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3 Commits

Author SHA1 Message Date
Filipi Fuchter
332a6b3669 Addressing the comments left in the PR review. 2025-09-23 18:33:55 -03:00
Filipi Fuchter
be570f593f Overriding the _start_interruption method. 2025-09-23 16:33:16 -03:00
Filipi Fuchter
12dcc87030 Fixed an issue in BaseOpenAILLMService that could cause the pipeline to freeze under certain race conditions. 2025-09-23 16:23:57 -03:00
54 changed files with 478 additions and 1294 deletions

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@@ -9,40 +9,6 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
### Added
- Added RTVI messages for user/bot audio levels and system logs.
- Include OpenAI-based LLM services cached tokens to `MetricsFrame`.
### Changed
- Updated the default model for `AnthropicLLMService` to
`claude-sonnet-4-5-20250929`.
### Deprecated
- `DailyUpdateRemoteParticipantsFrame` is deprecated and will be removed in a
future version. Instead, create your own custom frame and handle it in the
`@transport.output().event_handler("on_after_push_frame")` event handler or a
custom processor.
## Fixed
- Fixed a `PipelineTask` issue that could prevent the application to exit if
`task.cancel()` was called when the task was already finished.
- Fixed an issue where local SmartTurn was not being ran in a separate thread.
## [0.0.86] - 2025-09-24
### Added
- Added `HeyGenTransport`. This is an integration for HeyGen Interactive
Avatar. A video service that handles audio streaming and requests HeyGen to
generate avatar video responses. (see https://www.heygen.com/). When used, the
Pipecat bot joins the same virtual room as the HeyGen Avatar and the user.
- Added support to `TwilioFrameSerializer` for `region` and `edge` settings.
- Added support for using universal `LLMContext` with:
- `LLMLogObserver`
@@ -102,10 +68,6 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
### Changed
- Updated `aiortc` to 1.13.0.
- Updated `sentry` to 2.38.0.
- `BaseOutputTransport` methods `write_audio_frame` and `write_video_frame` now
return a boolean to indicate if the transport implementation was able to write
the given frame or not.
@@ -140,14 +102,8 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
### Fixed
- Fixed an issue where the pipeline could freeze if a task cancellation never
completed because a third-party library swallowed asyncio.CancelledError. We
now apply a timeout to task cancellations to prevent these freezes. If the
timeout is reached, the system logs warnings and leaves dangling tasks behind,
which can help diagnose where cancellation is being blocked.
- Fixed an `AudioBufferProcessor` issues that was causing user audio to be
missing in stereo recordings causing bot and user overlaps.
- Fixed an issue in BaseOpenAILLMService that could cause the pipeline to freeze
under certain race conditions.
- Fixed a `BaseOutputTransport` issue that could produce large saved
`AudioBufferProcessor` files when using an audio mixer.

View File

@@ -66,8 +66,8 @@ LMNT_VOICE_ID=...
PERPLEXITY_API_KEY=...
# PlayHT
PLAYHT_USER_ID=...
PLAYHT_API_KEY=...
PLAY_HT_USER_ID=...
PLAY_HT_API_KEY=...
# OpenAI
OPENAI_API_KEY=...

View File

@@ -5,7 +5,6 @@
#
import os
import wave
from dotenv import load_dotenv
from loguru import logger
@@ -14,14 +13,7 @@ from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
from pipecat.audio.turn.smart_turn.local_smart_turn_v3 import LocalSmartTurnAnalyzerV3
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
from pipecat.frames.frames import (
LLMFullResponseEndFrame,
LLMFullResponseStartFrame,
LLMRunFrame,
LLMTextFrame,
OutputAudioRawFrame,
TextFrame,
)
from pipecat.frames.frames import LLMRunFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
@@ -111,27 +103,7 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
logger.info(f"Client connected")
# Kick off the conversation.
messages.append({"role": "system", "content": "Please introduce yourself to the user."})
audio_file_path = os.path.join(os.path.dirname(__file__), "assets", "pre-recorded.wav")
with wave.open(audio_file_path, "rb") as wav_file:
llm_text_frame = TextFrame(text="This is a pre-recorded message.")
llm_text_frame.skip_tts = True
audio_data = wav_file.readframes(wav_file.getnframes())
output_audio_raw_frame = OutputAudioRawFrame(
audio=audio_data, sample_rate=44100, num_channels=1
)
await task.queue_frames(
[
LLMRunFrame(),
LLMFullResponseStartFrame(),
llm_text_frame,
output_audio_raw_frame,
LLMFullResponseEndFrame(),
]
)
await task.queue_frames([LLMRunFrame()])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):

View File

@@ -129,7 +129,7 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
async def on_client_connected(transport, client):
logger.info(f"Client connected")
# Kick off the conversation.
# An `LLMContextFrame` will be picked up by the LangchainProcessor using
# An `OpenAILLMContextFrame` will be picked up by the LangchainProcessor using
# only the content of the last message to inject it in the prompt defined
# above. So no role is required here.
messages = [({"content": "Please briefly introduce yourself to the user."})]

View File

@@ -9,12 +9,14 @@ from dotenv import load_dotenv
from loguru import logger
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import LLMMessagesAppendFrame, LLMRunFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.aggregators.llm_response import (
LLMAssistantContextAggregator,
LLMUserContextAggregator,
)
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.frameworks.strands_agents import StrandsAgentsProcessor
from pipecat.runner.types import RunnerArguments
from pipecat.runner.utils import create_transport
@@ -113,18 +115,19 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
)
# Setup context aggregators for message handling
context = LLMContext()
context_aggregator = LLMContextAggregatorPair(context)
context = OpenAILLMContext()
tma_in = LLMUserContextAggregator(context=context)
tma_out = LLMAssistantContextAggregator(context=context)
pipeline = Pipeline(
[
transport.input(), # Transport user input
stt, # Speech-to-text
context_aggregator.user(), # User responses
tma_in, # User context aggregator
llm, # Strands Agents processor
tts, # Text-to-speech
transport.output(), # Transport bot output
context_aggregator.assistant(), # Assistant spoken responses
tma_out, # Assistant context aggregator
]
)
@@ -140,20 +143,6 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
logger.info(f"Client connected")
# Kick off the conversation.
await task.queue_frames(
[
LLMMessagesAppendFrame(
messages=[
{
"role": "user",
"content": f"Greet the user and introduce yourself.",
}
],
run_llm=True,
)
]
)
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):

View File

@@ -9,9 +9,8 @@ import os
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from pipecat.adapters.schemas.function_schema import FunctionSchema
from pipecat.adapters.schemas.tools_schema import ToolsSchema
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
CancelFrame,
@@ -20,7 +19,6 @@ from pipecat.frames.frames import (
FunctionCallInProgressFrame,
FunctionCallResultFrame,
InterruptionFrame,
LLMContextFrame,
LLMRunFrame,
StartFrame,
SystemFrame,
@@ -34,8 +32,10 @@ from pipecat.pipeline.parallel_pipeline import ParallelPipeline
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.processors.filters.function_filter import FunctionFilter
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.user_idle_processor import UserIdleProcessor
@@ -66,13 +66,13 @@ class StatementJudgeContextFilter(FrameProcessor):
await self.push_frame(frame, direction)
return
# We only want to handle LLMContextFrames, and only want to push through a simplified
# We only want to handle OpenAILLMContextFrames, and only want to push through a simplified
# context frame that contains a system prompt and the most recent user messages,
# concatenated.
if isinstance(frame, LLMContextFrame):
if isinstance(frame, OpenAILLMContextFrame):
logger.debug(f"Context Frame: {frame}")
# Take text content from the most recent user messages.
messages = frame.context.get_messages()
messages = frame.context.messages
user_text_messages = []
last_assistant_message = None
for message in reversed(messages):
@@ -100,7 +100,7 @@ class StatementJudgeContextFilter(FrameProcessor):
if last_assistant_message:
messages.append(last_assistant_message)
messages.append({"role": "user", "content": user_message})
await self.push_frame(LLMContextFrame(LLMContext(messages)))
await self.push_frame(OpenAILLMContextFrame(OpenAILLMContext(messages)))
class CompletenessCheck(FrameProcessor):
@@ -231,26 +231,22 @@ class TurnDetectionLLM(Pipeline):
async def pass_only_llm_trigger_frames(frame):
return (
isinstance(frame, LLMContextFrame)
isinstance(frame, OpenAILLMContextFrame)
or isinstance(frame, InterruptionFrame)
or isinstance(frame, FunctionCallInProgressFrame)
or isinstance(frame, FunctionCallResultFrame)
)
async def filter_all(frame):
return False
super().__init__(
[
ParallelPipeline(
[
# Ignore everything except an LLMContextFrame. Pass a specially constructed
# Ignore everything except an OpenAILLMContextFrame. Pass a specially constructed
# simplified context frame to the statement classifier LLM. The only frame this
# sub-pipeline will output is a UserStoppedSpeakingFrame.
statement_judge_context_filter,
statement_llm,
completeness_check,
FunctionFilter(filter=filter_all, direction=FrameDirection.UPSTREAM),
],
[
# Block everything except frames that trigger LLM inference.
@@ -306,23 +302,30 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
async def on_function_calls_started(service, function_calls):
await tts.queue_frame(TTSSpeakFrame("Let me check on that."))
weather_function = FunctionSchema(
name="get_current_weather",
description="Get the current weather",
properties={
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
tools = [
ChatCompletionToolParam(
type="function",
function={
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location", "format"],
},
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
required=["location", "format"],
)
tools = ToolsSchema(standard_tools=[weather_function])
)
]
messages = [
{
@@ -331,8 +334,8 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
},
]
context = LLMContext(messages, tools)
context_aggregator = LLMContextAggregatorPair(context)
context = OpenAILLMContext(messages, tools)
context_aggregator = llm_main.create_context_aggregator(context)
# LLM + turn detection (with an extra LLM as a judge)
llm = TurnDetectionLLM(llm_main)
@@ -366,7 +369,7 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
await task.queue_frames([LLMRunFrame()])
@transport.event_handler("on_app_message")
async def on_app_message(transport, message, sender):
async def on_app_message(transport, message):
logger.debug(f"Received app message: {message}")
if "message" not in message:
return

View File

@@ -9,9 +9,8 @@ import os
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from pipecat.adapters.schemas.function_schema import FunctionSchema
from pipecat.adapters.schemas.tools_schema import ToolsSchema
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
CancelFrame,
@@ -20,7 +19,6 @@ from pipecat.frames.frames import (
FunctionCallInProgressFrame,
FunctionCallResultFrame,
InterruptionFrame,
LLMContextFrame,
LLMRunFrame,
StartFrame,
SystemFrame,
@@ -34,8 +32,10 @@ from pipecat.pipeline.parallel_pipeline import ParallelPipeline
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.processors.filters.function_filter import FunctionFilter
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.user_idle_processor import UserIdleProcessor
@@ -272,11 +272,11 @@ class StatementJudgeContextFilter(FrameProcessor):
await self.push_frame(frame, direction)
return
# We only want to handle LLMContextFrames, and only want to push through a simplified
# We only want to handle OpenAILLMContextFrames, and only want to push through a simplified
# context frame that contains a system prompt and the most recent user messages,
if isinstance(frame, LLMContextFrame):
if isinstance(frame, OpenAILLMContextFrame):
# Take text content from the most recent user messages.
messages = frame.context.get_messages()
messages = frame.context.messages
user_text_messages = []
last_assistant_message = None
for message in reversed(messages):
@@ -303,7 +303,7 @@ class StatementJudgeContextFilter(FrameProcessor):
if last_assistant_message:
messages.append(last_assistant_message)
messages.append({"role": "user", "content": user_message})
await self.push_frame(LLMContextFrame(LLMContext(messages)))
await self.push_frame(OpenAILLMContextFrame(OpenAILLMContext(messages)))
class CompletenessCheck(FrameProcessor):
@@ -425,15 +425,12 @@ class TurnDetectionLLM(Pipeline):
async def pass_only_llm_trigger_frames(frame):
return (
isinstance(frame, LLMContextFrame)
isinstance(frame, OpenAILLMContextFrame)
or isinstance(frame, InterruptionFrame)
or isinstance(frame, FunctionCallInProgressFrame)
or isinstance(frame, FunctionCallResultFrame)
)
async def filter_all(frame):
return False
super().__init__(
[
ParallelPipeline(
@@ -443,13 +440,12 @@ class TurnDetectionLLM(Pipeline):
FunctionFilter(filter=block_user_stopped_speaking),
],
[
# Ignore everything except an LLMContextFrame. Pass a specially constructed
# Ignore everything except an OpenAILLMContextFrame. Pass a specially constructed
# simplified context frame to the statement classifier LLM. The only frame this
# sub-pipeline will output is a UserStoppedSpeakingFrame.
statement_judge_context_filter,
statement_llm,
completeness_check,
FunctionFilter(filter=filter_all, direction=FrameDirection.UPSTREAM),
],
[
# Block everything except frames that trigger LLM inference.
@@ -509,23 +505,30 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
async def on_function_calls_started(service, function_calls):
await tts.queue_frame(TTSSpeakFrame("Let me check on that."))
weather_function = FunctionSchema(
name="get_current_weather",
description="Get the current weather",
properties={
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
tools = [
ChatCompletionToolParam(
type="function",
function={
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location", "format"],
},
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
required=["location", "format"],
)
tools = ToolsSchema(standard_tools=[weather_function])
)
]
messages = [
{
@@ -534,8 +537,8 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
},
]
context = LLMContext(messages, tools)
context_aggregator = LLMContextAggregatorPair(context)
context = OpenAILLMContext(messages, tools)
context_aggregator = llm_main.create_context_aggregator(context)
# LLM + turn detection (with an extra LLM as a judge)
llm = TurnDetectionLLM(llm_main)
@@ -574,7 +577,7 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
await task.queue_frames([LLMRunFrame()])
@transport.event_handler("on_app_message")
async def on_app_message(transport, message, sender):
async def on_app_message(transport, message):
logger.debug(f"Received app message: {message}")
if "message" not in message:
return

View File

@@ -9,6 +9,7 @@ import os
import time
from dotenv import load_dotenv
from google.genai.types import Content, Part
from loguru import logger
from pipecat.audio.vad.silero import SileroVADAnalyzer
@@ -20,7 +21,6 @@ from pipecat.frames.frames import (
FunctionCallResultFrame,
InputAudioRawFrame,
InterruptionFrame,
LLMContextFrame,
LLMFullResponseStartFrame,
LLMRunFrame,
StartFrame,
@@ -34,18 +34,20 @@ from pipecat.pipeline.parallel_pipeline import ParallelPipeline
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response import (
LLMAssistantAggregatorParams,
LLMAssistantResponseAggregator,
)
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.processors.filters.function_filter import FunctionFilter
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.runner.types import RunnerArguments
from pipecat.runner.utils import create_transport
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.google.llm import GoogleLLMService
from pipecat.services.google.llm import GoogleLLMContext, GoogleLLMService
from pipecat.services.llm_service import LLMService
from pipecat.sync.base_notifier import BaseNotifier
from pipecat.sync.event_notifier import EventNotifier
@@ -373,7 +375,7 @@ class AudioAccumulator(FrameProcessor):
await super().process_frame(frame, direction)
# ignore context frame
if isinstance(frame, LLMContextFrame):
if isinstance(frame, OpenAILLMContextFrame):
return
if isinstance(frame, TranscriptionFrame):
@@ -390,9 +392,9 @@ class AudioAccumulator(FrameProcessor):
f"Processing audio buffer seconds: ({len(self._audio_frames)}) ({len(data)}) {len(data) / 2 / 16000}"
)
self._user_speaking = False
context = LLMContext()
context = GoogleLLMContext()
context.add_audio_frames_message(audio_frames=self._audio_frames)
await self.push_frame(LLMContextFrame(context=context))
await self.push_frame(OpenAILLMContextFrame(context=context))
elif isinstance(frame, InputAudioRawFrame):
# Append the audio frame to our buffer. Treat the buffer as a ring buffer, dropping the oldest
# frames as necessary.
@@ -511,7 +513,7 @@ class LLMAggregatorBuffer(LLMAssistantResponseAggregator):
class ConversationAudioContextAssembler(FrameProcessor):
"""Takes the single-message context generated by the AudioAccumulator and adds it to the conversation LLM's context."""
def __init__(self, context: LLMContext, **kwargs):
def __init__(self, context: OpenAILLMContext, **kwargs):
super().__init__(**kwargs)
self._context = context
@@ -523,10 +525,11 @@ class ConversationAudioContextAssembler(FrameProcessor):
await self.push_frame(frame, direction)
return
if isinstance(frame, LLMContextFrame):
last_message = frame.context.get_messages()[-1]
if isinstance(frame, OpenAILLMContextFrame):
GoogleLLMContext.upgrade_to_google(self._context)
last_message = frame.context.messages[-1]
self._context._messages.append(last_message)
await self.push_frame(LLMContextFrame(context=self._context))
await self.push_frame(OpenAILLMContextFrame(context=self._context))
class OutputGate(FrameProcessor):
@@ -540,7 +543,7 @@ class OutputGate(FrameProcessor):
def __init__(
self,
notifier: BaseNotifier,
context: LLMContext,
context: OpenAILLMContext,
llm_transcription_buffer: LLMAggregatorBuffer,
**kwargs,
):
@@ -607,23 +610,19 @@ class OutputGate(FrameProcessor):
self._gate_task = None
async def _gate_task_handler(self):
while True:
try:
await self._notifier.wait()
await self._notifier.wait()
transcription = await self._transcription_buffer.wait_for_transcription() or "-"
self._context.add_message({"role": "user", "content": transcription})
transcription = await self._transcription_buffer.wait_for_transcription() or "-"
self._context.add_message(Content(role="user", parts=[Part(text=transcription)]))
self.open_gate()
for frame, direction in self._frames_buffer:
await self.push_frame(frame, direction)
self._frames_buffer = []
except asyncio.CancelledError:
break
self.open_gate()
for frame, direction in self._frames_buffer:
await self.push_frame(frame, direction)
self._frames_buffer = []
class TurnDetectionLLM(Pipeline):
def __init__(self, llm: LLMService, context: LLMContext):
def __init__(self, llm: LLMService, context: OpenAILLMContext):
# This is the LLM that will transcribe user speech.
tx_llm = GoogleLLMService(
name="Transcriber",
@@ -649,10 +648,10 @@ class TurnDetectionLLM(Pipeline):
# as complete or incomplete.
# statement_judge_context_filter = StatementJudgeAudioContextAccumulator(notifier=notifier)
audio_accumulator = AudioAccumulator()
audio_accumulater = AudioAccumulator()
# This sends a UserStoppedSpeakingFrame and triggers the notifier event
completeness_check = CompletenessCheck(
notifier=notifier, audio_accumulator=audio_accumulator
notifier=notifier, audio_accumulator=audio_accumulater
)
async def block_user_stopped_speaking(frame):
@@ -668,7 +667,7 @@ class TurnDetectionLLM(Pipeline):
super().__init__(
[
audio_accumulator,
audio_accumulater,
ParallelPipeline(
[
# Pass everything except UserStoppedSpeaking to the elements after
@@ -735,8 +734,8 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
system_instruction=conversation_system_instruction,
)
context = LLMContext()
context_aggregator = LLMContextAggregatorPair(context)
context = OpenAILLMContext()
context_aggregator = conversation_llm.create_context_aggregator(context)
llm = TurnDetectionLLM(conversation_llm, context)
@@ -762,10 +761,12 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
logger.info(f"Client connected")
# Kick off the conversation.
await task.queue_frames([LLMRunFrame()])
@transport.event_handler("on_app_message")
async def on_app_message(transport, message, sender):
logger.debug(f"Received app message: {message}, sender: {sender}") # TODO: revert
async def on_app_message(transport, message):
logger.debug(f"Received app message: {message}")
if "message" not in message:
return

View File

@@ -8,13 +8,13 @@ import os
from dataclasses import dataclass
from dotenv import load_dotenv
from google.genai.types import Content, Part
from loguru import logger
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
Frame,
InputAudioRawFrame,
LLMContextFrame,
LLMFullResponseEndFrame,
LLMRunFrame,
SystemFrame,
@@ -27,13 +27,15 @@ from pipecat.pipeline.parallel_pipeline import ParallelPipeline
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.processors.frame_processor import FrameProcessor
from pipecat.runner.types import RunnerArguments
from pipecat.runner.utils import create_transport
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.google.llm import GoogleLLMService
from pipecat.services.google.llm import GoogleLLMContext, GoogleLLMService
from pipecat.transports.base_transport import BaseTransport, TransportParams
from pipecat.transports.daily.transport import DailyParams
from pipecat.transports.websocket.fastapi import FastAPIWebsocketParams
@@ -99,7 +101,9 @@ class UserAudioCollector(FrameProcessor):
elif isinstance(frame, UserStoppedSpeakingFrame):
self._user_speaking = False
self._context.add_audio_frames_message(audio_frames=self._audio_frames)
await self._user_context_aggregator.push_frame(LLMContextFrame(context=self._context))
await self._user_context_aggregator.push_frame(
self._user_context_aggregator.get_context_frame()
)
elif isinstance(frame, InputAudioRawFrame):
if self._user_speaking:
self._audio_frames.append(frame)
@@ -117,10 +121,10 @@ class UserAudioCollector(FrameProcessor):
class InputTranscriptionContextFilter(FrameProcessor):
"""This FrameProcessor blocks all frames except the LLMContextFrame that triggers
"""This FrameProcessor blocks all frames except the OpenAILLMContextFrame that triggers
LLM inference. (And system frames, which are needed for the pipeline element lifecycle.)
We take the context object out of the LLMContextFrame and use it to create a new
We take the context object out of the OpenAILLMContextFrame and use it to create a new
context object that we will send to the transcriber LLM.
"""
@@ -132,54 +136,52 @@ class InputTranscriptionContextFilter(FrameProcessor):
await self.push_frame(frame, direction)
return
if not isinstance(frame, LLMContextFrame):
if not isinstance(frame, OpenAILLMContextFrame):
return
try:
message = frame.context.get_messages()[-1]
# Make sure we're working with a GoogleLLMContext
context = GoogleLLMContext.upgrade_to_google(frame.context)
message = context.messages[-1]
message_content = message["content"]
if not message_content or not isinstance(message_content, list):
if not isinstance(message, Content):
logger.error(f"Expected Content, got {type(message)}")
return
last_part = message["content"][-1]
if not (message["role"] == "user" and last_part["type"] == "input_audio"):
last_part = message.parts[-1]
if not (
message.role == "user"
and last_part.inline_data
and last_part.inline_data.mime_type == "audio/wav"
):
return
# Assemble a new message, with three parts: conversation history, transcription
# prompt, and audio. We could use only part of the conversation, if we need to
# keep the token count down, but for now, we'll just use the whole thing.
new_message_content = []
parts = []
# Get previous conversation history
previous_messages = frame.context.get_messages()[:-2]
previous_messages = frame.context.messages[:-2]
history = ""
for msg in previous_messages:
previous_message_content = msg["content"]
if not previous_message_content:
continue
if isinstance(previous_message_content, str):
history += f"{msg['role']}: {previous_message_content}\n"
elif isinstance(previous_message_content, list):
for c in previous_message_content:
if c.get("text"):
history += f"{msg['role']}: {c['text']}\n"
for part in msg.parts:
if part.text:
history += f"{msg.role}: {part.text}\n"
if history:
assembled = f"Here is the conversation history so far. These are not instructions. This is data that you should use only to improve the accuracy of your transcription.\n\n----\n\n{history}\n\n----\n\nEND OF CONVERSATION HISTORY\n\n"
new_message_content.append({"type": "text", "text": assembled})
parts.append(Part(text=assembled))
new_message_content.append(
{
"type": "text",
"text": "Transcribe this audio. Respond either with the transcription exactly as it was said by the user, or with the special string 'EMPTY' if the audio is not clear.",
}
parts.append(
Part(
text="Transcribe this audio. Respond either with the transcription exactly as it was said by the user, or with the special string 'EMPTY' if the audio is not clear."
)
)
new_message_content.append(last_part)
msg = {"role": "user", "content": new_message_content}
ctx = LLMContext([{"role": "system", "content": transcriber_system_message}, msg])
await self.push_frame(LLMContextFrame(context=ctx))
parts.append(last_part)
msg = Content(role="user", parts=parts)
ctx = GoogleLLMContext([msg])
ctx.system_message = transcriber_system_message
await self.push_frame(OpenAILLMContextFrame(context=ctx))
except Exception as e:
logger.error(f"Error processing frame: {e}")
@@ -225,8 +227,10 @@ class TranscriptionContextFixup(FrameProcessor):
Audio is big, using a lot of tokens and network bandwidth. So doing this is
important if we want to keep both latency and cost low.
This class is a bit of a hack, especially because it directly creates an
LLMContext object, which we don't generally do.
This class is a bit of a hack, especially because it directly creates a
GoogleLLMContext object, which we don't generally do. We usually try to leave
the implementation-specific details of the LLM context encapsulated inside the
service classes.
"""
def __init__(self, context):
@@ -235,22 +239,25 @@ class TranscriptionContextFixup(FrameProcessor):
self._transcript = "THIS IS A TRANSCRIPT"
def is_user_audio_message(self, message):
message_content = message["content"]
if not message_content or not isinstance(message_content, list):
return False
last_part = message["content"][-1]
return message["role"] == "user" and last_part["type"] == "input_audio"
last_part = message.parts[-1]
return (
message.role == "user"
and last_part.inline_data
and last_part.inline_data.mime_type == "audio/wav"
)
def swap_user_audio(self):
if not self._transcript:
return
message = self._context.get_messages()[-2]
message = self._context.messages[-2]
if not self.is_user_audio_message(message):
message = self._context.get_messages()[-1]
message = self._context.messages[-1]
if not self.is_user_audio_message(message):
return
message["content"] = self._transcript
audio_part = message.parts[-1]
audio_part.inline_data = None
audio_part.text = self._transcript
async def process_frame(self, frame, direction):
await super().process_frame(frame, direction)
@@ -320,8 +327,8 @@ async def run_bot(transport: BaseTransport, runner_args: RunnerArguments):
},
]
context = LLMContext(messages)
context_aggregator = LLMContextAggregatorPair(context)
context = OpenAILLMContext(messages)
context_aggregator = conversation_llm.create_context_aggregator(context)
audio_collector = UserAudioCollector(context, context_aggregator.user())
input_transcription_context_filter = InputTranscriptionContextFilter()
transcription_frames_emitter = InputTranscriptionFrameEmitter()

View File

@@ -206,14 +206,6 @@ async def bot(runner_args: RunnerArguments):
if __name__ == "__main__":
if not os.getenv("NASA_API_KEY"):
logger.error(
f"Please set NASA_API_KEY environment variable for this example. See https://api.nasa.gov"
)
import sys
sys.exit(1)
from pipecat.runner.run import main
main()

View File

@@ -141,14 +141,6 @@ async def bot(runner_args: RunnerArguments):
if __name__ == "__main__":
if not os.getenv("MCP_RUN_SSE_URL"):
logger.error(
f"Please set MCP_RUN_SSE_URL environment variable for this example. See https://mcp.run"
)
import sys
sys.exit(1)
from pipecat.runner.run import main
main()

View File

@@ -219,14 +219,6 @@ async def bot(runner_args: RunnerArguments):
if __name__ == "__main__":
if not os.getenv("NASA_API_KEY") or not os.getenv("MCP_RUN_SSE_URL"):
logger.error(
f"Please set NASA_API_KEY and MCP_RUN_SSE_URL environment variables. See https://api.nasa.gov and https://mcp.run"
)
import sys
sys.exit(1)
from pipecat.runner.run import main
main()

View File

@@ -145,14 +145,6 @@ async def bot(runner_args: RunnerArguments):
if __name__ == "__main__":
if not os.getenv("GITHUB_PERSONAL_ACCESS_TOKEN"):
logger.error(
f"Please set GITHUB_PERSONAL_ACCESS_TOKEN environment variable for this example."
)
import sys
sys.exit(1)
from pipecat.runner.run import main
main()

View File

@@ -1,113 +0,0 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
from pipecat.audio.turn.smart_turn.local_smart_turn_v3 import LocalSmartTurnAnalyzerV3
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
from pipecat.frames.frames import LLMRunFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_response_universal import (
LLMContext,
LLMContextAggregatorPair,
)
from pipecat.services.cartesia.tts import CartesiaTTSService
from pipecat.services.deepgram.stt import DeepgramSTTService
from pipecat.services.google.llm import GoogleLLMService
from pipecat.transports.heygen.transport import HeyGenParams, HeyGenTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
transport = HeyGenTransport(
api_key=os.getenv("HEYGEN_API_KEY"),
session=session,
params=HeyGenParams(
audio_in_enabled=True,
audio_out_enabled=True,
vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.2)),
turn_analyzer=LocalSmartTurnAnalyzerV3(params=SmartTurnParams()),
),
)
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="00967b2f-88a6-4a31-8153-110a92134b9f",
)
llm = GoogleLLMService(api_key=os.getenv("GOOGLE_API_KEY"))
messages = [
{
"role": "system",
"content": "You are a helpful assistant. Your output will be converted to audio so don't include special characters in your answers. Be succinct and respond to what the user said in a creative and helpful way.",
},
]
context = LLMContext(messages)
context_aggregator = LLMContextAggregatorPair(context)
pipeline = Pipeline(
[
transport.input(), # Transport user input
stt, # STT
context_aggregator.user(), # User responses
llm, # LLM
tts, # TTS
transport.output(), # Transport bot output
context_aggregator.assistant(), # Assistant spoken responses
]
)
task = PipelineTask(
pipeline,
params=PipelineParams(
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
logger.info(f"Client connected")
# Kick off the conversation.
messages.append(
{
"role": "system",
"content": "Start by saying 'Hello' and then a short greeting.",
}
)
await task.queue_frames([LLMRunFrame()])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
logger.info(f"Client disconnected")
await task.cancel()
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -14,12 +14,13 @@ from pipecat.audio.turn.smart_turn.local_smart_turn_v3 import LocalSmartTurnAnal
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
from pipecat.extensions.voicemail.voicemail_detector import VoicemailDetector
from pipecat.frames.frames import TTSSpeakFrame
from pipecat.frames.frames import EndTaskFrame, TTSSpeakFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.frame_processor import FrameDirection
from pipecat.runner.types import RunnerArguments
from pipecat.runner.utils import create_transport
from pipecat.services.cartesia.tts import CartesiaTTSService

View File

@@ -4,7 +4,7 @@ version = "0.1.0"
description = "Quickstart example for building voice AI bots with Pipecat"
requires-python = ">=3.10"
dependencies = [
"pipecat-ai[webrtc,daily,silero,deepgram,openai,cartesia,local-smart-turn-v3,runner]>=0.0.86",
"pipecat-ai[webrtc,daily,silero,deepgram,openai,cartesia,local-smart-turn-v3,runner]>=0.0.85",
"pipecatcloud>=0.2.4"
]

View File

@@ -93,7 +93,7 @@ riva = [ "nvidia-riva-client~=2.21.1" ]
runner = [ "python-dotenv>=1.0.0,<2.0.0", "uvicorn>=0.32.0,<1.0.0", "fastapi>=0.115.6,<0.117.0", "pipecat-ai-small-webrtc-prebuilt>=1.0.0"]
sambanova = []
sarvam = [ "pipecat-ai[websockets-base]" ]
sentry = [ "sentry-sdk>=2.28.0,<3" ]
sentry = [ "sentry-sdk~=2.23.1" ]
local-smart-turn = [ "coremltools>=8.0", "transformers", "torch>=2.5.0,<3", "torchaudio>=2.5.0,<3" ]
local-smart-turn-v3 = [ "transformers", "onnxruntime>=1.20.1,<2" ]
remote-smart-turn = []
@@ -107,7 +107,7 @@ tavus=[]
together = []
tracing = [ "opentelemetry-sdk>=1.33.0", "opentelemetry-api>=1.33.0", "opentelemetry-instrumentation>=0.54b0" ]
ultravox = [ "transformers>=4.48.0", "vllm>=0.9.0" ]
webrtc = [ "aiortc>=1.13.0,<2", "opencv-python>=4.11.0.86,<5" ]
webrtc = [ "aiortc~=1.13.0", "opencv-python~=4.11.0.86" ]
websocket = [ "pipecat-ai[websockets-base]", "fastapi>=0.115.6,<0.117.0" ]
websockets-base = [ "websockets>=13.1,<16.0" ]
whisper = [ "faster-whisper~=1.1.1" ]

View File

@@ -34,8 +34,7 @@ from pipecat.frames.frames import EndTaskFrame, LLMRunFrame, OutputImageRawFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.audio.audio_buffer_processor import AudioBufferProcessor
from pipecat.processors.frame_processor import FrameDirection
from pipecat.runner.types import RunnerArguments
@@ -284,8 +283,8 @@ async def run_eval_pipeline(
},
]
context = LLMContext(messages, tools)
context_aggregator = LLMContextAggregatorPair(context)
context = OpenAILLMContext(messages, tools)
context_aggregator = llm.create_context_aggregator(context)
audio_buffer = AudioBufferProcessor()

View File

@@ -83,7 +83,6 @@ TESTS_07 = [
("07k-interruptible-lmnt.py", PROMPT_SIMPLE_MATH, EVAL_SIMPLE_MATH, BOT_SPEAKS_FIRST),
("07l-interruptible-groq.py", PROMPT_SIMPLE_MATH, EVAL_SIMPLE_MATH, BOT_SPEAKS_FIRST),
("07m-interruptible-aws.py", PROMPT_SIMPLE_MATH, EVAL_SIMPLE_MATH, BOT_SPEAKS_FIRST),
("07m-interruptible-aws-strands.py", PROMPT_WEATHER, EVAL_WEATHER, BOT_SPEAKS_FIRST),
("07n-interruptible-gemini.py", PROMPT_SIMPLE_MATH, EVAL_SIMPLE_MATH, BOT_SPEAKS_FIRST),
("07n-interruptible-google.py", PROMPT_SIMPLE_MATH, EVAL_SIMPLE_MATH, BOT_SPEAKS_FIRST),
("07o-interruptible-assemblyai.py", PROMPT_SIMPLE_MATH, EVAL_SIMPLE_MATH, BOT_SPEAKS_FIRST),

View File

@@ -105,8 +105,6 @@ class OpenAILLMAdapter(BaseLLMAdapter[OpenAILLMInvocationParams]):
if item["type"] == "image_url":
if item["image_url"]["url"].startswith("data:image/"):
item["image_url"]["url"] = "data:image/..."
if item["type"] == "input_audio":
item["input_audio"]["data"] = "..."
if "mime_type" in msg and msg["mime_type"].startswith("image/"):
msg["data"] = "..."
msgs.append(msg)

View File

@@ -14,8 +14,6 @@ from abc import ABC, abstractmethod
from enum import Enum
from typing import Optional, Tuple
from pydantic import BaseModel
from pipecat.metrics.metrics import MetricsData
@@ -31,12 +29,6 @@ class EndOfTurnState(Enum):
INCOMPLETE = 2
class BaseTurnParams(BaseModel):
"""Base class for turn analyzer parameters."""
pass
class BaseTurnAnalyzer(ABC):
"""Abstract base class for analyzing user end of turn.
@@ -86,7 +78,7 @@ class BaseTurnAnalyzer(ABC):
@property
@abstractmethod
def params(self) -> BaseTurnParams:
def params(self):
"""Get the current turn analyzer parameters.
Returns:

View File

@@ -11,17 +11,15 @@ machine learning models to determine when a user has finished speaking, going
beyond simple silence-based detection.
"""
import asyncio
import time
from abc import abstractmethod
from concurrent.futures import ThreadPoolExecutor
from typing import Any, Dict, Optional, Tuple
import numpy as np
from loguru import logger
from pydantic import BaseModel
from pipecat.audio.turn.base_turn_analyzer import BaseTurnAnalyzer, BaseTurnParams, EndOfTurnState
from pipecat.audio.turn.base_turn_analyzer import BaseTurnAnalyzer, EndOfTurnState
from pipecat.metrics.metrics import MetricsData, SmartTurnMetricsData
# Default timing parameters
@@ -31,7 +29,7 @@ MAX_DURATION_SECONDS = 8 # Max allowed segment duration
USE_ONLY_LAST_VAD_SEGMENT = True
class SmartTurnParams(BaseTurnParams):
class SmartTurnParams(BaseModel):
"""Configuration parameters for smart turn analysis.
Parameters:
@@ -79,9 +77,6 @@ class BaseSmartTurn(BaseTurnAnalyzer):
self._speech_triggered = False
self._silence_ms = 0
self._speech_start_time = 0
# Thread executor that will run the model. We only need one thread per
# analyzer because one analyzer just handles one audio stream.
self._executor = ThreadPoolExecutor(max_workers=1)
@property
def speech_triggered(self) -> bool:
@@ -156,10 +151,7 @@ class BaseSmartTurn(BaseTurnAnalyzer):
Tuple containing the end-of-turn state and optional metrics data
from the ML model analysis.
"""
loop = asyncio.get_running_loop()
state, result = await loop.run_in_executor(
self._executor, self._process_speech_segment, self._audio_buffer
)
state, result = await self._process_speech_segment(self._audio_buffer)
if state == EndOfTurnState.COMPLETE or USE_ONLY_LAST_VAD_SEGMENT:
self._clear(state)
logger.debug(f"End of Turn result: {state}")
@@ -177,7 +169,9 @@ class BaseSmartTurn(BaseTurnAnalyzer):
self._speech_start_time = 0
self._silence_ms = 0
def _process_speech_segment(self, audio_buffer) -> Tuple[EndOfTurnState, Optional[MetricsData]]:
async def _process_speech_segment(
self, audio_buffer
) -> Tuple[EndOfTurnState, Optional[MetricsData]]:
"""Process accumulated audio segment using ML model."""
state = EndOfTurnState.INCOMPLETE
@@ -209,7 +203,7 @@ class BaseSmartTurn(BaseTurnAnalyzer):
if len(segment_audio) > 0:
start_time = time.perf_counter()
try:
result = self._predict_endpoint(segment_audio)
result = await self._predict_endpoint(segment_audio)
state = (
EndOfTurnState.COMPLETE
if result["prediction"] == 1
@@ -255,6 +249,6 @@ class BaseSmartTurn(BaseTurnAnalyzer):
return state, result_data
@abstractmethod
def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
"""Predict end-of-turn using ML model from audio data."""
pass

View File

@@ -104,15 +104,11 @@ class HttpSmartTurnAnalyzer(BaseSmartTurn):
logger.error(f"Failed to send raw request to Daily Smart Turn: {e}")
raise Exception("Failed to send raw request to Daily Smart Turn.")
def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
"""Predict end-of-turn using remote HTTP ML service."""
try:
serialized_array = self._serialize_array(audio_array)
loop = asyncio.get_running_loop()
future = asyncio.run_coroutine_threadsafe(
self._send_raw_request(serialized_array), loop
)
return future.result()
return await self._send_raw_request(serialized_array)
except Exception as e:
logger.error(f"Smart turn prediction failed: {str(e)}")
# Return an incomplete prediction when a failure occurs

View File

@@ -64,7 +64,7 @@ class LocalSmartTurnAnalyzer(BaseSmartTurn):
self._turn_model.eval()
logger.debug("Loaded Local Smart Turn")
def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
"""Predict end-of-turn using local PyTorch model."""
inputs = self._turn_processor(
audio_array,

View File

@@ -73,7 +73,7 @@ class LocalSmartTurnAnalyzerV2(BaseSmartTurn):
self._turn_model.eval()
logger.debug("Loaded Local Smart Turn v2")
def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
"""Predict end-of-turn using local PyTorch model."""
inputs = self._turn_processor(
audio_array,

View File

@@ -77,7 +77,7 @@ class LocalSmartTurnAnalyzerV3(BaseSmartTurn):
logger.debug("Loaded Local Smart Turn v3")
def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
async def _predict_endpoint(self, audio_array: np.ndarray) -> Dict[str, Any]:
"""Predict end-of-turn using local ONNX model."""
def truncate_audio_to_last_n_seconds(audio_array, n_seconds=8, sample_rate=16000):

View File

@@ -11,9 +11,7 @@ data structures for voice activity detection in audio streams. Includes state
management, parameter configuration, and audio analysis framework.
"""
import asyncio
from abc import ABC, abstractmethod
from concurrent.futures import ThreadPoolExecutor
from enum import Enum
from typing import Optional
@@ -86,10 +84,6 @@ class VADAnalyzer(ABC):
self._smoothing_factor = 0.2
self._prev_volume = 0
# Thread executor that will run the model. We only need one thread per
# analyzer because one analyzer just handles one audio stream.
self._executor = ThreadPoolExecutor(max_workers=1)
@property
def sample_rate(self) -> int:
"""Get the current sample rate.
@@ -171,7 +165,7 @@ class VADAnalyzer(ABC):
volume = calculate_audio_volume(audio, self.sample_rate)
return exp_smoothing(volume, self._prev_volume, self._smoothing_factor)
async def analyze_audio(self, buffer: bytes) -> VADState:
def analyze_audio(self, buffer) -> VADState:
"""Analyze audio buffer and return current VAD state.
Processes incoming audio data, maintains internal state, and determines
@@ -183,12 +177,6 @@ class VADAnalyzer(ABC):
Returns:
Current VAD state after processing the buffer.
"""
loop = asyncio.get_running_loop()
state = await loop.run_in_executor(self._executor, self._run_analyzer, buffer)
return state
def _run_analyzer(self, buffer: bytes) -> VADState:
"""Analyze audio buffer and return current VAD state."""
self._vad_buffer += buffer
num_required_bytes = self._vad_frames_num_bytes

View File

@@ -36,8 +36,7 @@ from pipecat.frames.frames import (
UserStoppedSpeakingFrame,
)
from pipecat.pipeline.parallel_pipeline import ParallelPipeline
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor, FrameProcessorSetup
from pipecat.services.llm_service import LLMService
from pipecat.sync.base_notifier import BaseNotifier
@@ -615,8 +614,8 @@ VOICEMAIL SYSTEM (respond "VOICEMAIL"):
]
# Create the LLM context and aggregators for conversation management
self._context = LLMContext(self._messages)
self._context_aggregator = LLMContextAggregatorPair(self._context)
self._context = OpenAILLMContext(self._messages)
self._context_aggregator = llm.create_context_aggregator(self._context)
# Create notification system for coordinating between components
self._gate_notifier = EventNotifier() # Signals classification completion

View File

@@ -13,7 +13,8 @@ including heartbeats, idle detection, and observer integration.
import asyncio
import time
from typing import Any, AsyncIterable, Dict, Iterable, List, Optional, Tuple, Type
from collections import deque
from typing import Any, AsyncIterable, Deque, Dict, Iterable, List, Optional, Tuple, Type
from loguru import logger
from pydantic import BaseModel, ConfigDict, Field
@@ -30,6 +31,7 @@ from pipecat.frames.frames import (
ErrorFrame,
Frame,
HeartbeatFrame,
InputAudioRawFrame,
InterruptionFrame,
InterruptionTaskFrame,
MetricsFrame,
@@ -393,8 +395,7 @@ class PipelineTask(BasePipelineTask):
Cancels all running tasks and stops frame processing without
waiting for completion.
"""
if not self._finished:
await self._cancel()
await self._cancel()
async def run(self, params: PipelineTaskParams):
"""Start and manage the pipeline execution until completion or cancellation.

View File

@@ -13,7 +13,6 @@ LLM processing, and text-to-speech components in conversational AI pipelines.
import asyncio
import json
from abc import abstractmethod
from typing import Any, Dict, List, Literal, Optional, Set
from loguru import logger
@@ -170,11 +169,6 @@ class LLMContextAggregator(FrameProcessor):
"""Reset the aggregation state."""
self._aggregation = ""
@abstractmethod
async def push_aggregation(self):
"""Push the current aggregation downstream."""
pass
class LLMUserAggregator(LLMContextAggregator):
"""User LLM aggregator that processes speech-to-text transcriptions.
@@ -307,7 +301,7 @@ class LLMUserAggregator(LLMContextAggregator):
frame = LLMContextFrame(self._context)
await self.push_frame(frame)
async def push_aggregation(self):
async def _push_aggregation(self):
"""Push the current aggregation based on interruption strategies and conditions."""
if len(self._aggregation) > 0:
if self.interruption_strategies and self._bot_speaking:
@@ -398,7 +392,7 @@ class LLMUserAggregator(LLMContextAggregator):
# pushing the aggregation as we will probably get a final transcription.
if len(self._aggregation) > 0:
if not self._seen_interim_results:
await self.push_aggregation()
await self._push_aggregation()
# Handles the case where both the user and the bot are not speaking,
# and the bot was previously speaking before the user interruption.
# So in this case we are resetting the aggregation timer
@@ -477,7 +471,7 @@ class LLMUserAggregator(LLMContextAggregator):
await self._maybe_emulate_user_speaking()
except asyncio.TimeoutError:
if not self._user_speaking:
await self.push_aggregation()
await self._push_aggregation()
# If we are emulating VAD we still need to send the user stopped
# speaking frame.
@@ -613,12 +607,12 @@ class LLMAssistantAggregator(LLMContextAggregator):
elif isinstance(frame, UserImageRawFrame) and frame.request and frame.request.tool_call_id:
await self._handle_user_image_frame(frame)
elif isinstance(frame, BotStoppedSpeakingFrame):
await self.push_aggregation()
await self._push_aggregation()
await self.push_frame(frame, direction)
else:
await self.push_frame(frame, direction)
async def push_aggregation(self):
async def _push_aggregation(self):
"""Push the current assistant aggregation with timestamp."""
if not self._aggregation:
return
@@ -650,7 +644,7 @@ class LLMAssistantAggregator(LLMContextAggregator):
await self.push_context_frame(FrameDirection.UPSTREAM)
async def _handle_interruptions(self, frame: InterruptionFrame):
await self.push_aggregation()
await self._push_aggregation()
self._started = 0
await self.reset()
@@ -784,7 +778,7 @@ class LLMAssistantAggregator(LLMContextAggregator):
text=frame.request.context,
)
await self.push_aggregation()
await self._push_aggregation()
await self.push_context_frame(FrameDirection.UPSTREAM)
async def _handle_llm_start(self, _: LLMFullResponseStartFrame):
@@ -792,7 +786,7 @@ class LLMAssistantAggregator(LLMContextAggregator):
async def _handle_llm_end(self, _: LLMFullResponseEndFrame):
self._started -= 1
await self.push_aggregation()
await self._push_aggregation()
async def _handle_text(self, frame: TextFrame):
if not self._started:

View File

@@ -12,14 +12,14 @@ in conversational pipelines.
"""
from pipecat.frames.frames import TextFrame
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response_universal import LLMUserAggregator
from pipecat.processors.aggregators.llm_response import LLMUserContextAggregator
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
class UserResponseAggregator(LLMUserAggregator):
class UserResponseAggregator(LLMUserContextAggregator):
"""Aggregates user responses into TextFrame objects.
This aggregator extends LLMUserAggregator to specifically handle
This aggregator extends LLMUserContextAggregator to specifically handle
user input by collecting text responses and outputting them as TextFrame
objects when the aggregation is complete.
"""
@@ -28,9 +28,9 @@ class UserResponseAggregator(LLMUserAggregator):
"""Initialize the user response aggregator.
Args:
**kwargs: Additional arguments passed to parent LLMUserAggregator.
**kwargs: Additional arguments passed to parent LLMUserContextAggregator.
"""
super().__init__(context=LLMContext(), **kwargs)
super().__init__(context=OpenAILLMContext(), **kwargs)
async def push_aggregation(self):
"""Push the aggregated user response as a TextFrame.

View File

@@ -229,12 +229,9 @@ class AudioBufferProcessor(FrameProcessor):
# Save time of frame so we can compute silence.
self._last_bot_frame_at = time.time()
if self._buffer_size > 0 and (
len(self._user_audio_buffer) >= self._buffer_size
or len(self._bot_audio_buffer) >= self._buffer_size
):
if self._buffer_size > 0 and len(self._user_audio_buffer) > self._buffer_size:
await self._call_on_audio_data_handler()
self._reset_primary_audio_buffers()
self._reset_recording()
# Process turn recording with preprocessed data.
if self._enable_turn_audio:
@@ -275,15 +272,9 @@ class AudioBufferProcessor(FrameProcessor):
async def _call_on_audio_data_handler(self):
"""Call the audio data event handlers with buffered audio."""
if not self._recording:
if not self.has_audio() or not self._recording:
return
if len(self._user_audio_buffer) == 0 and len(self._bot_audio_buffer) == 0:
return
self._align_track_buffers()
flush_time = time.time()
# Call original handler with merged audio
merged_audio = self.merge_audio_buffers()
await self._call_event_handler(
@@ -299,49 +290,23 @@ class AudioBufferProcessor(FrameProcessor):
self._num_channels,
)
self._last_user_frame_at = flush_time
self._last_bot_frame_at = flush_time
def _buffer_has_audio(self, buffer: bytearray) -> bool:
"""Check if a buffer contains audio data."""
return buffer is not None and len(buffer) > 0
def _reset_recording(self):
"""Reset recording state and buffers."""
self._reset_all_audio_buffers()
self._reset_audio_buffers()
self._last_user_frame_at = time.time()
self._last_bot_frame_at = time.time()
def _reset_all_audio_buffers(self):
def _reset_audio_buffers(self):
"""Reset all audio buffers to empty state."""
self._reset_primary_audio_buffers()
self._reset_turn_audio_buffers()
def _reset_primary_audio_buffers(self):
"""Clear user and bot buffers while preserving turn buffers and timestamps."""
self._user_audio_buffer = bytearray()
self._bot_audio_buffer = bytearray()
def _reset_turn_audio_buffers(self):
"""Clear user and bot turn buffers while preserving primary buffers and timestamps."""
self._user_turn_audio_buffer = bytearray()
self._bot_turn_audio_buffer = bytearray()
def _align_track_buffers(self):
"""Pad the shorter track with silence so both tracks stay in sync."""
user_len = len(self._user_audio_buffer)
bot_len = len(self._bot_audio_buffer)
if user_len == bot_len:
return
target_len = max(user_len, bot_len)
if user_len < target_len:
self._user_audio_buffer.extend(b"\x00" * (target_len - user_len))
self._last_user_frame_at = max(self._last_user_frame_at, self._last_bot_frame_at)
if bot_len < target_len:
self._bot_audio_buffer.extend(b"\x00" * (target_len - bot_len))
self._last_bot_frame_at = max(self._last_bot_frame_at, self._last_user_frame_at)
async def _resample_input_audio(self, frame: InputAudioRawFrame) -> bytes:
"""Resample audio frame to the target sample rate."""
return await self._input_resampler.resample(

View File

@@ -455,13 +455,9 @@ class FrameProcessor(BaseObject):
name = f"{self}::{coroutine.cr_code.co_name}"
return self.task_manager.create_task(coroutine, name)
async def cancel_task(self, task: asyncio.Task, timeout: Optional[float] = 1.0):
async def cancel_task(self, task: asyncio.Task, timeout: Optional[float] = None):
"""Cancel a task managed by this processor.
A default timeout if 1 second is used in order to avoid potential
freezes caused by certain libraries that swallow
`asyncio.CancelledError`.
Args:
task: The task to cancel.
timeout: Optional timeout for task cancellation.

View File

@@ -13,7 +13,6 @@ and frame observation for the RTVI protocol.
import asyncio
import base64
import time
from dataclasses import dataclass
from typing import (
Any,
@@ -30,7 +29,6 @@ from typing import (
from loguru import logger
from pydantic import BaseModel, Field, PrivateAttr, ValidationError
from pipecat.audio.utils import calculate_audio_volume
from pipecat.frames.frames import (
BotStartedSpeakingFrame,
BotStoppedSpeakingFrame,
@@ -54,7 +52,6 @@ from pipecat.frames.frames import (
SystemFrame,
TranscriptionFrame,
TransportMessageUrgentFrame,
TTSAudioRawFrame,
TTSStartedFrame,
TTSStoppedFrame,
TTSTextFrame,
@@ -616,9 +613,9 @@ class RTVIAppendToContextData(BaseModel):
Contains the role, content, and whether to run the message immediately.
.. deprecated:: 0.0.85
The RTVI message, append-to-context, has been deprecated. Use send-text
or custom client and server messages instead.
.. deprecated:: 0.0.85
The RTVI message, append-to-context, has been deprecated. Use send-text
or custom client and server messages instead.
"""
role: Literal["user", "assistant"] | str
@@ -842,36 +839,6 @@ class RTVIServerMessage(BaseModel):
data: Any
class RTVIAudioLevelMessageData(BaseModel):
"""Data format for sending audio levels."""
value: float
class RTVIUserAudioLevelMessage(BaseModel):
"""Message indicating user audio level."""
label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
type: Literal["user-audio-level"] = "user-audio-level"
data: RTVIAudioLevelMessageData
class RTVIBotAudioLevelMessage(BaseModel):
"""Message indicating bot audio level."""
label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
type: Literal["bot-audio-level"] = "bot-audio-level"
data: RTVIAudioLevelMessageData
class RTVISystemLogMessage(BaseModel):
"""Message including a system log."""
label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
type: Literal["system-log"] = "system-log"
data: RTVITextMessageData
@dataclass
class RTVIServerMessageFrame(SystemFrame):
"""A frame for sending server messages to the client.
@@ -891,36 +858,25 @@ class RTVIServerMessageFrame(SystemFrame):
class RTVIObserverParams:
"""Parameters for configuring RTVI Observer behavior.
.. deprecated:: 0.0.87
Parameter `errors_enabled` is deprecated. Error messages are always enabled.
Parameters:
bot_llm_enabled: Indicates if the bot's LLM messages should be sent.
bot_tts_enabled: Indicates if the bot's TTS messages should be sent.
bot_speaking_enabled: Indicates if the bot's started/stopped speaking messages should be sent.
bot_audio_level_enabled: Indicates if bot's audio level messages should be sent.
user_llm_enabled: Indicates if the user's LLM input messages should be sent.
user_speaking_enabled: Indicates if the user's started/stopped speaking messages should be sent.
user_transcription_enabled: Indicates if user's transcription messages should be sent.
user_audio_level_enabled: Indicates if user's audio level messages should be sent.
metrics_enabled: Indicates if metrics messages should be sent.
system_logs_enabled: Indicates if system logs should be sent.
errors_enabled: [Deprecated] Indicates if errors messages should be sent.
audio_level_period_secs: How often audio levels should be sent if enabled.
errors_enabled: Indicates if errors messages should be sent.
"""
bot_llm_enabled: bool = True
bot_tts_enabled: bool = True
bot_speaking_enabled: bool = True
bot_audio_level_enabled: bool = False
user_llm_enabled: bool = True
user_speaking_enabled: bool = True
user_transcription_enabled: bool = True
user_audio_level_enabled: bool = False
metrics_enabled: bool = True
system_logs_enabled: bool = False
errors_enabled: bool = True
audio_level_period_secs: float = 0.15
class RTVIObserver(BaseObserver):
@@ -936,11 +892,7 @@ class RTVIObserver(BaseObserver):
"""
def __init__(
self,
rtvi: Optional["RTVIProcessor"] = None,
*,
params: Optional[RTVIObserverParams] = None,
**kwargs,
self, rtvi: "RTVIProcessor", *, params: Optional[RTVIObserverParams] = None, **kwargs
):
"""Initialize the RTVI observer.
@@ -952,50 +904,9 @@ class RTVIObserver(BaseObserver):
super().__init__(**kwargs)
self._rtvi = rtvi
self._params = params or RTVIObserverParams()
self._frames_seen = set()
self._bot_transcription = ""
self._last_user_audio_level = 0
self._last_bot_audio_level = 0
if self._params.system_logs_enabled:
self._system_logger_id = logger.add(self._logger_sink)
if self._params.errors_enabled:
import warnings
with warnings.catch_warnings():
warnings.simplefilter("always")
warnings.warn(
"Parameter `errors_enabled` is deprecated. Error messages are always enabled.",
DeprecationWarning,
)
async def _logger_sink(self, message):
"""Logger sink so we cna send system logs to RTVI clients."""
message = RTVISystemLogMessage(data=RTVITextMessageData(text=message))
await self.send_rtvi_message(message)
async def cleanup(self):
"""Cleanup RTVI observer resources."""
await super().cleanup()
if self._params.system_logs_enabled:
logger.remove(self._system_logger_id)
async def send_rtvi_message(self, model: BaseModel, exclude_none: bool = True):
"""Send an RTVI message.
By default, we push a transport frame. But this function can be
overriden by subclass to send RTVI messages in different ways.
Args:
model: The message to send.
exclude_none: Whether to exclude None values from the model dump.
"""
if self._rtvi:
await self._rtvi.push_transport_message(model, exclude_none)
self._frames_seen = set()
rtvi.set_errors_enabled(self._params.errors_enabled)
async def on_push_frame(self, data: FramePushed):
"""Process a frame being pushed through the pipeline.
@@ -1037,58 +948,52 @@ class RTVIObserver(BaseObserver):
):
await self._handle_context(frame)
elif isinstance(frame, LLMFullResponseStartFrame) and self._params.bot_llm_enabled:
await self.send_rtvi_message(RTVIBotLLMStartedMessage())
await self.push_transport_message_urgent(RTVIBotLLMStartedMessage())
elif isinstance(frame, LLMFullResponseEndFrame) and self._params.bot_llm_enabled:
await self.send_rtvi_message(RTVIBotLLMStoppedMessage())
await self.push_transport_message_urgent(RTVIBotLLMStoppedMessage())
elif isinstance(frame, LLMTextFrame) and self._params.bot_llm_enabled:
await self._handle_llm_text_frame(frame)
elif isinstance(frame, TTSStartedFrame) and self._params.bot_tts_enabled:
await self.send_rtvi_message(RTVIBotTTSStartedMessage())
await self.push_transport_message_urgent(RTVIBotTTSStartedMessage())
elif isinstance(frame, TTSStoppedFrame) and self._params.bot_tts_enabled:
await self.send_rtvi_message(RTVIBotTTSStoppedMessage())
await self.push_transport_message_urgent(RTVIBotTTSStoppedMessage())
elif isinstance(frame, TTSTextFrame) and self._params.bot_tts_enabled:
if isinstance(src, BaseOutputTransport):
message = RTVIBotTTSTextMessage(data=RTVITextMessageData(text=frame.text))
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
else:
mark_as_seen = False
elif isinstance(frame, MetricsFrame) and self._params.metrics_enabled:
await self._handle_metrics(frame)
elif isinstance(frame, RTVIServerMessageFrame):
message = RTVIServerMessage(data=frame.data)
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
elif isinstance(frame, RTVIServerResponseFrame):
if frame.error is not None:
await self._send_error_response(frame)
else:
await self._send_server_response(frame)
elif isinstance(frame, InputAudioRawFrame) and self._params.user_audio_level_enabled:
curr_time = time.time()
diff_time = curr_time - self._last_user_audio_level
if diff_time > self._params.audio_level_period_secs:
level = calculate_audio_volume(frame.audio, frame.sample_rate)
message = RTVIUserAudioLevelMessage(data=RTVIAudioLevelMessageData(value=level))
await self.send_rtvi_message(message)
self._last_user_audio_level = curr_time
elif isinstance(frame, TTSAudioRawFrame) and self._params.bot_audio_level_enabled:
curr_time = time.time()
diff_time = curr_time - self._last_bot_audio_level
if diff_time > self._params.audio_level_period_secs:
level = calculate_audio_volume(frame.audio, frame.sample_rate)
message = RTVIBotAudioLevelMessage(data=RTVIAudioLevelMessageData(value=level))
await self.send_rtvi_message(message)
self._last_bot_audio_level = curr_time
if mark_as_seen:
self._frames_seen.add(frame.id)
async def push_transport_message_urgent(self, model: BaseModel, exclude_none: bool = True):
"""Push an urgent transport message to the RTVI processor.
Args:
model: The message model to send.
exclude_none: Whether to exclude None values from the model dump.
"""
frame = TransportMessageUrgentFrame(message=model.model_dump(exclude_none=exclude_none))
await self._rtvi.push_frame(frame)
async def _push_bot_transcription(self):
"""Push accumulated bot transcription as a message."""
if len(self._bot_transcription) > 0:
message = RTVIBotTranscriptionMessage(
data=RTVITextMessageData(text=self._bot_transcription)
)
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
self._bot_transcription = ""
async def _handle_interruptions(self, frame: Frame):
@@ -1100,7 +1005,7 @@ class RTVIObserver(BaseObserver):
message = RTVIUserStoppedSpeakingMessage()
if message:
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
async def _handle_bot_speaking(self, frame: Frame):
"""Handle bot speaking event frames."""
@@ -1111,12 +1016,12 @@ class RTVIObserver(BaseObserver):
message = RTVIBotStoppedSpeakingMessage()
if message:
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
async def _handle_llm_text_frame(self, frame: LLMTextFrame):
"""Handle LLM text output frames."""
message = RTVIBotLLMTextMessage(data=RTVITextMessageData(text=frame.text))
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
self._bot_transcription += frame.text
if match_endofsentence(self._bot_transcription):
@@ -1139,7 +1044,7 @@ class RTVIObserver(BaseObserver):
)
if message:
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
async def _handle_context(self, frame: OpenAILLMContextFrame | LLMContextFrame):
"""Process LLM context frames to extract user messages for the RTVI client."""
@@ -1159,7 +1064,7 @@ class RTVIObserver(BaseObserver):
text = "".join(part.text for part in message.parts if hasattr(part, "text"))
if text:
rtvi_message = RTVIUserLLMTextMessage(data=RTVITextMessageData(text=text))
await self.send_rtvi_message(rtvi_message)
await self.push_transport_message_urgent(rtvi_message)
# Handle OpenAI format (original implementation)
elif isinstance(message, dict):
@@ -1170,7 +1075,7 @@ class RTVIObserver(BaseObserver):
else:
text = content
rtvi_message = RTVIUserLLMTextMessage(data=RTVITextMessageData(text=text))
await self.send_rtvi_message(rtvi_message)
await self.push_transport_message_urgent(rtvi_message)
except Exception as e:
logger.warning(f"Caught an error while trying to handle context: {e}")
@@ -1197,7 +1102,7 @@ class RTVIObserver(BaseObserver):
metrics["characters"].append(d.model_dump(exclude_none=True))
message = RTVIMetricsMessage(data=metrics)
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
async def _send_server_response(self, frame: RTVIServerResponseFrame):
"""Send a response to the client for a specific request."""
@@ -1205,7 +1110,7 @@ class RTVIObserver(BaseObserver):
id=str(frame.client_msg.msg_id),
data=RTVIRawServerResponseData(t=frame.client_msg.type, d=frame.data),
)
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
async def _send_error_response(self, frame: RTVIServerResponseFrame):
"""Send a response to the client for a specific request."""
@@ -1213,7 +1118,7 @@ class RTVIObserver(BaseObserver):
message = RTVIErrorResponse(
id=str(frame.client_msg.msg_id), data=RTVIErrorResponseData(error=frame.error)
)
await self.send_rtvi_message(message)
await self.push_transport_message_urgent(message)
class RTVIProcessor(FrameProcessor):
@@ -1247,6 +1152,7 @@ class RTVIProcessor(FrameProcessor):
# Default to 0.3.0 which is the last version before actually having a
# "client-version".
self._client_version = [0, 3, 0]
self._errors_enabled = True
self._skip_tts: bool = False # Keep in sync with llm_service.py
self._registered_actions: Dict[str, RTVIAction] = {}
@@ -1316,6 +1222,14 @@ class RTVIProcessor(FrameProcessor):
await self._update_config(self._config, False)
await self._send_bot_ready()
def set_errors_enabled(self, enabled: bool):
"""Enable or disable error message sending.
Args:
enabled: Whether to send error messages.
"""
self._errors_enabled = enabled
async def interrupt_bot(self):
"""Send a bot interruption frame upstream."""
await self.push_interruption_task_frame_and_wait()
@@ -1344,11 +1258,6 @@ class RTVIProcessor(FrameProcessor):
"""
await self._send_error_frame(ErrorFrame(error=error))
async def push_transport_message(self, model: BaseModel, exclude_none: bool = True):
"""Push a transport message frame."""
frame = TransportMessageUrgentFrame(message=model.model_dump(exclude_none=exclude_none))
await self.push_frame(frame)
async def handle_message(self, message: RTVIMessage):
"""Handle an incoming RTVI message.
@@ -1369,7 +1278,7 @@ class RTVIProcessor(FrameProcessor):
args=params.arguments,
)
message = RTVILLMFunctionCallMessage(data=fn)
await self.push_transport_message(message, exclude_none=False)
await self._push_transport_message(message, exclude_none=False)
async def handle_function_call_start(
self, function_name: str, llm: FrameProcessor, context: OpenAILLMContext
@@ -1396,7 +1305,7 @@ class RTVIProcessor(FrameProcessor):
fn = RTVILLMFunctionCallStartMessageData(function_name=function_name)
message = RTVILLMFunctionCallStartMessage(data=fn)
await self.push_transport_message(message, exclude_none=False)
await self._push_transport_message(message, exclude_none=False)
async def process_frame(self, frame: Frame, direction: FrameDirection):
"""Process incoming frames through the RTVI processor.
@@ -1468,6 +1377,11 @@ class RTVIProcessor(FrameProcessor):
await self.cancel_task(self._message_task)
self._message_task = None
async def _push_transport_message(self, model: BaseModel, exclude_none: bool = True):
"""Push a transport message frame."""
frame = TransportMessageUrgentFrame(message=model.model_dump(exclude_none=exclude_none))
await self.push_frame(frame)
async def _action_task_handler(self):
"""Handle incoming action frames."""
while True:
@@ -1604,7 +1518,7 @@ class RTVIProcessor(FrameProcessor):
services = list(self._registered_services.values())
message = RTVIDescribeConfig(id=request_id, data=RTVIDescribeConfigData(config=services))
await self.push_transport_message(message)
await self._push_transport_message(message)
async def _handle_describe_actions(self, request_id: str):
"""Handle a describe-actions request."""
@@ -1619,7 +1533,7 @@ class RTVIProcessor(FrameProcessor):
actions = list(self._registered_actions.values())
message = RTVIDescribeActions(id=request_id, data=RTVIDescribeActionsData(actions=actions))
await self.push_transport_message(message)
await self._push_transport_message(message)
async def _handle_get_config(self, request_id: str):
"""Handle a get-config request."""
@@ -1633,7 +1547,7 @@ class RTVIProcessor(FrameProcessor):
)
message = RTVIConfigResponse(id=request_id, data=self._config)
await self.push_transport_message(message)
await self._push_transport_message(message)
def _update_config_option(self, service: str, config: RTVIServiceOptionConfig):
"""Update a specific configuration option."""
@@ -1758,7 +1672,7 @@ class RTVIProcessor(FrameProcessor):
# action responses (such as webhooks) don't set a request_id
if request_id:
message = RTVIActionResponse(id=request_id, data=RTVIActionResponseData(result=result))
await self.push_transport_message(message)
await self._push_transport_message(message)
async def _send_bot_ready(self):
"""Send the bot-ready message to the client."""
@@ -1769,21 +1683,23 @@ class RTVIProcessor(FrameProcessor):
id=self._client_ready_id,
data=RTVIBotReadyData(version=RTVI_PROTOCOL_VERSION, config=config),
)
await self.push_transport_message(message)
await self._push_transport_message(message)
async def _send_server_message(self, message: RTVIServerMessage | RTVIServerResponse):
"""Send a message or response to the client."""
await self.push_transport_message(message)
await self._push_transport_message(message)
async def _send_error_frame(self, frame: ErrorFrame):
"""Send an error frame as an RTVI error message."""
message = RTVIError(data=RTVIErrorData(error=frame.error, fatal=frame.fatal))
await self.push_transport_message(message)
if self._errors_enabled:
message = RTVIError(data=RTVIErrorData(error=frame.error, fatal=frame.fatal))
await self._push_transport_message(message)
async def _send_error_response(self, id: str, error: str):
"""Send an error response message."""
message = RTVIErrorResponse(id=id, data=RTVIErrorResponseData(error=error))
await self.push_transport_message(message)
if self._errors_enabled:
message = RTVIErrorResponse(id=id, data=RTVIErrorResponseData(error=error))
await self._push_transport_message(message)
def _action_id(self, service: str, action: str) -> str:
"""Generate an action ID from service and action names."""

View File

@@ -10,7 +10,6 @@ from loguru import logger
from pipecat.frames.frames import (
Frame,
LLMContextFrame,
LLMFullResponseEndFrame,
LLMFullResponseStartFrame,
LLMTextFrame,
@@ -72,11 +71,9 @@ class StrandsAgentsProcessor(FrameProcessor):
direction: The direction of frame flow in the pipeline.
"""
await super().process_frame(frame, direction)
if isinstance(frame, (LLMContextFrame, OpenAILLMContextFrame)):
messages = frame.context.get_messages()
if messages:
last_message = messages[-1]
await self._ainvoke(str(last_message["content"]).strip())
if isinstance(frame, OpenAILLMContextFrame):
text = frame.context.messages[-1]["content"]
await self._ainvoke(str(text).strip())
else:
await self.push_frame(frame, direction)

View File

@@ -61,8 +61,6 @@ class TwilioFrameSerializer(FrameSerializer):
call_sid: Optional[str] = None,
account_sid: Optional[str] = None,
auth_token: Optional[str] = None,
region: Optional[str] = None,
edge: Optional[str] = None,
params: Optional[InputParams] = None,
):
"""Initialize the TwilioFrameSerializer.
@@ -72,42 +70,13 @@ class TwilioFrameSerializer(FrameSerializer):
call_sid: The associated Twilio Call SID (optional, but required for auto hang-up).
account_sid: Twilio account SID (required for auto hang-up).
auth_token: Twilio auth token (required for auto hang-up).
region: Twilio region (e.g., "au1", "ie1"). Must be specified with edge.
edge: Twilio edge location (e.g., "sydney", "dublin"). Must be specified with region.
params: Configuration parameters.
"""
self._params = params or TwilioFrameSerializer.InputParams()
# Validate hangup-related parameters if auto_hang_up is enabled
if self._params.auto_hang_up:
# Validate required credentials
missing_credentials = []
if not call_sid:
missing_credentials.append("call_sid")
if not account_sid:
missing_credentials.append("account_sid")
if not auth_token:
missing_credentials.append("auth_token")
if missing_credentials:
raise ValueError(
f"auto_hang_up is enabled but missing required parameters: {', '.join(missing_credentials)}"
)
# Validate region and edge are both provided if either is specified
if (region and not edge) or (edge and not region):
raise ValueError(
"Both edge and region parameters are required if one is set. "
f"Twilio's FQDN format requires both: api.{{edge}}.{{region}}.twilio.com. "
f"Got: region='{region}', edge='{edge}'"
)
self._stream_sid = stream_sid
self._call_sid = call_sid
self._account_sid = account_sid
self._auth_token = auth_token
self._region = region
self._edge = edge
self._params = params or TwilioFrameSerializer.InputParams()
self._twilio_sample_rate = self._params.twilio_sample_rate
self._sample_rate = 0 # Pipeline input rate
@@ -189,14 +158,25 @@ class TwilioFrameSerializer(FrameSerializer):
account_sid = self._account_sid
auth_token = self._auth_token
call_sid = self._call_sid
region = self._region
edge = self._edge
region_prefix = f"{region}." if region else ""
edge_prefix = f"{edge}." if edge else ""
if not call_sid or not account_sid or not auth_token:
missing = []
if not call_sid:
missing.append("call_sid")
if not account_sid:
missing.append("account_sid")
if not auth_token:
missing.append("auth_token")
logger.warning(
f"Cannot hang up Twilio call: missing required parameters: {', '.join(missing)}"
)
return
# Twilio API endpoint for updating calls
endpoint = f"https://api.{edge_prefix}{region_prefix}twilio.com/2010-04-01/Accounts/{account_sid}/Calls/{call_sid}.json"
endpoint = (
f"https://api.twilio.com/2010-04-01/Accounts/{account_sid}/Calls/{call_sid}.json"
)
# Create basic auth from account_sid and auth_token
auth = aiohttp.BasicAuth(account_sid, auth_token)

View File

@@ -151,7 +151,7 @@ class AnthropicLLMService(LLMService):
self,
*,
api_key: str,
model: str = "claude-sonnet-4-5-20250929",
model: str = "claude-sonnet-4-20250514",
params: Optional[InputParams] = None,
client=None,
retry_timeout_secs: Optional[float] = 5.0,
@@ -162,7 +162,7 @@ class AnthropicLLMService(LLMService):
Args:
api_key: Anthropic API key for authentication.
model: Model name to use. Defaults to "claude-sonnet-4-5-20250929".
model: Model name to use. Defaults to "claude-sonnet-4-20250514".
params: Optional model parameters for inference.
client: Optional custom Anthropic client instance.
retry_timeout_secs: Request timeout in seconds for retry logic.

View File

@@ -429,7 +429,7 @@ class AWSNovaSonicLLMService(LLMService):
await self._finish_connecting_if_context_available()
except Exception as e:
logger.error(f"{self} initialization error: {e}")
await self._disconnect()
self._disconnect()
async def _finish_connecting_if_context_available(self):
# We can only finish connecting once we've gotten our initial context and we're ready to

View File

@@ -108,14 +108,12 @@ class HeyGenSession(BaseModel):
Parameters:
session_id (str): Unique identifier for the streaming session.
access_token (str): Token for accessing the session securely.
livekit_agent_token (str): Token for HeyGens audio agents(Pipecat).
realtime_endpoint (str): Real-time communication endpoint URL.
url (str): Direct URL for the session.
"""
session_id: str
access_token: str
livekit_agent_token: str
realtime_endpoint: str
url: str

View File

@@ -393,9 +393,7 @@ class HeyGenClient:
participant_id: Identifier of the participant to capture audio from
callback: Async function to handle received audio frames
"""
logger.debug(
f"capture_participant_audio: {participant_id}, sample_rate: {self._in_sample_rate}"
)
logger.debug(f"capture_participant_audio: {participant_id}")
self._audio_frame_callback = callback
if self._audio_task is not None:
logger.warning(
@@ -409,9 +407,7 @@ class HeyGenClient:
for track_pub in participant.track_publications.values():
if track_pub.kind == rtc.TrackKind.KIND_AUDIO and track_pub.track is not None:
logger.debug(f"Starting audio capture for existing track: {track_pub.sid}")
audio_stream = rtc.AudioStream(
track=track_pub.track, sample_rate=self._in_sample_rate
)
audio_stream = rtc.AudioStream(track_pub.track)
self._audio_task = self._task_manager.create_task(
self._process_audio_frames(audio_stream), name="HeyGenClient_Receive_Audio"
)
@@ -540,7 +536,7 @@ class HeyGenClient:
and self._audio_task is None
):
logger.debug(f"Creating audio stream processor for track: {publication.sid}")
audio_stream = rtc.AudioStream(track=track, sample_rate=self._in_sample_rate)
audio_stream = rtc.AudioStream(track)
self._audio_task = self._task_manager.create_task(
self._process_audio_frames(audio_stream), name="HeyGenClient_Receive_Audio"
)
@@ -563,7 +559,7 @@ class HeyGenClient:
)
await self._livekit_room.connect(
self._heyGen_session.url, self._heyGen_session.livekit_agent_token
self._heyGen_session.url, self._heyGen_session.access_token
)
logger.debug(f"Successfully connected to LiveKit room: {self._livekit_room.name}")
logger.debug(f"Local participant SID: {self._livekit_room.local_participant.sid}")

View File

@@ -110,7 +110,6 @@ class HeyGenVideoService(AIService):
api_key=self._api_key,
session=self._session,
params=TransportParams(
audio_in_sample_rate=48000,
audio_in_enabled=True,
video_in_enabled=True,
audio_out_enabled=True,

View File

@@ -41,6 +41,7 @@ from pipecat.processors.aggregators.openai_llm_context import (
)
from pipecat.processors.frame_processor import FrameDirection
from pipecat.services.llm_service import FunctionCallFromLLM, LLMService
from pipecat.utils.asyncio.cancellable_stream import CancellableStream
from pipecat.utils.tracing.service_decorators import traced_llm
@@ -138,6 +139,7 @@ class BaseOpenAILLMService(LLMService):
default_headers=default_headers,
**kwargs,
)
self._stream: Optional[CancellableStream[AsyncStream[AsyncStream]]] = None
def create_client(
self,
@@ -329,24 +331,20 @@ class BaseOpenAILLMService(LLMService):
await self.start_ttfb_metrics()
# Generate chat completions using either OpenAILLMContext or universal LLMContext
chunk_stream = await (
self._stream_chat_completions_specific_context(context)
if isinstance(context, OpenAILLMContext)
else self._stream_chat_completions_universal_context(context)
self._stream = CancellableStream(
await (
self._stream_chat_completions_specific_context(context)
if isinstance(context, OpenAILLMContext)
else self._stream_chat_completions_universal_context(context)
)
)
async for chunk in chunk_stream:
async for chunk in self._stream:
if chunk.usage:
cached_tokens = (
chunk.usage.prompt_tokens_details.cached_tokens
if chunk.usage.prompt_tokens_details
else None
)
tokens = LLMTokenUsage(
prompt_tokens=chunk.usage.prompt_tokens,
completion_tokens=chunk.usage.completion_tokens,
total_tokens=chunk.usage.total_tokens,
cache_read_input_tokens=cached_tokens,
)
await self.start_llm_usage_metrics(tokens)
@@ -395,6 +393,8 @@ class BaseOpenAILLMService(LLMService):
):
await self.push_frame(LLMTextFrame(chunk.choices[0].delta.audio["transcript"]))
self._stream = None
# if we got a function name and arguments, check to see if it's a function with
# a registered handler. If so, run the registered callback, save the result to
# the context, and re-prompt to get a chat answer. If we don't have a registered
@@ -422,6 +422,22 @@ class BaseOpenAILLMService(LLMService):
await self.run_function_calls(function_calls)
async def _start_interruption(self):
"""Start handling an interruption by cancelling current tasks."""
# OpenAI's client swallows asyncio.CancelledError internally, which prevents proper
# task cancellation propagation. To ensure proper cancellation behavior:
# 1. We check if there's an active chunk stream when receiving an interruption
# 2. We explicitly cancel the chunk stream first
# 3. This allows the task to be cancelled cleanly afterwards
# This approach ensures we don't get stuck in case there was a chunk processing loop
# when cancellation is requested.
if self._stream:
logger.debug(f"{self}: Cancelling chunk stream due to interruption")
await self._stream.cancel()
self._stream = None
await super()._start_interruption()
async def process_frame(self, frame: Frame, direction: FrameDirection):
"""Process frames for LLM completion requests.

View File

@@ -11,6 +11,7 @@ input processing, including VAD, turn analysis, and interruption management.
"""
import asyncio
from concurrent.futures import ThreadPoolExecutor
from typing import Optional
from loguru import logger
@@ -78,6 +79,10 @@ class BaseInputTransport(FrameProcessor):
# Track user speaking state for interruption logic
self._user_speaking = False
# We read audio from a single queue one at a time and we then run VAD in
# a thread. Therefore, only one thread should be necessary.
self._executor = ThreadPoolExecutor(max_workers=1)
# Task to process incoming audio (VAD) and push audio frames downstream
# if passthrough is enabled.
self._audio_task = None
@@ -393,7 +398,9 @@ class BaseInputTransport(FrameProcessor):
"""Analyze audio frame for voice activity."""
state = VADState.QUIET
if self.vad_analyzer:
state = await self.vad_analyzer.analyze_audio(audio_frame.audio)
state = await self.get_event_loop().run_in_executor(
self._executor, self.vad_analyzer.analyze_audio, audio_frame.audio
)
return state
async def _handle_vad(self, audio_frame: InputAudioRawFrame, vad_state: VADState) -> VADState:

View File

@@ -110,32 +110,12 @@ class DailyInputTransportMessageUrgentFrame(InputTransportMessageUrgentFrame):
class DailyUpdateRemoteParticipantsFrame(ControlFrame):
"""Frame to update remote participants in Daily calls.
.. deprecated:: 0.0.87
`DailyUpdateRemoteParticipantsFrame` is deprecated and will be removed in a future version.
Create your own custom frame and use a custom processor to handle it or use, for example,
`on_after_push_frame` event instead in the output transport.
Parameters:
remote_participants: See https://reference-python.daily.co/api_reference.html#daily.CallClient.update_remote_participants.
"""
remote_participants: Mapping[str, Any] = None
def __post_init__(self):
super().__post_init__()
import warnings
with warnings.catch_warnings():
warnings.simplefilter("always")
warnings.warn(
"DailyUpdateRemoteParticipantsFrame is deprecated and will be removed in a future version."
"Instead, create your own custom frame and handle it in the "
'`@transport.output().event_handler("on_after_push_frame")` event handler or a '
"custom processor.",
DeprecationWarning,
stacklevel=2,
)
class WebRTCVADAnalyzer(VADAnalyzer):
"""Voice Activity Detection analyzer using WebRTC.

View File

@@ -1,381 +0,0 @@
#
# Copyright (c) 20242025, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
"""HeyGen implementation for Pipecat.
This module provides integration with the HeyGen platform for creating conversational
AI applications with avatars. It manages conversation sessions and provides real-time
audio/video streaming capabilities through the HeyGen API.
The module consists of three main components:
- HeyGenInputTransport: Handles incoming audio and events from HeyGen conversations
- HeyGenOutputTransport: Manages outgoing audio and events to HeyGen conversations
- HeyGenTransport: Main transport implementation that coordinates input/output transports
"""
from typing import Any, Optional
import aiohttp
from loguru import logger
from pipecat.frames.frames import (
AudioRawFrame,
BotStartedSpeakingFrame,
BotStoppedSpeakingFrame,
CancelFrame,
EndFrame,
Frame,
InputAudioRawFrame,
InterruptionFrame,
OutputAudioRawFrame,
StartFrame,
UserStartedSpeakingFrame,
UserStoppedSpeakingFrame,
)
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor, FrameProcessorSetup
from pipecat.services.heygen.api import NewSessionRequest
from pipecat.services.heygen.client import HeyGenCallbacks, HeyGenClient
from pipecat.transports.base_input import BaseInputTransport
from pipecat.transports.base_output import BaseOutputTransport
from pipecat.transports.base_transport import BaseTransport, TransportParams
class HeyGenInputTransport(BaseInputTransport):
"""Input transport for receiving audio and events from HeyGen conversations.
Handles incoming audio streams from participants and manages audio capture
from the Daily room connected to the HeyGen conversation.
"""
def __init__(
self,
client: HeyGenClient,
params: TransportParams,
**kwargs,
):
"""Initialize the HeyGen input transport.
Args:
client: The HeyGen transport client instance.
params: Transport configuration parameters.
**kwargs: Additional arguments passed to parent class.
"""
super().__init__(params, **kwargs)
self._client = client
self._params = params
# Whether we have seen a StartFrame already.
self._initialized = False
async def setup(self, setup: FrameProcessorSetup):
"""Setup the input transport.
Args:
setup: The frame processor setup configuration.
"""
await super().setup(setup)
await self._client.setup(setup)
async def cleanup(self):
"""Cleanup input transport resources."""
await super().cleanup()
await self._client.cleanup()
async def start(self, frame: StartFrame):
"""Start the input transport.
Args:
frame: The start frame containing initialization parameters.
"""
await super().start(frame)
if self._initialized:
return
self._initialized = True
await self.set_transport_ready(frame)
async def stop(self, frame: EndFrame):
"""Stop the input transport.
Args:
frame: The end frame signaling transport shutdown.
"""
await super().stop(frame)
await self._client.stop()
async def cancel(self, frame: CancelFrame):
"""Cancel the input transport.
Args:
frame: The cancel frame signaling immediate cancellation.
"""
await super().cancel(frame)
await self._client.stop()
async def start_capturing_audio(self, participant_id: str):
"""Start capturing audio from a participant.
Args:
participant_id: The participant to capture audio from.
"""
if self._params.audio_in_enabled:
logger.info(f"HeyGenTransport start capturing audio for participant {participant_id}")
await self._client.capture_participant_audio(
participant_id, self._on_participant_audio_data
)
async def _on_participant_audio_data(self, audio_frame: AudioRawFrame):
"""Handle received participant audio data."""
frame = InputAudioRawFrame(
audio=audio_frame.audio,
sample_rate=audio_frame.sample_rate,
num_channels=audio_frame.num_channels,
)
await self.push_audio_frame(frame)
class HeyGenOutputTransport(BaseOutputTransport):
"""Output transport for sending audio and events to HeyGen conversations.
Handles outgoing audio streams to participants and manages the custom
audio track expected by the HeyGen platform.
"""
def __init__(
self,
client: HeyGenClient,
params: TransportParams,
**kwargs,
):
"""Initialize the HeyGen output transport.
Args:
client: The HeyGen transport client instance.
params: Transport configuration parameters.
**kwargs: Additional arguments passed to parent class.
"""
super().__init__(params, **kwargs)
self._client = client
self._params = params
# Whether we have seen a StartFrame already.
self._initialized = False
self._event_id = None
async def setup(self, setup: FrameProcessorSetup):
"""Setup the output transport.
Args:
setup: The frame processor setup configuration.
"""
await super().setup(setup)
await self._client.setup(setup)
async def cleanup(self):
"""Cleanup output transport resources."""
await super().cleanup()
await self._client.cleanup()
async def start(self, frame: StartFrame):
"""Start the output transport.
Args:
frame: The start frame containing initialization parameters.
"""
await super().start(frame)
if self._initialized:
return
self._initialized = True
await self._client.start(frame, self.audio_chunk_size)
await self.set_transport_ready(frame)
self._client.transport_ready()
async def stop(self, frame: EndFrame):
"""Stop the output transport.
Args:
frame: The end frame signaling transport shutdown.
"""
await super().stop(frame)
await self._client.stop()
async def cancel(self, frame: CancelFrame):
"""Cancel the output transport.
Args:
frame: The cancel frame signaling immediate cancellation.
"""
await super().cancel(frame)
await self._client.stop()
async def push_frame(self, frame: Frame, direction: FrameDirection = FrameDirection.DOWNSTREAM):
"""Push a frame to the next processor in the pipeline.
Args:
frame: The frame to push.
direction: The direction to push the frame.
"""
# The BotStartedSpeakingFrame and BotStoppedSpeakingFrame are created inside BaseOutputTransport
# This is a workaround, so we can more reliably be aware when the bot has started or stopped speaking
if direction == FrameDirection.DOWNSTREAM:
if isinstance(frame, BotStartedSpeakingFrame):
if self._event_id is not None:
logger.warning("self._event_id is already defined!")
self._event_id = str(frame.id)
elif isinstance(frame, BotStoppedSpeakingFrame):
await self._client.agent_speak_end(self._event_id)
self._event_id = None
await super().push_frame(frame, direction)
async def process_frame(self, frame: Frame, direction: FrameDirection):
"""Process frames and handle interruptions.
Handles various types of frames including interruption events and user speaking states.
Updates the HeyGen client state based on the received frames.
Args:
frame: The frame to process
direction: The direction of frame flow in the pipeline
Note:
Special handling is implemented for:
- InterruptionFrame: Triggers interruption of current speech
- UserStartedSpeakingFrame: Initiates agent listening mode
- UserStoppedSpeakingFrame: Stops agent listening mode
"""
await super().process_frame(frame, direction)
if isinstance(frame, InterruptionFrame):
await self._client.interrupt(self._event_id)
await self.push_frame(frame, direction)
if isinstance(frame, UserStartedSpeakingFrame):
await self._client.start_agent_listening()
await self.push_frame(frame, direction)
elif isinstance(frame, UserStoppedSpeakingFrame):
await self._client.stop_agent_listening()
await self.push_frame(frame, direction)
async def write_audio_frame(self, frame: OutputAudioRawFrame) -> bool:
"""Write an audio frame to the HeyGen transport.
Args:
frame: The audio frame to write.
"""
await self._client.agent_speak(bytes(frame.audio), self._event_id)
return True
class HeyGenParams(TransportParams):
"""Configuration parameters for the HeyGen transport.
Parameters:
audio_in_enabled: Whether to enable audio input from participants.
audio_out_enabled: Whether to enable audio output to participants.
"""
audio_in_enabled: bool = True
audio_out_enabled: bool = True
class HeyGenTransport(BaseTransport):
"""Transport implementation for HeyGen video calls.
When used, the Pipecat bot joins the same virtual room as the HeyGen Avatar and the user.
This is achieved by using `HeyGenTransport`, which initiates the conversation via
`HeyGenApi` and obtains a room URL that all participants connect to.
"""
def __init__(
self,
session: aiohttp.ClientSession,
api_key: str,
params: HeyGenParams = HeyGenParams(),
input_name: Optional[str] = None,
output_name: Optional[str] = None,
session_request: NewSessionRequest = NewSessionRequest(
avatar_id="Shawn_Therapist_public",
version="v2",
),
):
"""Initialize the HeyGen transport.
Sets up a new HeyGen transport instance with the specified configuration for
handling video calls between the Pipecat bot and HeyGen Avatar.
Args:
session: aiohttp session for making async HTTP requests
api_key: HeyGen API key for authentication
params: HeyGen-specific configuration parameters (default: HeyGenParams())
input_name: Optional custom name for the input transport
output_name: Optional custom name for the output transport
session_request: Configuration for the HeyGen session (default: uses Shawn_Therapist_public avatar)
Note:
The transport will automatically join the same virtual room as the HeyGen Avatar
and user through the HeyGenClient, which handles session initialization via HeyGenApi.
"""
super().__init__(input_name=input_name, output_name=output_name)
self._params = params
self._client = HeyGenClient(
api_key=api_key,
session=session,
params=params,
session_request=session_request,
callbacks=HeyGenCallbacks(
on_participant_connected=self._on_participant_connected,
on_participant_disconnected=self._on_participant_disconnected,
),
)
self._input: Optional[HeyGenInputTransport] = None
self._output: Optional[HeyGenOutputTransport] = None
self._HeyGen_participant_id = None
# Register supported handlers. The user will only be able to register
# these handlers.
self._register_event_handler("on_client_connected")
self._register_event_handler("on_client_disconnected")
async def _on_participant_disconnected(self, participant_id: str):
logger.debug(f"HeyGen participant {participant_id} disconnected")
if participant_id != "heygen":
await self._on_client_disconnected(participant_id)
async def _on_participant_connected(self, participant_id: str):
logger.debug(f"HeyGen participant {participant_id} connected")
if participant_id != "heygen":
await self._on_client_connected(participant_id)
if self._input:
await self._input.start_capturing_audio(participant_id)
def input(self) -> FrameProcessor:
"""Get the input transport for receiving media and events.
Returns:
The HeyGen input transport instance.
"""
if not self._input:
self._input = HeyGenInputTransport(client=self._client, params=self._params)
return self._input
def output(self) -> FrameProcessor:
"""Get the output transport for sending media and events.
Returns:
The HeyGen output transport instance.
"""
if not self._output:
self._output = HeyGenOutputTransport(client=self._client, params=self._params)
return self._output
async def _on_client_connected(self, participant: Any):
"""Handle client connected events."""
await self._call_event_handler("on_client_connected", participant)
async def _on_client_disconnected(self, participant: Any):
"""Handle client disconnected events."""
await self._call_event_handler("on_client_disconnected", participant)

View File

@@ -66,7 +66,7 @@ class SmallWebRTCCallbacks(BaseModel):
on_client_disconnected: Called when a client disconnects.
"""
on_app_message: Callable[[Any, str], Awaitable[None]]
on_app_message: Callable[[Any], Awaitable[None]]
on_client_connected: Callable[[SmallWebRTCConnection], Awaitable[None]]
on_client_disconnected: Callable[[SmallWebRTCConnection], Awaitable[None]]
@@ -254,7 +254,7 @@ class SmallWebRTCClient:
@self._webrtc_connection.event_handler("app-message")
async def on_app_message(connection: SmallWebRTCConnection, message: Any):
await self._handle_app_message(message, connection.pc_id)
await self._handle_app_message(message)
def _convert_frame(self, frame_array: np.ndarray, format_name: str) -> np.ndarray:
"""Convert a video frame to RGB format based on the input format.
@@ -512,9 +512,9 @@ class SmallWebRTCClient:
if not self._closing:
await self._callbacks.on_client_disconnected(self._webrtc_connection)
async def _handle_app_message(self, message: Any, sender: str):
async def _handle_app_message(self, message: Any):
"""Handle incoming application messages."""
await self._callbacks.on_app_message(message, sender)
await self._callbacks.on_app_message(message)
def _can_send(self):
"""Check if the connection is ready for sending data."""
@@ -935,11 +935,11 @@ class SmallWebRTCTransport(BaseTransport):
if self._output:
await self._output.queue_frame(frame, FrameDirection.DOWNSTREAM)
async def _on_app_message(self, message: Any, sender: str):
async def _on_app_message(self, message: Any):
"""Handle incoming application messages."""
if self._input:
await self._input.push_app_message(message)
await self._call_event_handler("on_app_message", message, sender)
await self._call_event_handler("on_app_message", message)
async def _on_client_connected(self, webrtc_connection):
"""Handle client connection events."""

View File

@@ -221,7 +221,6 @@ class TavusTransportClient:
),
on_joined=self._on_joined,
on_left=self._on_left,
on_before_leave=partial(self._on_handle_callback, "on_before_leave"),
on_error=partial(self._on_handle_callback, "on_error"),
on_app_message=partial(self._on_handle_callback, "on_app_message"),
on_call_state_updated=partial(self._on_handle_callback, "on_call_state_updated"),

View File

@@ -0,0 +1,87 @@
"""Provides a wrapper class for making async streams cancellable.
This module implements a CancellableStream class that wraps any async iterator,
adding the ability to safely cancel iteration at any point. This is particularly
useful for handling cleanup of async streams in scenarios where early termination
is required.
The module provides functionality for:
- Wrapping any async iterator with cancellation capabilities
- Safe termination of async streams
- Proper cleanup and synchronization during cancellation
"""
import asyncio
from typing import AsyncIterator, Generic, TypeVar
T = TypeVar("T")
class CancellableStream(Generic[T]):
"""A wrapper around an async stream that can be cancelled."""
def __init__(self, stream: AsyncIterator[T]) -> None:
"""Initialize a cancellable stream wrapper.
Creates a wrapper around an async iterator that can be cancelled mid-iteration.
The wrapper maintains state about cancellation requests and iteration status.
Args:
stream: The async iterator to wrap. This is the source stream that will
be iterated over until either exhaustion or cancellation.
Attributes:
_stream: The wrapped async iterator
_cancel_future: A future that completes when cancellation is acknowledged
_cancel_requested: Flag indicating if cancellation has been requested
_iter_started: Flag indicating if iteration has begun
"""
self._stream: AsyncIterator[T] = stream
self._cancel_future: asyncio.Future[None] | None = None
self._cancel_requested: bool = False
self._iter_started: bool = False
async def cancel(self) -> None:
"""Request stream cancellation and wait for acknowledgment.
Sets up cancellation state and waits until the cancellation is acknowledged
by the next iteration attempt. If iteration hasn't started yet, the
cancellation is acknowledged immediately.
The method will:
1. Create a cancellation future if one doesn't exist
2. Mark the stream as cancelled
3. If iteration hasn't started, complete the future immediately
4. Otherwise, wait for the next iteration to acknowledge cancellation
Returns:
None: The method returns when cancellation is acknowledged.
"""
if self._cancel_future is None:
self._cancel_future = asyncio.get_event_loop().create_future()
self._cancel_requested = True
# If iteration has not started, we complete the future immediately
if not self._iter_started:
self._cancel_future.set_result(None)
await self._cancel_future
def __aiter__(self) -> "CancellableStream[T]":
self._iter_started = True
return self
async def __anext__(self) -> T:
if self._cancel_requested:
# Complete the future if cancellation was requested
if self._cancel_future and not self._cancel_future.done():
self._cancel_future.set_result(None)
raise StopAsyncIteration
try:
return await self._stream.__anext__()
except StopAsyncIteration:
# also complete cancel future if iteration naturally ends
if self._cancel_future and not self._cancel_future.done():
self._cancel_future.set_result(None)
raise

View File

@@ -12,12 +12,14 @@ from dotenv import load_dotenv
from pipecat.adapters.schemas.function_schema import FunctionSchema
from pipecat.adapters.schemas.tools_schema import ToolsSchema
from pipecat.frames.frames import LLMContextFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.services.anthropic.llm import AnthropicLLMService
from pipecat.services.google.llm import GoogleLLMService
from pipecat.services.llm_service import FunctionCallParams, LLMService
from pipecat.services.llm_service import LLMService
from pipecat.services.openai.llm import OpenAILLMService
from pipecat.tests.utils import run_test
@@ -46,13 +48,8 @@ def standard_tools() -> ToolsSchema:
async def _test_llm_function_calling(llm: LLMService):
# Create a mock weather function
call_count = 0
async def mock_fetch_weather(params: FunctionCallParams):
nonlocal call_count
call_count += 1
pass
# Create an AsyncMock for the function
mock_fetch_weather = AsyncMock()
llm.register_function(None, mock_fetch_weather)
@@ -63,19 +60,21 @@ async def _test_llm_function_calling(llm: LLMService):
},
{"role": "user", "content": " How is the weather today in San Francisco, California?"},
]
context = LLMContext(messages, standard_tools())
context = OpenAILLMContext(messages, standard_tools())
# This is done by default inside the create_context_aggregator
context.set_llm_adapter(llm.get_llm_adapter())
pipeline = Pipeline([llm])
frames_to_send = [LLMContextFrame(context)]
frames_to_send = [OpenAILLMContextFrame(context)]
await run_test(
pipeline,
frames_to_send=frames_to_send,
expected_down_frames=None,
)
# Assert that the weather function was called once
assert call_count == 1
# Assert that the mock function was called
mock_fetch_weather.assert_called_once()
@pytest.mark.skipif(os.getenv("OPENAI_API_KEY") is None, reason="OPENAI_API_KEY is not set")

View File

@@ -1,117 +0,0 @@
#
# Copyright (c) 2024-2025 Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import struct
import unittest
from pipecat.frames.frames import InputAudioRawFrame, OutputAudioRawFrame, StartFrame
from pipecat.processors.audio.audio_buffer_processor import AudioBufferProcessor
class _PassthroughResampler:
async def resample(
self, audio: bytes, in_rate: int, out_rate: int
) -> bytes: # pragma: no cover - trivial
return audio
class TestAudioBufferProcessor(unittest.IsolatedAsyncioTestCase):
async def asyncSetUp(self):
self.processor = AudioBufferProcessor(sample_rate=16000, num_channels=2, buffer_size=4)
self.processor._input_resampler = _PassthroughResampler()
self.processor._output_resampler = _PassthroughResampler()
self.processor._update_sample_rate(StartFrame(audio_out_sample_rate=16000))
await self.processor.start_recording()
async def asyncTearDown(self):
if getattr(self.processor, "_recording", False):
await self.processor.stop_recording()
await self.processor.cleanup()
async def test_flush_user_audio_pads_bot_track(self):
user_audio = struct.pack("<hh", 1000, -1000)
audio_event = asyncio.Event()
track_event = asyncio.Event()
captured = {}
async def on_audio_data(_, audio: bytes, sample_rate: int, num_channels: int):
captured["merged"] = (audio, sample_rate, num_channels)
audio_event.set()
async def on_track_audio_data(
_, user: bytes, bot: bytes, sample_rate: int, num_channels: int
):
captured["tracks"] = (user, bot, sample_rate, num_channels)
track_event.set()
self.processor.add_event_handler("on_audio_data", on_audio_data)
self.processor.add_event_handler("on_track_audio_data", on_track_audio_data)
frame = InputAudioRawFrame(audio=user_audio, sample_rate=16000, num_channels=1)
await self.processor._process_recording(frame)
await asyncio.wait_for(audio_event.wait(), timeout=1)
await asyncio.wait_for(track_event.wait(), timeout=1)
merged_audio, merged_sr, merged_channels = captured["merged"]
user_track, bot_track, track_sr, track_channels = captured["tracks"]
self.assertEqual(merged_sr, 16000)
self.assertEqual(merged_channels, 2)
self.assertEqual(track_sr, 16000)
self.assertEqual(track_channels, 2)
self.assertEqual(user_track, user_audio)
self.assertEqual(bot_track, b"\x00" * len(user_audio))
self.assertEqual(len(merged_audio), len(user_audio) * 2)
self.assertEqual(merged_audio[0:2], user_audio[0:2])
self.assertEqual(merged_audio[2:4], b"\x00\x00")
self.assertEqual(merged_audio[4:6], user_audio[2:4])
self.assertEqual(merged_audio[6:8], b"\x00\x00")
self.assertEqual(len(self.processor._user_audio_buffer), 0)
self.assertEqual(len(self.processor._bot_audio_buffer), 0)
async def test_flush_bot_audio_pads_user_track(self):
bot_audio = struct.pack("<hh", -800, 400)
audio_event = asyncio.Event()
track_event = asyncio.Event()
captured = {}
async def on_audio_data(_, audio: bytes, sample_rate: int, num_channels: int):
captured["merged"] = (audio, sample_rate, num_channels)
audio_event.set()
async def on_track_audio_data(
_, user: bytes, bot: bytes, sample_rate: int, num_channels: int
):
captured["tracks"] = (user, bot, sample_rate, num_channels)
track_event.set()
self.processor.add_event_handler("on_audio_data", on_audio_data)
self.processor.add_event_handler("on_track_audio_data", on_track_audio_data)
frame = OutputAudioRawFrame(audio=bot_audio, sample_rate=16000, num_channels=1)
await self.processor._process_recording(frame)
await asyncio.wait_for(audio_event.wait(), timeout=1)
await asyncio.wait_for(track_event.wait(), timeout=1)
merged_audio, merged_sr, merged_channels = captured["merged"]
user_track, bot_track, track_sr, track_channels = captured["tracks"]
self.assertEqual(merged_sr, 16000)
self.assertEqual(merged_channels, 2)
self.assertEqual(track_sr, 16000)
self.assertEqual(track_channels, 2)
self.assertEqual(user_track, b"\x00" * len(bot_audio))
self.assertEqual(bot_track, bot_audio)
self.assertEqual(len(merged_audio), len(bot_audio) * 2)
self.assertEqual(merged_audio[0:2], b"\x00\x00")
self.assertEqual(merged_audio[2:4], bot_audio[0:2])
self.assertEqual(merged_audio[4:6], b"\x00\x00")
self.assertEqual(merged_audio[6:8], bot_audio[2:4])
self.assertEqual(len(self.processor._user_audio_buffer), 0)
self.assertEqual(len(self.processor._bot_audio_buffer), 0)

View File

@@ -10,21 +10,24 @@ from langchain.prompts import ChatPromptTemplate
from langchain_core.language_models import FakeStreamingListLLM
from pipecat.frames.frames import (
LLMContextAssistantTimestampFrame,
LLMContextFrame,
LLMFullResponseEndFrame,
LLMFullResponseStartFrame,
OpenAILLMContextAssistantTimestampFrame,
TextFrame,
TranscriptionFrame,
UserStartedSpeakingFrame,
UserStoppedSpeakingFrame,
)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.processors.aggregators.llm_context import LLMContext
from pipecat.processors.aggregators.llm_response import (
LLMAssistantAggregatorParams,
LLMAssistantContextAggregator,
LLMUserContextAggregator,
)
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.processors.aggregators.llm_response_universal import LLMContextAggregatorPair
from pipecat.processors.frame_processor import FrameProcessor
from pipecat.processors.frameworks.langchain import LangchainProcessor
from pipecat.tests.utils import SleepFrame, run_test
@@ -64,14 +67,13 @@ class TestLangchain(unittest.IsolatedAsyncioTestCase):
proc = LangchainProcessor(chain=chain)
self.mock_proc = self.MockProcessor("token_collector")
context = LLMContext()
context_aggregator = LLMContextAggregatorPair(
context, assistant_params=LLMAssistantAggregatorParams(expect_stripped_words=False)
context = OpenAILLMContext()
tma_in = LLMUserContextAggregator(context)
tma_out = LLMAssistantContextAggregator(
context, params=LLMAssistantAggregatorParams(expect_stripped_words=False)
)
pipeline = Pipeline(
[context_aggregator.user(), proc, self.mock_proc, context_aggregator.assistant()]
)
pipeline = Pipeline([tma_in, proc, self.mock_proc, tma_out])
frames_to_send = [
UserStartedSpeakingFrame(),
@@ -82,8 +84,8 @@ class TestLangchain(unittest.IsolatedAsyncioTestCase):
expected_down_frames = [
UserStartedSpeakingFrame,
UserStoppedSpeakingFrame,
LLMContextFrame,
LLMContextAssistantTimestampFrame,
OpenAILLMContextFrame,
OpenAILLMContextAssistantTimestampFrame,
]
await run_test(
pipeline,
@@ -92,6 +94,4 @@ class TestLangchain(unittest.IsolatedAsyncioTestCase):
)
self.assertEqual("".join(self.mock_proc.token), self.expected_response)
self.assertEqual(
context_aggregator.assistant().messages[-1]["content"], self.expected_response
)
self.assertEqual(tma_out.messages[-1]["content"], self.expected_response)

12
uv.lock generated
View File

@@ -4484,7 +4484,7 @@ requires-dist = [
{ name = "aioboto3", marker = "extra == 'aws'", specifier = "~=15.0.0" },
{ name = "aiofiles", specifier = ">=24.1.0,<25" },
{ name = "aiohttp", specifier = ">=3.11.12,<4" },
{ name = "aiortc", marker = "extra == 'webrtc'", specifier = ">=1.13.0,<2" },
{ name = "aiortc", marker = "extra == 'webrtc'", specifier = "~=1.13.0" },
{ name = "anthropic", marker = "extra == 'anthropic'", specifier = "~=0.49.0" },
{ name = "audioop-lts", marker = "python_full_version >= '3.13'", specifier = "~=0.2.1" },
{ name = "aws-sdk-bedrock-runtime", marker = "python_full_version >= '3.12' and extra == 'aws-nova-sonic'", specifier = "~=0.0.2" },
@@ -4522,7 +4522,7 @@ requires-dist = [
{ name = "onnxruntime", marker = "extra == 'local-smart-turn-v3'", specifier = ">=1.20.1,<2" },
{ name = "onnxruntime", marker = "extra == 'silero'", specifier = ">=1.20.1,<2" },
{ name = "openai", specifier = ">=1.74.0,<=1.99.1" },
{ name = "opencv-python", marker = "extra == 'webrtc'", specifier = ">=4.11.0.86,<5" },
{ name = "opencv-python", marker = "extra == 'webrtc'", specifier = "~=4.11.0.86" },
{ name = "openpipe", marker = "extra == 'openpipe'", specifier = "~=4.50.0" },
{ name = "opentelemetry-api", marker = "extra == 'tracing'", specifier = ">=1.33.0" },
{ name = "opentelemetry-instrumentation", marker = "extra == 'tracing'", specifier = ">=0.54b0" },
@@ -4557,7 +4557,7 @@ requires-dist = [
{ name = "python-dotenv", marker = "extra == 'runner'", specifier = ">=1.0.0,<2.0.0" },
{ name = "pyvips", extras = ["binary"], marker = "extra == 'moondream'", specifier = "~=3.0.0" },
{ name = "resampy", specifier = "~=0.4.3" },
{ name = "sentry-sdk", marker = "extra == 'sentry'", specifier = ">=2.28.0,<3" },
{ name = "sentry-sdk", marker = "extra == 'sentry'", specifier = "~=2.23.1" },
{ name = "simli-ai", marker = "extra == 'simli'", specifier = "~=0.1.10" },
{ name = "soundfile", marker = "extra == 'soundfile'", specifier = "~=0.13.0" },
{ name = "soxr", specifier = "~=0.5.0" },
@@ -6389,15 +6389,15 @@ wheels = [
[[package]]
name = "sentry-sdk"
version = "2.38.0"
version = "2.23.1"
source = { registry = "https://pypi.org/simple" }
dependencies = [
{ name = "certifi" },
{ name = "urllib3" },
]
sdist = { url = "https://files.pythonhosted.org/packages/b2/22/60fd703b34d94d216b2387e048ac82de3e86b63bc28869fb076f8bb0204a/sentry_sdk-2.38.0.tar.gz", hash = "sha256:792d2af45e167e2f8a3347143f525b9b6bac6f058fb2014720b40b84ccbeb985", size = 348116, upload-time = "2025-09-15T15:00:37.846Z" }
sdist = { url = "https://files.pythonhosted.org/packages/60/fd/2c5f7161dbea1fa03381f139c443b4524f3a15d58e50c96a65d19f454ba2/sentry_sdk-2.23.1.tar.gz", hash = "sha256:2288320465065f3f056630ce55936426204f96f63f1208edb79e033ed03774db", size = 316248, upload-time = "2025-03-17T12:52:34.14Z" }
wheels = [
{ url = "https://files.pythonhosted.org/packages/7a/84/bde4c4bbb269b71bc09316af8eb00da91f67814d40337cc12ef9c8742541/sentry_sdk-2.38.0-py2.py3-none-any.whl", hash = "sha256:2324aea8573a3fa1576df7fb4d65c4eb8d9929c8fa5939647397a07179eef8d0", size = 370346, upload-time = "2025-09-15T15:00:35.821Z" },
{ url = "https://files.pythonhosted.org/packages/4e/00/9a9a2ab9020ee824d787f7e82a539305bf926393fe139baedbcf34356770/sentry_sdk-2.23.1-py2.py3-none-any.whl", hash = "sha256:42ef3a6cc1db3d22cb2ab24163d75b23f291ad9892b1a8c44075ce809a32b191", size = 336327, upload-time = "2025-03-17T12:52:32.176Z" },
]
[[package]]