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Merge pull request #480 from pipecat-ai/aleix/input-output-frames

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introduce input/output audio and image frames
This commit is contained in:
Aleix Conchillo Flaqué
2024-09-20 14:44:37 -07:00
committed by GitHub
parent 58d9c84bc9 7e39d9ad3d
commit 14acf05a26
48 changed files with 410 additions and 258 deletions
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2
examples/dialin-chatbot/requirements.txt
View File

@@ -1,4 +1,4 @@
pipecat-ai[daily,openai,silero]
pipecat-ai[daily,elevenlabs,openai,silero]
fastapi
uvicorn
python-dotenv

12
examples/foundational/05a-local-sync-speech-and-image.py
View File

@@ -11,7 +11,13 @@ import sys
import tkinter as tk
from pipecat.frames.frames import AudioRawFrame, Frame, URLImageRawFrame, LLMMessagesFrame, TextFrame
from pipecat.frames.frames import (
Frame,
OutputAudioRawFrame,
TTSAudioRawFrame,
URLImageRawFrame,
LLMMessagesFrame,
TextFrame)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.sync_parallel_pipeline import SyncParallelPipeline
@@ -65,9 +71,9 @@ async def main():
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, AudioRawFrame):
if isinstance(frame, TTSAudioRawFrame):
self.audio.extend(frame.audio)
self.frame = AudioRawFrame(
self.frame = OutputAudioRawFrame(
bytes(self.audio), frame.sample_rate, frame.num_channels)
class ImageGrabber(FrameProcessor):

13
examples/foundational/06a-image-sync.py
View File

@@ -11,7 +11,7 @@ import sys
from PIL import Image
from pipecat.frames.frames import ImageRawFrame, Frame, SystemFrame, TextFrame
from pipecat.frames.frames import Frame, OutputImageRawFrame, SystemFrame, TextFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
@@ -52,9 +52,16 @@ class ImageSyncAggregator(FrameProcessor):
await super().process_frame(frame, direction)
if not isinstance(frame, SystemFrame) and direction == FrameDirection.DOWNSTREAM:
await self.push_frame(ImageRawFrame(image=self._speaking_image_bytes, size=(1024, 1024), format=self._speaking_image_format))
await self.push_frame(OutputImageRawFrame(
image=self._speaking_image_bytes,
size=(1024, 1024),
format=self._speaking_image_format)
)
await self.push_frame(frame)
await self.push_frame(ImageRawFrame(image=self._waiting_image_bytes, size=(1024, 1024), format=self._waiting_image_format))
await self.push_frame(OutputImageRawFrame(
image=self._waiting_image_bytes,
size=(1024, 1024),
format=self._waiting_image_format))
else:
await self.push_frame(frame)

25
examples/foundational/09-mirror.py
View File

@@ -8,9 +8,11 @@ import aiohttp
import asyncio
import sys
from pipecat.frames.frames import Frame, InputAudioRawFrame, InputImageRawFrame, OutputAudioRawFrame, OutputImageRawFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.transports.services.daily import DailyTransport, DailyParams
from runner import configure
@@ -24,6 +26,27 @@ logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
class MirrorProcessor(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, InputAudioRawFrame):
await self.push_frame(OutputAudioRawFrame(
audio=frame.audio,
sample_rate=frame.sample_rate,
num_channels=frame.num_channels)
)
elif isinstance(frame, InputImageRawFrame):
await self.push_frame(OutputImageRawFrame(
image=frame.image,
size=frame.size,
format=frame.format)
)
else:
await self.push_frame(frame, direction)
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
@@ -44,7 +67,7 @@ async def main():
async def on_first_participant_joined(transport, participant):
transport.capture_participant_video(participant["id"])
pipeline = Pipeline([transport.input(), transport.output()])
pipeline = Pipeline([transport.input(), MirrorProcessor(), transport.output()])
runner = PipelineRunner()

23
examples/foundational/09a-local-mirror.py
View File

@@ -10,9 +10,11 @@ import sys
import tkinter as tk
from pipecat.frames.frames import Frame, InputAudioRawFrame, InputImageRawFrame, OutputAudioRawFrame, OutputImageRawFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.transports.base_transport import TransportParams
from pipecat.transports.local.tk import TkLocalTransport
from pipecat.transports.services.daily import DailyParams, DailyTransport
@@ -27,6 +29,25 @@ load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
class MirrorProcessor(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, InputAudioRawFrame):
await self.push_frame(OutputAudioRawFrame(
audio=frame.audio,
sample_rate=frame.sample_rate,
num_channels=frame.num_channels)
)
elif isinstance(frame, InputImageRawFrame):
await self.push_frame(OutputImageRawFrame(
image=frame.image,
size=frame.size,
format=frame.format)
)
else:
await self.push_frame(frame, direction)
async def main():
async with aiohttp.ClientSession() as session:
@@ -52,7 +73,7 @@ async def main():
async def on_first_participant_joined(transport, participant):
transport.capture_participant_video(participant["id"])
pipeline = Pipeline([daily_transport.input(), tk_transport.output()])
pipeline = Pipeline([daily_transport.input(), MirrorProcessor(), tk_transport.output()])
task = PipelineTask(pipeline)

6
examples/foundational/11-sound-effects.py
View File

@@ -12,9 +12,9 @@ import wave
from pipecat.frames.frames import (
Frame,
AudioRawFrame,
LLMFullResponseEndFrame,
LLMMessagesFrame,
OutputAudioRawFrame,
)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
@@ -53,8 +53,8 @@ for file in sound_files:
filename = os.path.splitext(os.path.basename(full_path))[0]
# Open the image and convert it to bytes
with wave.open(full_path) as audio_file:
sounds[file] = AudioRawFrame(audio_file.readframes(-1),
audio_file.getframerate(), audio_file.getnchannels())
sounds[file] = OutputAudioRawFrame(audio_file.readframes(-1),
audio_file.getframerate(), audio_file.getnchannels())
class OutboundSoundEffectWrapper(FrameProcessor):

11
examples/moondream-chatbot/bot.py
View File

@@ -13,10 +13,11 @@ from PIL import Image
from pipecat.frames.frames import (
ImageRawFrame,
OutputImageRawFrame,
SpriteFrame,
Frame,
LLMMessagesFrame,
AudioRawFrame,
TTSAudioRawFrame,
TTSStoppedFrame,
TextFrame,
UserImageRawFrame,
@@ -59,7 +60,11 @@ for i in range(1, 26):
# Get the filename without the extension to use as the dictionary key
# Open the image and convert it to bytes
with Image.open(full_path) as img:
sprites.append(ImageRawFrame(image=img.tobytes(), size=img.size, format=img.format))
sprites.append(OutputImageRawFrame(
image=img.tobytes(),
size=img.size,
format=img.format)
)
flipped = sprites[::-1]
sprites.extend(flipped)
@@ -82,7 +87,7 @@ class TalkingAnimation(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, AudioRawFrame):
if isinstance(frame, TTSAudioRawFrame):
if not self._is_talking:
await self.push_frame(talking_frame)
self._is_talking = True

2
examples/moondream-chatbot/requirements.txt
View File

@@ -1,4 +1,4 @@
python-dotenv
fastapi[all]
uvicorn
pipecat-ai[daily,moondream,openai,silero]
pipecat-ai[daily,cartesia,moondream,openai,silero]

7
examples/patient-intake/bot.py
View File

@@ -10,7 +10,7 @@ import os
import sys
import wave
from pipecat.frames.frames import AudioRawFrame
from pipecat.frames.frames import OutputAudioRawFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
@@ -49,8 +49,9 @@ for file in sound_files:
filename = os.path.splitext(os.path.basename(full_path))[0]
# Open the sound and convert it to bytes
with wave.open(full_path) as audio_file:
sounds[file] = AudioRawFrame(audio_file.readframes(-1),
audio_file.getframerate(), audio_file.getnchannels())
sounds[file] = OutputAudioRawFrame(audio_file.readframes(-1),
audio_file.getframerate(),
audio_file.getnchannels())
class IntakeProcessor:

2
examples/patient-intake/requirements.txt
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@@ -1,4 +1,4 @@
python-dotenv
fastapi[all]
uvicorn
pipecat-ai[daily,openai,silero]
pipecat-ai[daily,cartesia,openai,silero]

12
examples/simple-chatbot/bot.py
View File

@@ -16,11 +16,11 @@ from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.llm_response import LLMAssistantResponseAggregator, LLMUserResponseAggregator
from pipecat.frames.frames import (
AudioRawFrame,
ImageRawFrame,
OutputImageRawFrame,
SpriteFrame,
Frame,
LLMMessagesFrame,
TTSAudioRawFrame,
TTSStoppedFrame
)
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
@@ -49,7 +49,11 @@ for i in range(1, 26):
# Get the filename without the extension to use as the dictionary key
# Open the image and convert it to bytes
with Image.open(full_path) as img:
sprites.append(ImageRawFrame(image=img.tobytes(), size=img.size, format=img.format))
sprites.append(OutputImageRawFrame(
image=img.tobytes(),
size=img.size,
format=img.format)
)
flipped = sprites[::-1]
sprites.extend(flipped)
@@ -72,7 +76,7 @@ class TalkingAnimation(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, AudioRawFrame):
if isinstance(frame, TTSAudioRawFrame):
if not self._is_talking:
await self.push_frame(talking_frame)
self._is_talking = True

2
examples/simple-chatbot/requirements.txt
View File

@@ -1,4 +1,4 @@
python-dotenv
fastapi[all]
uvicorn
pipecat-ai[daily,openai,silero]
pipecat-ai[daily,elevenlabs,openai,silero]

2
examples/storytelling-chatbot/requirements.txt
View File

@@ -2,4 +2,4 @@ async_timeout
fastapi
uvicorn
python-dotenv
pipecat-ai[daily,openai,fal]
pipecat-ai[daily,elevenlabs,openai,fal]

11
examples/storytelling-chatbot/src/utils/helpers.py
View File

@@ -2,7 +2,7 @@ import os
import wave
from PIL import Image
from pipecat.frames.frames import AudioRawFrame, ImageRawFrame
from pipecat.frames.frames import OutputAudioRawFrame, OutputImageRawFrame
script_dir = os.path.dirname(__file__)
@@ -16,7 +16,8 @@ def load_images(image_files):
filename = os.path.splitext(os.path.basename(full_path))[0]
# Open the image and convert it to bytes
with Image.open(full_path) as img:
images[filename] = ImageRawFrame(image=img.tobytes(), size=img.size, format=img.format)
images[filename] = OutputImageRawFrame(
image=img.tobytes(), size=img.size, format=img.format)
return images
@@ -30,8 +31,8 @@ def load_sounds(sound_files):
filename = os.path.splitext(os.path.basename(full_path))[0]
# Open the sound and convert it to bytes
with wave.open(full_path) as audio_file:
sounds[filename] = AudioRawFrame(audio=audio_file.readframes(-1),
sample_rate=audio_file.getframerate(),
num_channels=audio_file.getnchannels())
sounds[filename] = OutputAudioRawFrame(audio=audio_file.readframes(-1),
sample_rate=audio_file.getframerate(),
num_channels=audio_file.getnchannels())
return sounds

2
examples/twilio-chatbot/README.md
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@@ -55,7 +55,7 @@ This project is a FastAPI-based chatbot that integrates with Twilio to handle We
2. **Update the Twilio Webhook**:
Copy the ngrok URL and update your Twilio phone number webhook URL to `http://<ngrok_url>/start_call`.
3. **Update the streams.xml**:
3. **Update streams.xml**:
Copy the ngrok URL and update templates/streams.xml with `wss://<ngrok_url>/ws`.
## Running the Application

96
examples/twilio-chatbot/bot.py
View File

@@ -1,4 +1,3 @@
import aiohttp
import os
import sys
@@ -27,63 +26,62 @@ logger.add(sys.stderr, level="DEBUG")
async def run_bot(websocket_client, stream_sid):
async with aiohttp.ClientSession() as session:
transport = FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_out_enabled=True,
add_wav_header=False,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True,
serializer=TwilioFrameSerializer(stream_sid)
)
transport = FastAPIWebsocketTransport(
websocket=websocket_client,
params=FastAPIWebsocketParams(
audio_out_enabled=True,
add_wav_header=False,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True,
serializer=TwilioFrameSerializer(stream_sid)
)
)
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4o")
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4o")
stt = DeepgramSTTService(api_key=os.getenv('DEEPGRAM_API_KEY'))
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
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)
messages = [
{
"role": "system",
"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.",
},
]
messages = [
{
"role": "system",
"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.",
},
]
tma_in = LLMUserResponseAggregator(messages)
tma_out = LLMAssistantResponseAggregator(messages)
tma_in = LLMUserResponseAggregator(messages)
tma_out = LLMAssistantResponseAggregator(messages)
pipeline = Pipeline([
transport.input(), # Websocket input from client
stt, # Speech-To-Text
tma_in, # User responses
llm, # LLM
tts, # Text-To-Speech
transport.output(), # Websocket output to client
tma_out # LLM responses
])
pipeline = Pipeline([
transport.input(), # Websocket input from client
stt, # Speech-To-Text
tma_in, # User responses
llm, # LLM
tts, # Text-To-Speech
transport.output(), # Websocket output to client
tma_out # LLM responses
])
task = PipelineTask(pipeline, params=PipelineParams(allow_interruptions=True))
task = PipelineTask(pipeline, params=PipelineParams(allow_interruptions=True))
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
# Kick off the conversation.
messages.append(
{"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([LLMMessagesFrame(messages)])
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
# Kick off the conversation.
messages.append(
{"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([LLMMessagesFrame(messages)])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
await task.queue_frames([EndFrame()])
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, client):
await task.queue_frames([EndFrame()])
runner = PipelineRunner(handle_sigint=False)
runner = PipelineRunner(handle_sigint=False)
await runner.run(task)
await runner.run(task)

2
examples/twilio-chatbot/requirements.txt
View File

@@ -1,4 +1,4 @@
pipecat-ai[daily,openai,silero,deepgram]
pipecat-ai[daily,cartesia,openai,silero,deepgram]
fastapi
uvicorn
python-dotenv

86
examples/websocket-server/bot.py
View File

@@ -4,7 +4,6 @@
# SPDX-License-Identifier: BSD 2-Clause License
#
import aiohttp
import asyncio
import os
import sys
@@ -33,60 +32,59 @@ logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
transport = WebsocketServerTransport(
params=WebsocketServerParams(
audio_out_enabled=True,
add_wav_header=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True
)
transport = WebsocketServerTransport(
params=WebsocketServerParams(
audio_out_enabled=True,
add_wav_header=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True
)
)
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4o")
llm = OpenAILLMService(
api_key=os.getenv("OPENAI_API_KEY"),
model="gpt-4o")
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
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
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)
messages = [
{
"role": "system",
"content": "You are a helpful LLM in a WebRTC 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.",
},
]
messages = [
{
"role": "system",
"content": "You are a helpful LLM in a WebRTC 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.",
},
]
tma_in = LLMUserResponseAggregator(messages)
tma_out = LLMAssistantResponseAggregator(messages)
tma_in = LLMUserResponseAggregator(messages)
tma_out = LLMAssistantResponseAggregator(messages)
pipeline = Pipeline([
transport.input(), # Websocket input from client
stt, # Speech-To-Text
tma_in, # User responses
llm, # LLM
tts, # Text-To-Speech
transport.output(), # Websocket output to client
tma_out # LLM responses
])
pipeline = Pipeline([
transport.input(), # Websocket input from client
stt, # Speech-To-Text
tma_in, # User responses
llm, # LLM
tts, # Text-To-Speech
transport.output(), # Websocket output to client
tma_out # LLM responses
])
task = PipelineTask(pipeline)
task = PipelineTask(pipeline)
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
# Kick off the conversation.
messages.append(
{"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([LLMMessagesFrame(messages)])
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, client):
# Kick off the conversation.
messages.append(
{"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([LLMMessagesFrame(messages)])
runner = PipelineRunner()
runner = PipelineRunner()
await runner.run(task)
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

1
examples/websocket-server/frames.proto
View File

@@ -24,6 +24,7 @@ message AudioRawFrame {
bytes audio = 3;
uint32 sample_rate = 4;
uint32 num_channels = 5;
optional uint64 pts = 6;
}
message TranscriptionFrame {

2
examples/websocket-server/requirements.txt
View File

@@ -1,2 +1,2 @@
python-dotenv
pipecat-ai[openai,silero,websocket,whisper]
pipecat-ai[cartesia,openai,silero,websocket,whisper]