directly
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async def send_text_input_to_gemini(self, user_text):
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if not user_text or not self.gemini_session or not self.is_running:
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async def stream_media_to_gemini(self):
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logging.info("Task started: Stream media from WebRTC queues to Gemini.")
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async def get_media_from_queues():
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# Ensure queues are initialized before trying to get from them
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if video_frames_to_gemini_q is None or audio_chunks_to_gemini_q is None:
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await asyncio.sleep(0.1) # Wait for queues to be initialized
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return None
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try:
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video_frame = await asyncio.wait_for(video_frames_to_gemini_q.get(), timeout=0.02)
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video_frames_to_gemini_q.task_done()
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while self.is_running:
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if not self.gemini_session:
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await asyncio.sleep(0.1); continue
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if audio_from_gemini_playback_q is None: # Wait for queue init
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await asyncio.sleep(0.1); continue
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try:
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turn_response = self.gemini_session.receive()
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async for chunk in turn_response:
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async def play_gemini_audio(self):
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logging.info("Task started: Play Gemini audio responses.")
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try:
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# Wait for the playback queue to be initialized
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while audio_from_gemini_playback_q is None and self.is_running:
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await asyncio.sleep(0.1)
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if not self.is_running: return
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self.playback_stream = await asyncio.to_thread(
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pya.open, format=PYAUDIO_FORMAT, channels=PYAUDIO_CHANNELS, rate=GEMINI_AUDIO_RECEIVE_SAMPLE_RATE, output=True, frames_per_buffer=PYAUDIO_PLAYBACK_CHUNK_SIZE
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)
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def signal_stop(self):
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logging.info("Signal to stop GeminiInteractionLoop received.")
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self.is_running = False
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# Use global queue variables directly
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for q in [video_frames_to_gemini_q, audio_chunks_to_gemini_q, audio_from_gemini_playback_q]:
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if q: # Check if queue is initialized
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try: q.put_nowait(None)
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except asyncio.QueueFull: logging.warning(f"Queue was full when trying to put sentinel for stop signal.")
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except Exception as e: logging.error(f"Error putting sentinel in queue: {e}", exc_info=True)
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async def run_main_loop(self):
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global video_frames_to_gemini_q, audio_chunks_to_gemini_q, audio_from_gemini_playback_q # Allow modification of global vars
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self.async_event_loop = asyncio.get_running_loop()
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self.is_running = True
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logging.info("GeminiInteractionLoop run_main_loop starting...")
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# Initialize queues here, within the asyncio loop of this thread
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video_frames_to_gemini_q = asyncio.Queue(maxsize=MEDIA_TO_GEMINI_QUEUE_MAXSIZE)
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audio_chunks_to_gemini_q = asyncio.Queue(maxsize=MEDIA_TO_GEMINI_QUEUE_MAXSIZE)
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audio_from_gemini_playback_q = asyncio.Queue(maxsize=AUDIO_PLAYBACK_QUEUE_MAXSIZE)
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logging.info("Asyncio queues initialized in GeminiInteractionLoop.")
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if client is None:
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logging.critical("Gemini client is None in run_main_loop. Aborting.")
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return
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logging.info("GeminiInteractionLoop.run_main_loop() finishing...")
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self.is_running = False
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self.gemini_session = None
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# Clear queues on exit to prevent issues if loop restarts
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video_frames_to_gemini_q = None
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audio_chunks_to_gemini_q = None
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audio_from_gemini_playback_q = None
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logging.info("GeminiInteractionLoop finished and queues cleared.")
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# --- WebRTC Media Processors ---
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class VideoProcessor(VideoProcessorBase):
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self.last_gemini_send_time = time.monotonic()
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async def _process_and_queue_frame_async(self, frame_ndarray):
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if video_frames_to_gemini_q is None: # Wait for queue to be initialized
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logging.debug("VideoProcessor: video_frames_to_gemini_q is None, waiting...")
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return
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self.frame_counter += 1
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current_time = time.monotonic()
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if (current_time - self.last_gemini_send_time) < (1.0 / VIDEO_FPS_TO_GEMINI):
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async def recv(self, frame):
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img_bgr = frame.to_ndarray(format="bgr24")
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# Ensure an event loop is running in the current thread for create_task
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try:
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loop = asyncio.get_running_loop()
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loop.create_task(self._process_and_queue_frame_async(img_bgr))
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except RuntimeError: # No running loop in this thread (should not happen with streamlit-webrtc async_processing=True)
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logging.error("VideoProcessor.recv: No running asyncio loop in current thread for create_task.")
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# Fallback or log error, direct call might block WebRTC thread
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# await self._process_and_queue_frame_async(img_bgr) # Potentially blocking
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return frame
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class AudioProcessor(AudioProcessorBase):
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async def _process_and_queue_audio_async(self, audio_frames):
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if audio_chunks_to_gemini_q is None: # Wait for queue to be initialized
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logging.debug("AudioProcessor: audio_chunks_to_gemini_q is None, waiting...")
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return
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for frame in audio_frames:
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audio_data = frame.planes[0].to_bytes()
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mime_type = f"audio/L16;rate={frame.sample_rate};channels={frame.layout.channels}"
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except Exception as e: logging.error(f"Error queueing audio chunk: {e}", exc_info=True)
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async def recv(self, frames):
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try:
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loop = asyncio.get_running_loop()
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loop.create_task(self._process_and_queue_audio_async(frames))
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except RuntimeError:
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logging.error("AudioProcessor.recv: No running asyncio loop in current thread for create_task.")
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# await self._process_and_queue_audio_async(frames) # Potentially blocking
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return frames
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# --- Streamlit UI and Application Logic ---
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st.session_state.gemini_session_active = True
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st.session_state.chat_messages = [{"role": "system", "content": "Assistant activating. Please allow camera/microphone access in your browser if prompted."}]
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# Queues will be initialized inside GeminiInteractionLoop's thread
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gemini_loop = GeminiInteractionLoop()
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st.session_state.gemini_loop_instance = gemini_loop
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}
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}
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# Only render WebRTC streamer if queues are expected to be initialized by the Gemini loop
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# This is a bit of a race condition check, might need refinement
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# A better way would be for Gemini loop to signal when queues are ready.
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# For now, we assume if session is active, loop is trying to start and init queues.
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webrtc_ctx = webrtc_streamer(
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key=st.session_state.webrtc_component_key,
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mode=WebRtcMode.SENDONLY,
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rtc_configuration={
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"iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]
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},
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media_stream_constraints=MEDIA_STREAM_CONSTRAINTS,
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video_processor_factory=VideoProcessor, # Pass the class
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audio_processor_factory=AudioProcessor, # Pass the class
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async_processing=True,
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)
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st.caption("WebRTC connected. Streaming your camera and microphone.")
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elif st.session_state.gemini_session_active:
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st.caption("WebRTC attempting to connect. Ensure camera/microphone permissions are granted in your browser.")
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if hasattr(webrtc_ctx.state, 'error') and webrtc_ctx.state.error:
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st.error(f"WebRTC Connection Error: {webrtc_ctx.state.error}")
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else:
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st.info("Click 'Start Session' in the sidebar to enable the live feed and assistant.")
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