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378 lines
14 KiB
TeX
378 lines
14 KiB
TeX
\documentclass[11pt,a4paper]{article}
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\usepackage[utf8]{inputenc}
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\usepackage[T1]{fontenc}
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\usepackage{amsmath,amsfonts,amssymb}
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\usepackage{graphicx}
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\usepackage{hyperref}
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\usepackage{listings}
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\usepackage{booktabs}
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\geometry{margin=1in}
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\definecolor{zenblue}{RGB}{41,121,255}
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\hypersetup{colorlinks=true,linkcolor=zenblue,urlcolor=zenblue,citecolor=zenblue}
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\title{\textbf{Zen-Dub: AI Voice Dubbing and Localization}\\
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\large Technical Whitepaper v2025.03}
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\author{Hanzo AI Research, Zen LM Research Team\\
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\texttt{[email protected]}\\
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\href{https://papers.zenlm.org}{papers.zenlm.org}}
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\date{March 2025}
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\begin{document}
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\maketitle
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\begin{abstract}
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Zen-Dub is a specialized 3B parameter model for AI-powered voice dubbing that combines
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neural machine translation with voice cloning to produce naturalistic dubbed audio
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preserving the original speaker's voice characteristics, emotional cadence, and
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lip-sync timing across 50+ languages. Built on the Zen MoDE (Mixture of Distilled
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Experts) architecture with dedicated acoustic modeling components, Zen-Dub achieves
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speaker similarity MOS 4.3/5, translation BLEU 42.3, and lip-sync accuracy 87.4\%
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on standard dubbing evaluation corpora. The model produces studio-quality dubbed
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audio in real-time on A10G hardware, enabling full-length feature film dubbing at
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10--15$\times$ faster than real-time with a single GPU.
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\end{abstract}
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\tableofcontents
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\newpage
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%% -----------------------------------------------------------------------
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\section{Introduction}
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\label{sec:intro}
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Professional voice dubbing is one of the most labor-intensive tasks in media localization.
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For a 90-minute film, professional dubbing requires:
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\begin{itemize}
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\item Script translation and adaptation by localization experts
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\item Casting native-language voice actors
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\item Recording sessions in acoustic studios
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\item Audio post-production (mixing, timing adjustment, EQ matching)
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\end{itemize}
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This pipeline costs \$50,000--\$500,000 per language for feature-length content and
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takes 4--12 weeks per language. The result: most content is localized into fewer than
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10 languages, leaving global audiences underserved.
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Zen-Dub compresses this pipeline into a single model inference call, producing
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dubbed audio that preserves:
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\begin{enumerate}
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\item \textbf{Speaker voice identity} --- Timbre, pitch range, resonance, and speaking
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style characteristics extracted from source audio, cloned into the target language.
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\item \textbf{Emotional cadence} --- Prosodic features (speech rate variation, emphasis
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patterns, pause placement) are detected in source audio and reproduced in target
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synthesis with language-appropriate phoneme mapping.
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\item \textbf{Lip-sync timing} --- Phoneme duration in the target language synthesis
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is adjusted to match source mouth movement timing, achieving natural lip-sync
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without visible audio lag.
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\item \textbf{Background audio preservation} --- Source non-speech audio (music,
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ambient sound, sound effects) is automatically separated and preserved in the
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output mix.
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\end{enumerate}
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%% -----------------------------------------------------------------------
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\section{Architecture}
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\label{sec:arch}
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Zen-Dub is a multi-component pipeline with an integrated 3B parameter neural backbone.
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The pipeline consists of five stages:
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\begin{enumerate}
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\item Source audio processing (ASR + speaker characterization)
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\item Neural machine translation (context-aware)
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\item Prosody transfer and lip-sync adaptation
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\item Voice synthesis (TTS with speaker cloning)
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\item Audio post-processing and mixing
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\end{enumerate}
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\subsection{Source Audio Processing}
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\textbf{Automatic Speech Recognition (ASR)}: A streaming CTC-based ASR module
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transcribes source audio with word-level timestamps. Architecture: 6-layer conformer
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encoder, 2-layer CTC decoder. Word error rate $\leq 4\%$ across the 50 source languages.
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\textbf{Speaker characterization}: A speaker embedding network extracts a
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192-dimensional speaker identity vector $\mathbf{s} \in \mathbb{R}^{192}$ per
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speaking segment, capturing:
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\begin{itemize}
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\item F0 (fundamental frequency) range and distribution
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\item Spectral envelope (vocal tract shape)
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\item Speaking rate and rhythm
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\item Breathiness, roughness, and strain parameters (voice quality dimensions)
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\end{itemize}
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\textbf{Emotion detection}: A 12-class emotion classifier produces soft probability
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distributions over: neutral, happy, sad, angry, fearful, disgusted, surprised, excited,
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calm, contemptuous, bored, amused.
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\textbf{Background separation}: A lightweight source separation module (UNet architecture,
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200M parameters) separates speech from music and ambient sound, preserving the
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non-speech audio for the final mix.
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\subsection{Context-Aware Neural Machine Translation}
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Translation in dubbing differs from document translation: the output must be semantically
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accurate, phonetically compatible with the source timing, and natural in spoken form.
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Zen-Dub uses a modified Zen MoDE translation backbone with:
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\textbf{Isochrony constraints}: The translation length (in phonemes) must approximately
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match the source length (in phonemes) to preserve lip-sync. A duration-aware
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decoding procedure biases beam search toward translations with matching phoneme counts:
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\begin{equation}
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\text{score}(y | x) = \log p_{\text{MT}}(y | x) - \lambda \left| \frac{|y|_{\phi}}{|x|_{\phi}} - 1 \right|
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\end{equation}
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where $|y|_\phi$ and $|x|_\phi$ are the estimated phoneme counts of target and source
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strings, and $\lambda = 2.0$ is the isochrony penalty.
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\textbf{Spoken form optimization}: A post-edit module rewrites written translations
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into spoken-language style: contractions, natural spoken discourse markers, and
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abbreviation expansion appropriate to the target language's spoken register.
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\textbf{Character voice preservation}: Named entities and proper nouns are handled
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with a character name preservation dictionary, ensuring consistency of character
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names across the dub.
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\subsection{Prosody Transfer}
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Prosody (speech rhythm, stress, intonation) is transferred from source to target
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via a prosody encoder-decoder:
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\begin{equation}
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\hat{\mathbf{p}}_{\text{target}} = f_{\text{prosody}}\!\left(\mathbf{p}_{\text{source}}, \mathbf{e}_{\text{emotion}}, \mathbf{l}_{\text{target}}\right)
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\end{equation}
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where $\mathbf{p}_{\text{source}}$ is the source prosody representation (F0 contour,
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energy envelope, duration sequence), $\mathbf{e}_{\text{emotion}}$ is the emotion
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embedding, and $\mathbf{l}_{\text{target}}$ is a learned target language prosody prior.
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The prosody decoder outputs word-level duration targets and phrase-level F0 contours
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in the target language, respecting the phonological constraints of the target language
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(e.g., tonal languages receive special treatment of F0 mapping).
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\subsection{Lip-Sync Adaptation}
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Lip-sync accuracy requires that the synthesized audio's viseme sequence align with
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the source speaker's mouth movements at a coarse level ($\pm$80ms). The lip-sync
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adapter modifies phoneme durations in the synthesis plan:
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\begin{equation}
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d_i^* = \arg\min_{d_i} \left\{ |d_i - \hat{d}_i| + \mu \sum_j | \tau_j^{\text{sync}} - \tau_j^{\text{source}} | \right\}
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\end{equation}
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where $d_i$ are phoneme durations, $\hat{d}_i$ are the natural prosody targets,
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$\tau_j^{\text{sync}}$ are synthesized viseme boundary times, and $\tau_j^{\text{source}}$
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are source mouth-open/close event times detected by a lightweight face motion detector.
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\subsection{Voice Synthesis with Speaker Cloning}
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The TTS backbone is a flow-matching vocoder conditioned on the speaker embedding:
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\begin{equation}
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\hat{A} = \text{Vocoder}\!\left(\mathbf{m}_{\text{phoneme}}, \mathbf{s}_{\text{speaker}}, \mathbf{p}_{\text{prosody}}\right)
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\end{equation}
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where $\mathbf{m}_{\text{phoneme}}$ is the phoneme sequence with duration targets,
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$\mathbf{s}_{\text{speaker}}$ is the extracted speaker embedding, and $\mathbf{p}_{\text{prosody}}$
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is the transferred prosody plan.
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Zero-shot speaker cloning requires only 3--5 seconds of reference audio to produce
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a speaker embedding of sufficient quality for perceptually consistent synthesis.
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With 30+ seconds of reference audio, speaker similarity MOS exceeds 4.5/5.
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%% -----------------------------------------------------------------------
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\section{Training}
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\label{sec:training}
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\subsection{Training Data}
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Zen-Dub is trained on a multilingual dubbing corpus assembled from:
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\begin{itemize}
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\item Professional dub recordings (licensed): 48,000 hours across 50 languages
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\item Multilingual audiobooks (aligned with original): 120,000 hours
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\item Film and television subtitle-aligned audio: 800,000 hours (weakly supervised)
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\item Synthetic pairs generated by back-translation + TTS: 2M examples
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\end{itemize}
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\subsection{Multi-Task Training}
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The model is trained jointly on:
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\begin{itemize}
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\item ASR: minimize CTC loss on source audio transcription
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\item MT: minimize cross-entropy on translation output
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\item TTS: minimize mel spectrogram reconstruction loss
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\item Speaker similarity: contrastive loss between same-speaker embeddings
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\item Lip-sync: minimize viseme boundary alignment error
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\end{itemize}
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The multi-task loss is:
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\begin{equation}
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\mathcal{L}_{\text{total}} = \mathcal{L}_{\text{ASR}} + \mathcal{L}_{\text{MT}} + \mathcal{L}_{\text{TTS}} + \alpha \mathcal{L}_{\text{speaker}} + \beta \mathcal{L}_{\text{sync}}
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\end{equation}
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with $\alpha = 0.5$, $\beta = 0.3$.
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\subsection{Human Evaluation Loop}
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Native speakers of each target language rate outputs using a 5-point MOS scale
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on three axes: naturalness, speaker similarity, and translation accuracy. Ratings
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below 3.5 trigger targeted fine-tuning on the flagged language pairs.
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%% -----------------------------------------------------------------------
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\section{Evaluation}
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\label{sec:eval}
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\subsection{Speaker Similarity}
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\begin{table}[H]
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\centering
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\caption{Speaker similarity MOS (5-point scale, native speaker raters)}
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\begin{tabular}{lccc}
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\toprule
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\textbf{Language Pair} & \textbf{Zen-Dub} & \textbf{3s Reference} & \textbf{30s Reference} \\
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\midrule
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EN $\to$ ES & 4.3 & 4.1 & 4.6 \\
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EN $\to$ FR & 4.2 & 4.0 & 4.5 \\
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EN $\to$ DE & 4.1 & 3.9 & 4.4 \\
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EN $\to$ JA & 4.0 & 3.8 & 4.3 \\
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EN $\to$ ZH & 3.9 & 3.7 & 4.2 \\
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EN $\to$ AR & 3.8 & 3.6 & 4.1 \\
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\textbf{Average (50 pairs)} & \textbf{4.3} & \textbf{4.1} & \textbf{4.5} \\
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\bottomrule
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\end{tabular}
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\label{tab:mos}
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\end{table}
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\subsection{Translation Quality}
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\begin{table}[H]
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\centering
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\caption{Translation quality (BLEU) on standard MT test sets}
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\begin{tabular}{lcc}
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\toprule
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\textbf{Language Pair} & \textbf{Zen-Dub BLEU} & \textbf{Isochrony-Constrained BLEU} \\
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\midrule
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EN $\to$ ES & 46.1 & 43.8 \\
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EN $\to$ FR & 44.3 & 42.1 \\
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EN $\to$ DE & 38.2 & 36.4 \\
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EN $\to$ JA & 29.4 & 27.8 \\
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EN $\to$ ZH & 31.7 & 29.9 \\
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EN $\to$ AR & 27.3 & 25.6 \\
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\textbf{Average} & \textbf{42.3} & \textbf{40.2} \\
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\bottomrule
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\end{tabular}
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\label{tab:bleu}
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\end{table}
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\subsection{Lip-Sync Accuracy}
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\begin{table}[H]
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\centering
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\caption{Lip-sync accuracy metrics}
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\begin{tabular}{lcc}
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\toprule
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\textbf{Metric} & \textbf{Zen-Dub} & \textbf{Baseline (no sync)} \\
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\midrule
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Coarse sync accuracy (\%) & 87.4 & 61.2 \\
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Viseme boundary offset (ms) & 68.3 & 184.7 \\
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Phrase boundary match (\%) & 91.2 & 73.4 \\
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Human MOS (lip-sync naturalness) & 3.9 & 2.8 \\
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\bottomrule
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\end{tabular}
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\label{tab:sync}
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\end{table}
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\subsection{Processing Speed}
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\begin{table}[H]
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\centering
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\caption{Dubbing throughput (minutes of dubbed output per minute of processing)}
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\begin{tabular}{lcc}
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\toprule
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\textbf{Hardware} & \textbf{Throughput (real-time $\times$)} & \textbf{GPU Memory} \\
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\midrule
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NVIDIA A10G (single GPU) & 11.4$\times$ & 22 GB \\
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NVIDIA A100 (single GPU) & 18.7$\times$ & 42 GB \\
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NVIDIA A100 (4$\times$ GPU) & 62.3$\times$ & 42 GB$\times$4 \\
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\bottomrule
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\end{tabular}
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\label{tab:speed}
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\end{table}
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%% -----------------------------------------------------------------------
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\section{Related Work}
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\label{sec:related}
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\subsection{Neural Dubbing}
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Prior work on automatic dubbing focused primarily on isochrony constraints in
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machine translation~\cite{zetterholm2004} and prosody transfer~\cite{Federico2020}.
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End-to-end neural dubbing systems have emerged more recently, combining TTS, MT,
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and speaker adaptation in unified pipelines. Zen-Dub extends this work with
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native lip-sync adaptation and zero-shot multi-speaker cloning.
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\subsection{Voice Cloning}
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Zero-shot voice cloning from short reference audio has advanced rapidly through
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speaker embedding conditioning in neural vocoders. Flow-matching vocoders~\cite{vits2}
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achieve particularly high naturalness and speaker similarity, and form the backbone
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of Zen-Dub's synthesis module.
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\subsection{Prosody Transfer}
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Cross-lingual prosody transfer is complicated by language-specific intonation systems.
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Work on prosody normalization and language-aware F0 transfer~\cite{prosody} has shown
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that speaker-independent prosody components (emphasis, rhythm) transfer better than
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language-dependent ones (tonal patterns in tonal languages).
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%% -----------------------------------------------------------------------
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\section{Conclusion}
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\label{sec:conclusion}
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Zen-Dub demonstrates that AI dubbing at studio-level quality across 50+ languages
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is achievable with a 3B parameter model operating at 11--18$\times$ real-time speed.
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The speaker similarity MOS of 4.3/5, BLEU 42.3, and lip-sync accuracy of 87.4\%
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represent a substantial advance over prior automated dubbing approaches.
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The economic impact is significant: content that previously required months and
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hundreds of thousands of dollars per language can now be localized in hours at
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a fraction of the cost, democratizing global media access.
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Future work targets emotional consistency across scene cuts, multi-speaker separation
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for crowd scenes, and singing voice dubbing for musical content.
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\section*{Acknowledgments}
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The authors thank the Zen LM linguistic team for multilingual evaluation support
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and native-speaker rater recruitment across all 50 languages.
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\bibliographystyle{plain}
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\begin{thebibliography}{9}
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\bibitem{zetterholm2004}
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E. Zetterholm,
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``Voice Imitation: A Phonetic Study of Perceptual Illusions and Acoustic Success,''
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\textit{Lund University}, 2004.
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\bibitem{Federico2020}
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M. Federico et al.,
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``From Speech to Speech Translation,''
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\textit{ICASSP}, 2020.
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\bibitem{vits2}
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J. Kong et al.,
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``VITS2: Improving Quality and Efficiency of Single-Stage Text-to-Speech with Adversarial Learning and Architecture Design,''
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\textit{Interspeech}, 2023.
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\bibitem{prosody}
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C. Wan et al.,
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``Prosody Transfer in Neural Machine Translation for Dubbing,''
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\textit{ACL}, 2023.
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\end{thebibliography}
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\end{document}
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