The VMAF models are actually trained by humans who rated the quality on a 1-5 point scale, watching the movie either on an 1080p screen from 3x screen height away, or on a 4k screen from 1.5x screen height away. This rating is mapped to VMAF scores 0, 20, 40, 60, and 80, while 100 is when they compare the source video to itself. Your score of 94.240835 is ...
h264_nvenc is able to encode in constant quality, at least from Pascal onwards. I had to do quite some digging around the internet to figure out how to properly use it, but finally found a working formula.
A command like below will record in constant quality, and the CQ value 26 will produce about the same but slightly better quality, measured with VMAF, as ...
The concat filter uses the frame rate of the first input as the framerate for the output stream. Some MKV/WebM files don't write valid framerates. That can lead ffmpeg to assign a framerate of 1000 (the inverse of default timebase in Matroska), which can lead to huge frame duplication.
Set a sane output framerate for such scenarios.
ffmpeg -i intro.mkv -i ...
Yes, your observation is correct and everything works fine. Even if settings are identitcal, the file hashes will always be different.
The reason is quite simple: XMP metadata which is embedded in the files.
Such metadata entries which are often different are:
time stamps (not only the date but the time too)
To concatenate multiple files for expected playback in common players, following properties need to match for
video: codec, codec profile, codec level, resolution, reference count, pixel format, timebase/timescale.
audio: codec, codec profile, channel count & layout, sample format and sampling rate.
Advanced players can tolerate mid-stream changes in ...
If a video does not matter on a video... you could use an audio file like mp3.
But let us assume you are uploading it on a video platform like youtube. Then the video format should be accepted by the platform.
If you want to use it on your website, the most commonly used container is MP4, so use that.
But what you want to change are the video parameters. For ...
You have 37 minutes and 14 seconds of audio, so it's going to be at least as large as the original audio file (36Mb) plus one image and a few bytes for each frame saying "the image has not changed". You have some 67,000 frames if you build a 30 fps video.
So it will look larger than the input audio file. It could be that the settings are such that ...
No. Almost all video on the planet is limited YUV420 (all Blu-rays, even HEVC ones with Dolby Vision, all Youtube, only some satellite FEEDs not for consumers use 4:2:2). Sorry. Of course, there is RGB and even mathematically lossless mode in all common encoders, H.264/AV1/VP9/HEVC... For lossless AV1 see https://trac.ffmpeg.org/ticket/7600
But, alas. It is ...
You may check https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/nvenc.h#L186-L221
and the valid can be referenced from https://github.com/FFmpeg/nv-codec-headers/blob/master/include/ffnvcodec/nvEncodeAPI.h
!ffmpeg -y -i http://mirrors.standaloneinstaller.com/video-sample/Panasonic_HDC_TM_700_P_50i.mkv -c:v libx265 -x265-params no-info=1 output.mp4
would generate MediaInfo as
Video ID : 1
Format/Info: High EfficiencyVideo Coding Format
profile : Main@L4@Main
I've come up with a "good enough" solution to this problem, as it doesn't appear possible at the time of writing to actually pick up a partial encode that was cut off arbitrarily and accurately concatenate the rest of the video to it.
A Solution Using Segmenting
First, use FFmpeg's segmenting functionality to output segments at some interval for ...
To minimise the file size for distribution of something like this, you need a Variable Frame Rate encoder, which will effectively just encode 1 frame, and set the frame rate such that there's no further frames. I've not seen this available in Adobe (or any commercial product I've looked into), but definitely there are services I've seen encoding video this ...
About your other question regarding the -coder option, I was also looking for more info on this.
The current (Dec. 2020) IETF/cellar WG draft "FFV1 Video Coding Format Version 0, 1, and 3" mentions in it's "4.2.3. coder_type" chapter :
If coder_type is 0, then bits_per_raw_sample SHOULD NOT be > 8.
At the ...
I've been running some tests lately (2H2020) concerning reducing the size of some high quality movies of my collection, just for PC to TV 1080p viewing, from 20-30mbps videos to 2-3mbps ones. To that end, I run some ffmpeg (latest gyan.dev version at the time) tests comparing same-bitrate encodes between: codecs (x264,x265,VP9,AV1), presets (slow, slower etc)...