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.gitattributes

@@ -33,3 +33,4 @@ saved_model/**/* filter=lfs diff=lfs merge=lfs -text
 *.zip filter=lfs diff=lfs merge=lfs -text
 *.zst filter=lfs diff=lfs merge=lfs -text
 *tfevents* filter=lfs diff=lfs merge=lfs -text
+tokenizer.json filter=lfs diff=lfs merge=lfs -text

README.md

@@ -0,0 +1,130 @@
+---
+library_name: mlx-audio-plus
+base_model:
+- FunAudioLLM/Fun-ASR-Nano-2512
+tags:
+- mlx
+- funasr
+- speech-recognition
+- speech-to-text
+- stt
+pipeline_tag: automatic-speech-recognition
+language:
+- en
+- zh
+- ja
+- ko
+- es
+- fr
+- de
+- it
+- pt
+- ru
+- ar
+- th
+- vi
+---
+
+# mlx-community/Fun-ASR-Nano-2512-8bit
+
+This model was converted to MLX format from [FunAudioLLM/Fun-ASR-Nano-2512](https://huggingface.co/FunAudioLLM/Fun-ASR-Nano-2512) using [mlx-audio-plus](https://github.com/DePasqualeOrg/mlx-audio-plus) version **0.1.4**.
+
+## Features
+
+| Feature | Description |
+|---------|-------------|
+| **Multilingual** | Supports 13+ languages: English, Chinese, Japanese, Korean, Spanish, French, German, Italian, Portuguese, Russian, Arabic, Thai, Vietnamese |
+| **Translation** | Translate speech directly to English text |
+| **Custom prompting** | Guide recognition with domain-specific context |
+| **Streaming** | Real-time token-by-token output |
+
+## Installation
+
+```bash
+pip install -U mlx-audio-plus
+```
+
+## Usage
+
+### Basic Transcription
+
+```python
+from mlx_audio.stt.models.funasr import Model
+
+# Load the model
+model = Model.from_pretrained("mlx-community/Fun-ASR-Nano-2512-8bit")
+
+# Transcribe audio
+result = model.generate("audio.wav")
+print(result.text)
+# Output: "The quick brown fox jumps over the lazy dog."
+
+print(f"Duration: {result.duration:.2f}s")
+print(f"Language: {result.language}")
+```
+
+### Translation (Speech to English Text)
+
+```python
+# Translate Chinese/Japanese/etc. audio to English
+result = model.generate(
+    "chinese_speech.wav",
+    task="translate",
+    target_language="en"
+)
+print(result.text)  # English translation
+```
+
+### Custom Prompting
+
+Provide context to improve recognition accuracy for specialized domains:
+
+```python
+# Medical transcription
+result = model.generate(
+    "doctor_notes.wav",
+    initial_prompt="Medical consultation discussing cardiac symptoms and treatment options."
+)
+
+# Technical content
+result = model.generate(
+    "tech_podcast.wav",
+    initial_prompt="Discussion about machine learning, APIs, and software development."
+)
+```
+
+### Streaming Output
+
+Get real-time output as the model generates:
+
+```python
+# Print tokens as they're generated
+result = model.generate("audio.wav", verbose=True)
+# Tokens stream to stdout in real-time
+
+# Or use the streaming generator
+for chunk in model.generate("audio.wav", stream=True):
+    print(chunk, end="", flush=True)
+```
+
+### Batch Processing
+
+```python
+audio_files = ["meeting1.wav", "meeting2.wav", "meeting3.wav"]
+
+for audio_path in audio_files:
+    result = model.generate(audio_path)
+    print(f"{audio_path}: {result.text}")
+```
+
+## Supported Languages
+
+| Code | Language | Code | Language |
+|------|----------|------|----------|
+| `en` | English | `ru` | Russian |
+| `zh` | Chinese | `ar` | Arabic |
+| `ja` | Japanese | `th` | Thai |
+| `ko` | Korean | `vi` | Vietnamese |
+| `es` | Spanish | `de` | German |
+| `fr` | French | `it` | Italian |
+| `pt` | Portuguese | `auto` | Auto-detect |

config.json

@@ -0,0 +1,48 @@
+{
+  "model_type": "funasr",
+  "sample_rate": 16000,
+  "n_mels": 80,
+  "lfr_m": 7,
+  "lfr_n": 6,
+  "encoder": {
+    "input_dim": 560,
+    "encoder_dim": 512,
+    "num_heads": 4,
+    "ffn_dim": 2048,
+    "kernel_size": 11,
+    "num_encoders0": 1,
+    "num_encoders": 49,
+    "num_tp_encoders": 20,
+    "dropout": 0.0
+  },
+  "adaptor": {
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+    "llm_dim": 1024,
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+    "n_layer": 2,
+    "attention_heads": 8,
+    "dropout": 0.0
+  },
+  "llm": {
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+    "hidden_size": 1024,
+    "num_hidden_layers": 28,
+    "num_attention_heads": 16,
+    "num_key_value_heads": 8,
+    "intermediate_size": 3072,
+    "max_position_embeddings": 40960,
+    "rope_theta": 1000000.0,
+    "rms_norm_eps": 1e-06,
+    "tie_word_embeddings": false,
+    "head_dim": 128
+  },
+  "quantization": {
+    "bits": 8,
+    "group_size": 64,
+    "quantized_components": [
+      "llm.model.layers",
+      "audio_adaptor"
+    ]
+  }
+}

model.safetensors

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tokenizer.json

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tokenizer_config.json

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