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Update app.py
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app.py
CHANGED
@@ -17,13 +17,23 @@ import io
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import soundfile as sf
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from gtts import gTTS
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import re
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# Load environment variables
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load_dotenv()
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# Initialize clients
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elevenlabs_client = ElevenLabs(api_key=os.getenv("ELEVENLABS_API_KEY"))
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stt_model = get_stt_model()
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class DeepSeekAPI:
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def __init__(self, api_key):
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@@ -45,7 +55,7 @@ class DeepSeekAPI:
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# Check for error response
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if response.status_code != 200:
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-
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return {"choices": [{"message": {"content": "I'm sorry, I encountered an error processing your request."}}]}
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return response.json()
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@@ -60,8 +70,9 @@ def response(
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messages = [{"role": d["role"], "content": d["content"]} for d in chatbot]
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# Convert speech to text
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text = stt_model.stt(audio)
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-
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# Add user message to chat
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chatbot.append({"role": "user", "content": text})
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@@ -71,8 +82,10 @@ def response(
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messages.append({"role": "user", "content": text})
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# Call DeepSeek API
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response_data = deepseek_client.chat_completion(messages)
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response_text = response_data["choices"][0]["message"]["content"]
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# Add AI response to chat
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chatbot.append({"role": "assistant", "content": response_text})
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@@ -80,7 +93,7 @@ def response(
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# Convert response to speech
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if os.getenv("ELEVENLABS_API_KEY"):
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try:
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-
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# Use the streaming API for better experience
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for chunk in elevenlabs_client.text_to_speech.convert_as_stream(
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@@ -93,12 +106,12 @@ def response(
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yield (24000, audio_array)
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except Exception as e:
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# Fall back to gTTS
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yield from use_gtts_for_text(response_text)
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else:
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# Fall back to gTTS
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yield from use_gtts_for_text(response_text)
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yield AdditionalOutputs(chatbot)
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@@ -114,7 +127,7 @@ def use_gtts_for_text(text):
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continue
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mp3_fp = io.BytesIO()
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-
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tts = gTTS(text=sentence, lang='en-us', tld='com', slow=False)
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tts.write_to_fp(mp3_fp)
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mp3_fp.seek(0)
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@@ -147,48 +160,103 @@ def use_gtts_for_text(text):
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chunk = chunk.reshape(1, -1)
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yield (24000, chunk)
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except Exception as e:
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yield None
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#
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rtc_configuration = {
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"iceServers": [
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{
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"urls": ["turn:openrelay.metered.ca:80"],
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"username": "openrelayproject",
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"credential": "openrelayproject"
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},
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{
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"urls": ["turn:openrelay.metered.ca:443?transport=tcp"],
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"username": "openrelayproject",
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"credential": "openrelayproject"
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}
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],
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"iceCandidatePoolSize": 10
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}
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# Create
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# Launch the app
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if __name__ == "__main__":
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# Local development
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os.environ["GRADIO_SSR_MODE"] = "false"
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demo.launch(server_port=7860)
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import soundfile as sf
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from gtts import gTTS
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import re
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import logging
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# Set up logging for WebRTC debugging
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logging.basicConfig(level=logging.DEBUG)
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logger = logging.getLogger("fastrtc-voice-assistant")
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# Load environment variables
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load_dotenv()
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# Enable WebRTC debug tracing
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os.environ["WEBRTC_TRACE"] = "WEBRTC_TRACE_ALL"
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# Initialize clients
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logger.info("Initializing clients...")
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elevenlabs_client = ElevenLabs(api_key=os.getenv("ELEVENLABS_API_KEY"))
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stt_model = get_stt_model()
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logger.info("Clients initialized")
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class DeepSeekAPI:
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def __init__(self, api_key):
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# Check for error response
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if response.status_code != 200:
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logger.error(f"DeepSeek API error: {response.status_code} - {response.text}")
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return {"choices": [{"message": {"content": "I'm sorry, I encountered an error processing your request."}}]}
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return response.json()
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messages = [{"role": d["role"], "content": d["content"]} for d in chatbot]
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# Convert speech to text
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logger.info("Converting speech to text...")
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text = stt_model.stt(audio)
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logger.info(f"User said: {text}")
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# Add user message to chat
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chatbot.append({"role": "user", "content": text})
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messages.append({"role": "user", "content": text})
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# Call DeepSeek API
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logger.info("Calling DeepSeek API...")
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response_data = deepseek_client.chat_completion(messages)
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response_text = response_data["choices"][0]["message"]["content"]
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logger.info(f"DeepSeek response: {response_text[:50]}...")
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# Add AI response to chat
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chatbot.append({"role": "assistant", "content": response_text})
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# Convert response to speech
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if os.getenv("ELEVENLABS_API_KEY"):
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try:
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logger.info("Using ElevenLabs for speech generation")
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# Use the streaming API for better experience
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for chunk in elevenlabs_client.text_to_speech.convert_as_stream(
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yield (24000, audio_array)
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except Exception as e:
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logger.error(f"ElevenLabs error: {e}, falling back to gTTS")
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# Fall back to gTTS
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yield from use_gtts_for_text(response_text)
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else:
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# Fall back to gTTS
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logger.info("ElevenLabs API key not found, using gTTS...")
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yield from use_gtts_for_text(response_text)
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yield AdditionalOutputs(chatbot)
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continue
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mp3_fp = io.BytesIO()
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logger.info(f"Using gTTS for: {sentence[:30]}...")
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tts = gTTS(text=sentence, lang='en-us', tld='com', slow=False)
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tts.write_to_fp(mp3_fp)
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mp3_fp.seek(0)
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chunk = chunk.reshape(1, -1)
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yield (24000, chunk)
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except Exception as e:
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logger.error(f"gTTS error: {e}")
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yield None
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# Comprehensive WebRTC configuration with multiple STUN/TURN options
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rtc_configuration = {
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"iceServers": [
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# Google STUN servers
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{"urls": ["stun:stun.l.google.com:19302"]},
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{"urls": ["stun:stun1.l.google.com:19302"]},
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{"urls": ["stun:stun2.l.google.com:19302"]},
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{"urls": ["stun:stun3.l.google.com:19302"]},
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{"urls": ["stun:stun4.l.google.com:19302"]},
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# OpenRelay TURN servers
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{
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"urls": ["turn:openrelay.metered.ca:80"],
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"username": "openrelayproject",
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"credential": "openrelayproject"
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},
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{
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"urls": ["turn:openrelay.metered.ca:443"],
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"username": "openrelayproject",
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"credential": "openrelayproject"
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},
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{
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"urls": ["turn:openrelay.metered.ca:443?transport=tcp"],
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"username": "openrelayproject",
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"credential": "openrelayproject"
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},
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# Additional public STUN servers
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{"urls": ["stun:stun.stunprotocol.org:3478"]},
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{"urls": ["stun:stun.voip.blackberry.com:3478"]},
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{"urls": ["stun:stun.nextcloud.com:443"]}
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],
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"iceCandidatePoolSize": 10,
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"bundlePolicy": "max-bundle",
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"rtcpMuxPolicy": "require",
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"iceTransportPolicy": "all" # Try "relay" if "all" doesn't work
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}
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# Create a simple wrapper for the webchat UI
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with gr.Blocks(title="LLM Voice Chat (Powered by DeepSeek & ElevenLabs)") as demo:
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gr.Markdown("# LLM Voice Chat\nPowered by DeepSeek & ElevenLabs")
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with gr.Row():
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with gr.Column(scale=3):
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# Create the chatbot component
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chatbot = gr.Chatbot(type="messages")
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# For debugging, allow seeing connection status
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connection_status = gr.Textbox(label="Connection Status",
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value="Ready to connect. Click the microphone button to start.",
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interactive=False)
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# Display debugging information
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debug_info = gr.Textbox(label="Debug Info",
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value="WebRTC debug information will appear here.",
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interactive=False)
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# Button to manually refresh the page
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refresh_btn = gr.Button("Refresh Connection")
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def refresh_page():
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debug_info.value = f"Attempting to refresh connection at {time.time()}"
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return "Refreshed", f"Connection refresh attempted at {time.time()}"
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refresh_btn.click(
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refresh_page,
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outputs=[connection_status, debug_info]
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)
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logger.info("Creating Stream component...")
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# Initialize the stream (outside of the blocks context)
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stream = Stream(
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modality="audio",
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mode="send-receive",
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handler=ReplyOnPause(response, input_sample_rate=16000),
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additional_outputs_handler=lambda a, b: b,
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additional_inputs=[chatbot],
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additional_outputs=[chatbot],
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rtc_configuration=rtc_configuration,
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concurrency_limit=5 if get_space() else None,
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time_limit=90 if get_space() else None
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)
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# Mount the stream to the blocks interface
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stream.render()
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logger.info("Stream component created and rendered")
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# Launch the app
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if __name__ == "__main__":
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# Local development
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logger.info("Running in development mode")
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os.environ["GRADIO_SSR_MODE"] = "false"
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demo.launch(server_port=7860, share=True)
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else:
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# Hugging Face Spaces
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logger.info("Running in Hugging Face Spaces")
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demo.launch()
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