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Merge pull request #4450 from pipecat-ai/mb/gpt-realtime-whisper

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Default OpenAI Realtime transcription to gpt-realtime-whisper
This commit is contained in:
Mark Backman
2026-05-13 09:48:33 -04:00
committed by GitHub
parent 9148e307cc 3e8c5c08f4
commit 5fef239b68
11 changed files with 141 additions and 60 deletions
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10
src/pipecat/services/openai/_constants.py Normal file
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@@ -0,0 +1,10 @@
#
# Copyright (c) 2024-2026, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
"""Internal constants for OpenAI service integrations."""
OPENAI_SAMPLE_RATE = 24000
OPENAI_REALTIME_WHISPER_MODEL = "gpt-realtime-whisper"

10
src/pipecat/services/openai/realtime/events.py
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@@ -13,6 +13,7 @@ from typing import Any, Literal
from pydantic import BaseModel, ConfigDict, Field
from pipecat.adapters.schemas.tools_schema import ToolsSchema
from pipecat.services.openai._constants import OPENAI_REALTIME_WHISPER_MODEL, OPENAI_SAMPLE_RATE
#
# session properties
@@ -34,7 +35,7 @@ class PCMAudioFormat(AudioFormat):
"""
type: Literal["audio/pcm"] = "audio/pcm"
rate: Literal[24000] = 24000
rate: Literal[24000] = OPENAI_SAMPLE_RATE
class PCMUAudioFormat(AudioFormat):
@@ -60,20 +61,21 @@ class PCMAAudioFormat(AudioFormat):
class InputAudioTranscription(BaseModel):
"""Configuration for audio transcription settings."""
model: str = "gpt-4o-transcribe"
model: str = OPENAI_REALTIME_WHISPER_MODEL
language: str | None
prompt: str | None
def __init__(
self,
model: str | None = "gpt-4o-transcribe",
model: str | None = OPENAI_REALTIME_WHISPER_MODEL,
language: str | None = None,
prompt: str | None = None,
):
"""Initialize InputAudioTranscription.
Args:
model: Transcription model to use (e.g., "gpt-4o-transcribe", "whisper-1").
model: Transcription model to use (e.g., "gpt-realtime-whisper",
"gpt-4o-transcribe", "whisper-1").
language: Optional language code for transcription.
prompt: Optional transcription hint text.
"""

36
src/pipecat/services/openai/realtime/llm.py
View File

@@ -52,6 +52,7 @@ from pipecat.processors.aggregators import async_tool_messages
from pipecat.processors.aggregators.llm_context import LLMContext, LLMSpecificMessage
from pipecat.processors.frame_processor import FrameDirection
from pipecat.services.llm_service import FunctionCallFromLLM, LLMService
from pipecat.services.openai._constants import OPENAI_REALTIME_WHISPER_MODEL, OPENAI_SAMPLE_RATE
from pipecat.services.settings import (
NOT_GIVEN,
LLMSettings,
@@ -290,6 +291,8 @@ class OpenAIRealtimeLLMService(LLMService[OpenAIRealtimeLLMAdapter]):
if settings is not None:
default_settings.apply_update(settings)
self._omit_unsupported_input_audio_transcription_prompt(default_settings.session_properties)
# Build WebSocket URL with model query parameter
# Source: https://platform.openai.com/docs/guides/realtime-websocket
full_url = f"{base_url}?model={default_settings.model}"
@@ -330,6 +333,29 @@ class OpenAIRealtimeLLMService(LLMService[OpenAIRealtimeLLMAdapter]):
self._register_event_handler("on_conversation_item_updated")
self._retrieve_conversation_item_futures = {}
@staticmethod
def _omit_unsupported_input_audio_transcription_prompt(
session_properties: events.SessionProperties,
) -> bool:
"""Drop input transcription prompt settings unsupported by the selected model."""
transcription = (
session_properties.audio.input.transcription
if session_properties.audio
and session_properties.audio.input
and session_properties.audio.input.transcription
else None
)
if transcription and transcription.model == OPENAI_REALTIME_WHISPER_MODEL:
if transcription.prompt:
transcription.prompt = None
logger.warning(
f"{OPENAI_REALTIME_WHISPER_MODEL} does not support the prompt "
"parameter; omitting prompt from OpenAI Realtime input audio "
"transcription settings."
)
return True
return False
def can_generate_metrics(self) -> bool:
"""Check if the service can generate usage metrics.
@@ -487,7 +513,7 @@ class OpenAIRealtimeLLMService(LLMService[OpenAIRealtimeLLMAdapter]):
self._current_audio_response = None
def _calculate_audio_duration_ms(
self, total_bytes: int, sample_rate: int = 24000, bytes_per_sample: int = 2
self, total_bytes: int, sample_rate: int = OPENAI_SAMPLE_RATE, bytes_per_sample: int = 2
) -> int:
"""Calculate audio duration in milliseconds based on PCM audio parameters."""
samples = total_bytes / bytes_per_sample
@@ -656,8 +682,12 @@ class OpenAIRealtimeLLMService(LLMService[OpenAIRealtimeLLMAdapter]):
async def _update_settings(self, delta):
"""Apply a settings delta, sending a session update when needed."""
changed = await super()._update_settings(delta)
prompt_omitted = self._omit_unsupported_input_audio_transcription_prompt(
assert_given(self._settings.session_properties)
)
handled = {"session_properties", "system_instruction"}
if changed.keys() & handled:
handled_settings_changed = bool(changed.keys() & handled)
if handled_settings_changed or prompt_omitted:
await self._send_session_update()
self._warn_unhandled_updated_settings(changed.keys() - handled)
return changed
@@ -816,7 +846,7 @@ class OpenAIRealtimeLLMService(LLMService[OpenAIRealtimeLLMAdapter]):
self._current_audio_response.total_size += len(audio)
frame = TTSAudioRawFrame(
audio=audio,
sample_rate=24000,
sample_rate=OPENAI_SAMPLE_RATE,
num_channels=1,
)
await self.push_frame(frame)

45
src/pipecat/services/openai/stt.py
View File

@@ -36,6 +36,7 @@ from pipecat.frames.frames import (
VADUserStoppedSpeakingFrame,
)
from pipecat.processors.frame_processor import FrameDirection
from pipecat.services.openai._constants import OPENAI_REALTIME_WHISPER_MODEL, OPENAI_SAMPLE_RATE
from pipecat.services.settings import NOT_GIVEN, STTSettings, _NotGiven, assert_given
from pipecat.services.stt_latency import OPENAI_REALTIME_TTFS_P99, OPENAI_TTFS_P99
from pipecat.services.stt_service import WebsocketSTTService
@@ -178,15 +179,13 @@ class OpenAISTTService(BaseWhisperSTTService):
return await self._client.audio.transcriptions.create(**kwargs)
_OPENAI_SAMPLE_RATE = 24000
@dataclass
class OpenAIRealtimeSTTSettings(STTSettings):
"""Settings for OpenAIRealtimeSTTService.
Parameters:
prompt: Optional prompt text to guide transcription style.
prompt: Optional prompt text to guide transcription style. Not supported by
``"gpt-realtime-whisper"``.
noise_reduction: Noise reduction mode. ``"near_field"`` for close
microphones, ``"far_field"`` for distant microphones, or ``None``
to disable.
@@ -227,7 +226,7 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
stt = OpenAIRealtimeSTTService(
api_key="sk-...",
settings=OpenAIRealtimeSTTService.Settings(
model="gpt-4o-transcribe",
model="gpt-realtime-whisper",
noise_reduction="near_field",
),
)
@@ -255,7 +254,9 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
Args:
api_key: OpenAI API key for authentication.
model: Transcription model. Supported values are
model: Transcription model. For low-latency streaming
transcription, use ``"gpt-realtime-whisper"``. Other
supported transcription models include
``"gpt-4o-transcribe"`` and ``"gpt-4o-mini-transcribe"``.
.. deprecated:: 0.0.105
@@ -269,7 +270,8 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
Use ``settings=OpenAIRealtimeSTTService.Settings(language=...)`` instead.
prompt: Optional prompt text to guide transcription style
or provide keyword hints.
or provide keyword hints. Not supported by
``"gpt-realtime-whisper"``.
.. deprecated:: 0.0.105
Use ``settings=OpenAIRealtimeSTTService.Settings(prompt=...)`` instead.
@@ -303,7 +305,7 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
# --- 1. Hardcoded defaults ---
default_settings = self.Settings(
model="gpt-4o-transcribe",
model=OPENAI_REALTIME_WHISPER_MODEL,
language=Language.EN,
prompt=None,
noise_reduction=None,
@@ -329,6 +331,8 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
if settings is not None:
default_settings.apply_update(settings)
self._omit_unsupported_prompt(default_settings)
super().__init__(
ttfs_p99_latency=ttfs_p99_latency,
settings=default_settings,
@@ -349,6 +353,19 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
# Set to None or a dict to enable server-side VAD.
self._server_vad_enabled = turn_detection is not False
@staticmethod
def _omit_unsupported_prompt(settings: OpenAIRealtimeSTTSettings) -> dict[str, Any]:
"""Drop prompt settings that are not accepted by the selected model."""
if settings.model == OPENAI_REALTIME_WHISPER_MODEL and settings.prompt:
old_prompt = settings.prompt
settings.prompt = None
logger.warning(
f"{OPENAI_REALTIME_WHISPER_MODEL} does not support the prompt parameter; "
"omitting prompt from OpenAI Realtime transcription session."
)
return {"prompt": old_prompt}
return {}
@staticmethod
def _language_to_code(language: Language) -> str:
"""Convert a Language enum value to an ISO-639-1 code.
@@ -382,6 +399,8 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
Dict mapping changed field names to their previous values.
"""
changed = await super()._update_settings(delta)
for field, previous_value in self._omit_unsupported_prompt(self._settings).items():
changed.setdefault(field, previous_value)
if changed and self._session_ready:
await self._send_session_update()
@@ -550,7 +569,7 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
input_audio: dict = {
"format": {
"type": "audio/pcm",
"rate": _OPENAI_SAMPLE_RATE,
"rate": OPENAI_SAMPLE_RATE,
},
"transcription": transcription,
}
@@ -587,7 +606,7 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
Args:
audio: Raw audio bytes at the pipeline sample rate.
"""
audio = await self._resampler.resample(audio, self.sample_rate, _OPENAI_SAMPLE_RATE)
audio = await self._resampler.resample(audio, self.sample_rate, OPENAI_SAMPLE_RATE)
if not audio:
return
payload = base64.b64encode(audio).decode("utf-8")
@@ -676,9 +695,9 @@ class OpenAIRealtimeSTTService(WebsocketSTTService):
async def _handle_transcription_delta(self, evt: dict):
"""Handle incremental transcription text.
For ``gpt-4o-transcribe`` and ``gpt-4o-mini-transcribe``, deltas
contain streaming partial text. For ``whisper-1``, each delta
contains the full turn transcript.
For ``gpt-realtime-whisper``, ``gpt-4o-transcribe``, and
``gpt-4o-mini-transcribe``, deltas contain low-latency streaming
partial text.
Args:
evt: The delta event from the server.

11
src/pipecat/services/openai/tts.py
View File

@@ -24,6 +24,7 @@ from pipecat.frames.frames import (
StartFrame,
TTSAudioRawFrame,
)
from pipecat.services.openai._constants import OPENAI_SAMPLE_RATE
from pipecat.services.settings import NOT_GIVEN, TTSSettings, _NotGiven, assert_given
from pipecat.services.tts_service import TTSService
from pipecat.utils.tracing.service_decorators import traced_tts
@@ -85,8 +86,6 @@ class OpenAITTSService(TTSService):
Settings = OpenAITTSSettings
_settings: Settings
OPENAI_SAMPLE_RATE = 24000 # OpenAI TTS always outputs at 24kHz
class InputParams(BaseModel):
"""Input parameters for OpenAI TTS configuration.
@@ -150,9 +149,9 @@ class OpenAITTSService(TTSService):
parameters, ``settings`` values take precedence.
**kwargs: Additional keyword arguments passed to TTSService.
"""
if sample_rate and sample_rate != self.OPENAI_SAMPLE_RATE:
if sample_rate and sample_rate != OPENAI_SAMPLE_RATE:
logger.warning(
f"OpenAI TTS only supports {self.OPENAI_SAMPLE_RATE}Hz sample rate. "
f"OpenAI TTS only supports {OPENAI_SAMPLE_RATE}Hz sample rate. "
f"Current rate of {sample_rate}Hz may cause issues."
)
@@ -217,9 +216,9 @@ class OpenAITTSService(TTSService):
frame: The start frame containing initialization parameters.
"""
await super().start(frame)
if self.sample_rate != self.OPENAI_SAMPLE_RATE:
if self.sample_rate != OPENAI_SAMPLE_RATE:
logger.warning(
f"OpenAI TTS requires {self.OPENAI_SAMPLE_RATE}Hz sample rate. "
f"OpenAI TTS requires {OPENAI_SAMPLE_RATE}Hz sample rate. "
f"Current rate of {self.sample_rate}Hz may cause issues."
)