Merge pull request #2687 from pipecat-ai/memory_leak
Improving memory cleanup
This commit is contained in:
@@ -9,6 +9,8 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
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### Added
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### Added
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- Added memory cleanup improvements to reduce memory peaks.
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- Added `on_before_process_frame`, `on_after_process_frame`,
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- Added `on_before_process_frame`, `on_after_process_frame`,
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`on_before_push_frame` and `on_after_push_frame`. These are synchronous events
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`on_before_push_frame` and `on_after_push_frame`. These are synchronous events
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that get called before and after a frame is processed or pushed. Note that
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that get called before and after a frame is processed or pushed. Note that
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175
examples/foundational/46-video-processing.py
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175
examples/foundational/46-video-processing.py
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@@ -0,0 +1,175 @@
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#
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# Copyright (c) 2025, Daily
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#
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# SPDX-License-Identifier: BSD 2-Clause License
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#
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import os
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import cv2
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import numpy as np
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from dotenv import load_dotenv
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from loguru import logger
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from pipecat.audio.vad.silero import SileroVADAnalyzer
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from pipecat.frames.frames import Frame, InputImageRawFrame, LLMRunFrame, OutputImageRawFrame
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from pipecat.pipeline.pipeline import Pipeline
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from pipecat.pipeline.runner import PipelineRunner
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from pipecat.pipeline.task import PipelineParams, PipelineTask
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from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
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from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
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from pipecat.processors.frameworks.rtvi import RTVIObserver, RTVIProcessor
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from pipecat.runner.types import RunnerArguments
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from pipecat.runner.utils import create_transport
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from pipecat.services.gemini_multimodal_live import GeminiMultimodalLiveLLMService
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from pipecat.transports.base_transport import TransportParams
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from pipecat.transports.daily.transport import DailyParams, DailyTransport
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load_dotenv(override=True)
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transport_params = {
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"daily": lambda: DailyParams(
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audio_in_enabled=True,
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audio_out_enabled=True,
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audio_out_10ms_chunks=2,
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video_in_enabled=True,
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video_out_enabled=True,
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video_out_is_live=True,
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vad_analyzer=SileroVADAnalyzer(),
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),
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"webrtc": lambda: TransportParams(
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audio_in_enabled=True,
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audio_out_enabled=True,
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audio_out_10ms_chunks=2,
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video_in_enabled=True,
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video_out_enabled=True,
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video_out_is_live=True,
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vad_analyzer=SileroVADAnalyzer(),
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),
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}
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class EdgeDetectionProcessor(FrameProcessor):
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def __init__(self, video_out_width, video_out_height: int):
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super().__init__()
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self._video_out_width = video_out_width
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self._video_out_height = video_out_height
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async def process_frame(self, frame: Frame, direction: FrameDirection):
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await super().process_frame(frame, direction)
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# Send back the user's camera video with edge detection applied
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if isinstance(frame, InputImageRawFrame) and frame.transport_source == "camera":
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# Convert bytes to NumPy array
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img = np.frombuffer(frame.image, dtype=np.uint8).reshape(
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(frame.size[1], frame.size[0], 3)
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)
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# perform edge detection only on camera frames
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img = cv2.cvtColor(cv2.Canny(img, 100, 200), cv2.COLOR_GRAY2BGR)
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# convert the size if needed
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desired_size = (self._video_out_width, self._video_out_height)
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if frame.size != desired_size:
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resized_image = cv2.resize(img, desired_size)
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out_frame = OutputImageRawFrame(resized_image.tobytes(), desired_size, frame.format)
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await self.push_frame(out_frame)
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else:
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out_frame = OutputImageRawFrame(
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image=img.tobytes(), size=frame.size, format=frame.format
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)
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await self.push_frame(out_frame)
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else:
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await self.push_frame(frame, direction)
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SYSTEM_INSTRUCTION = f"""
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"You are Gemini Chatbot, a friendly, helpful robot.
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Your goal is to demonstrate your capabilities in a succinct way.
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Your output will be converted to audio so don't include special characters in your answers.
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Respond to what the user said in a creative and helpful way. Keep your responses brief. One or two sentences at most.
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"""
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async def run_bot(pipecat_transport):
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llm = GeminiMultimodalLiveLLMService(
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api_key=os.getenv("GOOGLE_API_KEY"),
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voice_id="Puck", # Aoede, Charon, Fenrir, Kore, Puck
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transcribe_user_audio=True,
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system_instruction=SYSTEM_INSTRUCTION,
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)
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messages = [
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{
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"role": "user",
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"content": "Start by greeting the user warmly and introducing yourself.",
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}
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]
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context = OpenAILLMContext(messages)
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context_aggregator = llm.create_context_aggregator(context)
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# RTVI events for Pipecat client UI
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rtvi = RTVIProcessor()
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pipeline = Pipeline(
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[
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pipecat_transport.input(),
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context_aggregator.user(),
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rtvi,
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llm, # LLM
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EdgeDetectionProcessor(
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pipecat_transport._params.video_out_width,
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pipecat_transport._params.video_out_height,
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), # Sending the video back to the user
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pipecat_transport.output(),
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context_aggregator.assistant(),
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]
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)
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task = PipelineTask(
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pipeline,
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params=PipelineParams(
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enable_metrics=True,
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enable_usage_metrics=True,
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),
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observers=[RTVIObserver(rtvi)],
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)
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@rtvi.event_handler("on_client_ready")
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async def on_client_ready(rtvi):
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logger.info("Pipecat client ready.")
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await rtvi.set_bot_ready()
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# Kick off the conversation.
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await task.queue_frames([LLMRunFrame()])
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@pipecat_transport.event_handler("on_client_connected")
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async def on_client_connected(transport, participant):
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logger.info("Pipecat Client connected")
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if isinstance(transport, DailyTransport):
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await pipecat_transport.capture_participant_video(participant["id"], framerate=30)
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else:
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await pipecat_transport.capture_participant_video("camera")
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@pipecat_transport.event_handler("on_client_disconnected")
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async def on_client_disconnected(transport, client):
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logger.info("Pipecat Client disconnected")
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await task.cancel()
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runner = PipelineRunner(handle_sigint=False, force_gc=True)
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await runner.run(task)
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async def bot(runner_args: RunnerArguments):
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"""Main bot entry point compatible with Pipecat Cloud."""
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transport = await create_transport(runner_args, transport_params)
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await run_bot(transport)
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if __name__ == "__main__":
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from pipecat.runner.run import main
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main()
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@@ -777,7 +777,6 @@ class PipelineTask(BasePipelineTask):
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"""
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"""
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running = True
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running = True
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last_frame_time = 0
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last_frame_time = 0
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frame_buffer = deque(maxlen=10) # Store last 10 frames
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while running:
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while running:
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try:
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try:
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@@ -785,9 +784,6 @@ class PipelineTask(BasePipelineTask):
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self._idle_queue.get(), timeout=self._idle_timeout_secs
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self._idle_queue.get(), timeout=self._idle_timeout_secs
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)
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)
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if not isinstance(frame, InputAudioRawFrame):
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frame_buffer.append(frame)
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if isinstance(frame, StartFrame) or isinstance(frame, self._idle_timeout_frames):
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if isinstance(frame, StartFrame) or isinstance(frame, self._idle_timeout_frames):
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# If we find a StartFrame or one of the frames that prevents a
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# If we find a StartFrame or one of the frames that prevents a
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# time out we update the time.
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# time out we update the time.
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@@ -798,7 +794,7 @@ class PipelineTask(BasePipelineTask):
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# valid frames.
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# valid frames.
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diff_time = time.time() - last_frame_time
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diff_time = time.time() - last_frame_time
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if diff_time >= self._idle_timeout_secs:
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if diff_time >= self._idle_timeout_secs:
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running = await self._idle_timeout_detected(frame_buffer)
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running = await self._idle_timeout_detected()
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# Reset `last_frame_time` so we don't trigger another
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# Reset `last_frame_time` so we don't trigger another
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# immediate idle timeout if we are not cancelling. For
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# immediate idle timeout if we are not cancelling. For
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# example, we might want to force the bot to say goodbye
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# example, we might want to force the bot to say goodbye
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@@ -808,14 +804,11 @@ class PipelineTask(BasePipelineTask):
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self._idle_queue.task_done()
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self._idle_queue.task_done()
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except asyncio.TimeoutError:
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except asyncio.TimeoutError:
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running = await self._idle_timeout_detected(frame_buffer)
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running = await self._idle_timeout_detected()
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async def _idle_timeout_detected(self, last_frames: Deque[Frame]) -> bool:
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async def _idle_timeout_detected(self) -> bool:
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"""Handle idle timeout detection and optional cancellation.
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"""Handle idle timeout detection and optional cancellation.
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Args:
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last_frames: Recent frames received before timeout for debugging.
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Returns:
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Returns:
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Whether the pipeline task should continue running.
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Whether the pipeline task should continue running.
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"""
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"""
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@@ -823,10 +816,7 @@ class PipelineTask(BasePipelineTask):
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if self._cancelled:
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if self._cancelled:
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return True
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return True
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logger.warning("Idle timeout detected. Last 10 frames received:")
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logger.warning("Idle timeout detected.")
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for i, frame in enumerate(last_frames, 1):
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logger.warning(f"Frame {i}: {frame}")
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await self._call_event_handler("on_idle_timeout")
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await self._call_event_handler("on_idle_timeout")
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if self._cancel_on_idle_timeout:
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if self._cancel_on_idle_timeout:
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logger.warning(f"Idle pipeline detected, cancelling pipeline task...")
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logger.warning(f"Idle pipeline detected, cancelling pipeline task...")
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@@ -281,8 +281,10 @@ class BaseOpenAILLMService(LLMService):
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# base64 encode any images
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# base64 encode any images
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for message in messages:
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for message in messages:
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if message.get("mime_type") == "image/jpeg":
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if message.get("mime_type") == "image/jpeg":
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encoded_image = base64.b64encode(message["data"].getvalue()).decode("utf-8")
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# Avoid .getvalue() which makes a full copy of BytesIO
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text = message["content"]
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raw_bytes = message["data"].read()
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encoded_image = base64.b64encode(raw_bytes).decode("utf-8")
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text = message.get("content", "")
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message["content"] = [
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message["content"] = [
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{"type": "text", "text": text},
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{"type": "text", "text": text},
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{
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{
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@@ -290,6 +292,7 @@ class BaseOpenAILLMService(LLMService):
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"image_url": {"url": f"data:image/jpeg;base64,{encoded_image}"},
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"image_url": {"url": f"data:image/jpeg;base64,{encoded_image}"},
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},
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},
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]
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]
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# Explicit cleanup
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del message["data"]
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del message["data"]
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del message["mime_type"]
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del message["mime_type"]
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@@ -659,6 +659,7 @@ class BaseOutputTransport(FrameProcessor):
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self._audio_queue.get(), timeout=vad_stop_secs
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self._audio_queue.get(), timeout=vad_stop_secs
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)
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)
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yield frame
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yield frame
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self._audio_queue.task_done()
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except asyncio.TimeoutError:
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except asyncio.TimeoutError:
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# Notify the bot stopped speaking upstream if necessary.
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# Notify the bot stopped speaking upstream if necessary.
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await self._bot_stopped_speaking()
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await self._bot_stopped_speaking()
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@@ -673,6 +674,7 @@ class BaseOutputTransport(FrameProcessor):
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frame.audio = await self._mixer.mix(frame.audio)
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frame.audio = await self._mixer.mix(frame.audio)
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last_frame_time = time.time()
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last_frame_time = time.time()
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yield frame
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yield frame
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self._audio_queue.task_done()
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except asyncio.QueueEmpty:
|
except asyncio.QueueEmpty:
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# Notify the bot stopped speaking upstream if necessary.
|
# Notify the bot stopped speaking upstream if necessary.
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diff_time = time.time() - last_frame_time
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diff_time = time.time() - last_frame_time
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@@ -309,7 +309,7 @@ class SmallWebRTCClient:
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# self._webrtc_connection.ask_to_renegotiate()
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# self._webrtc_connection.ask_to_renegotiate()
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frame = None
|
frame = None
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except MediaStreamError:
|
except MediaStreamError:
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logger.warning("Received an unexpected media stream error while reading the audio.")
|
logger.warning("Received an unexpected media stream error while reading the video.")
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frame = None
|
frame = None
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if frame is None or not isinstance(frame, VideoFrame):
|
if frame is None or not isinstance(frame, VideoFrame):
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@@ -321,15 +321,21 @@ class SmallWebRTCClient:
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# Convert frame to NumPy array in its native format
|
# Convert frame to NumPy array in its native format
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frame_array = frame.to_ndarray(format=format_name)
|
frame_array = frame.to_ndarray(format=format_name)
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frame_rgb = self._convert_frame(frame_array, format_name)
|
frame_rgb = self._convert_frame(frame_array, format_name)
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del frame_array # free intermediate array immediately
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|
image_bytes = frame_rgb.tobytes()
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del frame_rgb # free RGB array immediately
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|
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image_frame = UserImageRawFrame(
|
image_frame = UserImageRawFrame(
|
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user_id=self._webrtc_connection.pc_id,
|
user_id=self._webrtc_connection.pc_id,
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image=frame_rgb.tobytes(),
|
image=image_bytes,
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size=(frame.width, frame.height),
|
size=(frame.width, frame.height),
|
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format="RGB",
|
format="RGB",
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)
|
)
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image_frame.transport_source = video_source
|
image_frame.transport_source = video_source
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||||||
|
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del frame # free original VideoFrame
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del image_bytes # reference kept in image_frame
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||||||
|
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||||||
yield image_frame
|
yield image_frame
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||||||
|
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||||||
async def read_audio_frame(self):
|
async def read_audio_frame(self):
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@@ -364,23 +370,35 @@ class SmallWebRTCClient:
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resampled_frames = self._pipecat_resampler.resample(frame)
|
resampled_frames = self._pipecat_resampler.resample(frame)
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for resampled_frame in resampled_frames:
|
for resampled_frame in resampled_frames:
|
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# 16-bit PCM bytes
|
# 16-bit PCM bytes
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pcm_bytes = resampled_frame.to_ndarray().astype(np.int16).tobytes()
|
pcm_array = resampled_frame.to_ndarray().astype(np.int16)
|
||||||
|
pcm_bytes = pcm_array.tobytes()
|
||||||
|
del pcm_array # free NumPy array immediately
|
||||||
|
|
||||||
audio_frame = InputAudioRawFrame(
|
audio_frame = InputAudioRawFrame(
|
||||||
audio=pcm_bytes,
|
audio=pcm_bytes,
|
||||||
sample_rate=resampled_frame.sample_rate,
|
sample_rate=resampled_frame.sample_rate,
|
||||||
num_channels=self._audio_in_channels,
|
num_channels=self._audio_in_channels,
|
||||||
)
|
)
|
||||||
|
del pcm_bytes # reference kept in audio_frame
|
||||||
|
|
||||||
yield audio_frame
|
yield audio_frame
|
||||||
else:
|
else:
|
||||||
# 16-bit PCM bytes
|
# 16-bit PCM bytes
|
||||||
pcm_bytes = frame.to_ndarray().astype(np.int16).tobytes()
|
pcm_array = frame.to_ndarray().astype(np.int16)
|
||||||
|
pcm_bytes = pcm_array.tobytes()
|
||||||
|
del pcm_array # free NumPy array immediately
|
||||||
|
|
||||||
audio_frame = InputAudioRawFrame(
|
audio_frame = InputAudioRawFrame(
|
||||||
audio=pcm_bytes,
|
audio=pcm_bytes,
|
||||||
sample_rate=frame.sample_rate,
|
sample_rate=frame.sample_rate,
|
||||||
num_channels=self._audio_in_channels,
|
num_channels=self._audio_in_channels,
|
||||||
)
|
)
|
||||||
|
del pcm_bytes # reference kept in audio_frame
|
||||||
|
|
||||||
yield audio_frame
|
yield audio_frame
|
||||||
|
|
||||||
|
del frame # free original AudioFrame
|
||||||
|
|
||||||
async def write_audio_frame(self, frame: OutputAudioRawFrame):
|
async def write_audio_frame(self, frame: OutputAudioRawFrame):
|
||||||
"""Write an audio frame to the WebRTC connection.
|
"""Write an audio frame to the WebRTC connection.
|
||||||
|
|
||||||
|
|||||||
Reference in New Issue
Block a user