introduce PipelineParams audio input/output sample rates

This commit is contained in:
Aleix Conchillo Flaqué
2025-02-04 12:22:41 -08:00
parent cc54255c41
commit ab45e481be
61 changed files with 570 additions and 402 deletions

View File

@@ -17,6 +17,7 @@ from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.serializers.protobuf import ProtobufFrameSerializer
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.deepgram import DeepgramSTTService
from pipecat.services.openai import OpenAILLMService
@@ -80,7 +81,7 @@ class SessionTimeoutHandler:
async def main():
transport = WebsocketServerTransport(
params=WebsocketServerParams(
audio_out_sample_rate=16000,
serializer=ProtobufFrameSerializer(),
audio_out_enabled=True,
add_wav_header=True,
vad_enabled=True,
@@ -97,7 +98,6 @@ async def main():
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
sample_rate=16000,
)
messages = [
@@ -122,7 +122,12 @@ async def main():
]
)
task = PipelineTask(pipeline, params=PipelineParams(allow_interruptions=True))
task = PipelineTask(
pipeline,
params=PipelineParams(
audio_in_sample_rate=16000, audio_out_sample_rate=16000, allow_interruptions=True
),
)
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):