Alexandria Audiobook Generator

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A note to new users: Alexandria has recently seen a sudden surge of attention and new users. As a small project, I may not be able to respond to every issue promptly. Before opening an issue, please read this README and the Wiki thoroughly — most common questions are already answered there. Thank you for your patience!

Transform any book or novel into a fully-voiced audiobook using AI-powered script annotation and text-to-speech. Features a built-in Qwen3-TTS engine with batch processing and a browser-based editor for fine-tuning every line before final export.

Example: sample.mp3

Screenshots

Features

AI-Powered Pipeline

• Local & Cloud LLM Support - Use any OpenAI-compatible API (LM Studio, Ollama, OpenAI, etc.)
• Automatic Script Annotation - LLM parses text into JSON with speakers, dialogue, and TTS instruct directions
• LLM Script Review - Optional second LLM pass that fixes common annotation errors: strips attribution tags from dialogue, splits misattributed narration/dialogue, merges over-split narrator entries, and validates instruct fields
• Persona Generation - LLM analyzes the script to create voice descriptions for each character, then generates reference audio via VoiceDesign and assigns clone voices automatically — one click to go from script to fully-voiced cast
• Speaker Aliases - Map multiple speaker names to the same voice (e.g. "YOUNG ELENA" → "ELENA") so variants share a single voice configuration
• Smart Chunking - Groups consecutive lines by speaker (up to 500 chars) for natural flow
• Context Preservation - Passes character roster and last 3 script entries between chunks for name and style continuity

Voice Generation

• Built-in TTS Engine - Qwen3-TTS runs locally with no external server required
• External Server Mode - Optionally connect to a remote Qwen3-TTS Gradio server
• Multi-Language Support - English, Chinese, French, German, Italian, Japanese, Korean, Portuguese, Russian, Spanish, or Auto-detect
• Custom Voices - 9 pre-trained voices with instruct-based emotion/tone control
• Voice Cloning - Clone any voice from a 5-15 second reference audio sample
• Voice Designer - Create new voices from text descriptions (e.g. "A warm, deep male voice with a calm and steady tone")
• LoRA Voice Training - Fine-tune the Base model on custom voice datasets to create persistent voice identities with instruct-following
• Built-in LoRA Presets - Pre-trained voice adapters included out of the box, ready to assign to characters
• Dataset Builder - Interactive tool for creating LoRA training datasets with per-sample text, emotion, and audio preview
• Batch Processing - Generate dozens of chunks simultaneously with 3-6x real-time throughput
• Codec Compilation - Optional torch.compile optimization for 3-4x faster batch decoding
• Non-verbal Sounds - LLM writes natural vocalizations ("Ahh!", "Mmm...", "Haha!") with context-aware instruct directions
• Natural Pauses - Configurable silence between speakers (default 500ms) and same-speaker segments (default 250ms)

Web UI Editor

• Streamlined Interface - 5-step core pipeline (Setup, Script, Voices, Editor, Result) plus advanced tools (Designer, Dataset, Training)
• Chunk Editor - Edit speaker, text, and instruct for any line
• Selective Regeneration - Re-render individual chunks without regenerating everything
• Batch Processing - Optimized batch rendering with sub-batching for efficient GPU utilization
• Live Progress - Real-time logs and status tracking for all operations
• Audio Preview - Play individual chunks or sequence through the entire audiobook
• Script Library - Save and load annotated scripts with voice configurations

Export Options

• Combined Audiobook - Single MP3 with all voices and natural pauses
• Individual Voicelines - Separate MP3 per line for DAW editing (Audacity, etc.)
• Audacity Export - One-click zip with per-speaker WAV tracks, LOF project file, and labels for automatic multi-track import into Audacity
• M4B Audiobook - Chaptered M4B (AAC) with per-chunk or auto-detected chapter markers for audiobook players (Audiobookshelf, Apple Books, VLC, etc.)

Requirements

• Pinokio
• LLM server (one of the following):
  • LM Studio (local) - recommended: Qwen3 or similar
  • Ollama (local)
  • OpenAI API (cloud)
  • Any OpenAI-compatible API
• GPU: 8 GB VRAM minimum, 16 GB+ recommended — see compatibility table below
  • Each TTS model uses ~3.4 GB; remaining VRAM determines batch size
  • CPU mode available on all platforms but significantly slower
• RAM: 16 GB recommended (8 GB minimum)
• Disk: ~20 GB (8 GB venv/PyTorch, ~7 GB for model weights, working space for audio)

GPU Compatibility

Note: No external TTS server is required. Alexandria includes a built-in Qwen3-TTS engine that loads models directly. Model weights are downloaded automatically on first use (~3.5 GB per model variant).

Documentation: For in-depth guidance on voice types, LoRA training, batch generation, and more, see the Wiki.

Installation

Option A: Pinokio (Recommended)

1. Install Pinokio if you haven't already
2. Open Alexandria on Pinokio: Install via Pinokio
   • Or manually: in Pinokio, click Download and paste https://github.com/Finrandojin/alexandria-audiobook
3. Click Install to set up dependencies
4. Click Start to launch the web interface

Option B: Google Colab (No Install Required)

No GPU or wrong OS? Run Alexandria on a free T4 GPU in your browser:

Requires a free ngrok account for the web UI tunnel. See the notebook for full instructions.

Option C: Docker (NVIDIA GPU)

For integration into automated pipelines or server deployments:

git clone https://github.com/Finrandojin/alexandria-audiobook.git
cd alexandria-audiobook
docker compose up --build

Requires Docker with the NVIDIA Container Toolkit. The web UI is available at http://localhost:4200. TTS models download on first use and are cached in a Docker volume. User data (uploads, voice configs, trained LoRA adapters, audio output) persists via bind mounts to the project directory.

First Launch — What to Expect

If this is your first time running Alexandria, read this before anything else.

1. You Need an LLM Server Running First

Alexandria does not include an LLM — it connects to one over an API. Before generating a script, you must have one of these running:

If the LLM server isn't running when you click "Generate Script", the generation will fail. Check the Pinokio terminal for error details.

2. First TTS Generation Downloads ~3.5 GB

The TTS models are not included in the install. They download automatically from Hugging Face the first time you generate audio. This is normal:

• Each model variant is ~3.5 GB (CustomVoice, Base/Clone, VoiceDesign)
• Only the variant you use gets downloaded (most users start with CustomVoice)
• Downloads happen in the background — check the Pinokio terminal for progress
• The web UI may appear frozen during this time. It is not — it's waiting for the download to finish
• After the first download, models are cached locally and load in seconds

Tip: If the download seems stuck, check your internet connection. If it fails, restart the app and try again — it will resume from where it left off.

3. First Batch Has Extra Warmup Time

The very first batch generation in a session takes longer than subsequent ones:

• MIOpen autotuning (AMD GPUs): The GPU kernel optimizer runs once per session, adding 30-60 seconds
• Codec compilation (if enabled): One-time ~30-60 second warmup, then 3-4x faster for all remaining batches
• This is expected. After the first batch, generation speed stabilizes

4. VRAM Determines What You Can Do

• If you run out of VRAM, reduce Parallel Workers or Max Chars/Batch in the Setup tab
• Close other GPU applications (games, other AI tools) before generating
• Switching between voice types (Custom → Clone → LoRA) unloads and reloads models, which temporarily frees VRAM

5. Where to Look When Something Goes Wrong

The web UI shows high-level status, but detailed logs are in the Pinokio terminal:

• Click Terminal in the Pinokio sidebar to see real-time output
• Model loading, download progress, VRAM estimates, and errors all appear here
• If generation fails silently in the UI, the terminal will show why

For common issues and solutions, see Troubleshooting.

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Quick Start

The interface is split into a 5-step core pipeline (green tabs, numbered) and advanced tools (blue tabs, unnumbered). You only need the core pipeline to produce an audiobook.

Core Pipeline

Step 1 — Setup Configure your LLM connection and TTS engine. At minimum you need:

• LLM Base URL: http://localhost:1234/v1 (LM Studio) or http://localhost:11434/v1 (Ollama)
• LLM API Key: Your API key (use local for local servers)
• LLM Model Name: The model to use (e.g., qwen2.5-14b)
• TTS Mode: local (built-in, recommended) — loads models directly, no external server needed
• Click Save Configuration when done

Step 2 — Script

• Select your book file (.txt, .md, or .epub) using the file picker — it uploads automatically
• Click Generate Annotated Script — this sends the book to your LLM to split it into annotated chunks with speaker labels and voice directions
• (Optional) Click Review Script if the generated script has issues — this runs a second LLM pass to fix speaker misattributions or formatting problems
• You can save the script for later use with the Save feature below

Step 3 — Voices Each character detected in the script gets a voice card. For each speaker:

• Choose a voice type: Custom Voice (easiest), Clone Voice, LoRA Voice, or Voice Design
• For Custom Voice, pick from 9 presets (Ryan, Serena, Aiden, etc.) and optionally set a character style (e.g., "Heavy Scottish accent")
• Generate Personas — Click to have the LLM analyze the script, create voice descriptions for each character, generate reference audio, and assign clone voices automatically. Toggle "Advanced" for batch size control. This is the fastest way to assign unique voices to all characters
• Speaker Aliases — Use the "Alias of" dropdown on any voice card to map a speaker to another character's voice (e.g., set "YOUNG ELENA" as alias of "ELENA"). Aliased speakers use the target's voice config during generation
• Changes save automatically — see Voice Types for guidance on each type

Step 4 — Editor

• Click Render Pending to generate audio for all chunks in batch
• Listen to individual chunks or click Play Sequence to preview in order
• Edit any chunk's text, speaker, or instruct inline and regenerate it individually
• When satisfied, click Merge All to combine everything into the final audiobook

Step 5 — Result

• Listen to the finished audiobook in the browser
• Download as MP3, or click Export to Audacity for per-speaker WAV tracks

Advanced Tools (Optional)

These tabs are for power users who want more control over voice creation:

• Designer — Create new voices from text descriptions (e.g., "A warm elderly woman with a gentle raspy voice"). Save them to use as clone references in the Voices tab
• Dataset — Build LoRA training datasets interactively, one sample at a time with audio preview
• Training — Train LoRA adapters on voice datasets to create persistent voice identities that follow instruct directions

Web Interface

Setup Tab

Configure connections to your LLM and TTS engine.

TTS Settings:

• Mode - local (built-in engine) or external (connect to Gradio server)
• Device - auto (recommended), cuda, cpu, or mps
• Language - TTS synthesis language: English (default), Chinese, French, German, Italian, Japanese, Korean, Portuguese, Russian, Spanish, or Auto (let the model detect)
• Parallel Workers - Batch size for fast batch rendering (higher = more VRAM usage)
• Batch Seed - Fixed seed for reproducible batch output (leave empty for random)
• Compile Codec - Enable torch.compile for 3-4x faster batch decoding (adds ~30-60s warmup on first generation)
• Sub-batching - Split batches by text length to reduce wasted GPU compute on padding (enabled by default)
• Min Sub-batch Size - Minimum chunks per sub-batch before allowing a split (default: 4)
• Length Ratio - Maximum longest/shortest text length ratio before forcing a sub-batch split (default: 5)
• Speaker Change Pause - Silence in milliseconds between different speakers during merge (default: 500)
• Same Speaker Pause - Silence in milliseconds when the same speaker continues during merge (default: 250)

Prompt Settings (Advanced):

• Generation Settings - Chunk size and max tokens for LLM responses
• LLM Sampling Parameters - Temperature, Top P, Top K, Min P, and Presence Penalty
• Banned Tokens - Comma-separated list of tokens to ban from LLM output (useful for disabling thinking mode on models like GLM4, DeepSeek-R1, etc.)
• Prompt Customization - System and user prompts used for script generation. Defaults are loaded from default_prompts.txt and can be customized per-session in the UI. Click "Reset to Defaults" to reload the file-based defaults (picks up edits without restarting the app)

Script Tab

Upload a text file (.txt, .md, or .epub) and generate the annotated script. EPUB files are automatically converted to plain text on upload. The LLM converts your book into a structured JSON format with:

• Speaker identification (NARRATOR vs character names)
• Dialogue text with natural vocalizations (written as pronounceable text, not tags)
• Style directions for TTS delivery

Review Script - After generation, click "Review Script" to run a second LLM pass that detects and fixes common annotation errors:

1. Attribution tags left in dialogue ("said he", "she replied") are stripped
2. Narration mixed into character entries is split out as NARRATOR
3. Dialogue embedded in narrator entries is extracted as the correct speaker
4. Short consecutive narrator entries covering the same scene are merged
5. Invalid instruct fields (physical actions instead of voice directions) are corrected

Review prompts are customizable in review_prompts.txt (same format as default_prompts.txt).

Voices Tab

After script generation, voices are automatically loaded from the annotated script. For each speaker:

Persona Generation: Click Generate Personas to automatically assign voices to all characters. The LLM analyzes dialogue in the script and produces a voice description and sample text for each speaker. These are fed to the VoiceDesign model to generate reference audio, which is saved and assigned as a clone voice. The result is a fully-voiced cast with no manual configuration.

• Standard mode — Processes all speakers at once
• Advanced mode — Toggle "Advanced" to control batch size (default 40) for large casts. In advanced mode, speakers are processed in discovery batches, with the LLM receiving dialogue samples to infer voice characteristics

Speaker Aliases: Each voice card has an "Alias of" dropdown. Setting a speaker as an alias of another speaker means it will use the target's voice configuration during audio generation. Useful for:

• Character name variants (e.g., "DR. SMITH" → "SMITH")
• Age variants (e.g., "YOUNG ELENA" → "ELENA")
• Reducing the number of voices to configure

Aliases resolve transitively (A → B → C uses C's config) with cycle detection.

Custom Voice Mode:

• Select from 9 pre-trained voices: Aiden, Dylan, Eric, Ono_anna, Ryan, Serena, Sohee, Uncle_fu, Vivian
• Set a character style that appends persistent traits to every TTS instruct (e.g., "Heavy Scottish accent", "Refined aristocratic tone")
• Optionally set a seed for reproducible output

Clone Voice Mode:

• Select a designed voice or enter a custom reference audio path
• Provide the exact transcript of the reference
• Note: Instruct directions are ignored for cloned voices

LoRA Voice Mode:

• Select a trained LoRA adapter from the Training tab
• Set a character style (same as Custom — appended to every instruct)
• Combines voice identity from training with instruct-following from the Base model

Voice Design Mode:

• Set a base voice description (e.g., "Young strong soldier")
• Each line's instruct is appended as delivery/emotion direction
• Generates voice on-the-fly using the VoiceDesign model — ideal for minor characters

Voice Designer Tab

Create new voices from text descriptions without needing reference audio.

• Describe a voice in natural language (e.g., "A warm elderly woman with a gentle, raspy voice and a slight Southern drawl")
• Preview the voice with sample text before saving
• Save to library for use as clone voice references in the Voices tab
• Uses the Qwen3-TTS VoiceDesign model to synthesize voice characteristics from descriptions

Training Tab

Train LoRA adapters on the Base model to create custom voice identities. Several built-in LoRA presets are included out of the box and appear alongside your trained adapters.

Dataset:

• Upload ZIP — WAV files (24kHz mono) + metadata.jsonl with audio_filepath and text fields
• Generate Dataset — Auto-generate training samples from a Voice Designer description with custom sample texts
• Dataset Builder — Interactive tool in its own tab (see below) for building datasets sample-by-sample with preview

Training Configuration:

• Adapter Name — Identifier for the trained model
• Epochs — Full passes over the dataset (15-30 recommended for 20+ samples)
• Learning Rate — Default 5e-6 (conservative). Higher trains faster but risks instability
• LoRA Rank — Adapter capacity. High (64+) locks voice identity strongly but can flatten delivery. Low (8-16) preserves expressiveness
• LoRA Alpha — Scaling factor. Effective strength = alpha / rank. Common starting point: alpha = 2x rank
• Language — Language for the codec prefix token. Match this to your training data's language (English, Chinese, Korean, Japanese, etc.). Mismatched language can cause the adapter to lose speaker identity
• Batch Size / Grad Accum — Batch 1 with gradient accumulation 8 is typical for 24GB cards

Training tips:

• Include samples with varied emotions (happy, sad, angry, calm) for expressive voices
• Neutral-only training data produces flat voices that resist instruct prompting
• The settings info panel in the UI explains each parameter's effect on voice quality

Dataset Builder Tab

Build LoRA training datasets interactively, one sample at a time.

• Create a project with a voice description and optional global seed
• Define samples — Set text and emotion/style per row
• Preview audio — Generate and listen to individual samples or batch-generate all at once
• Cancel batch — Stop a running batch generation without losing completed samples
• Save as dataset — Export the project as a training-ready dataset that appears in the Training tab
• Designed voices and Voice Designer descriptions drive the audio generation via Qwen3-TTS VoiceDesign model

Editor Tab

Fine-tune your audiobook before export:

• View all chunks in a table with status indicators
• Edit inline - Click to modify speaker, text, or instruct
• Generate single - Regenerate just one chunk after editing
• Batch render - Process all pending chunks (see Render Modes below)
• Play sequence - Preview audio playback in order
• Merge all - Combine chunks into final audiobook

Render Modes

Alexandria offers two methods for batch rendering audio:

Render Pending (Standard)

The default rendering mode. Sends individual TTS calls in parallel using the configured worker count.

• Per-speaker seeds - Each voice uses its configured seed for reproducible output
• Voice cloning support - Works with both custom voices and cloned voices

Batch (Fast)

High-speed rendering that sends multiple lines to the TTS engine in a single batched call. Chunks are sorted by text length and processed in optimized sub-batches to minimize padding waste.

• 3-6x real-time throughput - With codec compilation enabled, batches of 20-60 chunks process at 3-6x real-time speed
• Sub-batching - Automatically groups similarly-sized chunks together for efficient GPU utilization
• Single seed - All voices share the Batch Seed from config (set empty for random)
• All voice types supported - Custom, Clone, and LoRA voices are batched; Voice Design is sequential
• Parallel Workers setting controls batch size (higher values use more VRAM)

Result Tab

Download your completed audiobook as MP3, export as M4B with chapter markers for audiobook players, or click Export to Audacity for per-speaker WAV tracks.

• Download MP3 - Standard merged audiobook
• Export M4B - AAC audiobook with chapter markers. By default, chapters are auto-detected from headings in the script (e.g. "Chapter 1", "Prologue"). Toggle Per-chunk chapters for fine-grained navigation where every line becomes a chapter.
• Export to Audacity - Zip with per-speaker WAV tracks. Unzip and open project.lof in Audacity to load all tracks, then import labels.txt via File > Import > Labels for chunk annotations.

Note: Some Linux audiobook players (e.g. Cozy) have limited M4B support and may not detect the file. The M4B output has been tested with VLC, Haruna, and Audiobookshelf.

Performance

Recommended Settings for Batch Generation

Benchmarks

Tested on AMD RX 7900 XTX (24 GB VRAM, ROCm 6.3/7.2):

A 273-chunk audiobook (~54 minutes of audio) generates in approximately 16 minutes with batch mode and codec compilation enabled.

ROCm (AMD GPU) Notes

Linux only. AMD GPU acceleration requires ROCm 6.3+ on Linux. AMD GPUs on Windows run in CPU mode — see GPU Compatibility.

Alexandria automatically applies ROCm-specific optimizations when running on AMD GPUs:

• MIOpen fast-find mode - Prevents workspace allocation failures that cause slow GEMM fallback
• Triton AMD flash attention - Enables native flash attention for the whisper encoder
• triton_key compatibility shim - Fixes torch.compile on pytorch-triton-rocm

These are applied transparently and require no configuration.

ROCm 7.x GPU downclocking fix: ROCm 7.x has a regression where the GPU's DPM controller aggressively downclocks the shader engine between autoregressive generation steps, causing batch generation to slow to a crawl or appear to hang. The fix is to set the GPU power profile to COMPUTE, which enforces a minimum clock frequency floor:

echo 5 | sudo tee /sys/class/drm/card1/device/pp_power_profile_mode

This needs to be run once per boot (it does not persist across reboots). You can add it to your system startup or run it manually before launching Alexandria. To verify it's active, check for COMPUTE* in the output of:

cat /sys/class/drm/card1/device/pp_power_profile_mode

ROCm 6.x users and NVIDIA users are not affected.

Script Format

The generated script is a JSON array with speaker, text, and instruct fields:

[
  {"speaker": "NARRATOR", "text": "The door creaked open slowly.", "instruct": "Calm, even narration."},
  {"speaker": "ELENA", "text": "Ah! Who's there?", "instruct": "Startled and fearful, sharp whispered question, voice cracking with panic."},
  {"speaker": "MARCUS", "text": "Haha... did you miss me?", "instruct": "Menacing confidence, low smug drawl with a dark chuckle, savoring the moment."}
]

• instruct — 2-3 sentence TTS voice direction sent directly to the engine. Set tone, describe delivery, then give specific references. Example: "Devastated by grief, Sniffing between words and pausing to collect herself, end with a wracking sob."

Non-verbal Sounds

Vocalizations are written as real pronounceable text that the TTS speaks directly — no bracket tags or special tokens. The LLM generates natural onomatopoeia with short instruct directions:

• Gasps: "Ah!", "Oh!" with instruct like "Fearful, sharp gasp."
• Sighs: "Haah...", "Hff..."
• Laughter: "Haha!", "Ahaha..."
• Crying: "Hic... sniff..."
• Exclamations: "Mmm...", "Hmm...", "Ugh..."

Output Files

Final Audiobook:

• cloned_audiobook.mp3 - Combined audiobook with natural pauses

Individual Voicelines (for DAW editing):

voicelines/
├── voiceline_0001_narrator.mp3
├── voiceline_0002_elena.mp3
├── voiceline_0003_marcus.mp3
└── ...

Files are numbered in timeline order with speaker names for easy:

• Import into Audacity or other DAWs
• Placement on separate character tracks
• Fine-tuning of timing and effects

Audacity Export (per-speaker tracks):

audacity_export.zip
├── project.lof       # Open this in Audacity to import all tracks
├── labels.txt        # Import via File > Import > Labels for chunk annotations
├── narrator.wav      # Full-length track with only NARRATOR audio
├── elena.wav         # Full-length track with only ELENA audio
├── marcus.wav        # Full-length track with only MARCUS audio
└── ...

Each WAV track is padded to the same total duration with silence where other speakers are talking. Playing all tracks simultaneously sounds identical to the merged MP3.

M4B Audiobook (chaptered):

• audiobook.m4b - AAC audiobook with embedded chapter markers
• Chapters auto-detected from script headings, or per-chunk when toggled
• Compatible with Audiobookshelf, Apple Books, VLC, Haruna, and most audiobook players

API Reference

Alexandria exposes a REST API for programmatic access:

Configuration

# Get current config (empty prompts fall through to file defaults)
curl http://127.0.0.1:4200/api/config

# Get file-based default prompts (hot-reloads from default_prompts.txt)
curl http://127.0.0.1:4200/api/default_prompts

# Save config
curl -X POST http://127.0.0.1:4200/api/config \
  -H "Content-Type: application/json" \
  -d '{
    "llm": {"base_url": "...", "api_key": "...", "model_name": "..."},
    "tts": {
      "mode": "local",
      "device": "auto",
      "language": "English",
      "parallel_workers": 25,
      "batch_seed": 12345,
      "compile_codec": true,
      "sub_batch_enabled": true,
      "sub_batch_min_size": 4,
      "sub_batch_ratio": 5,
      "pause_between_speakers_ms": 500,
      "pause_same_speaker_ms": 250
    }
  }'

Script Generation

# Upload text file (supports .txt, .md, .epub)
curl -X POST http://127.0.0.1:4200/api/upload \
  -F "file=@mybook.epub"

# Generate script (returns task ID)
curl -X POST http://127.0.0.1:4200/api/generate_script

# Check status
curl http://127.0.0.1:4200/api/status/script_generation

# Review script (fix attribution tags, misattributed lines, etc.)
curl -X POST http://127.0.0.1:4200/api/review_script

# Check review status
curl http://127.0.0.1:4200/api/status/review

Voice Management

# Get voices and config
curl http://127.0.0.1:4200/api/voices

# Parse voices from script
curl -X POST http://127.0.0.1:4200/api/parse_voices

# Save voice config
curl -X POST http://127.0.0.1:4200/api/save_voice_config \
  -H "Content-Type: application/json" \
  -d '{"NARRATOR": {"type": "custom", "voice": "Ryan", "character_style": "calm"}}'

Chunk Management

# Get all chunks
curl http://127.0.0.1:4200/api/chunks

# Update a chunk
curl -X POST http://127.0.0.1:4200/api/chunks/5 \
  -H "Content-Type: application/json" \
  -d '{"text": "Updated dialogue", "instruct": "Excited, bright energy."}'

# Generate audio for single chunk
curl -X POST http://127.0.0.1:4200/api/chunks/5/generate

# Standard batch render (parallel individual calls)
curl -X POST http://127.0.0.1:4200/api/generate_batch \
  -H "Content-Type: application/json" \
  -d '{"indices": [0, 1, 2, 3, 4]}'

# Fast batch render (batched TTS calls, much faster)
curl -X POST http://127.0.0.1:4200/api/generate_batch_fast \
  -H "Content-Type: application/json" \
  -d '{"indices": [0, 1, 2, 3, 4]}'

# Merge all chunks into final audiobook
curl -X POST http://127.0.0.1:4200/api/merge

Saved Scripts

# List saved scripts
curl http://127.0.0.1:4200/api/scripts

# Save current script
curl -X POST http://127.0.0.1:4200/api/scripts/save \
  -H "Content-Type: application/json" \
  -d '{"name": "my-novel"}'

# Load a saved script
curl -X POST http://127.0.0.1:4200/api/scripts/load \
  -H "Content-Type: application/json" \
  -d '{"name": "my-novel"}'

Persona Generation

# Generate personas (LLM + VoiceDesign, assigns clone voices automatically)
curl -X POST http://127.0.0.1:4200/api/generate_personas

# Generate personas in advanced mode with custom batch size
curl -X POST http://127.0.0.1:4200/api/generate_personas \
  -H "Content-Type: application/json" \
  -d '{"advanced": true, "batch_size": 40}'

# Check persona generation status
curl http://127.0.0.1:4200/api/status/persona

# Cancel persona generation
curl -X POST http://127.0.0.1:4200/api/cancel_persona

Voice Designer

# Preview a voice from text description
curl -X POST http://127.0.0.1:4200/api/voice_design/preview \
  -H "Content-Type: application/json" \
  -d '{"description": "A warm, deep male voice", "text": "Hello world."}'

# Save a designed voice
curl -X POST http://127.0.0.1:4200/api/voice_design/save \
  -H "Content-Type: application/json" \
  -d '{"name": "warm_narrator", "description": "A warm, deep male voice", "text": "Hello world."}'

# List saved designed voices
curl http://127.0.0.1:4200/api/voice_design/list

# Delete a designed voice
curl -X DELETE http://127.0.0.1:4200/api/voice_design/delete/voice_id_here

LoRA Training

# Upload a training dataset (ZIP with WAV + metadata.jsonl)
curl -X POST http://127.0.0.1:4200/api/lora/upload_dataset \
  -F "file=@dataset.zip" -F "name=my_voice"

# Generate a dataset from Voice Designer description
curl -X POST http://127.0.0.1:4200/api/lora/generate_dataset \
  -H "Content-Type: application/json" \
  -d '{"name": "warm_voice", "description": "A warm male voice", "texts": ["Hello.", "Goodbye."]}'

# List uploaded datasets
curl http://127.0.0.1:4200/api/lora/datasets

# Delete a dataset
curl -X DELETE http://127.0.0.1:4200/api/lora/datasets/dataset_id_here

# Start LoRA training
curl -X POST http://127.0.0.1:4200/api/lora/train \
  -H "Content-Type: application/json" \
  -d '{"name": "narrator_warm", "dataset_id": "my_voice", "epochs": 25, "lr": "5e-6", "lora_r": 32, "lora_alpha": 64}'

# Check training status
curl http://127.0.0.1:4200/api/status/lora_training

# List trained adapters
curl http://127.0.0.1:4200/api/lora/models

# Test a trained adapter
curl -X POST http://127.0.0.1:4200/api/lora/test \
  -H "Content-Type: application/json" \
  -d '{"adapter_id": "narrator_warm_1234567890", "text": "Test line.", "instruct": "Calm narration."}'

# Delete an adapter
curl -X DELETE http://127.0.0.1:4200/api/lora/models/adapter_id_here

Dataset Builder

# List all dataset builder projects
curl http://127.0.0.1:4200/api/dataset_builder/list

# Create a new project
curl -X POST http://127.0.0.1:4200/api/dataset_builder/create \
  -H "Content-Type: application/json" \
  -d '{"name": "my_voice_dataset"}'

# Update project metadata (description and global seed)
curl -X POST http://127.0.0.1:4200/api/dataset_builder/update_meta \
  -H "Content-Type: application/json" \
  -d '{"name": "my_voice_dataset", "description": "A warm male narrator", "global_seed": "42"}'

# Update sample rows
curl -X POST http://127.0.0.1:4200/api/dataset_builder/update_rows \
  -H "Content-Type: application/json" \
  -d '{"name": "my_voice_dataset", "rows": [{"text": "Hello world.", "emotion": "cheerful"}]}'

# Generate a single sample preview
curl -X POST http://127.0.0.1:4200/api/dataset_builder/generate_sample \
  -H "Content-Type: application/json" \
  -d '{"name": "my_voice_dataset", "description": "A warm male voice", "sample_index": 0, "samples": [{"text": "Hello.", "emotion": "cheerful"}]}'

# Batch generate all samples
curl -X POST http://127.0.0.1:4200/api/dataset_builder/generate_batch \
  -H "Content-Type: application/json" \
  -d '{"name": "my_voice_dataset", "description": "A warm male voice", "samples": [{"text": "Hello.", "emotion": "cheerful"}]}'

# Check batch generation status
curl http://127.0.0.1:4200/api/dataset_builder/status/my_voice_dataset

# Cancel a running batch generation
curl -X POST http://127.0.0.1:4200/api/dataset_builder/cancel \
  -H "Content-Type: application/json" \
  -d '{"name": "my_voice_dataset"}'

# Save project as a training dataset
curl -X POST http://127.0.0.1:4200/api/dataset_builder/save \
  -H "Content-Type: application/json" \
  -d '{"name": "my_voice_dataset", "ref_sample_index": 0}'

# Delete a project
curl -X DELETE http://127.0.0.1:4200/api/dataset_builder/my_voice_dataset

Audio Download

# Download audiobook (after merging in editor)
curl http://127.0.0.1:4200/api/audiobook --output audiobook.mp3

# Export to Audacity (per-speaker tracks + LOF + labels)
curl -X POST http://127.0.0.1:4200/api/export_audacity

# Poll for completion
curl http://127.0.0.1:4200/api/status/audacity_export

# Download the zip
curl http://127.0.0.1:4200/api/export_audacity --output audacity_export.zip

Python Integration

import requests

BASE = "http://127.0.0.1:4200"

# Upload and generate script
with open("mybook.txt", "rb") as f:
    requests.post(f"{BASE}/api/upload", files={"file": f})

requests.post(f"{BASE}/api/generate_script")

# Poll for completion
import time
while True:
    status = requests.get(f"{BASE}/api/status/script_generation").json()
    if status.get("status") in ["completed", "error"]:
        break
    time.sleep(2)

# Configure voices
voice_config = {
    "NARRATOR": {"type": "custom", "voice": "Ryan", "character_style": "calm narrator"},
    "HERO": {"type": "custom", "voice": "Aiden", "character_style": "brave, determined"}
}
requests.post(f"{BASE}/api/save_voice_config", json=voice_config)

# Fast batch render all chunks
chunks = requests.get(f"{BASE}/api/chunks").json()
indices = [c["id"] for c in chunks]
requests.post(f"{BASE}/api/generate_batch_fast", json={"indices": indices})
# ... poll until all chunks status == "done" ...
requests.post(f"{BASE}/api/merge")

# Download
with open("output.mp3", "wb") as f:
    f.write(requests.get(f"{BASE}/api/audiobook").content)

# Export to Audacity
requests.post(f"{BASE}/api/export_audacity")
# ... poll /api/status/audacity_export until not running ...
with open("audacity_export.zip", "wb") as f:
    f.write(requests.get(f"{BASE}/api/export_audacity").content)

JavaScript Integration

const BASE = "http://127.0.0.1:4200";

// Upload file
const formData = new FormData();
formData.append("file", fileInput.files[0]);
await fetch(`${BASE}/api/upload`, { method: "POST", body: formData });

// Generate script
await fetch(`${BASE}/api/generate_script`, { method: "POST" });

// Poll for completion
async function waitForTask(taskName) {
  while (true) {
    const res = await fetch(`${BASE}/api/status/${taskName}`);
    const data = await res.json();
    if (data.status === "completed" || data.status === "error") return data;
    await new Promise(r => setTimeout(r, 2000));
  }
}
await waitForTask("script_generation");

// Configure and generate
await fetch(`${BASE}/api/save_voice_config`, {
  method: "POST",
  headers: { "Content-Type": "application/json" },
  body: JSON.stringify({
    NARRATOR: { type: "custom", voice: "Ryan", character_style: "calm" }
  })
});

// Fast batch render all chunks
const chunks = await (await fetch(`${BASE}/api/chunks`)).json();
const indices = chunks.map(c => c.id);
await fetch(`${BASE}/api/generate_batch_fast`, {
  method: "POST",
  headers: { "Content-Type": "application/json" },
  body: JSON.stringify({ indices })
});
// ... poll until all chunks done ...

// Merge into final audiobook
await fetch(`${BASE}/api/merge`, { method: "POST" });

// Export to Audacity
await fetch(`${BASE}/api/export_audacity`, { method: "POST" });
// ... poll /api/status/audacity_export until not running ...
// Download zip from GET /api/export_audacity

Recommended LLM Models

For script generation, non-thinking models work best:

• Qwen3-next (80B-A3B-instruct) - Excellent JSON output and instruct directions
• Gemma3 (27B recommended) - Strong JSON output and instruct directions
• Qwen2.5 (any size) - Reliable JSON output
• Qwen3 (non-thinking variant)
• Llama 3.1/3.2 - Good character distinction
• Mistral/Mixtral - Fast and reliable

Thinking models (DeepSeek-R1, GLM4-air, etc.) can interfere with JSON output. If you must use one, add <think> to the Banned Tokens field in Setup to disable thinking mode.

Troubleshooting

Script generation fails

• Check LLM server is running and accessible
• Verify model name matches what's loaded
• Try a different model - some struggle with JSON output

Model download fails or is very slow

• TTS models (~3.5 GB each) are downloaded from Hugging Face on first use
• If downloads are slow or fail due to network restrictions (common in mainland China), set a Hugging Face mirror before launching:
  • Set the environment variable HF_ENDPOINT=https://hf-mirror.com before starting the app
  • Or in Pinokio, add it to start.js env field: env: { HF_ENDPOINT: "https://hf-mirror.com" }
• If you hit rate limits, create a free Hugging Face account and set HF_TOKEN to your access token
• Downloads resume automatically if interrupted — just restart the app

TTS generation fails

• Check the Pinokio terminal for model loading errors
• Ensure sufficient VRAM (16+ GB recommended for bfloat16)
• For external mode, ensure the Gradio TTS server is running at the configured URL
• Check voice_config.json has valid settings for all speakers
• For clone voices, verify reference audio exists and transcript is accurate

Slow batch generation

• Enable Compile Codec in Setup (adds warmup time but 3-4x faster after)
• Increase Parallel Workers (batch size) if VRAM allows
• Use Batch (Fast) render mode instead of Standard
• If you see MIOpen warnings on AMD, these are handled automatically

Out of memory errors

• Reduce Parallel Workers (batch size)
• Close other GPU-intensive applications
• Try device: cpu as a fallback (much slower)

Broken or tiny MP3 files (428 bytes)

Conda's bundled ffmpeg on Windows often lacks the MP3 encoder (libmp3lame). Alexandria now detects this and automatically falls back to WAV, but if you want MP3 output:

• Install ffmpeg with MP3 support: conda install -c conda-forge ffmpeg
• Or remove conda's ffmpeg to use your system one: conda remove ffmpeg
• Verify with: ffmpeg -encoders 2>/dev/null | grep mp3

Audio quality issues

• Use 5-15 second clear reference audio for cloning
• Avoid background noise in reference samples
• Try different seeds for custom voices

Mojibake characters in output

• The system automatically fixes common encoding issues
• If problems persist, ensure your input text is UTF-8 encoded

Prompt Customization

LLM prompts are stored in plain-text files at the project root, split into system prompt and user prompt sections by a ---SEPARATOR--- delimiter:

• default_prompts.txt — Prompts for script generation (annotation)
• review_prompts.txt — Prompts for script review (error correction)

How it works:

• app/default_prompts.py and app/review_prompts.py read their respective files and export the prompts
• Prompts hot-reload from disk on every request, so edits take effect immediately without restarting the app
• config.json stores user overrides for generation prompts — when its prompt fields are empty, the file defaults are used
• The "Reset to Defaults" button in the Web UI fetches the latest file defaults via /api/default_prompts

To customize prompts:

1. Temporary (per-session): Edit generation prompts directly in the Setup tab's Prompt Customization section
2. Permanent (all sessions): Edit default_prompts.txt or review_prompts.txt directly — changes are picked up on the next request

Non-English books: The default LLM prompts are written for English text and reference English-specific conventions (attribution tags like "said he", quotation marks, etc.). When processing books in other languages, you'll get better results by editing the prompts to match that language's dialogue conventions — for example, French guillemets (« »), Japanese brackets (「」), or language-appropriate attribution patterns. Set the TTS Language dropdown to match as well.

Project Structure

Alexandria/
├── app/
│   ├── app.py                 # FastAPI server
│   ├── tts.py                 # TTS engine (local + external backends)
│   ├── train_lora.py          # LoRA training subprocess script
│   ├── generate_script.py     # LLM script annotation
│   ├── generate_personas.py   # LLM persona generation + VoiceDesign voice assignment
│   ├── review_script.py       # LLM script review (second pass)
│   ├── utils.py               # Shared utilities (atomic JSON writes)
│   ├── default_prompts.py     # Generation prompt loader (reads default_prompts.txt)
│   ├── review_prompts.py      # Review prompt loader (reads review_prompts.txt)
│   ├── project.py             # Chunk management & batch generation
│   ├── parse_voices.py        # Voice extraction
│   ├── config.json            # Runtime configuration (gitignored)
│   ├── static/index.html      # Web UI
│   └── requirements.txt       # Python dependencies
├── builtin_lora/              # Pre-trained LoRA voice presets
├── dataset_builder/           # Dataset builder project workspace (gitignored)
├── designed_voices/           # Saved Voice Designer outputs (gitignored)
├── lora_datasets/             # Uploaded/generated training datasets (gitignored)
├── lora_models/               # Trained LoRA adapters (gitignored)
├── default_prompts.txt        # LLM prompts for script generation
├── review_prompts.txt         # LLM prompts for script review
├── install.js                 # Pinokio installer
├── start.js                   # Pinokio launcher
├── reset.js                   # Reset script
├── pinokio.js                 # Pinokio UI config
├── pinokio.json               # Pinokio metadata
└── README.md

Acknowledgements

• Ayush Naphade — Persona generation, speaker alias resolution, and contextual script review (PR #42). Check out his project Lily — looking forward to seeing where it goes!
• Michii — System Health Dashboard with real-time GPU/disk monitoring (PR #45), cross-platform subprocess runner (PR #46), Voice Training Dataset Preparer tab (PR #47)

License

MIT

Third-Party Licenses

• qwen_tts — Apache License 2.0, Copyright Alibaba Qwen Team