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t
Thanks for reaching out, Andrew. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Here are the answers and troubleshooting steps for your questions: 1. Could RoomOutputOptions(transcription_enabled=True) prevent audio from being published? No, enabling transcription with 
transcription_enabled=True
does not affect audio publishing. The transcription and audio settings are independent. You can enable both transcription and audio output simultaneously, or configure them separately. 2. What would cause the agent to process TTS but never transition to "speaking" state? The most common causes are: • Audio output disabled: If 
audio_enabled=False
in 
RoomOutputOptions
, TTS processing will occur, but no audio will be published, and the "speaking" state will not be triggered. • Audio track subscription issues: The agent's audio track must be subscribed to before audio can flow. If no subscription occurs, audio won't be published. • Speaking state transition logic: The transition to "speaking" state depends on the first audio frame being processed. If audio output is disabled, this state change won't occur. 3. Is there a specific order of operations (connect, start session, generate_reply) that must be followed? The order of operations is important. Ensure the following sequence: 1. Connect to the room. 2. Start the agent session with the correct 
RoomOutputOptions
. 3. Generate replies and process TTS. 4. Do we need to explicitly enable audio publishing somewhere? Yes, ensure that 
audio_enabled=True
(or not explicitly set to 
False
) in 
RoomOutputOptions
. This setting controls whether audio is published to the room. Troubleshooting Steps: 1. Verify that 
audio_enabled=True
in 
RoomOutputOptions
. 2. Check that clients are subscribing to the agent's audio track. 3. Monitor the initialization of the audio output to ensure the track is published successfully. 4. Ensure the TTS audio frames are being forwarded to the audio output without interruption. Let me know if you need further assistance with these steps. Sources: examples/other/transcription/translator.py
b
In our voice agent implementation, we're trying to ensure proper audio flow. Currently we have: # First connect to check if another agent is already present await ctx.connect() # Check for other agents... # Start session await session.start( agent=briseis_agent, room=ctx.room, room_input_options=RoomInputOptions( noise_cancellation=noise_cancellation.BVC(), ), room_output_options=RoomOutputOptions( audio_enabled=True, transcription_enabled=True ), ) # Wait for participants if needed... # Generate greeting await session.generate_reply(instructions=greeting_instruction) Questions: 1. Is it correct to call ctx.connect() BEFORE session.start()? Or should the connection happen after? 2. For the initial greeting, should we wait for at least one participant to be in the room before calling generate_reply()? 3. If no participants are connected when generate_reply() is called, will the audio still be published when someone joins later? 4. Is there a best practice for timing the initial agent greeting to ensure participants hear it? We want to make sure the greeting is heard by users when they join the room.
t
I don't have the answer you're looking for. You could also try asking your question: • in one of the other Slack channels or • to https://deepwiki.com/livekit/livekit_composite which is trained on all LiveKit source code If you find the answer, please post it here to help others!
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