Philipp Hancke tests why voice agents sometimes answer their own greetings. In his setup, echo reached transcription in 21% of Edge sessions and 4% of Chrome sessions. AEC dumps show how echo cancellation, gain control, and VAD thresholds interact.

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Chrome quietly delivers a neural model that changes part of its echo-cancellation pipeline. Philipp Hancke reverse engineers the model and compares echo suppression and speech recognition against AEC3.

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Stumbling over Microsoft’s Video Conferencing Dataset (VCD) I asked Claude to take on a video codec engineer’s favorite hot take: compare video codecs. It did a fairly good writeup using the tooling provided by libWebRTC. One of the most surprising results was that background blur makes H.264 a lot more effective. Other than that, the […]

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tl;dr: Opus encoding just got 20% faster on desktop Chromium by turning on AVX2 vectorization in libopus. Vectorization means one instruction doing the same arithmetic on a whole vector of values at once instead of one at a time. Codecs spend their hot loops running identical math across long buffers of samples, exactly where vectorization […]

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This might be my first editorial style post here. Fippo’s Is everyone switching to MoQ from WebRTC? started some threads on MoQ vs. WebRTC. I started to respond, but my responses quickly became too long so I decided to go even deeper with a post here. Fippo’s post shows hard data that Media over QUIC […]

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It is time for another edition of “Is everyone switching to…“. Cloudflare recently published a blog post about Media over Quic (MoQ) which made a number of statements about WebRTC that require some “clarification”. Let us start with that and look at MoQ and WebTransport after that. An odd understanding of WebRTC The blog post […]

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Earlier this month, OpenAI released the GA version of its realtime API. This includes many capabilities that the Beta didn’t have, including video support. I started out doing an update to the The Unofficial Guide to OpenAI’s Realtime WebRTC API I made for the Beta release last November. I discovered there were enough WebRTC updates […]

The post How OpenAI does WebRTC in the new gpt-realtime appeared first on webrtcHacks.

Audio jitter buffers are required 101 introductory material for understanding VoIP. libWebRTC’s audio jitter buffer implementation – the one in Chromium – is known as NetEQ. NetEQ is anything but basic. This is good from a user perspective since real-life networks conditions are often challenging. However, this means NetEQ’s esoteric code is complex and difficult […]

The post How WebRTC’s NetEQ Jitter Buffer Provides Smooth Audio appeared first on webrtcHacks.

OpenAI is utilizing WebRTC for its Realtime API! Even better, webrtcHacks friend and Pion founder Sean DuBois helped to develop it and agreed to a Q&A about the implementation. It is not often a massive WebRTC use case like this emerges so rapidly. In addition, Sean was extremely transparent about his work at OpenAI. In […]

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