DePasqualeOrg/mlx-audio-plus tiene un índice de salud de 53 sobre 100, lo que lo sitúa en la banda Moderado. Su puntuación más alta es Vitality (64/100) y la más baja, Community & Adoption (29/100). Se actualizó por última vez hace 82 días. 2 personas concentran la mayor parte del trabajo reciente.
53
global / 100
Moderado
Índice de salud del software
Las métricas se agrupan en categorías ponderadas sobre una escala estandarizada de 1 a 100. El resultado global parte de su media ponderada, calibrada contra la distribución del registro público para que las bandas tengan significado percentil; cuando la evidencia pública activa la Política de Jurisdicciones de Alto Riesgo, la calificación se ajusta y recibe un límite «En riesgo» de 34.
53
Excepcional93-100El nivel más alto del registro (≈ el 5% superior); cumple prácticamente todos los criterios evaluados
Excelente80-92Sólido en todos los frentes; carencias menores
Bueno65-79Saludable; carencias limitadas y manejables
Moderado50-64Aceptable con carencias notables; se recomienda revisión
Débil35-49Debilidades sustanciales en varias áreas
En riesgo20-34Debilidades significativas; su adopción exige cautela
Crítico1-19Problemas graves (proyecto abandonado, un solo mantenedor, sin higiene)
Perfil de puntuación
Cada eje es una categoría. La forma importa más que la media: un proyecto sano llena toda la figura, mientras que un perfil de picos y cráteres indica que la fortaleza en una dimensión enmascara el riesgo en otra.
El resultado global ponderado 52 se calibra a 53 en la escala publicada del índice (calibración del registro 2026-08-02).
0 seguidores189 repositorios públicosdesde ene 2017
Este repositorio pertenece a una cuenta personal. Un proyecto con un único propietario conlleva más riesgo de continuidad que uno respaldado por una organización.
Dependencias indirectas libres de avisos conocidos — ninguna dependencia indirecta tiene un aviso conocido
32.3/40
Sin avisos pendientes — 1 paquete(s) con avisos sin atender más allá de 90 días; el más antiguo publicado hace 219 días
Datos de entrada utilizados
source
osv
advisories
7
affected_packages
1
assessed_packages
50
unassessed_packages
0
affected_by_severity
critical 1
direct_affected_packages
1
Se cotejó el cierre de dependencias en tiempo de ejecución de pypi:mlx-audio-plus@0.1.8 —lo que arrastra la instalación del paquete publicado—: 50 paquetes. No se analiza la alcanzabilidad.
¿Hasta qué punto está el repositorio preparado para desarrollarse y mantenerse con agentes de codificación de IA? Tiene un peso deliberadamente pequeño (4%): las herramientas para agentes son una señal real de mantenimiento, pero un repositorio sin ninguna puede alcanzar igualmente 100/100.
Star history unavailable: GitHub GraphQL error: Resource not accessible by personal access token
Community profile unavailable
GitHub dependency-graph SBOM unavailable (404); the dependency graph may be disabled for this repository
Más detalle
Historial de estrellas y forks 0 ★ / 6 ⇿
0Estrellas
6Forks
2Versiones
Cuándo se añadió cada estrella y fork, recopilado de GitHub y agrupado por día. El crecimiento acumulado se sitúa justo encima de las adiciones diarias que lo componen, de modo que ambos se leen en conjunto: la acumulación orgánica sostenida no se parece en nada a un pico abrupto y efímero. Cuando esa diferencia es medible, se informa como autenticidad del crecimiento.
Mayor 0Menor 0Parche 2
OpenSSF Scorecard 3.3 / 10
3.3agregado
Evaluación de seguridad independiente y agnóstica en cuanto a herramientas, procedente del proyecto de código abierto OpenSSF Scorecard. Cada comprobación premia una práctica de seguridad, no la herramienta de un proveedor concreto. Las comprobaciones que Scorecard no pudo determinar se marcan como n/d y se excluyen de la puntuación de seguridad (nunca se cuentan como cero).Scorecard v5.5.0 · 2026-07-31 11:17 UTC
No fue posible recopilar el conjunto de dependencias resuelto para este informe: GitHub dependency-graph SBOM unavailable (404); the dependency graph may be disabled for this repository
Avisos de dependencias 1
Instalar pypi:mlx-audio-plus@0.1.8 arrastra 50 paquetes, directos y transitivos: 1 tienen avisos conocidos, de los cuales 1 son dependencias directas.
Paquete
Versión
Relación
Gravedad
Avisos
Corregido en
transformers
4.57.6
directa
crítica
7
5.5.0
Un aviso significa que la versión registrada en el grafo de dependencias cae dentro del rango afectado de un aviso. No se analiza la alcanzabilidad, y el grafo incluye fijaciones de desarrollo y prueba: un hallazgo puede referirse al utillaje y no al software distribuido.
Informe JSON sin procesar legible por máquina
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"body": "* fix long audio processing\n\n* Add token tracking and performance metrics to transcription generation\n\n* qwen3asr fix total tokens\n\n* format",
"is_bot": false,
"headline": "Fix ALMs max tokens and chunking (#474)",
"author_name": "Prince Canuma",
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"committed_at": "2026-02-28T12:59:03Z",
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"oid": "c7fe57473fb85c99735c77f02b45ea5d99eea086",
"body": null,
"is_bot": false,
"headline": "Restore encoder_config to preserve voice clone accents. (#461)",
"author_name": "orbitalquark",
"author_login": "orbitalquark",
"committed_at": "2026-02-28T12:58:13Z",
"body_truncated": false,
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"oid": "18b21e9651d3d840ca0a9b096d49a264fe3dff58",
"body": "* Add python-multipart dependency to pyproject.toml for multipart form handling\n\n* Refactor tokenizer usage in Whisper model files\n\n- Removed direct imports and references to the Tokenizer class in favor of using model methods for tokenizer retrieval.\n- Updated function signatures to accept a generi\n[…]\nrnal logic to reflect the new parameter names, ensuring proper handling of audio chunk durations.\n- Adjusted test cases to align with the updated parameter names, maintaining test integrity.\n\n* format",
"is_bot": false,
"headline": "Fix server (STT, TTS) (#460)",
"author_name": "Prince Canuma",
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"committed_at": "2026-02-28T12:58:08Z",
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"oid": "10470ee9e8f6618978bd0d6387422e84c21b3f9f",
"body": "* fix chatterbox load\n\n* format",
"is_bot": false,
"headline": "Fix chatterbox load (#455)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-02-28T12:55:24Z",
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"oid": "4970740aa66ae91040f1f1c12d5f0ff7bb19f132",
"body": "* Add Qwen3-ASR model and configurations\n\n- Introduced Qwen3-ASR model with audio encoder and transformer layers.\n- Added configuration classes for audio encoder, text decoder, and model settings.\n- Updated generate_transcription function to print segment text for debugging.\n- Enhanced model remappi\n[…]\nerstanding of the expected behavior for both aligned and non-aligned weights.\n- Removed redundant variable assignments to streamline the test code, improving maintainability and readability.\n\n* format",
"is_bot": false,
"headline": "Add Qwen3-ASR (#454)",
"author_name": "Prince Canuma",
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"committed_at": "2026-02-28T12:55:21Z",
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"oid": "3e9a3389d217dec303dc1e7dc56df77dfa07c075",
"body": null,
"is_bot": false,
"headline": "Update pyproject.toml for poetry support.",
"author_name": "Lucas Newman",
"author_login": "lucasnewman",
"committed_at": "2026-02-28T12:52:58Z",
"body_truncated": false,
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"body": "* Refactor audio utilities for volume normalization and random segment selection\n\n* support loading reference audio from file paths or mx.array\n\n* remove old examples\n\n* Allowing reference audio to be loaded from file paths or mx.array across tts\n\n* migrate to tfmx v5\n\n* revert\n\n* Refactor audio dec\n[…]\nor Qwen3 TTS model to track calls to speech_tokenizer.decode instead of streaming_decode\n\n* Add tests for Qwen3 TTS streaming decode behavior, including tracking of decode calls and chunk size control",
"is_bot": false,
"headline": "Refactor audio load (#445)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-02-28T12:29:01Z",
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"body": "* Update mel spectrogram padding calculation in Qwen3-TTS to align with PyTorch standards\n\n* Increase context size in Qwen3-TTS model for smoother audio transitions, improving decoding accuracy with minimal error.\n\n* Refactor text embedding handlingg\n\n- Updated the text embedding retrieval process t\n[…]\nhe deprecation of loading models from pretrained weights directly.\n\n* remove unused\n\n* Replaced custom sampling implementations with utility functions from `mlx_lm.sample_utils`\n\n* format\n\n* fix tests",
"is_bot": false,
"headline": "[Qwen3-TTS] Fix some Custom Voices producing silence with 0.6B (#444)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-02-28T12:28:58Z",
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"body": "* Fix: improve import error hadling\n\n* Update mlx_audio/utils.py\n\n---------\n\nCo-authored-by: Prince Canuma <prince.gdt@gmail.com>",
"is_bot": false,
"headline": "Fix: improve import error hadling (#443)",
"author_name": "zer0",
"author_login": "reinexworldc",
"committed_at": "2026-02-28T12:28:55Z",
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"body": "* Enhance Qwen3-TTS model with streaming audio decoding and memory optimizations\n\n- Implemented streaming decoding in the speech tokenizer to handle long sequences with reduced memory usage, yielding audio chunks as they're decoded.\n- Updated the audio decoding process in the Model class to utilize \n[…]\n\n- Updated GenerationResult to include streaming state indicators for better tracking of audio chunks.\n\n* Increase context size for smoother audio transitions in Qwen3-TTS streaming model\n\n* fix tests",
"is_bot": false,
"headline": "Qwen3-TTS: Add streaming and optimise peak usage (#435)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-02-28T12:28:52Z",
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"body": "…arkers (#432)\n\n- Adjusted Python version resolution markers to include support for versions 3.11 and 3.13.\n- Updated numpy dependency versions based on Python version markers, introducing version 2.3.5 for Python 3.11 and above, while retaining version 2.2.6 for earlier versions.\n- Updated scipy an\n[…]\nencies similarly to align with the new numpy versioning strategy.\n- Incremented huggingface-hub version to 1.3.3 and setuptools to 80.10.1, ensuring compatibility with the latest package requirements.",
"is_bot": false,
"headline": "Update uv.lock to reflect dependency version changes and resolution m…",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-31T07:02:25Z",
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"body": "…ing (#431)\n\n- Removed `setuptools-scm` from the build requirements in `pyproject.toml`.\n- Changed license format to a simpler string representation.\n- Updated the GitHub Actions workflow to use the latest version tag for the PyPI publish action and corrected the parameter name from `packages_dir` to `packages-dir`.",
"is_bot": false,
"headline": "Update pyproject.toml and GitHub Actions workflow for package publish…",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-31T07:02:21Z",
"body_truncated": false,
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"body": null,
"is_bot": false,
"headline": "Move audio path/format prints under verbose flag (#429)",
"author_name": "Wlad Paiva",
"author_login": "wladpaiva",
"committed_at": "2026-01-31T07:02:21Z",
"body_truncated": false,
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"oid": "6f1d01def48e93256996ffbedd9641fae4712d77",
"body": "…#428)",
"is_bot": false,
"headline": "Update MANIFEST.in to remove leading dot from requirements.txt path (…",
"author_name": "Prince Canuma",
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"committed_at": "2026-01-31T07:02:21Z",
"body_truncated": false,
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"oid": "d5906058cbb49a9957506e2b7fcbc1731914ff5b",
"body": "* Refactor audio input handling in transcription methods to support multiple formats\n\n- Updated the `generate_transcription` and `Model` class methods to accept audio input as either a file path (str/Path) or an audio waveform (mx.array).\n- Enhanced documentation to clarify the expected input types \n[…]\nc1` to `5.0.0rc3` and `mlx-lm` version from `0.30.2` to `0.30.5` for improved compatibility.\n- Updated `uv.lock` to reflect the new dependency versions and added upload timestamps for package sources.",
"is_bot": false,
"headline": "Fix stt array io (#426)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-31T07:02:20Z",
"body_truncated": true,
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"oid": "09f7c1268889ad6892e3d99772338b8854810370",
"body": "* Fix convert skipping subdirs\n\n* Add model_quant_predicate method to Qwen3-TTS and update apply_quantization logic to handle dictionary results. This enhances quantization control and improves model-specific predicate handling.\n\n* Addd ICL voice cloning support and improve audio processing [Qwen3-T\n[…]\ndicate for clarity.\n- Consolidated the logic for loading tokenizer weights by utilizing mx.load, improving efficiency.\n- Ensured proper evaluation of speech_tokenizer parameters after loading weights.",
"is_bot": false,
"headline": "Fix qwen3 tts quants (silence in VC and word precision) (#407)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-31T07:02:12Z",
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"oid": "6b41a8742c03e3ebadd5547f0a4e1e7d9c20f2a0",
"body": "…rs from quantization (#398)\n\nCo-authored-by: Prince Canuma <prince.gdt@gmail.com>",
"is_bot": false,
"headline": "fix: Add model_quant_predicate to Qwen3-TTS to exclude embedding laye…",
"author_name": "Aron Homberg",
"author_login": "kyr0",
"committed_at": "2026-01-31T07:02:12Z",
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"oid": "33e64f230189aecf0f8a5bafe1ae8b956096a042",
"body": "…udio (#417)\n\n* Manual depthwise conv1d via accumulation over kernel positions.\n\n Avoids the Metal depthwise_conv_1d kernel which may not be available\n on all Apple Silicon hardware. Accumulates one kernel position at a\n time to avoid materializing a large [B, T_out, K, C] tens\n[…]\n- Introduced a prefill step size parameter to optimize memory usage during token generation.\n- Updated progress bars for both encoding and generation processes to improve user experience and feedback.",
"is_bot": false,
"headline": "[VibeVoice-ASR] Fix Metal kernel crash and optimize memory for long a…",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-31T07:02:11Z",
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{
"oid": "790901a08717f64cba0b6a832c778e6bbb8bee92",
"body": "Signed-off-by: Salman Muin Kayser Chishti <13schishti@gmail.com>",
"is_bot": false,
"headline": "Upgrade GitHub Actions to latest versions (#419)",
"author_name": "Salman Chishti",
"author_login": "salmanmkc",
"committed_at": "2026-01-31T07:02:11Z",
"body_truncated": false,
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"oid": "a982744f914cb4a05bfa6172ffaa3e9a6ab89457",
"body": "Signed-off-by: Salman Muin Kayser Chishti <13schishti@gmail.com>",
"is_bot": false,
"headline": "Upgrade GitHub Actions for Node 24 compatibility (#418)",
"author_name": "Salman Chishti",
"author_login": "salmanmkc",
"committed_at": "2026-01-31T07:02:11Z",
"body_truncated": false,
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"oid": "db84261ffe852257a10c874c65697c9f14ebfce0",
"body": "* Enhance README with installation and usage examples\n\nAdded installation instructions for pip and uv tool, and included voice cloning example with Qwen-TTS.\n\n* Move voice cloning to be with the rest of the qwen3-tts part\n\nRemoved redundant voice cloning section and retained relevant information.\n\n* Move Qwen3-tts instructions to its own README\n\n* Correct speaker names in Qwen3-tts README\n\n---------\n\nCo-authored-by: Prince Canuma <prince.gdt@gmail.com>",
"is_bot": false,
"headline": "Enhance README with installation and usage examples (#404)",
"author_name": "Rahim Nathwani",
"author_login": "rahimnathwani",
"committed_at": "2026-01-31T07:02:11Z",
"body_truncated": false,
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"body": "…406)",
"is_bot": false,
"headline": "Fix issue of ref_audio not loading prior to inference with server. (#…",
"author_name": "byMAR.CO",
"author_login": "BuffMcBigHuge",
"committed_at": "2026-01-31T07:02:04Z",
"body_truncated": false,
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"oid": "f272c912dc9f941500136d92bbaa6ff3121aff59",
"body": null,
"is_bot": false,
"headline": "Unify Mimi implementation. (#403)",
"author_name": "Lucas Newman",
"author_login": "lucasnewman",
"committed_at": "2026-01-31T07:02:03Z",
"body_truncated": false,
"is_coding_agent": false
},
{
"oid": "be484555d5bda99f55cad403500f428de0a837e0",
"body": null,
"is_bot": false,
"headline": "Fix Whisper",
"author_name": "Anthony DePasquale",
"author_login": "DePasqualeOrg",
"committed_at": "2026-01-24T08:10:55Z",
"body_truncated": false,
"is_coding_agent": false
},
{
"oid": "3092c5d533e759a660ad864a2e82461acd32d42f",
"body": null,
"is_bot": false,
"headline": "Fix VibeVoice sanitization",
"author_name": "Anthony DePasquale",
"author_login": "DePasqualeOrg",
"committed_at": "2026-01-24T07:58:52Z",
"body_truncated": false,
"is_coding_agent": false
},
{
"oid": "1b2ae0b6c7241746e1ac433bb7340b5f30a437ab",
"body": null,
"is_bot": false,
"headline": "Update TTS commands in README to include language code option (#401)",
"author_name": "Rudolf Olah",
"author_login": "rudolfolah",
"committed_at": "2026-01-24T07:49:17Z",
"body_truncated": false,
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{
"oid": "df70ce3b3cd8f0eda12f34d437ecc2f740c9ca8b",
"body": "* Add unit tests for Qwen3-TTS reflect padding and mel spectrogram functionality\n\n* Enhance mel spectrogram tests in Qwen3-TTS\n\n- Introduced new tests to verify mel spectrogram output with center padding and slaney normalization.\n- Added checks for expected output shapes and values for various input\n[…]\nroduces correct boundary frame values and that a 1kHz sine wave activates the appropriate mel bins.\n- Updated existing tests to improve clarity and accuracy in verifying mel spectrogram functionality.",
"is_bot": false,
"headline": "Qwen3-TTS speaker embedding tests (#396)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:49:13Z",
"body_truncated": true,
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{
"oid": "e73d7e5fe21644a260a78903cc24594ee99ae490",
"body": "* Add vibevoice ASR\n\n* format\n\n* Refactor file copying logic in copy_model_files to exclude specific model files and remove redundant code for supporting file copying.\n\n* Refactor tokenizer usage in VibeVoice ASR model to standardize access and improve clarity. Introduce model_quant_predicate for qu\n[…]\narity and consistency. Add new test initialization file.\n\n* Add speaker_id to JSON output in save_as_json function\n\n* Add comprehensive unit tests for VibeVoice ASR model\n\n* format\n\n* update gitignore",
"is_bot": false,
"headline": "Add Vibevoice ASR (#389)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:49:13Z",
"body_truncated": true,
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{
"oid": "166332095a0e3f4774e53075f80456f34f36064a",
"body": "* Fix mel filters\n\n* Refactor speaker encoder to implement reflect padding",
"is_bot": false,
"headline": "Fix Qwen3-TTS Base Voice Cloning (#394)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:49:06Z",
"body_truncated": false,
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"oid": "787dfd5673e88ed17eefd040b1c13555907537c4",
"body": "* Refactor causal padding logic in CausalTransposeConv1d. Trim from the right for causal behavior instead of using left and right padding calculations.\n\n* Update package versions in uv.lock\n\n- Bump mlx-audio version from 0.3.0rc1 to 0.3.0.\n- Change mlx-lm version specifier from ==0.30.2 to >=0.30.2.\n- Update transformers version specifier from ==5.0.0rc1 to >=5.0.0rc1.",
"is_bot": false,
"headline": "Fix Qwen3-TTS tail artifacts (#391)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:49:06Z",
"body_truncated": false,
"is_coding_agent": false
},
{
"oid": "c652a6251e7bc6f0f09a9927061d2eecd83b8190",
"body": "* Fix speaker embedding extraction in Qwen3-TTS model to ensure correct input shape for speaker IDs\n\n* bump version",
"is_bot": false,
"headline": "Fix speaker embedding extraction in Qwen3-TTS model (#390)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:49:02Z",
"body_truncated": false,
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"oid": "bea96bbd032b8f071e30b33fc13313df6c67eb80",
"body": "* Add qwen3-tts\n\n* add custom generate and fix loading logic\n\n* fix loading\n\n* fix talker\n\n* update readme\n\n* refactor\n\n* refactor naming to qwen3_tts\n\n* Add 'instruct' parameter to audio generation functions for enhanced voice customization\n\n* Update README to reflect changes in model loading from \n[…]\no 'load_model'\n\n* Add numba dependency to pyproject.toml and uv.lock for librosa compatibility\n\n* add tests\n\n* Add 'instruct' parameter to generate_audio function for improved audio generation control",
"is_bot": false,
"headline": "Add Qwen3-TTS (#388)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:48:56Z",
"body_truncated": true,
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{
"oid": "fcb1ffd9e3cd09d0abd2e8459f5d5f402d97c540",
"body": "* Add output path parameter to generate_audio function\n\n- Introduced an optional output_path argument to specify the directory for saving audio files.\n- Updated documentation to include output_path in the function signature and argument parsing.\n- Implemented logic to create the output directory if it does not exist.\n\n* unpin mlx-lm and tfms\n\n* Update README to include example for saving audio to a specific directory",
"is_bot": false,
"headline": "Update generate output path (#385)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:48:50Z",
"body_truncated": false,
"is_coding_agent": false
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"oid": "2a282e04ecaeadbcad3f0b3eb8e00d8e0f3564fb",
"body": null,
"is_bot": false,
"headline": "add streaming decoding to snac and orpheus (#384)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:48:42Z",
"body_truncated": false,
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},
{
"oid": "43e12d80b59d4d236eb72cb05441d4aa07aff45f",
"body": null,
"is_bot": false,
"headline": "Refactor model path retrieval in get_model_path",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:48:42Z",
"body_truncated": false,
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{
"oid": "7c5d9a9d276fbf0fb7aefd8e8f26d85b113a5702",
"body": "* Enhance audio file support by adding M4A/AAC decoding using ffmpeg. Update read function to utilize ffmpeg for M4A/AAC formats while maintaining miniaudio for other formats. Improve error handling for ffmpeg availability and provide installation instructions.\n\n* Enhance transcription functionality\n[…]\nracking during audio chunk processing.\n- Introduced a verbose parameter to toggle progress display in the Model class.\n\n* fix stream logs\n\n* fix stream segments\n\n* remove text printing\n\n* bump version",
"is_bot": false,
"headline": "Fix STT stream (#382)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:47:57Z",
"body_truncated": true,
"is_coding_agent": false
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{
"oid": "10b42ca4f02bda4c8f1f1446a002d9fc38cdc433",
"body": null,
"is_bot": false,
"headline": "Add Pocket TTS model (#381)",
"author_name": "Lucas Newman",
"author_login": "lucasnewman",
"committed_at": "2026-01-24T07:47:51Z",
"body_truncated": false,
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},
{
"oid": "8a7d41d0235aab79ecd47224c97345225aedb809",
"body": "* replace soundfile with miniaudio + ffmpeg\n\n* bump version\n\n* fix tests",
"is_bot": false,
"headline": "Replace soundfile with miniaudio + ffmpeg (#379)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:47:51Z",
"body_truncated": false,
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{
"oid": "fd2d21d9c8a2f1a0b27c7ee68a565405a9ac4bc1",
"body": null,
"is_bot": false,
"headline": "fix lang codes kokoro (#380)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:47:34Z",
"body_truncated": false,
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{
"oid": "c305532338877513cb7f34868972e231b2fbea20",
"body": null,
"is_bot": false,
"headline": "update lfm readme (#377)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:47:33Z",
"body_truncated": false,
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"oid": "ca1978af20347b70468d671849e3f5f9e84b26f0",
"body": "* add lfm2-audio (tts, llm working)\n\n* Added 'normalized' parameter to the istft function to allow for window squared (COLA) normalization.\n\n* fix conformer and detokenizer\n\n* fix slide mask\n\n* Refactor audio decoding\n\n* Add sample_rate property and refactor weight loading in LFM2AudioModel\n\n* Add s\n[…]\nformat\n\n* Remove unused input projection in AudioHead class for improved performance and clarity.\n\n* Remove test script for allclose comparison\n\n* add tests\n\n* format\n\n* fix readme gen\n\n* bump version",
"is_bot": false,
"headline": "Add lfm2 audio (#370)",
"author_name": "Prince Canuma",
"author_login": null,
"committed_at": "2026-01-24T07:47:33Z",
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{
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{
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{
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{
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{
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"login": "joshwhiton",
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},
{
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{
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"documentation_url": "https://github.com/ossf/scorecard/blob/c395761df6afe1a69e476bc60a013a94bcbc153f/docs/checks.md#binary-artifacts"
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{
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"documentation_url": "https://github.com/ossf/scorecard/blob/c395761df6afe1a69e476bc60a013a94bcbc153f/docs/checks.md#cii-best-practices"
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{
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{
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{
"name": "Dangerous-Workflow",
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"documentation_url": "https://github.com/ossf/scorecard/blob/c395761df6afe1a69e476bc60a013a94bcbc153f/docs/checks.md#dangerous-workflow"
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{
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"documentation_url": "https://github.com/ossf/scorecard/blob/c395761df6afe1a69e476bc60a013a94bcbc153f/docs/checks.md#dependency-update-tool"
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{
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{
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{
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{
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{
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{
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"documentation_url": "https://github.com/ossf/scorecard/blob/c395761df6afe1a69e476bc60a013a94bcbc153f/docs/checks.md#sast"
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{
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"documentation_url": "https://github.com/ossf/scorecard/blob/c395761df6afe1a69e476bc60a013a94bcbc153f/docs/checks.md#security-policy"
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{
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{
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"documentation_url": "https://github.com/ossf/scorecard/blob/c395761df6afe1a69e476bc60a013a94bcbc153f/docs/checks.md#token-permissions"
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{
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{
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{
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{
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{
"key": "tests_present",
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{
"key": "linter_config",
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],
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{
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{
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{
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},
{
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"notes": [],
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"inputs": {
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"warnings": [
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}
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