It only takes a minute to sign up. What is the default output frame rate chosen by ffmpeg to encode MP4s? Where is this specified on the man page? After demuxing, decoding splitting rescaling, encoding and muxing the video stream I end up with has the following stats:. I started with frames and ended up with should have been The input average fps was So, I imagine ffmpeg picks the highest one from the input How does one go simply matching the input fps?
And why does I have a variable fps to begin with? It will then duplicate or drop frames to keep that rate. Use -vsync vfr to keep the variable rate.
Sign up to join this community. The best answers are voted up and rise to the top. Home Questions Tags Users Unanswered. Asked 3 years, 1 month ago. Active 3 years, 1 month ago. Viewed 5k times. Still, is this simply a consequence of this paragraph? It picks the "best" of each based upon the following criteria: for video, it is the stream with the highest resolution, for audio, it is the stream with the most channels, for subtitles, it is the first subtitle stream.
In the case where several streams of the same type rate equally, the stream with the lowest index is chosen. Atcold Atcold 1 1 silver badge 5 5 bronze badges. Active Oldest Votes. Gyan Gyan You're impressive. The first line comes from the FAQ. Any idea where this is mentioned in the man page? Yes, I'm encoding in MP4. Where did you get that info about the constant frame rate picking? My client expect temporal constancy between the frames.
But in ubuntu I find it very difficult to find out this basic information. Any help is appreciated. Someone edited with one that didn't quite work the way I wanted. It's referenced here Additional edit If you want the tbr value this sed line works. Here is a python function based on Steven Penny's answer using ffprobe that gives exact frame rate.
The alternative to command line, is looking at the properties of your file via context menu in Nautilus graphical file manager. This is a python script to do this using mplayer, in case anyone is interested. To get a precise framerate you can install MediaInfo. I usually use exiftool to get info of any file type For example with command exiftool file. Ubuntu Community Ask! Sign up to join this community. The best answers are voted up and rise to the top.
Home Questions Tags Users Unanswered. How to find frames per second of any video file? Ask Question. Asked 8 years, 1 month ago. Active 17 days ago. Viewed 92k times. Vivek Vivek 2, 10 10 gold badges 21 21 silver badges 30 30 bronze badges.
This is not possible, because not all video files have a "fps" because VFR encoding exists. VFR videos still have an average frame rate - whether or not this is useful depends on the application. Active Oldest Votes. This will tell you the framerate if it's not variable framerate: ffmpeg -i filename Sample output with filename obscured: Input 0, matroska,webm, from 'somerandom.PREMIERE PRO FINALLY SUPPORTS VARIABLE FRAME RATE (VFR) - CC 2018 12.0.1 Update Details
RobotHumans RobotHumans I needed to use tb instead of fp in the one-liner. Seems not all video files report fps but all autput something like tbr tbc which has the same value. I'ld rather it fail in a really noticeable way than a way that isn't noticed at all. What if it is variable frame rate?
It only takes a minute to sign up. I have a video in mp4 format with a frame rate of. I have tried the below command but it does not do any thing:. As noted in the comments there is a way to do this where the video does not have to be re-encoded. It requires remuxing the file to a different containter format MKV and then remuxing it back into an MP4.
If the video contains audio you can also slow that down without changing the pitch, but it is not a lossless conversion. This is the guidance from the ffmpeg wiki. Note that all of these options do require re-encoding the video.
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Note that in the following examples, the audio stream is not changed, so it should ideally be disabled with -an. The filter works by changing the presentation timestamp PTS of each video frame. For example, if there are two succesive frames shown at timestamps 1 and 2, and you want to speed up the video, those timestamps need to become 0.
Thus, we have to multiply them by 0. Note that this method will drop frames to achieve the desired speed. You can avoid dropped frames by specifying a higher output frame rate than the input. This is also known as "motion interpolation" or "optical flow".
The atempo filter is limited to using values between 0. If you need to, you can get around this limitation by stringing multiple atempo filters together.
The following with quadruple the audio speed:. I could only get the changed framerate to take effect if the input file was classed as a "raw" file:.
Without specifying -f h it would default to 25 fps and it could not be changed. Apparently this was because the stream lacked any framerate information at all and this is ffmpeg's default framerate. Apparently when you use -r as an output option it duplicates or drops frames so the video plays at the same speed - in this case, not what you want!
But changing the input framerate as above will cause the video to speed up or slow down, with no frames lost or duplicated. Such a feature - of changing framerate - is called "conforming" and is often used to produce slow-motion or fast-forward like showing a plant's growth in minutes insted of days. If it does so, duration will change and audio would be out of sync unless separately mended.
But I'm afraid audio is not of interest in your case with framerate of.
You want conforming because you just want to change framerate, but ffmpeg ignores -r silently if framerate is specified in the input file. Since your input file is in. For this you need a different tool: mencoder. Assuming your input file contains no sound - likely true for any video with.
Stack Overflow for Teams is a private, secure spot for you and your coworkers to find and share information. I'd like to use ffmpeg, mencoder, or some other command-line video transcoder to re-sample this video to a lower framerate without loss of image quality. That is, each frame should remain as crisp as possible. This seems like a simple enough use case. I'm very surprised that obvious things are not working. Is there something wrong with my approach? A lot has changed since this posting in I am adding this answer for people like me who find this from the search engines.
I had good luck with the following:. For more details see the official documentation. As Andy T indicated, you absolutely must re-encode the video, but that does not mean that the quality must be reduced in any noticeable manner. First, ensure you are not using old software.
Changing the frame rate
Video codecs are a fast moving field with significant advanced every few weeks. It's probably what mplayer uses, but you can get the most recent real release from www.
As I said, these are fast-moving fields! First, I would use VirtualDub to decompress to a lossless video type, which will make for a pretty large file.
VirtualDub can also reduce the frame rate. See here. Next, use this as the input to x If you want the absolute best quality with smallest file size where you are unlikely to notice the difference, use a command line like this with x :.
A quick guide to using FFmpeg to convert media files | Opensource.com
Staxrip has options switch the preset to placebo, tune for film sources assuming this isn't an animation of courseand to reduce thread count to 1. You can change "--crf 22" to 21 for a larger, better quality video, but I've found CRF22 to be about the point where I have a very hard time noticing the difference, even when comparing frame-by-frame.
Any change in the the other settings will probably reduce quality or increase file size without boost in quality. Most companies that encode video or that make products that encode video really have no idea what they are doing, and even if they did, they don't have the CPU power needed to do an encode with these settings, so in addition to the modest reduction in file size by frame rate reduction, you will also get a sharp drop in file size from using a smart encoder with the strictest settings.
Lossless (video) remuxing
If you want to make absolutely certain that the resulting video looks as close to VirtualDub's output as possible, no one can tell the difference using CRF17, but the file will be pretty large.
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Change your preferences any time. Stack Overflow for Teams is a private, secure spot for you and your coworkers to find and share information. I am trying to convert a video clip MP4, yuvp from 30 fps to 24 fps. The number of frames is correct so my output should change from 20 minutes at 30fps to 25 minutes at 24fps. Everything else should remain the same. Try as I might everything I try with ffmpeg converts the frame rate but changes the number of frames to keep the same duration or changes the duration without altering the framerate.
I'm doing this on windows but normally would be on linux. That converts the framerate but drops frames so the total duration is unaltered. Surely I should be able to do this with a single ffmpeg command without having to reencode or even as some people suggested going back to the original raw frames. No recompression is necessary. There was a small utility avifrate. To the best of my knowledge you can't do this with ffmpeg without re-encoding.
I had a 24fps file I wanted at 25fps to match some other material I was working with. I used the command ffmpeg -i inputfile -r 25 outputfile which worked perfectly with a webm,matroska input and resulted in an h, matroska output utilizing encoder: Lavc You can accomplish the same thing at 6fps but as you noted the duration will not change which in most cases is a good thing as otherwise you will lose audio sync. If this doesn't fit your requirements I suggest that you try this answer although my experience has been that it still re-encodes the output file.
For the best frame accuracy you are still better off decoding to raw streams as previously suggested. I use a script for this as reproduced below:. Clearly this script expects all files in the current directory to be media files but can easily be changed to restrict processing to a specific extension of your choosing.
Be aware that your file size will increase by a rather large factor when you decompress into raw streams. Learn more.The output duration of the video will stay the same.
This is useful when working with, for example, high-framerate input video that needs to be temporally scaled down for devices that do not support high FPS. When the frame rate is changed, ffmpeg will drop or duplicate frames as necessary to achieve the targeted output frame rate. Note: Changing frame rates requires the video to be re-encoded. Without setting appropriate output quality or bit rate, the video quality may be degraded.
Please look at the respective encoding guides for the codec you've chosen. In order to verify which frames are duplicated or dropped by a frame rate change, you can first generate a sample video:. This 10 second, 60 fps video called test. By playing the output in a player that allows seeking frame-by-frame, you can inspect which frames have been dropped or duplicated. Powered by Trac 1. Changing the frame rate Contents How to change the frame rate Verifying frame rate changes.
Last modified 18 months ago Last modified on Oct 13,PM. Download in other formats: Plain Text.Jump to navigation. Tools like Audacity or Handbrake are fantastic, but sometimes you just want to change a file from one format into another quickly. Enter FFmpeg. FFmpeg is a collection of different projects for handling multimedia files. It's often used behind the scenes in many other media-related projects.
Despite its name, it has nothing to do with the Moving Picture Experts Group or the myriad multimedia formats it has created. In this article I'll be using FFmpeg through the command-line tool ffmpegwhich is only a single, small piece of the FFmpeg project. It's available on many different operating systems and is included in some operating systems by default.
It can be downloaded from the FFmpeg website or through most package managers. FFmpeg is a powerful tool that can do almost anything you can imagine with multimedia files. In this article, we are interested in using it to convert files, so we won't be taking a deep dive into its entire feature set.
At a very high-level view, a media file is broken up into a container and its streams. The streams include the actual AV components, such as a movie's audio or video, and are encoded using a particular media encoding, or codec.
Each codec has its own properties, strengths, and weaknesses. For example, the FLAC codec is good for high-quality lossless audio, whereas Vorbis is designed to compete with MP3 in file size while offering better audio quality.
This means a FLAC-formatted file will be much larger than a Vorbis audio stream but should sound better. Neither is inherently better than the other, as each is trying to do different things. The container is the wrapper for the streams.
Some containers are highly advanced and allow for any sort of stream, including multiple video and audio streams inside a single container. The streams in a container don't have to be just audio or video though.
It all depends on what the container is set to allow. This is an abstract representation of media files and skips over a lot of the differences between containers. Many require certain streams and metadata or put restrictions on the codecs or contents allowed. This explanation is enough to get you through this article. To learn more, click on the links above. Be aware that video and audio encoding can take a very long time to run. You should be prepared to settle in for a while when you use FFmpeg.
The thing that trips up most people when it comes to converting audio and video is selecting the correct formats and containers. Luckily, FFmpeg is pretty clever with its default settings. Usually it automatically selects the correct codecs and container without any complex configuration. This command takes an MP3 file called input. You didn't have to specify stream or container types, because FFmpeg figured it out for you.
Because WebM is a well-defined format, FFmpeg automatically knows what video and audio it can support and will convert the streams to be a valid WebM file. Depending on your container of choice, this won't always work. For instance, containers like Matroska are designed to handle almost any stream you care to put in them, whether they're valid or not. This means the command:.