remove extra LLMFullResponseEndFrame

CHANGELOG.md

@@ -9,12 +9,6 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 
 ### Added
 
-- Introduce audio resamplers (`BaseAudioResampler`). This is just a base class
-  to implement audio resamplers. Currently, two implementations are provided
-  `SOXRAudioResampler` and `ResampyResampler`. A new
-  `create_default_resampler()` has been added (replacing the now deprecated
-  `resample_audio()`).
-
 - It is now possible to specify the asyncio event loop that a `PipelineTask` and
   all the processors should run on by passing it as a new argument to the
   `PipelineRunner`. This could allow running pipelines in multiple threads each
@@ -47,9 +41,6 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 
 ### Changed
 
-- `GatedOpenAILLMContextAggregator` now require keyword arguments. Also, a new
-  `start_open` argument has been added to set the initial state of the gate.
-
 - Added `organization` and `project` level authentication to
   `OpenAILLMService`.
 
@@ -65,27 +56,8 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - `InputDTMFFrame` is now based on `DTMFFrame`. There's also a new
   `OutputDTMFFrame` frame.
 
-### Deprecated
-
-- `resample_audio()` is now deprecated, use `create_default_resampler()`
-  instead.
-
-### Removed
-
-- `AudioBufferProcessor.reset_audio_buffers()` has been removed, use
-  `AudioBufferProcessor.start_recording()` and
-  ``AudioBufferProcessor.stop_recording()` instead.
-
 ### Fixed
 
-- Fixed a `AudioBufferProcessor` that would cause crackling in some recordings.
-
-- Fixed an issue in `AudioBufferProcessor` where user callback would not be
-  called on task cancellation.
-
-- Fixed an issue in `AudioBufferProcessor` that would cause wrong silence
-  padding in some cases.
-
 - Fixed an issue where `ElevenLabsTTSService` messages would return a 1009
   websocket error by increasing the max message size limit to 16MB.
 
@@ -101,11 +73,6 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 
 ### Other
 
-- Updated `twilio-chatbot` with a few new features: use 8000 sample rate and
-  avoid resampling, a new client useful for stress testing and testing locally
-  without the need to make phone calls. Also, added audio recording on both the
-  client and the server to make sure the audio sounds good.
-
 - Updated examples to use `task.cancel()` to immediately exit the example when a
   participant leaves or disconnects, instead of pushing an `EndFrame`. Pushing
   an `EndFrame` causes the bot to run through everything that is internally
@@ -1560,9 +1527,6 @@ async def on_connected(processor):
 
 ### Changed
 
-- `FrameSerializer.serialize()` and `FrameSerializer.deserialize()` are now
-  `async`.
-
 - `Filter` has been renamed to `FrameFilter` and it's now under
   `processors/filters`.
 

examples/canonical-metrics/bot.py

@@ -124,7 +124,6 @@ async def main():
 
         @transport.event_handler("on_first_participant_joined")
         async def on_first_participant_joined(transport, participant):
-            await audio_buffer_processor.start_recording()
             await transport.capture_participant_transcription(participant["id"])
             await task.queue_frames([context_aggregator.user().get_context_frame()])
 

examples/chatbot-audio-recording/bot.py

@@ -109,9 +109,8 @@ async def main():
         context = OpenAILLMContext(messages)
         context_aggregator = llm.create_context_aggregator(context)
 
-        # NOTE: Watch out! This will save all the conversation in memory. You
-        # can pass `buffer_size` to get periodic callbacks.
-        audiobuffer = AudioBufferProcessor()
+        # Save audio every 10 seconds.
+        audiobuffer = AudioBufferProcessor(buffer_size=480000)
 
         pipeline = Pipeline(
             [
@@ -133,7 +132,6 @@ async def main():
 
         @transport.event_handler("on_first_participant_joined")
         async def on_first_participant_joined(transport, participant):
-            await audiobuffer.start_recording()
             await transport.capture_participant_transcription(participant["id"])
             await task.queue_frames([context_aggregator.user().get_context_frame()])
 

examples/foundational/14e-function-calling-gemini.py

@@ -63,7 +63,7 @@ async def main():
         )
 
         llm = GoogleLLMService(
-            model="gemini-2.0-flash-exp",
+            model="gemini-1.5-flash-latest",
             # model="gemini-exp-1114",
             api_key=os.getenv("GOOGLE_API_KEY"),
         )

examples/foundational/22-natural-conversation.py

@@ -104,11 +104,8 @@ async def main():
         )
 
         # This processor keeps the last context and will let it through once the
-        # notifier is woken up. We start with the gate open because we send an
-        # initial context frame to start the conversation.
-        gated_context_aggregator = GatedOpenAILLMContextAggregator(
-            notifier=notifier, start_open=True
-        )
+        # notifier is woken up.
+        gated_context_aggregator = GatedOpenAILLMContextAggregator(notifier)
 
         # Notify if the user hasn't said anything.
         async def user_idle_notifier(frame):

examples/foundational/22b-natural-conversation-proposal.py

@@ -129,9 +129,9 @@ class CompletenessCheck(FrameProcessor):
 
 
 class OutputGate(FrameProcessor):
-    def __init__(self, *, notifier: BaseNotifier, start_open: bool = False, **kwargs):
+    def __init__(self, notifier: BaseNotifier, **kwargs):
         super().__init__(**kwargs)
-        self._gate_open = start_open
+        self._gate_open = False
         self._frames_buffer = []
         self._notifier = notifier
 
@@ -252,9 +252,7 @@ async def main():
         # sentence, this will wake up the notifier if that happens.
         user_idle = UserIdleProcessor(callback=user_idle_notifier, timeout=5.0)
 
-        # We start with the gate open because we send an initial context frame
-        # to start the conversation.
-        bot_output_gate = OutputGate(notifier=notifier, start_open=True)
+        bot_output_gate = OutputGate(notifier=notifier)
 
         async def block_user_stopped_speaking(frame):
             return not isinstance(frame, UserStoppedSpeakingFrame)

examples/foundational/22c-natural-conversation-mixed-llms.py

@@ -333,9 +333,9 @@ class CompletenessCheck(FrameProcessor):
 
 
 class OutputGate(FrameProcessor):
-    def __init__(self, *, notifier: BaseNotifier, start_open: bool = False, **kwargs):
+    def __init__(self, notifier: BaseNotifier, **kwargs):
         super().__init__(**kwargs)
-        self._gate_open = start_open
+        self._gate_open = False
         self._frames_buffer = []
         self._notifier = notifier
 
@@ -461,9 +461,7 @@ async def main():
         # sentence, this will wake up the notifier if that happens.
         user_idle = UserIdleProcessor(callback=user_idle_notifier, timeout=5.0)
 
-        # We start with the gate open because we send an initial context frame
-        # to start the conversation.
-        bot_output_gate = OutputGate(notifier=notifier, start_open=True)
+        bot_output_gate = OutputGate(notifier=notifier)
 
         async def block_user_stopped_speaking(frame):
             return not isinstance(frame, UserStoppedSpeakingFrame)

examples/storytelling-chatbot/Dockerfile

@@ -1,11 +1,54 @@
-FROM pipecatai/cloud-base:latest
+FROM python:3.11-slim-bookworm
 
+ARG DEBIAN_FRONTEND=noninteractive
+ARG USE_PERSISTENT_DATA
+ENV PYTHONUNBUFFERED=1
+ENV NODE_MAJOR=20
+
+# Expose FastAPI port
+ENV FAST_API_PORT=7860
+EXPOSE 7860
+
+# Install system dependencies
+RUN apt-get update && apt-get install --no-install-recommends -y \
+    build-essential \
+    git \
+    ffmpeg \
+    google-perftools \
+    ca-certificates curl gnupg \
+    && apt-get clean && rm -rf /var/lib/apt/lists/*
+
+# Install Node.js
+RUN mkdir -p /etc/apt/keyrings 
+RUN curl -fsSL https://deb.nodesource.com/gpgkey/nodesource-repo.gpg.key | gpg --dearmor -o /etc/apt/keyrings/nodesource.gpg
+RUN echo "deb [signed-by=/etc/apt/keyrings/nodesource.gpg] https://deb.nodesource.com/node_${NODE_MAJOR}.x nodistro main" | tee /etc/apt/sources.list.d/nodesource.list > /dev/null
+RUN apt-get update && apt-get install nodejs -y
+
+# Set up a new user named "user" with user ID 1000
+RUN useradd -m -u 1000 user
+
+# Set home to the user's home directory
+ENV HOME=/home/user \
+    PATH=/home/user/.local/bin:$PATH \
+    PYTHONPATH=$HOME/app \
+    PYTHONUNBUFFERED=1
+
+# Switch to the "user" user
+USER user
+
+# Set the working directory to the user's home directory
+WORKDIR $HOME/app
+
+# Install Python dependencies
 COPY ./requirements.txt requirements.txt
+RUN pip3 install --no-cache-dir --upgrade -r requirements.txt
 
-RUN pip install --no-cache-dir --upgrade -r requirements.txt
+# Copy everything else
+COPY --chown=user ./src/ src/
 
-COPY ./src/bot.py bot.py
-COPY ./src/processors.py processors.py
-COPY ./src/prompts.py prompts.py
-COPY ./src/assets assets
-COPY ./src/utils utils
+# Copy frontend app and build
+COPY --chown=user ./frontend/ frontend/
+RUN cd frontend && npm install && npm run build
+
+# Start the FastAPI server
+CMD python3 src/bot_runner.py --port ${FAST_API_PORT}

examples/storytelling-chatbot/frontend/app/api/route.ts

@@ -1,27 +0,0 @@
-// [POST] /api
-
-export async function POST(request: Request) {
-  const params = await request.json();
-    console.log("in POST, params is ", params)
-    const url = process.env.BOT_START_URL || "http://localhost:7860"
-  const req = await fetch(url, {
-    method: "POST",
-    headers: {
-      "Content-Type": "application/json",
-      Authorization: `Bearer ${process.env.PCC_API_KEY}`,
-    },
-    body: JSON.stringify(params),
-  });
-
-  const res = await req.json();
-
-  if (req.status !== 200) {
-    return Response.json(res, { status: req.status });
-  }
-  console.log({res});
-  return Response.json(res);
-}
-
-export async function GET(request: Request) {
-    return Response.json({message: "Hello World"});
-}

examples/storytelling-chatbot/frontend/components/App.tsx

@@ -27,26 +27,22 @@ export default function Call() {
 
     // Create a new room for the story session
     try {
-      console.log("POSTing to /api")
-      const response = await fetch("/api", {
+      const response = await fetch("/", {
         method: "POST",
         headers: {
           "Content-Type": "application/json",
         },
-        body: JSON.stringify({
-          "createDailyRoom": true
-        })
       });
 
-      const {dailyRoom, dailyToken} = await response.json();
-      console.log({dailyRoom, dailyToken})
+      const { room_url, token } = await response.json();
+
       // Keep a reference to the room url for later
-      setRoom(dailyRoom);
+      setRoom(room_url);
 
       // Join the WebRTC session
       await daily.join({
-        url: dailyRoom,
-        token: dailyToken,
+        url: room_url,
+        token,
         videoSource: false,
         startAudioOff: true,
       });
@@ -58,7 +54,6 @@ export default function Call() {
 
       setState("started");
     } catch (error) {
-      console.log("caught error:", error)
       setState("error");
     }
   }

examples/storytelling-chatbot/frontend/example.env.local

@@ -1,3 +0,0 @@
-SITE_URL=
-PCC_API_KEY=
-BOT_START_URL=

examples/storytelling-chatbot/frontend/next.config.mjs

@@ -0,0 +1,15 @@
+/** @type {import('next').NextConfig} */
+const nextConfig = {
+  output: "export",
+
+  async rewrites() {
+    return [
+      {
+        source: "/:path*",
+        destination: "http://localhost:7860/:path*",
+      },
+    ];
+  },
+};
+
+export default nextConfig;

examples/storytelling-chatbot/frontend/package.json

@@ -9,7 +9,7 @@
     "lint": "next lint"
   },
   "dependencies": {
-    "@daily-co/daily-js": "^0.74.0",
+    "@daily-co/daily-js": "^0.62.0",
     "@daily-co/daily-react": "^0.18.0",
     "@radix-ui/react-select": "^2.1.2",
     "@radix-ui/react-slot": "^1.0.2",
@@ -33,7 +33,6 @@
     "eslint-config-next": "14.1.4",
     "postcss": "^8.4.47",
     "tailwindcss": "^3.4.13",
-    "typescript": "^5.6.2",
-    "vercel": "^41.0.1"
+    "typescript": "^5.6.2"
   }
 }

examples/storytelling-chatbot/requirements.txt

@@ -2,5 +2,5 @@ async_timeout
 fastapi
 uvicorn
 python-dotenv
-pipecat-ai[daily,silero,openai,fal,cartesia,google]~=0.0.55
-pipecatcloud @ git+https://github.com/daily-co/pipecat-cloud@main
+-e "../..[daily,silero,openai,fal,cartesia,google]"
+-e "../../../python-genai"

examples/storytelling-chatbot/src/bot.py

@@ -12,22 +12,19 @@ import sys
 import aiohttp
 from dotenv import load_dotenv
 from loguru import logger
-from processors import StoryBreakReinsertProcessor, StoryImageProcessor, StoryProcessor
+from processors import StoryImageProcessor, StoryProcessor
 from prompts import CUE_USER_TURN, LLM_BASE_PROMPT
 from utils.helpers import load_images, load_sounds
 
 from pipecat.audio.vad.silero import SileroVADAnalyzer
 from pipecat.frames.frames import EndFrame
-from pipecat.pipeline.parallel_pipeline import ParallelPipeline
 from pipecat.pipeline.pipeline import Pipeline
 from pipecat.pipeline.runner import PipelineRunner
-from pipecat.pipeline.sync_parallel_pipeline import SyncParallelPipeline
 from pipecat.pipeline.task import PipelineParams, PipelineTask
 from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
-from pipecat.processors.logger import FrameLogger
-from pipecat.services.elevenlabs import ElevenLabsHttpTTSService, ElevenLabsTTSService
+from pipecat.services.elevenlabs import ElevenLabsTTSService
 from pipecat.services.fal import FalImageGenService
-from pipecat.services.google import GoogleImageGenService, GoogleLLMService
+from pipecat.services.google import GoogleLLMService
 from pipecat.transports.services.daily import (
     DailyParams,
     DailyTransport,
@@ -66,20 +63,13 @@ async def main(room_url, token=None):
 
         # -------------- Services --------------- #
 
-        llm_service = GoogleLLMService(
-            api_key=os.getenv("GOOGLE_API_KEY"),
-            model="gemini-2.0-flash-exp",
+        llm_service = GoogleLLMService(api_key=os.getenv("GOOGLE_API_KEY"))
+
+        tts_service = ElevenLabsTTSService(
+            api_key=os.getenv("ELEVENLABS_API_KEY"), voice_id=os.getenv("ELEVENLABS_VOICE_ID")
         )
 
-        tts_service = ElevenLabsHttpTTSService(
-            aiohttp_session=session,
-            api_key=os.getenv("ELEVENLABS_API_KEY"),
-            voice_id=os.getenv("ELEVENLABS_VOICE_ID"),
-        )
-
-        image_gen = GoogleImageGenService(
-            api_key=os.getenv("GOOGLE_API_KEY"),  # model="imagen-3.0-fast-generate-001"
-        )
+        image_gen = GoogleImageGenService(api_key=os.getenv("GOOGLE_API_KEY"))
 
         # --------------- Setup ----------------- #
 
@@ -109,14 +99,13 @@ async def main(room_url, token=None):
                 image_processor,
                 tts_service,
                 transport.output(),
-                StoryBreakReinsertProcessor(),
                 context_aggregator.assistant(),
             ]
         )
 
         main_task = PipelineTask(
-            pipeline=main_pipeline,
-            params=PipelineParams(
+            main_pipeline,
+            PipelineParams(
                 allow_interruptions=True,
                 enable_metrics=True,
                 enable_usage_metrics=True,
@@ -132,6 +121,7 @@ async def main(room_url, token=None):
                     images["book1"],
                     context_aggregator.user().get_context_frame(),
                     DailyTransportMessageFrame(CUE_USER_TURN),
+                    # sounds["listening"],
                     images["book2"],
                 ]
             )
@@ -150,10 +140,6 @@ async def main(room_url, token=None):
         await runner.run(main_task)
 
 
-async def bot(data, daily_room, daily_token):
-    await main(daily_room, daily_token)
-
-
 if __name__ == "__main__":
     parser = argparse.ArgumentParser(description="Daily Storyteller Bot")
     parser.add_argument("-u", type=str, help="Room URL")

examples/storytelling-chatbot/src/bot_runner.py

@@ -127,8 +127,8 @@ async def start_bot(request: Request) -> JSONResponse:
 
     return JSONResponse(
         {
-            "dailyRoom": room.url,
-            "dailyToken": user_token,
+            "room_url": room.url,
+            "token": user_token,
         }
     )
 

examples/storytelling-chatbot/src/processors.py

@@ -44,15 +44,6 @@ class StoryPromptFrame(TextFrame):
     pass
 
 
-class StoryBreakFrame(Frame):
-    """Frame for storing story text that needs a [break] tag reinserted.
-    Does not inherit from TextFrame to avoid TTS processing.
-    """
-
-    def __init__(self):
-        super().__init__()
-
-
 # ------------ Frame Processors ----------- #
 
 
@@ -71,10 +62,7 @@ class StoryImageProcessor(FrameProcessor):
         super().__init__()
         self._image_gen_service = image_gen_service
         # Create a new LLM service to use a different system prompt, etc
-        self._llm_service = GoogleLLMService(
-            api_key=os.getenv("GOOGLE_API_KEY"),
-            model="gemini-2.0-flash-exp",
-        )
+        self._llm_service = GoogleLLMService(api_key=os.getenv("GOOGLE_API_KEY"))
 
         self.pages = []
         self.image_descriptions = []
@@ -200,25 +188,8 @@ class StoryProcessor(FrameProcessor):
                 if len(before_break) > 2:
                     self._story.append(before_break)
                     await self.push_frame(StoryPageFrame(before_break))
-                    await self.push_frame(StoryBreakFrame())
+                    # await self.push_frame(sounds["ding"])
                     await self.push_frame(DailyTransportMessageFrame(CUE_ASSISTANT_TURN))
 
                 # Keep the remainder (if any) in the buffer
                 self._text = parts[1].strip() if len(parts) > 1 else ""
-
-
-class StoryBreakReinsertProcessor(FrameProcessor):
-    """Re-inserts [break] tags into story text before it reaches the assistant context aggregator.
-
-    This processor looks for StoryBreakFrames (which aren't processed by TTS) and creates
-    TextFrames with [break] tags for the context aggregator.
-    """
-
-    async def process_frame(self, frame: Frame, direction: FrameDirection):
-        await super().process_frame(frame, direction)
-
-        if isinstance(frame, StoryBreakFrame):
-            # Create a new TextFrame with [break] tag
-            await self.push_frame(TextFrame(" [break]"))
-        else:
-            await self.push_frame(frame)

examples/twilio-chatbot/README.md

@@ -107,34 +107,3 @@ The server will start on port 8765. Keep this running while you test with Twilio
 ## Usage
 
 To start a call, simply make a call to your configured Twilio phone number. The webhook URL will direct the call to your FastAPI application, which will handle it accordingly.
-
-## Testing
-
-It is also possible to automatically test the server without making phone calls by using a software client.
-
-First, update `templates/streams.xml` to point to your server's websocket endpoint. For example:
-
-```
-<?xml version="1.0" encoding="UTF-8"?>
-<Response>
-  <Connect>
-    <Stream url="ws://localhost:8765/ws"></Stream>
-  </Connect>
-  <Pause length="40"/>
-</Response>
-```
-
-Then, start the server with `-t` to indicate we are testing:
-
-```sh
-# Make sure you’re in the project directory and your virtual environment is activated
-python server.py -t
-```
-
-Finally, just point the client to the server's URL:
-
-```sh
-python client.py -u http://localhost:8765 -c 2
-```
-
-where `-c` allows you to create multiple concurrent clients.

examples/twilio-chatbot/bot.py

@@ -4,16 +4,10 @@
 # SPDX-License-Identifier: BSD 2-Clause License
 #
 
-import datetime
-import io
 import os
 import sys
-import wave
 
-import aiofiles
-from deepgram import LiveOptions
 from dotenv import load_dotenv
-from fastapi import WebSocket
 from loguru import logger
 
 from pipecat.audio.vad.silero import SileroVADAnalyzer
@@ -21,7 +15,6 @@ from pipecat.pipeline.pipeline import Pipeline
 from pipecat.pipeline.runner import PipelineRunner
 from pipecat.pipeline.task import PipelineParams, PipelineTask
 from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
-from pipecat.processors.audio.audio_buffer_processor import AudioBufferProcessor
 from pipecat.serializers.twilio import TwilioFrameSerializer
 from pipecat.services.cartesia import CartesiaTTSService
 from pipecat.services.deepgram import DeepgramSTTService
@@ -36,73 +29,39 @@ load_dotenv(override=True)
 logger.remove(0)
 logger.add(sys.stderr, level="DEBUG")
 
-SAMPLE_RATE = 8000
 
-
-async def save_audio(server_name: str, audio: bytes, sample_rate: int, num_channels: int):
-    if len(audio) > 0:
-        filename = (
-            f"{server_name}_recording_{datetime.datetime.now().strftime('%Y%m%d_%H%M%S')}.wav"
-        )
-        with io.BytesIO() as buffer:
-            with wave.open(buffer, "wb") as wf:
-                wf.setsampwidth(2)
-                wf.setnchannels(num_channels)
-                wf.setframerate(sample_rate)
-                wf.writeframes(audio)
-            async with aiofiles.open(filename, "wb") as file:
-                await file.write(buffer.getvalue())
-        logger.info(f"Merged audio saved to {filename}")
-    else:
-        logger.info("No audio data to save")
-
-
-async def run_bot(websocket_client: WebSocket, stream_sid: str, testing: bool):
+async def run_bot(websocket_client, stream_sid):
     transport = FastAPIWebsocketTransport(
         websocket=websocket_client,
         params=FastAPIWebsocketParams(
-            audio_in_enabled=True,
             audio_out_enabled=True,
-            audio_out_sample_rate=SAMPLE_RATE,
             add_wav_header=False,
             vad_enabled=True,
-            vad_analyzer=SileroVADAnalyzer(sample_rate=SAMPLE_RATE),
+            vad_analyzer=SileroVADAnalyzer(),
             vad_audio_passthrough=True,
-            serializer=TwilioFrameSerializer(
-                stream_sid, TwilioFrameSerializer.InputParams(sample_rate=SAMPLE_RATE)
-            ),
+            serializer=TwilioFrameSerializer(stream_sid),
         ),
     )
 
     llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
 
-    stt = DeepgramSTTService(
-        api_key=os.getenv("DEEPGRAM_API_KEY"),
-        live_options=LiveOptions(sample_rate=SAMPLE_RATE),
-        audio_passthrough=True,
-    )
+    stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
 
     tts = CartesiaTTSService(
         api_key=os.getenv("CARTESIA_API_KEY"),
         voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22",  # British Lady
-        sample_rate=SAMPLE_RATE,
-        push_silence_after_stop=testing,
     )
 
     messages = [
         {
             "role": "system",
-            "content": "You are an elementary teacher in an audio call. Your output will be converted to audio so don't include special characters in your answers. Respond to what the student said in a short short sentence.",
+            "content": "You are a helpful LLM in an audio call. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way.",
         },
     ]
 
     context = OpenAILLMContext(messages)
     context_aggregator = llm.create_context_aggregator(context)
 
-    # NOTE: Watch out! This will save all the conversation in memory. You can
-    # pass `buffer_size` to get periodic callbacks.
-    audiobuffer = AudioBufferProcessor(sample_rate=SAMPLE_RATE)
-
     pipeline = Pipeline(
         [
             transport.input(),  # Websocket input from client
@@ -111,7 +70,6 @@ async def run_bot(websocket_client: WebSocket, stream_sid: str, testing: bool):
             llm,  # LLM
             tts,  # Text-To-Speech
             transport.output(),  # Websocket output to client
-            audiobuffer,  # Used to buffer the audio in the pipeline
             context_aggregator.assistant(),
         ]
     )
@@ -120,8 +78,6 @@ async def run_bot(websocket_client: WebSocket, stream_sid: str, testing: bool):
 
     @transport.event_handler("on_client_connected")
     async def on_client_connected(transport, client):
-        # Start recording.
-        await audiobuffer.start_recording()
         # Kick off the conversation.
         messages.append({"role": "system", "content": "Please introduce yourself to the user."})
         await task.queue_frames([context_aggregator.user().get_context_frame()])
@@ -130,15 +86,6 @@ async def run_bot(websocket_client: WebSocket, stream_sid: str, testing: bool):
     async def on_client_disconnected(transport, client):
         await task.cancel()
 
-    @audiobuffer.event_handler("on_audio_data")
-    async def on_audio_data(buffer, audio, sample_rate, num_channels):
-        server_name = f"server_{websocket_client.client.port}"
-        await save_audio(server_name, audio, sample_rate, num_channels)
-
-    # We use `handle_sigint=False` because `uvicorn` is controlling keyboard
-    # interruptions. We use `force_gc=True` to force garbage collection after
-    # the runner finishes running a task which could be useful for long running
-    # applications with multiple clients connecting.
-    runner = PipelineRunner(handle_sigint=False, force_gc=True)
+    runner = PipelineRunner(handle_sigint=False)
 
     await runner.run(task)

examples/twilio-chatbot/client.py

@@ -1,203 +0,0 @@
-#
-# Copyright (c) 2025, Daily
-#
-# SPDX-License-Identifier: BSD 2-Clause License
-#
-
-import argparse
-import asyncio
-import datetime
-import io
-import os
-import sys
-import wave
-import xml.etree.ElementTree as ET
-from uuid import uuid4
-
-import aiofiles
-import aiohttp
-from deepgram import LiveOptions
-from dotenv import load_dotenv
-from loguru import logger
-
-from pipecat.audio.vad.silero import SileroVADAnalyzer
-from pipecat.audio.vad.vad_analyzer import VADParams
-from pipecat.frames.frames import EndFrame, TransportMessageUrgentFrame
-from pipecat.pipeline.pipeline import Pipeline
-from pipecat.pipeline.runner import PipelineRunner
-from pipecat.pipeline.task import PipelineParams, PipelineTask
-from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
-from pipecat.processors.audio.audio_buffer_processor import AudioBufferProcessor
-from pipecat.serializers.twilio import TwilioFrameSerializer
-from pipecat.services.cartesia import CartesiaTTSService
-from pipecat.services.deepgram import DeepgramSTTService
-from pipecat.services.openai import OpenAILLMService
-from pipecat.transports.network.websocket_client import (
-    WebsocketClientParams,
-    WebsocketClientTransport,
-)
-
-load_dotenv(override=True)
-
-logger.remove(0)
-logger.add(sys.stderr, level="DEBUG")
-
-
-DEFAULT_CLIENT_DURATION = 30
-SAMPLE_RATE = 8000
-
-
-async def download_twiml(server_url: str) -> str:
-    # TODO(aleix): add error checking.
-    async with aiohttp.ClientSession() as session:
-        async with session.post(server_url) as response:
-            return await response.text()
-
-
-def get_stream_url_from_twiml(twiml: str) -> str:
-    root = ET.fromstring(twiml)
-    # TODO(aleix): add error checking.
-    stream_element = root.find(".//Stream")  # Finds the first <Stream> element
-    url = stream_element.get("url")
-    return url
-
-
-async def save_audio(client_name: str, audio: bytes, sample_rate: int, num_channels: int):
-    if len(audio) > 0:
-        filename = (
-            f"{client_name}_recording_{datetime.datetime.now().strftime('%Y%m%d_%H%M%S')}.wav"
-        )
-        with io.BytesIO() as buffer:
-            with wave.open(buffer, "wb") as wf:
-                wf.setsampwidth(2)
-                wf.setnchannels(num_channels)
-                wf.setframerate(sample_rate)
-                wf.writeframes(audio)
-            async with aiofiles.open(filename, "wb") as file:
-                await file.write(buffer.getvalue())
-        logger.info(f"Merged audio saved to {filename}")
-    else:
-        logger.info("No audio data to save")
-
-
-async def run_client(client_name: str, server_url: str, duration_secs: int):
-    twiml = await download_twiml(server_url)
-
-    stream_url = get_stream_url_from_twiml(twiml)
-
-    stream_sid = str(uuid4())
-
-    transport = WebsocketClientTransport(
-        uri=stream_url,
-        params=WebsocketClientParams(
-            audio_in_enabled=True,
-            audio_out_enabled=True,
-            audio_out_sample_rate=SAMPLE_RATE,
-            add_wav_header=False,
-            serializer=TwilioFrameSerializer(
-                stream_sid, params=TwilioFrameSerializer.InputParams(sample_rate=SAMPLE_RATE)
-            ),
-            vad_enabled=True,
-            vad_analyzer=SileroVADAnalyzer(
-                params=VADParams(stop_secs=1.5), sample_rate=SAMPLE_RATE
-            ),
-            vad_audio_passthrough=True,
-        ),
-    )
-
-    llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
-
-    # We let the audio passthrough so we can record the conversation.
-    stt = DeepgramSTTService(
-        api_key=os.getenv("DEEPGRAM_API_KEY"),
-        live_options=LiveOptions(sample_rate=SAMPLE_RATE),
-        audio_passthrough=True,
-    )
-
-    tts = CartesiaTTSService(
-        api_key=os.getenv("CARTESIA_API_KEY"),
-        voice_id="e13cae5c-ec59-4f71-b0a6-266df3c9bb8e",  # Madame Mischief
-        sample_rate=SAMPLE_RATE,
-        push_silence_after_stop=True,
-    )
-
-    messages = [
-        {
-            "role": "system",
-            "content": "You are an 8 year old child. A teacher will explain you new concepts you want to know about. Feel free to change topics whnever you want. Once you are taught something you need to keep asking for clarifications if you think someone your age would not understand what you are being taught.",
-        },
-    ]
-
-    context = OpenAILLMContext(messages)
-    context_aggregator = llm.create_context_aggregator(context)
-
-    # NOTE: Watch out! This will save all the conversation in memory. You can
-    # pass `buffer_size` to get periodic callbacks.
-    audiobuffer = AudioBufferProcessor(sample_rate=SAMPLE_RATE)
-
-    pipeline = Pipeline(
-        [
-            transport.input(),  # Websocket input from server
-            stt,  # Speech-To-Text
-            context_aggregator.user(),
-            llm,  # LLM
-            tts,  # Text-To-Speech
-            transport.output(),  # Websocket output to server
-            audiobuffer,  # Used to buffer the audio in the pipeline
-            context_aggregator.assistant(),
-        ]
-    )
-
-    task = PipelineTask(pipeline, params=PipelineParams(allow_interruptions=True))
-
-    @transport.event_handler("on_connected")
-    async def on_connected(transport: WebsocketClientTransport, client):
-        # Start recording.
-        await audiobuffer.start_recording()
-
-        message = TransportMessageUrgentFrame(
-            message={"event": "connected", "protocol": "Call", "version": "1.0.0"}
-        )
-        await transport.output().send_message(message)
-
-        message = TransportMessageUrgentFrame(
-            message={"event": "start", "streamSid": stream_sid, "start": {"streamSid": stream_sid}}
-        )
-        await transport.output().send_message(message)
-
-    @audiobuffer.event_handler("on_audio_data")
-    async def on_audio_data(buffer, audio, sample_rate, num_channels):
-        await save_audio(client_name, audio, sample_rate, num_channels)
-
-    async def end_call():
-        await asyncio.sleep(duration_secs)
-        await task.queue_frame(EndFrame())
-
-    runner = PipelineRunner()
-
-    await asyncio.gather(runner.run(task), end_call())
-
-
-async def main():
-    parser = argparse.ArgumentParser(description="Pipecat Twilio Chatbot Client")
-    parser.add_argument("-u", "--url", type=str, required=True, help="specify the server URL")
-    parser.add_argument(
-        "-c", "--clients", type=int, required=True, help="number of concurrent clients"
-    )
-    parser.add_argument(
-        "-d",
-        "--duration",
-        type=int,
-        default=DEFAULT_CLIENT_DURATION,
-        help=f"duration of each client in seconds (default: {DEFAULT_CLIENT_DURATION})",
-    )
-    args, _ = parser.parse_known_args()
-
-    clients = []
-    for i in range(args.clients):
-        clients.append(asyncio.create_task(run_client(f"client_{i}", args.url, args.duration)))
-    await asyncio.gather(*clients)
-
-
-if __name__ == "__main__":
-    asyncio.run(main())

examples/twilio-chatbot/server.py

@@ -4,7 +4,6 @@
 # SPDX-License-Identifier: BSD 2-Clause License
 #
 
-import argparse
 import json
 
 import uvicorn
@@ -39,16 +38,8 @@ async def websocket_endpoint(websocket: WebSocket):
     print(call_data, flush=True)
     stream_sid = call_data["start"]["streamSid"]
     print("WebSocket connection accepted")
-    await run_bot(websocket, stream_sid, app.state.testing)
+    await run_bot(websocket, stream_sid)
 
 
 if __name__ == "__main__":
-    parser = argparse.ArgumentParser(description="Pipecat Twilio Chatbot Server")
-    parser.add_argument(
-        "-t", "--test", action="store_true", default=False, help="set the server in testing mode"
-    )
-    args, _ = parser.parse_known_args()
-
-    app.state.testing = args.test
-
     uvicorn.run(app, host="0.0.0.0", port=8765)

pyproject.toml

@@ -32,7 +32,6 @@ dependencies = [
     "protobuf~=5.29.3",
     "pydantic~=2.10.5",
     "pyloudnorm~=0.1.1",
-    "resampy~=0.4.3",
     "soxr~=0.5.0"
 ]
 

src/pipecat/audio/filters/krisp_filter.py

@@ -20,22 +20,6 @@ except ModuleNotFoundError as e:
     raise Exception(f"Missing module: {e}")
 
 
-class KrispProcessorManager:
-    """
-    Ensures that only one KrispAudioProcessor instance exists for the entire program.
-    """
-
-    _krisp_instance = None
-
-    @classmethod
-    def get_processor(cls, sample_rate: int, sample_type: str, channels: int, model_path: str):
-        if cls._krisp_instance is None:
-            cls._krisp_instance = KrispAudioProcessor(
-                sample_rate, sample_type, channels, model_path
-            )
-        return cls._krisp_instance
-
-
 class KrispFilter(BaseAudioFilter):
     def __init__(
         self, sample_type: str = "PCM_16", channels: int = 1, model_path: str = None
@@ -64,7 +48,7 @@ class KrispFilter(BaseAudioFilter):
 
     async def start(self, sample_rate: int):
         self._sample_rate = sample_rate
-        self._krisp_processor = KrispProcessorManager.get_processor(
+        self._krisp_processor = KrispAudioProcessor(
             self._sample_rate, self._sample_type, self._channels, self._model_path
         )
 

src/pipecat/audio/resamplers/__init__.py

@@ -1 +0,0 @@
-

src/pipecat/audio/resamplers/base_audio_resampler.py

@@ -1,30 +0,0 @@
-#
-# Copyright (c) 2024–2025, Daily
-#
-# SPDX-License-Identifier: BSD 2-Clause License
-#
-
-from abc import ABC, abstractmethod
-
-
-class BaseAudioResampler(ABC):
-    """Abstract base class for audio resampling. This class defines an
-    interface for audio resampling implementations.
-    """
-
-    @abstractmethod
-    async def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
-        """
-        Resamples the given audio data to a different sample rate.
-
-        This is an abstract method that must be implemented in subclasses.
-
-        Parameters:
-            audio (bytes): The audio data to be resampled, represented as a byte string.
-            in_rate (int): The original sample rate of the audio data (in Hz).
-            out_rate (int): The desired sample rate for the resampled audio data (in Hz).
-
-        Returns:
-            bytes: The resampled audio data as a byte string.
-        """
-        pass

src/pipecat/audio/resamplers/resampy_resampler.py

@@ -1,25 +0,0 @@
-#
-# Copyright (c) 2024–2025, Daily
-#
-# SPDX-License-Identifier: BSD 2-Clause License
-#
-
-import numpy as np
-import resampy
-
-from pipecat.audio.resamplers.base_audio_resampler import BaseAudioResampler
-
-
-class ResampyResampler(BaseAudioResampler):
-    """Audio resampler implementation using the resampy library."""
-
-    def __init__(self, **kwargs):
-        pass
-
-    async def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
-        if in_rate == out_rate:
-            return audio
-        audio_data = np.frombuffer(audio, dtype=np.int16)
-        resampled_audio = resampy.resample(audio_data, in_rate, out_rate, filter="kaiser_fast")
-        result = resampled_audio.astype(np.int16).tobytes()
-        return result

src/pipecat/audio/resamplers/soxr_resampler.py

@@ -1,25 +0,0 @@
-#
-# Copyright (c) 2024–2025, Daily
-#
-# SPDX-License-Identifier: BSD 2-Clause License
-#
-
-import numpy as np
-import soxr
-
-from pipecat.audio.resamplers.base_audio_resampler import BaseAudioResampler
-
-
-class SOXRAudioResampler(BaseAudioResampler):
-    """Audio resampler implementation using the SoX resampler library."""
-
-    def __init__(self, **kwargs):
-        pass
-
-    async def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
-        if in_rate == out_rate:
-            return audio
-        audio_data = np.frombuffer(audio, dtype=np.int16)
-        resampled_audio = soxr.resample(audio_data, in_rate, out_rate, quality="VHQ")
-        result = resampled_audio.astype(np.int16).tobytes()
-        return result

src/pipecat/audio/utils.py

@@ -10,24 +10,8 @@ import numpy as np
 import pyloudnorm as pyln
 import soxr
 
-from pipecat.audio.resamplers.base_audio_resampler import BaseAudioResampler
-from pipecat.audio.resamplers.soxr_resampler import SOXRAudioResampler
-
-
-def create_default_resampler(**kwargs) -> BaseAudioResampler:
-    return SOXRAudioResampler(**kwargs)
-
 
 def resample_audio(audio: bytes, original_rate: int, target_rate: int) -> bytes:
-    import warnings
-
-    with warnings.catch_warnings():
-        warnings.simplefilter("always")
-        warnings.warn(
-            "'resample_audio()' is deprecated, use 'create_default_resampler()' instead.",
-            DeprecationWarning,
-        )
-
     if original_rate == target_rate:
         return audio
     audio_data = np.frombuffer(audio, dtype=np.int16)
@@ -91,45 +75,41 @@ def exp_smoothing(value: float, prev_value: float, factor: float) -> float:
     return prev_value + factor * (value - prev_value)
 
 
-async def ulaw_to_pcm(
-    ulaw_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler
-):
+def ulaw_to_pcm(ulaw_bytes: bytes, in_sample_rate: int, out_sample_rate: int):
     # Convert μ-law to PCM
     in_pcm_bytes = audioop.ulaw2lin(ulaw_bytes, 2)
 
     # Resample
-    out_pcm_bytes = await resampler.resample(in_pcm_bytes, in_rate, out_rate)
+    out_pcm_bytes = resample_audio(in_pcm_bytes, in_sample_rate, out_sample_rate)
 
     return out_pcm_bytes
 
 
-async def pcm_to_ulaw(pcm_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler):
+def pcm_to_ulaw(pcm_bytes: bytes, in_sample_rate: int, out_sample_rate: int):
     # Resample
-    in_pcm_bytes = await resampler.resample(pcm_bytes, in_rate, out_rate)
+    in_pcm_bytes = resample_audio(pcm_bytes, in_sample_rate, out_sample_rate)
 
     # Convert PCM to μ-law
-    out_ulaw_bytes = audioop.lin2ulaw(in_pcm_bytes, 2)
+    ulaw_bytes = audioop.lin2ulaw(in_pcm_bytes, 2)
 
-    return out_ulaw_bytes
+    return ulaw_bytes
 
 
-async def alaw_to_pcm(
-    alaw_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler
-) -> bytes:
+def alaw_to_pcm(alaw_bytes: bytes, in_sample_rate: int, out_sample_rate: int) -> bytes:
     # Convert a-law to PCM
     in_pcm_bytes = audioop.alaw2lin(alaw_bytes, 2)
 
     # Resample
-    out_pcm_bytes = await resampler.resample(in_pcm_bytes, in_rate, out_rate)
+    out_pcm_bytes = resample_audio(in_pcm_bytes, in_sample_rate, out_sample_rate)
 
     return out_pcm_bytes
 
 
-async def pcm_to_alaw(pcm_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler):
+def pcm_to_alaw(pcm_bytes: bytes, in_sample_rate: int, out_sample_rate: int):
     # Resample
-    in_pcm_bytes = await resampler.resample(pcm_bytes, in_rate, out_rate)
+    in_pcm_bytes = resample_audio(pcm_bytes, in_sample_rate, out_sample_rate)
 
     # Convert PCM to μ-law
-    out_alaw_bytes = audioop.lin2alaw(in_pcm_bytes, 2)
+    alaw_bytes = audioop.lin2alaw(in_pcm_bytes, 2)
 
-    return out_alaw_bytes
+    return alaw_bytes

src/pipecat/processors/aggregators/gated_openai_llm_context.py

@@ -16,10 +16,9 @@ class GatedOpenAILLMContextAggregator(FrameProcessor):
 
     """
 
-    def __init__(self, *, notifier: BaseNotifier, start_open: bool = False, **kwargs):
+    def __init__(self, notifier: BaseNotifier, **kwargs):
         super().__init__(**kwargs)
         self._notifier = notifier
-        self._start_open = start_open
         self._last_context_frame = None
 
     async def process_frame(self, frame: Frame, direction: FrameDirection):
@@ -32,11 +31,7 @@ class GatedOpenAILLMContextAggregator(FrameProcessor):
             await self._stop()
             await self.push_frame(frame)
         elif isinstance(frame, OpenAILLMContextFrame):
-            if self._start_open:
-                self._start_open = False
-                await self.push_frame(frame, direction)
-            else:
-                self._last_context_frame = frame
+            self._last_context_frame = frame
         else:
             await self.push_frame(frame, direction)
 

src/pipecat/processors/async_generator.py

@@ -30,7 +30,7 @@ class AsyncGeneratorProcessor(FrameProcessor):
         if isinstance(frame, (CancelFrame, EndFrame)):
             await self._data_queue.put(None)
         else:
-            data = await self._serializer.serialize(frame)
+            data = self._serializer.serialize(frame)
             if data:
                 await self._data_queue.put(data)
 

src/pipecat/processors/audio/audio_buffer_processor.py

@@ -4,12 +4,8 @@
 # SPDX-License-Identifier: BSD 2-Clause License
 #
 
-import time
-
-from pipecat.audio.utils import create_default_resampler, interleave_stereo_audio, mix_audio
+from pipecat.audio.utils import interleave_stereo_audio, mix_audio, resample_audio
 from pipecat.frames.frames import (
-    AudioRawFrame,
-    CancelFrame,
     EndFrame,
     Frame,
     InputAudioRawFrame,
@@ -43,13 +39,6 @@ class AudioBufferProcessor(FrameProcessor):
         self._user_audio_buffer = bytearray()
         self._bot_audio_buffer = bytearray()
 
-        self._last_user_frame_at = 0
-        self._last_bot_frame_at = 0
-
-        self._recording = False
-
-        self._resampler = create_default_resampler()
-
         self._register_event_handler("on_audio_data")
 
     @property
@@ -75,76 +64,43 @@ class AudioBufferProcessor(FrameProcessor):
         else:
             return b""
 
-    async def start_recording(self):
-        self._recording = True
-        self._reset_recording()
-
-    async def stop_recording(self):
-        await self._call_on_audio_data_handler()
-        self._recording = False
+    def reset_audio_buffers(self):
+        self._user_audio_buffer = bytearray()
+        self._bot_audio_buffer = bytearray()
 
     async def process_frame(self, frame: Frame, direction: FrameDirection):
         await super().process_frame(frame, direction)
 
-        if self._recording and isinstance(frame, InputAudioRawFrame):
-            # Add silence if we need to.
-            silence = self._compute_silence(self._last_user_frame_at)
-            self._user_audio_buffer.extend(silence)
-            # Add user audio.
-            resampled = await self._resample_audio(frame)
+        # Include all audio from the user.
+        if isinstance(frame, InputAudioRawFrame):
+            resampled = resample_audio(frame.audio, frame.sample_rate, self._sample_rate)
             self._user_audio_buffer.extend(resampled)
-            # Save time of frame so we can compute silence.
-            self._last_user_frame_at = time.time()
-        elif self._recording and isinstance(frame, OutputAudioRawFrame):
-            # Add silence if we need to.
-            silence = self._compute_silence(self._last_bot_frame_at)
-            self._bot_audio_buffer.extend(silence)
-            # Add bot audio.
-            resampled = await self._resample_audio(frame)
+            # Sync the bot's buffer to the user's buffer by adding silence if needed
+            if len(self._user_audio_buffer) > len(self._bot_audio_buffer):
+                silence = b"\x00" * len(resampled)
+                self._bot_audio_buffer.extend(silence)
+        # If the bot is speaking, include all audio from the bot.
+        elif isinstance(frame, OutputAudioRawFrame):
+            resampled = resample_audio(frame.audio, frame.sample_rate, self._sample_rate)
             self._bot_audio_buffer.extend(resampled)
-            # 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:
             await self._call_on_audio_data_handler()
 
-        if isinstance(frame, (CancelFrame, EndFrame)):
-            await self.stop_recording()
+        if isinstance(frame, EndFrame):
+            await self._call_on_audio_data_handler()
 
         await self.push_frame(frame, direction)
 
     async def _call_on_audio_data_handler(self):
-        if not self.has_audio() or not self._recording:
+        if not self.has_audio():
             return
 
         merged_audio = self.merge_audio_buffers()
         await self._call_event_handler(
             "on_audio_data", merged_audio, self._sample_rate, self._num_channels
         )
-        self._reset_audio_buffers()
+        self.reset_audio_buffers()
 
     def _buffer_has_audio(self, buffer: bytearray) -> bool:
         return buffer is not None and len(buffer) > 0
-
-    def _reset_recording(self):
-        self._reset_audio_buffers()
-        self._last_user_frame_at = time.time()
-        self._last_bot_frame_at = time.time()
-
-    def _reset_audio_buffers(self):
-        self._user_audio_buffer = bytearray()
-        self._bot_audio_buffer = bytearray()
-
-    async def _resample_audio(self, frame: AudioRawFrame) -> bytes:
-        return await self._resampler.resample(frame.audio, frame.sample_rate, self._sample_rate)
-
-    def _compute_silence(self, from_time: float) -> bytes:
-        quiet_time = time.time() - from_time
-        # We should get audio frames very frequently. We pick 100ms because
-        # that's big enough, but it could be even a bit slower since we usually
-        # do 20ms audio frames.
-        if from_time == 0 or quiet_time < 0.1:
-            return b""
-        num_bytes = int(quiet_time * self._sample_rate) * 2
-        silence = b"\x00" * num_bytes
-        return silence

src/pipecat/serializers/base_serializer.py

@@ -22,9 +22,9 @@ class FrameSerializer(ABC):
         pass
 
     @abstractmethod
-    async def serialize(self, frame: Frame) -> str | bytes | None:
+    def serialize(self, frame: Frame) -> str | bytes | None:
         pass
 
     @abstractmethod
-    async def deserialize(self, data: str | bytes) -> Frame | None:
+    def deserialize(self, data: str | bytes) -> Frame | None:
         pass

src/pipecat/serializers/livekit.py

@@ -25,7 +25,7 @@ class LivekitFrameSerializer(FrameSerializer):
     def type(self) -> FrameSerializerType:
         return FrameSerializerType.BINARY
 
-    async def serialize(self, frame: Frame) -> str | bytes | None:
+    def serialize(self, frame: Frame) -> str | bytes | None:
         if not isinstance(frame, OutputAudioRawFrame):
             return None
         audio_frame = AudioFrame(
@@ -36,7 +36,7 @@ class LivekitFrameSerializer(FrameSerializer):
         )
         return pickle.dumps(audio_frame)
 
-    async def deserialize(self, data: str | bytes) -> Frame | None:
+    def deserialize(self, data: str | bytes) -> Frame | None:
         audio_frame: AudioFrame = pickle.loads(data)["frame"]
         return InputAudioRawFrame(
             audio=bytes(audio_frame.data),

src/pipecat/serializers/protobuf.py

@@ -41,7 +41,7 @@ class ProtobufFrameSerializer(FrameSerializer):
     def type(self) -> FrameSerializerType:
         return FrameSerializerType.BINARY
 
-    async def serialize(self, frame: Frame) -> str | bytes | None:
+    def serialize(self, frame: Frame) -> str | bytes | None:
         proto_frame = frame_protos.Frame()
         if type(frame) not in self.SERIALIZABLE_TYPES:
             logger.warning(f"Frame type {type(frame)} is not serializable")
@@ -57,7 +57,26 @@ class ProtobufFrameSerializer(FrameSerializer):
 
         return proto_frame.SerializeToString()
 
-    async def deserialize(self, data: str | bytes) -> Frame | None:
+    def deserialize(self, data: str | bytes) -> Frame | None:
+        """Returns a Frame object from a Frame protobuf.
+
+        Used to convert frames
+        passed over the wire as protobufs to Frame objects used in pipelines
+        and frame processors.
+
+        >>> serializer = ProtobufFrameSerializer()
+        >>> serializer.deserialize(
+        ...     serializer.serialize(OutputAudioFrame(data=b'1234567890')))
+        InputAudioFrame(data=b'1234567890')
+
+        >>> serializer.deserialize(
+        ...     serializer.serialize(TextFrame(text='hello world')))
+        TextFrame(text='hello world')
+
+        >>> serializer.deserialize(serializer.serialize(TranscriptionFrame(
+        ...     text="Hello there!", participantId="123", timestamp="2021-01-01")))
+        TranscriptionFrame(text='Hello there!', participantId='123', timestamp='2021-01-01')
+        """
         proto = frame_protos.Frame.FromString(data)
         which = proto.WhichOneof("frame")
         if which not in self.DESERIALIZABLE_FIELDS:

src/pipecat/serializers/telnyx.py

@@ -9,13 +9,7 @@ import json
 
 from pydantic import BaseModel
 
-from pipecat.audio.utils import (
-    alaw_to_pcm,
-    create_default_resampler,
-    pcm_to_alaw,
-    pcm_to_ulaw,
-    ulaw_to_pcm,
-)
+from pipecat.audio.utils import alaw_to_pcm, pcm_to_alaw, pcm_to_ulaw, ulaw_to_pcm
 from pipecat.frames.frames import (
     AudioRawFrame,
     Frame,
@@ -46,23 +40,21 @@ class TelnyxFrameSerializer(FrameSerializer):
         params.inbound_encoding = inbound_encoding
         self._params = params
 
-        self._resampler = create_default_resampler()
-
     @property
     def type(self) -> FrameSerializerType:
         return FrameSerializerType.TEXT
 
-    async def serialize(self, frame: Frame) -> str | bytes | None:
+    def serialize(self, frame: Frame) -> str | bytes | None:
         if isinstance(frame, AudioRawFrame):
             data = frame.audio
 
             if self._params.inbound_encoding == "PCMU":
-                serialized_data = await pcm_to_ulaw(
-                    data, frame.sample_rate, self._params.telnyx_sample_rate, self._resampler
+                serialized_data = pcm_to_ulaw(
+                    data, frame.sample_rate, self._params.telnyx_sample_rate
                 )
             elif self._params.inbound_encoding == "PCMA":
-                serialized_data = await pcm_to_alaw(
-                    data, frame.sample_rate, self._params.telnyx_sample_rate, self._resampler
+                serialized_data = pcm_to_alaw(
+                    data, frame.sample_rate, self._params.telnyx_sample_rate
                 )
             else:
                 raise ValueError(f"Unsupported encoding: {self._params.inbound_encoding}")
@@ -79,7 +71,7 @@ class TelnyxFrameSerializer(FrameSerializer):
             answer = {"event": "clear"}
             return json.dumps(answer)
 
-    async def deserialize(self, data: str | bytes) -> Frame | None:
+    def deserialize(self, data: str | bytes) -> Frame | None:
         message = json.loads(data)
 
         if message["event"] == "media":
@@ -87,18 +79,12 @@ class TelnyxFrameSerializer(FrameSerializer):
             payload = base64.b64decode(payload_base64)
 
             if self._params.outbound_encoding == "PCMU":
-                deserialized_data = await ulaw_to_pcm(
-                    payload,
-                    self._params.telnyx_sample_rate,
-                    self._params.sample_rate,
-                    self._resampler,
+                deserialized_data = ulaw_to_pcm(
+                    payload, self._params.telnyx_sample_rate, self._params.sample_rate
                 )
             elif self._params.outbound_encoding == "PCMA":
-                deserialized_data = await alaw_to_pcm(
-                    payload,
-                    self._params.telnyx_sample_rate,
-                    self._params.sample_rate,
-                    self._resampler,
+                deserialized_data = alaw_to_pcm(
+                    payload, self._params.telnyx_sample_rate, self._params.sample_rate
                 )
             else:
                 raise ValueError(f"Unsupported encoding: {self._params.outbound_encoding}")

src/pipecat/serializers/twilio.py

@@ -9,7 +9,7 @@ import json
 
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler, pcm_to_ulaw, ulaw_to_pcm
+from pipecat.audio.utils import pcm_to_ulaw, ulaw_to_pcm
 from pipecat.frames.frames import (
     AudioRawFrame,
     Frame,
@@ -32,22 +32,18 @@ class TwilioFrameSerializer(FrameSerializer):
         self._stream_sid = stream_sid
         self._params = params
 
-        self._resampler = create_default_resampler()
-
     @property
     def type(self) -> FrameSerializerType:
         return FrameSerializerType.TEXT
 
-    async def serialize(self, frame: Frame) -> str | bytes | None:
+    def serialize(self, frame: Frame) -> str | bytes | None:
         if isinstance(frame, StartInterruptionFrame):
             answer = {"event": "clear", "streamSid": self._stream_sid}
             return json.dumps(answer)
         elif isinstance(frame, AudioRawFrame):
             data = frame.audio
 
-            serialized_data = await pcm_to_ulaw(
-                data, frame.sample_rate, self._params.twilio_sample_rate, self._resampler
-            )
+            serialized_data = pcm_to_ulaw(data, frame.sample_rate, self._params.twilio_sample_rate)
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
                 "event": "media",
@@ -59,15 +55,15 @@ class TwilioFrameSerializer(FrameSerializer):
         elif isinstance(frame, (TransportMessageFrame, TransportMessageUrgentFrame)):
             return json.dumps(frame.message)
 
-    async def deserialize(self, data: str | bytes) -> Frame | None:
+    def deserialize(self, data: str | bytes) -> Frame | None:
         message = json.loads(data)
 
         if message["event"] == "media":
             payload_base64 = message["media"]["payload"]
             payload = base64.b64decode(payload_base64)
 
-            deserialized_data = await ulaw_to_pcm(
-                payload, self._params.twilio_sample_rate, self._params.sample_rate, self._resampler
+            deserialized_data = ulaw_to_pcm(
+                payload, self._params.twilio_sample_rate, self._params.sample_rate
             )
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._params.sample_rate

src/pipecat/services/aws.py

@@ -10,7 +10,7 @@ from typing import AsyncGenerator, Optional
 from loguru import logger
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import resample_audio
 from pipecat.frames.frames import (
     ErrorFrame,
     Frame,
@@ -148,8 +148,6 @@ class PollyTTSService(TTSService):
             "volume": params.volume,
         }
 
-        self._resampler = create_default_resampler()
-
         self.set_voice(voice_id)
 
     def can_generate_metrics(self) -> bool:
@@ -195,7 +193,8 @@ class PollyTTSService(TTSService):
             response = self._polly_client.synthesize_speech(**args)
             if "AudioStream" in response:
                 audio_data = response["AudioStream"].read()
-                return audio_data
+                resampled = resample_audio(audio_data, 16000, self._settings["sample_rate"])
+                return resampled
             return None
 
         logger.debug(f"Generating TTS: [{text}]")
@@ -226,10 +225,6 @@ class PollyTTSService(TTSService):
                 yield None
                 return
 
-            audio_data = await self._resampler.resample(
-                audio_data, 16000, self._settings["sample_rate"]
-            )
-
             await self.start_tts_usage_metrics(text)
 
             yield TTSStartedFrame()

src/pipecat/services/canonical.py

@@ -117,6 +117,7 @@ class CanonicalMetricsService(AIService):
         try:
             await self._multipart_upload(filename)
             await aiofiles.os.remove(filename)
+            audio_buffer_processor.reset_audio_buffers()
         except FileNotFoundError:
             pass
         except Exception as e:

src/pipecat/services/cartesia.py

@@ -250,9 +250,7 @@ class CartesiaTTSService(WordTTSService, WebsocketService):
                 # because we are likely still playing out audio and need the
                 # timestamp to set send context frames.
                 self._context_id = None
-                await self.add_word_timestamps(
-                    [("TTSStoppedFrame", 0), ("LLMFullResponseEndFrame", 0), ("Reset", 0)]
-                )
+                await self.add_word_timestamps([("TTSStoppedFrame", 0), ("Reset", 0)])
             elif msg["type"] == "timestamps":
                 await self.add_word_timestamps(
                     list(zip(msg["word_timestamps"]["words"], msg["word_timestamps"]["start"]))

src/pipecat/services/elevenlabs.py

@@ -284,7 +284,7 @@ class ElevenLabsTTSService(WordTTSService, WebsocketService):
         if isinstance(frame, (TTSStoppedFrame, StartInterruptionFrame)):
             self._started = False
             if isinstance(frame, TTSStoppedFrame):
-                await self.add_word_timestamps([("LLMFullResponseEndFrame", 0), ("Reset", 0)])
+                await self.add_word_timestamps([("Reset", 0)])
 
     async def process_frame(self, frame: Frame, direction: FrameDirection):
         await super().process_frame(frame, direction)

src/pipecat/services/tavus.py

@@ -12,7 +12,7 @@ import base64
 import aiohttp
 from loguru import logger
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import resample_audio
 from pipecat.frames.frames import (
     CancelFrame,
     EndFrame,
@@ -47,8 +47,6 @@ class TavusVideoService(AIService):
 
         self._conversation_id: str
 
-        self._resampler = create_default_resampler()
-
     async def initialize(self) -> str:
         url = "https://tavusapi.com/v2/conversations"
         headers = {"Content-Type": "application/json", "x-api-key": self._api_key}
@@ -91,10 +89,12 @@ class TavusVideoService(AIService):
         async with self._session.post(url, headers=headers) as r:
             r.raise_for_status()
 
-    async def _encode_audio_and_send(self, audio: bytes, in_rate: int, done: bool) -> None:
+    async def _encode_audio_and_send(
+        self, audio: bytes, original_sample_rate: int, done: bool
+    ) -> None:
         """Encodes audio to base64 and sends it to Tavus"""
         if not done:
-            audio = await self._resampler.resample(audio, in_rate, 16000)
+            audio = resample_audio(audio, original_sample_rate, 16000)
         audio_base64 = base64.b64encode(audio).decode("utf-8")
         logger.trace(f"{self}: sending {len(audio)} bytes")
         await self._send_audio_message(audio_base64, done=done)

src/pipecat/services/xtts.py

@@ -9,7 +9,7 @@ from typing import Any, AsyncGenerator, Dict
 import aiohttp
 from loguru import logger
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import resample_audio
 from pipecat.frames.frames import (
     ErrorFrame,
     Frame,
@@ -89,8 +89,6 @@ class XTTSService(TTSService):
         self._studio_speakers: Dict[str, Any] | None = None
         self._aiohttp_session = aiohttp_session
 
-        self._resampler = create_default_resampler()
-
     def can_generate_metrics(self) -> bool:
         return True
 
@@ -163,7 +161,7 @@ class XTTSService(TTSService):
                         buffer = buffer[48000:]
 
                         # XTTS uses 24000 so we need to resample to our desired rate.
-                        resampled_audio = await self._resampler.resample(
+                        resampled_audio = resample_audio(
                             bytes(process_data), 24000, self._sample_rate
                         )
                         # Create the frame with the resampled audio
@@ -172,9 +170,7 @@ class XTTSService(TTSService):
 
             # Process any remaining data in the buffer.
             if len(buffer) > 0:
-                resampled_audio = await self._resampler.resample(
-                    bytes(buffer), 24000, self._sample_rate
-                )
+                resampled_audio = resample_audio(bytes(buffer), 24000, self._sample_rate)
                 frame = TTSAudioRawFrame(resampled_audio, self._sample_rate, 1)
                 yield frame
 

src/pipecat/transports/network/fastapi_websocket.py

@@ -91,7 +91,7 @@ class FastAPIWebsocketInputTransport(BaseInputTransport):
     async def _receive_messages(self):
         try:
             async for message in self._iter_data():
-                frame = await self._params.serializer.deserialize(message)
+                frame = self._params.serializer.deserialize(message)
 
                 if not frame:
                     continue
@@ -163,7 +163,7 @@ class FastAPIWebsocketOutputTransport(BaseOutputTransport):
 
     async def _write_frame(self, frame: Frame):
         try:
-            payload = await self._params.serializer.serialize(frame)
+            payload = self._params.serializer.serialize(frame)
             if payload and self._websocket.client_state == WebSocketState.CONNECTED:
                 await self._send_data(payload)
         except Exception as e:

src/pipecat/transports/network/websocket_client.py

@@ -138,7 +138,7 @@ class WebsocketClientInputTransport(BaseInputTransport):
         await self._session.disconnect()
 
     async def on_message(self, websocket, message):
-        frame = await self._params.serializer.deserialize(message)
+        frame = self._params.serializer.deserialize(message)
         if not frame:
             return
         if isinstance(frame, InputAudioRawFrame) and self._params.audio_in_enabled:
@@ -200,7 +200,7 @@ class WebsocketClientOutputTransport(BaseOutputTransport):
         await self._write_audio_sleep()
 
     async def _write_frame(self, frame: Frame):
-        payload = await self._params.serializer.serialize(frame)
+        payload = self._params.serializer.serialize(frame)
         if payload:
             await self._session.send(payload)
 

src/pipecat/transports/network/websocket_server.py

@@ -105,7 +105,7 @@ class WebsocketServerInputTransport(BaseInputTransport):
         # Handle incoming messages
         try:
             async for message in websocket:
-                frame = await self._params.serializer.deserialize(message)
+                frame = self._params.serializer.deserialize(message)
 
                 if not frame:
                     continue
@@ -193,7 +193,7 @@ class WebsocketServerOutputTransport(BaseOutputTransport):
 
     async def _write_frame(self, frame: Frame):
         try:
-            payload = await self._params.serializer.serialize(frame)
+            payload = self._params.serializer.serialize(frame)
             if payload and self._websocket:
                 await self._websocket.send(payload)
         except Exception as e:

src/pipecat/transports/services/livekit.py

@@ -11,7 +11,7 @@ from typing import Any, Awaitable, Callable, List, Optional
 from loguru import logger
 from pydantic import BaseModel
 
-from pipecat.audio.utils import create_default_resampler
+from pipecat.audio.utils import resample_audio
 from pipecat.audio.vad.vad_analyzer import VADAnalyzer
 from pipecat.frames.frames import (
     AudioRawFrame,
@@ -349,7 +349,6 @@ class LiveKitInputTransport(BaseInputTransport):
         self._client = client
         self._audio_in_task = None
         self._vad_analyzer: VADAnalyzer | None = params.vad_analyzer
-        self._resampler = create_default_resampler()
 
     async def start(self, frame: StartFrame):
         await super().start(frame)
@@ -385,9 +384,7 @@ class LiveKitInputTransport(BaseInputTransport):
             audio_data = await self._client.get_next_audio_frame()
             if audio_data:
                 audio_frame_event, participant_id = audio_data
-                pipecat_audio_frame = await self._convert_livekit_audio_to_pipecat(
-                    audio_frame_event
-                )
+                pipecat_audio_frame = self._convert_livekit_audio_to_pipecat(audio_frame_event)
                 input_audio_frame = InputAudioRawFrame(
                     audio=pipecat_audio_frame.audio,
                     sample_rate=pipecat_audio_frame.sample_rate,
@@ -395,12 +392,12 @@ class LiveKitInputTransport(BaseInputTransport):
                 )
                 await self.push_audio_frame(input_audio_frame)
 
-    async def _convert_livekit_audio_to_pipecat(
+    def _convert_livekit_audio_to_pipecat(
         self, audio_frame_event: rtc.AudioFrameEvent
     ) -> AudioRawFrame:
         audio_frame = audio_frame_event.frame
 
-        audio_data = await self._resampler.resample(
+        audio_data = resample_audio(
             audio_frame.data.tobytes(), audio_frame.sample_rate, self._params.audio_in_sample_rate
         )
 

tests/test_protobuf_serializer.py

@@ -20,19 +20,17 @@ class TestProtobufFrameSerializer(unittest.IsolatedAsyncioTestCase):
 
     async def test_roundtrip(self):
         text_frame = TextFrame(text="hello world")
-        frame = await self.serializer.deserialize(await self.serializer.serialize(text_frame))
+        frame = self.serializer.deserialize(self.serializer.serialize(text_frame))
         self.assertEqual(text_frame, frame)
 
         transcription_frame = TranscriptionFrame(
             text="Hello there!", user_id="123", timestamp="2021-01-01"
         )
-        frame = await self.serializer.deserialize(
-            await self.serializer.serialize(transcription_frame)
-        )
+        frame = self.serializer.deserialize(self.serializer.serialize(transcription_frame))
         self.assertEqual(frame, transcription_frame)
 
         audio_frame = OutputAudioRawFrame(audio=b"1234567890", sample_rate=16000, num_channels=1)
-        frame = await self.serializer.deserialize(await self.serializer.serialize(audio_frame))
+        frame = self.serializer.deserialize(self.serializer.serialize(audio_frame))
         self.assertEqual(frame.audio, audio_frame.audio)
         self.assertEqual(frame.sample_rate, audio_frame.sample_rate)
         self.assertEqual(frame.num_channels, audio_frame.num_channels)