* studio: extract param count from model name as fallback When HuggingFace API doesn't return totalParams for a model, extract the param count from the model name (e.g. "Qwen3-0.6B" -> "0.6B", "Llama-3.2-1B-Instruct" -> "1B"). Applied to both the recommended list and HF search results. * studio: read GGUF context_length via fast header parser, set max tokens - Fast GGUF metadata reader (~30-55ms) parses only KV header, skips tensor data and large arrays (tokenizer vocab etc) - Extracts context_length and chat_template from GGUF metadata - Returns context_length in LoadResponse for frontend to use - Frontend sets maxTokens to actual context_length for GGUFs (e.g. 262144 for Qwen3.5-9B, 131072 for Qwen2.5-7B) - Max Tokens slider shows "Max" and is locked for GGUFs - Auto-load path also uses actual context_length from load response - Toast auto-dismiss (5s) and close button for auto-load toast * studio: GGUF TTS audio support (from PR #4318) Add GGUF TTS audio generation via llama-server. When a GGUF model loads, the backend probes its vocabulary to detect audio codecs (SNAC/BiCodec/DAC/CSM/Whisper). If detected, the codec is pre-loaded and the model is reported as audio to the frontend. During chat, TTS models route to the audio generation path which sends a per-codec prompt to llama-server's /completion endpoint, extracts generated tokens/text, and decodes to WAV using AudioCodecManager. Also strips base64 audio data from prior assistant messages to prevent context overflow. Co-authored-by: Manan Shah <mananshah511@gmail.com> * [pre-commit.ci] auto fixes from pre-commit.com hooks for more information, see https://pre-commit.ci * Remove package-lock.json from tracking * studio: per-model inference defaults, GGUF max tokens fix, reasoning toggle - Add inference_defaults.json with per-model-family sampling parameters for ~50 families (Qwen3.5, Qwen3, Gemma-3, Llama-3, DeepSeek, etc.). Values sourced from unslothai/docs and Ollama params blobs. - Family-based lookup in inference_config.py: extracts model family from identifier, matches against patterns (longest match first), merges with priority: model-specific YAML > family JSON > default.yaml. - Fix GGUF Max Tokens slider locked at "Max": store ggufContextLength separately from maxTokens so the slider is adjustable (step=64). - Fix Ministral YAML: top_p was literal string "default", now 0.95. - Add reasoning toggle for thinking models (Qwen3.5, Qwen3, DeepSeek-R1, DeepSeek-V3.1, etc.): detect enable_thinking support from GGUF chat template metadata, pass --jinja to llama-server, send chat_template_kwargs per-request. Frontend shows "Reasoning is ON/OFF" pill button next to attachment button in composer. * [pre-commit.ci] auto fixes from pre-commit.com hooks for more information, see https://pre-commit.ci * studio: remove default system prompt injection Backend was injecting "You are a helpful AI assistant." when no system prompt was provided. Neither unslothai/docs nor Ollama specify a default system prompt for most models. Now defaults to empty string, letting the model's own chat template handle system behavior. * [pre-commit.ci] auto fixes from pre-commit.com hooks for more information, see https://pre-commit.ci * studio: use lightbulb icons and "Think" label for reasoning toggle Lightbulb on when thinking enabled, lightbulb-off when disabled. Label is just "Think" in both states; grayed out styling when off. * studio: fix HTML file upload breaking chat Replace SimpleTextAttachmentAdapter with custom TextAttachmentAdapter (excludes text/html) and HtmlAttachmentAdapter that strips tags via DOMParser, removing scripts/styles and extracting readable text content instead of dumping raw HTML markup into the conversation. * studio: show chat template in Configuration panel Display the model's Jinja2 chat template in a new "Chat Template" section under Settings (now open by default). For GGUFs, reads from GGUF metadata; for safetensors, reads from tokenizer.chat_template. Template is editable with a "Restore default chat template" button that appears when modified. Section only shows when a model with a chat template is loaded. * studio: editable chat template with Apply & Reload Chat template section now functional: - Editing the template shows "Apply & Reload" (reloads model with custom template) and "Revert changes" buttons - For GGUFs: writes template to temp .jinja file, passes --chat-template-file to llama-server on reload - For non-GGUF: passes chat_template_override in load request - Settings section now open by default - selectModel supports forceReload to reload same model * [pre-commit.ci] auto fixes from pre-commit.com hooks for more information, see https://pre-commit.ci * studio: fix DeepSeek reasoning detection and auto-load metadata - Set _model_identifier before _read_gguf_metadata so DeepSeek "thinking" template detection works (was always None before) - Populate ggufContextLength, supportsReasoning, reasoningEnabled, defaultChatTemplate in autoLoadSmallestModel GGUF path * studio: add spacing before BETA badge in navbar Add gap-1.5 on the logo Link container to space the BETA label from the wordmark. Co-authored-by: Imagineer99 <Imagineer99@users.noreply.github.com> * studio: vertically center BETA badge with logo --------- Co-authored-by: Manan Shah <mananshah511@gmail.com> Co-authored-by: pre-commit-ci[bot] <66853113+pre-commit-ci[bot]@users.noreply.github.com> Co-authored-by: Imagineer99 <Imagineer99@users.noreply.github.com>
341 lines
13 KiB
Python
341 lines
13 KiB
Python
# SPDX-License-Identifier: AGPL-3.0-only
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# Copyright 2026-present the Unsloth AI Inc. team. All rights reserved. See /studio/LICENSE.AGPL-3.0
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"""
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Audio codec loading and decoding for TTS inference.
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Supports: SNAC (Orpheus), CSM (Sesame), BiCodec (Spark), DAC (OuteTTS)
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"""
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import io
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import re
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import wave
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import structlog
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from loggers import get_logger
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from typing import Optional, Tuple
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import numpy as np
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import torch
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logger = get_logger(__name__)
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def _numpy_to_wav_bytes(waveform: np.ndarray, sample_rate: int) -> bytes:
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"""Convert a float32 numpy waveform to WAV bytes (16-bit PCM)."""
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waveform = waveform.flatten()
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peak = max(abs(waveform.max()), abs(waveform.min()))
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if peak > 1.0:
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waveform = waveform / peak
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pcm = (waveform * 32767).astype(np.int16)
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buf = io.BytesIO()
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with wave.open(buf, "wb") as wf:
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wf.setnchannels(1)
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wf.setsampwidth(2)
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wf.setframerate(sample_rate)
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wf.writeframes(pcm.tobytes())
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return buf.getvalue()
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class AudioCodecManager:
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"""Manages loading and caching of audio codec models for TTS decoding."""
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def __init__(self):
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self._snac_model = None
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self._bicodec_tokenizer = None
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self._bicodec_repo_path = None
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self._dac_audio_codec = None
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def load_codec(
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self,
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audio_type: str,
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device: str = "cuda",
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model_repo_path: Optional[str] = None,
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) -> None:
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"""Load the appropriate codec for the given audio type."""
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if audio_type == "snac":
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self._load_snac(device)
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elif audio_type == "bicodec":
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self._load_bicodec(device, model_repo_path)
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elif audio_type == "dac":
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self._load_dac(device)
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elif audio_type == "csm":
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pass # CSM decoding is built into the model (output_audio=True)
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else:
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raise ValueError(f"Unknown audio_type: {audio_type}")
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# ── Lazy loaders ─────────────────────────────────────────────
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def _load_snac(self, device: str) -> None:
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if self._snac_model is not None:
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return
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from snac import SNAC
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self._snac_model = (
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SNAC.from_pretrained("hubertsiuzdak/snac_24khz").to(device).eval()
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)
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logger.info("Loaded SNAC codec (24kHz)")
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def _load_bicodec(self, device: str, model_repo_path: Optional[str] = None) -> None:
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if self._bicodec_tokenizer is not None:
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return
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import os
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import sys
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import subprocess
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# Clone SparkAudio/Spark-TTS GitHub repo for the sparktts Python package
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# (same approach as training — the HF model repos don't contain the package)
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spark_code_dir = os.path.join(
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os.path.dirname(model_repo_path or "."), "Spark-TTS"
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)
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sparktts_pkg = os.path.join(spark_code_dir, "sparktts")
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if not os.path.isdir(sparktts_pkg):
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logger.info(f"Cloning SparkAudio/Spark-TTS to {spark_code_dir}...")
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subprocess.run(
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[
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"git",
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"clone",
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"--depth",
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"1",
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"https://github.com/SparkAudio/Spark-TTS",
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spark_code_dir,
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],
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check = True,
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)
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if spark_code_dir not in sys.path:
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sys.path.insert(0, spark_code_dir)
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from sparktts.models.audio_tokenizer import BiCodecTokenizer
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# BiCodecTokenizer needs the MODEL repo path (contains BiCodec/ weights)
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tokenizer_path = model_repo_path or spark_code_dir
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self._bicodec_repo_path = tokenizer_path
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self._bicodec_tokenizer = BiCodecTokenizer(tokenizer_path, device)
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logger.info(f"Loaded BiCodec tokenizer from {tokenizer_path}")
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def _load_dac(self, device: str) -> None:
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if self._dac_audio_codec is not None:
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return
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import os
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import sys
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import subprocess
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# Clone OuteTTS repo (same pattern as Spark-TTS / BiCodec)
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# The pip package has problematic dependencies; the notebook clones and
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# removes gguf_model.py, interface.py, __init__.py before importing.
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base_dir = os.path.dirname(os.path.abspath(__file__))
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outetts_code_dir = os.path.join(base_dir, "OuteTTS")
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outetts_pkg = os.path.join(outetts_code_dir, "outetts")
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if not os.path.isdir(outetts_pkg):
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logger.info(f"Cloning edwko/OuteTTS to {outetts_code_dir}...")
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subprocess.run(
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[
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"git",
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"clone",
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"--depth",
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"1",
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"https://github.com/edwko/OuteTTS",
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outetts_code_dir,
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],
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check = True,
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)
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# Remove files that pull in heavy / incompatible dependencies
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# (matches notebook: gguf_model.py is under models/, others under outetts/)
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remove_paths = [
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os.path.join(outetts_pkg, "models", "gguf_model.py"),
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os.path.join(outetts_pkg, "interface.py"),
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os.path.join(outetts_pkg, "__init__.py"),
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]
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for fpath in remove_paths:
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if os.path.exists(fpath):
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os.remove(fpath)
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logger.info(f"Removed {fpath}")
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if outetts_code_dir not in sys.path:
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sys.path.insert(0, outetts_code_dir)
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from outetts.version.v3.audio_processor import AudioProcessor
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from outetts.models.config import ModelConfig as OuteTTSModelConfig
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dummy_config = OuteTTSModelConfig(
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tokenizer_path = "OuteAI/Llama-OuteTTS-1.0-1B",
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device = device,
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audio_codec_path = None,
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)
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processor = AudioProcessor(config = dummy_config)
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self._dac_audio_codec = processor.audio_codec
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logger.info("Loaded DAC audio codec")
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# ── Decoders ─────────────────────────────────────────────────
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def decode_snac(
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self, generated_ids: torch.Tensor, device: str
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) -> Tuple[bytes, int]:
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"""
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Decode SNAC tokens (Orpheus) into WAV bytes.
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generated_ids: full model output including prompt tokens.
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Looks for START_OF_SPEECH (128257) marker, extracts codes after it,
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strips EOS (128258), redistributes 7-per-frame codes into 3 SNAC layers.
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Returns (wav_bytes, 24000).
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"""
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# Find START_OF_SPEECH token (128257)
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token_indices = (generated_ids == 128257).nonzero(as_tuple = True)
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if len(token_indices[1]) > 0:
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cropped = generated_ids[:, token_indices[1][-1] + 1 :]
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else:
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# Gracefully fall back to using entire output if marker not found
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logger.warning(
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"No START_OF_SPEECH token (128257) found — using full generated output"
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)
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cropped = generated_ids
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row = cropped[0]
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# Remove EOS tokens (128258)
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row = row[row != 128258]
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# Trim to multiple of 7
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row = row[: (len(row) // 7) * 7]
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if len(row) == 0:
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raise ValueError("No valid audio codes found after START_OF_SPEECH token")
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codes = [t.item() - 128266 for t in row]
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# Redistribute into 3 SNAC layers (7 codes per frame → 1+2+4)
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layer_1, layer_2, layer_3 = [], [], []
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for i in range(len(codes) // 7):
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layer_1.append(codes[7 * i])
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layer_2.append(codes[7 * i + 1] - 4096)
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layer_3.append(codes[7 * i + 2] - 8192)
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layer_3.append(codes[7 * i + 3] - 12288)
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layer_2.append(codes[7 * i + 4] - 16384)
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layer_3.append(codes[7 * i + 5] - 20480)
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layer_3.append(codes[7 * i + 6] - 24576)
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snac_codes = [
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torch.tensor(layer).unsqueeze(0).to(device)
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for layer in [layer_1, layer_2, layer_3]
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]
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with torch.no_grad():
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audio = self._snac_model.decode(snac_codes)
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waveform = audio.squeeze().cpu().numpy()
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return _numpy_to_wav_bytes(waveform, 24000), 24000
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def decode_csm(self, audio_values: torch.Tensor) -> Tuple[bytes, int]:
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"""
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Decode CSM output (already a waveform from model.generate(output_audio=True)).
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Returns (wav_bytes, 24000).
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"""
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waveform = audio_values[0].to(torch.float32).cpu().numpy()
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return _numpy_to_wav_bytes(waveform, 24000), 24000
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def decode_bicodec(self, generated_text: str, device: str) -> Tuple[bytes, int]:
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"""
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Decode BiCodec tokens (Spark-TTS) from generated text.
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Extracts bicodec_semantic_N and bicodec_global_N tokens via regex.
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Returns (wav_bytes, sample_rate).
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"""
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semantic_matches = re.findall(r"<\|bicodec_semantic_(\d+)\|>", generated_text)
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global_matches = re.findall(r"<\|bicodec_global_(\d+)\|>", generated_text)
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logger.info(
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f"BiCodec decode: {len(global_matches)} global tokens, {len(semantic_matches)} semantic tokens"
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)
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if len(global_matches) < 10:
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logger.info(
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f"BiCodec generated text (first 500 chars): {generated_text[:500]}"
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)
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if not semantic_matches:
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raise ValueError("No bicodec_semantic tokens found in generated output")
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semantic_ids = (
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torch.tensor([int(t) for t in semantic_matches]).long().unsqueeze(0)
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)
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# Speaker encoder expects exactly 32 global tokens (token_num=32 in BiCodec config).
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# Pad with zeros or truncate to 32.
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GLOBAL_TOKEN_NUM = 32
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if global_matches:
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raw = [int(t) for t in global_matches]
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else:
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raw = []
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if len(raw) < GLOBAL_TOKEN_NUM:
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raw = raw + [0] * (GLOBAL_TOKEN_NUM - len(raw))
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raw = raw[:GLOBAL_TOKEN_NUM]
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global_ids = torch.tensor(raw).long().unsqueeze(0) # (1, 32)
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self._bicodec_tokenizer.device = device
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self._bicodec_tokenizer.model.to(device)
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wav_np = self._bicodec_tokenizer.detokenize(
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global_ids.to(device),
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semantic_ids.to(device),
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)
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sr = self._bicodec_tokenizer.config.get("sample_rate", 16000)
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return _numpy_to_wav_bytes(wav_np, sr), sr
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def decode_dac(self, generated_text: str, device: str) -> Tuple[bytes, int]:
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"""
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Decode DAC tokens (OuteTTS) from generated text.
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Extracts c1_N and c2_N codec code tokens via regex.
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Returns (wav_bytes, 24000).
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"""
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c1 = list(map(int, re.findall(r"<\|c1_(\d+)\|>", generated_text)))
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c2 = list(map(int, re.findall(r"<\|c2_(\d+)\|>", generated_text)))
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if not c1 or not c2:
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raise ValueError("No DAC code tokens (c1/c2) found in generated output")
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t = min(len(c1), len(c2))
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c1 = c1[:t]
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c2 = c2[:t]
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codes = torch.tensor([[c1, c2]], dtype = torch.int64).to(device)
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with torch.no_grad():
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audio = self._dac_audio_codec.decode(codes)
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waveform = audio.squeeze().cpu().numpy()
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return _numpy_to_wav_bytes(waveform, 24000), 24000
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def decode(
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self,
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audio_type: str,
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device: str,
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token_ids: Optional[list] = None,
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text: Optional[str] = None,
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) -> Tuple[bytes, int]:
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"""Unified decode — dispatches to the right codec decoder."""
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if audio_type == "snac":
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if not token_ids:
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raise ValueError("SNAC decoding requires token_ids")
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return self.decode_snac(torch.tensor([token_ids], dtype = torch.long), device)
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elif audio_type == "bicodec":
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if not text:
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raise ValueError("BiCodec decoding requires text")
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return self.decode_bicodec(text, device)
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elif audio_type == "dac":
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if not text:
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raise ValueError("DAC decoding requires text")
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return self.decode_dac(text, device)
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raise ValueError(f"Cannot decode audio_type: {audio_type}")
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# ── Cleanup ──────────────────────────────────────────────────
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def unload(self) -> None:
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"""Release all codec models from memory."""
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if self._snac_model is not None:
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del self._snac_model
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self._snac_model = None
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if self._bicodec_tokenizer is not None:
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del self._bicodec_tokenizer
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self._bicodec_tokenizer = None
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self._bicodec_repo_path = None
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if self._dac_audio_codec is not None:
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del self._dac_audio_codec
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self._dac_audio_codec = None
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logger.info("Unloaded all audio codecs")
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