Audio Processing

This skill should be used when working with pre-trained transformer models for natural language processing, computer vision, audio, or multimodal tasks. Use for text generation, classification, question answering, translation, summarization, image classification, object detection, speech recognition, and fine-tuning models on custom datasets.

Create SEO-optimized marketing content with consistent brand voice. Includes brand voice analyzer, SEO optimizer, content frameworks, and social media templates. Use when writing blog posts, creating social media content, analyzing brand voice, optimizing SEO, planning content calendars, or when user mentions content creation, brand voice, SEO optimization, social media marketing, or content strategy.

This skill should be used when the user asks to "write a post", "check my voice", "look up contact", "prepare for meeting", "weekly review", "track goals", or mentions personal brand, content creation, network management, or voice consistency.

This skill should be used when working with pre-trained transformer models for natural language processing, computer vision, audio, or multimodal tasks. Use for text generation, classification, question answering, translation, summarization, image classification, object detection, speech recognition, and fine-tuning models on custom datasets.

Process and generate multimedia content using Google Gemini API. Capabilities include analyze audio files (transcription with timestamps, summarization, speech understanding, music/sound analysis up to 9.5 hours), understand images (captioning, object detection, OCR, visual Q&A, segmentation), process videos (scene detection, Q&A, temporal analysis, YouTube URLs, up to 6 hours), extract from documents (PDF tables, forms, charts, diagrams, multi-page), generate images (text-to-image, editing, composition, refinement). Use when working with audio/video files, analyzing images or screenshots, processing PDF documents, extracting structured data from media, creating images from text prompts, or implementing multimodal AI features. Supports multiple models (Gemini 2.5/2.0) with context windows up to 2M tokens.

Create compelling story-format summaries using UltraThink to find the best narrative framing. Support multiple formats - 3-part narrative, n-length with inline links, abridged 5-line, or comprehensive via Foundry MCP. USE WHEN user says 'create story explanation', 'narrative summary', 'explain as a story', or wants content in Daniel's conversational first-person voice.

Ensure all communication matches brand voice and tone guidelines. Use when creating marketing copy, customer communications, public-facing content, or when users mention brand voice, tone, or writing style.

Emulate the user's personal writing voice and style patterns. Use when the user asks to write content in their voice, draft documents, compose messages, or requests "write this like me" or "in my style."

PyTorch library for audio generation including text-to-music (MusicGen) and text-to-sound (AudioGen). Use when you need to generate music from text descriptions, create sound effects, or perform melody-conditioned music generation.

Transcribe audio via OpenAI Audio Transcriptions API (Whisper).

ElevenLabs text-to-speech with mac-style say UX.

Corrects speech-to-text transcription errors in meeting notes, lectures, and interviews using dictionary rules and AI. Learns patterns to build personalized correction databases. Use when working with transcripts containing ASR/STT errors, homophones, or Chinese/English mixed content requiring cleanup.

Use when implementing speech-to-text, live transcription, or audio transcription. Covers SpeechAnalyzer (iOS 26+), SpeechTranscriber, volatile/finalized results, AssetInventory model management, audio format handling.

Reference — AVFoundation audio APIs, AVAudioSession categories/modes, AVAudioEngine pipelines, bit-perfect DAC output, iOS 26+ spatial audio capture, ASAF/APAC, Audio Mix with Cinematic framework

Build with OpenAI's stateless APIs - Chat Completions (GPT-5.2, GPT-5.1, o3, o4-mini), Realtime API (voice), Batch API (50% savings), Embeddings, Images (DALL-E 3), Audio (Whisper + TTS), and Moderation. Node.js SDK and fetch for Cloudflare Workers. Use when: implementing chat with GPT-5.2/5.1/o3/o4-mini, Realtime voice API (WebSocket), Batch API for cost savings, xhigh reasoning (GPT-5.2), streaming responses, function calling/tools, structured outputs, embeddings for RAG, DALL-E 3 images, Whisper transcription, TTS (13 voices), or troubleshooting rate limits (429), API keys (401), streaming errors.

Build conversational AI voice agents with ElevenLabs Platform using React, JavaScript, React Native, or Swift SDKs. Configure agents, tools (client/server/MCP), RAG knowledge bases, multi-voice, and Scribe real-time STT. Use when: building voice chat interfaces, implementing AI phone agents with Twilio, configuring agent workflows or tools, adding RAG knowledge bases, testing with CLI "agents as code", or troubleshooting deprecated @11labs packages, Android audio cutoff, CSP violations, dynamic variables, or WebRTC config. Keywords: ElevenLabs Agents, ElevenLabs voice agents, AI voice agents, conversational AI, @elevenlabs/react, @elevenlabs/client, @elevenlabs/react-native, @elevenlabs/elevenlabs-js, @elevenlabs/agents-cli, elevenlabs SDK, voice AI, TTS, text-to-speech, ASR, speech recognition, turn-taking model, WebRTC voice, WebSocket voice, ElevenLabs conversation, agent system prompt, agent tools, agent knowledge base, RAG voice agents, multi-voice agents, pronunciation dictionary, voice speed control, elevenlabs scribe, @11labs deprecated, Android audio cutoff, CSP violation elevenlabs, dynamic variables elevenlabs, case-sensitive tool names, webhook authentication, post-call webhook, webhook payload schema, ElevenLabs-Signature header, transcript null message, call_successful string, webhook cost credits USD, charging llm_price, user context extraction, llm_usage tokens, data_collection_results, evaluation_criteria_results, feedback thumb_rating, interrupted turn, source_medium, rag_retrieval_info, has_audio has_user_audio has_response_audio

Progressive disclosure guide to TabzChrome capabilities. This skill should be used when users ask about profiles, terminal management, browser automation, MCP tools, audio/TTS notifications, integration, debugging, API, or setup. Provides on-demand help organized by topic with references to detailed documentation.

Guide for implementing Google Gemini API audio capabilities - analyze audio with transcription, summarization, and understanding (up to 9.5 hours), plus generate speech with controllable TTS. Use when processing audio files, creating transcripts, analyzing speech/music/sounds, or generating natural speech from text.

Transcribes video audio using WhisperX, preserving original timestamps. Creates JSON transcript with word-level timing. Use when you need to generate audio transcripts for videos.

Write content that sounds natural, conversational, and authentically human - avoiding AI-generated patterns, corporate speak, and generic phrasing