标签：#ffmpeg

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

Originally published at ffmpeg-micro.com If you've tried running FFmpeg inside a Pipedream workflow, you've probably hit one of two walls: the step timed out before processing finished, or the FFmpeg binary wasn't available. These are the most common complaints in Pipedream community threads, and neither has a clean workaround. Why FFmpeg Breaks in Pipedream Pipedream workflows run Node.js steps with a 30-second default execution timeout . Paid plans extend that to 300 seconds. But even five minutes isn't enough to transcode most videos. A 10-minute 1080p file can take 3-8 minutes to process depending on the codec and output settings. Longer videos or higher-quality encodes blow past that limit every time. The timeout kills your step mid-execution. No partial output. No graceful failure. Just a dead workflow. Then there's the binary problem. FFmpeg isn't available in Pipedream's runtime environment. Developers on the Pipedream community forums have tried downloading the static binary at runtime, setting PATH variables, and running chmod inside a Node.js step. Some of these hacks work intermittently. Most break the next time Pipedream updates its execution environment. And even if you solve both problems, Pipedream steps have memory constraints that make video processing unreliable. A single high-resolution transcode can exhaust available RAM and crash silently. The Fix: Call an FFmpeg API Instead The timeout issue goes away when you stop running FFmpeg inside the workflow. Make an HTTP request to an external API instead. The API processes the video on its own infrastructure with no time limit. Your Pipedream step sends the request, gets back a job ID, and moves on. FFmpeg Micro processes video through a standard REST API, so any Pipedream HTTP step can call it. No marketplace plugin to install. No binary to configure. Just a POST request and a polling loop. This is different from tools like Rendi or Renderio.dev that require a native Pipedream marketplace integratio

Originally published at ffmpeg-micro.com You need server-side video processing in your Swift app. Maybe you're building a Vapor backend that transcodes user uploads, a macOS utility that batch-converts media files, or a command-line tool that generates thumbnails. FFmpeg is the standard tool for the job, but getting it into a Swift project isn't as simple as adding a package dependency. Running FFmpeg from Swift with Process Swift's Foundation framework provides the Process class for running external commands. If FFmpeg is installed on the machine, you can shell out to it directly: import Foundation let process = Process () process . executableURL = URL ( fileURLWithPath : "/opt/homebrew/bin/ffmpeg" ) process . arguments = [ "-i" , "input.mp4" , "-c:v" , "libx264" , "-crf" , "23" , "-preset" , "medium" , "-c:a" , "aac" , "-b:a" , "128k" , "output.mp4" ] let pipe = Pipe () process . standardOutput = pipe process . standardError = pipe try process . run () process . waitUntilExit () let data = pipe . fileHandleForReading . readDataToEndOfFile () let output = String ( data : data , encoding : . utf8 ) ?? "" print ( output ) guard process . terminationStatus == 0 else { fatalError ( "FFmpeg failed with exit code \( process . terminationStatus ) " ) } This works on macOS and Linux. Install FFmpeg with brew install ffmpeg on macOS or apt-get install ffmpeg on Ubuntu, point executableURL at the binary, and you're running. But you own that FFmpeg install on every machine. On Linux servers, you're managing the binary across deploys. On macOS CI runners, you're adding Homebrew steps to your build pipeline. And on iOS, Process doesn't exist at all. Processing Video via Cloud API (No FFmpeg Install) Skip the local binary entirely. FFmpeg Micro exposes full FFmpeg capabilities through a REST API. Send a video URL, pick your settings, get processed video back. If you're familiar with how this works in Node.js or Kotlin , the pattern is identical. Here's the basic flow using URLSe

📦 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.

TL;DR Converting HDR10 to SDR with a naive FFmpeg command gives you grey, washed-out video. The fix is tone mapping. We will detect HDR with ffprobe , run two working tone-map chains ( zscale on CPU, libplacebo on GPU) in FFmpeg 8.0, compare operators, and batch it. Test the commands on your own build before shipping. 📦 Code: github.com/USER/hdr-to-sdr, replace before publishing If you have ever run an HDR clip through your normal pipeline and gotten back something flat and foggy, this post is for you. The bug is that HDR and SDR are different color systems, and "just converting" reinterprets one as the other. We will use FFmpeg 8.0 "Huffman" (8.0.2 is current as of May 2026). Why naive conversion fails HDR10 SDR Transfer function PQ (SMPTE ST 2084) gamma 2.4 / BT.1886 Color primaries Rec.2020 (wide) BT.709 (narrow) Peak luminance ~1,000 to 4,000 nits ~100 nits A command that ends in -pix_fmt yuv420p with no tone mapping reads PQ-encoded, Rec.2020 values as if they were SDR. The gamut gets crushed with no intelligence and the brightness curve is misread. Hence the fog. 1. Detect whether a file is even HDR 🔍 Do not tone-map SDR files. Check first: # detect transfer characteristics and primaries ffprobe -v error -select_streams v:0 \ -show_entries stream = color_transfer,color_primaries,color_space \ -of default = noprint_wrappers = 1 input.mkv HDR10 content reports something like: color_space = bt2020nc color_transfer = smpte2084 color_primaries = bt2020 If color_transfer is smpte2084 (PQ) or arib-std-b67 (HLG), you have HDR and you need to tone-map. If it says bt709 , leave it alone. 2. The libplacebo path (GPU, my default) 🚀 libplacebo is the Vulkan-accelerated filter in FFmpeg 8.0. It follows the ITU tone-mapping recommendations and handles the color conversions internally, so the command is short: ffmpeg -i input.mkv \ -vf "libplacebo=tonemapping=bt.2390:colorspace=bt709:color_primaries=bt709:color_trc=bt709:format=yuv420p" \ -c :v libx264 -crf 20 -c :a copy \ ou

This is a sanitized engineering note about server-side audio noise reduction for WebRTC calls. Source article: https://www.lodan.me/posts/server-side-webrtc-noise-reduction-pion-ffmpeg-rnn/ What the prototype tests The goal is not to replace WebRTC's built-in audio processing. The narrower test is: receive a WebRTC Opus track with Pion read RTP packets in OnTrack decode Opus payloads to PCM pipe raw PCM into FFmpeg apply the arnndn RNN noise reduction filter validate the output as a file before considering real-time forwarding Why this boundary matters RTP, Opus, PCM, and FFmpeg raw audio input are different boundaries. If the PCM format is wrong, FFmpeg may still produce a file, but the result should not be trusted. For example, if the Go side writes int16 PCM, the FFmpeg input format should be reviewed as s16le , not casually treated as s32le . Production concerns The prototype is useful because it isolates the audio path, but production use needs more work: buffering and latency CPU and memory isolation FFmpeg process lifecycle model choice packet loss and jitter RTP timestamps audio/video sync whether the processed audio is returned to WebRTC or only recorded The full article has diagrams and the longer explanation: https://www.lodan.me/posts/server-side-webrtc-noise-reduction-pion-ffmpeg-rnn/