Spaces:
Sleeping
Sleeping
going to use chatgpt code and see if it works
Browse files
flow.txt
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| 1 |
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Create a gradio which acts a basic voice assient capable of executing task
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| 2 |
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make the applicatio in python and deplyoble on hugging face spaces
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| 3 |
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use the transformers api from hugging face where every possible
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| 4 |
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| 5 |
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Start by using fasterwhisper to convert speach into text; do this in chunks
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| 6 |
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detect if the user stops talking and use senteinet anaylsis to figure if the user stopped to think or asked a question
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| 7 |
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if the user aksed a question feed it to chat gpt-oss-120b
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| 8 |
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give chat gpt acees to a bunch of get info; get booking dates; create; booking; take order
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get a resoponce from chat gpt and play and conver it audio using Higgs Audio V2 [https://huggingface.co/bosonai/higgs-audio-v2-generation-3B-base]
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feed this audio responce back to gradio app; make sure all the models are avible and ready at the same time for real time conversation
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Creating a Voice Assistant with Gradio for Hugging Face Spaces
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This explanation details how to build a sophisticated voice assistant using Gradio that can be deployed on Hugging Face Spaces. The assistant will provide real-time interaction capabilities with speech-to-text conversion, intelligent conversation analysis, and natural-sounding responses.
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+
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## System Architecture
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The voice assistant consists of several key components working together:
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1. **Speech-to-Text (STT)** - Using Faster Whisper for efficient and accurate transcription
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2. **Conversation Analysis** - Sentiment analysis to detect user intent and conversation flow
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3. **Natural Language Processing** - GPT model integration for intelligent responses
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4. **Text-to-Speech (TTS)** - Higgs Audio V2 for high-quality voice synthesis
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5. **User Interface** - Gradio for a clean, accessible web interface
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## Detailed Implementation Steps
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### 1. Speech-to-Text Processing
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The system will use Faster Whisper, an optimized version of OpenAI's Whisper model, to convert user speech to text:
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- Implement chunk-based audio processing to handle continuous speech
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- Configure the model to process audio in near real-time (16kHz sampling rate)
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- Optimize for latency by using a smaller model variant for initial deployment
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- Implement adaptive silence detection to identify when a user has finished speaking
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- Batch process audio frames to balance accuracy and responsiveness
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### 2. Conversation Flow Analysis
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The system will use sentiment analysis and conversational cues to determine user intent:
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- Implement a pause detection algorithm that analyzes audio for natural breaks
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- Use transformer-based sentiment analysis to distinguish between:
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- Questions requiring information
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- Thinking pauses (user contemplating)
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- Statements or commands
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- Track conversation context to improve understanding of follow-up queries
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- Calculate confidence scores for detected intents to handle ambiguous cases
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```
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### 3. GPT Model Integration
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The system will use a powerful language model to generate contextually relevant responses:
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- Integrate with Hugging Face's `gpt-oss-120b` or equivalent open-source large language model
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- Provide the model with specialized system prompts for different tasks:
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- Information retrieval (get_info)
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- Booking management (get_booking_dates, create_booking)
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- Order processing (take_order)
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- Implement context management to maintain conversation history
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- Use function calling capabilities to execute specific tasks based on user requests
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- Apply rate limiting and token optimization for efficient resource usage
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### 4. Text-to-Speech Synthesis
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The system will convert text responses to natural-sounding speech:
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- Integrate Higgs Audio V2 model from Hugging Face for high-quality voice synthesis
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- Configure voice parameters (pitch, speed, style) for natural conversation
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- Implement streaming audio playback to minimize perceived latency
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- Cache common responses to improve performance
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- Add prosody and emphasis based on sentiment and content type
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### 5. Gradio Interface Implementation
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The system will use Gradio to create an intuitive, accessible user interface:
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- Design a clean interface with audio input and output components
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- Implement WebRTC for low-latency audio streaming
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- Add visual feedback indicators for system status (listening, processing, speaking)
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- Include text display of transcriptions and responses for accessibility
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- Provide controls for adjusting voice parameters and conversation settings
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- Ensure responsive design for both desktop and mobile use
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## Hugging Face Spaces Deployment
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To deploy the application on Hugging Face Spaces:
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1. Create a `requirements.txt` file with all necessary dependencies:
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- gradio
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- transformers
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- torch
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- faster-whisper
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- numpy
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- scipy
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- ffmpeg-python
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3. Implement model caching:
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- Use Hugging Face's model caching mechanisms to improve loading times
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- Implement progressive loading to make the interface available quickly
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4. Create a `README.md` with clear usage instructions and capabilities
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5. Add a Spaces SDK configuration file to specify resource requirements
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