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1 Commits

Author SHA1 Message Date
Aleix Conchillo Flaqué
dcf7e454f6 services(tts): don't wait for BotStoppedSpeakingFrame to resume input queue 2024-12-02 19:00:47 -08:00
206 changed files with 1602 additions and 12439 deletions

9
.gitignore vendored
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@@ -28,11 +28,4 @@ share/python-wheels/
MANIFEST
.DS_Store
.env
fly.toml
# Example files
pipecat/examples/twilio-chatbot/templates/streams.xml
# Documentation
docs/api/_build/
docs/api/api
fly.toml

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@@ -1,36 +0,0 @@
version: 2
build:
os: ubuntu-22.04
tools:
python: '3.12'
apt_packages:
- portaudio19-dev
- python3-dev
- libasound2-dev
jobs:
pre_build:
- python -m pip install --upgrade pip
- pip install wheel setuptools
post_build:
- echo "Build completed"
sphinx:
configuration: docs/api/conf.py
fail_on_warning: false
python:
install:
- requirements: docs/api/requirements.txt
- method: pip
path: .
search:
ranking:
api/*: 5
getting-started/*: 4
guides/*: 3
submodules:
include: all
recursive: true

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@@ -9,178 +9,20 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
### Added
- Add support for more languages to ElevenLabs (Arabic, Croatian, Filipino,
Tamil) and PlayHT (Afrikans, Albanian, Amharic, Arabic, Bengali, Croatian,
Galician, Hebrew, Mandarin, Serbian, Tagalog, Urdu, Xhosa).
### Changed
- It's no longer necessary to call `super().start/stop/cancel(frame)` if you
subclass and implement `AIService.start/stop/cancel()`. This is all now done
internally and will avoid possible issues if you forget to add it.
- It's no longer necessary to call `super().process_frame(frame, direction)` if
you subclass and implement `FrameProcessor.process_frame()`. This is all now
done internally and will avoid possible issues if you forget to add it.
### Deprecated
- `AWSTTSService` is now deprecated, use `PollyTTSService` instead.
### Fixed
- Fixed a `BaseOutputTransport` issue that was causing non-audio frames being
processed before the previous audio frames were played. This will allow, for
example, sending a frame `A` after a `TTSSpeakFrame` and the frame `A` will
only be pushed downstream after the audio generated from `TTSSpeakFrame` has
been spoken.
## [0.0.50] - 2024-12-11
### Added
- Added `GeminiMultimodalLiveLLMService`. This is an integration for Google's
Gemini Multimodal Live API, supporting:
- Real-time audio and video input processing
- Streaming text responses with TTS
- Audio transcription for both user and bot speech
- Function calling
- System instructions and context management
- Dynamic parameter updates (temperature, top_p, etc.)
- Added `AudioTranscriber` utility class for handling audio transcription with
Gemini models.
- Added new context classes for Gemini:
- `GeminiMultimodalLiveContext`
- `GeminiMultimodalLiveUserContextAggregator`
- `GeminiMultimodalLiveAssistantContextAggregator`
- `GeminiMultimodalLiveContextAggregatorPair`
- Added new foundational examples for `GeminiMultimodalLiveLLMService`:
- `26-gemini-multimodal-live.py`
- `26a-gemini-multimodal-live-transcription.py`
- `26b-gemini-multimodal-live-video.py`
- `26c-gemini-multimodal-live-video.py`
- Added `SimliVideoService`. This is an integration for Simli AI avatars.
(see https://www.simli.com)
- Added NVIDIA Riva's `FastPitchTTSService` and `ParakeetSTTService`.
(see https://www.nvidia.com/en-us/ai-data-science/products/riva/)
- Added `IdentityFilter`. This is the simplest frame filter that lets through
all incoming frames.
- New `STTMuteStrategy` called `FUNCTION_CALL` which mutes the STT service
during LLM function calls.
- `DeepgramSTTService` now exposes two event handlers `on_speech_started` and
`on_utterance_end` that could be used to implement interruptions. See new
example `examples/foundational/07c-interruptible-deepgram-vad.py`.
- Added `GroqLLMService`, `GrokLLMService`, and `NimLLMService` for Groq, Grok,
and NVIDIA NIM API integration, with an OpenAI-compatible interface.
- New examples demonstrating function calling with Groq, Grok, Azure OpenAI,
Fireworks, and NVIDIA NIM: `14f-function-calling-groq.py`,
`14g-function-calling-grok.py`, `14h-function-calling-azure.py`,
`14i-function-calling-fireworks.py`, and `14j-function-calling-nvidia.py`.
- In order to obtain the audio stored by the `AudioBufferProcessor` you can now
also register an `on_audio_data` event handler. The `on_audio_data` handler
will be called every time `buffer_size` (a new constructor argument) is
reached. If `buffer_size` is 0 (default) you need to manually get the audio as
before using `AudioBufferProcessor.merge_audio_buffers()`.
```
@audiobuffer.event_handler("on_audio_data")
async def on_audio_data(processor, audio, sample_rate, num_channels):
await save_audio(audio, sample_rate, num_channels)
```
- Added a new RTVI message called `disconnect-bot`, which when handled pushes
an `EndFrame` to trigger the pipeline to stop.
### Changed
- `STTMuteFilter` now supports multiple simultaneous muting strategies.
- `XTTSService` language now defaults to `Language.EN`.
- `SoundfileMixer` doesn't resample input files anymore to avoid startup
delays. The sample rate of the provided sound files now need to match the
sample rate of the output transport.
- Input frames (audio, image and transport messages) are now system frames. This
means they are processed immediately by all processors instead of being queued
internally.
- Expanded the transcriptions.language module to support a superset of
languages.
- Updated STT and TTS services with language options that match the supported
languages for each service.
- Updated the `AzureLLMService` to use the `OpenAILLMService`. Updated the
`api_version` to `2024-09-01-preview`.
- Updated the `FireworksLLMService` to use the `OpenAILLMService`. Updated the
default model to `accounts/fireworks/models/firefunction-v2`.
- Updated the `simple-chatbot` example to include a Javascript and React client
example, using RTVI JS and React.
### Removed
- Removed `AppFrame`. This was used as a special user custom frame, but there's
actually no use case for that.
### Fixed
- Fixed a `ParallelPipeline` issue that would cause system frames to be queued.
- Fixed `FastAPIWebsocketTransport` so it can work with binary data (e.g. using
the protobuf serializer).
- Fixed an issue in `CartesiaTTSService` that could cause previous audio to be
received after an interruption.
- Fixed Cartesia, ElevenLabs, LMNT and PlayHT TTS websocket
reconnection. Before, if an error occurred no reconnection was happening.
- Fixed a `BaseOutputTransport` issue that was causing audio to be discarded
after an `EndFrame` was received.
- Fixed an issue in `WebsocketServerTransport` and `FastAPIWebsocketTransport`
that would cause a busy loop when using audio mixer.
- Fixed a `DailyTransport` and `LiveKitTransport` issue where connections were
being closed in the input transport prematurely. This was causing frames
queued inside the pipeline being discarded.
- Fixed an issue in `DailyTransport` that would cause some internal callbacks to
not be executed.
- Fixed an issue where other frames were being processed while a `CancelFrame`
was being pushed down the pipeline.
- `AudioBufferProcessor` now handles interruptions properly.
- Fixed a `WebsocketServerTransport` issue that would prevent interruptions with
`TwilioSerializer` from working.
- `DailyTransport.capture_participant_video` now allows capturing user's screen
share by simply passing `video_source="screenVideo"`.
- Fixed Google Gemini message handling to properly convert appended messages to
Gemini's required format.
- Fixed an issue with `FireworksLLMService` where chat completions were failing
by removing the `stream_options` from the chat completion options.
- Fixed Google Gemini message handling to properly convert appended messages to Gemini's required format
## [0.0.49] - 2024-11-17

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@@ -55,17 +55,17 @@ pip install "pipecat-ai[option,...]"
Available options include:
| Category | Services | Install Command Example |
| ------------------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- | --------------------------------------- |
| Speech-to-Text | [AssemblyAI](https://docs.pipecat.ai/api-reference/services/stt/assemblyai), [Azure](https://docs.pipecat.ai/api-reference/services/stt/azure), [Deepgram](https://docs.pipecat.ai/api-reference/services/stt/deepgram), [Gladia](https://docs.pipecat.ai/api-reference/services/stt/gladia), [Whisper](https://docs.pipecat.ai/api-reference/services/stt/whisper) | `pip install "pipecat-ai[deepgram]"` |
| LLMs | [Anthropic](https://docs.pipecat.ai/api-reference/services/llm/anthropic), [Azure](https://docs.pipecat.ai/api-reference/services/llm/azure), [Fireworks AI](https://docs.pipecat.ai/api-reference/services/llm/fireworks), [Gemini](https://docs.pipecat.ai/api-reference/services/llm/gemini), [Grok](https://docs.pipecat.ai/api-reference/services/llm/grok), [Groq](https://docs.pipecat.ai/api-reference/services/llm/groq), [NVIDIA NIM](https://docs.pipecat.ai/api-reference/services/llm/nim), [Ollama](https://docs.pipecat.ai/api-reference/services/llm/ollama), [OpenAI](https://docs.pipecat.ai/api-reference/services/llm/openai), [Together AI](https://docs.pipecat.ai/api-reference/services/llm/together) | `pip install "pipecat-ai[openai]"` |
| Text-to-Speech | [AWS](https://docs.pipecat.ai/api-reference/services/tts/aws), [Azure](https://docs.pipecat.ai/api-reference/services/tts/azure), [Cartesia](https://docs.pipecat.ai/api-reference/services/tts/cartesia), [Deepgram](https://docs.pipecat.ai/api-reference/services/tts/deepgram), [ElevenLabs](https://docs.pipecat.ai/api-reference/services/tts/elevenlabs), [Google](https://docs.pipecat.ai/api-reference/services/tts/google), [LMNT](https://docs.pipecat.ai/api-reference/services/tts/lmnt), [OpenAI](https://docs.pipecat.ai/api-reference/services/tts/openai), [PlayHT](https://docs.pipecat.ai/api-reference/services/tts/playht), [Rime](https://docs.pipecat.ai/api-reference/services/tts/rime), [XTTS](https://docs.pipecat.ai/api-reference/services/tts/xtts) | `pip install "pipecat-ai[cartesia]"` |
| Speech-to-Speech | [Gemini Multimodal Live](https://docs.pipecat.ai/server/services/s2s/gemini), [OpenAI Realtime](https://docs.pipecat.ai/api-reference/services/s2s/openai) | `pip install "pipecat-ai[openai]"` |
| Transport | [Daily (WebRTC)](https://docs.pipecat.ai/api-reference/services/transport/daily), WebSocket, Local | `pip install "pipecat-ai[daily]"` |
| Video | [Tavus](https://docs.pipecat.ai/api-reference/services/video/tavus), [Simli](https://docs.pipecat.ai/api-reference/services/video/simli) | `pip install "pipecat-ai[tavus,simli]"` |
| Vision & Image | [Moondream](https://docs.pipecat.ai/api-reference/services/vision/moondream), [fal](https://docs.pipecat.ai/api-reference/services/image-generation/fal) | `pip install "pipecat-ai[moondream]"` |
| Audio Processing | [Silero VAD](https://docs.pipecat.ai/api-reference/utilities/audio/silero-vad-analyzer), [Krisp](https://docs.pipecat.ai/api-reference/utilities/audio/krisp-filter), [Noisereduce](https://docs.pipecat.ai/api-reference/utilities/audio/noisereduce-filter) | `pip install "pipecat-ai[silero]"` |
| Analytics & Metrics | [Canonical AI](https://docs.pipecat.ai/api-reference/services/analytics/canonical), [Sentry](https://docs.pipecat.ai/api-reference/services/analytics/sentry) | `pip install "pipecat-ai[canonical]"` |
| Category | Services | Install Command Example |
| ------------------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- | ------------------------------------- |
| Speech-to-Text | [AssemblyAI](https://docs.pipecat.ai/api-reference/services/stt/assemblyai), [Azure](https://docs.pipecat.ai/api-reference/services/stt/azure), [Deepgram](https://docs.pipecat.ai/api-reference/services/stt/deepgram), [Gladia](https://docs.pipecat.ai/api-reference/services/stt/gladia), [Whisper](https://docs.pipecat.ai/api-reference/services/stt/whisper) | `pip install "pipecat-ai[deepgram]"` |
| LLMs | [Anthropic](https://docs.pipecat.ai/api-reference/services/llm/anthropic), [Azure](https://docs.pipecat.ai/api-reference/services/llm/azure), [Fireworks AI](https://docs.pipecat.ai/api-reference/services/llm/fireworks), [Gemini](https://docs.pipecat.ai/api-reference/services/llm/gemini), [Ollama](https://docs.pipecat.ai/api-reference/services/llm/ollama), [OpenAI](https://docs.pipecat.ai/api-reference/services/llm/openai), [Together AI](https://docs.pipecat.ai/api-reference/services/llm/together) | `pip install "pipecat-ai[openai]"` |
| Text-to-Speech | [AWS](https://docs.pipecat.ai/api-reference/services/tts/aws), [Azure](https://docs.pipecat.ai/api-reference/services/tts/azure), [Cartesia](https://docs.pipecat.ai/api-reference/services/tts/cartesia), [Deepgram](https://docs.pipecat.ai/api-reference/services/tts/deepgram), [ElevenLabs](https://docs.pipecat.ai/api-reference/services/tts/elevenlabs), [Google](https://docs.pipecat.ai/api-reference/services/tts/google), [LMNT](https://docs.pipecat.ai/api-reference/services/tts/lmnt), [OpenAI](https://docs.pipecat.ai/api-reference/services/tts/openai), [PlayHT](https://docs.pipecat.ai/api-reference/services/tts/playht), [Rime](https://docs.pipecat.ai/api-reference/services/tts/rime), [XTTS](https://docs.pipecat.ai/api-reference/services/tts/xtts) | `pip install "pipecat-ai[cartesia]"` |
| Speech-to-Speech | [OpenAI Realtime](https://docs.pipecat.ai/api-reference/services/s2s/openai) | `pip install "pipecat-ai[openai]"` |
| Transport | [Daily (WebRTC)](https://docs.pipecat.ai/api-reference/services/transport/daily), WebSocket, Local | `pip install "pipecat-ai[daily]"` |
| Video | [Tavus](https://docs.pipecat.ai/api-reference/services/video/tavus) | `pip install "pipecat-ai[tavus]"` |
| Vision & Image | [Moondream](https://docs.pipecat.ai/api-reference/services/vision/moondream), [fal](https://docs.pipecat.ai/api-reference/services/image-generation/fal) | `pip install "pipecat-ai[moondream]"` |
| Audio Processing | [Silero VAD](https://docs.pipecat.ai/api-reference/utilities/audio/silero-vad-analyzer), [Krisp](https://docs.pipecat.ai/api-reference/utilities/audio/krisp-filter), [Noisereduce](https://docs.pipecat.ai/api-reference/utilities/audio/noisereduce-filter) | `pip install "pipecat-ai[silero]"` |
| Analytics & Metrics | [Canonical AI](https://docs.pipecat.ai/api-reference/services/analytics/canonical), [Sentry](https://docs.pipecat.ai/api-reference/services/analytics/sentry) | `pip install "pipecat-ai[canonical]"` |
📚 [View full services documentation →](https://docs.pipecat.ai/api-reference/services/supported-services)

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@@ -1,5 +1,5 @@
build~=1.2.1
grpcio-tools~=1.65.4
grpcio-tools~=1.62.2
pip-tools~=7.4.1
pyright~=1.1.376
pytest~=8.3.2

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@@ -1,20 +0,0 @@
# Minimal makefile for Sphinx documentation
#
# You can set these variables from the command line, and also
# from the environment for the first two.
SPHINXOPTS ?=
SPHINXBUILD ?= sphinx-build
SOURCEDIR = .
BUILDDIR = _build
# Put it first so that "make" without argument is like "make help".
help:
@$(SPHINXBUILD) -M help "$(SOURCEDIR)" "$(BUILDDIR)" $(SPHINXOPTS) $(O)
.PHONY: help Makefile
# Catch-all target: route all unknown targets to Sphinx using the new
# "make mode" option. $(O) is meant as a shortcut for $(SPHINXOPTS).
%: Makefile
@$(SPHINXBUILD) -M $@ "$(SOURCEDIR)" "$(BUILDDIR)" $(SPHINXOPTS) $(O)

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@@ -1,109 +0,0 @@
# Pipecat Documentation
This directory contains the source files for auto-generating Pipecat's server API reference documentation.
## Setup
1. Install documentation dependencies:
```bash
pip install -r requirements.txt
```
2. Make the build scripts executable:
```bash
chmod +x build-docs.sh rtd-test.py
```
## Building Documentation
From this directory, you can build the documentation in several ways:
### Local Build
```bash
# Using the build script (automatically opens docs when done)
./build-docs.sh
# Or directly with sphinx-build
sphinx-build -b html . _build/html -W --keep-going
```
### ReadTheDocs Test Build
To test the documentation build process exactly as it would run on ReadTheDocs:
```bash
./rtd-test.py
```
This script:
- Creates a fresh virtual environment
- Installs all dependencies as specified in requirements files
- Handles conflicting dependencies (like grpcio versions for Riva and PlayHT)
- Builds the documentation in an isolated environment
- Provides detailed logging of the build process
Use this script to verify your documentation will build correctly on ReadTheDocs before pushing changes.
## Viewing Documentation
The built documentation will be available at `_build/html/index.html`. To open:
```bash
# On MacOS
open _build/html/index.html
# On Linux
xdg-open _build/html/index.html
# On Windows
start _build/html/index.html
```
## Directory Structure
```
.
├── api/ # Auto-generated API documentation
├── _build/ # Built documentation
├── _static/ # Static files (images, css, etc.)
├── conf.py # Sphinx configuration
├── index.rst # Main documentation entry point
├── requirements-base.txt # Base documentation dependencies
├── requirements-riva.txt # Riva-specific dependencies
├── requirements-playht.txt # PlayHT-specific dependencies
├── build-docs.sh # Local build script
└── rtd-test.py # ReadTheDocs test build script
```
## Notes
- Documentation is auto-generated from Python docstrings
- Service modules are automatically detected and included
- The build process matches our ReadTheDocs configuration
- Warnings are treated as errors (-W flag) to maintain consistency
- The --keep-going flag ensures all errors are reported
- Dependencies are split into multiple requirements files to handle version conflicts
## Troubleshooting
If you encounter missing service modules:
1. Verify the service is installed with its extras: `pip install pipecat-ai[service-name]`
2. Check the build logs for import errors
3. Ensure the service module is properly initialized in the package
4. Run `./rtd-test.py` to test in an isolated environment matching ReadTheDocs
For dependency conflicts:
1. Check the requirements files for version specifications
2. Use `rtd-test.py` to verify dependency resolution
3. Consider adding service-specific requirements files if needed
For more information:
- [ReadTheDocs Configuration](.readthedocs.yaml)
- [Sphinx Documentation](https://www.sphinx-doc.org/)

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@@ -1,10 +0,0 @@
#!/bin/bash
# Clean previous build
rm -rf _build
# Build docs matching ReadTheDocs configuration
sphinx-build -b html -d _build/doctrees . _build/html -W --keep-going
# Open docs (MacOS)
open _build/html/index.html

View File

@@ -1,252 +0,0 @@
import logging
import sys
from pathlib import Path
# Configure logging
logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
logger = logging.getLogger("sphinx-build")
# Add source directory to path
docs_dir = Path(__file__).parent
project_root = docs_dir.parent.parent
sys.path.insert(0, str(project_root / "src"))
# Project information
project = "pipecat-ai"
copyright = "2024, Daily"
author = "Daily"
# General configuration
extensions = [
"sphinx.ext.autodoc",
"sphinx.ext.napoleon",
"sphinx.ext.viewcode",
"sphinx.ext.intersphinx",
]
# Napoleon settings
napoleon_google_docstring = True
napoleon_numpy_docstring = False
napoleon_include_init_with_doc = True
# AutoDoc settings
autodoc_default_options = {
"members": True,
"member-order": "bysource",
"special-members": "__init__",
"undoc-members": True,
"exclude-members": "__weakref__",
"no-index": True,
"show-inheritance": True,
}
# Mock imports for optional dependencies
autodoc_mock_imports = [
"riva",
"livekit",
"pyht", # Base PlayHT package
"pyht.async_client", # PlayHT specific imports
"pyht.client",
"pyht.protos",
"pyht.protos.api_pb2",
"pipecat_ai_playht", # PlayHT wrapper
"anthropic",
"assemblyai",
"boto3",
"azure",
"cartesia",
"deepgram",
"elevenlabs",
"fal",
"gladia",
"google",
"krisp",
"langchain",
"lmnt",
"noisereduce",
"openai",
"openpipe",
"simli",
"soundfile",
# Existing mocks
"pipecat_ai_krisp",
"pyaudio",
"_tkinter",
"tkinter",
"daily",
"daily_python",
"pydantic.BaseModel",
"pydantic.Field",
"pydantic._internal._model_construction",
"pydantic._internal._fields",
]
# HTML output settings
html_theme = "sphinx_rtd_theme"
html_static_path = ["_static"]
autodoc_typehints = "description"
html_show_sphinx = False
def verify_modules():
"""Verify that required modules are available."""
required_modules = {
"services": [
"assemblyai",
"aws",
"cartesia",
"deepgram",
"google",
"lmnt",
"riva",
"simli",
],
"serializers": ["livekit"],
"vad": ["silero", "vad_analyzer"],
"transports": {
"services": ["daily", "livekit"],
"local": ["audio", "tk"],
"network": ["fastapi_websocket", "websocket_server"],
},
}
missing = []
for category, modules in required_modules.items():
if isinstance(modules, dict):
# Handle nested structure
for subcategory, submodules in modules.items():
for module in submodules:
try:
__import__(f"pipecat.{category}.{subcategory}.{module}")
logger.info(
f"Successfully imported pipecat.{category}.{subcategory}.{module}"
)
except (ImportError, TypeError, NameError) as e:
missing.append(f"pipecat.{category}.{subcategory}.{module}")
logger.warning(
f"Optional module not available: pipecat.{category}.{subcategory}.{module} - {str(e)}"
)
else:
# Handle flat structure
for module in modules:
try:
__import__(f"pipecat.{category}.{module}")
logger.info(f"Successfully imported pipecat.{category}.{module}")
except (ImportError, TypeError, NameError) as e:
missing.append(f"pipecat.{category}.{module}")
logger.warning(
f"Optional module not available: pipecat.{category}.{module} - {str(e)}"
)
if missing:
logger.warning(f"Some optional modules are not available: {missing}")
def clean_title(title: str) -> str:
"""Automatically clean module titles."""
# Remove everything after space (like 'module', 'processor', etc.)
title = title.split(" ")[0]
# Get the last part of the dot-separated path
parts = title.split(".")
title = parts[-1]
# Special cases for service names and common acronyms
special_cases = {
"ai": "AI",
"aws": "AWS",
"api": "API",
"vad": "VAD",
"assemblyai": "AssemblyAI",
"deepgram": "Deepgram",
"elevenlabs": "ElevenLabs",
"openai": "OpenAI",
"openpipe": "OpenPipe",
"playht": "PlayHT",
"xtts": "XTTS",
"lmnt": "LMNT",
}
# Check if the entire title is a special case
if title.lower() in special_cases:
return special_cases[title.lower()]
# Otherwise, capitalize each word
words = title.split("_")
cleaned_words = []
for word in words:
if word.lower() in special_cases:
cleaned_words.append(special_cases[word.lower()])
else:
cleaned_words.append(word.capitalize())
return " ".join(cleaned_words)
def setup(app):
"""Generate API documentation during Sphinx build."""
from sphinx.ext.apidoc import main
docs_dir = Path(__file__).parent
project_root = docs_dir.parent.parent
output_dir = str(docs_dir / "api")
source_dir = str(project_root / "src" / "pipecat")
# Clean existing files
if Path(output_dir).exists():
import shutil
shutil.rmtree(output_dir)
logger.info(f"Cleaned existing documentation in {output_dir}")
logger.info(f"Generating API documentation...")
logger.info(f"Output directory: {output_dir}")
logger.info(f"Source directory: {source_dir}")
excludes = [
str(project_root / "src/pipecat/pipeline/to_be_updated"),
str(project_root / "src/pipecat/processors/gstreamer"),
str(project_root / "src/pipecat/services/to_be_updated"),
str(project_root / "src/pipecat/vad"), # deprecated
"**/test_*.py",
"**/tests/*.py",
]
try:
main(
[
"-f", # Force overwriting
"-e", # Don't generate empty files
"-M", # Put module documentation before submodule documentation
"--no-toc", # Don't create a table of contents file
"--separate", # Put documentation for each module in its own page
"--module-first", # Module documentation before submodule documentation
"--implicit-namespaces", # Added: Handle implicit namespace packages
"-o",
output_dir,
source_dir,
]
+ excludes
)
logger.info("API documentation generated successfully!")
# Process generated RST files to update titles
for rst_file in Path(output_dir).glob("**/*.rst"): # Changed to recursive glob
content = rst_file.read_text()
lines = content.split("\n")
# Find and clean up the title
if lines and "=" in lines[1]: # Title is typically the first line
old_title = lines[0]
new_title = clean_title(old_title)
content = content.replace(old_title, new_title)
rst_file.write_text(content)
logger.info(f"Updated title: {old_title} -> {new_title}")
except Exception as e:
logger.error(f"Error generating API documentation: {e}", exc_info=True)
# Run module verification
verify_modules()

View File

@@ -1,77 +0,0 @@
Pipecat API Reference Docs
==========================
Welcome to Pipecat's API reference documentation!
Pipecat is an open source framework for building voice and multimodal assistants.
It provides a flexible pipeline architecture for connecting various AI services,
audio processing, and transport layers.
Quick Links
-----------
* `GitHub Repository <https://github.com/pipecat-ai/pipecat>`_
* `Website <https://pipecat.ai>`_
API Reference
-------------
Core Components
~~~~~~~~~~~~~~~
* :mod:`Frames <pipecat.frames>`
* :mod:`Processors <pipecat.processors>`
* :mod:`Pipeline <pipecat.pipeline>`
Audio Processing
~~~~~~~~~~~~~~~~
* :mod:`Audio <pipecat.audio>`
Services
~~~~~~~~
* :mod:`Services <pipecat.services>`
Transport & Serialization
~~~~~~~~~~~~~~~~~~~~~~~~~
* :mod:`Transports <pipecat.transports>`
* :mod:`Local <pipecat.transports.local>`
* :mod:`Network <pipecat.transports.network>`
* :mod:`Services <pipecat.transports.services>`
* :mod:`Serializers <pipecat.serializers>`
Utilities
~~~~~~~~~
* :mod:`Clocks <pipecat.clocks>`
* :mod:`Metrics <pipecat.metrics>`
* :mod:`Sync <pipecat.sync>`
* :mod:`Transcriptions <pipecat.transcriptions>`
* :mod:`Utils <pipecat.utils>`
.. toctree::
:maxdepth: 3
:caption: API Reference
:hidden:
Audio <api/pipecat.audio>
Clocks <api/pipecat.clocks>
Frames <api/pipecat.frames>
Metrics <api/pipecat.metrics>
Pipeline <api/pipecat.pipeline>
Processors <api/pipecat.processors>
Serializers <api/pipecat.serializers>
Services <api/pipecat.services>
Sync <api/pipecat.sync>
Transcriptions <api/pipecat.transcriptions>
Transports <api/pipecat.transports>
Utils <api/pipecat.utils>
Indices and tables
==================
* :ref:`genindex`
* :ref:`modindex`
* :ref:`search`

View File

@@ -1,35 +0,0 @@
@ECHO OFF
pushd %~dp0
REM Command file for Sphinx documentation
if "%SPHINXBUILD%" == "" (
set SPHINXBUILD=sphinx-build
)
set SOURCEDIR=.
set BUILDDIR=_build
%SPHINXBUILD% >NUL 2>NUL
if errorlevel 9009 (
echo.
echo.The 'sphinx-build' command was not found. Make sure you have Sphinx
echo.installed, then set the SPHINXBUILD environment variable to point
echo.to the full path of the 'sphinx-build' executable. Alternatively you
echo.may add the Sphinx directory to PATH.
echo.
echo.If you don't have Sphinx installed, grab it from
echo.https://www.sphinx-doc.org/
exit /b 1
)
if "%1" == "" goto help
%SPHINXBUILD% -M %1 %SOURCEDIR% %BUILDDIR% %SPHINXOPTS% %O%
goto end
:help
%SPHINXBUILD% -M help %SOURCEDIR% %BUILDDIR% %SPHINXOPTS% %O%
:end
popd

View File

@@ -1,40 +0,0 @@
# Sphinx dependencies
sphinx>=8.1.3
sphinx-rtd-theme
sphinx-markdown-builder
sphinx-autodoc-typehints
toml
# Install all extras individually to ensure they're properly resolved
pipecat-ai[anthropic]
pipecat-ai[assemblyai]
pipecat-ai[aws]
pipecat-ai[azure]
pipecat-ai[canonical]
pipecat-ai[cartesia]
pipecat-ai[daily]
pipecat-ai[deepgram]
pipecat-ai[elevenlabs]
pipecat-ai[fal]
pipecat-ai[fireworks]
pipecat-ai[gladia]
pipecat-ai[google]
pipecat-ai[grok]
pipecat-ai[groq]
# pipecat-ai[krisp] # Mocked instead
pipecat-ai[langchain]
pipecat-ai[livekit]
pipecat-ai[lmnt]
pipecat-ai[local]
pipecat-ai[moondream]
pipecat-ai[nim]
pipecat-ai[noisereduce]
pipecat-ai[openai]
# pipecat-ai[openpipe]
# pipecat-ai[playht] # Mocked due to grpcio conflict with riva
pipecat-ai[riva]
pipecat-ai[silero]
pipecat-ai[simli]
pipecat-ai[soundfile]
pipecat-ai[websocket]
pipecat-ai[whisper]

View File

@@ -1,38 +0,0 @@
#!/bin/bash
set -e
# Configuration
DOCS_DIR=$(pwd)
PROJECT_ROOT=$(cd ../../ && pwd)
TEST_DIR="/tmp/rtd-test-$(date +%Y%m%d_%H%M%S)"
echo "Creating test directory: $TEST_DIR"
mkdir -p "$TEST_DIR"
cd "$TEST_DIR"
# Create virtual environment
python -m venv venv
source venv/bin/activate
echo "Installing build dependencies..."
pip install --upgrade pip wheel setuptools
echo "Installing documentation dependencies..."
pip install -r "$DOCS_DIR/requirements.txt"
echo "Building documentation..."
cd "$DOCS_DIR"
sphinx-build -b html . "_build/html"
echo "Build complete. Check _build/html directory for output."
# Print summary
echo -e "\n=== Build Summary ==="
echo "Documentation: $DOCS_DIR/_build/html"
echo "Test environment: $TEST_DIR"
echo -e "\nTo view the documentation:"
echo "open $DOCS_DIR/_build/html/index.html"
# Print installed packages for verification
echo -e "\n=== Installed Packages ==="
pip freeze | grep -E "sphinx|pipecat"

View File

@@ -96,6 +96,9 @@ Notable control frames:
## 7. Special Purpose Frames
### AppFrame
Base class for application-specific custom frames.
### MetricsFrame
Contains performance metrics data.

View File

@@ -54,9 +54,5 @@ TAVUS_API_KEY=...
TAVUS_REPLICA_ID=...
TAVUS_PERSONA_ID=...
# Simli
SIMLI_API_KEY=...
SIMLI_FACE_ID=...
# Krisp
KRISP_MODEL_PATH=...
#Krisp
KRISP_MODEL_PATH=...

View File

@@ -102,6 +102,7 @@ async def main():
audio_buffer_processor=audio_buffer_processor,
aiohttp_session=session,
api_key=os.getenv("CANONICAL_API_KEY"),
api_url=os.getenv("CANONICAL_API_URL"),
call_id=str(uuid.uuid4()),
assistant="pipecat-chatbot",
assistant_speaks_first=True,

View File

@@ -4,9 +4,7 @@
# SPDX-License-Identifier: BSD 2-Clause License
#
import aiofiles
import asyncio
import io
import os
import sys
@@ -34,17 +32,15 @@ logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def save_audio(audio: bytes, sample_rate: int, num_channels: int):
if len(audio) > 0:
async def save_audio(audiobuffer):
if audiobuffer.has_audio():
merged_audio = audiobuffer.merge_audio_buffers()
filename = f"conversation_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())
with wave.open(filename, "wb") as wf:
wf.setnchannels(2)
wf.setsampwidth(2)
wf.setframerate(audiobuffer._sample_rate)
wf.writeframes(merged_audio)
print(f"Merged audio saved to {filename}")
else:
print("No audio data to save")
@@ -110,9 +106,7 @@ async def main():
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
# Save audio every 10 seconds.
audiobuffer = AudioBufferProcessor(buffer_size=480000)
audiobuffer = AudioBufferProcessor()
pipeline = Pipeline(
[
transport.input(), # microphone
@@ -127,10 +121,6 @@ async def main():
task = PipelineTask(pipeline, PipelineParams(allow_interruptions=True))
@audiobuffer.event_handler("on_audio_data")
async def on_audio_data(buffer, audio, sample_rate, num_channels):
await save_audio(audio, sample_rate, num_channels)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
@@ -140,6 +130,7 @@ async def main():
async def on_participant_left(transport, participant, reason):
print(f"Participant left: {participant}")
await task.queue_frame(EndFrame())
await save_audio(audiobuffer)
runner = PipelineRunner()

View File

@@ -1,4 +1,3 @@
aiofiles
python-dotenv
fastapi[all]
uvicorn

View File

@@ -2,4 +2,4 @@ python-dotenv==1.0.1
modal==0.65.48
pipecat-ai[daily,silero,cartesia,openai]==0.0.48
fastapi==0.115.4
aiohttp==3.11.9
aiohttp==3.10.10

View File

@@ -1,56 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import aiohttp
import os
import sys
from pipecat.frames.frames import EndFrame, TTSSpeakFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.task import PipelineTask
from pipecat.pipeline.runner import PipelineRunner
from pipecat.services.riva import FastPitchTTSService
from pipecat.transports.services.daily import DailyParams, DailyTransport
from runner import configure
from loguru import logger
from dotenv import load_dotenv
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
(room_url, _) = await configure(session)
transport = DailyTransport(
room_url, None, "Say One Thing", DailyParams(audio_out_enabled=True)
)
tts = FastPitchTTSService(api_key=os.getenv("NVIDIA_API_KEY"))
runner = PipelineRunner()
task = PipelineTask(Pipeline([tts, transport.output()]))
# Register an event handler so we can play the audio when the
# participant joins.
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
participant_name = participant.get("info", {}).get("userName", "")
await task.queue_frames([TTSSpeakFrame(f"Aloha, {participant_name}!"), EndFrame()])
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -12,7 +12,7 @@ import sys
from dataclasses import dataclass
from pipecat.frames.frames import (
DataFrame,
AppFrame,
Frame,
LLMFullResponseStartFrame,
LLMMessagesFrame,
@@ -42,7 +42,7 @@ logger.add(sys.stderr, level="DEBUG")
@dataclass
class MonthFrame(DataFrame):
class MonthFrame(AppFrame):
month: str
def __str__(self):
@@ -56,6 +56,8 @@ class MonthPrepender(FrameProcessor):
self.prepend_to_next_text_frame = False
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, MonthFrame):
self.most_recent_month = frame.month
elif self.prepend_to_next_text_frame and isinstance(frame, TextFrame):

View File

@@ -62,6 +62,8 @@ async def main():
self.text = ""
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TextFrame):
self.text = frame.text
await self.push_frame(frame, direction)
@@ -73,6 +75,8 @@ async def main():
self.frame = None
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TTSAudioRawFrame):
self.audio.extend(frame.audio)
self.frame = OutputAudioRawFrame(
@@ -86,6 +90,8 @@ async def main():
self.frame = None
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, URLImageRawFrame):
self.frame = frame
await self.push_frame(frame, direction)

View File

@@ -47,6 +47,8 @@ class ImageSyncAggregator(FrameProcessor):
self._waiting_image_bytes = self._waiting_image.tobytes()
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if not isinstance(frame, SystemFrame) and direction == FrameDirection.DOWNSTREAM:
await self.push_frame(
OutputImageRawFrame(

View File

@@ -1,105 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from deepgram import LiveOptions
from dotenv import load_dotenv
from loguru import logger
from runner import configure
from pipecat.frames.frames import (
BotInterruptionFrame,
LLMMessagesFrame,
StopInterruptionFrame,
UserStartedSpeakingFrame,
UserStoppedSpeakingFrame,
)
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.services.deepgram import DeepgramSTTService, DeepgramTTSService
from pipecat.services.openai import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
(room_url, _) = await configure(session)
transport = DailyTransport(
room_url,
None,
"Respond bot",
DailyParams(
audio_in_enabled=True,
audio_out_enabled=True,
),
)
stt = DeepgramSTTService(
api_key=os.getenv("DEEPGRAM_API_KEY"),
live_options=LiveOptions(vad_events=True, utterance_end_ms="1000"),
)
tts = DeepgramTTSService(api_key=os.getenv("DEEPGRAM_API_KEY"), voice="aura-helios-en")
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
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.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(), # Transport user input
stt, # STT
context_aggregator.user(), # User responses
llm, # LLM
tts, # TTS
transport.output(), # Transport bot output
context_aggregator.assistant(), # Assistant spoken responses
]
)
task = PipelineTask(pipeline, PipelineParams(allow_interruptions=True))
@stt.event_handler("on_speech_started")
async def on_speech_started(stt, *args, **kwargs):
await task.queue_frames([BotInterruptionFrame(), UserStartedSpeakingFrame()])
@stt.event_handler("on_utterance_end")
async def on_utterance_end(stt, *args, **kwargs):
await task.queue_frames([StopInterruptionFrame(), UserStoppedSpeakingFrame()])
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
# Kick off the conversation.
messages.append({"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([LLMMessagesFrame(messages)])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -50,6 +50,7 @@ async def main():
tts = XTTSService(
aiohttp_session=session,
voice_id="Claribel Dervla",
language="en",
base_url="http://localhost:8000",
)

View File

@@ -19,7 +19,7 @@ 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.services.aws import PollyTTSService
from pipecat.services.aws import AWSTTSService
from pipecat.services.deepgram import DeepgramSTTService
from pipecat.services.openai import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
@@ -48,12 +48,12 @@ async def main():
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
tts = PollyTTSService(
tts = AWSTTSService(
api_key=os.getenv("AWS_SECRET_ACCESS_KEY"),
aws_access_key_id=os.getenv("AWS_ACCESS_KEY_ID"),
region=os.getenv("AWS_REGION"),
voice_id="Amy",
params=PollyTTSService.InputParams(engine="neural", language="en-GB", rate="1.05"),
params=AWSTTSService.InputParams(engine="neural", language="en-GB", rate="1.05"),
)
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")

View File

@@ -82,6 +82,8 @@ class UserAudioCollector(FrameProcessor):
self._user_speaking = False
async def process_frame(self, frame, direction):
await super().process_frame(frame, direction)
if isinstance(frame, TranscriptionFrame):
# We could gracefully handle both audio input and text/transcription input ...
# but let's leave that as an exercise to the reader. :-)
@@ -124,6 +126,7 @@ class TranscriptExtractor(FrameProcessor):
self._accumulating_transcript = False
async def process_frame(self, frame, direction):
await super().process_frame(frame, direction)
if isinstance(frame, LLMFullResponseStartFrame):
self._processing_llm_response = True
self._accumulating_transcript = True
@@ -177,6 +180,8 @@ class TanscriptionContextFixup(FrameProcessor):
self._context.messages[-1].parts[-1].text += f"\n\n{marker}\n{self._transcript}\n"
async def process_frame(self, frame, direction):
await super().process_frame(frame, direction)
if isinstance(frame, MagicDemoTranscriptionFrame):
self._transcript = frame.text
elif isinstance(frame, LLMFullResponseEndFrame) or isinstance(

View File

@@ -1,92 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import LLMMessagesFrame
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.services.nim import NimLLMService
from pipecat.services.riva import FastPitchTTSService, ParakeetSTTService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
(room_url, _) = await configure(session)
transport = DailyTransport(
room_url,
None,
"Respond bot",
DailyParams(
audio_out_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True,
),
)
stt = ParakeetSTTService(api_key=os.getenv("NVIDIA_API_KEY"))
llm = NimLLMService(
api_key=os.getenv("NVIDIA_API_KEY"), model="meta/llama-3.1-405b-instruct"
)
tts = FastPitchTTSService(api_key=os.getenv("NVIDIA_API_KEY"))
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.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(), # Transport user input
stt, # STT
context_aggregator.user(), # User responses
llm, # LLM
tts, # TTS
transport.output(), # Transport bot output
context_aggregator.assistant(), # Assistant spoken responses
]
)
task = PipelineTask(pipeline, PipelineParams(allow_interruptions=True))
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
# Kick off the conversation.
messages.append({"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([LLMMessagesFrame(messages)])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -35,6 +35,8 @@ 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(

View File

@@ -39,6 +39,8 @@ 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(

View File

@@ -14,18 +14,16 @@ from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
Frame,
LLMFullResponseEndFrame,
LLMMessagesFrame,
OutputAudioRawFrame,
)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineTask
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.logger import FrameLogger
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.cartesia import CartesiaHttpTTSService
from pipecat.services.openai import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
@@ -60,6 +58,8 @@ for file in sound_files:
class OutboundSoundEffectWrapper(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, LLMFullResponseEndFrame):
await self.push_frame(sounds["ding1.wav"])
# In case anything else downstream needs it
@@ -70,7 +70,9 @@ class OutboundSoundEffectWrapper(FrameProcessor):
class InboundSoundEffectWrapper(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
if isinstance(frame, OpenAILLMContextFrame):
await super().process_frame(frame, direction)
if isinstance(frame, LLMMessagesFrame):
await self.push_frame(sounds["ding2.wav"])
# In case anything else downstream needs it
await self.push_frame(frame, direction)
@@ -96,7 +98,7 @@ async def main():
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
tts = CartesiaTTSService(
tts = CartesiaHttpTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)

View File

@@ -42,6 +42,8 @@ class UserImageRequester(FrameProcessor):
self._participant_id = participant_id
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if self._participant_id and isinstance(frame, TextFrame):
await self.push_frame(
UserImageRequestFrame(self._participant_id), FrameDirection.UPSTREAM

View File

@@ -42,6 +42,8 @@ class UserImageRequester(FrameProcessor):
self._participant_id = participant_id
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if self._participant_id and isinstance(frame, TextFrame):
await self.push_frame(
UserImageRequestFrame(self._participant_id), FrameDirection.UPSTREAM

View File

@@ -42,6 +42,8 @@ class UserImageRequester(FrameProcessor):
self._participant_id = participant_id
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if self._participant_id and isinstance(frame, TextFrame):
await self.push_frame(
UserImageRequestFrame(self._participant_id), FrameDirection.UPSTREAM

View File

@@ -42,6 +42,8 @@ class UserImageRequester(FrameProcessor):
self._participant_id = participant_id
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if self._participant_id and isinstance(frame, TextFrame):
await self.push_frame(
UserImageRequestFrame(self._participant_id), FrameDirection.UPSTREAM

View File

@@ -30,6 +30,8 @@ logger.add(sys.stderr, level="DEBUG")
class TranscriptionLogger(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TranscriptionFrame):
print(f"Transcription: {frame.text}")

View File

@@ -28,6 +28,8 @@ logger.add(sys.stderr, level="DEBUG")
class TranscriptionLogger(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TranscriptionFrame):
print(f"Transcription: {frame.text}")

View File

@@ -31,6 +31,8 @@ logger.add(sys.stderr, level="DEBUG")
class TranscriptionLogger(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TranscriptionFrame):
print(f"Transcription: {frame.text}")

View File

@@ -29,6 +29,8 @@ logger.add(sys.stderr, level="DEBUG")
class TranscriptionLogger(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TranscriptionFrame):
print(f"Transcription: {frame.text}")

View File

@@ -29,6 +29,8 @@ logger.add(sys.stderr, level="DEBUG")
class TranscriptionLogger(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TranscriptionFrame):
print(f"Transcription: {frame.text}")

View File

@@ -5,15 +5,10 @@
#
import asyncio
import aiohttp
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
@@ -23,6 +18,14 @@ from pipecat.services.openai import OpenAILLMContext
from pipecat.services.together import TogetherLLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
from openai.types.chat import ChatCompletionToolParam
from runner import configure
from loguru import logger
from dotenv import load_dotenv
load_dotenv(override=True)
logger.remove(0)
@@ -122,7 +125,7 @@ async def main():
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
# Kick off the conversation.
await task.queue_frames([context_aggregator.user().get_context_frame()])
# await tts.say("Hi! Ask me about the weather in San Francisco.")
runner = PipelineRunner()

View File

@@ -1,139 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.groq import GroqLLMService
from pipecat.services.openai import OpenAILLMContext
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def start_fetch_weather(function_name, llm, context):
# note: we can't push a frame to the LLM here. the bot
# can interrupt itself and/or cause audio overlapping glitches.
# possible question for Aleix and Chad about what the right way
# to trigger speech is, now, with the new queues/async/sync refactors.
# await llm.push_frame(TextFrame("Let me check on that."))
logger.debug(f"Starting fetch_weather_from_api with function_name: {function_name}")
async def fetch_weather_from_api(function_name, tool_call_id, args, llm, context, result_callback):
await result_callback({"conditions": "nice", "temperature": "75"})
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_out_enabled=True,
transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)
llm = GroqLLMService(
api_key=os.getenv("GROQ_API_KEY"), model="llama3-groq-70b-8192-tool-use-preview"
)
# Register a function_name of None to get all functions
# sent to the same callback with an additional function_name parameter.
llm.register_function(None, fetch_weather_from_api, start_callback=start_fetch_weather)
tools = [
ChatCompletionToolParam(
type="function",
function={
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"unit": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location"],
},
},
)
]
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.",
},
]
context = OpenAILLMContext(messages, tools)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
# Kick off the conversation.
await task.queue_frames([context_aggregator.user().get_context_frame()])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,137 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.grok import GrokLLMService
from pipecat.services.openai import OpenAILLMContext
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def start_fetch_weather(function_name, llm, context):
# note: we can't push a frame to the LLM here. the bot
# can interrupt itself and/or cause audio overlapping glitches.
# possible question for Aleix and Chad about what the right way
# to trigger speech is, now, with the new queues/async/sync refactors.
# await llm.push_frame(TextFrame("Let me check on that."))
logger.debug(f"Starting fetch_weather_from_api with function_name: {function_name}")
async def fetch_weather_from_api(function_name, tool_call_id, args, llm, context, result_callback):
await result_callback({"conditions": "nice", "temperature": "75"})
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_out_enabled=True,
transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)
llm = GrokLLMService(api_key=os.getenv("GROK_API_KEY"))
# Register a function_name of None to get all functions
# sent to the same callback with an additional function_name parameter.
llm.register_function(None, fetch_weather_from_api, start_callback=start_fetch_weather)
tools = [
ChatCompletionToolParam(
type="function",
function={
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location", "format"],
},
},
)
]
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.",
},
]
context = OpenAILLMContext(messages, tools)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
# Kick off the conversation.
await task.queue_frames([context_aggregator.user().get_context_frame()])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,141 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.services.azure import AzureLLMService
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.openai import OpenAILLMContext
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def start_fetch_weather(function_name, llm, context):
# note: we can't push a frame to the LLM here. the bot
# can interrupt itself and/or cause audio overlapping glitches.
# possible question for Aleix and Chad about what the right way
# to trigger speech is, now, with the new queues/async/sync refactors.
# await llm.push_frame(TextFrame("Let me check on that."))
logger.debug(f"Starting fetch_weather_from_api with function_name: {function_name}")
async def fetch_weather_from_api(function_name, tool_call_id, args, llm, context, result_callback):
await result_callback({"conditions": "nice", "temperature": "75"})
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_out_enabled=True,
transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)
llm = AzureLLMService(
api_key=os.getenv("AZURE_CHATGPT_API_KEY"),
endpoint=os.getenv("AZURE_CHATGPT_ENDPOINT"),
model=os.getenv("AZURE_CHATGPT_MODEL"),
)
# Register a function_name of None to get all functions
# sent to the same callback with an additional function_name parameter.
llm.register_function(None, fetch_weather_from_api, start_callback=start_fetch_weather)
tools = [
ChatCompletionToolParam(
type="function",
function={
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location", "format"],
},
},
)
]
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.",
},
]
context = OpenAILLMContext(messages, tools)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
# Kick off the conversation.
await task.queue_frames([context_aggregator.user().get_context_frame()])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,140 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.fireworks import FireworksLLMService
from pipecat.services.openai import OpenAILLMContext
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def start_fetch_weather(function_name, llm, context):
# note: we can't push a frame to the LLM here. the bot
# can interrupt itself and/or cause audio overlapping glitches.
# possible question for Aleix and Chad about what the right way
# to trigger speech is, now, with the new queues/async/sync refactors.
# await llm.push_frame(TextFrame("Let me check on that."))
logger.debug(f"Starting fetch_weather_from_api with function_name: {function_name}")
async def fetch_weather_from_api(function_name, tool_call_id, args, llm, context, result_callback):
await result_callback({"conditions": "nice", "temperature": "75"})
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_out_enabled=True,
transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)
llm = FireworksLLMService(
api_key=os.getenv("FIREWORKS_API_KEY"),
model="accounts/fireworks/models/firefunction-v2",
)
# Register a function_name of None to get all functions
# sent to the same callback with an additional function_name parameter.
llm.register_function(None, fetch_weather_from_api, start_callback=start_fetch_weather)
tools = [
ChatCompletionToolParam(
type="function",
function={
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location", "format"],
},
},
)
]
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.",
},
]
context = OpenAILLMContext(messages, tools)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
# Kick off the conversation.
await task.queue_frames([context_aggregator.user().get_context_frame()])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,140 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.nim import NimLLMService
from pipecat.services.openai import OpenAILLMContext
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def start_fetch_weather(function_name, llm, context):
# note: we can't push a frame to the LLM here. the bot
# can interrupt itself and/or cause audio overlapping glitches.
# possible question for Aleix and Chad about what the right way
# to trigger speech is, now, with the new queues/async/sync refactors.
# await llm.push_frame(TextFrame("Let me check on that."))
logger.debug(f"Starting fetch_weather_from_api with function_name: {function_name}")
async def fetch_weather_from_api(function_name, tool_call_id, args, llm, context, result_callback):
await result_callback({"conditions": "nice", "temperature": "75"})
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_out_enabled=True,
transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
# text_filter=MarkdownTextFilter(),
)
llm = NimLLMService(
api_key=os.getenv("NVIDIA_API_KEY"), model="meta/llama-3.1-405b-instruct"
)
# Register a function_name of None to get all functions
# sent to the same callback with an additional function_name parameter.
llm.register_function(None, fetch_weather_from_api, start_callback=start_fetch_weather)
tools = [
ChatCompletionToolParam(
type="function",
function={
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location", "format"],
},
},
)
]
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.",
},
]
context = OpenAILLMContext(messages, tools)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
tts,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
# Kick off the conversation.
await task.queue_frames([context_aggregator.user().get_context_frame()])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -9,10 +9,8 @@ import aiohttp
import os
import sys
from deepgram import LiveOptions
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import LLMMessagesFrame
from pipecat.frames.frames import LLMMessagesFrame, TTSUpdateSettingsFrame
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.parallel_pipeline import ParallelPipeline
from pipecat.pipeline.runner import PipelineRunner
@@ -20,7 +18,6 @@ from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.processors.filters.function_filter import FunctionFilter
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.deepgram import DeepgramSTTService
from pipecat.services.openai import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
@@ -64,16 +61,13 @@ async def main():
"Pipecat",
DailyParams(
audio_out_enabled=True,
transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True,
),
)
stt = DeepgramSTTService(
api_key=os.getenv("DEEPGRAM_API_KEY"), live_options=LiveOptions(language="multi")
)
english_tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
@@ -119,7 +113,6 @@ async def main():
pipeline = Pipeline(
[
transport.input(), # Transport user input
stt, # STT
context_aggregator.user(), # User responses
llm, # LLM
ParallelPipeline( # TTS (bot will speak the chosen language)

View File

@@ -53,7 +53,7 @@ async def main():
out_params=GStreamerPipelineSource.OutputParams(
video_width=1280,
video_height=720,
audio_sample_rate=24000,
audio_sample_rate=16000,
audio_channels=1,
),
)

View File

@@ -64,6 +64,7 @@ class StatementJudgeContextFilter(FrameProcessor):
self._notifier = notifier
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
# We must not block system frames.
if isinstance(frame, SystemFrame):
await self.push_frame(frame, direction)
@@ -117,6 +118,7 @@ class CompletenessCheck(FrameProcessor):
self._notifier = notifier
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TextFrame) and frame.text == "YES":
logger.debug("Completeness check YES")
await self.push_frame(UserStoppedSpeakingFrame())
@@ -139,6 +141,8 @@ class OutputGate(FrameProcessor):
self._gate_open = True
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
# We must not block system frames.
if isinstance(frame, SystemFrame):
if isinstance(frame, StartFrame):

View File

@@ -101,12 +101,12 @@ HIGH PRIORITY SIGNALS:
Examples:
# Complete Wh-question
[{"role": "assistant", "content": "I can help you learn."},
[{"role": "assistant", "content": "I can help you learn."},
{"role": "user", "content": "What's the fastest way to learn Spanish"}]
Output: YES
# Complete Yes/No question despite STT error
[{"role": "assistant", "content": "I know about planets."},
[{"role": "assistant", "content": "I know about planets."},
{"role": "user", "content": "Is is Jupiter the biggest planet"}]
Output: YES
@@ -118,12 +118,12 @@ Output: YES
Examples:
# Direct instruction
[{"role": "assistant", "content": "I can explain many topics."},
[{"role": "assistant", "content": "I can explain many topics."},
{"role": "user", "content": "Tell me about black holes"}]
Output: YES
# Action demand
[{"role": "assistant", "content": "I can help with math."},
[{"role": "assistant", "content": "I can help with math."},
{"role": "user", "content": "Solve this equation x plus 5 equals 12"}]
Output: YES
@@ -134,12 +134,12 @@ Output: YES
Examples:
# Specific answer
[{"role": "assistant", "content": "What's your favorite color?"},
[{"role": "assistant", "content": "What's your favorite color?"},
{"role": "user", "content": "I really like blue"}]
Output: YES
# Option selection
[{"role": "assistant", "content": "Would you prefer morning or evening?"},
[{"role": "assistant", "content": "Would you prefer morning or evening?"},
{"role": "user", "content": "Morning"}]
Output: YES
@@ -153,17 +153,17 @@ MEDIUM PRIORITY SIGNALS:
Examples:
# Self-correction reaching completion
[{"role": "assistant", "content": "What would you like to know?"},
[{"role": "assistant", "content": "What would you like to know?"},
{"role": "user", "content": "Tell me about... no wait, explain how rainbows form"}]
Output: YES
# Topic change with complete thought
[{"role": "assistant", "content": "The weather is nice today."},
[{"role": "assistant", "content": "The weather is nice today."},
{"role": "user", "content": "Actually can you tell me who invented the telephone"}]
Output: YES
# Mid-sentence completion
[{"role": "assistant", "content": "Hello I'm ready."},
[{"role": "assistant", "content": "Hello I'm ready."},
{"role": "user", "content": "What's the capital of? France"}]
Output: YES
@@ -175,12 +175,12 @@ Output: YES
Examples:
# Acknowledgment
[{"role": "assistant", "content": "Should we talk about history?"},
[{"role": "assistant", "content": "Should we talk about history?"},
{"role": "user", "content": "Sure"}]
Output: YES
# Disagreement with completion
[{"role": "assistant", "content": "Is that what you meant?"},
[{"role": "assistant", "content": "Is that what you meant?"},
{"role": "user", "content": "No not really"}]
Output: YES
@@ -194,12 +194,12 @@ LOW PRIORITY SIGNALS:
Examples:
# Word repetition but complete
[{"role": "assistant", "content": "I can help with that."},
[{"role": "assistant", "content": "I can help with that."},
{"role": "user", "content": "What what is the time right now"}]
Output: YES
# Missing punctuation but complete
[{"role": "assistant", "content": "I can explain that."},
[{"role": "assistant", "content": "I can explain that."},
{"role": "user", "content": "Please tell me how computers work"}]
Output: YES
@@ -211,12 +211,12 @@ Output: YES
Examples:
# Filler words but complete
[{"role": "assistant", "content": "What would you like to know?"},
[{"role": "assistant", "content": "What would you like to know?"},
{"role": "user", "content": "Um uh how do airplanes fly"}]
Output: YES
# Thinking pause but incomplete
[{"role": "assistant", "content": "I can explain anything."},
[{"role": "assistant", "content": "I can explain anything."},
{"role": "user", "content": "Well um I want to know about the"}]
Output: NO
@@ -241,17 +241,17 @@ DECISION RULES:
Examples:
# Incomplete despite corrections
[{"role": "assistant", "content": "What would you like to know about?"},
[{"role": "assistant", "content": "What would you like to know about?"},
{"role": "user", "content": "Can you tell me about"}]
Output: NO
# Complete despite multiple artifacts
[{"role": "assistant", "content": "I can help you learn."},
[{"role": "assistant", "content": "I can help you learn."},
{"role": "user", "content": "How do you I mean what's the best way to learn programming"}]
Output: YES
# Trailing off incomplete
[{"role": "assistant", "content": "I can explain anything."},
[{"role": "assistant", "content": "I can explain anything."},
{"role": "user", "content": "I was wondering if you could tell me why"}]
Output: NO
"""
@@ -268,6 +268,7 @@ class StatementJudgeContextFilter(FrameProcessor):
self._notifier = notifier
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
# We must not block system frames.
if isinstance(frame, SystemFrame):
await self.push_frame(frame, direction)
@@ -319,6 +320,8 @@ class CompletenessCheck(FrameProcessor):
self._notifier = notifier
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TextFrame) and frame.text == "YES":
logger.debug("!!! Completeness check YES")
await self.push_frame(UserStoppedSpeakingFrame())
@@ -341,6 +344,8 @@ class OutputGate(FrameProcessor):
self._gate_open = True
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
# We must not block system frames.
if isinstance(frame, SystemFrame):
if isinstance(frame, StartFrame):

View File

@@ -90,6 +90,8 @@ class StatementJudgeAudioContextAccumulator(FrameProcessor):
self._user_speaking = False
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
# ignore context frame
if isinstance(frame, OpenAILLMContextFrame):
return
@@ -131,6 +133,8 @@ class CompletenessCheck(FrameProcessor):
self._audio_accumulator = audio_accumulator
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TextFrame) and frame.text.startswith("YES"):
logger.debug("Completeness check YES")
await self.push_frame(UserStoppedSpeakingFrame())
@@ -155,6 +159,8 @@ class OutputGate(FrameProcessor):
self._gate_open = True
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
# We must not block system frames.
if isinstance(frame, SystemFrame):
if isinstance(frame, StartFrame):

View File

@@ -11,11 +11,12 @@ import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from openai.types.chat import ChatCompletionToolParam
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import LLMMessagesFrame
from pipecat.frames.frames import (
LLMMessagesFrame,
)
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
@@ -31,18 +32,6 @@ logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def start_fetch_weather(function_name, llm, context):
logger.debug(f"Starting fetch_weather_from_api with function_name: {function_name}")
async def fetch_weather_from_api(function_name, tool_call_id, args, llm, context, result_callback):
# Add a delay to test interruption during function calls
logger.info("Weather API call starting...")
await asyncio.sleep(5) # 5-second delay
logger.info("Weather API call completed")
await result_callback({"conditions": "nice", "temperature": "75"})
async def main():
async with aiohttp.ClientSession() as session:
(room_url, _) = await configure(session)
@@ -60,52 +49,23 @@ async def main():
)
stt = DeepgramSTTService(api_key=os.getenv("DEEPGRAM_API_KEY"))
# Configure the mute processor with both strategies
# Configure the mute processor to mute only during first speech
stt_mute_processor = STTMuteFilter(
stt_service=stt,
config=STTMuteConfig(
strategies={STTMuteStrategy.FIRST_SPEECH, STTMuteStrategy.FUNCTION_CALL}
),
stt_service=stt, config=STTMuteConfig(strategy=STTMuteStrategy.FIRST_SPEECH)
)
tts = DeepgramTTSService(api_key=os.getenv("DEEPGRAM_API_KEY"), voice="aura-helios-en")
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
llm.register_function(None, fetch_weather_from_api, start_callback=start_fetch_weather)
tools = [
ChatCompletionToolParam(
type="function",
function={
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location", "format"],
},
},
)
]
messages = [
{
"role": "system",
"content": "You are a helpful assistant who can check the weather. Always check the weather when a location is mentioned. Respond concisely and naturally. Your output will be converted to audio so use only simple words and punctuation.",
"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.",
},
]
context = OpenAILLMContext(messages, tools)
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
@@ -125,13 +85,8 @@ async def main():
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
# Kick off the conversation with a weather-related prompt
messages.append(
{
"role": "system",
"content": "Ask the user what city they'd like to know the weather for.",
}
)
# Kick off the conversation.
messages.append({"role": "system", "content": "Please introduce yourself to the user."})
await task.queue_frames([LLMMessagesFrame(messages)])
runner = PipelineRunner()

View File

@@ -1,366 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import aiohttp
import asyncio
import os
import sys
import google.ai.generativelanguage as glm
from dataclasses import dataclass
from dotenv import load_dotenv
from loguru import logger
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.parallel_pipeline import ParallelPipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.processors.aggregators.openai_llm_context import (
OpenAILLMContext,
OpenAILLMContextFrame,
)
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.google import GoogleLLMService, GoogleLLMContext
from pipecat.processors.frame_processor import FrameProcessor
from pipecat.transports.services.daily import DailyParams, DailyTransport
from pipecat.frames.frames import (
Frame,
InputAudioRawFrame,
LLMFullResponseEndFrame,
MetricsFrame,
SystemFrame,
TextFrame,
TranscriptionFrame,
UserStartedSpeakingFrame,
UserStoppedSpeakingFrame,
)
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
#
# The system prompt for the main conversation.
#
conversation_system_message = """
You are a helpful LLM in a WebRTC call. Your goals are to be helpful and brief in your responses. Respond with one or two sentences at most, unless you are asked to
respond at more length. Your output will be converted to audio so don't include special characters in your answers.
"""
#
# The system prompt for the LLM doing the audio transcription.
#
# Note that we could provide additional instructions per-conversation, here, if that's helpful
# for our use case. For example, names of people so that the transcription gets the spelling
# right.
#
# A possible future improvement would be to use structured output so that we can include a
# language tag and perhaps other analytic information.
#
transcriber_system_message = """
You are an audio transcriber. You are receiving audio from a user. Your job is to
transcribe the input audio to text exactly as it was said by the user..
You will receive the full conversation history before the audio input, to help with context. Use the full history only to help improve the accuracy of your transcription.
Rules:
- Respond with an exact transcription of the audio input.
- Do not include any text other than the transcription.
- Do not explain or add to your response.
- Transcribe the audio input simply and precisely.
- If the audio is not clear, emit the special string "EMPTY".
- No response other than exact transcription, or "EMPTY", is allowed.
"""
class UserAudioCollector(FrameProcessor):
"""
This FrameProcessor collects audio frames in a buffer, then adds them to the
LLM context when the user stops speaking.
"""
def __init__(self, context, user_context_aggregator):
super().__init__()
self._context = context
self._user_context_aggregator = user_context_aggregator
self._audio_frames = []
self._start_secs = 0.2 # this should match VAD start_secs (hardcoding for now)
self._user_speaking = False
async def process_frame(self, frame, direction):
if isinstance(frame, TranscriptionFrame):
# We could gracefully handle both audio input and text/transcription input ...
# but let's leave that as an exercise to the reader. :-)
return
if isinstance(frame, UserStartedSpeakingFrame):
self._user_speaking = True
elif isinstance(frame, UserStoppedSpeakingFrame):
self._user_speaking = False
self._context.add_audio_frames_message(audio_frames=self._audio_frames)
await self._user_context_aggregator.push_frame(
self._user_context_aggregator.get_context_frame()
)
elif isinstance(frame, InputAudioRawFrame):
if self._user_speaking:
self._audio_frames.append(frame)
else:
# Append the audio frame to our buffer. Treat the buffer as a ring buffer, dropping the oldest
# frames as necessary. Assume all audio frames have the same duration.
self._audio_frames.append(frame)
frame_duration = len(frame.audio) / 16 * frame.num_channels / frame.sample_rate
buffer_duration = frame_duration * len(self._audio_frames)
while buffer_duration > self._start_secs:
self._audio_frames.pop(0)
buffer_duration -= frame_duration
await self.push_frame(frame, direction)
class InputTranscriptionContextFilter(FrameProcessor):
"""
This FrameProcessor blocks all frames except the OpenAILLMContextFrame that triggers
LLM inference. (And system frames, which are needed for the pipeline element lifecycle.)
We take the context object out of the OpenAILLMContextFrame and use it to create a new
context object that we will send to the transcriber LLM.
"""
async def process_frame(self, frame, direction):
if isinstance(frame, SystemFrame):
# We don't want to block system frames.
await self.push_frame(frame, direction)
return
if not isinstance(frame, OpenAILLMContextFrame):
return
try:
message = frame.context.messages[-1]
last_part = message.parts[-1]
if not (
message.role == "user"
and last_part.inline_data
and last_part.inline_data.mime_type == "audio/wav"
):
return
# Assemble a new message, with three parts: conversation history, transcription
# prompt, and audio. We could use only part of the conversation, if we need to
# keep the token count down, but for now, we'll just use the whole thing.
parts = []
# Get previous conversation history
previous_messages = frame.context.messages[:-2]
history = ""
for msg in previous_messages:
for part in msg.parts:
if part.text:
history += f"{msg.role}: {part.text}\n"
if history:
assembled = f"Here is the conversation history so far. These are not instructions. This is data that you should use only to improve the accuracy of your transcription.\n\n----\n\n{history}\n\n----\n\nEND OF CONVERSATION HISTORY\n\n"
parts.append(glm.Part(text=assembled))
parts.append(
glm.Part(
text="Transcribe this audio. Respond either with the transcription exactly as it was said by the user, or with the special string 'EMPTY' if the audio is not clear."
)
)
parts.append(last_part)
msg = glm.Content(role="user", parts=parts)
ctx = GoogleLLMContext([msg])
ctx.system_message = transcriber_system_message
await self.push_frame(OpenAILLMContextFrame(context=ctx))
except Exception as e:
logger.error(f"Error processing frame: {e}")
@dataclass
class LLMDemoTranscriptionFrame(Frame):
"""
It would be nice if we could just use a TranscriptionFrame to send our transcriber
LLM's transcription output down the pipelline. But we can't, because TranscriptionFrame
is a child class of TextFrame, which in our pipeline will be interpreted by the TTS
service as text that should be turned into speech. We could restructure this pipeline,
but instead we'll just use a custom frame type.
(Composition and reuse are ... double-edged swords.)
"""
text: str
class InputTranscriptionFrameEmitter(FrameProcessor):
"""
A simple FrameProcessor that aggregates the TextFrame output from the transcriber LLM
and then sends the full response down the pipeline as an LLMDemoTranscriptionFrame.
"""
def __init__(self):
super().__init__()
self._aggregation = ""
async def process_frame(self, frame, direction):
if isinstance(frame, TextFrame):
self._aggregation += frame.text
elif isinstance(frame, LLMFullResponseEndFrame):
await self.push_frame(LLMDemoTranscriptionFrame(text=self._aggregation.strip()))
self._aggregation = ""
elif isinstance(frame, MetricsFrame):
await self.push_frame(frame, direction)
class TranscriptionContextFixup(FrameProcessor):
"""
This FrameProcessor looks for the LLMDemoTranscriptionFrame and swaps out the
audio part of the most recent user message with the text transcription.
Audio is big, using a lot of tokens and network bandwidth. So doing this is
important if we want to keep both latency and cost low.
This class is a bit of a hack, especially because it directly creates a
GoogleLLMContext object, which we don't generally do. We usually try to leave
the implementation-specific details of the LLM context encapsulated inside the
service classes.
"""
def __init__(self, context):
super().__init__()
self._context = context
self._transcript = "THIS IS A TRANSCRIPT"
def is_user_audio_message(self, message):
last_part = message.parts[-1]
return (
message.role == "user"
and last_part.inline_data
and last_part.inline_data.mime_type == "audio/wav"
)
def swap_user_audio(self):
if not self._transcript:
return
message = self._context.messages[-2]
if not self.is_user_audio_message(message):
message = self._context.messages[-1]
if not self.is_user_audio_message(message):
return
audio_part = message.parts[-1]
audio_part.inline_data = None
audio_part.text = self._transcript
async def process_frame(self, frame, direction):
if isinstance(frame, LLMDemoTranscriptionFrame):
logger.info(f"Transcription from Gemini: {frame.text}")
self._transcript = frame.text
self.swap_user_audio()
self._transcript = ""
await self.push_frame(frame, direction)
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_out_enabled=True,
# No transcription at all. just audio input to Gemini!
# transcription_enabled=True,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
vad_audio_passthrough=True,
),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="79a125e8-cd45-4c13-8a67-188112f4dd22", # British Lady
)
conversation_llm = GoogleLLMService(
name="Conversation",
model="gemini-1.5-flash-latest",
# model="gemini-exp-1121",
api_key=os.getenv("GOOGLE_API_KEY"),
# we can give the GoogleLLMService a system instruction to use directly
# in the GenerativeModel constructor. Let's do that rather than put
# our system message in the messages list.
system_instruction=conversation_system_message,
)
input_transcription_llm = GoogleLLMService(
name="Transcription",
model="gemini-1.5-flash-latest",
# model="gemini-exp-1121",
api_key=os.getenv("GOOGLE_API_KEY"),
system_instruction=transcriber_system_message,
)
messages = [
{
"role": "user",
"content": "Start by saying hello.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = conversation_llm.create_context_aggregator(context)
audio_collector = UserAudioCollector(context, context_aggregator.user())
input_transcription_context_filter = InputTranscriptionContextFilter()
transcription_frames_emitter = InputTranscriptionFrameEmitter()
fixup_context_messages = TranscriptionContextFixup(context)
pipeline = Pipeline(
[
transport.input(),
audio_collector,
context_aggregator.user(),
ParallelPipeline(
[ # transcribe
input_transcription_context_filter,
input_transcription_llm,
transcription_frames_emitter,
],
[ # conversation inference
conversation_llm,
],
),
tts,
transport.output(),
context_aggregator.assistant(),
fixup_context_messages,
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
# Kick off the conversation.
await task.queue_frames([context_aggregator.user().get_context_frame()])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,82 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import aiohttp
import asyncio
import os
import sys
from dotenv import load_dotenv
from loguru import logger
from runner import configure
from pipecat.services.gemini_multimodal_live.gemini import GeminiMultimodalLiveLLMService
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_in_sample_rate=16000,
audio_out_sample_rate=24000,
audio_out_enabled=True,
vad_enabled=True,
vad_audio_passthrough=True,
# set stop_secs to something roughly similar to the internal setting
# of the Multimodal Live api, just to align events. This doesn't really
# matter because we can only use the Multimodal Live API's phrase
# endpointing, for now.
vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.5)),
),
)
llm = GeminiMultimodalLiveLLMService(
api_key=os.getenv("GOOGLE_API_KEY"),
# system_instruction="Talk like a pirate."
)
pipeline = Pipeline(
[
transport.input(),
llm,
transport.output(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,111 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
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.services.gemini_multimodal_live.gemini import GeminiMultimodalLiveLLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_in_sample_rate=16000,
audio_out_sample_rate=24000,
audio_out_enabled=True,
vad_enabled=True,
vad_audio_passthrough=True,
# set stop_secs to something roughly similar to the internal setting
# of the Multimodal Live api, just to align events. This doesn't really
# matter because we can only use the Multimodal Live API's phrase
# endpointing, for now.
vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.5)),
),
)
llm = GeminiMultimodalLiveLLMService(
api_key=os.getenv("GOOGLE_API_KEY"),
voice_id="Aoede", # Puck, Charon, Kore, Fenrir, Aoede
# system_instruction="Talk like a pirate."
transcribe_user_audio=True,
transcribe_model_audio=True,
# inference_on_context_initialization=False,
)
context = OpenAILLMContext(
[
{
"role": "user",
"content": "Say hello. Then ask if I want to hear a joke.",
},
# {"role": "assistant", "content": "Hello! Why don't scientists trust atoms?"},
# {
# "role": "user",
# "content": [
# {
# "type": "text",
# "text": "Oh, I know this one: because they make up everything.",
# }
# ],
# },
],
)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await task.queue_frames([context_aggregator.user().get_context_frame()])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,142 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
from datetime import datetime
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
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.services.gemini_multimodal_live.gemini import GeminiMultimodalLiveLLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def fetch_weather_from_api(function_name, tool_call_id, args, llm, context, result_callback):
temperature = 75 if args["format"] == "fahrenheit" else 24
await result_callback(
{
"conditions": "nice",
"temperature": temperature,
"format": args["format"],
"timestamp": datetime.now().strftime("%Y%m%d_%H%M%S"),
}
)
tools = [
{
"function_declarations": [
{
"name": "get_current_weather",
"description": "Get the current weather",
"parameters": {
"type": "object",
"properties": {
"location": {
"type": "string",
"description": "The city and state, e.g. San Francisco, CA",
},
"format": {
"type": "string",
"enum": ["celsius", "fahrenheit"],
"description": "The temperature unit to use. Infer this from the users location.",
},
},
"required": ["location", "format"],
},
},
]
}
]
system_instruction = """
You are a helpful assistant who can answer questions and use tools.
You have a tool called "get_current_weather" that can be used to get the current weather. If the user asks
for the weather, call this function.
"""
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_in_sample_rate=16000,
audio_out_sample_rate=24000,
audio_out_enabled=True,
vad_enabled=True,
vad_audio_passthrough=True,
# set stop_secs to something roughly similar to the internal setting
# of the Multimodal Live api, just to align events. This doesn't really
# matter because we can only use the Multimodal Live API's phrase
# endpointing, for now.
vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.5)),
),
)
llm = GeminiMultimodalLiveLLMService(
api_key=os.getenv("GOOGLE_API_KEY"),
system_instruction=system_instruction,
tools=tools,
)
llm.register_function("get_current_weather", fetch_weather_from_api)
context = OpenAILLMContext(
[{"role": "user", "content": "Say hello."}],
)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
context_aggregator.assistant(),
transport.output(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await task.queue_frames([context_aggregator.user().get_context_frame()])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,115 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import os
import sys
import aiohttp
from dotenv import load_dotenv
from loguru import logger
from runner import configure
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.audio.vad.vad_analyzer import VADParams
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.services.gemini_multimodal_live.gemini import GeminiMultimodalLiveLLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
transport = DailyTransport(
room_url,
token,
"Respond bot",
DailyParams(
audio_in_sample_rate=16000,
audio_out_sample_rate=24000,
audio_out_enabled=True,
vad_enabled=True,
vad_audio_passthrough=True,
# set stop_secs to something roughly similar to the internal setting
# of the Multimodal Live api, just to align events. This doesn't really
# matter because we can only use the Multimodal Live API's phrase
# endpointing, for now.
vad_analyzer=SileroVADAnalyzer(params=VADParams(stop_secs=0.5)),
start_audio_paused=True,
start_video_paused=True,
),
)
llm = GeminiMultimodalLiveLLMService(
api_key=os.getenv("GOOGLE_API_KEY"),
voice_id="Aoede", # Puck, Charon, Kore, Fenrir, Aoede
# system_instruction="Talk like a pirate."
transcribe_user_audio=True,
transcribe_model_audio=True,
# inference_on_context_initialization=False,
)
context = OpenAILLMContext(
[
{
"role": "user",
"content": "Say hello.",
},
],
)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
# Enable both camera and screenshare. From the client side
# send just one.
await transport.capture_participant_video(
participant["id"], framerate=1, video_source="camera"
)
await transport.capture_participant_video(
participant["id"], framerate=1, video_source="screenVideo"
)
await task.queue_frames([context_aggregator.user().get_context_frame()])
await asyncio.sleep(3)
logger.debug("Unpausing audio and video")
llm.set_audio_input_paused(False)
llm.set_video_input_paused(False)
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -1,105 +0,0 @@
#
# Copyright (c) 2024, Daily
#
# SPDX-License-Identifier: BSD 2-Clause License
#
import asyncio
import aiohttp
import os
import sys
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.pipeline.pipeline import Pipeline
from pipecat.pipeline.runner import PipelineRunner
from pipecat.pipeline.task import PipelineParams, PipelineTask
from pipecat.frames.frames import LLMMessagesFrame
from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
from pipecat.services.cartesia import CartesiaTTSService
from pipecat.services.openai import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
from runner import configure
from loguru import logger
from dotenv import load_dotenv
from simli import SimliConfig
from pipecat.services.simli import SimliVideoService
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
async def main():
async with aiohttp.ClientSession() as session:
room, token = await configure(session)
transport = DailyTransport(
room,
token,
"Simli",
DailyParams(
audio_out_enabled=True,
camera_out_enabled=True,
camera_out_width=512,
camera_out_height=512,
vad_enabled=True,
vad_analyzer=SileroVADAnalyzer(),
transcription_enabled=True,
),
)
tts = CartesiaTTSService(
api_key=os.getenv("CARTESIA_API_KEY"),
voice_id="a167e0f3-df7e-4d52-a9c3-f949145efdab",
)
simli_ai = SimliVideoService(
SimliConfig(os.getenv("SIMLI_API_KEY"), os.getenv("SIMLI_FACE_ID"))
)
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o-mini")
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.",
},
]
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
pipeline = Pipeline(
[
transport.input(),
context_aggregator.user(),
llm,
tts,
simli_ai,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
),
)
@transport.event_handler("on_first_participant_joined")
async def on_first_participant_joined(transport, participant):
await transport.capture_participant_transcription(participant["id"])
await task.queue_frames([LLMMessagesFrame(messages)])
runner = PipelineRunner()
await runner.run(task)
if __name__ == "__main__":
asyncio.run(main())

View File

@@ -13,13 +13,13 @@ from PIL import Image
from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
BotStartedSpeakingFrame,
BotStoppedSpeakingFrame,
ImageRawFrame,
OutputImageRawFrame,
SpriteFrame,
Frame,
LLMMessagesFrame,
TTSAudioRawFrame,
TTSStoppedFrame,
TextFrame,
UserImageRawFrame,
UserImageRequestFrame,
@@ -81,15 +81,16 @@ class TalkingAnimation(FrameProcessor):
self._is_talking = False
async def process_frame(self, frame: Frame, direction: FrameDirection):
if isinstance(frame, BotStartedSpeakingFrame):
await super().process_frame(frame, direction)
if isinstance(frame, TTSAudioRawFrame):
if not self._is_talking:
await self.push_frame(talking_frame)
self._is_talking = True
elif isinstance(frame, BotStoppedSpeakingFrame):
elif isinstance(frame, TTSStoppedFrame):
await self.push_frame(quiet_frame)
self._is_talking = False
await self.push_frame(frame, direction)
await self.push_frame(frame)
class UserImageRequester(FrameProcessor):
@@ -101,6 +102,8 @@ class UserImageRequester(FrameProcessor):
self.participant_id = participant_id
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if self.participant_id and isinstance(frame, TextFrame):
if frame.text == user_request_answer:
await self.push_frame(
@@ -117,17 +120,21 @@ class TextFilterProcessor(FrameProcessor):
self.text = text
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if isinstance(frame, TextFrame):
if frame.text != self.text:
await self.push_frame(frame)
else:
await self.push_frame(frame, direction)
await self.push_frame(frame)
class ImageFilterProcessor(FrameProcessor):
async def process_frame(self, frame: Frame, direction: FrameDirection):
await super().process_frame(frame, direction)
if not isinstance(frame, ImageRawFrame):
await self.push_frame(frame, direction)
await self.push_frame(frame)
async def main():

View File

@@ -4,8 +4,6 @@
This project implements an AI-powered chatbot designed to streamline the medical intake process for Tri-County Health Services. The chatbot, named Jessica, interacts with patients to collect essential information before their doctor's visit, enhancing efficiency and improving the patient experience.
💡 Looking to build structured conversations? Check out [Pipecat Flows](https://github.com/pipecat-ai/pipecat-flows) for managing complex conversational states and transitions.
## Features
Identity Verification: Confirms patient identity by verifying their date of birth.
@@ -64,32 +62,3 @@ Then, visit `http://localhost:7860/` in your browser to start a chatbot session.
docker build -t chatbot .
docker run --env-file .env -p 7860:7860 chatbot
```
## Cartesia best practices
Since this example is using Cartesia, checkout the best practices given in Cartesia's docs. LLM prompts should be modified accordingly.
<https://docs.cartesia.ai/build-with-sonic/formatting-text-for-sonic/best-practices>
<https://docs.cartesia.ai/build-with-sonic/formatting-text-for-sonic/inserting-breaks-pauses>
<https://docs.cartesia.ai/build-with-sonic/formatting-text-for-sonic/spelling-out-input-text>
### Example
```python
messages = [
{
"role": "system",
"content": '''You are a helpful AI assistant. Format all responses following these guidelines:
1. Use proper punctuation and end each response with appropriate punctuation
2. Format dates as MM/DD/YYYY
3. Insert pauses using - or <break time='1s' /> for longer pauses
4. Use ?? for emphasized questions
5. Avoid quotation marks unless citing
6. Add spaces between URLs/emails and punctuation marks
7. For domain-specific terms or proper nouns, provide pronunciation guidance in [brackets]
8. Keep responses clear and concise
9. Use appropriate voice/language pairs for multilingual content
Your goal is to demonstrate these capabilities in a succinct way. Your output will be converted to audio, so maintain natural communication flow. Respond creatively and helpfully, but keep responses brief. Start by introducing yourself.'''
}
]
```

View File

@@ -1,51 +1,161 @@
# Python
# Byte-compiled / optimized / DLL files
__pycache__/
*.py[cod]
*$py.class
# C extensions
*.so
# Distribution / packaging
.Python
build/
develop-eggs/
dist/
downloads/
eggs/
.eggs/
lib/
lib64/
parts/
sdist/
var/
wheels/
share/python-wheels/
*.egg-info/
.installed.cfg
*.egg
.pytest_cache/
MANIFEST
# PyInstaller
# Usually these files are written by a python script from a template
# before PyInstaller builds the exe, so as to inject date/other infos into it.
*.manifest
*.spec
# Installer logs
pip-log.txt
pip-delete-this-directory.txt
# Unit test / coverage reports
htmlcov/
.tox/
.nox/
.coverage
.coverage.*
.cache
nosetests.xml
coverage.xml
*.cover
*.py,cover
.hypothesis/
.pytest_cache/
cover/
# Translations
*.mo
*.pot
# Django stuff:
*.log
local_settings.py
db.sqlite3
db.sqlite3-journal
# Flask stuff:
instance/
.webassets-cache
# Scrapy stuff:
.scrapy
# Sphinx documentation
docs/_build/
# PyBuilder
.pybuilder/
target/
# Jupyter Notebook
.ipynb_checkpoints
# IPython
profile_default/
ipython_config.py
# pyenv
# For a library or package, you might want to ignore these files since the code is
# intended to run in multiple environments; otherwise, check them in:
# .python-version
# pipenv
# According to pypa/pipenv#598, it is recommended to include Pipfile.lock in version control.
# However, in case of collaboration, if having platform-specific dependencies or dependencies
# having no cross-platform support, pipenv may install dependencies that don't work, or not
# install all needed dependencies.
#Pipfile.lock
# poetry
# Similar to Pipfile.lock, it is generally recommended to include poetry.lock in version control.
# This is especially recommended for binary packages to ensure reproducibility, and is more
# commonly ignored for libraries.
# https://python-poetry.org/docs/basic-usage/#commit-your-poetrylock-file-to-version-control
#poetry.lock
# pdm
# Similar to Pipfile.lock, it is generally recommended to include pdm.lock in version control.
#pdm.lock
# pdm stores project-wide configurations in .pdm.toml, but it is recommended to not include it
# in version control.
# https://pdm.fming.dev/#use-with-ide
.pdm.toml
# PEP 582; used by e.g. github.com/David-OConnor/pyflow and github.com/pdm-project/pdm
__pypackages__/
# Celery stuff
celerybeat-schedule
celerybeat.pid
# SageMath parsed files
*.sage.py
# Environments
.env
.venv
env/
venv/
ENV/
env.bak/
venv.bak/
# Spyder project settings
.spyderproject
.spyproject
# Rope project settings
.ropeproject
# mkdocs documentation
/site
# mypy
.mypy_cache/
.dmypy.json
dmypy.json
# JavaScript/Node.js
node_modules/
dist/
dist-ssr/
*.local
.env.local
.env.development.local
.env.test.local
.env.production.local
# Pyre type checker
.pyre/
# Logs
logs/
*.log
npm-debug.log*
yarn-debug.log*
yarn-error.log*
pnpm-debug.log*
# pytype static type analyzer
.pytype/
# Editor/IDE
.vscode/*
!.vscode/extensions.json
.idea/
*.swp
*.swo
.DS_Store
# Cython debug symbols
cython_debug/
# Project specific
runpod.toml
# PyCharm
# JetBrains specific template is maintained in a separate JetBrains.gitignore that can
# be found at https://github.com/github/gitignore/blob/main/Global/JetBrains.gitignore
# and can be added to the global gitignore or merged into this file. For a more nuclear
# option (not recommended) you can uncomment the following to ignore the entire idea folder.
#.idea/
runpod.toml

View File

@@ -2,96 +2,36 @@
<img src="image.png" width="420px">
This repository demonstrates a simple AI chatbot with real-time audio/video interaction, implemented in three different ways. The bot server supports multiple AI backends, and you can connect to it using three different client approaches.
This app connects you to a chatbot powered by GPT-4, complete with animations generated by Stable Video Diffusion.
## Two Bot Options
See a video of it in action: https://x.com/kwindla/status/1778628911817183509
1. **OpenAI Bot** (Default)
And a quick video walkthrough of the code: https://www.loom.com/share/13df1967161f4d24ade054e7f8753416
- Uses gpt-4o for conversation
- Requires OpenAI API key
The first time, things might take extra time to get started since VAD (Voice Activity Detection) model needs to be downloaded.
2. **Gemini Bot**
- Uses Google's Gemini Multimodal Live model
- Requires Gemini API key
## Get started
## Three Ways to Connect
```python
python3 -m venv venv
source venv/bin/activate
pip install -r requirements.txt
1. **Daily Prebuilt** (Simplest)
- Direct connection through a Daily Prebuilt room
- For demo purposes only; handy for quick testing
2. **JavaScript**
- Basic implementation using [Pipecat JavaScript SDK](https://docs.pipecat.ai/client/reference/js/introduction)
- No framework dependencies
- Good for learning the fundamentals
3. **React**
- Basic impelmentation using [Pipecat React SDK](https://docs.pipecat.ai/client/reference/react/introduction)
- Demonstrates the basic client principles with Pipecat React
## Quick Start
### First, start the bot server:
1. Navigate to the server directory:
```bash
cd server
```
2. Create and activate a virtual environment:
```bash
python3 -m venv venv
source venv/bin/activate # On Windows: venv\Scripts\activate
```
3. Install requirements:
```bash
pip install -r requirements.txt
```
4. Copy env.example to .env and configure:
- Add your API keys
- Choose your bot implementation:
```ini
BOT_IMPLEMENTATION= # Options: 'openai' (default) or 'gemini'
```
5. Start the server:
```bash
python server.py
```
### Next, connect using your preferred client app:
- [Daily Prebuilt](examples/prebuilt/README.md)
- [JavaScript Guide](examples/javascript/README.md)
- [React Guide](examples/react/README.md)
## Important Note
The bot server must be running for any of the client implementations to work. Start the server first before trying any of the client apps.
## Requirements
- Python 3.10+
- Node.js 16+ (for JavaScript and React implementations)
- Daily API key
- OpenAI API key (for OpenAI bot)
- Gemini API key (for Gemini bot)
- ElevenLabs API key
- Modern web browser with WebRTC support
## Project Structure
cp env.example .env # and add your credentials
```
simple-chatbot/
├── server/ # Bot server implementation
│ ├── bot-openai.py # OpenAI bot implementation
│ ├── bot-gemini.py # Gemini bot implementation
│ ├── runner.py # Server runner utilities
│ ├── server.py # FastAPI server
│ └── requirements.txt
└── examples/ # Client implementations
├── prebuilt/ # Daily Prebuilt connection
├── javascript/ # Pipecat JavaScript client
└── react/ # Pipecat React client
## Run the server
```bash
python server.py
```
Then, visit `http://localhost:7860/` in your browser to start a chatbot session.
## Build and test the Docker image
```
docker build -t chatbot .
docker run --env-file .env -p 7860:7860 chatbot
```

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@@ -4,19 +4,6 @@
# SPDX-License-Identifier: BSD 2-Clause License
#
"""OpenAI Bot Implementation.
This module implements a chatbot using OpenAI's GPT-4 model for natural language
processing. It includes:
- Real-time audio/video interaction through Daily
- Animated robot avatar
- Text-to-speech using ElevenLabs
- Support for both English and Spanish
The bot runs as part of a pipeline that processes audio/video frames and manages
the conversation flow.
"""
import asyncio
import os
import sys
@@ -31,7 +18,6 @@ from pipecat.audio.vad.silero import SileroVADAnalyzer
from pipecat.frames.frames import (
BotStartedSpeakingFrame,
BotStoppedSpeakingFrame,
EndFrame,
Frame,
LLMMessagesFrame,
OutputImageRawFrame,
@@ -42,24 +28,19 @@ 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.frame_processor import FrameDirection, FrameProcessor
from pipecat.processors.frameworks.rtvi import (
RTVIBotTranscriptionProcessor,
RTVIMetricsProcessor,
RTVISpeakingProcessor,
RTVIUserTranscriptionProcessor,
)
from pipecat.services.elevenlabs import ElevenLabsTTSService
from pipecat.services.openai import OpenAILLMService
from pipecat.transports.services.daily import DailyParams, DailyTransport
load_dotenv(override=True)
logger.remove(0)
logger.add(sys.stderr, level="DEBUG")
sprites = []
script_dir = os.path.dirname(__file__)
# Load sequential animation frames
for i in range(1, 26):
# Build the full path to the image file
full_path = os.path.join(script_dir, f"assets/robot0{i}.png")
@@ -68,20 +49,18 @@ for i in range(1, 26):
with Image.open(full_path) as img:
sprites.append(OutputImageRawFrame(image=img.tobytes(), size=img.size, format=img.format))
# Create a smooth animation by adding reversed frames
flipped = sprites[::-1]
sprites.extend(flipped)
# Define static and animated states
quiet_frame = sprites[0] # Static frame for when bot is listening
talking_frame = SpriteFrame(images=sprites) # Animation sequence for when bot is talking
# When the bot isn't talking, show a static image of the cat listening
quiet_frame = sprites[0]
talking_frame = SpriteFrame(images=sprites)
class TalkingAnimation(FrameProcessor):
"""Manages the bot's visual animation states.
Switches between static (listening) and animated (talking) states based on
the bot's current speaking status.
"""
This class starts a talking animation when it receives an first AudioFrame,
and then returns to a "quiet" sprite when it sees a TTSStoppedFrame.
"""
def __init__(self):
@@ -89,18 +68,12 @@ class TalkingAnimation(FrameProcessor):
self._is_talking = False
async def process_frame(self, frame: Frame, direction: FrameDirection):
"""Process incoming frames and update animation state.
await super().process_frame(frame, direction)
Args:
frame: The incoming frame to process
direction: The direction of frame flow in the pipeline
"""
# Switch to talking animation when bot starts speaking
if isinstance(frame, BotStartedSpeakingFrame):
if not self._is_talking:
await self.push_frame(talking_frame)
self._is_talking = True
# Return to static frame when bot stops speaking
elif isinstance(frame, BotStoppedSpeakingFrame):
await self.push_frame(quiet_frame)
self._is_talking = False
@@ -109,19 +82,9 @@ class TalkingAnimation(FrameProcessor):
async def main():
"""Main bot execution function.
Sets up and runs the bot pipeline including:
- Daily video transport
- Speech-to-text and text-to-speech services
- Language model integration
- Animation processing
- RTVI event handling
"""
async with aiohttp.ClientSession() as session:
(room_url, token) = await configure(session)
# Set up Daily transport with video/audio parameters
transport = DailyTransport(
room_url,
token,
@@ -145,7 +108,6 @@ async def main():
),
)
# Initialize text-to-speech service
tts = ElevenLabsTTSService(
api_key=os.getenv("ELEVENLABS_API_KEY"),
#
@@ -159,7 +121,6 @@ async def main():
# voice_id="gD1IexrzCvsXPHUuT0s3",
)
# Initialize LLM service
llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"), model="gpt-4o")
messages = [
@@ -176,53 +137,24 @@ async def main():
},
]
# Set up conversation context and management
# The context_aggregator will automatically collect conversation context
context = OpenAILLMContext(messages)
context_aggregator = llm.create_context_aggregator(context)
ta = TalkingAnimation()
#
# RTVI events for Pipecat client UI
#
# This will send `user-*-speaking` and `bot-*-speaking` messages.
rtvi_speaking = RTVISpeakingProcessor()
# This will emit UserTranscript events.
rtvi_user_transcription = RTVIUserTranscriptionProcessor()
# This will emit BotTranscript events.
rtvi_bot_transcription = RTVIBotTranscriptionProcessor()
# This will send `metrics` messages.
rtvi_metrics = RTVIMetricsProcessor()
pipeline = Pipeline(
[
transport.input(),
rtvi_speaking,
rtvi_user_transcription,
context_aggregator.user(),
llm,
rtvi_bot_transcription,
tts,
ta,
rtvi_metrics,
transport.output(),
context_aggregator.assistant(),
]
)
task = PipelineTask(
pipeline,
PipelineParams(
allow_interruptions=True,
enable_metrics=True,
enable_usage_metrics=True,
),
)
task = PipelineTask(pipeline, PipelineParams(allow_interruptions=True))
await task.queue_frame(quiet_frame)
@transport.event_handler("on_first_participant_joined")
@@ -230,11 +162,6 @@ async def main():
await transport.capture_participant_transcription(participant["id"])
await task.queue_frames([LLMMessagesFrame(messages)])
@transport.event_handler("on_participant_left")
async def on_participant_left(transport, participant, reason):
print(f"Participant left: {participant}")
await task.queue_frame(EndFrame())
runner = PipelineRunner()
await runner.run(task)

View File

@@ -1,6 +1,4 @@
DAILY_SAMPLE_ROOM_URL=https://yourdomain.daily.co/yourroom # (for joining the bot to the same room repeatedly for local dev)
DAILY_API_KEY=7df...
OPENAI_API_KEY=sk-PL...
GEMINI_API_KEY=AIza...
ELEVENLABS_API_KEY=aeb...
BOT_IMPLEMENTATION= # Options: 'openai' or 'gemini'
ELEVENLABS_API_KEY=aeb...

View File

@@ -1,27 +0,0 @@
# JavaScript Implementation
Basic implementation using the [Pipecat JavaScript SDK](https://docs.pipecat.ai/client/reference/js/introduction).
## Setup
1. Run the bot server. See the [server README](../../README).
2. Navigate to the `examples/javascript` directory:
```bash
cd examples/javascript
```
3. Install dependencies:
```bash
npm install
```
4. Run the client app:
```
npm run dev
```
5. Visit http://localhost:5173 in your browser.

View File

@@ -1,40 +0,0 @@
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<title>AI Chatbot</title>
</head>
<body>
<div class="container">
<div class="status-bar">
<div class="status">
Status: <span id="connection-status">Disconnected</span>
</div>
<div class="controls">
<button id="connect-btn">Connect</button>
<button id="disconnect-btn" disabled>Disconnect</button>
</div>
</div>
<div class="main-content">
<div class="bot-container">
<div id="bot-video-container">
</div>
<audio id="bot-audio" autoplay></audio>
</div>
</div>
<div class="debug-panel">
<h3>Debug Info</h3>
<div id="debug-log"></div>
</div>
</div>
<script type="module" src="/src/app.js"></script>
<link rel="stylesheet" href="/src/style.css">
</body>
</html>

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View File

@@ -1,21 +0,0 @@
{
"name": "client",
"version": "1.0.0",
"main": "index.js",
"scripts": {
"dev": "vite",
"build": "vite build",
"preview": "vite preview"
},
"keywords": [],
"author": "",
"license": "ISC",
"description": "",
"dependencies": {
"@daily-co/realtime-ai-daily": "^0.2.1",
"realtime-ai": "^0.2.1"
},
"devDependencies": {
"vite": "^6.0.2"
}
}

View File

@@ -1,314 +0,0 @@
/**
* Copyright (c) 2024, Daily
*
* SPDX-License-Identifier: BSD 2-Clause License
*/
/**
* RTVI Client Implementation
*
* This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
* It handles audio/video streaming and manages the connection lifecycle.
*
* Requirements:
* - A running RTVI bot server (defaults to http://localhost:7860)
* - The server must implement the /connect endpoint that returns Daily.co room credentials
* - Browser with WebRTC support
*/
import { RTVIClient, RTVIEvent } from 'realtime-ai';
import { DailyTransport } from '@daily-co/realtime-ai-daily';
/**
* ChatbotClient handles the connection and media management for a real-time
* voice and video interaction with an AI bot.
*/
class ChatbotClient {
constructor() {
// Initialize client state
this.rtviClient = null;
this.setupDOMElements();
this.setupEventListeners();
}
/**
* Set up references to DOM elements and create necessary media elements
*/
setupDOMElements() {
// Get references to UI control elements
this.connectBtn = document.getElementById('connect-btn');
this.disconnectBtn = document.getElementById('disconnect-btn');
this.statusSpan = document.getElementById('connection-status');
this.debugLog = document.getElementById('debug-log');
this.botVideoContainer = document.getElementById('bot-video-container');
// Create an audio element for bot's voice output
this.botAudio = document.createElement('audio');
this.botAudio.autoplay = true;
this.botAudio.playsInline = true;
document.body.appendChild(this.botAudio);
}
/**
* Set up event listeners for connect/disconnect buttons
*/
setupEventListeners() {
this.connectBtn.addEventListener('click', () => this.connect());
this.disconnectBtn.addEventListener('click', () => this.disconnect());
}
/**
* Add a timestamped message to the debug log
*/
log(message) {
const entry = document.createElement('div');
entry.textContent = `${new Date().toISOString()} - ${message}`;
// Add styling based on message type
if (message.startsWith('User: ')) {
entry.style.color = '#2196F3'; // blue for user
} else if (message.startsWith('Bot: ')) {
entry.style.color = '#4CAF50'; // green for bot
}
this.debugLog.appendChild(entry);
this.debugLog.scrollTop = this.debugLog.scrollHeight;
console.log(message);
}
/**
* Update the connection status display
*/
updateStatus(status) {
this.statusSpan.textContent = status;
this.log(`Status: ${status}`);
}
/**
* Check for available media tracks and set them up if present
* This is called when the bot is ready or when the transport state changes to ready
*/
setupMediaTracks() {
if (!this.rtviClient) return;
// Get current tracks from the client
const tracks = this.rtviClient.tracks();
// Set up any available bot tracks
if (tracks.bot?.audio) {
this.setupAudioTrack(tracks.bot.audio);
}
if (tracks.bot?.video) {
this.setupVideoTrack(tracks.bot.video);
}
}
/**
* Set up listeners for track events (start/stop)
* This handles new tracks being added during the session
*/
setupTrackListeners() {
if (!this.rtviClient) return;
// Listen for new tracks starting
this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
// Only handle non-local (bot) tracks
if (!participant?.local) {
if (track.kind === 'audio') {
this.setupAudioTrack(track);
} else if (track.kind === 'video') {
this.setupVideoTrack(track);
}
}
});
// Listen for tracks stopping
this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
this.log(
`Track stopped event: ${track.kind} from ${
participant?.name || 'unknown'
}`
);
});
}
/**
* Set up an audio track for playback
* Handles both initial setup and track updates
*/
setupAudioTrack(track) {
this.log('Setting up audio track');
// Check if we're already playing this track
if (this.botAudio.srcObject) {
const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
if (oldTrack?.id === track.id) return;
}
// Create a new MediaStream with the track and set it as the audio source
this.botAudio.srcObject = new MediaStream([track]);
}
/**
* Set up a video track for display
* Handles both initial setup and track updates
*/
setupVideoTrack(track) {
this.log('Setting up video track');
const videoEl = document.createElement('video');
videoEl.autoplay = true;
videoEl.playsInline = true;
videoEl.muted = true;
videoEl.style.width = '100%';
videoEl.style.height = '100%';
videoEl.style.objectFit = 'cover';
// Check if we're already displaying this track
if (this.botVideoContainer.querySelector('video')?.srcObject) {
const oldTrack = this.botVideoContainer
.querySelector('video')
.srcObject.getVideoTracks()[0];
if (oldTrack?.id === track.id) return;
}
// Create a new MediaStream with the track and set it as the video source
videoEl.srcObject = new MediaStream([track]);
this.botVideoContainer.innerHTML = '';
this.botVideoContainer.appendChild(videoEl);
}
/**
* Initialize and connect to the bot
* This sets up the RTVI client, initializes devices, and establishes the connection
*/
async connect() {
try {
// Create a new Daily transport for WebRTC communication
const transport = new DailyTransport();
// Initialize the RTVI client with our configuration
this.rtviClient = new RTVIClient({
transport,
params: {
// The baseURL and endpoint of your bot server that the client will connect to
baseUrl: 'http://localhost:7860',
endpoints: {
connect: '/connect',
},
},
enableMic: true, // Enable microphone for user input
enableCam: false,
callbacks: {
// Handle connection state changes
onConnected: () => {
this.updateStatus('Connected');
this.connectBtn.disabled = true;
this.disconnectBtn.disabled = false;
this.log('Client connected');
},
onDisconnected: () => {
this.updateStatus('Disconnected');
this.connectBtn.disabled = false;
this.disconnectBtn.disabled = true;
this.log('Client disconnected');
},
// Handle transport state changes
onTransportStateChanged: (state) => {
this.updateStatus(`Transport: ${state}`);
this.log(`Transport state changed: ${state}`);
if (state === 'ready') {
this.setupMediaTracks();
}
},
// Handle bot connection events
onBotConnected: (participant) => {
this.log(`Bot connected: ${JSON.stringify(participant)}`);
},
onBotDisconnected: (participant) => {
this.log(`Bot disconnected: ${JSON.stringify(participant)}`);
},
onBotReady: (data) => {
this.log(`Bot ready: ${JSON.stringify(data)}`);
this.setupMediaTracks();
},
// Transcript events
onUserTranscript: (data) => {
// Only log final transcripts
if (data.final) {
this.log(`User: ${data.text}`);
}
},
onBotTranscript: (data) => {
this.log(`Bot: ${data.text}`);
},
// Error handling
onMessageError: (error) => {
console.log('Message error:', error);
},
onError: (error) => {
console.log('Error:', error);
},
},
});
// Set up listeners for media track events
this.setupTrackListeners();
// Initialize audio/video devices
this.log('Initializing devices...');
await this.rtviClient.initDevices();
// Connect to the bot
this.log('Connecting to bot...');
await this.rtviClient.connect();
this.log('Connection complete');
} catch (error) {
// Handle any errors during connection
this.log(`Error connecting: ${error.message}`);
this.log(`Error stack: ${error.stack}`);
this.updateStatus('Error');
// Clean up if there's an error
if (this.rtviClient) {
try {
await this.rtviClient.disconnect();
} catch (disconnectError) {
this.log(`Error during disconnect: ${disconnectError.message}`);
}
}
}
}
/**
* Disconnect from the bot and clean up media resources
*/
async disconnect() {
if (this.rtviClient) {
try {
// Disconnect the RTVI client
await this.rtviClient.disconnect();
this.rtviClient = null;
// Clean up audio
if (this.botAudio.srcObject) {
this.botAudio.srcObject.getTracks().forEach((track) => track.stop());
this.botAudio.srcObject = null;
}
// Clean up video
if (this.botVideoContainer.querySelector('video')?.srcObject) {
const video = this.botVideoContainer.querySelector('video');
video.srcObject.getTracks().forEach((track) => track.stop());
video.srcObject = null;
}
this.botVideoContainer.innerHTML = '';
} catch (error) {
this.log(`Error disconnecting: ${error.message}`);
}
}
}
}
// Initialize the client when the page loads
window.addEventListener('DOMContentLoaded', () => {
new ChatbotClient();
});

View File

@@ -1,98 +0,0 @@
body {
margin: 0;
padding: 20px;
font-family: Arial, sans-serif;
background-color: #f0f0f0;
}
.container {
max-width: 1200px;
margin: 0 auto;
}
.status-bar {
display: flex;
justify-content: space-between;
align-items: center;
padding: 10px;
background-color: #fff;
border-radius: 8px;
margin-bottom: 20px;
}
.controls button {
padding: 8px 16px;
margin-left: 10px;
border: none;
border-radius: 4px;
cursor: pointer;
}
#connect-btn {
background-color: #4caf50;
color: white;
}
#disconnect-btn {
background-color: #f44336;
color: white;
}
button:disabled {
opacity: 0.5;
cursor: not-allowed;
}
.main-content {
background-color: #fff;
border-radius: 8px;
padding: 20px;
margin-bottom: 20px;
}
.bot-container {
display: flex;
flex-direction: column;
align-items: center;
}
#bot-video-container {
width: 640px;
height: 360px;
background-color: #e0e0e0;
border-radius: 8px;
margin: 20px auto;
overflow: hidden;
display: flex;
align-items: center;
justify-content: center;
}
#bot-video-container video {
width: 100%;
height: 100%;
object-fit: cover;
}
.debug-panel {
background-color: #fff;
border-radius: 8px;
padding: 20px;
}
.debug-panel h3 {
margin: 0 0 10px 0;
font-size: 16px;
font-weight: bold;
}
#debug-log {
height: 200px;
overflow-y: auto;
background-color: #f8f8f8;
padding: 10px;
border-radius: 4px;
font-family: monospace;
font-size: 12px;
line-height: 1.4;
}

View File

@@ -1,15 +0,0 @@
# Daily Prebuilt Connection
The simplest way to connect to the chatbot using Daily's Prebuilt UI.
1. Start the bot server
```bash
python server/server.py
```
2. Visit http://localhost:7860
3. Allow microphone access when prompted
4. Start talking with the bot

View File

@@ -1,24 +0,0 @@
# Logs
logs
*.log
npm-debug.log*
yarn-debug.log*
yarn-error.log*
pnpm-debug.log*
lerna-debug.log*
node_modules
dist
dist-ssr
*.local
# Editor directories and files
.vscode/*
!.vscode/extensions.json
.idea
.DS_Store
*.suo
*.ntvs*
*.njsproj
*.sln
*.sw?

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