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fix: send sample rate to STT services

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This commit is contained in:
Abhishek Kumar 2026-02-09 16:25:09 +05:30
parent 4c936ae57d
commit 7a102026fb
6 changed files with 16 additions and 11 deletions
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2
api/services/looptalk/core/pipeline_builder.py
View file

@ -91,7 +91,7 @@ class LoopTalkPipelineBuilder:
logger.info(f"Using {len(keyterms)} keyterms for STT: {keyterms}")
# Create services
stt = create_stt_service(user_config, keyterms=keyterms)
stt = create_stt_service(user_config, audio_config, keyterms=keyterms)
llm = create_llm_service(user_config)
tts = create_tts_service(user_config, audio_config)

8
api/services/pipecat/audio_config.py
View file

@ -104,7 +104,7 @@ def create_audio_config(transport_type: str) -> AudioConfig:
transport_out_sample_rate=8000,
vad_sample_rate=8000, # Use matching VAD rate
pipeline_sample_rate=8000, # Keep at 8kHz to avoid resampling
buffer_size_seconds=1.0,
buffer_size_seconds=5.0,
)
elif transport_type == WorkflowRunMode.VONAGE.value:
# Vonage uses 16kHz Linear PCM
@ -113,7 +113,7 @@ def create_audio_config(transport_type: str) -> AudioConfig:
transport_out_sample_rate=16000,
vad_sample_rate=16000, # Use matching VAD rate
pipeline_sample_rate=16000, # Keep at 16kHz to avoid resampling
buffer_size_seconds=1.0,
buffer_size_seconds=5.0,
)
elif transport_type in [
WorkflowRunMode.WEBRTC.value,
@ -126,7 +126,7 @@ def create_audio_config(transport_type: str) -> AudioConfig:
transport_out_sample_rate=16000, # Transport will resample to 24kHz
vad_sample_rate=16000, # VAD native rate
pipeline_sample_rate=16000, # Keep pipeline at 16kHz
buffer_size_seconds=1.0,
buffer_size_seconds=5.0,
)
else:
# Default configuration
@ -138,5 +138,5 @@ def create_audio_config(transport_type: str) -> AudioConfig:
transport_out_sample_rate=16000,
vad_sample_rate=16000,
pipeline_sample_rate=16000,
buffer_size_seconds=1.0,
buffer_size_seconds=5.0,
)

2
api/services/pipecat/run_pipeline.py
View file

@ -494,7 +494,7 @@ async def _run_pipeline(
]
# Create services based on user configuration
stt = create_stt_service(user_config, keyterms=keyterms)
stt = create_stt_service(user_config, audio_config, keyterms=keyterms)
tts = create_tts_service(user_config, audio_config)
llm = create_llm_service(user_config)

11
api/services/pipecat/service_factory.py
View file

@ -30,7 +30,7 @@ if TYPE_CHECKING:
from api.services.pipecat.audio_config import AudioConfig
def create_stt_service(user_config, keyterms: list[str] | None = None):
def create_stt_service(user_config, audio_config: "AudioConfig", keyterms: list[str] | None = None):
"""Create and return appropriate STT service based on user configuration
Args:
@ -53,6 +53,7 @@ def create_stt_service(user_config, keyterms: list[str] | None = None):
keyterm=keyterms or [],
),
should_interrupt=False, # Let UserAggregator take care of sending InterruptionFrame
sample_rate=audio_config.transport_in_sample_rate
)
# Other models than flux
@ -63,20 +64,21 @@ def create_stt_service(user_config, keyterms: list[str] | None = None):
profanity_filter=False,
endpointing=100,
model=user_config.stt.model,
keyterm=keyterms or [],
keyterm=keyterms or []
)
logger.debug(f"Using DeepGram Model - {user_config.stt.model}")
return DeepgramSTTService(
live_options=live_options,
api_key=user_config.stt.api_key,
should_interrupt=False, # Let UserAggregator take care of sending InterruptionFrame
sample_rate=audio_config.transport_in_sample_rate
)
elif user_config.stt.provider == ServiceProviders.OPENAI.value:
return OpenAISTTService(
api_key=user_config.stt.api_key, model=user_config.stt.model
)
elif user_config.stt.provider == ServiceProviders.CARTESIA.value:
return CartesiaSTTService(api_key=user_config.stt.api_key)
return CartesiaSTTService(api_key=user_config.stt.api_key, sample_rate=audio_config.transport_in_sample_rate)
elif user_config.stt.provider == ServiceProviders.DOGRAH.value:
base_url = MPS_API_URL.replace("http://", "ws://").replace("https://", "wss://")
language = getattr(user_config.stt, "language", None) or "multi"
@ -86,6 +88,7 @@ def create_stt_service(user_config, keyterms: list[str] | None = None):
model=user_config.stt.model,
language=language,
keyterms=keyterms,
sample_rate=audio_config.transport_in_sample_rate
)
elif user_config.stt.provider == ServiceProviders.SARVAM.value:
# Map Sarvam language code to pipecat Language enum
@ -109,6 +112,7 @@ def create_stt_service(user_config, keyterms: list[str] | None = None):
api_key=user_config.stt.api_key,
model=user_config.stt.model,
params=SarvamSTTService.InputParams(language=pipecat_language),
sample_rate=audio_config.transport_in_sample_rate
)
elif user_config.stt.provider == ServiceProviders.SPEECHMATICS.value:
from pipecat.services.speechmatics.stt import (
@ -134,6 +138,7 @@ def create_stt_service(user_config, keyterms: list[str] | None = None):
operating_point=operating_point,
additional_vocab=additional_vocab,
),
sample_rate=audio_config.transport_in_sample_rate
)
else:
raise HTTPException(

2
api/services/pipecat/transport_setup.py
View file

@ -368,7 +368,7 @@ def create_stasis_transport(
audio_out_enabled=True,
audio_out_sample_rate=audio_config.transport_out_sample_rate,
audio_in_sample_rate=audio_config.transport_in_sample_rate,
audio_out_10ms_chunks=2, # Send 20ms packets
# audio_out_10ms_chunks=2, # ToDo: Check if we cant support 40 ms packets?
audio_out_mixer=(
SoundfileMixer(
sound_files={

2
pipecat

@ -1 +1 @@
Subproject commit 5313e8cd94443f220cc56c10cc2fc2aa98d8b6ba
Subproject commit d67983b3b165f945a93e5ce594f47781a96bff9b