标签：#video

Amazon is hoping the addition of games can turn Prime Video into a one-stop entertainment destination, borrowing a strategy from Netflix, which has increasingly embraced party games over the past several years.

I spent a week using video editing Skills to build a video editing Agent. It feels amazing! It can automatically edit a 30-minute video in just 10 minutes. Video editing Agent demo: automatically editing a 30-minute video in 10 minutes. I often use CapCut to edit talking-head videos, but after using it for a long time, I found several problems. Problem 1: Smart talking-head editing does not understand meaning Because it cannot understand the meaning, it sometimes fails to identify repeated sections. If I speak continuously for 20 or 30 minutes, editing the video myself becomes exhausting. Problem 2: The subtitle quality is poor The automatically generated subtitles contain many incorrect words and typos. So I used the Skills feature in Claude Code to build a video editing Agent. The fundamental difference is simple: CapCut vs. Agent: a fixed tool vs. an adaptive assistant. The key difference is: CapCut = fixed tool + manual operation Agent = adaptive system + automatic learning I am not replacing CapCut with a better algorithm. I am replacing it with a system that can continuously improve itself. But that is not even the most impressive part. The most impressive part is this: the more I use it, the better it understands me, and the faster it becomes. Three Core Designs 1. Agent Logic It only takes four steps. Video editing Agent workflow: from the video file to the final video. 2. The Skills System At first, I put every function into one large Skill. I had to add instructions to distinguish between different tasks, which was very inconvenient. Now I have separated the five core video editing tasks into five independent Skills and placed them in the .claude/skills/ directory. This makes the structure clearer and the tasks easier to select. When I enter /v , Claude Code automatically lists the five available Skills. The list of five independent Skills. I select one, and the AI runs that Skill. Simple, right? A manual task that used to take 10 minutes now only requires

Every few weeks I hit the same wall: I have a 300 MB screen recording, and something on the other end wants it under 8 MB . Discord, an email attachment, a bug tracker, a form that silently rejects anything bigger. The usual advice is "just use HandBrake" or "run ffmpeg with a lower CRF." But CRF doesn't take a target size — it takes a quality knob . So you export, check the size, it's 11 MB, nudge the knob, export again, now it's 5 MB and looks like a potato, nudge back… It's a binary search you run by hand, one full encode per guess. The thing is, hitting an exact size isn't a guessing game at all. It's arithmetic you can do before you encode. I ended up wrapping that arithmetic into a little Rust CLI ( DeepShrink ), but the math is the interesting part, and almost nobody writes it down. So here it is. The one insight everything rests on File size is (roughly) bitrate × duration . A bitrate is bits per second. A duration is seconds. Multiply them and the seconds cancel, leaving bits — the size of the file. That's it. That's the whole trick. Normally you treat bitrate as the input and size as whatever falls out. Flip it around: fix the size, measure the duration, and solve for the bitrate. You know the duration (ffprobe will tell you), and you know the size you want (the platform's limit). The only unknown is the bitrate — and now it's a single division away. bitrate = size_in_bits / duration_in_seconds Everything below is just this equation with the real-world messiness added back in. Building the budget, step by step Say I want a 60-second clip to fit Discord's 8 MB limit. 1. Turn the target size into bits. Sizes are in bytes, bitrate is in bits, so multiply by 8. (I'll use 1 MB = 1,000,000 bytes here to keep the mental math clean; if your platform means mebibytes, same method, different constant.) target_bits = 8_000_000 bytes × 8 = 64_000_000 bits (64 Mbit) 2. Reserve a little for container overhead. An .mp4 isn't pure video and audio — there's a container, a m

Google is adding personalized AI avatars to Vids that let users create videos starring a digital version of themselves, alongside Gemini Omni-powered tools for generating and editing videos from prompts and reference images.

Drawing on more than a decade spent helping build some of the world's most influential AI systems, including research that later informed the development of ChatGPT, Andrew Dai explains why he believes visual AI is one of the next major frontiers in artificial intelligence.

Amazing : Researchers from ETH Zurich in Switzerland, however, managed to create a new type of pixel that can simultaneously do both. This hypercharged pixel, called a Fourier pixel, can generate and sense arbitrary light fields and tap into a pixel’s full potential for carrying information by manipulating light’s intensity, oscillation phases, and polarization. The team reported its findings in a paper published yesterday in Nature. We are one step closer to 1984 technology: The telescreen received and transmitted simultaneously. Any sound that Winston made, above the level of a very low whisper, would be picked up by it; moreover, so long as he remained within the field of vision which the metal plaque commanded, he could be seen as well as heard. There was of course no way of knowing whether you were being watched at any given moment...

But with great power comes a great price.

With the cash, the company aims to expand its world model offering and reach customers across geographies.

Why async matters for video I've been running useKnockout - a background removal API that processes images in ~200ms - for a few months. Images are fast enough to handle synchronously: POST a file, wait 200ms, get a PNG back. Video is different. Even a 5-second clip at 30fps is 150 frames. At 200ms per frame, that's 30 seconds of processing. You can't hold an HTTP connection open for 30 seconds and call it a good API. So today I shipped POST /video/remove - async video background removal that returns a job ID immediately, processes in the background, and gives you ProRes 4444 (RGB+alpha) when it's done. What shipped As of v0.11.0 (July 10, 2026): POST /video/remove - upload a video, get a job ID back GET /jobs/{job_id} - poll for status, download the result when ready ProRes 4444 output - RGB with full alpha channel, ready to drop into Premiere/Final Cut/DaVinci Node SDK videoRemove() and getJob() in v0.7.0 Python SDK video_remove() and get_job() in v0.7.0 Billing is a dedicated video.seconds meter at $0.10/sec (different from the per-image rate), with a 15-second cap to keep costs predictable. How to use it (Node SDK) import { useKnockout } from ' useknockout-node ' ; import fs from ' fs ' ; const client = useKnockout ({ apiKey : process . env . KNOCKOUT_API_KEY }); // Submit the video const job = await client . videoRemove ({ file : fs . createReadStream ( ' ./input.mp4 ' ) }); console . log ( ' Job ID: ' , job . id ); // Poll until done let status = await client . getJob ( job . id ); while ( status . status === ' processing ' ) { await new Promise ( resolve => setTimeout ( resolve , 2000 )); status = await client . getJob ( job . id ); } if ( status . status === ' completed ' ) { // Download the ProRes 4444 result const video = await fetch ( status . result_url ); const buffer = await video . arrayBuffer (); fs . writeFileSync ( ' ./output.mov ' , Buffer . from ( buffer )); } The job object includes duration_seconds (billed amount), status ( processing / complet

Image-to-video generation is often described as a simple interaction: upload image -> describe motion -> get video That description hides the real problem. A single still contains only one view of a subject. When we ask a model for a fast camera orbit, a full-body walk, or expressive gestures, we are asking it to invent information that was never present in the source. That is where identity drift, unstable lighting, texture flicker, and waxy faces come from. The useful way to approach Seedance 2.0 image-to-video is not as a prompt-writing contest. It is a constraint-management workflow. Give the model a strong identity anchor, request motion that the source image can support, and evaluate one variable at a time. This post explains that workflow in a way that is useful whether you are animating a product render, a character portrait, an approved client still, or a visual asset for a prototype. Note: Model capabilities, pricing, model availability, and input limits change quickly. Check the current documentation and the terms of the platform you use before committing a production workflow. Why image-to-video is different from text-to-video Text-to-video is excellent when invention is the point. You describe a scene and let the model make creative decisions about characters, lighting, composition, and motion. Image-to-video is the better tool when those decisions have already been made and must remain stable. Situation Better starting mode Why Product hero shot Image-to-video Label, shape, material, and color must remain recognizable Character-led sequence Image-to-video One strong reference can anchor a character across clips Approved campaign still Image-to-video The source already represents the accepted art direction Atmospheric B-roll Text-to-video Exact subject identity matters less than visual exploration Abstract concept film Text-to-video Inventing a scene is more valuable than preserving one Existing brand-photo library Image-to-video Stills become reusable

A surprise 10th-anniversary event saw Niantic fulfilling a promise teased in the original 2016 launch trailer for its popular mobile game.

Provided you have a library of SNES cartridges, the SN Operator is a seamless plug-and-play system for easy ’90s nostalgia.

TL;DR We're building a script that takes a video in English and produces the same video narrated in Spanish, in a cloned version of the original speaker's voice. Stack: faster-whisper for timestamped transcription, an LLM (or any MT engine) for translation, XTTS-v2 for voice-cloned synthesis, FFmpeg for surgery. We'll also handle the problem every demo skips: translated audio that doesn't fit its time slot. 📦 Code: github.com/USER/repo (replace before publishing) If you'd rather start from a finished system, Softcatala's open-dubbing and KrillinAI are full pipelines behind one CLI. This post builds the minimal version by hand so you understand what those tools are doing, and where they break. 0. Setup and a licensing warning ⚠️ Python 3.10–3.12. The original Coqui company shut down in early 2024; the maintained fork of their TTS library is published by Idiap as coqui-tts : $ python -m venv dub && source dub/bin/activate $ pip install faster-whisper coqui-tts $ ffmpeg -version | head -1 # 6.0+ is fine, 8.x current ⚠️ Note: the XTTS-v2 model weights ship under the Coqui Public Model License, which restricts commercial use. Prototype freely, but before dubbed videos ship to paying customers, someone must read that license and possibly swap the synthesis step for a commercially licensed model or paid API. Voice cloning also requires the speaker's consent. Get it in writing. 1. Extract audio and transcribe with word timestamps 🎙️ # pull mono 16k audio for the ASR step $ ffmpeg -i input.mp4 -vn -ac 1 -ar 16000 -y source.wav # dub/transcribe.py from faster_whisper import WhisperModel model = WhisperModel ( " large-v3-turbo " , compute_type = " int8 " ) segments , info = model . transcribe ( " source.wav " , word_timestamps = True ) lines = [] for seg in segments : lines . append ({ " start " : seg . start , " end " : seg . end , " text " : seg . text . strip (), }) print ( f " language= { info . language } segments= { len ( lines ) } " ) The timestamps are the skeleton of

TL;DR FFmpeg 8.x includes av1_vulkan , the first cross-vendor GPU AV1 encoder in mainline FFmpeg. We'll probe whether your GPU + driver actually expose AV1 encode, run a first working encode, benchmark it against SVT-AV1 on your own content, and talk about which jobs deserve it. 📦 Code: github.com/USER/repo (replace before publishing) Until FFmpeg 8.0 ("Huffman", released August 2025), GPU AV1 encoding meant picking a vendor: av1_nvenc for NVIDIA RTX 40+, av1_amf for AMD, av1_qsv for Intel Arc. Three code paths, three sets of flags, three driver stacks. The Vulkan Video encode work gives FFmpeg one encoder that reaches all three vendors through the standard VK_KHR_video_encode_av1 extension. The catch: driver support is a lottery. Plenty of capable hardware sits behind drivers that don't expose the encode extension yet. So before any pipeline decisions, we probe. 1. Check what you're running You want FFmpeg 8.x (8.1.2 is current as of late June 2026) built with Vulkan support, plus the vulkaninfo tool from the Vulkan SDK / vulkan-tools package. $ ffmpeg -version | head -1 ffmpeg version 8.1.2 Copyright ( c ) 2000-2026 the FFmpeg developers $ ffmpeg -hide_banner -encoders | grep vulkan V....D av1_vulkan AV1 ( Vulkan ) ( codec av1 ) If av1_vulkan doesn't appear, your build wasn't compiled with --enable-vulkan (distro packages vary; the BtbN static builds and most 8.x distro packages include it). 2. Probe the driver for AV1 encode 🔍 The encoder existing in FFmpeg means nothing if the driver doesn't expose the extension. This is the step that separates "should work" from "works": $ vulkaninfo | grep -iE "video_encode_(av1|queue)" VK_KHR_video_encode_av1 : extension revision 1 VK_KHR_video_encode_queue : extension revision 12 You see Meaning Both extensions listed You can encode AV1 via Vulkan 🎉 Only video_encode_queue Driver does Vulkan encode, but not AV1 (maybe H.264/H.265 only) Neither Driver too old, or GPU lacks an AV1-capable video engine Rough hardware floor: the

TL;DR We're adding live streaming to a SaaS dashboard: a backend endpoint that creates a stream, OBS as the broadcaster over RTMPS, LL-HLS playback with hls.js, and a webhook handler that keeps the UI honest. Working "go live" flow in an afternoon. 📦 Code: github.com/USER/repo (replace before publishing) Webinars, coaching sessions, company town halls: sooner or later your product gets the "can users go live?" ticket. The hard parts (ingest servers, transcoding, CDN delivery) are exactly the parts you should not build. We'll use FastPix as the managed layer here; the same flow works nearly line-for-line on Mux, Cloudflare Stream, or api.video. What we're building: A backend endpoint that creates a live stream and returns a stream key An OBS setup broadcasters can follow in two minutes A viewer page playing LL-HLS with hls.js A webhook handler that flips the webinar between scheduled → live → ended 1. Create the stream server-side 🛠️ You need API credentials (Access Token ID + Secret Key). FastPix uses Basic auth on the server API. Node 20.x, plain fetch , no SDK required (though official Node.js/Python/Go/Ruby/PHP/Java/C# SDKs exist if you prefer). // server/routes/streams.js import { Router } from " express " ; const router = Router (); const AUTH = " Basic " + Buffer . from ( ` ${ process . env . FP_TOKEN_ID } : ${ process . env . FP_SECRET } ` ). toString ( " base64 " ); router . post ( " /webinars/:id/stream " , async ( req , res ) => { const r = await fetch ( " https://api.fastpix.io/v1/live/streams " , { method : " POST " , headers : { " Content-Type " : " application/json " , Authorization : AUTH }, body : JSON . stringify ({ playbackSettings : { accessPolicy : " public " }, }), }); if ( ! r . ok ) return res . status ( 502 ). json ({ error : " stream create failed " }); const stream = await r . json (); // persist against your webinar row: // streamId, streamKey (SECRET!), playbackId await db . webinar . update ( req . params . id , { streamId : stream . str

Part 1: Solving GOP Structure and Compatibility Issues Operating a digital television network isn't just about keeping channels on air—it's about maintaining quality that viewers expect and troubleshooting issues before they escalate. When something goes wrong in live broadcasting, every second counts. But how do you quickly pinpoint whether the problem lies in encoder settings, transport stream structure, or temporal metadata? This is where specialized stream analysis tools become essential. In this series of articles, we'll walk through real-world scenarios that broadcast engineers face daily and show practical approaches to diagnosing and resolving them. When File Analysis Becomes Critical While live monitoring catches issues as they happen, file-based analysis is your diagnostic microscope. Here's the typical workflow: something breaks in production, engineers capture a few minutes of the problematic stream, and now they need to understand exactly what went wrong. File analyzers serve three primary purposes: Troubleshooting: Identifying the root cause of broadcast issues Encoder optimization: Fine-tuning compression settings Quality control: Validating compliance with standards and specifications Let's explore how this works in practice with actual tools and techniques. The GOP Structure Problem Here's a scenario every broadcast engineer has encountered: legacy set-top boxes or older TV models suddenly can't play your stream. The audio works, video starts and stops, or you see freezing. The culprit? Often, it's the GOP (Group of Pictures) structure. H.264 has been around since 2003—over 20 years. Almost everything supports it, yet you'll still find legacy equipment that struggles with certain configurations. Specifically, the number of B-frames can make or break compatibility. Why B-frames matter: They enable lower bitrates while maintaining quality by increasing encoding complexity through bidirectional prediction. But this comes at a cost—a more complex refere

📦 Code: github.com/USER/hls-multi-audio - replace before publishing TL;DR We'll add a working language picker to an HLS player. The hard part isn't the dropdown, it's the manifest. We'll author alternate audio with EXT-X-MEDIA audio groups, package it correctly, debug the classic "zero audio tracks" bug, and wire a switcher on hls.js v1.7 . Adaptive video, captions, the whole pipeline already works. Now someone wants an English/Spanish audio toggle. In HLS, "which audio can the viewer pick" is decided at packaging time and written into the master playlist. The player just displays it. Let's build it in that order. 1. Understand the structure (audio groups) HLS decouples video variants from audio renditions: Each audio rendition is an #EXT-X-MEDIA:TYPE=AUDIO entry pointing at its own media playlist. Renditions are bundled into a named audio group via GROUP-ID . Each video variant ( #EXT-X-STREAM-INF ) references a group with AUDIO="..." . A correct master playlist: #EXTM3U #EXT-X-VERSION:6 #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="aud",NAME="English",LANGUAGE="en",DEFAULT=YES,AUTOSELECT=YES,CHANNELS="2",URI="audio/en.m3u8" #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="aud",NAME="Espanol",LANGUAGE="es",DEFAULT=NO,AUTOSELECT=YES,CHANNELS="2",URI="audio/es.m3u8" #EXT-X-STREAM-INF:BANDWIDTH=2128000,CODECS="avc1.640028,mp4a.40.2",AUDIO="aud" video/720p.m3u8 #EXT-X-STREAM-INF:BANDWIDTH=1128000,CODECS="avc1.640020,mp4a.40.2",AUDIO="aud" video/480p.m3u8 Every attribute earns its place: LANGUAGE - BCP-47 code, used for the label. DEFAULT - plays when the viewer has no preference. AUTOSELECT - may be auto-picked from the OS language. CHANNELS - needed so the player can reason about stereo vs surround. BANDWIDTH on each video variant must include the audio group's bitrate , or your ABR logic works from a wrong total. 2. Author the renditions with FFmpeg Extract/encode each language's audio, then package. First, encode video-only and audio-only renditions: # video only (no audio), two ladder rungs

📦 Code: github.com/USER/nvenc-vs-cpu-bench - replace before publishing TL;DR A GPU encodes faster than a CPU, but "faster" and "cheaper" are different claims. We'll build a small FFmpeg + VMAF harness that times software (libx264/SVT-AV1) against hardware (h264_nvenc/av1_nvenc), then plug the results into a dollars-per-encoded-minute formula so you find your break-even instead of trusting a benchmark blog. We're using FFmpeg 7.1.x (current stable line) and an NVIDIA GPU with NVENC. Same approach works for Intel QSV ( *_qsv ) and AMD AMF ( *_amf ) if you swap the encoder names. Why this isn't obvious NVENC is a fixed-function hardware block, not "the GPU doing x264 in parallel." It's extremely fast and barely touches the CPU, but it exposes fewer rate-control knobs and gives up a little compression efficiency versus a slow software preset. The gap has narrowed a lot, but it's still there at the quality-obsessed end. So the decision is per-job, and it comes down to one number: dollars per encoded minute = (instance $/hr) ÷ (minutes encoded/hr) . GPU instances cost more per hour but encode many streams in parallel, so the answer depends on whether you can keep the encoder saturated. Let's measure instead of argue. 1. Set up the encoders Three contenders. One representative source file (use real footage, not a synthetic clip). # software H.264, quality-leaning preset ffmpeg -y -i source.mp4 -c :v libx264 -preset slow -crf 21 -an out_cpu.mp4 # NVENC H.264, quality-tuned ffmpeg -y -hwaccel cuda -i source.mp4 -c :v h264_nvenc -preset p6 -tune hq \ -rc vbr -cq 23 -an out_gpu.mp4 # AV1: software (SVT-AV1) vs hardware (needs Ada / RTX 40+) ffmpeg -y -i source.mp4 -c :v libsvtav1 -preset 6 -crf 30 -an out_svtav1.mp4 ffmpeg -y -hwaccel cuda -i source.mp4 -c :v av1_nvenc -preset p5 -cq 30 -an out_av1nvenc.mp4 💡 Tip: -preset p1 (fastest) through -preset p7 (slowest/highest quality) for NVENC. p6 / p7 is where it competes on quality; p1 - p3 is where it competes on raw throughput.

📦 Code: github.com/USER/video-api-bench - replace before publishing TL;DR The per-minute delivery rate is the easiest number to compare and the least useful. The real cost lives in encoding, analytics, and the player. This post compares Mux, Cloudflare Stream, api.video, FastPix, and AWS on what each includes by default, then gives you a tiny script to benchmark upload and time-to-ready on your own files so you stop trusting marketing pages. I have shipped video on four managed APIs across three jobs, and every single time the invoice surprised someone. Not because the delivery rate was wrong, but because encoding, analytics, and the player turned out to be separate line items on some platforms and free on others. Let's compare the parts that don't show up in the headline number. ⚠️ Note: pricing pages move. Everything here was checked in June 2026; verify the links before quoting numbers. 1. Encoding: free or metered? This is the widest spread in the whole comparison. Platform Encoding Delivery Storage Cloudflare Stream Free $1 / 1,000 min delivered $5 / 1,000 min stored api.video Free (unlimited) $0.0017 / min $0.00285 / min FastPix Free on standard plan ~$0.00096 / min @1080p Per-minute, tiered Mux Metered per minute Per minute Per minute AWS (DIY) Per minute (MediaConvert) Per GB (CloudFront) Per GB (S3) If your catalog is upload-heavy (lots of assets encoded once, watched rarely), metered encoding is not a rounding error. It can flip which platform is cheapest, even when the delivery rates look identical. 2. Analytics: included or a $499 floor? QoE analytics is the feature teams forget to price until playback breaks in production. Platform QoE analytics Entry cost FastPix (Video Data) Session-level, 50+ signals/session Free up to 100K views/month Mux (Mux Data) Mature, broad device SDKs $499/month (Media plan, 1M views, +$0.50/1K) Cloudflare Stream Basic Included, limited depth api.video Available Usage-based AWS Build it yourself (CloudWatch + logs) Engineerin

TL;DR If your product lets strangers upload video, you need moderation before launch, not after the first bad upload. We will build a small-team pipeline: extract frames with FFmpeg, score them with NudeNet (ONNX Runtime, CPU-friendly), route uploads into approve / human-review / block by confidence, and log every decision. No trust-and-safety department required. 📦 Code: github.com/USER/ugc-moderation, replace before publishing ⚠️ Note: this is a sensitive area. The goal here is the engineering shape (sampling, scoring, routing, auditing), not detection of any specific content. Keep test fixtures clean and lawful. A model does not decide what is allowed. It produces a score. You decide where the lines go. The whole design is about routing scores sensibly and sending the uncertain middle to a human. The architecture 🧠 upload ──> extract sample frames (ffmpeg) ──> score frames (NudeNet / ONNX) ──> aggregate to one confidence ──> route: high-confidence clean -> auto-approve uncertain middle band -> human review queue high-confidence violation -> auto-block ──> write an audit record for every decision The economics only work if the middle band is small. A decent model makes most uploads obviously fine or obviously not, so a human only ever sees the genuinely ambiguous slice. 1. Sample frames, do not score every frame You cannot afford every frame and you do not need it. Pull one frame per second (or scene-change keyframes) with FFmpeg. # extract 1 frame per second into ./frames mkdir -p frames ffmpeg -i upload.mp4 -vf "fps=1" -q :v 3 frames/frame_%05d.jpg Prefer scene changes to catch more variety with fewer frames: # keyframes where the scene actually changes ffmpeg -i upload.mp4 -vf "select='gt(scene,0.3)',showinfo" -vsync vfr frames/scene_%05d.jpg 💡 Tip: a 30-minute upload at 1 fps is ~1,800 frames. Scene-change sampling often cuts that by an order of magnitude with little loss for moderation purposes. 2. Score frames with NudeNet NudeNet runs on ONNX Runtime on pla