Client-Side Audio/Video Development Engineer (Android / iOS / Web)
at Binance
Posted 7 hours ago
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Develop and optimize client-side RTC and live streaming interactions, including co-streaming, audio routing, in-ear monitoring, weak network strategies, and video quality/bitrate adaptation. Deeply integrate third-party RTC SDKs (e.g., Agora) or in-house SDKs, covering integration, upgrades, parameter management, compatibility/testing, and online issue troubleshooting. Focus on client stability and performance, addressing audio/video threading, CPU/memory/power optimization, end-to-end latency reduction, and issues such as stuttering or no audio. Build client observability with instrumentation, QoS reporting, exception capture, and automatic issue attribution, and collaborate with server/testing/operations teams on stress testing, rehearsals, incident responses, and post-mortems.
Responsibilities
- 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.
Minimum Qualifications
- 3+ years of Android/iOS/Web (one or multiple platforms) R&D experience; experience with audio/video/live streaming/IM/conference products is preferred.
- 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).
Preferred qualifications
- 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.

