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Adding support to WebsocketTransport #1911
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}, | ||
"dependencies": { | ||
"@pipecat-ai/client-js": "^0.3.5", | ||
"@pipecat-ai/realtime-websocket-transport": "^0.3.5", |
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I still need to publish a new version which will include the PipecatWebsocketTransport
.
Right now I am using the version from this PR:
Codecov ReportAttention: Patch coverage is
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In terms of organization for the examples, we already have a websocket-server example. Should this new example replace the current one? |
@markbackman, I have removed the
What do you think ? |
This makes sense! |
pipeline, | ||
params=PipelineParams( | ||
allow_interruptions=True, | ||
observers=[RTVIObserver(rtvi)], |
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observers
should be a PipelineTask param. It's deprecated as a PipelineParams param.
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done.
), | ||
) | ||
|
||
@rtvi.event_handler("on_client_ready") |
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We do this for simple-chatbot:
@rtvi.event_handler("on_client_ready")
async def on_client_ready(rtvi):
await rtvi.set_bot_ready()
# Kick off the conversation
await task.queue_frames([context_aggregator.user().get_context_frame()])
Does this make sense here too?
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done.
@@ -0,0 +1,110 @@ | |||
# |
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Same comments as the bot_fast_api.py example.
- observer
- on_client_ready
- GeminiMultimodalLiveLLMService args
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done.
llm = GeminiMultimodalLiveLLMService( | ||
api_key=os.getenv("GOOGLE_API_KEY"), | ||
voice_id="Puck", # Aoede, Charon, Fenrir, Kore, Puck | ||
transcribe_user_audio=True, |
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This now happens automatically.
transcribe_user_audio=True, |
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done.
api_key=os.getenv("GOOGLE_API_KEY"), | ||
voice_id="Puck", # Aoede, Charon, Fenrir, Kore, Puck | ||
transcribe_user_audio=True, | ||
transcribe_model_audio=True, |
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Same here.
transcribe_model_audio=True, |
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done.
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Looks really good! Just a few comments, but overall this is really nice to have.
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…o start the bot inside on_client_ready.
Thank you for the review @markbackman 🎉 |
ProtobufSerializer
.WebsocketTransport