tracks show two different sampling rates. (...) Can anyone explain what this means? I always thought one audio track can only have one sampling rate. Is this just a different way of describing one sampling rate?

MediaInfo reports what is played by a decoder, depending of its capabilities:
- if your decoder is able to play an HE-AAC stream, the stream will be played at 48 kHz (the frequecy of the "core" + the SBR data).
- if your decoder is not able to play an HE-AAC stream, the stream will be played at 24 kHz (the frequency of the "core" AAC)

  I always thought one audio track can only have one sampling rate.

No. You can have a "core" stream with a specific sampling rate, decodable by old players, and an additionnal sub-stream in the stream with additionnal data for having 48 kHz for decoders capable of decoding such additional sub-stream.

Note that it is same for e.g. channels, with HE-AACv2 you'll get a mono "core" + PS for upgrading to stereo. or AC-3 (and DTS) can have a 5.1 "core" lossy stream for old decoders + extra data for newest decoder in order to get 7.1 and/or lossless.

  At the moment, the the software just expects single values for the meta data, so storing two different values for the sampling rate would make things more complicated.

If you are interested only in the "maximum" capability, don't care of the second value.
If you use the command line or DLL, you can say to MediaInfo to not show info about legacy players with --LegacyStreamDisplay=0 option.

Note : this is not "24-bit" as I read in another comment, bit depth in AAC is "dynamic" so no bit depth is shown for AAC (it would be shown for e.g. PCM). This output with several values is wanted by design (showing how an old decoder would decode the stream), and you can disable such behavior with the option provided.


Is there an online video server that can show paneable extremely large videos?

With current technology no, you "can not"; and with more powerful technology it is a waste of resources.

You do not need to stream a gigapixel video, you need to stream a video, receive the new "interest" zone, process the coordinates and display a new version of the video. Even arrange different chunks of different videos.

This is what google maps, bing maps or other applications do. They do not send you all the photos of the planet, just the ones you need.


Is there a way to pause and resume FFmpeg encoding?

Suspend

A simple method is to suspend it with ctrl+z. Or you could get the PID with pgrep ffmpeg then use kill -s SIGSTOP <PID> to suspend.

Then resume with fg command or kill -s SIGCONT <PID>.

Unfortunately this will not survive a reboot.

VM

If you use a virtual machine, with something like VirtualBox, you could perform your encoding in a guest VM. It will allow you to "save the machine state" at any time which can survive a reboot. It can also allow you to assign max CPU resources for the guest so your host will always have available resources.

Possibly overkill, but if you're familiar with VMs or using one already it is a possible solution.


How to *stream* audio over multiple scenes in Flash?

I faced a similar problem with a music video. Multiple scenes trying to keep every thing lined up and some lip scync toward the end.  Had a hell of a time trying to get it all to match up with the music- scene changes, action, and lip scync. As the audio was located in scene 1 I did not want to try to cut up the song and try to get it to play correctly over numberous scenes. since the audio disappears when you're working on a different scene it all becomes guesswork trying to estimate the number of frames required for every scene in order to come out where you need to be.
Then it accurred to me to use a line timer to solve the problem. Here is what I did and kept the scenes rather than moving at all to one timeline:

If you are using music in your animation, determine the beat of the audio.  lets say in this case 14 frames per beat (Use an audio editor to determine beat rate.)
Make sure your frame rate in flash is the same as your audio in the audio editor. My case was 29.97 fps
Create a small graphic lets say a red square. Have it appear every 14 frames for a duration of 2 frames.  Repeate pattern until you reach the end of the scene. Always
make sure you end the pattern with it one frame short of repeating the marker.  The rest of your scene layer/s should fall somewhere within this last timing pattern. Don't worry if they are not lined up yet.
Begin new scene. Start your timing pattern at the first frame with a timing marker. Repeat #3, as per need.  (Short cut: copy and paste timing layer using copy frames command by right clicking on layer. Note you will need to add 1 frame to the layer right after the first new timing mark per paste if your time line is longer than your intial paste & if you paste more than one time it will need to be done for each paste.) Saves a ton of time.

5 Remove extra timing marks. End as you would in #3.

6 Here is the tricky part and requirews a little thinking, but this will line every thing up and get it to run to the beat. Add or remove frames in the rest of your layers to the end of your marker frames but just before a new timeing marker appears.  in other words if your final timing marker in the scene is only 4 frames from the end of your current layer subtract the frames so the layer ends before the last timeing marker. so you now at some point need to add these frames back in.  You may be able to do this in the next scene. The way to keep track of this:

formula for tracking:
**

1. A# = added number of frames;
2. R# = removed number of frames;
3. NA# = need to add number of frames;
4. NR# = need to remove number of frames;

**

Sample of how it gets used:
**

1. N0  Scene 1 R4  NA4;
2. NA4 Scene 2 A6  NR2;
3. NR2 Scene 3 R10 NA8;
4. NA8 Scene 4 A8  NR0;

**

Okay here is the logic behind this chart. Starting at the begining of frame 1 scene 1 you are at 0 (N0)**You do not need to add frames to the beginning of this scene. If you remove 4 frames to get to where you need to be in association with your timing marks(R4)** Then you need to add them back in (NA4). Carry over the NA4 to the next line on the chart or scene (NA4 Scene 2). next you had to add 6 frames in scene 2 (A6) since you already needed to add 4 frames and you added 2 more than what you needed you now need to subract these 2 extra frames at some point Need to Remove 2 frames (NR2)
 hence (NA4 Scene 2 A6  NR2)
next line:
NR2 Scene 3 R10 NA8 Needed to remove 2 frames in scene 3, you removed 10 frames you now need to add 8 frames back in.
next line:
NA8 Scene 4 A8  NR0 needed to add 8 frames in scene 4, you added 8 frames which just happens to end at the right spot on the timing marker layer.   Now everything is synced up you can add additional scenes making sure that you end your scene as described in number 3 of these instructions.  Everything should sync up perfectly.  If you have any problems you may have to adjust one or two frames either direction in previous scenes to make up for the lag that flash has.
you always want to get back to a 0 point or (N0)
Also now that the time markers are blinking during your animation you can also add lineup/sync other animations that occur within the scene to the beat of the music.


How to record video locally during a video conference on a mac

I found Kazam amazing in Linux but it ain't mac app; hope the link Kazam Alternatives for Mac would help. Good luck


Custom timeline settings - how to use that?

That particular menu is only available on the cut page, AFAIK.  The BMD website is referring to the fact that as of R16, projects can contain multiple timelines with different resolutions.  In previous versions of Resolve, timeline resolution was set at the project level; you could have multiple timelines in a project, but they all had to be the same resolution.

But if all you need is that little menu, here are the steps to find it:

Switch to the "Cut" page.  It's icon looks like this:

Then click on the Icon in the upper-right side of the screen that looks like this:

This little window is really just a shortcut, which sets a few things in the project settings to commonly used values.  To put it another way, changing the "Timeline resolution" dropdown menu in Project Settings->Master Settings (gear menu) to 1920x1080, 3840x2160, etc, does exactly the same thing.

There are a few other places in Resolve's interface where you'll find shortcuts to the project settings, but all the magic really happens under the gear icon.  For example, when you create a new timeline via File->New Timeline, then click the "Custom Settings" button, and switch to the "Format" tab, you'll get a dialoge that looks like this:

Again, these settings are just shortcuts to the main project settings, and you can change everything besides timeline framerate after the fact (which actually is possible, but that's another topic.  Best to just get it right the first time).

And just for good measure, here's a screenshot of the master "Project Settings", which you'll find under the gear menu on the bottom right of the UI:


Renaming the files to .m2ts seemed to have solved the issue.


You can use the Bezier command that will help you a lot.


Why is my video file so large when there isn't any video?

You have 37 minutes and 14 seconds of audio, so it's going to be at least as large as the original audio file (36Mb) plus one image and a few bytes for each frame saying "the image has not changed". You have some 67,000 frames if you build a 30 fps video.
So it will look larger than the input audio file. It could be that the settings are such that the tool you're using adds full frames every now and then, like say once every second. These full frames are called key frames and allow one to start the video midway without having to replay everything from the start. So that could easily add a few Mb of data in a 37 minutes video. A 640x480 frame is about 1Mb uncompressed, probably about 0.01Mb once compressed (since it's just text). Your video has about 2,234 seconds, that would mean 0.01Mb x 2,234 = 22 Mb of key frames only and 36Mb + 22Mb = 58Mb.
Some tools can remove all the key frames. I've used ffmpeg and generate mp4 without key frames. I don't know that the tool you're using would be capable of doing such, though.


Efficient decoding of 10-bit 4k h264 from GH5S?

Nvdec and qsv do not have support for 10 bit AVC. And if there was a more “efficient” software decoder available, it would already be in ffmpeg.


It's a matter of semantics. Both audio and video are expressed as samples, where a continuous analog signal is encoded into discrete chunks. These chunks are expressed as numbers, and the numbers are composed of bytes, and the bytes are composed of bits. But saying that doesn't really give you any deeper understanding of the process.

Audio is encoded in samples where each sample has a magnitude, representing the sound pressure on the microphone at a particular time.

Video is also encoded as samples, where each sample is a representation of the amount of light that falls on a sensor in a certain period. Due to the fact that light is coloured, each sample is usually composed of three components, representing the three visible primaries, red, green and blue (often encoded as YUV, but that's beyond the scope here).

Unlike sound, more than one sample is recorded at a time; an array of them (a grid, aka a "raster") makes up a frame, which is the same as the picture a stills camera would record if it was fired at that particular time. By taking successive still pictures the camera is able to create the illusion of movement due to the persistence of vision. All of the pixels in each picture are recorded at (more or less, search for "rolling shutter" and "interlacing" if you want to dig deeper) the same time.


How to independently produce 3D Movies when 3D-TV's are discontinued?

Some are using VR headsets, some work with anaglyph glasses. With some plug-ins (such as the Dashwood 360 VR Toolbox, as one example), you can use popular headsets like the Oculus Rift. But many will convert the left and right eye to red/cyan and blend them so you can use those silly paper glasses from your childhood. Simple, but effective!


When making an animation how can I get a zooming effect?

Drag your composition (in the project window) onto the Make new Composition button, everything inside that first composition will be added as one layer in the new composition. Now you can just zoom this one layer!

(Say your original comp. is called 1. You drag 1 onto the button. Now there will be a New Composition comp. with one layer inside. This layer is called 1.)


How to change the resolution/size of iMovie so I can export square videos for Instagram?

You might crop each slide.  I haven't looked but if you have a lot of slide thee might be a way to set the crop as a 'default'


After Effects, can I parent specific points on a path to a null object?

Use the puppet tool to do this. Place puppet pins on the rectangle and they will allow you to distort the rectangle to follow them. You may need other puppet pins or starch to hold the rest of the rectangle in place.

Then connect their position to the null using an expression:

thisComp.layer("<the name of the null>").transform.position

or if the nulls are parented to something else:

thisComp.layer("<the name of the null>").toWorld(thisComp.layer("<the name of the null>").transform.anchorPoint)

EDIT Are you asking whether you can attach specific points on a vector path to a null? Not natively, but there is a third party plugin that lets you animate individual path points: http://aescripts.com/bao-mask-avenger/ Fairly pricey though.


After Effects: Strange motion tracking problem

some possibilities:

You are trying to track a very general area. Try to track a smaller
area with well differentiated colors (maybe another footage).
You are using CC 2014. There are bugs that have been solved since
then. Why don't you try to update your After Effects version to
2015.3.


Can i record continuously, keeping just the last N minutes?

Short answer: no.
Long answer, maybe. You have very limited scripting ability via the auto update function of the camera. You simply drop a text file called "autoexec.sh" on the SD card before turning the camera on and the camera will execute the script. Though I haven't seen a script that would do something like infinite recording by erasing the oldest part of the recording.

Maybe you are able to delete certain files from the SD card via the scripting interface, in that case you wouldn't be able to do seamless infinite recordings but you could record in N minute intervals and delete the oldest file after x recordings when you know that you probably have nearly used up the sdcard space.

You might also be able to do this by doing said interval recordings and have the GoPro's wifi webserver monitored by a computer that deletes the oldest file on the camera once you reach critical storage space.
In case deleting via the camera script interface isn't doable.

All needed info about this hack can be found here (horrible site but lots of information) and some existing scripts here.


Can I split audio before it gets to my sound recorder?

Technically if you want to be sure that you are getting the highest possible audio quality you need a pre-amp that you can use to split the signal. But the effect is going to be very small.

When you split a signal you are relying on the impedance of the two input devices being high, and the output impedance of the source being low, and everything being even across the frequency spectrum. Most modern equipment has high input impedance and low output impedance, so that's generally fine–you will get a couple of dB drop because you're splitting the signal across two outputs–but what could go wrong is that you will get e.q. shifts because the impedances might not be perfectly flat across the whole range of frequencies.

However it is unlikely to cause much of a problem for recording interviews, most likely it would only be a real issue if you were recording something like musical instruments. In an interview things like the room acoustics are going to have a far greater effect on the audio quality. Either way, a splitter cable is only going to cost a few bucks, so it's worth a test.


What codec should I edit in on Premiere Pro?

Premiere Pro supports the use of "proxies" that allow you to edit the video using low resolution files (and thus making edits very fast)

Once you finish, you export using the original file and thus the quality is what you expect.

It is impossible to explain proxies in a single answer. Search the official documentation or one of the many tutorials that exist on this topic.


How to change output filename in ffmpeg batch?

You can extract the file name directly from the for replacement parameter. No need to set a new variable at all.

for %f in (*.mp4) do ffmpeg -i "%f" -vn -ar 44100 -ac 1 -b:a 32k -f mp3 "%~nf.mp3"

If you want to do more complicated string munging, it's much more intuitive in Powershell:

ls *.mp4|foreach{
    ffmpeg -i $_ -vn -ar 44100 -ac 1 -b:a 32k -f mp3 $_.name.replace("mp4", "mp3")
}


What cable for 200' run between camera (video/audio) and ATEM switcher

You need to use cable specifically designed for SDI. RG59 essentially only defines the impedance and capacitance of the cable, not the return loss and attenuation characteristics. You can use reference charts from cable manufacturers like Belden and Gepco to select the right cable for your situation.


The answer to your question depends on the software you will use for editing.

Uncompressed RAW video is rarely used for editing for performance reasons. If you have actually recorded your media in 10-bit RAW format (some cameras can do that), then it can be subsequently used for processes that require as much colour information as possible (colour grading, for example).

Modern non-linear editors can be capable of working with H.264 (see Adobe Premiere and its Mercury engine), but you waste a lot of processing power that way.

At the time of writing, the most widely used codecs in the broadcast industry are:

Avid DNxHD
Apple ProRes HQ
AVC-intra

So depending on your operating system and software you may want to transcode to one of these codecs and then re-link your edit to the source H.264 files, to ensure that you get the maximum quality.

Additionally, if you don't have a powerful machine for video editing, you can convert your media to a low resolution codec (to make the edit with it), to later re-link the sequence to high-res media. This process is explained here


Adding date/time information on video for ffmpeg

I think adding:

-vf "drawtext=fontfile=/usr/share/fonts/dejavu/DejaVuSans-Bold.ttf: text='%{localtime\:%T}': fontcolor=white@0.8: x=7: y=700"

should work. See Adding time stamp overlay to video stream using ffmpeg


Are most scenes in movies voiced over in post production?

It's generally preferred to use the actual dialog audio from the take.  It matches the movement of the lips, it sounds right for the environment, the actor is giving an authentic performance, etc.  And, you don't have to pay the actor to come back for an extra day re-recording audio in a rented sound booth.

"Looping" in audio after a scene is shot is certainly not unheard of.  If a line needs to be changed, or there was an error on set, or there was some unavoidable noise going on during filming.  It'll also happen in very wide shots where it's hard to get a microphone close to an actor.  If this happens, it's called ADR -- Additional Dialog Recording, or sometimes Automatic Dialog Replacement.  Though as common as that latter definition of the acronym seems to be, the process really isn't particularly automated.

As for how to avoid needing it, you want to record clean sound from the start.  This is why you'll have a boom operator pointing a boom mic on a pole as close to the actor as possible -- even at the risk of accidentally getting the microphone in the shot.  If they kept the microphone far away, the actor would be quieter relative to all the other sounds going on in the room, so you'd have to crank up the gain and get more of those other sounds.  You can also have the actor wear a hidden lavalier microphone.  It's usually poking out near the collar close to their mouth, and then the wire runs inside the costume shirt so you can't see it.  Occasionally if the character has a big hat or helmet, a lav mic can be hidden in there instead of by their neck.

While you are in the location, you also record some "room tone."  Whatever the room sounds like when it's quiet -- because no place is actually 100% quiet.  There's some wind, or the neighbor's air conditioner or the house creaks slightly.  Whatever.  Get a few minutes of that.  And, if you have any crowd or whatever, you can get some of them talking or walking, or whatever it is that the other people are doing while the main characters are talking about the plot.

Then, when you get to editing the scene, you have a track of that "silent" room tone as a background noise.  And you have you crowd noise.  Maybe you use some actual recordings of the crowd from the set, maybe you just wind up using some stock sound, or do a separate audio recording session.  If it's a cafe or something, you'll have some clinking cups and dishes and silverware, and some babble of people "talking" but you want to make sure that it's not any recognizable words that would catch your ear while you are supposed to be paying attention to what the main characters are saying.  That can be just two or three tracks, or a whole bunch of tracks.  If you are on a space ship, you may have a bunch of tracks of engine sounds and beeping sensors and whatnot.  This bed of sound is constant through your scene, and is pretty much independent of the main character dialog that you get from the actual editing.  (Unless the sensors need to beep extra loud when the aliens approach or whatever, that sort of thing obviously needs to be in sync with the video.)

With that quiet but consistent bed of audio, you can cut the scene and get the best takes, and use the dialog audio from the relevant takes without to much trouble.  You don't notice any slight shifts in the audio from cut to cut because there is enough other audio that's consistent in that quiet background noise that it doesn't matter.


Preventing flicker when recording live presentations

The reason why this happen has to do with synchronization, or rather, lack there-of.

The projector displays its video-stream at one frame-rate and starting at one point, while your Canon records at a perhaps different frame-rate and starting when the projector image maybe has drawn part of the image.

The ideal approach is to use a common sync - in this case that is very unlikely to help in any case as the Canon don't have a sync input and the projector probably doesn't have one either.

To reduce the problem you need to make sure as a minimum that the Canon and the projector work on the same frame-rate. Ask someone or ask to look at the projector if possible. The start-point (for when the image lines start to draw) is worse - here you need to cross fingers and hope that when you turn on your camera that it will sync with the vertical blanking. As a rule of thumb: use 60i or 30p fps in America and Japan, 50i or 25p in Europe. However, some projectors can show at an internal rate rather than the one based on the electrical system.

Secondly, lower shutter-time (slower) and of course increase the f-stop (add ND filter if necessary). This will reduce the flicker but you can still get "bands" in the video. It will also be more sensitive to movement so here you need to balance/compromise. Too low and you get a strobo-effect instead.

Also try progressive record mode to see if that helps.

However, the best option in cases such as these is to ask for a copy of the presentation file so you can grab stills/video from that in post and super-impose those on the video, or cut to the presentation when editing.

Ask in advance so you can rig your camera in a better position to capture the presenter rather then the screen, but keep track of where in the presentation he is so you get the correct "slide" in post.


ffmpeg with this filterchain does it:

ffmpeg -i input -vf "tblend=average,framestep=2,tblend=average,framestep=2,setpts=0.25*PTS" -r srcfps -{encoding parameters} output

srcfps should be replaced with your input's framerate.


You will generally get the following guidance if you take courses on presentation skills (this list from write-out-loud.com)

Slow speech is usually regarded as less than 110 wpm, or words per minute.
Conversational speech generally falls between 120 wpm at the slow end, to 150 - 200 wpm in the fast range.
People who read books for radio or podcasts are often asked to speak at 150-160 wpm.
Auctioneers or commentators who practice speed speech are usually in the 250 to 400 wpm range.

So you'll probably want to aim for that 150 wpm range to be able to get a reasonable amount across, but not sound like an auctioneer


Noise (from autofocus?) during Video on DSLR

The noise you talk about is usually caused by lens motors related to autofocus and motors, related to image stabilization. The best you can do is to use external microphone and connect it to microphone port of the camera. This will give you better isolation of the recorded sound from camera noises and better sound quality.

Of course you can always use external sound recorder and sync video and sound in post.

EDIT: to avoid the noise from focus motors you can use manual focus. To avoid noise from stabilization motors (if any) you can switch off image stabilization (VR, OS, IS or whatever it is named) and use rig with stabilization or tripod.


Stream from several android phones to pc

You can use streaming apps for android which transmit the screen to a desktop-machine from which you can capture the image. I wouldn't recommend it though, as the aforementioned streaming will make your image-quality suffer big-time.

Let me propose a different solution:
using the cable of the phone in question, you can directly capture the image from the device. The only issue would be bringing the recordings in sync (if that's important). To fix this, you could open up an online clock for example which displays the exact time, so when stitching together the recordings in post, you'll be able to synchronize the recordings.


Scaling video lowers quality in Premiere Pro

Why this is happening: because an interpolation method the Premiere uses for scaling kills high frequency details (ie., sharp edges, as text or some thin elements of the GUI are).

A scaling interpolation method (in our case, Lanczos method, as an Adobe employee mentions here) is an algorithm used to map an array of pixels of one size into an array of a different size. You have a 200x200 image and want it to be 400x400. What do you do? Quadruple each pixel of an original image so it will take 4 pixels in an upscaled image? Or blend the neighboring ones somehow?

There are a couple of algorithms developed, like Nearest Neighbor, Bilinear, Bicubic and Lanczos. They differ in computational complexity and in the quality of the results they produce. And the quality (here we finally come to your question) also depends of the nature of the source material. Adobe chose Lanczos because it does a decent job at scaling natural images taken by camera. If this was an image of a person, you would expect that a scaled down version will not have the individual hairs and the little marks on the skin to be seen in the same detail as they are seen on an original larger image.

Yet, you are scaling down a different kind of footage – the screen capture of a software GUI. The one you'd want to see all the sharp details even on a scaled down image. As you see, the Lanczos method does a bad job here.

And we come your second question: what algorithm would do a better job, or how to fix this?

Short answer: in Premiere, you can't.
Long answer: a scaling algorithm we need is a Nearest Neighbor algorithm. This one preserves the sharp edges of the GUI just as we like it. Adobe After Effects has it (called "Draft" scaling quality). Premiere doesn't (natively). You can try your luck searching for "Nearest Neighbor scaling plugin" for Premiere.


How do I crop a video with a particular point of the video always in the centre of the frame?

The action you're trying to perform is called “stabilization,” which is usually performed by analyzing the entire frame when the goal is to eliminate camera shake, but when you want a single object to stay in place, you'd track a single point, or small area of the target, and apply the same counter-movement that you would to eliminate shake.
In AE, point trackers are under the animation menu→Track Motion.  After you analyze the subject, click the “Stabilize Motion” button on the “Tracker” tab.
If you don't have AE, the free version of DaVinci Resolve can do the same thing, and there are several ways to do it.
Either:

On the Color Page, go to the “Window” tab, add a shape to track, then switch to the “Tracker” tab, select “Stabilizer” from the dropdown, and click “Stabilize.”

or

On the Fusion Page, hit shift-space to call the select tool menu, and start typing “tracker.”
Click OK, hold Shift and drag this new node onto the line connecting “Media In” to “Media Out.”
Move the green box that appears over the area of the footage you want to track, click the “Track Forward” button.
When it finishes, switch to the “Operation” tab and select “Match Move” from the dropdown.

The Fusion method won't zoom in on the result by default, so you'll see black boarders around the edges as the footage moves to compensate, but you can easily zoom in on the edit page, or do it “properly” in fusion by adding a Transform node, and scaling there.


Difference between duration_ts and duration in ffprobe output?

duration is in seconds, e.g. 15.9
duration_ts is in the media's time_base, e.g. 1/30014 * 479000 = 15.959219


Build linux render farm in order to render adobe premiere projects

You can do this by using ffmpeg. Here are the steps:

First do steps of the second method ("Use a frameserver") here.
Beginning this step you should be streaming your Premier timeline through Debugmode and Avisynth on a specific IP and port. Now open a command line and run this ffmpeg command:

ffmpeg -i frameserver.avs -f mpegts  tcp://[IP address of your server]:[open port on your server]

By running this command, you are sending the Avisynth output (streaming) to a destination machine (your Linux server) through ffmpeg.
After this step all you have to do on the client server (where you edit your videos) is done. Next, set up the server side.

On the server (your Linux server), make sure you have ffmpeg installed and run the following command:
ffmpeg -i tcp://[your server IP]:[The same port you entered in step 2]?listen -c:v libx264 -preset medium -crf 23 -pix_fmt yuv420p -c:a libfdk_aac -vbr 4 output2.mp4

Through this command, ffmpeg will be listening on a specified IP and port and receiving sent packets and encode them with your favorite options!


There is no professional tool that does this, as far as I know. It is never done, presumably because it would not be good enough and would need to be corrected by a human anyway.

There are of course many subtitling applications into which you can import your transcript and then do the spotting by hand. And they have varying features to make it easier/faster/more precise.

However, Youtube does offer automatic timing of transcripts. You may want to try it.

Using that, you could:

upload your video to youtube
upload your transcript
wait for their software to do the automatic timing
download the youtube captions
... then probably still do manual corrections.


How to set up streaming from two webcams or stereo camera (2x lens) to VR headset with minimal latency?

Wireless will always give you some latency (if you're not spending a fortune). I would recommend you try using sdi cables. In video-production, sdi is often used for immediate feedback to the directors monitor etc. - so you should have less of a problem with latency there.


There are several factors that comes to play with recording computer screens.

As filzilla already mentions, sync is a major factor. In the old days CRT monitors was very challenging to film even with sync-locking. Depending on if you filmed on silver or on video you did get different results. With the exception of LED screens, LCDs also offer sync problems.
The other factor is light conditions. For movie scene you usually have a large amount of light which a computer screen is unable to produce - so the screen turns very dark. You also get problems with color-correction/white-balance as the monitor typically have a different color temperature than the lights used to lit the scene.
The third factor is of more recent nature and has to do with the nature of LCD screen and it's polarization, referring to the limited angle they are visible within. You will get "gradient" effects and on more poor screens also have limited visibility from the sides, top and bottom.
Moiré can be a problem. For video this isn't such a big problem because the light coming from the screen will blend the pixels. Unless you have a high resolution camera filming in low(er) light conditions, or are filming close-up (but not so close that you can actually see the screen pixels), the camera won't be able to separate the individual pixels. For movie cameras this is more present though due the higher resolutions and the requirement to light conditions.
Then there is content synchronization with the actor's interactions or the scene in general, the potential of typos, wrong clicks, having to reset for each take and so forth.

It's more easy to put in a video in post which you can color correct and adjust, synchronize and so forth to the movie.

The process isn't very complicated. The computer screen is often recorded with a green or blue backdrop to eliminate the need for mask/matte rotoscoping. You just track the corners of the inner part of the screen and drop in a keyed replacement for the screen.

There are plugins too that do a pretty decent job automating this process (or instance the Mocha AE and its planar tracking).


How long should the maximum length of an exr sequence be?

Maybe check the frame after it stops loading, it could be a corrupt file that causes After Effects to stop loading the rest of the files.


How to create film sounds like those booms?

A boom could be done a zillion ways. If you have a composer and a full orchestra: Timpani coupled with several double basses playing a quick bowed low note or longer depending on how long you want the boom. A "boom" could be interpreted many ways including collecting real sounds and in post production lower the pitch and cut the sound envelope to make a fast attack. For example, you record a washing machine just before it finishes the spin cycle take the track and cut it right before the final thud, drop the pitch to make it deeper.

The key here is to get a low percussive sound which has the following characteristics: loud, very low pitches, fast attack, and a natural release, even adding a bit of Hall reverb or delay or both will help sweeten up a bit.

Natural boom sounds might be found in thunder, earthquakes, landslides, rock slides, a large tree falling over, etc.

Man made boom sounds can be found in jets breaking the sound barrier, cannons, fireworks, a large truck in a collision, a train wreck, etc.

Alternately, if you have a friend with a synthesizer you might explore some of the pre-programmed samples such as timpani, orchestra hit, Taiko drums, etc.

What I have mentioned are the basic ingredients, stir well, and season to taste. In other words you will likely want to mix a number of sounds to get it just the way you want it.


Adobe Premiere not reading audio from .MTS file

Apparently its a problem with Premiere pro Cs6 that haven't been updated or fixed as yet there are 3 ways that might help you resolve your work;

Install an earlier version of premiere pro like 4 - 5 depending what version fits what you do......(DO NOT UPDATE!!!)
Convert all your videos to mp4 720 or 1080, what every quality your working with. Might be a bit time consuming but it works.
Import the ENTIRE CARD, very important, all the files from your card to your pc/mac, the metadata should be located somewhere in a folder within the card, it's not on the .mts file.

Method 3. has been reported to work so try it and let me know.
installing new codecs wont work.


ffmpeg - trascoding multiple streams from stdin errors out: pipe:: Invalid data found when processing input

I fixed your command

       cat a2.mkv | ffmpeg -nostdin -i - -y -map 0:0 -c:0 copy -f matroska -y 0.mkv -map 0:0 -vf "select=gt(scene\\,0.4)" -frames:v 1 -vsync vfr -vf fps=fps=1/60 -codec png -f image2pipe 0.png


Is there any use in archiving graded 8-bit footage as 10-bit file?

Yes, you will lose information from the grade if you save as 8-bit.

Here's a thought experiment to illustrate. Say your original red level for a pixel was 128 out of 256 (00001111 / 11111111), and then in Resolve you grade it (using the internal 32-bit floating point colour in Resolve) to be a little bit brighter, say 128.25. Now you'll need to save it in 10 bits  or more to keep that information (e.g 000111101 / 1111111111 where the two right-hand-side bits 01 / 11 = 1/4 = 0.25). If you saved it as 8-bit it would have to be rounded to 00001111, meaning it would go back to being 128.

The upshot is, if your new colours don't fit into the 256 values per channel of 8-bit colour they will get rounded to the closest fit. So while you don't gain anything from up-depthing (that's a thing now, because I just invented it) ungraded footage, you do gain by not collapsing the higher-bit-depth graded footage back down to 8-bit.


As long as the HDMI output is configured properly, it should be the same as what is recorded.  I'm not sure about that particular model, but most higher end video cameras support clean HDMI or SDI output.  A clean output means it is free of any of the on screen displays.  They may also be capable of outputting different resolutions between the HDMI/SDI outputs and the recording.

As long as the HDMI output is configured to match the settings of the recording, then it should be the same signal.  Since it is digital, as long as the signal isn't dropping bits of data entirely, there won't be the traditional type of quality loss from analog transmissions, but you'll want to make sure the HDMI run isn't longer than necessary.


How to configure crtmpserver for live streaming?

I'm pretty sure you can use vlc to both stream and restream it. I use vlc to stream all my tv channels to all my devices(rtsp, mp4, html5).

Search for vlc media player streaming like

https://www.videolan.org/doc/streaming-howto/en/ch04.html


Cool Edit Pro is now Adobe Audition. They have started a beta, so you're in luck.

http://labs.adobe.com/technologies/audition/


Assuming an irregular shape, here's a manual method to do it.

Create a solid layer of any color, stacked above the shape layer. Change its opacity to around 30%, enough to see the shape layer underneath. Unlink the scale controllers (X and Y) so that they can be scaled independently. Move and rescale the solid so that it can serve as a bounding box for the shape layer. Your original solid size should have been the same as your comp size, so the dimensions of your shape layer are

Shape width = Composition width * 'X' scale % of solid layer
Shape height = Composition height * 'Y' scale % of solid layer

Of course, this is a crude method for one-time polling. There's probably an expression for dynamically updated values if the shape is animated.


Easiest way to create a continuous shot effect using multiple cameras?

CG is probably the best bet.  Motion controlled cameras would be the second best bet, but it would still be pretty tricky as the real world has a remarkable ability to change subtly, but substantially enough to result in a jump cut pretty quickly.  Even if you could manage to stop the camera on something that is static, resuming the movement at the same speed would be difficult at best without a computer controlled camera.

If you are able to have the camera stop movement on scenery in between, that might make a possible stop point, but it would still have to be done pretty quickly to avoid changes in lighting conditions that would be noticeable in the final shot.


Lossy codecs such as x264 degrade the image quality by design, if you don't want to lose image quality you either have to use a lossless codec or better yet, not re-encode at all.

I assume the source material is either H.264 or MPEG-2, so the best solution in your case would be to simply copy all the streams into the new container:

ffmpeg -i input.flv -c copy output.mp4

Here's the command for using x264's lossless mode if you have to re-encode for whatever reason:

ffmpeg -i input.flv -c:a copy -c:v libx264 -crf 0 output.mp4


What's a good, cheap, piece of gear to start with when adding lighting equpment to your toolset?

Lighting is not trivial and deserves attention and training to master. I certainly have lots to learn even after more than 25 years with working with studio lighting. There is no one thing you can buy that will work in every situation beyond understanding and mastering white balance and exposure. Nothing to invest in on this step. However, if you want to illuminate an outdoor subject that is silhouetted, then a round reflector held by a camera assistant will make a huge difference and this won't cost you much, under $75. Here you won't have to worry as much about color temperature as you are reflecting the same light as the sun provides to cast a fill in light on the subject.

For most studio lighting, you need to learn the basics of 3 point lighting. Here are three Vimeo School tutorials on basic lighting that I recommend you review before buying or making anything:

http://vimeo.com/videoschool/archive/lighting

While I shoot in my studio, I am generally shooting 1:12 scale subjects so my lighting can be somewhat small. I have two 100 watt tungsten lamps with big reflectors (available at Ace Hardware for about $12 each) mounted on vertical rails on my bench.

I have metal clips holding plastic diffusers made out of discarded computer plotter opaque plastic. I also have an array of smaller 40 watt tungsten lamps that I can place in strategic areas such as under a lift frame that has a diffuser on top for under neath lighting.


Pan & zoom from still image with green screen

If you want parallax, you can do it without a video, a still image of the phone could work, and would make things a lot easier and faster.

If you want a video you need to add markers to the green screen so the program can track them.

You could also have just simple moves, for example, your object only rotating, so you do not need to track the background. Some simulated moves for the new background on post could be enough.

For the DOF you can potentially add one problem if you want the DOF on your product. It is better if you film your product very sharply, so the green screen cuts nicely. The DOF would be a nightmare.

Then using a real 3D program you can add your video on a layer, track the green screen, generate a 3D scene, and replace the green screen with another plane with a photo.

I would use Blender for that. But if you want simple moves and not a complex 3D tracking you could go away with HitFilm, which has 3D planes too.


I use a script for After Effects (CompsFromSpreadsheet) which generates compositions using the data from a spreadsheet. We use it to batch create graphics with different values - like in the ad.

The script can add different files into the animation at different times, so they probably have a database of all the possible phrases.

The script as it stands couldnt be used to fully automate the process as you still have to specify where the spreadsheet is, but you could easily create a simmilar script which would look for new spreadsheets, apply the data to a template and start a render.

That being said, I wouldn't be surprised if they had their own in-house software.


Kdenlive composite effect not stretching

I had the same issue.  Use the "Composite and Transform" transition instead.


How does YouTube encode my uploads and what codec should I use to upload?

Some general info about the formats used:
YouTube uses 4 container formats and 4 different codecs. It depends on the popularity of the video what codecs are used for your video (see below why). Generally, every of your uploaded videos will be encoded in h.264 and will be muxed into an .flv and .mp4 container. That's the standard and this will happen for every video. Though .flv will only be used for resolutions below 720p. Meaning only 360p and 480p will exist in an .flv container. Though every video below 720p will also have a mp4 version available. For 240p YouTube is also using 3gp which is a rather old codec (based on MPEG-4 Part 2 (not to be confused with MPEG4 Part 10 aka h.264) meant for mobile devices (long before the smartphone era), it comes in the container .3gpp.
The other codec used is VP8 which is coming in the WebM container format. WebM is a format developed by Google and was meant as a standard video codec for HTML5, support for it is by now pretty good in most modern browsers.
WebM was introduced with the HTML5 version of YouTube. YouTube only encodes some videos in WebM after they got uploaded and mostly only popular videos (going by the videos I saw encoded in WebM) so it's not certain your video will be present in WebM. Though this is changing with WebM gaining some more support.
YouTube also supports VP8/VP9 in the WebM container. This is supported by all major browsers for several years now. With WebM, they also introduced support for the Opus audio codec (besides AAC which is used in all other containers).
In regard of your linked question (maybe they should be merged?)
What codec/container should you use for uploading?
That depends, if you are limited/concerned by your upload speed then use h.264 Level 3.1/4.1 with Main Profile for SD or High Profile for HD. YouTube will accept this just fine and it will look nice after it got encoded by the YouTube servers. I would usually recommend a bitrate of ~4-5 Mbit/s for 720p material and ~8-9 Mbit/s for 1080p. For 4k go for ~15-25Mbit/s or more depending on the complexity of the footage. "Cartoon" content with few colours can usually have a much lower bitrate in general.
This is usually a good balance between size and quality. If you want better quality go with a higher bitrate, and if you want a smaller video go with a lower bitrate.
But be aware that YouTube will ALWAYS encode your video once it's uploaded, doesn't matter what codec and settings you use. So if you want the theoretically best quality for your uploads, choose a lossless codec for uploading or at least visually lossless.
See YouTube as the final output into a delivery/consumer format and the upload to YouTube is the last step in production and during production you want to stay lossless.
But please note this is all just a theoretical thing, practically I'd say it really doesn't matter as we are talking about YouTube and not about TV broadcast or Cinema. In almost all cases a very high bitrate h264 encode (20-40mbit/s) will suffice even for high-end content.
But if you really want to do it the "perfect" way, use a production codec and not a consumer codec like h.264.
MJPEG would be a good codec for that, YouTube definitely supports that in an .avi or .mov container. MJPEG is a lossy codec but the visual quality will be the same as the source (if you chose a high enough quality setting, this is practically JPEG as the video codec).
Going with a real lossless codec would be a waste of hard-drive space and bandwidth in my opinion.
But if you want to upload your video truly lossless and don't care about the upload time, I recommend using a standard QuickTime codec as they should be nearly all supported by YouTube (note that not all of them are lossless, h264 is also a standard QuickTime codec). Though YouTube doesn't state which QuickTime codecs are supported, unfortunately. Animation or JPEG2000 should work I guess. Both codecs can be 100% lossless depending on settings in case of JPEG2000.
When it comes to frame rates, if you can choose then use 25FPS or 30FPS  during recording/animating (by now YouTube also supports 50 and 60FPS+ video, so use that if your content benefits from a higher framerate), YouTube likes standard framerates like 25 or 30FPS the most (or again one of the higher FPS options) but if your footage comes already in another frame rate like 24FPS then stick with that and don't interpolate up or down. YouTube will handle the conversion for you and generally does that better then your average encoder. They have to deal with all kind of framerates every day and solved this (actually very complicated) issue really well.
Audio:
For the audio use PCM if you want to stay lossless with the audio as well but again this is really just a theoretical quality improvement. AAC will usually do the same job quality wise (subjective) and will be smaller. I recommend a bitrate of at 128-192kbps for AAC. The size impact is usually not as big as the video codec so you can also just go with 192kbps or higher if you feel your audio content is very sensitive. YouTube will convert audio to ~24kbps(Mobile aka 3gp) ~64kbps(240p), ~128kbps(360p/480p) and ~192kbps(720p+) using the AAC and Opus (WebM only) codec.


Replace the video edit with another video keeping the in and out timings

A possible workaround:

Open the new clip in its own project.
Set the "in" point and the "out" point.
Export that as a new clip (with lossless video quality, if you can pull that off).
Open your other project.
Use Replace Clip from Bin, and replace it with the clip you just exported.  In the new clip, the "in" point is the beginning of the clip!


What is/are the best free tool/tools for editing my videos?

You can download a FREE 30 trail version of Vegas Pro 11 from Sony.

http://www.sonycreativesoftware.com/download/trials/vegaspro

I've have Vegas Pro 10 and I have used these features: Stabilize, Convert resolution down to 720p, and convert (render) to MP4.

I never convert resolution up, it just looks terrible (see my comments above).

By "Crop to Anamorphic", I am assuming you mean to convert 4:3 to 16:9, I have not tried this with Vegas as all my originals are already 16:9 HD. So you might want to google for that part.

If you go this route search YouTube for tutorials on each item as it will speed up your learning curve considerably.

Caution: if you want to take a 4:3 SD video and make it into a 720p 16:9 HD it's just gonna look terrible. Why? Because you are asking the software to expand the resolution, this never looks good. It's ok to reduce resolution say go from 1920 X 1080 and step it down to 720p, that's just fine, but when you try to go the other way you are asking the software to essentially make up material that is not there.


Techniques for compressing a slideshow presentation video?

x264 is the best-in-class encoder for generating H.264 video streams and you can use it via ffmpeg, a command-line tool, to encode to a small-sized file. Get a binary from here - opt for the latest nightly/snapshot release - and run

ffmpeg -i input.mov -c:v libx264 -crf 23 output.mov

The output will be the same constant frame rate as the input but x264 is very good at minimizing space consumed by frames which are the same as preceding ones. Lower CRF values produce better quality files at the cost of larger file sizes. 18 to 28 is a decent range to stick within.


Is it possible to capture video signal waveform as audio for video replay?

The short answer to your question is "no".  The sample rate of video is typically measured in MHz or GHz.  If you have a 1080p24 signal at 8 bits per pixel, the actual frame is 2750 x 1125 pixels (according to digital test equipment manufacturer Kramer).  Multiplied by a 24 fps refresh rate, this results in 74,250,000 pixels / second.  Multiplied by 10 bits per clock tick per pixel, this is a bandwidth of 0.74 Gbps.  As a "sample rate", that is approximately 1000x higher than 16-bit at 48K.  Hence the answer "no".


H.264 slow scrubbing and playback performance

Usually in these situations I'd say maybe you have a bottleneck from your storage media. But with those bit rates any Ssd or hdd should be fine unless the files are really large. Thus it sounds to me that you have either a bottleneck with your GPU, Cpu, or Ram. All are equally important for smooth playback. I'm guessing the issue is the increased resolution of the second clip.

Without knowing more about your specs I would say your best options are to create low res Proxies. In premiere it's pretty easy, you just have to modify your project ingest settings. You might as well create Proxies for all the files, it won't matter anyways as it will relink on export.


How can I convert frame rates without changing speed?

In order to convert frame rates without impacting the speed of playback, you have to remove frames to decrease the frame rate or create new frames to speed it up.  If the frame rate is an even multiple, then this process can be done losslessly for reduction in frame rate since you simply drop every unneeded frame.

If they are not multiples or you are increasing the frame rate, then it is necessary to interpolate new frames.  Interpolation is a process by which the computer looks at the next and previous frames and invents one that looks like it would fit between them.

If you shoot at 60fps and drop it to 24fps, it is not an even multiple and frames have to be interpolated or taken at the wrong time in order to make up for the difference.  This results in a lower quality of video than shooting in 24fps to start.


looking for some advise on setting up a sports event livestream with local playback

Yes, there will be problems trying to stream the live-image from a dslr. Usually, you would use a multicam-setup for a livestream such as this, including scenes for intermissions, transitions, etc.

Normally, you would use a mixer like the ones from black-magic, here's a link to it. Using it, you would want to feed an sdi signal from the camera into the mixer and send the image from the mixer to a distributer (like a pc or laptop) which will then be able to stream the image.

Unfortunately, you can't just plug a gopro into your pc and press "stream image".


Is there any method to create an audio visualizer as seen in video below

Audio Spectrum effect that comes with After Effects. Just apply the effect to a solid layer and there will be a dropdown of the layers in the current composition, select the audio layer.


Create video file with multiple audio tracks?

FFMPEG can do it. FFMPEG is command line video (and audio) processing software, available here.

If you have FFMPEG here is the command you would use:

ffmpeg -i video.avi -i audio1.mp3 -i audio2.mp3 -map 0 -map 1 -map 2 -codec copy output


make a stroke from a path in after effect

There are many ways to do this - here is one:

Make a null object
Make a mask path on the null layer using the pen
Set a single keyframe on the mask's path
Go to your object (e.g. a star in this example)
Make a single keyframe on its position
Copy your mask path keyframe
Select your position keyframe
Paste, and you should see your object now moves along the path

To make a tail that follows the star:

Add a new shape layer with a single line in it
Make a keyframe on that line's path
Copy the path from your null
Paste it into your new shape layer.
Now you will have a line the same shape as your star's path.
Add a Trim Paths effect to your Line layer
Animate the 'End' control on the Trim Paths effect to get the path you need.

This set of scripts is excellent, and free. http://www.redefinery.com/ae/rd_scripts/
It includes scripts such as "shapes to masks" and "masks to shape".
Highly recommended


Sound and image are shifted after importing them into Adobe Premiere

If you recorded using the IPhone, it could be due to the fact that the camera records in a variable frame-rate or VFR. This means that if the camera can't record a frame, it will skip it and so that second will have 59 instead of 60, and sometimes this can be very bad for longer footage and older phones. After time it those skipped frames will add up.

Adobe programs don't like variable frame-rates, I believe this is because they are "Professional" and don't think they need to support this feature because it's only a problem with low level cameras and recording software. They intend all footage to be Constant Frame Rate or CFR, meaning there is no skipped frames at all. When it comes into contact with VFR they essentially stretch out the footage and frames to fit the entire footage time, while keeping it at a CFR.

I personally don't know any amazing way of fixing this, other than re-compressing it using a program like Handbrake. Using handbrake you can change the footage frame-rate to a CFR. Handbrake is my solution, however it only supports H.264 and H.265, etc so it can be limiting. If you want to make sure you don't lose quality, try finding a different program and rendering it with a lossless codec.


Probably the best thing you can do is get an external monitor. the electronic viewfinders on most DSLRs are too low resolution to be able to judge focus critically. Many of these monitors have built in edge peaking, or focus detection tools as well to make it even easier. Unfortunately to get a decent high resolution external monitor is going to cost the price of the camera again.


Frame rate conversion during playback for maximal smoothness

The problem I think you are seeing is not because of the difference in framerate directly because this is inperceptable but how it is displayed.  You need to enable vsync to stop tearing and then it should play smoothly.


How to normalize 360-degree videos so that certain object / person remains in the middle?

You may have sorted this out by now, but I saw the open question, so I thought I'd answer it.

If you have a subscription to the Adobe Creative Cloud, you can now do this reorientation in Premiere.  In June, Adobe bought out the Skybox plugin from Mettle.  Essentially, that plugin allows you to re-orient your 360 video in AE or PR among other things.

There's now an effect in Premiere called 'VR Projection' and it allows you to adjust the pan, tilt, and roll of your projection.


How to link an FX property of multiple layers?

I think so.  Add a Slider (in the Effects window) to one of your tracks, then use the pickwhip to parent the value of the slider to each of the other Brightness controls.

Examples here of using Sliders and pick whips:
http://helpx.adobe.com/after-effects/using/expression-basics.html


Benro and Manfrotto Monopods - QR System

Have you considered buying an entirely separate set of QR plates to unify your entire setup? For instance, in our setup, we use a tripod, a monopod, and a GlideCam with various attachments for each of these (rail systems, teleprompters, etc.). By adding this (relatively) cheap QR plate to the top of each stabilization method, we can switch our cameras around all systems with ease, and the QR plate can go on any standard 3/8" or 1/4" screw.


Is there any way to stop Media Encoder creating this launch preventing file?

Solution presented at https://forums.adobe.com/thread/1268390:

  Open the file (C:\Program Files (x86)\Common
  Files\Adobe\SLCache\X19zbGNfZmVhdHVyZXNfXw==.slc) in any text editor,
  delete all of the contents, save it, and set the file to read-only.


AviSynth vs AviSynth+ vs VapourSynth: which one should I choose?

As a summary of the differences:
AviSynth

Outdated and no longer recommended for use
Only available in 32bit – not able to use 64bit plugins/filters
No multi-threading (unless you install AviSynth-MT, another fork, which is prone to errors and issues and no updates over last 4+ years)
Older and only supports 8bit video
Lacks support for some color spaces (YV16, YV24 etc.)
Latest release over 4 years ago (May 2016)

AviSynth+

Has both a 32bit and 64bit version, although the 32bit version apparently has some issues
Supports multi-threading which can significantly improve rendering performance for some functions (similar to AviSynth-MT, another fork, but AviSynth+’s appears more stable)
Be aware that using the 64bit version will require all plugins and filters to also be 64bit – and not all of them have 64bit versions. There are some filters that are 64bit only, however, which means you’re able to use those.
QTGMC runs faster in multi-threaded mode, and also apparently in 64bit mode
AviSynth+ specific documentation is limited, but most of the great AviSynth wiki also applies to AviSynth+ and many pages now have notes mentioning any specific instructions for working with AviSynth+.
Also runs on Linux, which is useful when you need to run it on a server
Regularly updated, with latest release Jun 2020

Note: be sure to get a recent version of AviSynth+ from https://github.com/AviSynth/AviSynthPlus/releases. There was a pinterf fork and it has been merged back to upstream.
Or you can use the Universal AviSynth Installer which allows you to easily switch between the different versions of AviSynth, AviSynth-MT, and AviSynth+ (32bit and 64bit).
VapourSynth

Rewritten to use Python, a much more powerful and flexible scripting language
Multi-threaded
Supports generalized colorspaces whereby new or custom colorspaces can be specified at runtime
Also runs on Linux, which is useful when you need to run it on a server
Documentation isn't as detailed as AviSynth (it's not a community wiki)
Regularly updated, with latest release Aug 2020


Ok, so it turns out that after a large amount of research into error codes, the Error: e:31:23 means that there is a problem with the right hand side tape reel cog. It appears that the cog cannot move the spool, as it clicks a bit before displaying the error when I insert a tape. I think this can only be fixed by a specialist, but I included this answer to give someone in the same situation a resolution.


Change project settings after footage is all cut up on Premiere Elements

you should be able to adjust everything just from the project panel.
See this, I dropped mp3 into 'new timeline' button and got this sequence -

All you need to do is go to sequence and select desired output

You can set everything here

For me, making the MP3 file into 'New sequence' defaulted in 1920x1080 29.97 sequence, but if you have issue that it made different format and you want to skip changing scale of every individual file (I'm assuming you might have got different format, like 1280x720, although unlikely), you can just setup an adjustment file above your edit, select 'distort - > transform' from effects and adjust scale to fit to your desired resolution.


What are the pros and cons of using a 4k to HD digital extender zoom

I've shot 4k footage since 2014, and when mastering in 1080p, I often use digital zoom in post. There are multiple factors I would look at.

Cameras that use a single sensor will display debeyering artefacts when examining the direct pixel readout (mushier detail and reduced colour information). This issue is nicely improved when downsampling 4k to 1080p, but is revealed when digitally zooming. The Z190, however, uses a 3-chip design. If each chip is full 4k (which the marketing copy certainly implies), the quality difference between downsampled and a direct pixel crop should be significantly less noticeable than on single-chip competitors.
Lens resolution can be a contributing factor as well. If the lens barely keeps up with the 4k resolution of the sensors, or the three-chip design introduces optical issues, then downsampling will likely hide minor softness. Digital zoom will reveal these optical shortcomings simply by magnifying what's being projected onto the sensor. On the upside, lens performance is typically best near the centre, so a centre crop is at least likely to be optically stronger than the edges.
While exposure and colour should look identical because nothing is changing (aperture, shutter speed, gain, pixel pitch), you're right that the exposure performance will change, as measured by the amount or type of visible grain. From user reports I've read, this is best demonstrated in the Sony A7 series of cameras where similar light sensitivity (based on the amount of visual noise) can be seen in direct pixel readouts from the A7s (giant pixels, lower resolution) and downsampled readouts from the A7r (smaller pixels, higher resolution, but the subsequent higher noise is reduced during downsampling). Digital zoom will reveal the per-pixel noise that downsampling would have averaged out, which can impact perceptual sharpness.
If you decide to record in 4k and downsample to 1080p afterward, zooming as needed in post, compression may come into play as well. While the Z190 records 4:2:2 10-bit, digital zoom can reveal minor compression softness in colour detail that would have been reduced during downsampling. However, I would expect this to apply only to downsampling/zooming 4k footage in post. If the camera is recording a 1080p 4:2:2 image directly, switching between a downsampled full sensor readout and a crop of the centre pixels should have little to no impact on compression results.
Some cameras may simply process things slightly differently when performing downsampling versus direct pixel readout.

When tracking down the source of the softness, I'd recommend starting by recording 4k footage and cropping it in post, then comparing it to footage shot using the 1080p zoom mode. Though highly unlikely, if poor camera processing of 1080p footage is to blame you'll see the difference here.

It's more likely a combination of lens resolution, optical performance of the three-chip design, and perceptual sharpness improvements from reduced noise when downsampling. But by all means, try some locked down resolution tests first! After tracking down the source(s) of the softness, then it's a question of how to combat it.

My approach has always been to shoot in 4k and rely on "digital zoom" in post while never reaching a full 2x zoom (direct pixel crop). This ensures the footage is always downsampled to some extent, maintaining the higher relative quality we're used to seeing now.


Making Premiere Pro Project Codecs Cross-Platform Compatible (Windows & Mac)

It has nothing to do with a codec or Apple (I assume he/she has QT installed on the PC anyway).

Here's what's going on: the PC version of Premiere is cracked and/or was not fully installed/updated. As such it is indeed missing the "Sequence Presets" that you have on your Mac. Hence the translation problem.

By "Sequence Preset" I mean the DSLR, RED, etc presets you see when you decide to "Create a new sequence". A cracked and/or poorly installed copy of Premiere will have all the important presets missing.

I recommend getting your hands of a legit copy of the program or running a thorough update of the program (it should solve your issue then, if the PC version is legal). Et oui!

The "Sequence Presets" look like this on Premiere CC (and are very similar on CS6 and previous versions):


Which video editing software can create this style of title graphic?

If you are just looking for what video editor can do a title over a video, the answer is any of them that are worth paying for and most of them that are free.  That's the majority of the point of a video editor is to blend video together and that more or less inherently requires supporting transparency.

If you are looking for the particular template used in that video, I'd suggest asking the people whose YouTube videos used it.  There are many motion graphics template vendors who provide templates that can be used for similar looks if you don't know how (or don't want to bother) to make one yourself.  There are often even some that are free.

That particular effect could be for any number of editors, composition programs or titlers though.  It might come with one directly as an included template or it might come separately.  Nobody is going to know all templates everywhere, so the best bet is to ask the people that used them.

If you are just looking for software in general, I use Adobe After Effects for most things like that, but it's a professional level product so you may not want to go that high end.  Premiere Pro is also able to do a level of motion graphics, though it is more limited.  I'm not sure on the lower levels, but if text overlay or layer transparency or something similar is listed on the features of a product, it should be able to do something reasonably similar.


How to figure out video format by video sample?

If you want to identify video / audio file i recommend you to download K-Lite codec pack. It always works for me.

http://www.codecguide.com/changelogs_full.htm
"•Updated MediaInfo Lite to version 0.7.53"

MediaInfo - that's the one you can use, after installing it's just right clicking on the file or opening it and it should be recognized, via the tray icons (if not disabled) you can change various layout and information settings.


Add black (or other color) padding when doing vstack (or hstack)

First, setup a color filter of the same height, and then add it to the hstack

-filter_complex "color=black:5x240[c];[v1][c][v2]hstack=inputs=3:shortest=1[videoout]"

See the options for the color filter here.

Alternate method is

-filter_complex "[0]pad=iw+5:ih:color=black[0v];[0v][v2]hstack[videoout]"


Using ffmpeg,

ffmpeg -i video.avi -vf field=top top%d.png

and

ffmpeg -i video.avi -vf field=bottom bottom%d.png


Why does variable bitrate still have a bitrate specified?

For CBR encodings, the bitrate is always kept at the bitrate specified regardless of if it is needed.  For VBR encoding, the bitrate is an average target, however the media stream will use more or less data rate when it needs to but will try to average to the target.

This is why you see 1 pass and 2 pass VBR.  2 pass VBR first makes a pass to estimate where more or less data will be needed and then makes a second pass to make best use of the available data rate.  1 pass does a lot more guessing.

For an example of how this works, lets say we had a video that had 10 seconds of rapid action followed by 10 seconds of a still image.

With CBR, the same data rate would be used the entire time, the 10 seconds of still image would be crazy high quality since nothing is changing, but the action would then fall apart because it is changing too much.

With 1 pass VBR, the algorithm would adjust to use more data for the high level of action, but it might limit itself since the entire video might be like that.  Then, when it got to the still shot, it would use almost nothing, but wouldn't hit the average rate because it didn't need the storage space.

With 2 pass VBR, the first pass would discover that there is no movement at the end, so then during the second pass, a lot of data rate would be used to handle the motion really well and then very little would be used for the still image, the average data rate would be hit and the high movement portion of the video would have the highest quality of any of the encodings while the still portion wouldn't really suffer.


After Effects: Jagged edges on shape layer and white borders in general

It's embarrasing, but it was a blending mode issue.

I must have accidentaly fiddled with blending and couldnt percieve it.

sorry for the dumbness.


Same exposure settings on 2 different DLSRs (Canon SL1 and 70d) created differently exposed image?

Simple answer is No.

Now to be a little more technical. It is all down to build, sensor type, style, size etc and of course the camera being stuff like, consumer, semi-pro, professional etc.

A little more.
Cameras like the SL1 which would be classed as consumer would not have the same build or sensor quality as a professional or possibly semi-pro same goes for sensor sizes etc.

I have a Canon 7D and a Canon 100D both of which I can use but I have to expose slightly differently on the 100D due to it being a lesser quality camera, even though it has basically the same sensor and specs (which is why I bought it). Most things you can correct it in post as long as I've thought about a little bits before hand.

This would also be the same for the built in colour profiles in the camera. Which  is why a lot of people start to build there own or put on preset ones like cinestyle.


How does ffprobe identifies I frame and IDR, when it sets key_frame=1

It sounds like you're analyzing a HEVC bitstream. HEVC has multiple types of Random Access Pictures, only one of which clears the buffer. But ffmpeg flags all of them as keyframes.

You have to use the HEVC decoder to parse this info out. The skip_frame option, as implemented in the HEVC decoder, only decodes IDR frames.

Use

ffmpeg -skip_frame nokey -i in.mp4 -vf showinfo -vsync 0 -f null -

The number of frames output is the IDR count.

For the sample file, you can use the filter_units bitstream filter to keep only IDR frames

ffmpeg -copyts -i HEVC-1080i25-STB_02_576i25.ts -an -c copy -bsf:v filter_units="19-20" -f mpegts - | ffmpeg -copyts -i - -vf showinfo -f null NUL

This will show the data of 118 frames, same as the analyzer.


Invalid DICT array size (83::2) in After Effects

This in essence is a font system error.

If your system was working fine and then after installing some fonts you are getting this error, I would uninstall the fonts in question.

Fonts that have special characters in their internal font names (not the fontfile name) like accents, umlauts or trademark signs will cause this issue and need to be removed.

The update to After Effects CC 12.1 should fix this error. http://blogs.adobe.com/aftereffects/2013/09/after-effects-cc-12-1-whats-new-and-changed.html


I would get as far away from EE as possible. Using the x264 tool, and mp4box, you can convert and segment out the files which are ready to be streamed to any dash compatible players. Especially since you mentioned using batch scripts, this is a great solution I think.

This is a good guide: http://www.dash-player.com/blog/2014/11/mpeg-dash-content-generation-using-mp4box-and-x264/


Just use hstack:

ffmpeg -i video.avi -i video2.avi -filter_complex "[0:v] setpts=PTS-STARTPTS, scale=360x288 [left]; [1:v] setpts=PTS-STARTPTS, scale=360x288 [right]; [left][right]hstack=inputs=2" -c:v libx264 -f flv rtmp://192.168.1.1/live/videocrop


ffmpeg -to option acting like -t option?

Yes, it does. -to looks at the timestamps. And input ss resets them to zero from the seek point. Either use -ss as output option (after -i) or calculate and use duration -t. The former will slow operation, as it performs a slow seek (decode and discard till inpoint).


Does youtube store different video files for different bit rate?

Yes

  is the technique called "resampling" ?

No

  Does youtube make live resampling for every user ?

No

  All this things happens live ?

No, not on youtube. But Yes on twitch.tv

  Or it encode and save different video files for different bitrate, so
  that when a user changes bitrate it servers that file.

Yes

  Isn't that a disk space destroyer too ?

Yes


Add overlay video and animated text (textbox and shadow) with FFmpeg

You only have two inputs, so 2:v can't exist. Instead, you'll need a base canvas to draw the text upon. I've used the color source filter for that. And you probably want that canvas to be the same size as the video. I've used scale2ref for that.

ffmpeg -i input -i overlayvideo \
-filter_complex \
"[0:v] setpts=PTS-STARTPTS [v01]; \
[1:v] setpts=PTS-STARTPTS [gradient]; \
color=black@0:duration=8,format=yuva444p[txtbase];[txtbase][v01]scale2ref[txtbase][v01];[txtbase] drawtext=fontfile=...font.ttf: text='Test': fontcolor=white:fontsize=54: shadowcolor=black@0.6:shadowx=4:shadowy=2: \
box=1: boxcolor=a60000@0.4:boxborderw=10:x='310':y='350',fade=t=in:st=1:d=1:alpha=1,fade=t=out:st=7:d=1:alpha=1[txt]; \
[v01][gradient] overlay [v02]; \
[v02][txt] overlay[outv]" \
-c:v libx264 -profile:v main -level:v 3.1 -pix_fmt yuv420p -c:a aac -map [outv] \
-f mp4 -y 'output.mp4'


Handbrake x265 Optimal settings for ripping DVD

Summary response for summary question :)

CRF 20 will look really nice
8-bit (no advantages over 10-bit)
No tuning. Grain tuning is for grainy source (old content). SSIM is for testing.
No additional parameters needed.
VFR (same as source)

I could really, really, reallly go down the rabbit hole with a more detailed answer, but I'll leave it at this for now.


gopro hero vs gopro hero4 black video quality diference

There are charts on their website, but in short, the Black allows for 4k recording, 60 mbps bandwidth, higher frame rates at various resolutions, simply put, it has a larger bottleneck enabling it to capture more data.

I would suspect the cameras may be perhaps identical, and are firmware locked down into lower levels for the less expensive versions. Bypassing that would be, well very difficult if not impossible.

The Blacks are great, I own a handful and use them all the time.


VHS and capture: what is the real aspect ratio?

I have tested a 4:3 vhs, and using 5:4 a/r making screenshot report bad final a/r(1:22)
mplayer -aspect 5:4 -vf screenshot

using 4:3 a/r report in screenshot the correct ar(1:31:1)
mplayer -aspect 4:3 -vf screenshot

So I think the correct answer is 4:3.


How to link an audio mp3 file created on soundcloud to your Youtube account

You may want to reupload the file to Youtube.

Convert the audio file (preferably your original, uncompressed file) to video with avconv -loop 1 -i cover.jpg -i original-track.wav -tune stillimage -r 1 -s 1280x720 -c:a flac -shortest -v error out-video.ogv (where cover.jpg is the still image you want to use for the video) or using a media converter like WinFF, and upload it to your youtube account.

You can place a link to the soundcloud page in your video description.


Why does the color red always appear pixelated on TV and videos on the PC?

It's not an illusion - it's called chroma subsampling.

Most video codecs do not represent colour in full resolution. This allows for more efficient "lossy" compression because it takes advantage of the fact that the human eye is more sensitive to brightness ("luma") than colour ("chroma"). Most lossy codecs lower the chroma resolution to half or one quarter of the overall resolution, so you may only get one pixel's worth of colour for every four pixels of brightness. This dramatically decreases the amount of data needed, with only a small loss of apparent quality.

It's slightly more complicated though: the brightness is actually made up of the sum of the three colour components Red, Green and Blue. And they're not encoded as RGB, that would be require more bandwidth, they're encoded as YUV. Y corresponds roughly to the green component, and the U and V are Y minus the red component and Y minus the blue component (a gross approximation, actually - if you want the whole formula look here).

In most codecs the U and V components are sampled at a lower resolution than the Y. this is expressed in the three-way ratio you often see if you hang around video forums too much, e.g. 4:2:2 or 4:2:0. For a two row rectangle of pixels the numbers represent:

  "width of sample region (Y samples)":"UV samples in the first
  row":"additional UV samples in the second row"

A common example of this notation is in the codec name "proRes422" the 422 part of the name comes from 4:2:2 meaning for every a 4x2 rectangle there will be 4 Y samples in each row 2 UV samples in the first row (half the horizontal resolution) and 2 UV samples in the second row. So proRes422 has half the chroma resolution of the luma.

On the internet and on telly you're most likely seeing everything in a 4:2:0 codec. In every 4x2 rectangle of picture there are only two UV samples (the 0 means that there are no additional samples on the second row). So the colour part of the image is composed of chunks 2x2 pixels in size, in other words, one quarter the resolution.

This means the red channel on its own has one quarter the resolution of the overall picture.

TL;DR the red looks pixelated - because it actually is.


What is the difference between ffmpeg and ffmbc now?

Usage differences

Note: ffmbc/ffmpeg will refer to the tools, while FFmbc/FFmpeg will refer to the projects themselves and/or each whole collection of tools and libraries.

There are syntax differences including:

ffmbc uses the old, confusing -newaudio option while ffmpeg can just use -map.
ffmbc does not support -codec or -c, so instead all stream types must be explicitly defined if you do not want to rely on the defaults. For example -vcodec copy -acodec copy -scodec copy vs just -c copy.
Probably lots more that I didn't think of. Feel free to edit answer and add more.

ffmbc will automatically relocate some data after encoding to MOV, MP4, etc while ffmpeg requires the -movflags +faststart option.
ffmbc does not support -filter_complex so filtering is different. For example, using the overlay video filter:

ffmbc: -i video.mp4 -vf "movie='overlay.png' [movie]; [in][movie] overlay [out]"
ffmpeg: -i video.mp4 -i image.png -filter_complex "[0:v][1:v]overlay"

FFmbc seems to support more broadcast formats.
FFmpeg has more features and filters.
FFmbc dropped ffplay and ffserver.

Development differences

FFmbc is GPL, and FFmpeg is mostly LGPL which makes porting features from FFmbc to FFmpeg more difficult.
FFmbc is basically the project of a former FFmpeg developer with a few contributions from others. FFmpeg has many developers and contributors.
FFmpeg often merges things form Libav, while FFmbc ignores Libav. Libav cherry-picks occasionally from FFmpeg, but ignores most stuff from FFmpeg (Libav is often joked to have "NIH syndrome").


Adobe after effects showing green background with nothing every time I import hd videos

It should be problem with codec. Found two solutions:

First — recompile footage to different codec.
Second — try to install software with codecs.Like Auicktime. Or if it will not help, some codeck pack.


When making an educational video, does citation need to be in the video itself?

The big disadvantage of citations that are part of the video itself is that they can't easily be a link, or even copy-pasted into a search engine like a text citation. So if it's a URL, the viewer will have to pause the video, and type the URL manually. You could use YouTubes "cards" function to add links as pop-up interactive overlays, but I think you are limited to 5 links per video.

I would put my citations in the description, and in case the video gets embedded somewhere else, give them at the end of the video as well, perhaps with a link to a page where they could be found.

There is no set convention that I know of, unlike citations in papers or books. There is a podcast that I listen to that does give citations, but they just tell you how many at the end and send you to their website to read the transcript to find them. I don't think most viewers are expecting them, except for the implicit citation you give when you say something like "in a 2019 study by Dr Stib, it was noted that…" when citing other's research.


Recording Video Games and uploading to YouTube

1) Each software has its own advantages and disadvantages, there isn't a right answer.  If you have ShadowPlay support, the hardware leverage is nice.  FRAPS is good at dumping a really high quality video quickly, but it needs lots of memory and high speed disks to deal with the amount of data it dumps.  I've not personally worked with DxTory, so I can't speak to that, though I believe it offers a slightly more balanced level of data rate (can use slower hdd) but at a cost of CPU performance.  I could be wrong on that last part though.

2) There are two things you can do to help reduce the size.  You can edit the video in a video editor such as Premiere or Windows Movie Maker in order to remove portions of the video that you do not need.  This trims file size by reducing the amount of time the video covers.

You can also use what is known as a video encoder, such as Quicktime, Adobe Media Encoder or ffMPEG.  A video encoder stores the video in a more compressed way that allows for smaller files size while minimizing the quality loss.  Using a format like 2-pass VBR h.264 will produce very high quality files with a minimal size, however it is time consuming and CPU intensive to produce such files, so the live encoding tools can not use that level of compression.  Running the slower encoding will produce smaller files as it can take the time to find more ways to save space.

The Youtube recommendations that you talk about are the preferred encoder settings that Youtube wants a source file to have, however Youtube itself will still transcode (make additional encodings) the file for streaming at different resolutions and speeds.

3) Your preference.  Video standards tend to be either 30fps (NTSC TV (America/Japan)), 25fps (PAL TV (most of the rest of the world)) or 24fps (Cinema, used frequently by everyone).  Choose based on what your system can handle and what you like the most.  If you can't manage 24fps, then an even multiple tends to work best since frames will match up at the higher rate. (For example, 15fps on a 30fps video will result in each original frame being used twice, so the computer doesn't have to guess at any frames [called interpolation].)


Is it possible to verify if mpeg dash video was correctly transcoded

This is not an issue of the player, but it appears to be related to your Cloudfront setup and/or your S3 bucket, as it doesn't provide the required CORS configuration, therefore the player can't download your video/audio content.

You can check that yourself by opening the debug console of your browser (F12 on Windows, or CMD SHIFT I on Mac). In the console tab you will find error messages like that:

Access to XMLHttpRequest at 'https://d19cwas8lp55sz.cloudfront.net/archivos/1048/mpegDash/mpegDash4800k/3d3b437408996f42ed668f17cbe10408e0059e8914fcb12c775ad92d002cc88f' from origin 'https://bitmovin.com' has been blocked by CORS policy: Response to preflight request doesn't pass access control check: No 'Access-Control-Allow-Origin' header is present on the requested resource.

CORS is a security policy enforced by the browser in order to allow/deny XHR requests to resources which are hosted on a different domain than the site the request is coming from. XHR requests are commonly used by javascript applications web apps as well as HTML5 players to request your video content.

The decision of the browser, if it blocks this request or not, depends on additional headers which are returned by the server the XHR request is sent to (Cloudfront and/or your S3 bucket in this case). If the CORS configuration whitelists the domain the request is coming from, the browser will allow this request.

In your case this CORS configuration seems to be missing. Therefore, please see your S3 CORS settings:

S3 Bucket CORS configuration: https://docs.aws.amazon.com/AmazonS3/latest/user-guide/add-cors-configuration.html

This CORS configuration allows every origin to access your content

<?xml version="1.0" encoding="UTF-8"?>
<CORSConfiguration xmlns="http://s3.amazonaws.com/doc/2006-03-01/">
<CORSRule>
    <AllowedOrigin>*</AllowedOrigin>
    <AllowedMethod>GET</AllowedMethod>
    <MaxAgeSeconds>1800</MaxAgeSeconds>
    <AllowedHeader>*</AllowedHeader>
</CORSRule>
</CORSConfiguration>

This CORS configuration only allows bitmovin.com to access your content

<?xml version="1.0" encoding="UTF-8"?>
<CORSConfiguration xmlns="http://s3.amazonaws.com/doc/2006-03-01/">
<CORSRule>
    <AllowedOrigin>https://bitmovin.com</AllowedOrigin>
    <AllowedMethod>GET</AllowedMethod>
    <MaxAgeSeconds>1800</MaxAgeSeconds>
    <AllowedHeader>*</AllowedHeader>
</CORSRule>
</CORSConfiguration>

How to configure Cloudfront to respect CORS:
https://medium.com/@dmitter/amazon-cloudfront-and-cors-57dd43cdfd30
OR
https://docs.aws.amazon.com/AmazonCloudFront/latest/DeveloperGuide/header-caching.html#header-caching-web-cors

As soon as this is done, the playback at this demo page should work just fine. https://bitmovin.com/demos/stream-test?format=dash&manifest=https%3A%2F%2Fbitmovin-a.akamaihd.net%2Fcontent%2FMI201109210084_1%2Fmpds%2Ff08e80da-bf1d-4e3d-8899-f0f6155f6efa.mpd

I hope this is helpful and understandable :)


Is footage recorded through external recorder really better?

Unless the DSLR/Mirrorless camera you are using outputs 10-bit 4:2:2, using an external recorder to up-convert the footage to ProRes HQ will not result in a better image quality than what the camera is able to produce/output.

If anything it will allow you to edit and re-compress the footage multiple times without losing the original quality.

Using an external recorder may result in a better picture quality compared to the internal encoding, because there is more bitrate available per frame, more robust codecs, and also being able to remove the 5, 15, or 30 minutes recording limitation the camera may have.

A bigger plus using an external recorder (or display) is the ability to nail the focus, assess proper exposure, and having multiple eyes monitoring the recording.


After Effects Crashes When Exporting To Media Encoder

Media encoder takes a real long time to open, I think this is the problem.  If I open media encoder first then after effects second it will export properly.  Just got to wait 5 mins for media encoder to open fully first.


How to BATCH remux combine video and audio to one file?

for f in .mkv;do ffmpeg -i "$f" -map 0:1 "${f%mkv}wav";done && for i in .mkv; do mkdir -p "${i%.}" && mkvmerge -o "${i%.}/${i%.}.mkv" -A "$i" "${i%.}.wav"; done


show the process in a football feint or motion

You would do this kind of thing in After-Effects or a similar compositing program (like Nuke or Fusion). Simply duplicate your video as many times as you want the move to be repeated, then offset the layers in time, and lastly roughly mask out the person from every shot except the very bottom layer.


Lightworks problems importing, editing, and exporting using Canon EOS Rebel T3i

So after 8 hours of research, trial and error, I found a successful workflow for .mov footage from my Canon EOS Rebel T3i.

The first part is that you can't import .mov "Apple / H264" files with Lightworks without purchasing the "Pro" version.  The pro version costs $60 per year (which is actually a good price for what you get).  So I bought that and was able to import my .mov files.

The next problem I had was that playing the imported footage was horribly slow and choppy.  I wanted to "mark and park" to get clips added to my editor but when I would play the video, it would only play a few frames and then get horribly choppy (audio would still play).  By the way, I have a laptop with an Intel Core i5 (2nd gen) and 4GB of RAM.

After some research, I found the solution to this was to import the video while transcoding it to AVI format with "DVCPRO100".  This made the video playable within Lightworks for editing.

Next I had some real problems exporting my edits into a codec/format that would actually work.  Some formats would just error out (post an error and exit).  Others would succeed, but the final video would not have audio (I think this happened with MP4).  Finally I found that exporting as "Quicktime Movie" (not to be confused with the "MOV" option!) worked.  It worked with the following options:

Quicktime Movie
H.264, Quality = High, Key framerate = 24
Frame Recording = Yes, Encoding Mode = Multi-pass

Sound:  Format = Uncompressed, 44.1 kHz, Channels = 2

For the other Lightworks export options, I chose:

HD 1080, 29.97 fps, and Progressive button checked.

I sure hope this helps other people save 8 hours of frustration!  With this workflow, I can now be efficient and productive and make some great videos.


Is automatic switching of audio stereo balance while cutting between camera angles possible?

Is it possible?  Sure, just setup each source clip as a sequence that has the audio feed you want for that shot and leave the tracks linked so that you cut between both video and audio.

Is it advisable?  Probably not.  People are far less used to changing of the audio scene they are watching.  Video creates kind of a disconnect between the scene we are seeing and the audio we are hearing and it may be a bit jarring and break continuity to try radically changing the audio image every time you make a cut.

Go ahead and try it, I may be incorrect on how it will come off since I have not personally tried it, but in my experience with both video and audio, it doesn't seem like it will work well.


FFmpeg, encode video for 720p @ 4mbps playback

The ffprobe command below will spit out the bitrate:

ffprobe in.mp4 -select_streams v -show_entries stream=bit_rate -of compact=p=0:nk=1

e.g.

4118620

If the result is greater than 4000000 then you can add -b:v 4000k -bufsize 4000k -maxrate:v 4000k. Or you can add it anyway. Set -c:v to copy if the stream is under 4000000 else to libx264. The three values will be ignored just like -b:a:0 448k in your command is.


Environments in which to stage an animated scene

Have you looked at Maya?

Lovely showreel here.

It's an excellent platform for animation, rendering and development.


quickly extract video frames with ffmpeg

I have no idea if what you're doing is optimal, but "playing the video" to extract frames is essentially what has to happen, at least a little, if the frame you're extracting is not an I-frame (most frames are not).

The chosen frame is likely represented as the accumulated differences since the last complete frame, which is what interframe compression does. In order to extract it, a program must read and reconstruct the intervening frames -- not from the beginning of the video, just from the most recent complete frame

Depending on how many frames you need from each video, it might make sense to first convert to an "all-I" codec like MJPEG, where every frame stands alone. Then extracting any given frame would be much faster. When done you can delete the converted file.


Can i use my laptop as an external recorder

Not really. Not unless your laptop happens to contain a 4K capture card.  Most laptops don’t have video signal inputs.  You can do HD with a Blackmagic Ultrastudio Mini Recorder for a little over a hundred dollars, as long as you can find capture+encode software, but putting the encoding onus on your laptop isn’t very reliable.
If recording space is your major concern, many cameras are now capable of writing directly to externally attached SSDs, presuming the data rate generated by the codec you’re recording is lower than the write speed of the SSD.  I’m not sure about the R6, but if it doesn’t, you might consider trading it for one that does, depending on your priorities.


Name of the software used for the given animation

The short film, Wildebeest was rendered in Adobe After Effects and Flash.

This is the quote from an interview by the director Ant Blades.
"The film is animated in Flash and AfterEffects. The main aim was to be funny. The secondary aim was to develop a style I hadn’t tried before."
Source:"Filmmaker Ant Blades, Wildebeest"

Workflow of Mr Blades is to first sketch in Adobe Flash. As he puts it, it is an easy way to sketch out an "animatic" of an idea. He then renders it with After Effects, I assume he would be using the plug-in Cinema 4D to render the 3D work but that is a guess on my part. A more complete interview and the process in how he creates his works for Bird Box Studio. "Interview Ant Blades, Bird Box Studio"


I think browsers usually don't have configurable buffer sizes for streaming.  If you can get the video URL out of the browser, you could stream with mpv or something.

mpv --cache-secs=15 http://example.org/video.mp4

mpv has several other cache-tuning options.

As far as using a VPN: your ISP still sees the traffic.  It just doesn't see that it's an HTTP request, and even deep packet inspection won't be able to tell that it's streaming video.  I think most ISPs, if they do any QoS-based traffic shaping, would prioritize streaming audio/video to prevent annoyances when realtime matters, unlike torrent downloads.  So putting your traffic inside a VPN would if anything make it worse, in most cases, I'd expect.


Is it possible to store subtitle srt files in an m3u playlist file?

m3u playlists only contains links to the media itself that you're playing.

It doesn't actually consider the srt file at all. If the software used to play back the m3u file support srt files it will look in the same folder for the same file as the video only with the extension switched to srt.

You will therefor always need to store the srt together with the video file.