Binance39mo ago

Client-Side Audio/Video Development Engineer (Android / iOS / Web)

AsiaRemoteFull-time: Remotemid

OtherDevelopment Engineer

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Binance is a leading global blockchain ecosystem behind the world’s largest cryptocurrency exchange by trading volume and registered users. We are trusted by 300+ million people in 100+ countries for our industry-leading security, user fund transparency, trading engine speed, deep liquidity, and an unmatched portfolio of digital-asset products. Binance offerings range from trading and finance to education, research, payments, institutional services, Web3 features, and more. We leverage the power of digital assets and blockchain to build an inclusive financial ecosystem to advance the freedom of money and improve financial access for people around the world.

Develop and optimize client-side RTC/live streaming interaction capabilities: co-streaming (joining/leaving the stream), audio routing, in-ear monitoring/audio mixing, weak network strategies, video quality/bitrate adaptation, and foreground/background/system interruption handling.

Deeply integrate third-party RTC SDKs (e.g., Agora) or in-house SDKs: integration, version upgrades, parameter management, compatibility and regression testing, online issue troubleshooting (via logs/callbacks/state machines).

Responsible for client stability and performance: audio/video threading models, CPU/memory/power optimization, end-to-end latency reduction, troubleshooting of complex issues such as stuttering, no audio/black screen.

Build client observability: key path instrumentation, QoS reporting, exception capture, automatic issue attribution (network/device/permissions/state machine/SDK).

Collaborate with server-side, testing, and operations teams: formulate stress testing plans, conduct rehearsals, and handle online incident emergency response and post-mortems.

3+ years of Android/iOS/Web (one or multiple platforms) R&D experience; experience with audio/video/live streaming/IM/conference products is preferred.

Familiar with audio/video fundamentals on your platform:

Android: AudioRecord/AudioTrack, AudioFocus, permissions & background restrictions, device compatibility.

iOS: AVAudioSession, audio routing/interruptions, ReplayKit/background strategies.

Web: WebRTC APIs, getUserMedia, RTCPeerConnection, browser compatibility & security policies.

Understanding of real-time audio/video principles and common issues: packet loss/jitter/latency, echo & howling, audio routing errors, permission issues, state machine errors leading to "mutual hearing failure," etc.

Strong engineering skills: modular design, testability, stability governance (Crash/ANR), performance profiling and optimization.

Ability to understand and utilize core callbacks and diagnostic information from RTC SDKs; capable of online issue localization (building evidence chains using QoS metrics and logs).

In-depth experience with WebRTC (e.g., having read/modified parts of the source code, performed custom builds).

Experience adapting to complex network environments: overseas networks, weak network strategies, UDP restrictions, NAT traversal related to enterprise/campus networks (knowledge of ICE/STUN significantly enhances troubleshooting efficiency).

Specialization in audio: 3A (AEC/NS/AGC), audio mixing/voice changing/spatial audio, on-device noise reduction, or AI enhancement.

Client-side low-level capabilities: C++ core libraries, JNI/NDK, Metal/OpenGL, hardware encoder/decoder tuning, optimization for low-end devices.

Experience supporting large-scale events: pre-event rehearsals, gradual rollouts (gray releases), feature toggles, emergency strategies, and rapid rollback plans.

Experience developing real-time live streaming filters.