It sounds like the problem is probably what is known as dropped frames.  There is unfortunately not any particularly good fix.  With dropped frames, information is actually completely missing for the frames where there was no update, thus there is nothing to play during those frames.

Depending on the length of time that is dropped, it may be possible to interpolate frames, but this is not a perfect process and will generally produce substandard results.

Basically, the idea with interpolation is that the computer tries to guess what things are the same in the frame before and the frame after the missing information and then invents frames that go in-between, but since it doesn't know what objects are, it has to make a lot of guesses and motion isn't always linear, so if it is more than a frame or two, it won't be particularly good results.

Generally, if possible, the recommendation is re-shoot rather than try to interpolate, but you can interpolate if you don't have the ability to re-shoot.  This problem also generally occurs when you have insufficient write speed to stay ahead of the video, so you may want to consider a faster memory card for future video so that you can avoid dropped frames in the future.

This question over on superuser talks about doing so with some free tools.


How to calculate the length of the video file that was encoded so far?

Short answer is No.

Longer answer is, it depends.

If you're encoding a file, then generally the output is the duration of the input, unless there's speed change or trim filters or -ss, -to, -t options applied. For a live input, FFmpeg will stop the encode when it encounters EOF on the input, so unless you know that, you won't know the output duration. For multiple inputs of differing duration, it will depend on what the command is e.g. is there a -shortest in there, is there an amix, which defaults to the longest of the inputs, or amerge which terminates upon the shortest input.

By default, the ffmpeg console readout will display the progress of the encoding in realtime i.e. the duration encoded, so far.


FFmpeg add logo watermark to live stream

Use the overlay video filter:

find "/video path" -name "*.mp4" | xargs -I $ "/ffmpeg path" -y -re -i $ -i watermark.png -filter_complex overlay -c:a copy -f flv rtmp://xxx.xxx.xx/live/

You can't filter video (-filter_complex or -vf) and stream copy (-vcodec copy) the same stream. If you try you'll get an error:

  Streamcopy requested for output stream 0:0, which is fed from a
  complex filtergraph. Filtering and streamcopy cannot be used together

So remove -vcodec copy when filtering the video.


VHS tape has one "cog" blocked off - can't play it

So, I posted these images to Reddit under /r/whatisthisthing, and the answer that came back was obvious once it was offered:

Someone has taken a normal tape apart and re-assembled it with the take up spool upside-down.
It can't be a cleaning tape because it will not engage with the take up spool drive. A machine might just spit it back out but it could slam into the drive mechanism and cause some damage.

You will have to take the cartridge apart and carefully flip the spool over. No telling what other nonsense might be going on inside that cassette, however. So standard YMMV disclaimer applies.


Does rackmount hardware attach only on the faceplate?

Yes, almost all rackmount devices are designed to bear their loads on the front plate. Only in very rare circumstances have I seen a piece of rack gear that has not remained rigid while mounted in a rack, and that was only after years of being in one. There are also some rack pieces that have a second set of mounts in the back, but I haven't seen that in a long time.

My Crown power amp, which has got to be 50+ pounds, is only front mounted, as is my Akai S5000, which is about as unwieldy as it gets. You should be fine. Make sure you use four rack screws, and be careful not to overtighten them. You may also want to use rubber or nylon washers, but that's a whole other topic in itself.


Benefits of component/RGB connector (vs. S-video or composite) for VHS capture?

Component Y/Pb/Pr is better than both S-video and composite.  It supports higher resolution, and progressive scan (480p,720p,1080p).  The Y is luminance and sync signaling, the Pb is blue minus, and the Pr is red minus.  Since its progressive, it uses more bandwidth than the former's.
It's easy to get "component" mixed up for "composite"/"rca".  Component is the Red/Blue/Green cables, composite (or RCA) is the Yellow/Red/White cables.
The S-video and composite were made for standard definition video, which is lower resolution interlaced, as opposed to progressive.  interlaced video has less bandwidth needs.
Normally you would use Component when you have it available over composite and s-video alternatives, but don't yet have HDMI capabilities.
If you can use HDMI, it is better than component, since HDMI is digital, component is not.
VCR's
The goodwill I found is one good source for VCR's.  The Time-sync ones, which you need, are about $400.  Too expensive, when you can get a scaler that has the same frame correction features in it on evil bay for cheaper.  I got one used for $45 (Extron System 7SC, uses Composite, up to 1080p).  The VCR isn't as important as its head is in good alignment (take the cover off and adjust it if not) and VCR has more than a few decent heads on it.
Logically, if you can snatch a good deal on a VCR that has component, then make sure your scaler (or cap card if not using one) has component input.  Scaler's with HDMI aren't really useful for VCR capture IMO.   Im also not sure the component ones are in fact either, since its possible that somehow they have upconverted the signal to make it component, which you can just eliminate, and run it straight into your scaler with svideo or composite.
Out of the scaler it can be scaled up 720p/1080p for raw capture.  So if your capture card has component in, then make sure the scaler has component out.  I'm still a bit yellow in the area if the RGBHV on the back of my scaler can be converted to Y/Pb/Pr easily or not, im not sure.  Im testing instead using composite out of scaler, to composite in to USB capture.  The results are maybe more low light detail than the S-video signal, so it has its benefits.
Scaler
Look for one that takes care of the frame corrections for you, so that you wont have sync issues, when you capture (VirtualDub/HuffYUV I assume?)

https://www.extron.com/article/syst7sc32pulldown
I see this one on evilbay now for $50, https://www.extron.com/product/vsc200 has an S-video output also, which i needed, for my capture card to test over composite, although I'm not sure capturing from S-video is all its chalked up to me, for this purpose.  (not sure about the outputs on this one though)
This one looks decent, has svideo https://www.ebay.com/itm/Extron-IN1508-Video-Converter-Scaling-Switcher/124668934942


How to assign roto to another layer in After Effects

Put a layer underneath it and set the lower layer's "TrakMat" to "Alpha Matte".  If you don't see the "TrkMat" dropdown menu to the right of the source name, click the "Toggle Switches/Modes" button at the bottom of the screen.


How to achieve this "trail effect" - After Effects

A quick and dirty solution would be to try applying a fast blur effect over the particle layer to the point they look like smoke.


use ffmpeg to convert interleaved raw yuv/pcm to compressed

Here's a method that should, in principle, work. It assumes VAVA interleaving and for somewhat efficient processing requires the packet sizes for both video and audio having a reasonably large greatest common divisor.

The examples below assume a video packet size of 1382400 bytes and audio of 7680 bytes. As it happens, the former is exactly 180 times the latter. The idea is we can ingest the raw stream as a raw "audio" stream and deinterleave using a selection expression, which routes 180 "frames" to one output and the 181st frame to another. Then the separated outputs are dumped into regular or fifo files which can then be read by another ffmpeg process with a sane interpretation of the input.

Command 1:

ffmpeg -f u8 -channels 1 -i raw-input-url -filter_complex "asetnsamples=7680,aselect='1+not(mod(n+1,181))':n=2[vid][aud]" -map [vid] -f u8 file-vid -map [aud] -f u8 file-aud

Command 1 with input offset:

ffmpeg -f u8 -channels 1 -i "subfile,,start,3248615,end,0,,:raw-input-url" -filter_complex "asetnsamples=7680,aselect='1+not(mod(n+1,181))':n=2[vid][aud]" -map [vid] -f u8 file-vid -map [aud] -f u8 file-aud

where 3248615 is in bytes and should point to the start of a video frame for this to work.

Command 2:

ffmpeg -f rawvideo -pixel_format yuv420p -framerate F -video_size WxH -i file-vid -f f32le -channels N -sample_rate R -i file-aud { -encoding parameters } output


If it sounds good, don't change it. You could try to change it to 44.1kHz-16bit and see if it's really that off from a CD. Otherwise leave it as it is. The whole loudness war is lame anyway. http://productionadvice.co.uk/dynamic-range-day/


FFmpeg with OpenCL support - Ubuntu 18.04

OpenCL is not what you need.

In https://trac.ffmpeg.org/wiki/HWAccelIntro, see the section on “FFmpeg API Implementation Status” — Notice that OpenCL is not applicable for encoding and decoding, only for use in filters.

That's because OpenCL is a low level API used for writing code in such a way that it can be accelerated by GPUs or CPU SIMD instructions depending on the platform/driver. It is used in FFmpeg to implement various filters, but you aren't using any filters. If you really want to use OpenCL for encoding/decoding, you will have to start writing your own encoder/decoder first.

For your use case (encoding a video from source frames), you are better off using an API specifically designed for video encoding. These use the separate special purpose video encoding/decoding functionality found on most modern GPUs (including integrated GPUs). The best choice depends on your platform and use case, but for example this page has some examples for using VAAPI: https://trac.ffmpeg.org/wiki/Hardware/VAAPI. Usually this means selecting a different encoder, for example h264_vaapi instead of x264 (which is a purely CPU-based encoder). To ensure good performance, avoid copying frames between host/CPU memory and device/GPU memory as much as possible.


Reveal entire layer in after effects with mask partial overlap

So, if we will continue analogy with Lego blocks:

First you need add On & Off effect for your blocks. You can find here how.
Then you shall add Collision detection. You can find about it here.
Now you need to trigger On & Off effect when your object (mask?) collide with block.

I never tried it myself, but now really want to test it!


How does TV video quality remain same on both small and large TVs?

The quality does differ. If both TVs have the same resolution and specs, the image on the bigger one will be more pixelated if viewed from the same distance. The last part of the sentence is the key. You will probably sit farther away from the 64" television. The recommended viewing distance increases proportionally in relation to the screen size (for example, here is a calculator).
If you watch the same source video from the same viewing distance on both TVs, you will notice more pixelation on the bigger one.

You can also notice this effect when looking at any (pixel-based) image on your screen. Open any image on your computer. Looks smooth? Good, then zoom in by a few hundred percent. Now you will notice pixelation to the point where you can make out the individual pixels. Conversely, if you have a pixelated image and zoom out enough, it will look smooth. The same thing happens when you move your chair farther away from/closer to your TV.


After subtitle video one of my audio track is missing

FFmpeg, by default, only picks one stream of each type (video, audio, subtitles). The user needs to set map option to select tracks for an output.
ffmpeg -i /sdcard/input.mkv -vf subtitles=/sdcard/input.ass -map 0 -c:a copy /sdcard/hrdcoded.mp4

See https://ffmpeg.org/ffmpeg.html#Stream-selection


Can't add Motion Blur to layer imported from Photoshop

Thats because you have the collapse transformations button ticked:

When collapse transformations is ticked After Effects takes the options of the layers inside the 'loader' composition. So you will need to open up the 'loader' composition (double click) and then you can enable motion blur on the layers inside.

When you import Photoshop documents into After Effects as compositions, any groups in the PSD become pre-comps within AE.


Okay, so it turns out that the shutter change is automatic depending on the amount of light available. You cannot alter the shutter manually. What this means in practical terms is that, with room lightning you are 99% guaranteed that you're shooting wide open, which just blurs everything and makes the video mode pretty much unusable. Thanks Canon.


Is it possible to have paragraph text as an After Effects template (in Premiere Pro)?

Just to sync our watches: Both point text and paragraph text have the potential to be either single or multiline.

If you create multi-line text using so-called point text like this (ie, single-click of the Text tool, w/ multiple lines created by hitting the return key) ...

..then Premiere will display the multiple lines like this:

On the other hand, if you create multi-line text using so-called paragraph text like this (ie, click-drag a rectanglar area with the Text tool, w/ multiple lines created by hitting return key or word-wrapping) ...

..then Premiere will display editable fields like this:

So basically Premiere will add additional editable text fields whenever it sees linebreak characters in your text, whether it's point text or paragraph text. Hopefully that helps!


The obligatory ffmpeg answer (not a very good one, other answers still needed):

You could feed your video through ffmpeg -vf drawtext=textfile=onscreen.txt:reload=1

With whatever other drawtext options you need to put your text where you want it, with the font you want.

reload=1 makes it check the text file for changes every frame, but doesn't give you a way to update the formatting, screen position, or anything else on the fly.  (and you'd have to build your own custom thing to render your data into line-wrapped text.  You could maybe control the on-screen position with leading spaces and newlines.)  And -vf drawtext isn't very nice to use anyway.


Corrupt video when playing through JW Player

Seems like the framerate was the problem. JW Player seems to have problems playing videos with high framerates (like 1k).

A simple -r 30 in the FFMpeg command seems to have solved it


Yes, there's a private option for the MOV demuxer.

ffmpeg -ignore_editlist 1 -i VIDEO.mp4 ...


FFmpeg mosaic with hardware acceleration on Quadro P4000

Regarding,

Cannot find a matching stream for unlabeled input pad 0 on filter Parsed_hwupload_cuda_17

in

[tmp3][lowerright] overlay=shortest=1:x=720:y=480;hwupload_cuda

you have to replace the ; with a , else the overlay filterchain is terminated and the hwupload starts a new chain.

Input options like -hwaccel cuvid -c:v h264_cuvid only apply to the input immediately following them. These options have to be prefixed before each input for which they're intended.

scale_npp is a CUDA scaler, so for a hardware-decoded stream, no filter is required to transfer frames. After scaling, frames will have to be transported to system memory using hwdownload,format=nv12.


There are several programs available that allow you to do that.
One very popular tool in the live visual industry is MAX (especially Max for Live).
Its a visual programming language to create visuals for live performances that also integrates into a few tools like Ableton Live.

A program that I personally use at work is called vvvv, its a visual programming tool aswell that allows for very complex programs and is more "multipurpose" than MAX and offers a huge collection of add-ons to work with all kinds of peripherals and input devices.
For getting started I highly recommend having a look at this cheat sheet, vvvv has a VERY minimalistic GUI and heavily relies on shortcuts and hidden menus for more advanced settings and features.

Unfortunately only available for Windows but unlike Max its absolutely free to use unless you want to use it commercially, then you have to buy a commercial license (no differnces to the free version, you still just download the same version but have permission to earn money with it).

A also a free solution but only available on Mac would be Quartz Composer which is coming with Xcode. Its also a visual programming language that works in a similar fashion like vvvv and MAX.

E: Another solution would be MadMapper, its more meant for projection mapping but also does video with filters that can react to sound. In can also integrate into Live afaik.


picking a frame and an overlapping audio from a video

The inverse way of doing this is faster.

Get file duration

ffprobe in.mp4 -show_entries format=duration -of compact=p=0:nk=1 -v 0

This will print the duration in fractional seconds.

Pick a random value between (0, duration - 1)

Run

ffmpeg -ss value -t 1 -i in.mp4 -map 0:a -c:a copy audio.m4a -vf thumbnail -vframes 1 img.png

The thumbnail filter will pick one frame out of the 1 second of video provided to it.


Tool to transcode Canon PF24 MTS files to native 24p?

The usual tool recommended for this is JES Deinterlacer, which apparently has an adaptive method of finding and removing the 'smear' frames. I haven't used it myself.

The reason this is an issue is that the field cadence used for PF24 is (stupidly, IMO) different from the typical telecine cadence, which is what most 'inverse telecine' algos detect and remove. So applying standard inverse telecine just adds another layer of mess-up to the video.


What are the minimum camera characteristics required to record a movie for cinema?

If I understood your question correctly, you want to project a movie on a cinema.

This would depend on the projector used by the movie theatre. Most movie theatres use film projectors, for which there really isn't a "quality", but rather you'd have to worry about getting a decent camera with decent film at the correct size. In this case I don't know much about transferring from digital to film.

If the movie theatre uses a digital projector though, they usually come in 2K (2048×1080) at 24 or 48 frames per second, or 4K (4096×2160) at 24 fps if they follow the DCI specification. Basically, get the best camera you can get. 2K should be enough to have a pretty good quality experience at the distance a cinema projection is seen from.

Also remember though, the quality does not only depend on the amount of pixels on screen, but also on the compression used, and the sensor of the camera itself, so be prepared for large files or a lot of processing to compress your movie to a high quality profile.

If you can't get a 2K camera, 1080p should do: it's not too far from 2k: it the same vertical size, just a little less wide.

Links that might help:

http://www.mkpe.com/digital_cinema/faqs/tech_faqs.php#2K+4K
http://en.wikipedia.org/wiki/Digital_cinema

Hope it helps!


One approach is to change the speed of the clip slightly.  This will change the pitch; speeds above 100% will make the pitch higher, and speeds under 100% will make the pitch lower. You can fiddle around with the percentages until the pitch matches.  Obviously the clip will visually run faster or slower as well, so hopefully you won't need to change the speed too much. If it makes the clip look funny, you may want to separate the audio track from the video track so you can just adjust the audio speed while leaving the video speed at 100%.

As for pitch vs. key, I can tell you what they mean from a musician's point of view. A "pitch" is the vibrations per second (hertz) of a note.  The more cycles per second, the higher the pitch. The word "key" has two meanings.  On a piano, each of the things you can hit is a "key".  There are white keys and black keys.  Each one has a different pitch.  The other meaning of "key" pertains to composition. The "key" a song is written in defines the scale it uses (for the most part). Songs in the key of C major or A minor will be played on all the white keys (unless there is what's called an "accidental" note, which would be one of the black keys).  Songs in the key of G major or E minor are played on all the white notes except for F# ("F sharp"), unless there are "accidental" notes.  There is a major and minor "key" for each of the 12 "keys" in a scale.


ffmpeg: How to avoid "start: 0.033333" when extraction audio from mp4

Correction: instead of

ffmpeg -i output-audio.mp4 -i output-video.mp4 test-combine.mp4

it should be

ffmpeg -i output-audio.mp4 -i output-video.mp4 -shortest output-combine.mp4

Kind of answer:

ffmpeg -i output-video.mp4 -itsoffset 0.033333 -t 9.966666 -i output-audio.mp4  output-combine.mp4

seems to do the job, although the result differs in file-size slightly from an direct extraction (direct is bigger than combine) done with

ffmpeg -accurate_seek -ss 30 -t 10 -i input.mp4 output-direct.mp4


<link href="http://mysite.net/sites/all/libraries/video-js/video-js.css" rel="stylesheet" />
<script src="http://mysite.net/sites/all/libraries/video-js/video.js"></script>
<script>
  videojs.options.flash.swf = "http://mysite.net/sites/all/libraries/video-js/video-js.swf"
</script>
<script src="http://mysite.net/sites/all/libraries/video-js/videosub.js"/></script>

<video class="video-js vjs-default-skin"
       controls=""
       data-setup="{&quot;example_option&quot;:true}"
       height="264"
       id="example_video_1"
       poster="http://video-js.zencoder.com/oceans-clip.png"
       preload="auto"
       width="640">
  <source src="rtmp://mysite.net/oflaDemo/takeout.flv" type="rtmp/flv"/>
  <track kind="subtitle" label="English" src="http://mysite.net/takeout.srt" srclang="en-US" />
</video>


Highlighting current speaker on multi person podcast?

Easy way: patch some client for voice conferences, save info about speakers. I did it with Mumble. Then write simple app for rendering such movies, read/write of images and footages can be done with ffmpeg, fftw3 can calculate spectrum for us, grab some filters for image scaling...

Hard way: write each speaker in separate channel (mumble can do this), then place all tracks in After Effects, use there tracks for scaling avatars and adding spectrograms. You can done it without writing code, but if you have many-many podcasts, you will suffer. Also, this not applicable if you want broadcast such video in realtime, for example, with questions and calls of guests.


Do cartoon creators have to pay software owners?

As a media creator, you pay for use of the tool, not for the use of the result from using the tool.  Different companies have different systems, but you either buy the software up front or pay a fixed subscription to have the tool available to you.

It does not matter if you are using it for personal use (though sometimes educational copies will get you a discount for non-commercial use), making a video for your ad supported youtube channel or producing the super bowl.  The cost of the software remains the same.

It's similar to buying any other tool (say a hammer).  You can either buy the hammer or rent it from a hardware store, but you don't pay for every house you sell that you build with the hammer, you just pay for the hammer.


How should I interpret the results from Netflix VMAF

The numbers don’t mean anything. They only have value when compared to other values. You can compare the results between two encodes of the same file and determine which encode produced better results.


According to the product comparison source, the free version does not allow to "User definable project locations". This is only possible for the Pro version.


FFmpeg raspberry pi OMX hardware encoder with inline_headers enabled = Increasing latency over the time

Gyan is right :

I just need to add +global header flag and only the "dump_extra" bit stream filter

My patch become useless, ffmpeg can add SPS/PPS itself inside the stream.

My old cmdline is - missing inline SPS/PPS for infinite streaming - :

ffmpeg -f v4l2 -i /dev/video0 -vf hqdn3d=0:0:50:50 -c:v h264_omx -profile:v baseline -b:v 1500000 -f rawvideo tcp://127.0.0.1:8003

I just need to add "-flags:v +global_header -bsf:v dump_extra" the cmdline become :

ffmpeg -f v4l2 -i /dev/video0 -vf hqdn3d=0:0:50:50 -c:v h264_omx -profile:v baseline -b:v 1500000 -flags:v +global_header -bsf:v dump_extra -f rawvideo tcp://127.0.0.1:8003

(live result here : https://www.vigibot.com/ click on the "Astro" public robot)


Gimbals are good for adding movement without shake, they're brilliant for run-n-gun documentary work where lugging a tripod slows you down. They're perfect when you're in the thick of things, using a wide lens to shoot close to the action.

They take a bit of time to master, you need to practise steering them and think about the movement. You also need a camera with good autofocus and you probably want to shoot with more depth of field; while you can focus-pull on a gimbal-mounted camera, particularly with a remote motor driven follow focus, it's challenging.

A major disadvantage of a gimbal is that you're limited in terms of the weight of the camera. Unless you're shelling out a lot, the cheaper gimbals are limited to something the weight of a DSLR. So if you're looking at using a cinema camera or shoulder-mount style video camera you're also going to be spending more on your gimbal.

A gimbal is fairly useless if you're shooting interviews, unless you're a fan of the camera-always-needlessly-moving style of interview shooting, and if you are, damn your eyes, you're a disgrace. For static interview set-ups a tripod will win hands-down.

You also wouldn't take a gimbal if you were trying to get shots of snow leopards in the wild, live sports, or candid long-lens documentary footage. You can't really stick long lenses on them, not just because of the weight, but also because even with the stabilisation the constant movement is still too much.

Another consideration is ergonomics. Even though the average video tripod weighs more than the average gimbal when you're lugging it to the gig, once you're there you put the tripod on the floor. Because you have to hold it while shooting, shooting for extended amounts of time with a gimbal can be exhausting. Also, the batteries on a tripod never go flat and the software doesn't ever need resetting.

Finally, a bottom of the range gimbal is going to cost you the same as a pretty good tripod. Worth considering if money is tight.

TL;DR Both are useful parts of the kit, but they have different uses.


Change Output file name in AfterFX .jsx script

I found the right property that has to be changed and wrote a function that does the job:

function renameComposition(newName) {
    var curComp = null;
    for (var i = 1; i < app.project.numItems +1; i++) {
        if(app.project.item(i) instanceof CompItem) {
            curComp = app.project.item(i);
            curComp.name = newName;
        }
    }
}


How to balance audio across multiple clips?

You could create a compound clip of your voice over and then add a Compressor effect onto this compound clip.


Remove:

-profile:v high -level 4.0 - You don't need to set a profile or level because you're not targeting any specific, limited, or outdated devices. These options don't matter because YouTube is going to re-encode whatever you give it and set its own profile and level.
-pix_fmt yuv420p - You can let YouTube deal with the pixel format.
-bf 2 -coder 1 - These are already handled automatically by -preset.
-b:v 15M - This is mutually exclusive with -crf. Just use -crf instead. It will let you set the quality instead of you trying to apply an arbitrary bitrate.
-ar 48000 -ac 2 - No need to change the audio rate or channel layout. If you do what to change those do it via the dshow -sample_rate and -channels input options.

So your command can look like:

ffmpeg -f gdigrab -framerate 60 -i desktop -f dshow -i audio="Microphone (2- Realtek Audio)" -c:v libx264rgb -preset slow -crf 18 -b:a 384k output.mp4

The input is RGB, so you can avoid the RGB to YUV conversion by using -c:v libx264rgb.
If your computer can handle it then consider using a slower -preset.
You can use flac or libopus for audio (change output.mp4 to output.mkv), but for microphone it will hardly matter.
You can try -crf 17 if you think you can see a difference. I wouldn't bother with the huge file increase from using lossless (-crf 0).
You can see if dshow is faster than gdigrab for capturing the desktop. I don't use Windows so I can't test for you.

Note that a non-FFmpeg based player may not be able to decode the file, but I'm fairly certain YouTube will have no issues with it.


After Effects rendering a video at base 99, but outputs as a 60 one

Since your source is in h264 I'm guessing you are encoding to mp4/h264 again?
The format doesn't support a frame rate of 99, you will have to use a less constrained video codec like mjpeg in an avi or mov container.

Whenever you are violating certain video or audio specs After Effects will tell you that in your export settings like shown here (see the little warning triangle in the left corner). AME doesn't do that for some reason but will show the adjusted settings in the source output info field.


How to remove honking noise in room for voice over recording

When recording, as soon as you hear a honk or another noise, stop and go back to the beginning of the sentence you were saying. After your done, go back through your recording in a programme like Audition or Audacity and remove all the half-sentences that were interrupted by a honk, so only complete sentences remain.


If you're comfortable with the command-line, FFmpeg will do the job.

Since these are VHS rips, 5-10 mbps is way overkill when using the H.264 codec.

Suggested command:

ffmpeg -i rip.mpg -vf yadif,scale=iw*sar:ih,setsar=1 -crf 18 -c:a aac -b:a 128k rip.mp4

This will deinterlace the video, then convert it to a square pixel representation. Audio codec is AAC @ 128 kbps. Video bitrate control method is CRF, which strives to allocate bits as per content complexity i.e. tries to maintain uniform quality throughout. 18 is a good value for SD content. Lower values produce better quality (and larger files) but shouldn't be necessary here.

You can create a batch file to automate the conversions or use a GUI like Avanti, which has a job control manager.


Best way to edit multi screen footage into one clip.

Adobe Premiere Pro actually has a multicamera mode you can use where it will display all the videos playing together and you can switch between them.  You just line up the time and then can swap.  If they aren't time synced, then you can simply cut clips out and place them at will.  That's why it is called non-linear editing (NLE), because you can take clips and put them together without having to work linearly (deck to deck) like you used to.

While links are generally discouraged, I think this link to Adobe's help pages is probably the most ideal explanation of how to do multicamera editing.


How to extract and modify data rate and time scale from OpenDML (AVI) file

ffprobe can supply the frame rate as a rational value

ffprobe in.avi -show_entries stream=r_frame_rate,avg_frame_rate -select_streams v -v 0 -of compact=p=0:nk=1

The first rational value is from the stream header; the 2nd is the computed value which will differ for a variable frame rate stream


How peoples create video highlights from matches?

Alot of times there is someone who tracks the game and writes down specific times. Say there was a goal at the 35 min mark, they jot down "Goal - team name - 35:26" or something related to that. Then later on they have a list of different times as well as different events (goals, saves, close shots, etc.) and choose the best ones. Then the editor opens a recording of the whole game in Final Cut Pro or Adobe Premiere and goes to those times, trims them quickly, add cool transitions and exports. I have experience with it and for a 3 period hockey game, I can normally edit and export in under an hour. The log file saves you from having to scrub through and find the best shots.


Repeating Flashing Text in After Effects

This can be done easily with a mixture of time remapping and an expresison. Simply precomp the text animated as once on, once off (keyed out manually), then drop it into your main comp and apply the following expression:

loop_out("cycle",0)

Then, just extend the composition for as long as you would like it to loop.


To create video from hand drawn images, you will need to prepare yourself for drawing a lot more images, as you likely need one for every frame, which you then photograph as a single image. When there images are combined you get the effect of movement.

It's a lot like the old fashioned flick books we used to draw at school- with a slightly changed picture at the corner of each page so that when you riffle the pages you get the effect of continuous movement.

There are some video packages that will let you build a virtual model and overlay your images onto the model so you can animate it etc. But you will be talking big bucks, and steep learning curves.


Can this be created in a free animation software like Synfig or OpenToonz?

I think you can use Krita 4.0+ for this animation.


How to setup Adobe Premiere Pro CC to edit Interview type videos for YouTube

What presets do you recommend when creating Premiere project for a YouTube video? I read that YouTube recommends uploading 720p videos, but nothing like that exists in Premiere presets.

Any preset works as long as your export settings correspond to what YouTube can ingest.

By the way, 720p presets exist in Premiere, for most codecs.
Here's a screen capture for the DNxHD codec:

What settings should I use to render the sequence to have good quality but acceptable file size?

For a 720p file, make sure your data rate is no smaller than 5 Mbps.

For a 1080p file, make sure your data rate is no smaller than 10 Mbps.

Codec: H264.

Field order: progressive usually (depending on your raw footage).

Aspect: square pixel usually (depending on your raw footage).

Audio: AAC, 4100Hz, Maximum Bitrate.

Maximum render quality.


How to optimize already-encoded videos of still images (e.g. presentation slides)?

(Ideally unaffected parts of the stream would be copied directly without being re-encoded, but if this isn't possible then I can live with a bit of quality loss.)

First of all, I doubt there's a way to do this losslessly. Very few things that touch the actual, encoded video content in any way can be done without re-encoding.

That said, I tried out a few things just now and found that a very effective way of doing this is making the video frame rate variable and dropping duplicate frames using FFmpeg's mpdecimate filter:

ffmpeg -i input.mp4 -vsync vfr -vf mpdecimate output.mp4

The mpdecimate filter has a bunch of additional options that can be set but I found the default worked just fine. I downloaded this video with youtube-dl, encoded the first 5 minutes of it with standard options -c:v libx264 -an -t 300 -crf 22 -preset medium. I then encoded the same part of the video with the additional options -vsync vfr -vf mpdecimate and the resulting file was 42% smaller than the one encoded with a constant framerate. 30.8 MB vs 18.0 MB. The encoding process was also significantly faster since there were much fewer frames to encode.

However, I've heard that some players may have trouble playing variable frame rate video and that it can also cause audio sync issues. So I also encoded a version with the same settings but included audio as 64k Opus. I played the video in mpv and Firefox and didn't observe any audio synchronisation issues.

What I did notice was that this does affect seeking, sometimes quite dramatically. Firefox was able to seek fine for the most part, but when seeking to some parts of the video FF would display a frame that was long past (like a picture of the speaker at 2:03) and only start playing from a more recent frame about two seconds later but sometimes this would continue for a long time, like 40 seconds. However, when playing the video without seeking playback was perfect.

mpv played the file perfectly too and seemed more adept at seeking. However, some parts of the video couldn't be seeked to (presumably due to the same issue/property of the file which caused FF to have trouble with seeking) and would jump ahead or behind the spot I attempted to seek to. Both of these player issues probably have something to do with the general lack of frames inside the video. AFAIK most players use keyframes for seeking and services like Youtube use a keyframe interval of about two seconds to make videos easier to seek. So this issue probably can't be fixed without sacrificing the file size decrease gained from throwing away duplicate frames.

Due to these issues I wouldn't really recommend this method for video that's intended for wide distribution. However, it's probably fine for video that's inteded for personal use or archival. With the latter purpose in mind I attempted to re-encode the variable framerate video back into a constant framerate one using the -r 30000/1001 switch, just to check that it works, and ran into this issue with the audio. I used the -max_muxing_queue_size 9999 switch and the encode worked just fine and the resulting file played without issue. So as long as your tools are up to the task this should be a viable method of archiving video material.

Since this method does involve re-encoding you're going to lose some quality. However, since getting rid of duplicate frames significantly reduces the file size with this type of content, you can probably lower the CRF value to a point where the visible quality loss is almost insignificant and still end up saving a lot of space.


The worst part of transferring old VHS to DVD is that it has to be done in real time as VHS is linear access and analog. Here are two ways I know of that work. One will eat your time the other will eat your wallet.

1) for about $35 or less you can get a transfer device which connects the output of your VHS machine into a USB interface to your computer. There are a bunch of these out there, they are all about that same as this:

http://www.newegg.com/Product/Product.aspx?Item=N82E16815306013

It comes with software to capture the VHS program and "software included to burn full screen video capture, edit and create DVD" but you can take the captured mpeg files and put them in any video DVD authoring suite too. The software is easy to learn, you can start transferring quickly.

I did this process 3 years ago, I must have saved over 50 VHS tapes. So the cost was less than $40 but I surely put more than 40 hours into that project.
:>

2) Costco and many other large chains have video transfer services, e.g.:

http://www.costcodvd.com/videotape_to_dvd.aspx

Prices range from vendor to vendor, $9 to $20 per transfer.


Is it possible to convert output of TV Channel (set top box 720p) to 1080p?

What you are talking about is upscaling and any current HDTV will do upscaling automatically.  Upscaling doesn't work miracles though, it will only make it so that the lower quality signal can be watched on a higher quality display.  It just multiplies the pixels so that a 720 by 480 (.9 pixel compressed) signal for example doesn't end up only taking up 1/4 of your screen.  It doesn't make the video any higher quality (it in fact will be slightly lower quality than if you were to watch it at the native resolution).

Your problem however is not lack of upscaling.  Your problem is using the wrong signal.  You are currently connecting your cable box to your TV through what is known as a composite cable.  It's an old type of analog connection that includes all the video information on the single yellow pin and sends left and right audio on the white and red pins respectively.  It is only capable of reproducing a standard definition signal so you are not actually using HD signals.

To benefit from 720 or 1080 signals, you will have to either use analog Component video (the red/green/blue connectors) or use an HDMI cable.  It's worth noting that as long as your TV and cable box both have component inputs/outputs, you can actually re-purpose the existing Composite cable to carry a component signal, but there will be no audio (you need a separate white/red pair for the audio, which is the same format as with the composite signal).

Once you fix the cable and hookup issue, you should see better image quality.  Personally, I'd recommend using HDMI if it is available on both your cable box and TV as that is the easiest to setup and will give a nice digital signal.


Getting audio from a Bell & Howell Filmosound 8 Projector

The projector you have is for use with the Bell and Howell model 450 cassette recorder. This projector has no optical or magnetic means of playing or recording sound. IN this system, which is a "Double Sound System", the sound is recorded on the 450 player/recorder via a special cable that goes between the camera and the recorder. The film itself has no magnetic or optical sound track on it. The recorder records both the audio from a microphone and sync information from the camera, hence the special cable running between camera and recorder. After the film is processed and returned, it threaded into the projector and then the cassette player/recorder is connected by means of another special cable to the projector, via the 2 jacks pointed out in your photo. The jack questioned is a size 3/32.
This special cable is used to deliver sync information that was originally recorded on the tape to the projector for lip sync sound reproduction


Stripping metadata sometimes doubles video, the second half of (doubled) video is without sound

This is the Gyan's solution, and it works:

ffmpeg -ignore_editlist 1 -i input.mp4 -map_metadata -1 -c copy output.mp4

The important part here is the -ignore_editlist 1 option  (yet undocumented).

The reason is that the original video contains for its video stream an edit list (elst box) with entryCount value set to 2 (which is ignored by some media players — but not by FFmpeg, which interprets it as repeating the video stream twice).

The Gyan's explanation (see ticket #8456):

  The video stream in this file has an edit list that says that the video should be displayed twice. So actually looping the video is the correct behavior. But I think the file is actually invalid, as the duration of the track in the tkhd and the mdhd box both shows 6s, although it is double that after applying the edit list.

Note:
 I tried several videos converted by Any Video Converter into the H.264 format (as input files), and the same problem occurred with most of them.


How to make this kind of animated video?

This is called a "write on" effect and is typically accomplished with either a 2d animation package such as Adobe Flash or with a motion graphics software package such as Adobe After Effects or Apple Motion.

It generally involves building the imagery as vector graphics.  Any lines that will be animated are generally drawn as paths that can be "written on" by starting at the beginning of the path and drawing the line until it reaches the end of the path.

It can also be accomplished using motion recording and a digitizer tablet which will allow the writing to be done in real time and recorded, it can then be accelerated or decelerated as needed, or the paths created and timings can be tweaked to smooth things out using typical keyframe animation.


disabling streams in quicktime movies with ffmpeg

QTCoffee let's you edit track information in QuickTime files, but it's OS X only and donationware:

$ modmovie -list clip.mov
Track list for clip.mov:
    * Video Track
    * Timecode Track
    * Sound Track
$ modmovie -disable "Sound Track" -save-in-place clip.mov

If you are on OS X, and if you have QuickTime Pro (included in the pro apps), I would rather use that and AppleScript.

For other platforms: libquicktime can also edit track properties through it's API, so you will need to write a command line wrapper first ...


Is there a way to basically start a counter from a given frame in ffmpeg?

Just subtract the first frame index from the expression:

overlay=x='n-438':y='n-438':enable='between(t,438/25,450/25)'


Append part of the audio at the end of another audio

Assuming the duration of B is X (in seconds), you would use

ffmpeg -i A -i B -filter_complex "[0]volume=0:enable='lt(t,X)'[a];[a][1]amix=duration=first:dropout_transition=0,volume=2" out


how to change volume of audio at specific time in audio and add that audio to video(with audio) using ffmpeg command?

Use

ffmpeg -i video.mp4 -i audio.mp3 -filter_complex \
"[1:a]volume=150:enable='between(t,10,20)'[1a];[0:a][1a]amerge=inputs=2[a]" \
-map 0:v -map "[a]" -c:v copy -c:a libvorbis -ac 2 -shortest output.mp4


Try

ffmpeg -i video.mp4 -filter_complex
"[0]trim=start_frame=0:end_frame=1[c];[c][0]blend=all_mode=difference,blackframe" -f null -

As long as the very first frame of the video is not something like a static logo screen or similar, this should help you locate the loop joint. If it is, change the start frame value to a higher number. The end_frame should be 1 greater.  Basically, one instance of the video is trimmed to just one frame, then it's composited with the full video frame-by-frame, in difference mode. When the blend filter encounters an identical frame i.e. the point of loop, the result will be a black frame. The blackframe filter should then identify that frame (and all others at each loop turnover), with a readout looking like this:

[Parsed_blackframe_2 @ 0000000003607d20] frame:325 pblack:100 pts:13837 t:13.837000 type:I last_keyframe:325

That t value represents the duration of a single loop.


How can loop animations with a different start and end point in after effects?

Using time remapping could be a solution. You animate one full circle of the animation and precompose this. Then you time remap that layer and use loopOut('cycle').


How to use Avisynth+AvsP with VirtualDub frame server?

Did you install the frameserver support with auxsetup.exe? You can test if it is working by trying to open your video.vdr with another instance of virtualdub.exe.

From your description it doesn't look you really need the frameserver. You can trim the video with avs itself (e.g. Trim(100,200)), this is more robust and efficient.

Btw did you know there is active avisynth community on forum.doom9.org, forum.videohelp.com


The fs parameter will stop the encode once it hits its value. So, if the output hits the 10MB mark while encoding the 15th second, then that's the duration of your output file.

If you want to make sure that the entire file is encoded but it doesn't cross the set target size, then use the bitrate method. To accommodate the muxing overhead and other data within the target size, set your video bitrate slightly lower by, say, 1-2%


Is it possible to create keyboard shortcuts for applying MULTIPLE/DIFFERENT effects?

Click in the Menu Bar on “Final Cut Pro”. A pull down menu opens, then click on “Commands” then “Customise”
A window will open with all available commands possible. Many won’t have a shortcut and you can set keyboard shortcuts for them. Mind the search bar in the top right corner.

I suggest to duplicate the Default command set before start changing.

If this is not enough then you could look into make a command recording with Automator.


Video with transparent background file-size issue

Transparent video on the web is possible, but it's going to only work for users with modern browsers.  You'll need to export in two different codecs - one for Chrome users, one for Safari.
From this site:

Safari supports HEVC with alpha, Chrome does not.
Chrome supports VP9 with alpha, Safari does not.

You need to export an HEVC with alpha channel for Safari.
You need a .webm VP9 codec with alpha for Chrome.
The animation codec you've exported at is not designed for web publishing.  It's a lossless very large codec designed for editing or archiving your finished project before you use it to make a smaller lossy version to go on the web.
This is how to make the webm using ffmpeg
ffmpeg -i "movie-prores.mov" -c:v libvpx-vp9 movie-webm.webm

I would recommend also exporting a "fallback version" for browsers that can't cope with transparent video (as a normal h264 mp4 or webm).


Check whether the equalizer effect you made runs through to the end of the song you are using. If that is not the case, just stretch the animation to the end of the song and you should be good to go.

If that does not work, create a new project or sequence and import your current files to the new project.


What causes these lines in the top left of some video recordings?

That's EIA-608 analog closed captioning data. It should be on line 21, which is within the vertical blanking interval of the SD signal. Either the line was shifted down or the VBI was captured as being active video.


Is it possible to ripple edit Adobe Premier?

Yes, you can use the Ripple Edit Tool. See here (Tools panel and Options panel section)


How is it customary to pay a video editor - global or per-hour?

My experience is that it really depends on the situation and the editor.  I personally have done both hourly and project rate editing.  Typically I'll only take project rates if it is clear how much time it will take (ie, a simpler project), but it will really depend on your area and what particular editor you are working with.

I'd suggest having estimates of what you are willing to pay for either option and then simply ask them which method they prefer and then supply them with that number.  If you have a preference one way or the other, you might also ask them if they would consider the method you prefer.


Why does Premiere Pro 6 Generate Peak Files?

Peak file creation as well as creating alternate forms of various assets is perfectly normal background behavior for Premiere.  When you load new assets into a project, Premiere automatically starts a number of background processes to analyze the clip and make it easier for Premiere to do its job smoothly.

This should not cause any trouble and should not be a concern to you.  It is just the program functioning as it should.  It shouldn't cause any problems if you start working with it prior to completion of the work, but some functions may work more slowly or provide incomplete information while the processes are still going on.


The 4k videos from Sony a7iii are quite large by default, suggestions on HQ ways to compress?

Well first, the a7iii at 100mbps is far from lossless or even minimally lossless. It's highly compressed in a 4:2:0 or 4:2:2 colorspace depending on your recording selection. The files are incredibly compressed when compared to what's considered minimally lossless or lossless compression algorithms. Shooting 4K at a "near" lossless level would require 4:4:4 and a codec such as REDCode at a compression of 4:1 or 3:1. These files are astronomically larger.

As far as in camera, you're limited to what the camera provides as options, in terms of 50mbps, 100mbps, and so on.

Your only real option to compress down and reduce file size would be to run the files through an encoder, either a real time "live" encoder which captures the video from the camera's HDMI out, or in post production, using Adobe Media Encoder or a similar application.

With 4K, you could compress down, if you are happy with the color and don't plan to do any additional color correction later, to a 4:2:0 H.264 or H.265 codec using an MP4 wrapper. Using a Variable Bitrate, depending on your content/how much motion/detail/etc, you could likely get very good results with bitrates in the 25-50mbps VBR, allowing a Max Bitrate to be set up to 100mbps.

Adobe Media Encoder is available from Adobe and comes with Adobe Premiere. Your best bet if this is something you need, is to run some trials, and to find the best "acceptable" bitrate you can live with for archival purposes.


To determine if a channel has SCTE-35, you will need to have the Program Map Table (PMT) recorded. From the PMT, you should be able to find the Packet Identifiers (PIDs) of all of the streams associated with the channel you are interested in. There will be a PID for the video and at least one audio PID. If SCTE-35 is present, there will be at least one PID with a stream type of 0x52.

Keep in mind, the PMT and the SCTE-35 PID will either need to be in the clear or you will need to be able to decrypt the conditional access.

The easiest way to check all of this is with a transport stream analyzer. Most of the freely available ones, like DVB Inspector, will show you the PMT. However, you will probably need a commercial analyzer to get it to display something more than just the raw SCTE-35 bytes.


I think this article should help you:
https://support.google.com/youtube/answer/1722171?hl=en


What solution do you use to bring a udp, rtmp, ... stream back into the hd-sdi format

While I am pretty sure there are a few stand alone devices that can perform this process, they are effectively doing the same thing as you are doing by playing it back through the HD-SDI output of a computer.  RTMP is a very different type of signal than SDI and requires fairly complex translation to get it back to SDI.


Let's call the video frame rate F. And the start time of the overlay t

Use

ffmpeg -i in.mp4 -i ol.png -filter_complex
         "[0]trim=0:t,loop=X*F:1:t*F,setpts=N/FRAME_RATE/TB[fol];
          [0]trim=t,setpts=N/FRAME_RATE/TB[post]
          [fol][1]overlay[pre];[pre][post]concat" out.mp4

The loop filter accepts frame index numbers, so you have to convert your start time and loop time to frame counts and substitute with those values.


Why hasn't "still image camera matching" technology been used for tracking moving shots?

You're describing augmented reality.

Try something like Carrot Weather's AR Mode or Memoji?

The answer to your question is "Many vendors have implemented 'Augmented Reality' or AR, which works similarly to fSpy.


"Swimming" texture artefact on a drone video footage

This is due to an issue with resolving an image to a set of pixels called moire.  Basically, due to the regular pattern, some parts of a straight line end up on one line of sensors and the other end up on the other.  This results in a jaggy line and the line moves around as the image moves across the sensor, producing the effect you see.  It can occur when shooting any regular repeating pattern off angle to vertical or horizontal.

The "solution" is to apply a gaussian blur (or moire reduction filter if you have one available, but it will basically do the same thing) to normalize the lines enough that they don't flicker back and forth anymore.  It results in a loss of crispness of the image, but will also deal with making the shifting less pronounced.


ffmpeg: Placing an overlay over multiple images

If all TIFFs have the same properties, and are named sequentially like image0001.tif, then you can use
ffmpeg -i image%04d.tif -i my.png -filter_complex "[1][0]scale2ref[i][m];[m][i]overlay" out%04d.tif

This will generate images with the overlay named as follows
out0001.tif
out0002.tif
...


Keep residual image of an element in a video

You'll need to separate the spider from the background somehow. Then you can stack successive images of the spider on top of the initial background, and the previous spider image frames.
here's an ascii art representation
        sp5------------->
      sp4--------------->
    sp3----------------->
  sp2------------------->
sp1--------------------->
bg---------------------->

So on the first frame we see the BG with the first Spider frame composited above it. This frame is held, then on the next frame we add spider frame 2 and hold, then on the next spider frame 3 is added, and so on.
How you actually achieve this depends on the software you're using.


Use ffplay to check how much to brighten video before using ffmpeg?

You can use simple filtergraphs easily with ffplay.

Example using colorlevels filter in ffplay:

ffplay -vf colorlevels=rimax=0.902:gimax=0.902:bimax=0.902 input.mp4

Then you can use the same filter in ffmpeg:

ffmpeg -i input.mp4 -vf colorlevels=rimax=0.902:gimax=0.902:bimax=0.902 -c:a copy output.mp4

This example will stream copy the audio.


Shortcut to extend length of still image clip in Premiere?

As happens, the act of asking a questions leads me to the answer faster.  I can do this with a combination of two shortcuts:

Down Arrow (Go to Next Edit Point)
Shift + Q (Extend Previous Edit To Playhead)

(Still be interested if there's a way to do this with a single shortcut)


Want to try green screen with my elementary students. What do we need. Using Mac OS 10.5.8

Summary:
Green screen work is not trivial, the quality largely depends on technique and good lighting, and at least enough space for doing the work you suggested. It does require software although iMovie bundled in most MACs has a chroma key effect (the software that allows you to composite your subject with whatever background you want including maps, other videos, pictures etc).

Things you will need to purchase:

A pro screen.

Enough lights (with stands and diffusers) with the correct color temperature to fill the screen evenly as well as light the subject appropriately.

For your situation I think you might get there on $200, assuming you have iMovie on your MAC, spend under $100 on a screen, and another $100 on enough lights do do the job.

Details:

Here is a good tutorial on how to shoot with a green screen:
http://www.youtube.com/user/tubetape?v=q3PZO_lCBkw

Here are some tutorials showing you how to composite your green screen with iMovie:

Things to remember:

1) The green screen has to cover the entire area of the frame you are shooting. That is you can't have the green screen half way in the shot or showing any edges.

2) The green screen has to be flat, no wrinkles, no reflections, no hot spots, and no shadows.

3) The green screen has to be lit evenly and bright enough to register light in the frame.

4) The green screen should be at least 5 feet behind the subject and appear diffused not in focus. (The subject should be in tight focus, and the background green screen out of focus to optimize how it blends in the chroma keyer in your editing software).

5) Your subject can not wear anything green and should not have anything reflective on them, e.g. jewelery, watches, glasses.

As you can see from the tutorials a 6 x 9 room may be too small. Alternately if you purchase a portable screen, you can clear the classroom, move your lights in, set up the screen and you are good to go. Tearing down the set should take even less time. Better to have plenty of space to do this right then jam this into a space that doesn't allow for the minimum screen to subject distance.

Alternately, if you shoot outside with indirect sunlight or on a cloudy day, you won't have to buy lights.

Suggestion 1: Take it in two steps. Get a screen first, shoot outside when the light is flat and diffused to get the hang of it, then spend the money for lights and go for the studio thing.

Suggestion 2: Should you use the portable set it up as you go method, once you achieve optimum results document the studio set up, where the lights are placed and aimed, where the subject is, and so on so you can repeat this next time.


Are mp4 bit rates the same as Flash bytes per frame?

Comparable yes but unlike Flash you have a very consistent average size over the timeline of your content. So you don't need to download as much to show content but can actually just start right away to play the video if your bandwidth is equal to (in practice it should be a bit higher of course) the average bitrate of your video. You only need to pre-load a tiny bit to account for inconsistency in the download rate.

Edit: What you have to consider when using looking at the graph you posted, you have quite a bit of very low bitrate frames, so the peak is not really that important, they somewhat cancel each other out. Whats important when streaming is the average bitrate as this is the bitrate that is needed to stream the video consistently without stopping to fill the buffer a bit more.
In reality you usually need a little more than the average to cope with very "noisy" bitrates that go up and down for long periods, your video is already a rather extreme example, usually you have a much smoother graph. It always depends on the complexity of your video to how well you can compress a frame.


Adobe Premiere Pro CC(2015): Cache edited clips

maybe if you used the Adobe Creative Cloud you could upload your footage to it ... if you have an account with them and if you a fast-enough internet connection. otherwise, make lo-res proxies and edit from a USB-3 external drive (since it sounds like you have limited space)


What software is used to produce this video?

It honestly looks hand animated to me, however the digital equivalent would be to start with vector graphics and animate them in something like Flash or After Effects.  Either product is capable of keyframe animation of various assets including vector graphics.

Personally, I'd rank them by likelihood as by hand and scanned, followed by Flash, followed by After Effects as the possibilities.  Keyframe animation allows for objects to be set at a start point and an end point and the computer fills in the values in-between.


After Effects expression - define two positions, tween back and forth

Darby Edelen posted an answer on Creative Cow.

  You won't be able to define any spatial paths any other way, but this
  should work:

d = 5;
fd = d * 2;
p1 = [200, 500];
p2 = [500, 200];
t = (time - inPoint) % fd;
if(t <= d) linear(t, 0, d, p1, p2);
else linear(t, d, fd, p2, p1);

  This starts from p1 at the beginning of the layer and runs to p2 over
  5 seconds, then goes back to p1 and begins again. If you want the
  animation to ease in to each point you could use the ease() function
  instead of linear() providing the same arguments.

  -Darby


Use a filter to reduce the framerate beforehand

ffmpeg -i in.mpg -vf "fps=24,scale=-2:360,drawtext=timecode='00\:00\:00\:00': r=24" out.mpg


Linux command-line option to do video lens correction?

Frei0r effects has a lenscorrection module:
https://ffmpeg.org/ffmpeg-filters.html#frei0r-1

this is an example to remove the fisheye distortion of a samyang 7.5mm

  -filter_complex 'frei0r=lenscorrection:0.5:0.5:0.16:0.6'

this is the docs from the mlt framework for frei0r lenscorrection
https://www.mltframework.org/bin/view/MLT/FilterFrei0r-lenscorrection


Tips to make professional looking hand-held videos using smart phones?

Don't shoot handheld and don't use a phone.

Baring that, use stabilization such as a steadicam or steadicam jr (if using a tripod really isn't an option), use lots of well controlled lighting (phone cameras suck at low light) and plan your shots out very carefully.  Use external audio recording rather than the phone's built in mic and do a lot of grading of the color in post.  By the time all is said and done, getting a better camera is likely a cheaper option for the most bang for your buck, though any camera would benefit from the other mentioned options.

It is exceptionally hard to get a professional quality result out of using a phone camera.  There is a very valid reason that most decent video camera costs more than your average smartphone, and the camera in a smart phone is a very small portion of the actual cost of the phone.

There have been projects where smartphones were used to produce professional results, but these have more been marketing gimmicks done by experienced professionals and it made their lives more challenging rather than easier.  Trying to get similar results without a lot of experience is going to be quite challenging.


Resize video while live streaming with ffmpeg

You wanted to resize your video (-vf scale="480:-1"), and at the same time you wanted don't to use a video codec for encoding such resized video (-vcodec copy).

It's impssible — video codec copy means no decoding, no encoding, no filtering - simply copying without any change. It is not a real codec.

Try omitting -vcodec copy to enable FFmpeg to choose an appropriate codec.


Am I increasing dynamic range by increasing the bit rate of video?

It won't increase dynamic range per se like using a 10-bit encoder would, but it will throw out less information in shadows and highlights when it compresses, so the extremes of the dynamic range will look better.


Convert fisheye videos to equirectangular videos

You can do that very easily in After Effects. Import your footage/images, create a new composition that matches your footage/images specifications and then apply Optics Compensation (Effect> Distort > Optics Compensation.
Tweak Field of View values until you have the desired result. In some cases you will need to check the Reverse Lens Distortion box.

Cheers!


ffmpeg operations yielding changing fps values

What the fps and tbr values indicate is the average and peak framerate detected within the file. When the container doesn't have valid metadata, either or both of these values can be wrong.

A tbr like 26.51 indicates a variable frame-rate file. After you extracted a segment with -c copy, ffmpeg will display the rates detected in the new clip. Since the source is VFR, the rate may not match but the individual frame timings won't have changed, and can't, since no frame operations can occur*.

When you transcoded instead, ffmpeg sets the output rate to the detected tbr and drops frames in regions with a greater framerate. (When saving to a constant rate muxer like MP4, it will clone frames if input rate is lower).

*true within this context


Android apk for (nearly) professional video edit

Is it the Adobe Premiere Clip apk you said you used, or is it Premiere on desktop?

For professional video edits it is recommended you use the desktop as most of the Video Editing apps come with tons of effects,fancy items for quick, easy use. I personally would recommend to try out Quik Video Editor over below mentioned Video Editors,when it comes to Android.

Quik Video Editor features:

Free
Photos and Videos limit - 75
Carefully designed Video transitions and graphics
Beat sync transitions
Output quality ( 1080p/720p)
Video styles (Raw/stylish )

The other Apps that you can use (Not recommended - but you can try)
FilmoraGo Video Editor
Viva Video editor
Kine Master
Magisto Video Editor & Maker (Second suggestion)
Adobe Premiere Clip

All of the aforementioned apps offer basic video-editing features. Some excel at social media integration, some are clearly designed to be a desktop replacement. Which one works best for you will depend on your needs which is "Professional video edit and quality output".

The best thing is that all of the apps mentioned are either completely free or offer a free version. Give them a try and see which is the best fit for you.

For a fact: "Since some apps are clearly designed to be a desktop replacement,we cannot expect desktop quality professional edits,outputs (codecs,fps) from the apps which come for the mobile devices"


Use FFmpeg:

ffmpeg -i input.mp4 -map 0 -c copy -disposition:a:0 0 -disposition:a:1 default output.mp4

Audio tracks (streams) are numbered from 0.

Supposing the input video has 2 audio streams, and the default audio stream is 0, we need to

remove the “default” disposition from it (by setting it to 0: -disposition:a:0 0) and
set it to the audio stream 1 (by setting it to “default”: -disposition:a:1 default).

In general, remove the “default” disposition from all audio streams except one, and set it for it.

Note:

Not all media players honor default audio stream — for example Media Player Classic don't honor it (but you may set the preference by the language of audio streams), while VLC Media Player and FFplay do.

Proof (note the location of the “(default)” tag):

ffprobe -hide_banner input.mp4

gives (among others)

Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 69 kb/s (default)
Metadata:
  handler_name    : SoundHandler
Stream #0:2(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 74 kb/s
Metadata:
  handler_name    : SoundHandler

while

ffprobe -hide_banner output.mp4

gives

Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 69 kb/s
Metadata:
  handler_name    : SoundHandler
Stream #0:2(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 74 kb/s (default)
Metadata:
  handler_name    : SoundHandler

The numbering here numbers all tracks, so #0:0 (omitted from the FFprobe outputs) is the first (video) track, and #0:1 and #0:2 are audio tracks.


How to do video streaming with a canon EOS M?

There's a couple ways I know of to do this that might work depending on what you need.  The first way would be to use an HDMI capture card like the Blackmagic Intensity.  The second way would be to use the Canon EOS utility over USB.


Audio Not Playing on Imported Video Files

Check for a couple things, as there's more than one thing to cause this. First, make sure sure in your preview preferences that audio is enabled. Also, check that the duration isn't set to zero (default should be 30s, set it appropriately). Then RAM Preview with 0 on the number pad. You can scrub through audio only in the timeline by ctrl+alt+dragging (Windows) or cmd+option+dragging (Mac) to double-check existence of audio. You can also preview audio only by choosing Composition > Preview > Audio Preview.

More detail can be found at these two links:

Adobe help document on previewing audio and video.

Adobe forum post about this issue.


Is there a way to find out if the current frame is a keyframe in Premiere Elements 10?

The Help specifies how to view keyframes.


Change output using a .jsx script for After Effects CS6

So, the answer to your question is 'yes, but...'

It's perfectly feasible, but you have to create an output module with the settings that you want, and then save it to your computer.  You can create this by going to Edit -> Templates -> Output Module.  Click on the 'New' button in the pop-up window, and then Edit it for the settings you need (quicktime, h.264, etc).  Save it as a useful name ("H264HighQuality" or something like that).  Save the output module, and you'll be able to access it from your scripts from there on out.

In order to use it in your script, just use:
app.project.renderQueue.items.add(myComp).applyTemplate("H264HighQuality");

As long as you've made the template first, you can access it in your scripts as many times as you want.  However, After Effects does NOT allow you to programmatically create Output Modules or alter them on the fly, unfortunately.  But you only need to make the template once -- as long as you have that template on the computer you're running the script, it should work just fine.


How to exit fullscreen monitor playback in Premiere?

As far as I am aware, there isn't a way to disable Mercury Transmit if the viewer covers the UI, unless you set up a key binding. You can assign a keyboard shortcut to the "Enable Transmit" function in the keyboard preferences (I use ctrl+shift+' as it's similar to the maximise window shortcut). You can then use this key combination to toggle the full screen display on and off.


Is video stabilization during recording better than stabilizing a shot in post-production?

I would decide by running some recording tests, and deciding which image you actually prefer - the on-board stabilisation, or doing it in post on the PC. There will be pros and cons both ways - although I wouldn't worry about things like CPU overhead, unless you've actually read somewhere of this being an issue?

This article gives a little bit of a run down of the stabilisation feature, if you haven't seen it.
https://havecamerawilltravel.com/gopro/gopro-hero5-video-stabilization/

Based on the article, it looks like the stabilisation is being done in the sensor. In a lot of cameras, the full sensor isn't used for recording, certainly not in all recording modes. For stabilisation, these cameras sample a larger area of sensor elements than what is being recorded, work out where your image has moved between frames, and record the area/window they consider to be your intended image, smoothing out your movements.

The article notes that you can't record a full resolution image in some modes when using the stabilisation, so it's almost certainly sampling the full sensor, and then stabilising by reducing the number of sensor elements being recorded, changing where the recording window is, based on your movement. Note with this method that the camera might not always stabilise in a way you consider to be "correct", and you'd have no control over that.

Stabilising with the PC will give you more control over what is done, but the PC software will also be interpolating (educated guessing) where required to create extra image data for your stabilised image. If you don't need the highest camera resolution, the stabilised visual data from the GoPro might be preferable to stabilisation corrections generated in PC software. But if you want to record in the highest resolutions, you'll be cropping images from the GoPro to do that, which may not be ideal - taking the unstabilised GoPro image and stabilising on a PC may be the better option for some resolutions.

Like I said, I would test both, and see what you think of the results for yourself. It's useful to know how both stabilisation techniques work, and other than that there's no correct answer for everyone - it will depend on exactly what you're recording, what quality it needs to be recorded in, and which method produces the best looking result to you when completed. If you don't need full resolution, and you're happy with how the stabilised GoPro image looks, I'd probably go with the camera stabilisation - but it was certainly noted in the article above that in some cases the stabilised image didn't look that great, so I'd definitely test it on what you're doing first.


How to live feed video from a Nikon D3200 to a Mac?

Since your camera supports NTSC and PAL video over HDMI, recording video from a mac won't be much of a problem.  You'll just need a small conversion box to get the video in.  For our macs with thunderbolt, we use an Ultra Studio Mini Recorder.  If you don't have thunderbolt, Grass Valley has a product that works over firewire.  Once the feed's connected, you can capture from within Final Cut.  Just open the import window (cmd-i) and you'll see a list of cameras at the top left.  In fact you can probably try this out now, because if your monitor has a camera in it, it'll be listed as a facetime camera.  Select the facetime camera or ultra studio or whatever your input source is, and click import.  When you're done, click stop import.  The recording will show up in your events browser.

As far as taking pictures from your mac, I know there's an app for Cannon cameras that lets you adjust settings, trigger the shutter, and download the image over USB.  Nikon probably has one too, but I don't know for sure.


What exactly is a "sample" in audio/video context?

If you are working in a video editing app like Premiere or Final Cut, when you specify a sample rate, you are specifying the audio sampling rate.  This is how many times per second the audio is going to be converted into a numeric value and stored.  So typically with video, it is 48,000 times per second (a frequency of 48kHz), or 44,100 = 44.1kHz. You need two samples to correctly measure a frequency. This means that the maximum frequency you can record in an uncompressed audio recording is usually just under half the sample rate.  This is called the Nyquist frequency.  So at a sample rate of 48kHz, you cannot correctly record any frequencies about 24kHz without hearing alias frequencies (incorrectly recorded lower frequencies appearing in error in your recording).
Pixels are not commonly referred to as samples, and a frame (whether defined as an audio frame in a file format, or as a frame of video) is different to a sample.
It would improve your question to say which program your code snippet comes from - someone might be able to help directly.
Also - bit depth is not the same as sampling - sampling is how often you measure something. Bit depth is how many numbers you use to record the measurement.  So most audio is recorded at 16, 24, or 32 bits.  So each sample in a 16 bit recording could be measured between zero and 65,536 amplitude levels.


Is it possible to speed up footage by more than 4x in Sony Vegas Movie Studio?

The only workaround I know of is to:

Speed up the footage by the desired amount (in this case 4X)
Render the clip.
Import the clip
Repeat steps 1-3 until satisfied.

This isn't ideal as repetitively rendering the clip repeatedly is fairly time consuming.


Although probably not best practice, this seems to do the trick:

Take a NUT file, same codec and dimensions;
ffmpeg -i nutfile1.nut -frames 2 -c:a copy -c:v copy -f nut nutheader.nut

And then a little piping hijinks:

nc 192.168.1.161 6600 | cat nutheader.nut - | ffmpeg -i pipe:0 -c:a copy -c:v copy -map 0:a -map 0:v -f fifo -fifo-format nut -drop_pkts_on_overflow 1 pipe:1 | nc -L -p 6601

You get a 'discontinuous DTS' when you connect the first time and headers are duplicated, but whatever.

And you get one 'packet damaged' when you start the standard input after the headers, but it immediately and with extremely low latency resynchronizes.


How is "definition" (SD, HD, UHD) actually defined?

The problem is that definitions such as HD, UHD and 4k etc. are partly hype. Marketers for AV equipment like to be rather elastic with the truth, hence the rather rubbery definitions. They're not standards such as ATSC or DVB, or graphic display resolutions such as VGA, WXGA etc., they're really just buzzwords.

There are other complications - like the different broadcast standards for SD — so that SD PAL is 576 visible lines and SD NTSC is 480 lines — or non-square pixels in digital video meaning that even a 16:9 HD 1080p image can be either 1440x1080 or 1920x1080*. So it's pretty understandable that you're confused.

For what it's worth, the general consensus is that:

SD refers to 576i / 480i (with the unusual EDTV variant being 576p and 480p). Display resolution is 768 × 576 PAL or 720 × 534 NTSC, often the actual pixel resolution is reduced, as in the case of DV: 720 × 576 PAL or 720 × 480 NTSC, with non-square pixels
HD refers to 720p and 1080i / 1080p. The display resolution is 1280 × 720 or 1920 × 1080, but sometimes the horizontal pixel resolution is reduced, e.g. HDV which is 1440 × 1080 with non-square pixels. You could also say that resolutions higher than this are also High Definition. So in one sense it's anything that isn't standard definition. Note that there is no 720i.
Full HD or FHD means the 1080 flavour of HD (either interlaced or progressive)
2K describes the Digital Cinema Initiative resolution of 2048 × 1080, this resolution is based on the historical resolution of scanned super 35mm film. As with 4K it is a little slippery, and can sometimes be used to describe FHD. As the wikipedia article states it describes content "having horizontal resolution on the order of 2,000 pixels". So in other words: near enough is good enough.
QHD or Quad HD - 2560 × 1440 is a whacky one, it's twice the resolution of 720p. The "quad" refers to there being four times the pixels of a 720p frame. Not to be confused with QFHD - see below.
UHD refers to 3840 × 2160, and like HD also sometimes to higher resolutions like 4K and UHD 8K
QFHD is 3840 x 2160 "Quad Full HD", the quad referring to the fact that it has four times the pixels of a FHD frame, which sounds better than saying it has twice the horizontal and vertical resolution. QFHD = UHD
UHD 4K - 3840 × 2160, is double the HD 1080 frame size, and is also sometimes called 4K, even though the horizontal resolution is less than 4000 pixels. UHD 4K = UHD
UHD 8K or FUHD is 7680 × 4320. I'm guessing that as the FUHD sets hit the shops they'll be called "8K" by the salespeople.
4K usually refers to Digital Cinema Initiatives 4K which is 4096 × 2160 but the term is sometimes used to refer to UHD 4K
Similarly 8K should properly refer to 8192 × 4320 but is also applied to FUHD

Note that there are cameras such as the Red cameras that shoot in-between resolutions such as 3K, 5K and 6K. But that's just for production, the finished media will usually be cropped / scaled to a more standard resolution for display or broadcast.

TL;DR If you're trying to define anything in technical terms and want to be unambiguous just give the pixel dimensions; terms like HD, UHD etc are really FUD. They're more for marketing than anything else.

Edit: Here's a handy chart:

pdf version available here


What deinterlacing filter should I use to encode/compress a PAL MPEG-2 576i 25fps raw file?

I don't agree that the other answer is a good idea in general, because "-vf yadif" will drop half the frames, thus resulting in a poor substitute of the original footage. Here is the thing: 576i25 means that you have 50 interlaced fields per second, rendering motion of objects fluid and natural to the human eye. You can get a high speed video at 50 frames per second easily by means of this:

ffmpeg -i in.mpg -vf yadif=1 out.mp4

Please note the difference, i.e. yadif=1 is specified. If you don't give a value, the yadif filter defaults to yadif=0 which cuts the frame rate of TV footage in 2.

Some people think that, because 25 fps is close to the theatrical movie speed (24p), the resulting video should look better, right?. Big mistake. The director of any project for TV had in mind that the target medium was TV and acted accordingly, using different techniques and taking a different artistic approach, allowing for instance a lot of fast movement, pan and zoom, or a hand-held camera to give a more subjective point of view. Also, a higher shutter speed is usual in TV without producing the nasty stroboscopic look because the frame rate is high. But later, when that footage gets converted to half frame rate the result is not "cinema look" but simply bad TV footage with stroboscopic movement.


The effect is known as a "difference" key. It's also known as a "not very good" key, because unless your footage is shot on a rock-solid locked off tripod and you're shooting uncompressed 4:4:4, it works a lot less well than you'd think.

After effects has a Colour Difference Key effect, as does Premiere, and this forum post explains how to do it in Apple Motion - though the principle will work in most compositing applications.

And of course, if you're after free as in speech as well as in beer Blender has a difference key node too.


-filter_script and -filter_complex_script simply read a filtergraph from a file, and there is no support for comments within filtergraphs.


Can a 1D LUT be used in ffmpeg using lutrgb?

The LUT filters don't have cross-referencing of channels. Use geq instead

geq=r='if(gt(g(X,Y),145),255,p(X,Y))':g='if(gt(g(X,Y),145),255,p(X,Y))':b='if(gt(g(X,Y),145),255,p(X,Y))'


GoPro Hero 5 Session camera for long shot (crowd shot)

Yes, useful for wide crowd shots.
Yes, you can crop down/scale up 1080p selections from the 4K to simulate additional camera angles.

Consider using the gopro app or a wifi remote to stop/start the gopro so you don't have to physically access the device while monitoring your main camera(s).

Not sure how long the event will run for but you might want to look at an external USB power source too.


How h265 settings / presets affects the target device cpu while decoding and how can I tweak them?

Try Encoding using "--tune fastdecode" and also check if "Average Bitrate" with "2 Pass" is better than "Constant Quality" with your particular Decoder; trying to maintain Quality over Bitrate can spike the Bitrate (holding a particular Bitrate will drop the Quality on complex Scenes).

Try a few Test Files (short Clips from your Source) and determine which Encode Settings produce the lowest CPU load when Decoding.

Decoding x265 works best with Hardware support which is not going to be available in most Phones, the Phone will probably decode x264 much easier since so many people/years have gone into decoding x264.

Reference: http://x265.readthedocs.io/en/default/presets.html

See also: http://www.techspot.com/article/1131-hevc-h256-enconding-playback/page4.html


TV trope name for monologue to unexpected off-camera recipient?
Theres a definite technique where the cameraman is focusing solely on character A who is speaking to character B, and character B is not within view for the TV viewer. Then, when it is revealed who person B is, it is completely surprising and often to comedic effect.

It is close to the Trick Dialogue, with a bit of Subverted Trope, not a Fourth Wall Psych.

Trick Dialogue:

A subversion of the most fundamental aspect of Dialogue: whether a character is talking to someone else or not.
Commonly, a character will appear to be making a brave confession, and then be revealed to have just been practicing before a mirror. Or, he will seem to be explaining his life to someone, and then be shown to be talking to a gravestone.
Alternatively, a character will clearly be talking to himself, but we find out that someone just happened to overhear. (Bonus points if it was the intended recipient of a difficult request.) This can follow on the first kind for a Double Subversion of dialogue.
Visual media typically use a Reveal Shot, often a Close-Up on Head, to execute Trick Dialogue. Compare Fourth Wall Psych (where the character appears to be speaking to the audience but is actually talking to another person). See also Not So Dire.

Subverted Trope:

Basically, this is playing bait and switchnote  with a trope. A work makes you think a trope is going to happen, but it doesn't.
But how could people know a trope is going to happen? Well, tropes live in the minds of the audience. As such, sufficiently Trope Savvy audience members can predict a familiar trope coming based on the hints dropped by the writer. So when the writer decides to build on this expectation, only to reveal that the expected "trope" was a Red Herring while an entirely different situation results, you have a Subverted Trope.


Hardware or software Gen-Lock for different actioncams

I'm not sure there is really a good answer to this problem without custom hardware.  General wireless products aren't really going to be a great option as wifi and bluetooth will both have variable latency issues and missed packets.  What you'd really want is something akin to flash triggers, but capable of firing repeatedly at your frame rate.  It would need to be a custom wireless solution for the problem though.

On the software side, you can try frame interpolation, but I'm not sure that the quality loss associated wouldn't be worse than the minor timing difference of not being gen-locked.  It still isn't going to match up exactly as it is a guess rather than what actually happened.

This is also ignoring the fact that rolling shutter greatly reduces the advantage of gen-lock to begin with.  The sensor itself is taking time to read the scene and so not even the top and bottom of the frame are truly in sync.  Genlocking is really something that more mattered for doing live switching where you needed to be able to keep a constant frame sequence and needed a frame to send.  It can also be useful in fine tuning audio/video sync, but that's pretty secondary to the technical needs to it, especially if you are shooting high framerate video.

That may actually be the real solution to your problem too.  If you shoot at a higher frame rate than your intended final product, you'll have frames much closer in timing to each other and can pick which frames to use.  This will work best if it's an even multiple, but pull downs may also work acceptably.  So if you shot 60 fps and were planning on going with 30, you could pick your frames to pull out to make sure you were at most half a frame off and on average be only 1/4 frame off.  That's a pretty tight sync without needing any weird magic.


Hacking up a podcast functionality with the Behringer XENYX 1202FX.

Why don't you do the music as a post.. Then you can just add the music track in the background and make it 'Duck' behind the voices. And if you really need the music to keep the tempo of the podcast conversation, you can just have it play on another device (iPod) and have it mixed into just the headphones, but not record it at all.. Or am I missing something?


Speed up a video 30x using only key frames with ffmpeg?

For a H264/5 stream, this can be done in two steps:

ffmpeg -discard nokey -i t1.mp4 -c copy t1.264

ffmpeg -r 30 -i t1.264 -c copy t1keys.mp4


What is the technique behind mkv. video format?

This question is a bit nonsensical as Matroska is only a container format, however I think it does bring up a deeper question on how video files work in general.  Matroska doesn't actually define the way anything inside it is encoded or encrypted, it just stores the information that says what was used on the content so it can be decoded.

While Matroska does support encrypted content streams, generally video and audio is encoded, not encrypted.  Encoding is when a format is used to store data in a more efficient manner than a simple representation of the data, such as MPEG2, h.264, etc.  Encryption is when a key is used to protect data from unauthorized access or tampering and uses algorithms like DES, 3DES, TwoFish, Blowfish or AES.  In the special cases where video is encrypted, it is also generally still encoded prior to encryption to make it a manageable file size.

Either way, it is kind of a moot point though, because matroska can use any of a large number of encoders and/or encryptors.  As an open standard, the details of the container format are publicly available on the Matroska website here.  More precisely, this page describes the different codec formats that are currently suppoted by the container.

As far as the differences between Matroska and other formats, there isn't any meaningful difference.  It is a different way of storing the information about the streams, but the streams themselves are exactly the same as if they were in an AVI, MOV or m4v file (provided they are encoded in a codec that is supported in those containers.)  Transitioning between containers with appropriate tools is a lossless operation as the media stream itself is not materially altered.


Blu-ray transcode to H.265 using Intel Quick Sync Video

Wrong container format (MP4)

I omitted the information about the container format. I used MP4 in preliminary tests in an effort to increase compatibility with portable devices.

It did not cross my mind that this would be the cause for the missing audio in PowerDVD 17 Ultra player. So the container has been changed to MKV. It did not, however, change anything on the artifacts in Media Player Classic - Home Cinema.

Probably buggy implementation of Native DXVA2 in Media Player Classic - Home Cinema

Thanks to Mulvya, we have found a work-around (not a solution). It seems to be buggy implementation of Native DXVA2 in Media Player Classic - Home Cinema. Switching to Video Mixing Renderer 9 (renderless) works well.


How can I extract clips from a streaming video feed in near real-time?

This is fairly straightforward if you save to a streaming format like Flash Video (FLV) or MPEG-Transport Stream (MPEG-TS).

Let's say this is your full feed capture command

ffmpeg -i source {-transcoding options} localfile.flv

You can then run

ffmpeg -ss 6:12 -i localfile.flv -t 1:05 -c copy extract.mp4

Ideally, the stream should be past 7:17 or close to it when you run the latter command. Note that since the 2nd command doesn't transcode, inpoint seek isn't precise. The extract will start from the nearest keyframe earlier to the specified inpoint.


How do I find out about the shutter speed of frames in a video?

If the data was recorded by the camera as EXIF MetaData, Adobe Bridge would reveal the camera settings under the Metadata Panel. You may have to customize within Bridge WHICH metadata you want to look at; because there are literally hundreds, thousands, of types of EXIF data; which could or could not have been embedded; AND at times; you will find what you need hidden in a strange EXIF container.

Simply put; if you don't see a shutter speed setting under MAIN, or CAMERA metadata; the brand camera/type, could have written it elsewhere; so show all; expand all, and scroll down to see if the data is possibly there; hidden in a collapsed field.

If you shot the video at 24fps; you are right in thinking that it likely was recorded at 1/48 / 180 shutter. However; I would think; just based off the engineering / camera / lens type which is built onto the iPhone 7; and how I would engineer the camera to work using, which it does by default-  auto-exposure; that it would rely on aperture priority for exposure; and would thus constantly adjust your shutter speed on the fly based on it's exposure method. So if you did a constant shot from a dark area to bright area; you'd literally see an on the fly adjustment from 1/48 all the way up to 1/4000 or higher. Then aperture might kick in, as the phone as no ND filters to fall back on.

I only add that because usually iPhone footage shot outdoors in bright sunlight looks like it was recorded at HIGH shutter (strobe effect) rather than iris down. The diameter on the lens of an iPhone 7 also would make me think they would use Aperture Priority (constant) as it's practically a pin-hole camera to begin with.

Cheers!


This is effectively a streaming media server tied to a live renderer.  Rather than save a recorded video, when you make a request for the video, it accesses a live stream.  That live stream uses the contents of the request to run a template video and encode in real time.  That real time stream is then played for the user.

You could probably manage to wire something like Red5 up to be able to do this with some alterations.  The big trick would be kicking off the live stream as part of the processes.  You also need sufficient rendering hardware to be able to generate a distinct live stream for each request.  You also need a real time video rendering system that can work with templates.  There is tech like this for video mixers, but I'm not sure of any particular software that does it in software in real time.  Some streaming oriented software may be able to do it though.


How to create circle gradient in Final Cut Pro?

Add a new custom generator and set the colour to white.

Add your media above. In this example, I will use a photo.
Apply the Draw Mask to your image.
Click to add 4 control points, then set the control point positions using the inspector.

Choose B-Spline as the Shape Type, then set the Feather and Falloff as appropriate.


Is CPU speed important for uploading a youtube video

Sending your files to YouTube (once rendered and or exported) will be down to the upload speed of your internet service provide (ISP) / Broadband supplier. The faster the upload speed, the faster it can get to the YouTube servers for processing.

So the answer to the question is no, but there are other factors involved like those mentioned above.


Windows 7 - multiple programs with conflicting sound output

Driver type ASIO Audio device ASIO4ALL v2

Aha, there we go. ASIO is designed to give only one program at a time access to the audio hardware. As I said, this is in principle a good thing, because it ensures the DAW can run trouble-free and without overly long latency or crippled sound introduced when the OS tries to combine the audio from different programs so some composite that's then usable for even the worse hardware that can be assumed. However, for just testing out the program, learning features etc. none of this is really a concern, so you might as well use one of the non-professional alternatives. Microsoft DirectSound should always work.


How to get a TV channel to stream as IP camera

I think I can see what you are asking, and Ken seems to have missed a big part of the question.

So assuming you are saying that the camera you want to watch is being broadcast as a TV channel that is connected to via an aerial (or equiv).  Then as you have said, you will need something to capture that with. That's going to depend a little on the hardware you are using with the capture device. But I am capturing local TV channels on a xbox usb tuner through a Ubuntu box.  But there are loads of low cost devices out there.

I am then using a bit of software called TVHeadEnd to grab that and spit it out as an IP stream. That can be watched on loads of devices. Of course if you want to watch this outside you home network, you are going to have to open ports on your router, and likely use something like DDNS (no-ip is a good provider for this) to find your router.  Just make sure what ever device you get to capture the broadcast and stream it, is compatible with the software you use to broadcast the stream.  Research before you purchase is recommended.

Lastly I think you will probably get away with running all this on something like a rasberry pi, depending on if you have to transcode the stream (which you would use FFMPEG to do, over an above most internal transcoders, but that's a whole different story).

There may be much simpler solutions out there, with everything just built in, perhaps a HDHomeRun, but I have not used one, worth a looks to see if it fits your need, it may remove the need for a PC all together.

Without more details this is all the info I can offer at the mo. I hope it's enough to get you started.


Most-efficient inexpensive way to speed/slow video track to sync with audio track?

Well you could download a trial 30 day copy of the legacy version of Adobe After Effects CS6. It allows a full unrestricted 30 day install trial.

From there, you could take your footage in, place the master clip in a composition, and instead of cutting it, use the the time warp feature, which will give you smooth keyframes where you can adjust up the speed to X, then place a keyframe if you notice it begin to slip, and set a keyframe to slow it down a bit. This will provide ramping, which basically "cuts" the clip in half over and over again, and does all the math for you, to make the smooth changes in speed. Then render it out. Premiere might also have that function- it's also included in the CS6 Production Premium Bundle, google the download link, and you can install and try for 30 days without purchase.