Loading...
Loading...
Found 52 Skills
Terminal transcription of audio files and URLs with the Gladia CLI (gladia speech-to-text). Use when the user has gladia-cli installed, wants shell-based transcription, or asks an agent to transcribe audio then answer questions about the content. For audio intelligence features not available as CLI flags, use the SDK skills instead.
Automatic Speech Recognition (ASR). Uses Volcano Engine BigModel ASR for speech recognition, with two available modes: Express Edition (≤2h/100MB, synchronous fast response) and Standard Edition (≤5h, asynchronous recognition). It supports Feishu voice messages, local audio files and audio URLs. Use this skill when you receive voice messages or audio attachments (.ogg/.mp3/.wav).
Use when live speech translation is needed with Alibaba Cloud Model Studio Qwen LiveTranslate models, including bilingual meetings, realtime interpretation, and speech-to-speech or speech-to-text translation flows.
Minimal realtime ASR smoke test for Model Studio Qwen ASR Realtime.
Use when creating cloned voices with Alibaba Cloud Model Studio CosyVoice customization models, especially cosyvoice-v3.5-plus or cosyvoice-v3.5-flash, from reference audio and then reusing the returned voice_id in later TTS calls.
ML-powered Karaoke app in Rust using Bevy, WhisperX, and Demucs for stem separation, lyrics transcription, and pitch scoring.
Use when asked to normalize audio volume, match loudness, or apply peak/RMS normalization to audio files.
Production-grade FFmpeg video/audio processing. Convert, compress, trim, merge, resize, and extract audio from media files with progress tracking, comprehensive error handling, and safety limits.
Assemble final video from generated clips, audio, and assets using FFmpeg or Remotion. Handles concatenation, audio mixing, transitions, titles, and export. Use when combining multiple production outputs into a final deliverable.
Audio spectrograms/features (mel, chroma, MFCC) via CLI.
Text-to-speech synthesis via Google Cloud Text-to-Speech API — MP3 output, configurable language and voice, voice listing.
Speaker diarization — identifies and tracks who is speaking at each moment in an audio stream, using provider-delegated labels or local offline clustering.