diff --git a/gallery/index.yaml b/gallery/index.yaml index 9776061a2..8244716e7 100644 --- a/gallery/index.yaml +++ b/gallery/index.yaml @@ -46720,3 +46720,348 @@ - model: gemma-4-26b-a4b-it-heretic-apex-i-balanced - model: gemma-4-26b-a4b-it-heretic-apex-i-compact - model: gemma-4-26b-a4b-it-heretic-apex-i-mini +- name: audio-cpp-supertonic + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://huggingface.co/audio-cpp/audio.cpp-gguf + - https://github.com/0xShug0/audio.cpp + description: | + Supertonic 3 (audio.cpp, GGUF orig) - fast English text to speech with no + reference clip required, served by the audio-cpp backend. Covers both the + TTS and the TTSStream RPC, so /v1/audio/speech works streaming and + non-streaming. + + The orig (F32) package is deliberate. Upstream's tested-status table in + docs/gguf.md records supertonic as Pass on orig, untested on 16-bit, and + "No (unsupported weight dtype)" on q8_0; the f16 package aborted the + process inside ggml_concat when tested locally. The backend refuses + anything but f32/i64 weights for this family, so a wrong build fails at + load with a clear message rather than crashing. + license: apache-2.0 + tags: + - audio-cpp + - supertonic + - tts + - text-to-speech + - gguf + - ggml + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - tts + name: audio-cpp-supertonic + parameters: + model: audio-cpp/supertonic-3-orig.gguf + files: + - filename: audio-cpp/supertonic-3-orig.gguf + sha256: af814486a0bc9513fb36afabd9b1155ad14fb2c36a107ac6ffe62ea9adafb662 + uri: huggingface://audio-cpp/audio.cpp-gguf/Supertonic-3-GGUF/supertonic-3-orig.gguf +- name: audio-cpp-chatterbox + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://huggingface.co/audio-cpp/audio.cpp-gguf + - https://github.com/0xShug0/audio.cpp + description: | + Chatterbox (audio.cpp, Q8_0) - multilingual zero-shot voice cloning, served + by the audio-cpp backend. + + A reference clip is REQUIRED. This build advertises only the cloning and + voice-conversion tasks, not plain TTS, so every request must set the TTS + voice field to the path of a WAV file on the server; a voice that is not a + file on disk is treated as a named preset and the request is refused with a + message saying so. + + Q8_0 rather than 16-bit because upstream's tested-status table records the + same result for both (Pass, ASR match with drift) and this build is half + the size. + license: mit + tags: + - audio-cpp + - chatterbox + - tts + - text-to-speech + - voice-cloning + - gguf + - ggml + - quantized + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - tts + name: audio-cpp-chatterbox + parameters: + model: audio-cpp/chatterbox-q8_0.gguf + files: + - filename: audio-cpp/chatterbox-q8_0.gguf + sha256: d586dd1aa59613cab8046176fb7ca5ba191c02a9b10ffa5b0d892ed22b470656 + uri: huggingface://audio-cpp/audio.cpp-gguf/Chatterbox-GGUF/chatterbox-q8_0.gguf +- name: audio-cpp-citrinet-asr + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://huggingface.co/audio-cpp/audio.cpp-gguf + - https://github.com/0xShug0/audio.cpp + description: | + Citrinet ASR (audio.cpp, Q8_0) - compact English speech to text, served by + the audio-cpp backend through /v1/audio/transcriptions. 40 MB, offline + transcription only. Q8_0 is the only package upstream publishes and tests + for this family, and it is recorded as a clean Pass. + license: apache-2.0 + tags: + - audio-cpp + - citrinet + - asr + - speech-recognition + - transcription + - gguf + - ggml + - quantized + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - transcript + name: audio-cpp-citrinet-asr + parameters: + model: audio-cpp/citrinet-asr-q8_0.gguf + files: + - filename: audio-cpp/citrinet-asr-q8_0.gguf + sha256: e1b9187ecc1c5770f5785f1d1e797ff2bf1f3692c78a5328684275aa729c6368 + uri: huggingface://audio-cpp/audio.cpp-gguf/Citrinet-ASR-GGUF/citrinet-asr-q8_0.gguf +- name: audio-cpp-nemotron-asr + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://huggingface.co/audio-cpp/audio.cpp-gguf + - https://github.com/0xShug0/audio.cpp + description: | + Nemotron 3.5 ASR Streaming 0.6B (audio.cpp, F16) - streaming English speech + to text, served by the audio-cpp backend. This is the family behind the + live transcription RPC, so it drives the realtime transcription session as + well as ordinary /v1/audio/transcriptions calls. + + F16 rather than Q8_0 because upstream's tested-status table records Q8_0 as + "Pass (minor filler drift)" while 16-bit is a clean Pass, and drift in a + transcript is drift in the output a caller reads. + license: other + tags: + - audio-cpp + - nemotron + - asr + - speech-recognition + - transcription + - streaming + - gguf + - ggml + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - transcript + name: audio-cpp-nemotron-asr + parameters: + model: audio-cpp/nemotron-3.5-asr-streaming-0.6b-f16.gguf + files: + - filename: audio-cpp/nemotron-3.5-asr-streaming-0.6b-f16.gguf + sha256: 8bef32306425ebe62484160ee1450fa477f92d0e6e03b13cf4e5fa158254137e + uri: huggingface://audio-cpp/audio.cpp-gguf/Nemotron-3.5-ASR-Streaming-0.6B-GGUF/nemotron-3.5-asr-streaming-0.6b-f16.gguf +- name: audio-cpp-sortformer-diarization + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://huggingface.co/audio-cpp/audio.cpp-gguf + - https://github.com/0xShug0/audio.cpp + description: | + Sortformer Diarization 4-speaker v1 (audio.cpp, Q8_0) - speaker diarization + for up to four speakers, served by the audio-cpp backend through + /v1/audio/diarization. Returns per-segment start, end and speaker label; it + does not transcribe, so pair it with an ASR model for text. + + Q8_0 because upstream records it as a clean Pass, the same as 16-bit, at + two thirds of the size. + license: other + tags: + - audio-cpp + - sortformer + - diarization + - speaker-diarization + - gguf + - ggml + - quantized + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - diarization + name: audio-cpp-sortformer-diarization + parameters: + model: audio-cpp/sortformer-diar-4spk-v1-q8_0.gguf + files: + - filename: audio-cpp/sortformer-diar-4spk-v1-q8_0.gguf + sha256: 84744b047296daf2177aed9dd2735497152ff83f49ec170d2a6a18cbcb93ef19 + uri: huggingface://audio-cpp/audio.cpp-gguf/Sortformer-Diar-4spk-v1-GGUF/sortformer-diar-4spk-v1-q8_0.gguf +- name: audio-cpp-htdemucs + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://huggingface.co/audio-cpp/audio.cpp-gguf + - https://github.com/0xShug0/audio.cpp + description: | + HTDemucs (audio.cpp, F16) - four-stem music source separation (drums, bass, + other, vocals), served by the audio-cpp backend through /audio/transform. + One run produces all four stems; ask for a single one with the + params[stem] form field, and read the X-Audio-Stems response header for the + URLs of the rest. + + F16 rather than Q8_0 because upstream's tested-status table records Q8_0 as + "Pass (drift)" while 16-bit is a clean Pass, and separation drift is + audible in an isolated stem in a way it is not in a mix. + license: mit + tags: + - audio-cpp + - htdemucs + - demucs + - source-separation + - stem-separation + - audio-transform + - gguf + - ggml + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - audio_transform + name: audio-cpp-htdemucs + parameters: + model: audio-cpp/htdemucs-f16.gguf + files: + - filename: audio-cpp/htdemucs-f16.gguf + sha256: 27763c59d7e65f64bbb326dc0c15beec952291954b70a813b7f4b83ebe6bd0c9 + uri: huggingface://audio-cpp/audio.cpp-gguf/HTDemucs-GGUF/htdemucs-f16.gguf +- name: audio-cpp-stable-audio-sfx + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://huggingface.co/audio-cpp/audio.cpp-gguf + - https://github.com/0xShug0/audio.cpp + description: | + Stable Audio 3 Small SFX (audio.cpp, Q8_0) - text to sound effects, served + by the audio-cpp backend through /v1/sound-generation. Set + duration_seconds to control the length. + + Text prompt only. The family also has an audio-conditioned editing route + upstream, which is not wired here: it aborts the process on the pinned + revision, and no field on the sound-generation request can reach it in any + case. + + Q8_0 because upstream records both 16-bit and Q8_0 as "Pass (drift)", so + the smaller build costs nothing extra. + license: other + tags: + - audio-cpp + - stable-audio + - sound-generation + - sfx + - text-to-audio + - gguf + - ggml + - quantized + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - sound_generation + name: audio-cpp-stable-audio-sfx + parameters: + model: audio-cpp/stable-audio-3-small-sfx-q8_0.gguf + files: + - filename: audio-cpp/stable-audio-3-small-sfx-q8_0.gguf + sha256: 5bb1ec653134e63dac46264e336ae198a2590195fc996fcd0605021215e9b26f + uri: huggingface://audio-cpp/audio.cpp-gguf/Stable-Audio-3-Small-SFX-GGUF/stable-audio-3-small-sfx-q8_0.gguf +- name: audio-cpp-forced-aligner + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://huggingface.co/audio-cpp/audio.cpp-gguf + - https://github.com/0xShug0/audio.cpp + description: | + Qwen3 Forced Aligner 0.6B (audio.cpp, Q8_0) - aligns a transcript you + already have against the audio, served by the audio-cpp backend. + + It is driven through /v1/audio/transcriptions, not a dedicated endpoint: + put the known transcript in the prompt field and the model returns it with + word timings instead of recognising anything. Both a transcript and a + language are required, so this entry defaults language to en; override it + with the language form field for other languages. + + Q8_0 because upstream records both 16-bit and Q8_0 as a clean Pass. + license: apache-2.0 + tags: + - audio-cpp + - qwen3 + - forced-alignment + - alignment + - transcription + - gguf + - ggml + - quantized + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - transcript + name: audio-cpp-forced-aligner + parameters: + language: en + model: audio-cpp/qwen3-forced-aligner-0.6b-q8_0.gguf + files: + - filename: audio-cpp/qwen3-forced-aligner-0.6b-q8_0.gguf + sha256: 75209490b11cec2b0db749ca5f4ff92266f58efd30f7fd04d9eb2a3ac9cc929f + uri: huggingface://audio-cpp/audio.cpp-gguf/Qwen3-ForcedAligner-0.6B-GGUF/qwen3-forced-aligner-0.6b-q8_0.gguf +- name: audio-cpp-silero-vad + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://github.com/0xShug0/audio.cpp + - https://github.com/snakers4/silero-vad + description: | + Silero VAD (audio.cpp) - voice activity detection over /v1/vad. The model + ships inside the audio-cpp backend package, so installing this entry + downloads nothing. + license: mit + tags: + - audio-cpp + - silero + - vad + - voice-activity-detection + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - vad + name: audio-cpp-silero-vad + options: + - family:silero_vad + parameters: + model: bundled:silero_vad +- name: audio-cpp-marblenet-vad + url: github:mudler/LocalAI/gallery/virtual.yaml@master + urls: + - https://github.com/0xShug0/audio.cpp + description: | + MarbleNet VAD (audio.cpp) - NeMo voice activity detection over /v1/vad, an + alternative to Silero with different segment boundaries. The model ships + inside the audio-cpp backend package, so installing this entry downloads + nothing. + license: other + tags: + - audio-cpp + - marblenet + - nemo + - vad + - voice-activity-detection + last_checked: "2026-07-27" + overrides: + backend: audio-cpp + known_usecases: + - vad + name: audio-cpp-marblenet-vad + options: + - family:marblenet_vad + parameters: + model: bundled:marblenet_vad