New opportunity

Audio AI Engineer

Zoom · Singapore

About this role

Zoom’s Audio team is seeking an Audio AI Engineer in Singapore to develop AI-driven features for real-time audio and video communication. The role focuses on low-latency streaming speech models that improve intelligibility and naturalness while preserving speaker identity and expressiveness.

Responsibilities

  • Research, design and develop algorithms for accent conversion, voice conversion, speech synthesis and automatic speech recognition.
  • Prototype end-to-end audio models and work with product and platform teams to integrate them into real-time communication systems.
  • Evaluate and optimize speech quality, latency, robustness and scalability. Keep current with speech-processing research and contribute through patents and internal knowledge sharing.

Qualifications

  • A PhD or equivalent experience in a relevant field involving streaming, accent conversion, voice conversion, text-to-speech or automatic speech recognition.
  • Proficiency with deep learning frameworks such as PyTorch or TensorFlow, and programming skills in Python, C/C++ or similar languages.
  • Understanding of sequence-modeling architectures, including Transformers, RNNs, diffusion models or conformers.
  • Experience developing and deploying low-latency, real-time speech or audio models using streaming architectures and optimized pipelines; familiarity with compression and acceleration methods including quantization, pruning and distillation; and experience with real-time audio systems in networked communication environments.
  • The posting calls for publication in top-tier conferences such as ICASSP, Interspeech, NeurIPS and ICLR. More than 2 years of relevant industry experience is considered a plus.

The position is full-time in Singapore. Zoom describes an office-and-remote hybrid approach generally but does not specify this role’s work style. Its benefits program offers options supporting health, work-life balance and community involvement.

For full details about this role, click “Apply on company site” to view the employer’s original posting.

Skills for this role

Accent conversionVoice conversionSpeech synthesisAutomatic speech recognitionStreaming architecturesReal-time audio systemsPyTorchTensorFlowPythonCC++TransformersRNNsDiffusion modelsConformersSequence modelingModel compressionQuantizationPruningKnowledge distillationSpeech processingDeep learningModel deploymentPerformance optimization

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Get interview-ready for this role

A focused preparation guide, built around this job’s responsibilities and requirements.

✦ AI-generated guide
Preparation suggestions, not the employer’s actual interview questions. Always check the original posting for current requirements.
01

Understand the role

What matters most

  • Develop streaming speech algorithms

    The role researches and develops accent conversion, voice conversion, speech synthesis, and automatic speech recognition for low-latency streaming architectures.

  • Preserve the speaker while improving speech

    End-to-end audio models are expected to improve intelligibility and naturalness without losing speaker identity or expressiveness.

  • Bring models into live communication systems

    The engineer prototypes and refines models, then works with product and platform teams to integrate them into real-time audio and video communication.

  • Optimize across competing performance measures

    The posting calls for evaluating speech quality, latency, robustness, and scalability, alongside experience with optimized streaming pipelines and familiarity with model compression and acceleration.

  • Contribute research knowledge

    The role includes staying current with speech-processing research and contributing through patents and internal knowledge sharing; the posting also calls for top-tier conference publications.

Likely challenges to think through

  • Balancing transformation with speaker preservation

    A likely challenge is making converted speech clearer or more natural without weakening identity or expressiveness. Prepare to explain how you would compare those outcomes and recognize when an improvement in one harms another.

  • Meeting latency limits without sacrificing quality

    Streaming and compression can reduce computation but may affect speech quality or robustness. Prepare a concrete account of how you would measure these trade-offs and choose among architectural and acceleration options.

  • Moving from prototype to networked deployment

    A model that works in a prototype may be harder to integrate into a real-time communication pipeline while maintaining robustness and scalability. Prepare to discuss how you would work with product and platform teams to test and resolve integration constraints; the posting does not identify a specific existing problem.

03

Practice your answers

Technical practice

How would you design accent conversion for low-latency communication?

What to coverThe posting calls for accent conversion on streaming architectures that improve intelligibility while preserving speaker identity and expressiveness. Suggested answer focus: explain how you would balance conversion strength, naturalness and latency, and how you would check for unwanted changes to the speaker’s voice.

How would you evaluate a voice-conversion model?

What to coverThe role includes voice conversion and evaluation of speech quality, robustness and latency. Suggested answer focus: distinguish whether the intended voice change succeeds from whether speech remains intelligible and natural, then describe how you would examine failure cases.

What makes speech synthesis suitable for streaming use?

What to coverThe posting includes speech synthesis and low-latency streaming models. Suggested answer focus: discuss when a model can start producing audio, how it maintains continuity as more input arrives, and the trade-off between responsiveness and naturalness.

How would you approach streaming automatic speech recognition?

What to coverAutomatic speech recognition on low-latency streaming architectures is a stated responsibility. Suggested answer focus: explain how limited future context affects recognition and how you would balance response time, recognition quality and stability of partial results.

How would you structure and profile an end-to-end streaming audio pipeline?

What to coverThe role calls for prototyping end-to-end models and deploying them through optimized, real-time pipelines. Suggested answer focus: trace where buffering, model inference and other processing contribute to latency, then explain how measurements would guide changes without sacrificing quality.

How would you choose among sequence-modeling architectures for this role?

What to coverThe posting names Transformers, RNNs, diffusion models and conformers. Suggested answer focus: compare their suitability for streaming speech in terms of available context, computation, output quality and latency rather than claiming one is always best.

When would you use quantization, pruning or distillation?

What to coverThe posting explicitly names these model-compression and acceleration techniques. Suggested answer focus: explain what each changes, how you would measure speed and resource gains, and how you would detect unacceptable losses in speech quality or robustness.

How would you assess quality, latency, robustness and scalability together?

What to coverThe role requires evaluation and optimization across all four dimensions. Suggested answer focus: propose separate measurements for each and explain how you would compare model versions when an improvement in one dimension worsens another.

How would you troubleshoot an audio model in networked communication?

What to coverThe posting calls for experience with real-time audio systems in networked communication environments. Suggested answer focus: consider, as possible diagnostic factors rather than stated system details, variable arrival times or missing audio, and explain how you would distinguish input or pipeline problems from model failures.

How would you move a research prototype into a real-time communication system?

What to coverThe role involves PyTorch or TensorFlow, Python or C/C++ or similar languages, and collaboration with product and platform teams on integration. Suggested answer focus: explain how you would make model interfaces and performance measurable, validate the optimized implementation against the prototype, and resolve integration trade-offs with partner teams.

Behavioural practice

Tell me about a time you collaborated across teams to integrate an audio or ML model.

What to coverThe posting calls for close work with product and platform teams. Suggested answer focus: use a real situation, your specific actions to resolve differing constraints, and the resulting integration or decision.

Tell me about a time performance measurements changed your technical approach.

What to coverThe role includes optimizing speech quality, latency, robustness and scalability. Suggested answer focus: describe the situation and evidence, the trade-off you chose and actions you took, and the measured result.

Tell me about a time you turned speech-processing research into a practical prototype.

What to coverThe posting combines research with end-to-end model prototyping. Suggested answer focus: describe the research question and practical constraint, your experimental actions, and what the prototype demonstrated or failed to demonstrate.

Tell me about a time you shared technical findings with others.

What to coverThe role mentions staying current with speech research, patents and internal knowledge sharing. Suggested answer focus: give a real situation, explain how you communicated evidence and limitations, and describe the resulting decision or learning without assuming a patent or publication was required for your example.

02

Sharpen your knowledge

Topics to prepare

  • Speech-task trade-offs

    Posting: The role covers accent conversion, voice conversion, speech synthesis, and ASR, with goals of intelligibility, naturalness, speaker identity, and expressiveness. Revision exercise: Compare how changing an accent versus converting a voice could affect each goal, and explain the trade-offs.

  • Streaming sequence models

    Posting: Low-latency streaming architectures and understanding of Transformers, RNNs, diffusion models, or conformers are specified. Revision exercise: Sketch how two of these architectures would process successive audio chunks; identify where each might add delay.

  • Speech-model evaluation

    Posting: Models are evaluated on quality, latency, robustness, and scalability. Revision exercise: Design a comparison table for two streaming models, specifying what evidence you would gather for each dimension and how you would investigate a quality–latency trade-off.

  • Compression and acceleration

    Posting: Familiarity with quantization, pruning, and distillation is requested. Revision exercise: Explain how each could reduce inference cost and what speech-quality or speaker-identity regressions you would check afterward.

  • Networked real-time audio

    Posting: The role involves real-time audio systems in networked communication environments. Revision exercise: Trace an audio chunk from input through model inference to playback, identifying where buffering or variable network conditions could affect responsiveness. The specific failure cases are preparation suggestions, not stated requirements.

Skills to revise

  • Prototype a streaming model

    Posting: The role prototypes end-to-end audio models on streaming architectures. Practice building a small chunk-by-chunk speech-model prototype in PyTorch or TensorFlow and verify that its output depends only on audio available at each step. This is a suggested exercise, not a specified employer project.

  • Implement an optimized inference path

    Posting: Python and C/C++ or similar programming skills, optimized pipelines, and low-latency deployment experience are requested. Practice implementing and timing an audio inference path in a listed language, then document where computation or buffering adds latency.

  • Benchmark quality against latency

    Posting: The role analyzes speech quality, latency, robustness, and scalability. Practice running a repeatable comparison across audio segments and chunk sizes, recording output quality observations alongside processing time and explaining the resulting trade-offs.

  • Test a compression change

    Posting: Quantization, pruning, and distillation are named acceleration techniques. Apply one to a speech-model prototype, compare inference cost and output before and after, and note any intelligibility or speaker-identity change.

  • Explain integration boundaries

    Posting: The engineer works with product and platform teams to integrate models into real-time video and audio communication systems. As preparation, draw a proposed model input/output interface and explain its streaming behavior, latency assumptions, and failure handling; this interface is an exercise, not a stated Zoom design.

04

Your preparation checklist

  • Map the role’s requirements to evidence

    The posting calls for a PhD or equivalent experience, PyTorch or TensorFlow, programming in Python or C/C++, and top-tier publications. Prepare a one-page evidence map with specific work and publications you can substantiate; leave gaps explicit.

  • Build a speech-model walkthrough

    The role covers accent conversion, voice conversion, speech synthesis and ASR. Select a relevant project, if available, and prepare a concise account of the problem, your contribution, model choices, results and limitations.

  • Practice a streaming-system design

    Sketch how an end-to-end speech model could run in real-time communication. Explain chunking and state handling, compare relevant sequence architectures such as conformers and RNNs, and identify latency–quality trade-offs. Treat this as a preparation exercise, not a description of Zoom’s implementation.

  • Prepare an evaluation plan

    The posting names intelligibility, naturalness, speaker identity, expressiveness, latency, robustness and scalability. Draft a test matrix showing how you would assess these dimensions and diagnose a regression.

  • Show an optimization trade-off

    Prepare one example or design exercise comparing quantization, pruning or distillation for a streaming model. Show how you would measure speed, memory use and speech-quality changes rather than assume an optimization will help.

  • Explain a path from prototype to integration

    The role involves deploying optimized pipelines and working with product and platform teams on networked audio/video systems. Prepare a handoff diagram and discuss interfaces, network-related failure cases and what you would validate before release.

  • Bring targeted questions

    Ask which speech tasks are priorities, what latency and quality measures matter most, how research moves into product, and how patents or internal knowledge sharing fit the team. The page describes a general hybrid approach but does not specify this role’s work style, so ask about that if relevant.

AI-generated from the job information · Prepared Sep 30, 2026 (UTC)Useful preparation, not a guarantee of eligibility or an offer.

Availability last checked: 1 Oct 2026. Status can change after a check; confirm on the employer’s website. Report an issue · Our checking policy

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