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This is the official repository for the paper *"**MuQ**: Self-Supervised **Mu**sic Representation Learning with Mel Residual Vector **Q**uantization"*. For more detailed information, we strongly recommend referring to https://github.com/tencent-ailab/MuQ and the paper). - **MuQ**(see this link): A large music foundation model pre-trained via Self-Supervised Learning (SSL), achieving SOTA in various MIR tasks. - **MuQ-MuLan**(see this link): A music-text joint embedding model trained via contrastive learning, supporting both English and Chinese texts. To begin with, please use pip to install the official `muq` lib, and ensure that your `python>=3.8`:

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This is the official repository for the paper *"**MuQ**: Self-Supervised **Mu**sic Representation Learning with Mel Residual Vector **Q**uantization"*. For more detailed information, we strongly recommend referring to https://github.com/tencent-ailab/MuQ and the paper). - **MuQ**(see this link): A large music foundation model pre-trained via Self-Supervised Learning (SSL), achieving SOTA in various MIR tasks. - **MuQ-MuLan**(see this link): A music-text joint embedding model trained via contrastive learning, supporting both English and Chinese texts. To begin with, please use pip to install the official `muq` lib, and ensure that your `python>=3.8`:

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This model detects the presence of Southern Resident Killer Whale (SRKW) calls in audio recordings from hydrophone networks. It was trained on labeled audio data from the Orcasound hydrophone network in Puget Sound, Washington. **Architecture**: ResNet50 with custom classification head **Input**: Mel spectrogram (1 channel, 256 mel bins, 312 time frames) **Output**: Binary classification (orca call present / not present) **Framework**: PyTorch (ported from FastAI) **Note**: This is a port of the original FastAI model to pure PyTorch for easier usage and deployment.

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> **Common-Voice-Gender-Detection** is a fine-tuned version of `facebook/wav2vec2-base-960h` for **binary audio classification**, specifically trained to detect speaker gender as **female** or **male**. This model leverages the `Wav2Vec2ForSequenceClassification` architecture for efficient and accurate voice-based gender classification. > [!note] Wav2Vec2: Self-Supervised Learning for Speech Recognition : https://arxiv.org/pdf/2006.11477 * **Speech Analytics** – Assist in analyzing speaker demographics in call centers or customer service recordings. * **Conversational AI Personalization** – Adjust tone or dialogue based on gender detection for more personalized voice assistants. * **Voice Dataset Curation** – Automatically tag or filter voice datasets by speaker gender for better dataset management. * **Research Applications** – Enable linguistic and acoustic research involving gender-specific speech patterns. * **Multimedia Content Tagging** – Automate metadata generation for gender identification in podcasts, interviews, or video content.

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This is an ONNX version of prithivMLmods/Common-Voice-Gender-Detection. It was automatically converted and uploaded using this space. > **Common-Voice-Gender-Detection** is a fine-tuned version of `facebook/wav2vec2-base-960h` for **binary audio classification**, specifically trained to detect speaker gender as **female** or **male**. This model leverages the `Wav2Vec2ForSequenceClassification` architecture for efficient and accurate voice-based gender classification. > [!note] Wav2Vec2: Self-Supervised Learning for Speech Recognition : https://arxiv.org/pdf/2006.11477

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Qwen3-TTS-Tokenizer-12Hz

阿里巴巴 · Qwen

This repository contains the **Qwen3-TTS-Tokenizer-12Hz**, as presented in the paper Qwen3-TTS Technical Report. Qwen3-TTS-Tokenizer-12Hz achieves extreme bitrate reduction and ultra-low-latency streaming, enabling immediate first-packet emission through its 12.5 Hz, 16-layer multi-codebook design and a lightweight causal ConvNet. * **Paper:** Qwen3-TTS Technical Report * **GitHub Repository:** QwenLM/Qwen3-TTS * **Demo:** Hugging Face Space

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A fine-tuned **AutoencoderOobleck** decoder with an intent to improve audio fidelity for the ACE-Step 1.5 music generation pipeline. Drop-in compatible with all existing ACE-Step DiT checkpoints. ACE-Step 1.5 uses a VAE (Variational Autoencoder) to convert between audio waveforms and the latent space that the DiT diffusion model operates in. The original VAE decoder attenuates high-frequency content, resulting in audio with reduced clarity and detail above 6kHz. ScragVAE retrains the decoder half of the VAE to better reconstruct upper harmonics, transient detail, and spectral "air" — while keeping the encoder frozen so all existing DiT models remain fully compatible.

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sidon-coreml

seanll95

基础档案已收录,详细介绍后续补充。可先查看厂商、分类与来源。

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DF_Arena_1B_V_1

Speech-Arena-2025

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DF_Arena_500M_V_1

Speech-Arena-2025

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This repository provides all the necessary tools to perform emotion recognition with a fine-tuned wav2vec2 (base) model using SpeechBrain. It is trained on IEMOCAP training data. For a better experience, we encourage you to learn more about SpeechBrain. The model performance on IEMOCAP test set is: This system is composed of an wav2vec2 model. It is a combination of convolutional and residual blocks. The embeddings are extracted using attentive statistical pooling. The system is trained with Additive Margin Softmax Loss. Speaker Verification is performed using cosine distance between speaker embeddings.

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This is a spoken language recognition model trained on the VoxLingua107 dataset using SpeechBrain. The model uses the ECAPA-TDNN architecture that has previously been used for speaker recognition. However, it uses more fully connected hidden layers after the embedding layer, and cross-entropy loss was used for training. We observed that this improved the performance of extracted utterance embeddings for downstream tasks. The system is trained with recordings sampled at 16kHz (single channel). The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling *classify_file* if needed. The model can classify a speech utterance according to the language spoken. It covers 107 different languages ( Abkhazian, Afrikaans, Amharic, Arabic, Assamese, Azerbaijani, Bashkir, Belarusian, Bulgarian, Bengali, Tibetan, Breton, Bosnian, Catalan, Cebuano, Czech, Welsh, Danish, German, Greek, English, Esperanto, Spanish, Estonian, Basque, Persian, Finnish, Faroese, French, Galician, Guarani, Gujarati, Manx, Hausa, Hawaiian, Hindi, Croatian, Haitian, Hungarian, Armenian, Interlingua, Indonesian, Icelandic, Italian, Hebrew, Japanese, Javanese, Georgian, Kazakh, Central Khmer, Kannada, Korean, Latin, Luxembourgish, Lingala, Lao, Lithuanian, Latvian, Malagasy, Maori, Macedonian, Malayalam, Mongolian, Marathi, Malay, Maltese, Burmese, Nepali, Dutch, Norwegian Nynorsk, Norwegian, Occitan, Panjabi, Polish, Pushto, Portuguese, Romanian, Russian, Sanskrit, Scots, Sindhi, Sinhala, Slovak, Slovenian, Shona, Somali, Albanian, Serbian, Sundanese, Swedish, Swahili, Tamil, Telugu, Tajik, Thai, Turkmen, Tagalog, Turkish, Tatar, Ukrainian, Urdu, Uzbek, Vietnamese, Waray, Yiddish, Yoruba, Mandarin Chinese).

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This repository provides all the necessary tools to perform enhancement with SpeechBrain. For a better experience we encourage you to learn more about SpeechBrain. The model performance is: The system is trained with recordings sampled at 16kHz (single channel). The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling *enhance_file* if needed. Make sure your input tensor is compliant with the expected sampling rate if you use *enhance_batch* as in the example. To perform inference on the GPU, add `run_opts={"device":"cuda"}` when calling the `from_hparams` method.

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