This article provides a comprehensive overview of the Linux time subsystem, covering its hardware foundations, time concepts, user‑space and kernel APIs across multiple generations, timer implementations, scheduler tick interaction, and the overall design of the system clock.
To solve duplicated functionality, legacy code, and scalability limits in Bilibili’s live‑streaming interaction services, the team created a unified Interaction Center that abstracts RTC, consolidates session, link, UI, scoring and role management, introduces a shared state machine and tracing, and evolves through phased, extensible architecture for higher performance and maintainability.
This article introduces a data‑driven, objective evaluation method for Acoustic Echo Cancellation (AEC), detailing test environments, hardware setups, core metrics, single‑talk and double‑talk scenarios, scoring models, and result analysis to help developers assess and improve AEC algorithms across devices.
This article explores the challenges of real‑time communication audio problems, outlines their common manifestations and characteristics, and presents a comprehensive troubleshooting framework with standardized processes, automation tools, and perception models to improve efficiency and service quality.
This article introduces real‑time communication (RTC) and quality‑of‑service (QoS) concepts, then details audio QoS methods such as bandwidth estimation, DTX, FEC, RED, RTX/NACK, acceleration, deceleration, and PLC, while highlighting NetEase Cloud Communication's optimizations and practical considerations.
This article details OPPO's low‑latency audio‑video technology for cloud rendering, covering transmission requirements, the integration of RTC with cloud rendering, streaming architecture, optimization techniques across encoding, transport, and reception, and future outlooks for ultra‑low‑delay applications.
This article examines the unique technical hurdles of video‑conference RTC—such as free mic control, flexible layouts, screen sharing, weak networks, diverse devices and poor lighting—and outlines the edge‑rendering, smart encoding, multi‑level simulcast and reliable signaling solutions we’ve built to tackle them.
The article describes NetEase Cloud Music’s unified real‑time communication framework, which consolidates disparate audio/video features into a single, container‑based architecture using a common IPlayer interface, dynamic method generation, switchable players, and state‑machine management to streamline development, ensure business isolation, and provide seamless, controllable RTC services across all apps.
This article explains why real‑time audio/video (RTC) quality testing is critical in the 5G era, outlines an ideal communication model, details common audio issues, and presents both subjective and objective evaluation methods along with a fully automated laboratory setup for reproducible results.
This article traces the 25‑year evolution of real‑time communication (RTC) across five generations—from the early MCU era to P2P, SFU, cloud‑SFU, and the current global cloud RTC stage—highlighting key protocols, architectural shifts, advantages, drawbacks, and emerging AI‑driven media processing techniques.
The presentation details the TalRTC real‑time communication platform used in online education, covering its product overview, three‑layer architecture, high‑availability and weak‑network strategies, as well as special optimizations for teaching scenarios that improve audio‑video quality and reliability.
This article explains the fundamentals of RTC, distinguishes it from traditional streaming, highlights WebRTC's cross‑platform, plugin‑free benefits, and shows how low‑latency real‑time communication can enhance interactive applications.
The article outlines XdfRTC's high‑concurrency, global coverage, low‑latency, elastic scaling and multi‑platform support features, then details the evolution of RTC media forwarding architectures—including MESH, MCU, SFU, mesh‑based, and hierarchical designs—highlighting their schemes, advantages, and drawbacks.
This article provides an in‑depth technical analysis of integrating real‑time audio/video (RTC) with legacy PSTN, covering latency sources, protocol and codec differences, adaptation layers, system architecture, and optimization techniques such as jitter buffering, ARQ/FEC, and automatic failover.
At a Cloud+ Community salon on June 29, nearly 300 developers heard Tencent Cloud detail mobile co‑hosting live streaming, P2P‑accelerated CDN, globally distributed overseas streaming architecture, RTC‑to‑PSTN integration, and AI‑driven video services such as recognition, moderation, compression, editing, and highlight extraction.
This technical sharing session by a senior audio‑video engineer from 360 Video Cloud explains core concepts of video and audio, their encoding pipelines, media processing techniques, streaming protocols, and the challenges and key technologies behind real‑time communication (RTC).
The article chronicles how Tencent Video Cloud built a low‑latency audio/video SDK for WeChat mini‑programs—using live‑pusher and live‑player components to capture, process, encode, and transmit streams via TCP/UDP, adding echo cancellation, QoS, and room‑based signaling to enable real‑time chat and multi‑party conferencing within a 500 ms end‑to‑end delay.
