text:
How to spread light?
I am looking to film near a window. I was wondering if there was a certain material that can take the sun light source and spread it evenly and wider. I find that a diffuser it isn't enough is there another type of material that would spread and expand it even?

Is the window in shot?

Define "isn't enough"--what specific issue are you having? Light level? Color? Contrast?

A prism would separate the light into a spectrum, which probably isn't what you want. If the window isn't visible in your shot, you could use what's sometimes called a silk, which is more often made of muslin or other semi-translucent material.

A "silk" (whatever its actual fiber composition) is exactly a "diffuser" which armoose said "isn't enough".

@RichardCrowley  Right. But he didn't say what he used as a diffuser and given the lack of other details it certainly bears suggesting.

You did not mention: size/area, permanent/temporary, budget, etc. etc.  So, in the absense of any practical constraints, letting my imagination run wild....
I would investigate hanging a row of round acrylic ("Plexiglas") rods like a curtain which would serve rather like a big amorphous "Fresnel lens".
Else, you could use the traditional "brute-force" method used by big-budget production companies:

Giant lamps outside the window focused through the opening. They commonly use expensive (and very bright) HMI lights for "punching through" sunlight levels.
Big reflectors that catch extra sunlight and reflect it in through the window.  Metalized reflective fabric stretched on a big frame

How about a bedsheet over the outside of the window?  Possibly along with some correcting gel to bring the color temperature closer to any other sources you want to use inside?


text:
Lens that doesn't make faces look fat
I'm using a "PANASONIC LUMIX G II Lens, 20mm, F1.7 ASPH" (MFT) lens to shoot interviews.
However, it distorts faces a bit when within 10 feet. I've read (after I bought it, unfortunately) that this lens isn't too well suited to interviews for that reason.
What would be a better fit? A wider lens would also be nice.
Any advice appreciated!

This question is better suited on http://photo.stackexchange.com/. Actually, it has been asked there numerous times and answered as well; see for instance Which focal-length lens is usually used for portrait photography, and why?

Another complete comparison of shorter focal lengths at Photo.SE.

Typically recommended portrait lenses are around 85mm to 135m. These numbers refer to full frame cameras.
Since you are using a m43 camera with a crop factor of 2x you should investigate lenses from 40mm to 70mm that fit your Panasonic

70mm to 100mm is the ideal portrait range. Although I prefer a farther throw many times simply for the look.
Wider will distort, longer will compress.
20mm, that's wide- very for full frame or even a crop sensor. Are you sure of that?

Thanks coinrocket, I'll investigate lenses in that range. Mine is a micro-four-thirds lens- I think 20mm is equivalent to 50mm.

I have not used this lens but from my research it is a great lens for portraits/interviews. The Lumix 42.5mm F1.7 can be found on ebay for around $300 to $250 USD.

source: Stack Exchange

license: Creative Commons - Attribution Share-Alike - https://creativecommons.org/licenses/by-sa/3.0/

all_licenses: Creative Commons - Attribution Share-Alike - https://creativecommons.org/licenses/by-sa/4.0/

sort: votes


text:
Halve framerate without complete re-encoding
Is there any tool that can manipulate / transcode h264 or h265 video streams throwing away approximately half of the frames, without performing a full re-encode?
I'm thinking it would be theoretically possible to preserve motion vectors and such computed data, preserve all I-frames, and just throw away approximately half of the B- and P-frames (possibly adjusting the remaining ones.) The result does not need to be CFR.
Is there a way to do this with ffmpeg? (or any other tool)

The solution at question " Lower framerate without re-encode " looks promising:
ffmpeg -discard nokey -i input.mp4 -c copy -vsync 0 I-only.mp4

By manpage there should be 5 levels of frame discarding: none, noref, bidir, nokey, all.
I tested this with older ffmpeg 4.2.4 and h265 and it had no effect. Probably not implemented yet.

Comment for the OP question (can't comment yet):  I came looking for the same thing. I need to play 1920x1080@60 files on a hardware player that can't effectively drop frames. I don't want to buy a new box just because people started sending me unnecessarily high quality files.

It is just a matter of time when people start sending me 4k. Do you think it is technically possible to do something like this for halving the resolution?

I personally think it's technically possible (and feasible) to halve the framerate and/or the resolution without a full re-encode, by analyzing and transforming the already encoded data, in much the same way as JPEG files can be rotated by 90 degrees and/or mirrored without re-encoding, or MP3 files could have their volume normalized without a full re-encode. Whether such a tool already exist is another question. I personally bought new hardware, so this question is not relevant to me anymore, but I'll keep it open because if there is a solution, it can be useful to others.

I think the term for this effort is statistical multiplexing where your proposed framework would be an efficient implementation thereof.

license: Creative Commons - Attribution Share-Alike - https://creativecommons.org/licenses/by-sa/4.0/


text:
Can I reproduce video in 1440×1080 12M MP4 on a Full HD TV?
I am interesting in shooting some amatorial videos (family) with a digital camera.
At the moment my camera support AVCHD and MP4.
I found AVCHD less compatible and more time consuming to process so I was thinking if MP4 would be a more convenient options.
In that format my digital camera shoot at 1440×1080 12M MP4
I would like to know:

Can I reproduce video on a FULL HD TV (1920x1080) recorded with 1440×1080 12M MP4?
Would you suggest AVCHD instead for my use scenario?

Her some technical details of my camera:
http://docs.esupport.sony.com/dvimag/DSCRX100_guide/en/contents/04/03/02/02.html
http://www.sony.jp/products/overseas/contents/pickup/english/110324_promotion/DSC-TX100_Full_HD/
http://www.dpreview.com/reviews/sony-cybershot-dsc-rx100-m2/5

Maybe related: http://www.frankschrader.us/2013/05/pixel-aspect-ratio-video-resolution-of-1440x1080-considered-a-widescreen-format-169/

You don't mention either how (or IF?) you are editing anything here, or what forms of storage, distribution, archiving, etc you anticipate?
You are probably correct that MP4 is a more compatible distribution format, but distribution format does not necessarily have to be identical with recording and editing format.
I would not anticipate any issue with displaying a 1440×1080 frame on a 1920x1080 screen. Modern digital displays typically handle scaling (UP or DOWN) pretty seamlessly.  In any case, shouldn't it be easy to actually try it for yourself?  No amount of anonymous advice from the internet beats actual first-hand experiment.

Thanks for your answer, I would like to ask you... By scaling do you mean that the resolution 1440x1080 as a different aspect ratio for pixel dimensions. Example square vs rectangle? Thanks for your time on this.

The digital video contains "meta information" about the video so that different screens can display it properly.  Note that 1440x1080 is the old 4x3 frame standard, and not the modern 16x9 frame.

all_licenses: Creative Commons - Attribution Share-Alike - https://creativecommons.org/licenses/by-sa/3.0/


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How to DISABLE--NOT REMOVE subtitle tracks from an MKV file from the command line?

ATTENTION:
THIS QUESTION IS ABOUT DISABLING--NOT REMOVING SUBTITLE TRACKS FROM MKV FILES FROM THE COMMAND LINE. PERIOD.
It IS NOT about REMOVING subtitles by REMUXING with mkvmerge or any other tool. There are literally thousands of threads about removal by remuxing. That's precisely why I started this thread about DISABLING NOT REMOVING subtiles. I could not find a single thread about that.
It is NOT about GLOBALLY DISABLING subtitles in any PLAYER. That is obviously trivial and doesn't need any explanation. I don't need/want to do that and, if I did, I obviously wouldn't have taken the time to research the topic at hand and publish my findings in this thread.
I shouldn't have to add this disclaimer but there have been a number of, in most cases demeaning, 'answers' that reveal a distinct misunderstanding of the OP, suggesting irrelevant 'solutions', and/or criticizing this hack as being 'O-h-h-h-so-difficult' if you want to re-enable the subtitle track at a later date. As I pointed out numerous times, anyone who can understand my answer and who knows how to use the command line will find this task trivial. Those who don't SHOULD NOT ATTEMPT this hack.
The answer has already been given as this is the entire point of this thread. So any new answers are either superfluous or, in SOME cases, irrelevant. I will no longer respond to irrelevant 'answers' or comments or demeaning remarks by users who, judging by their reputations, should know better. Any such abuse will be flagged accordingly.
I welcome any CONSTRUCTIVE suggestions that ADHERE TO THE TOPIC, in a COMMENT, but any 'answers' or negative comments will be unanswered and flagged as inappropriate.

Now that THAT is out of the way, let us begin…
I couldn't find out how to do this anywhere so I thought I'd share this hack with others who, like me, want to DISABLE, NOT REMOVE, subtitles in MKV files.
There are times when you don't want the subtitles that are embedded in a file and you don't want to have to disable subtitles every time you play such a file or globally disable subtitles in every player you happen to have. Or you may want to automatically use a subtitle (srt) file that is preferable.
Sure, you can completely REMOVE subtitles with mkvmerge:
mkvmerge --no-subtitles "inputfile.mkv" -o "outputfile.mkv"

But this is heavy-handed and a real pain if you have a bunch of multi-GB files. It takes lots of time and disk space to de- and re-mux every file. So I turned to mkvpropedit which edits the properties of the subtitle track header in milliseconds. Note that this tool is installed when you install  MKVToolNix. Unfortunately, it is not at all obvious how to use this wonderful tool for the task at hand. Here's an example of the properties of a subtitle track as given by mkvinfo:
| + A track
|  + Track number: 4 (track ID for mkvmerge & mkvextract: 3)
|  + Track UID: 11363869179551629452
|  + Track type: subtitles
|  + Default flag: 0
|  + Lacing flag: 0
|  + Codec ID: S_TEXT/UTF8
|  + Name: CC

Despite the default flag setting 0 (false), the subtitle track is still active by default in players. I later read in the mkvmerge documentation "If no track is set to be the default track then mkvmerge will promote the first track of each type that it finds to be the default track. This is consistent with the behavior of various media players." I'm not sure if this means that it sets the default flag to 1 or somehow otherwise arranges for the track to be the default, but it seems maybe the latter.
I tried adding the track enabled property set to false but it didn't work.
mkvpropedit --edit track:4 --set flag-enabled=0 "inputfile.mkv"

| + A track
|  + Track number: 4 (track ID for mkvmerge & mkvextract: 3)
|  + Track UID: 11363869179551629452
|  + Track type: subtitles
|  + Default flag: 0
|  + Lacing flag: 0
|  + Codec ID: S_TEXT/UTF8
|  + Name: CC
|  + Enabled: 0

You'd be (we would all be) better served if you split this into a separate question and answer. Also, nice trick but it might be better to just prefix the codec name with # or ! (eg) to disable it but still make it obvious to undo.

Yeah, I did that and it ended up eliciting spam answers and a mean-spirited attack on me personally by AJ Henderson who didn't even understand the question. I notice that her cronies actually downvoted my QUESTION! WOW!

"Despite the default flag setting 0 (false), the subtitle track is still active by default in players" which players do this?! All the ones I've tried respect the default flag without an issue.

I gave up and contacted the author of mkvpropedit to ask if it was possible to do what I wanted to do. The first answer I got was 'No'. But I persevered and asked if there was any setting that will trick the player into not recognizing a subtitle track? The response was to set the Codec ID to something the player doesn't support. Eureka!
mkvpropedit --edit track:4 --set codec-id=DISABLED "inputfile.mkv"

| + A track
|  + Track number: 4 (track ID for mkvmerge & mkvextract: 3)
|  + Track UID: 11363869179551629452
|  + Track type: subtitles
|  + Default flag: 0
|  + Lacing flag: 0
|  + Codec ID: DISABLED
|  + Name: CC

This worked! The player doesn't recognize any subtitle track. In milliseconds I have achieved my goal. No time-consuming remux. Nice.
The procedure for doing this hack is as follows:

Get track number of the subtitle track
mkvinfo "inputfile.mkv"
…
| + A track
|  + Track number: 4 (track ID for mkvmerge & mkvextract: 3)
|  + Track UID: 11363869179551629452
|  + Track type: subtitles
|  + Default flag: 0
|  + Lacing flag: 0
|  + Codec ID: S_TEXT/UTF8
|  + Name: CC
…

OR
mkvmerge --identify "inpufile.mkv" | grep subtitles
Track ID 3: subtitles (SubRip/SRT)

Note that mkvinfo gives the proper 1-indexed track number that mkvpropedit uses, BUT mkvmerge gives a 0-indexed number so you have to add 1.  Be sure to do this or you'll mess up the wrong track!
Change the Codec ID to some nonsense name so the track is not recognized as a subtitle track. (In this case the track number is 4 but it will be different for other files.)
mkvpropedit --edit track:4 --set codec-id=DISABLED "inputfile.mkv"

| + A track
|  + Track number: 4 (track ID for mkvmerge & mkvextract: 3)
|  + Track UID: 11363869179551629452
|  + Track type: subtitles
|  + Default flag: 0
|  + Lacing flag: 0
|  + Codec ID: DISABLED
|  + Name: CC

Track ID 3: subtitles (DISABLED)

If you think you may at some time want to enable the track, keep a copy of the original Codec ID. As suggested by Jim Mack, it would be expedient to just add an obvious character to the existing Codec ID, thus preserving the original ID.
This hack works on files that have multiple subtitle tracks. Just repeat the renaming procedure for each track.

This is the answer. All other answers given do not respect the conditions of the OP.

I renamed Codec ID to DISABLED, but VLC still displays all subtitles in the dropdown menu (selecting returns an error). In addition Popcorn AudioConverter or rather one of its used batch tools returns an error while remuxing the file.

Not enough karma to post a proper answer. But, I found my way here because I was wondering why vlc insisted on showing English subs in some English-audio mkv files I had.  Turns out, players are smarter than we give them credit for, they just need all the information.  By properly setting the language on the audio and subs tracks and default the English subs track to off, the subs are still there but not shown. Assuming your file only has 1 audio and 1 sub track: mkvpropedit file.mkv --edit track:a1 --set language=eng --edit track:s1 --set language=eng --edit track:s1 --set flag-default=0

I don't know when this was added, but MKV actually has a track-flag dictating whether it is enabled or not, flag-enabled in mkvpropedit. According to MKVToolNix GUI "Players should only consider enabled tracks for playback." The problem here is: should. I tested on MPV (0.39), VLC (3.0.21), and Kodi (21.1), and they all seem to just ignore the flag :( Maybe the support will get better eventually, but for now it's just a note that this is technically possible natively. Unfortunately the codec-id method segfaults mpv.

What player are you using?
Unless you have the "forced" flag set, it's a player option whether subtitles are displayed automatically or not and in which language if you have several subtitle tracks.
First, set the "default-flag" to false:
mkvpropedit video.mkv --edit track:s1 --set flag-default=0

Then lookup your player preferences.
In VLC (2.0.8), I had to follow these instructions:

To disable subtitles by default, select "Preferences", then "Show
  All". Select "Input/Codecs". On the "Subtitle Track ID" selection
  window, change the value to "-1".

and also set the preferred language to "none".
In SMplayer, uncheck Preferences / Subtitles / "Select first available subtitle".
Once your "default" flag is set to false for all subtitle tracks, it only depends on your player preferences.

You didn't read the OP. Disabling subtitles in the player is obvious but not at all what I wanted. I want English subtitles to display but not any other languages. Setting the default flag does not work. That is why I went to the trouble of finding a method that works universally and published my findings here.

Using MakeMKV for some reason subtitles are always enabled (shows up in both VLC and apple tv) I used your script do make this one liner to set the default flag on all mkv files in a given directory. Thanks! Works great.
for FILE in *.mkv; do mkvpropedit $FILE --edit track:s1 --set flag-default=0; done

While your self-selected answer will work, it requires modifying the file again when you do want the subtitles.
Why not just create an empty but syntactically valid SRT file and mux that as the first subtitle stream (leaving the original as a secondary stream)? Even if the player is set to display subs, nothing will show up. No need to mess with ID strings.
And when you do want to see subtitles, you should be able to switch to the (populated) secondary subtitle stream, from within the player. No need to modify the file.

Because it is much more work  and goes against the first point in the OP. This is about disabling subtitles WITHOUT remuxing.

Exactly!  I'm specifically looking for something just like the op described and remuxing is not an option.
To provide more context: I'd like to disable subtitles in a very large number of large files. They're on a networked file system so I'd like to do this with minimal network traffic.
Fortunately, the file system can apply binary diffs for small changes to large files.  Remuxing would still require downloading the entire file so that isn't ideal.  The accepted answer is hacky, but works great so far!


text:
Brightness/Contrast Difference from PC to YouTube
So I have a trailer I've been working on, I did everything in Davinci Resolve 12. When I convert the video using Handbrake everything looks fine, but when I upload it to YouTube, either there's a ton more contrast, or gray is being remaped to black, not sure, but the photos below should show what I'm talking about.
I've done quite a bit of research, and the closest I've come is there might be something wrong with the way YouTube treats color spaces in conjunction with how I converted it. Any ideas? Thanks.
EDIT: Just finished uploading a version I converted with MPEG Streamclip, and the color looks great, though it was washed out on my PC. It came out as a ~200MB file for just a few seconds at 100% though, so not fun to upload. Looked just as crisp with Handbrake at ~27MB. Any way to take the color/brightness stuff from Streamclip and use it with Handbrake?
YouTube:

VLC on my PC:

Did you find out what was causing the shift? I have the same problem - Prores mov looks great in QT and VLC but washed out when uploaded to YT.

@Pedr No, but what I ended up doing is to use DaVinci's YouTube output setting on the deliver page, which seems to work great. There's also a "Data level" setting on that page under video which makes the footage look different.

Thanks. I had assumed that converting to H264 before upload would result in a lower quality video because YouTube will re-encode it.

Looks like Handbrake is outputting full range but flagging it as video levels, so the Handbrake output actually has Y=0 as black and Y=255 as white but the flag is telling Youtube that Y=16 is black and Y=235 is white.
If you upload a short segment (3-5s) of both your Handbrake input and output, that can be confirmed.

That was what I thought. I just uploaded the original MOV from Davinci and it looks just like the Handbrake file.

Not clear to me. You said that Handbrake looked fine locally but expanded on Youtube, so does the MOV on YT look like Handbrake on PC or YT?

Sorry, the MOV acts the same as the Handbrake file, in that locally it looks fine, but on YouTube it looks the same as the Handbrake file does when uploaded to YouTube.


text:
Non-Monotonous DTS on concat (ffmpeg)
After running this command ffmpeg -f concat -i mylist.txt -c copy output.mp4 - I'm getting corrupted output.mp4 file and this message:
ffmpeg -f concat -i mylist.txt -c copy output.mp4
ffmpeg version 2.6.2 Copyright (c) 2000-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-vda
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Input #0, concat, from 'mylist.txt':
  Duration: N/A, start: 0.000000, bitrate: 829 kb/s
    Stream #0:0: Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1440x900, 701 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc
    Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
Output #0, mp4, to 'output.mp4':
  Metadata:
    encoder         : Lavf56.25.101
    Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1440x900, q=2-31, 701 kb/s, 30 fps, 30 tbr, 15360 tbn, 15360 tbc
    Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo, 128 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x7f897a01bc00] Non-monotonous DTS in output stream 0:0; previous: 598061, current: 467644; changing to 598062. This may result in incorrect timestamps in the output file.
[mp4 @ 0x7f897a01bc00] Non-monotonous DTS in output stream 0:0; previous: 598062, current: 468044; changing to 598063. This may result in incorrect timestamps in the output file.
[mp4 @ 0x7f897a01bc00] Non-monotonous DTS in output stream 0:0; previous: 598063, current: 468444; changing to 598064. This may result in incorrect timestamps in the output file.
...
[mp4 @ 0x7f897a01bc00] Non-monotonous DTS in output stream 0:0; previous: 598362, current: 588044; changing to 598363. This may result in incorrect timestamps in the output file.
frame= 1472 fps=0.0 q=-1.0 Lsize=    5825kB time=00:00:49.04 bitrate= 973.0kbits/s
video:4903kB audio:877kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.776358%

Content of mylist.txt
file 'cut.mp4'
file 'cut2.mp4'

cut.mp4 output from ffmpeg:
ffmpeg -i cut.mp4
ffmpeg version 2.6.2 Copyright (c) 2000-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-vda
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'cut.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf56.25.101
  Duration: 00:00:39.04, start: 0.036281, bitrate: 837 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1440x900, 701 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      handler_name    : SoundHandler

cut2.mp4 output from ffmpeg:
ffmpeg -i cut2.mp4
ffmpeg version 2.6.2 Copyright (c) 2000-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-vda
  libavutil      54. 20.100 / 54. 20.100
  libavcodec     56. 26.100 / 56. 26.100
  libavformat    56. 25.101 / 56. 25.101
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 11.102 /  5. 11.102
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc    53.  3.100 / 53.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'cut2.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf56.25.101
  Duration: 00:00:10.07, start: 0.000000, bitrate: 1498 kb/s
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1440x900, 1271 kb/s, 30 fps, 30 tbr, 12k tbn, 60 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 218 kb/s (default)
    Metadata:
      handler_name    : SoundHandler

cut.mp4 I got by this command ffmpeg -ss 00:00:11 -i myfile.mp4 -to 00:00:39 -vf 'drawbox= : x=0 : y=0 : color=invert' cut.mp4
cut2.mp4 I got by this command ffmpeg -ss 00:00:00 -i myfile.mp4 -to 00:00:10 -c copy cut2.mp4
I searched a lot - didn't find any solution, maybe, someone can help me out  with this one.
output.mp4 is playable, but looks weird.

Can you provide both input files so I can perform tests?

@LordNeckbeard yes, I sent it to you, thank you very much.

I can't seem to duplicate the exact issue. I'm using the most recent static build available from tessus. The sample files seem to be different than your screenshot, but it appears to be working fine in VLC and QuickTime, except seeking seems messed up. I tested on an OS X machine because that looks like what you're using.

@LordNeckbeard Thank you very much, yes with this version ffmpeg version N-71996-g372aa07-tessus  it does work great. Thank you.

tl;dr; remove -c copy to allow re-encoding or use -c:a copy only.  Then concat step can still use -c copy since all videos already re-encoded. Note that some videos with this warning still looks ok after concat and so -c copy is fine.

Using the latest ffmpeg
General users should always use ffmpeg from the current git master branch (the latest code available):

When encountering an issue the first thing to do is check to see if you are using a build from git master.
It is considered stable.
It will have more bug fixes and features.
FFmpeg development is very active.
If you want to get support from official help resources you must use git master.
Releases are for distributors.

See the FFmpeg Download page for various options, or refer to FFmpeg Wiki: Compile Guides.
Other stuff

In your command with drawbox you can steam copy the audio with -c:a copy instead of re-encoding it.
When using -ss, -to only works as expected if -ss is used as an output option.

Thank you very much, It does work better & faster.
Can you please check this one too http://video.stackexchange.com/questions/15477/can-i-run-this-in-one-command-line-ffmpeg  I'm just curious if it can be simplified.

@whitesiroi Sure. Answered, although it appears I made a commenter to that question unhappy.

Sorry for that.

As per https://trac.ffmpeg.org/wiki/Concatenate
I had this same issue.  I'm running ffmpeg version 2.6.3 on a mac.  I converted my source clips to the .MTS format with an added tag of -q 0, updated the extensions in the text file and set my output file to a .MTS when I ran the concat command in ffmpeg.  It worked like a charm.  Here is my workflow.
ffmpeg -i clip-1.MOV -q 0 clip-1.MTS

ffmpeg -i clip-2.MOV -q 0 clip-2.MTS

updated the text file (mylist.txt) to read:
file clip-1.MTS
file clip-2.MTS

then ran the concat command as follows:
ffmpeg -f concat -i mylist.txt -c copy output.MTS

Worked for me! You can convert the final output.MTS to whatever format you need.

is this protocol lossless?

No, the first command converts to MPEG-2 which doesn't have a lossless mode.

Ok, here's an answer that has worked well for me (as well as others: see link to helpful post below).
As far as my knowledge goes, there are two main things that need to match up to make the videos reasonably playable after concatenation:

The video timebase
The audio sampling frequency

Before you can match those though, you need to know where your videos are starting from.  Run ffprobe on your videos to get the video timebase value (e.g. "30k tbn") and the audio sample value (e.g. "44100 Hz"), by running:
ffprobe -i yourvideo.mp4

Now match the timebase value of your odd videos (those with a minority tbn value) as described in this post using the video_track_timescale parameter of ffmpeg to match the timebase (tbn) of your other videos.  For example:
ffmpeg -i input.mp4 -c:v libx264 -c:a copy -pix_fmt yuv420p -video_track_timescale 30000 ouput.mp4

Now for the audio, you can match the sampling frequency of the audio using the ar parameter in ffmpeg.  For example:
ffmpeg -i input.mp4 -acodec aac -vcodec copy -ar 32000 output.mp4

If your run ffprobe now on your output file, your video timebase and audio sampling frequency should match your other video, and an attempt to run concat on your videos should now work fairly well.

Upvoted for the technically correct answer but this approach - pre-encoding the inputs before running them through the concat demuxer - loses the main benefit of using the concat demuxer, which is to avoid another re-encode by using -c copy. If willing to re-encode anyway you may as well just the concat filter, which doesn't require that inputs have the same TBN/frame rate and is just one simple command.

Based on Duncan's really helpful answer:
for f in *.mp4; do ffmpeg -i ${f} -q 0 "${f%.*}.mts"; done

Then this:
ffmpeg -f concat -safe 0 -i <(for f in *.mts; do echo "file '$PWD/${f}'"; done) -c copy All.mts

Or if videos are all the same format:
ffmpeg -f concat -safe 0 -i <(find . -name '*.mp4' -printf "file '$PWD/%p'\n" | sort) -c copy all.mp4

That final command should search for the .mts files made in step one, not the original mp4s. So: ffmpeg -f concat -safe 0 -i <(find . -name '*.mts' -printf "file '$PWD/%p'\n" | sort) -c copy all.mp4

I got similar errors, and the audio in my output videos was sometimes out of sync and sometimes missing completely, but I ended up using mkvmerge from mkvtoolnix instead.
My input videos had been encoded with different settings and they had different dimensions, so I first resized and re-encoded them using ffmpeg:
for f in *.mp4;do w=1280;h=720;ffmpeg -i $f -filter:v "scale=iw*min($w/iw\,$h/ih):ih*min($w/iw\,$h/ih),pad=$w:$h:($w-iw*min($w/iw\,$h/ih))/2:($h-ih*min($w/iw\,$h/ih))/2" -c:v libx264 -crf 22 -preset slow -pix_fmt yuv420p -c:a libfdk_aac -vbr 3 -ac 2 -ar 44100 ${f%mp4}mkv;done
Some of my input videos didn't have an audio channel, so I used a command like this to add a silent audio channel to them:
for f in *.mkv;do ffprobe $f|&grep -q '1: Audio'||ffmpeg -i $f -f lavfi -i anullsrc -c:a libfdk_aac -shortest -c:v copy temp-$f;mv {temp-,}$f;done
After that I concatenated the videos:
mkvmerge -o /tmp/output.mkv $(printf %s\\n *.mkv|sed '1!s/^/+/')

If your input files have the same dimensions, the same number of audio channels, the same audio sample rate, and so on, you can use just mkvmerge to concatenate the files losslessly. The input files for mkvmerge can also have an MP4 container but the output file will have an MKV container even if its filename extension is specified as .mp4. You can run ffmpeg -i output.mkv -c copy output.mp4 to change the container.
mkvmerge has a syntax like mkvmerge -o output.mkv file1.mkv + file2.mkv + file3.mkv, where mkvmerge -o output.mkv file1.mkv +file2.mkv +file3.mkv is also accepted.

When I use ffmpeg to remux some recorded video (can be TS, MP4 or MKV) and get the non monotonous dts in output stream - timestamps etc etc error, the resulting video either appears to have audio only or plays jerkily. The only way I have found of reliably dealing with this is to use AviDemux. With Copy settings for both audio and video and selecting MP4V2 Muxer it is possible,, under the Configure option to select Optimize For Streaming. The resulting MP4 file can now be used in ffmpeg without generating error messages. Trust this is of use.


text:
Can I run this in one command line (ffmpeg)
I used this command ffmpeg -i thevideo.mp4 -c:a copy -vf 'drawbox= : x=0 : y=0 : color=invert:enable=between(t\,11\,39)' output.mp4 but it runs very slow.
So, I took another approach, I copy first and last part and run "drawbox" only on selected length, and then adding them all together.
ffmpeg -i thevideo.mp4 -ss 00:00:00 -to 00:00:10 -c copy cut.mp4

ffmpeg -i thevideo.mp4 -ss 00:00:11 -to 00:00:39 -c:a copy -vf 'drawbox= : x=0 : y=0 : color=invert' cut2.mp4

ffmpeg -i thevideo.mp4 -ss 00:00:40 -c copy cut3.mp4

ffmpeg -f concat -i mylist.txt -c copy output.mp4

content of mylist.txt:
file 'cut1.mp4'
file 'cut2.mp4'
file 'cut3.mp4'

Can I run it in one command and have the same speed or even faster but without cutting and concatenating the parts of the video?

Your current method using the concat demuxer while stream copying is probably the fastest (I'm just counting the ffmpeg time and not any time you take to enter the additional commands). Unfortunately, you have to perform several steps, but the advantage is that your non-filtered segments are not being re-encoded which preserves quality and results in fast processing.
It is possible to do it all in one command with the trim, atrim, and concat filters, but this will re-encode everything and possibly be slower. You will have to test. However, it may create a more "stable" output if you find your current method to be problematic (which would likely be fairly obvious upon playback). The filter is generally used instead of the demuxer when all streams are being filtered.
Example using concat filter:
ffmpeg -i input.mp4 -filter_complex \
"[0:v]trim=end=10[v0]; \
 [0:a]atrim=end=10[a0]; \
 [0:v]trim=10:40,drawbox=color=invert[v1]; \
 [0:a]atrim=10:40[a1]; \
 [0:v]trim=start=40[v2]; \
 [0:a]atrim=start=40[a2]; \
 [v0][a0][v1][a1][v2][a2]concat=n=3:v=1:a=1[v][a]" \
-map "[v]" -map "[a]" -movflags +faststart output.mp4

Notes

If you get a weird output refer to the setpts, asetpts filters (search for PTS-STARTPTS).
If it is still too slow use a faster x264 preset. See FFmpeg Wiki: H.264 Video Encoding Guide.
I assume you're using OS X with the "tessus" build of ffmpeg from your previous FFmpeg question. The default AAC encoder for the tessus build is libvo_aacenc. The native FFmpeg AAC encoder is generally regarded as being slightly better. If you want to use it add -c:a aac -strict experimental. If you want the best supported AAC encoder, libfdk_aac, then use homebrew to compile ffmpeg. See FFmpeg Wiki: AAC Encoding Guide.

Really, great answer, thank you very much for your great explanation, you did help me a lot.


text:
Who owns the RAW video, audio and photo footage in case the client does not pay the agreed price?
Hello to all here is my burning question: I came from Europe with all my equipment and worked for 3 months for a client. Prior to my coming there I send my terms for working and the agreed price and no complicated contract, just saying that a down payment will be needed of 30% and that food and lodging, airplane ticket and expensive s as such is not included on the price we agreed.
Now until today the client did not pay me anything and they claim I must hand over all RAW footage to them. I don't know about the USA federal law but here in Europe the video maker and in my case also photos in raw is the owner before a agreement for handing off is made.
During my stay there one of my camera's a ProHD JVC went broken the fire-wire board needs to be replaced and it will cost me over $1K
And I am more than willing to give them all the footage, including audio and photos in the case they pay. I think I am in my right.

Generally, if a person pays full price for a product (the raw footage), federal law would say that the person paying has a right to the product. And if a person does not pay for a product, they have no right to it. However, a contract between the two people could override this. It sounds as though your contract does not state what happens if there is partial payment. Does your contract state what that 30% payment is for?

Copyright law here says that the person who shoots the footage owns the footage. The only way to change this is if you sign a work for hire contract which says that the production company owns the copyright and that they'll pay you in exchange for your work and copyright. If they didn't pay you then they breached their contract. If there was no contract then you own the footage and don't have to give them anything. Something as simple as an email could be considered a legally binding contract though. I suggest you contact a lawyer who knows entertainment law in the US.
Save everything, emails, phone call records, written notes, receipts, plane tickets, reservations, anything that demonstrates your working relationship with this company and how much they have paid you and how much they owe you or should have paid you.
Sorry this sort of thing happens.
See here:
http://en.wikipedia.org/wiki/Work_for_hire


text:
How to play a clip only till an Out mark in Adobe Premiere?
In Adobe Premiere there are shortcut keys to play a clip right from the In mark to Out mark. Shortcut on a PC  is Ctrl + Shift + Space.
However, how can I play a clip only till my Out mark? Not my full selection from In to Out, but from an arbitrary point of a selection only to my Out mark? It should stop playing on my Out mark.

To "Play from Playhead to Out Point":

PC: Ctrl + Spacebar
Mac: Ctrl + Spacebar

To "Play In to Out":

PC: Ctrl + Shift + Spacebar
Mac: Opt + K

from: https://helpx.adobe.com/premiere-pro/using/keyboard-shortcuts.html

Wow, I didn't know this was possible, +1! I'll edit that into my answer as well

So there is no button for this? That's odd.

This is fantastic. Why this isn't more discoverable in the UI is beyond me.

I don't think this is possible. Edit: Turns out I was wrong. As Green mentions, CTRL+Spacebar will do exactly what you asked for. I'll leave the rest of my answer as alternatives to that.
Another option is the Play Around Shortcut, Shift+K, which will start playback a couple of seconds before your current timemarker position and stop a couple of seconds behind it. You can customize the Preroll and Postroll seconds under Preferences → Playback.


text:
Best softwares for heavy pan/zoom editing?
I use to do a lot of pan/zoom editing with Sony Vegas Pro.
Now that I read that Sony plans to discontinue Vegas (they said it will remain supported but there will never be a new version again), I am checking other softwares, so far I have tested those but I would like to hear recommandations.
(I am also making heavy use of "Velocity envelope" in Vegas, I have not tested it in other softwares yet but I'm interested in feedback about similar or alternative features through different softwares. Maybe I should make it a different question.)

Sony Vegas: Maybe it's because I used it for more than 10 years now but I think its pan/zoom tool is extremely flexible, and you can
set transition speed/acceleration on keyframes.
Adobe Premiere Pro: In their pan/zoom tools, it is possible (and seems to happen quite a lot if you do a lot of pan) to lose the corner
outside of the interface. I don't find it intuitive at all. Maybe
Premiere users are used to edit numbers rather than to manipulate the
visual tool?
Adobe After Effects: Looks like the best alternative so far as the tool's corners remain visible even when they are "behind" the
interace. Though it is very close to that of Premiere, I found this one more intuitive. From a tutorial I watched this software's tool actually looks more precise than that of Vegas, but quite more complex to manipulate and obtain similar results.
Corel VideoStudio X8: I am quite lost here and the interface gives me the feeling of a toy.

(Note: I use Windows so Final Cut is not an option.)

There's nothing firmly stated by Sony about the future of Vegas. The Catalyst suite still hasn't found its niche yet, and if all you're doing is Pan/Zoom then Vegas will keep on working regardless of future releases.

Thank you. I saw so many posts about that on Sony's forum that I though it was definitive. By the way I do other things than Pan/Zoom and Velocity envelope, but for some of my videos they're the most important parts of editing.

Update: Vegas Pro software has been acquired by Magix and they will release a new version in the Autumn.

Go with After Effects, it's a tool that will most likely stick around. If necessary you can also take advantage of the more advanced tracking an compositing features.
If you end up editing in Premiere (which pretty much is the best option if you don't want to dive into Avid) the integration between AE and Pr is crazy useful. I find myself even making titles there because it's just a faster and lets me do prettier stuff.
Let me know if you have any questions.


text:
SD card for Sony FDR-AX100E 4K?
I recently acquired a Sony FDR-AX100E.
Samsung 64 Pro+, SDXC I 10 U3 says:
Cannot record. Please use UHS-I U3 compatible memory ...

Sandisk Extreme Pro SDHC-1 U3 10 32Gb says:
This memory card does not supoprt does not support recording ofXAVC S movies.

Lexar Professional SDXC I U1 says
Cannot record. Please use UHS-I U3 compatible memory ...

But this latest is a U1 so this one makes sense. (I only recently discovered about U1 and U3.)
The manual says that to record XAVC S I need
Class 10 or faster, up to 64GB

Which card am I supposed to use ?
The camera has the latest firmware (3.00).

Solution found regarding Sandisk Extreme Pro SDHC-1 U3 10
The card was formatted to FAT32, I reformatted the card to exFat (unit size 128kb, though I am not sure at all about the impact of unit size, that's something I'll have to check) .
What is really "fun" is that the camera itself can only format the card to FAT32 so I need to format it using a computer.
I'll keep this up to date.
Update:
Even re-formatted:  Samsung 64 Pro+, SDXC I 10 U3 look like the camera won't recognize it as U3.


text:
GPU problem/replacement ? For Lightroom and Vegas
I use both Sony Vegas Pro and Adobe Lightroom.
In my new computer is an Asus Strix GTX980 GPU card.
In Lightroom: I had to disable GPU acceleration
In Vegas: I have crashes as soon as the preview windows is resized of plays video. Even with GPU acceleration disabled. (Rendering works fine but I cannot work the video that way.
From what I read, there may be something wrong with applications support of Nvidia GTX9xx .
Maybe I can exchange the GPU for another.
Do you know if that GPU may be the problem ?
What GPU in the same price range would you recommend (with 3 screens in mind, 2x4K and 1xFullHD).
Should I go AMD?
Thank you for you help.

Problem solved for Vegas
Solution A

(optional) I have installed thedriver version I wanter (in my case I took the latest drivers,359.00 currently).
CTRL+Shift on Vegas icon, checked clear cache data. (Beware that you may lose some configuration in the process.)
After maybe a few starts (can't tell precisely), it finally works, WITH GPU acceleration.

Solution B
The problem happened again. Solution this time. It might look like a random thing but I've done it multiple times and tested alternatives.
(There was no driver update this time.)

Start Vegas with a blank projet.
Display Preview Window (content should be all black).
Resize preview window (not sure if this is optional, but if it doesn't crash at this point, that's a good sign).
Open the project you initially intended to work with.

Regarding Lightroom: no solution yet.


text:
Is possible to use this "map" as a mask? (After Effects)
I'm creating animations like the hearthstone golden cards,and I'm tired of masking everything bit by bit.
I know that the hearthstone team used masks (with colors) in order to make the animations,they used a map of colors that was telling the after effects where to apply the effects and where to ignore the effects. Exemples:
Image 1:

Mask 1:

In this case,the blue area had a particles effect,the yellow/green had a glow and the redish was some rotating light effects.
In a nutshell:
How can I tell after effects to make some effects visible only in a certain spot using a color map?

Create an adjustment layer above the base image and apply one of the effects to it (you will have to do this whole process several times with one effect per adjustment layer).
Add the colour mask layer above the adjustment layer and use a key effect (Effects>Keying>whatever…) to make the relevant colour for the effect transparent.
Set the track matte for the adjustment layer to "Alpha Inverted Matte the colour mask layer".

Using an alpha inverted track matte will make the adjustment layer opaque only where the Alpha Matte is transparent, so the effect will only be visible in the areas defined by the relevant colour. Note that when you apply the track matte the layer that is acting as a matte is made invisible (eye icon switched off).


text:
Premiere Pro: export individual clips from sequence with all effects applied
I have several long videos that I need to export as individual clips, instead of a completed sequence. The goal is to have a library of clips that a coworker of mine can pull from to make Tik Tok or Instagram videos.
I've tried Render & Replace and the Project Manager's Consolidate & Transcode option, but neither achieve what I'm after. The exported clips have no color correction, no stabilization, no keyframes, and they don't reflect how they were sized/cropped within the sequence (for example the clips I filmed 16x9 but were cropped in the 9x16 sequence export as their 16x9 originals).
The only real workaround I can think of is to turn each clip into a nested sequence and then select/export those from the Project window. That will become a time consuming process and I'm hoping there's a faster way.
TL:DR - Is it possible to batch export individual clips from a sequence with the effects, sizing, etc. applied?

The nesting method I mentioned does work...and it's as time consuming as I figured it'd be.

This would be a great feature to include with media management - I'm very surprised it doesn't work with "render and replace". If R&R does work on a nested clip/effect though, there may be a faster spin on your workaround: Could you nest the entire timeline, run 'scene edit detection' on the long nested sequence to recreate your cut points, and then render/replace to get a collection of individual clips with effects applied?


text:
Easily make loudness more uniform in Premiere Pro
I'm having to edit some speech videos (60+ min each) I was given from a conference and the audio levels vary pretty wildly for some of them, due to different speakers and the sound guy not doing a good job with the levels.
In FCPX, the Loudness effect has a slider for Uniformity, but I'm not sure how to easily accomplish a similar task in Pr.
I set the audio type as dialogue and then clicked Auto-Match under the Loudness tab.  I also added a Multiband Compressor and enabled Loudness Radar, but I'm having difficulty balancing the loudness in a track where it varies so greatly, as you can see below

My Question:
Is there a way to automatically (or easily) make the loudness of this clip more uniform?
I know I can go through and manually keyframe it or split the track into sections of equal loudness then balance them all, but I was hoping for quicker approach.

Dynamics Processing effect, maybe?   I'm not really an audio guy.

This is a copy and paste from a similar previous question which I answered:
In order to even out the audio of a single stream; you will want to either mix by hand as noted above; or you can use dynamics; specifically Compression.
Within the PPro audio mixer, apply dynamics to the audio tracks, or master track.
You can then use the compressor within the dynamics filter, which you can use on all your tracks, to get your dynamic range all in line.
The simplest way to learn this is just trial and error; but looking at your waveform; you will want to:
Adjust the gain up- it's a pot/knob center column bottom. Use a ratio of about 2:1 to 6:1.
Last, click the checkbox on the bottom right of the dynamics filter for each track to enable the limiter. The limiter; a good setting is -2 db.
Then you want to adjust your threshhold in terms of attack and release, what you want to see on your audio (do your setup/testing on the loud sections and soft sections to find the sweet spot); is your compressor VU meter should just barely be jumping... when say someone is speaking. If it's spiking to high; gain down. If it's not jumping at all; gain up.
This will balance out your audio so the soft sounds are adjusted up and can be heard, and the loud sections are not adjusted up; resulting in overmodulation. Its a much better way to smooth out your audio, decrease the dynamic range, and end up with audio that sounds far more balance.
Once you learn how to effectively use the dynamics filter on tracks, I think you will find you will be far more effecient in mixing your audio on the fly. It really does a decent job to just get everything in balance. Then; after you've done that, you can do back and sweeten the audio with a full mix; using a control surface or the pots, recording the pot movements via Touch Setting.

I'm not sure about leveling the audio automatically. You could probably find an effect that does this, after all audio normalisation is a pretty common feature even in audio players, let alone editors. However, I doubt that it would give the exact result you want and given that you are using Premiere Pro, you will probably want more flexibility and manual control.
There is an easy way to do this on the go while watching the video though, which is the Audio Track Mixer. Note that using this method requires you to use one audio track per audio type (e.g. one track each for voice over, ambience and music), which is advisable anyway.

First, open the Audio Track Mixer for your sequence by selecting it in the menu under Window → Audio Track Mixer. There you will see all the tracks in your sequence, along with volume controls for every one of them and the master track (all individual audio channels are mixed down into a master track for the export, but that is another topic entirely). Now, you want to change the Automation Mode of the track with the audio you want to adjust from the default setting Read to Latch, Touch or Write. Then you can simply start playback using the contols on the bottom of the panel and use the volume controls to adjust the volume on the fly. Premiere will save the set volume levels over time.
The Automation Mode setting changes how the volume controls behave:

Off Ignores the track’s stored settings during playback. Off allows real-time use of Audio Track Mixer controls without interference from existing keyframes. However, changes to the audio track aren’t recorded in Off mode.
Read Reads the track’s keyframes and uses them to control the track during playback. If a track has no keyframes, adjusting a track option (such as volume) affects the entire track uniformly. If you adjust an option for a track that’s set to Read automation, the option returns to its former value (before the current automated changes were recorded) when you stop adjusting it. The rate of return is determined by the Automatch Time preference.
Write Records adjustments you make to any automatable track settings that aren’t set to Safe During Write, and creates corresponding track keyframes in a Timeline panel. Write mode writes automation as soon as playback starts without waiting for a setting to change. You can modify this behavior by choosing the Switch To Touch After Write command from the Audio Track Mixer menu. After playback stops or a playback loop cycle is completed, the Switch To Touch After Write command switches all Write mode tracks to Touch mode.
Latch Identical to Write, except that automation doesn’t start until you begin adjusting a property. The initial property settings are from the previous adjustment.
Touch Identical to Write, except that automation doesn’t start until you begin adjusting a property. When you stop adjusting a property, its option settings return to their previous state before the current automated changes were recorded. The rate of return is determined by the Automatch Time audio preference.

Source: Documentation
For speeches, I would recommend you use Write, Touch is better suited for music or other more dynamic audio.

Great answer, although it's still more manual than I was hoping for.  I think I might just end up splitting the long audio track into smaller sections based on their loudness, then normalizing all of the individual clips, so they are at the same level.  My issue with this method is that the speeches are all over an hour long and I have about 10 to do, so I want a quicker method.  If no one else comes up with anything, I'll mark this as accepted in a few days.


text:
FFmpeg for fade in doesn't get applied to the final video
This is the way I'm trying to apply fade in inside a filtercomplex of FFmpeg command.
[v]fade=t=in:s=0:n=30[vfadein]

This command doesn't return error but the result is a video without fade in effect. What's the problem?

What's the value of startCut.ToString()

Gyan thanks for the reply. It is 00:00:34.2210000

So your fade happens in the first 30 frames, but you only encode output from 34 seconds. Add a trim filter before the fade, trim='00:00:34.221000'.

and duration is 00:00:06.1600000.

Gyan I've added this line to the command but the output is only audio no video ,[finalv]trim="+startCut.ToString()+ "[finalvtrimmed]"+

What's your full cmd?

Thanks, it works. [v]fade=t=in:st=34:n=30[vfadein]


text:
FFMpeg concat demuxer corrupted output (freezed video on some concatenated parts)
I have big videos with different types of properties and trying to collect some part of these videos by ffmpeg.
I use this command to cut specific parts:
scale video >> scale png >> pad scaled_video >> add icon to padded_scaled_video >> encode.

ffmpeg.StartInfo.Arguments = @"/c ffmpeg.exe -i " + VideoIn + " -i " +
  imageIn + " -filter_complex \"[0:v]
  scale='if(gt(a,6/4),480,-2)':'if(gt(a,6/4),-2,320)' [scaledv], [1:v]
  scale=iw*.5:ih*.5 [scaledicon], [scaledv]
  pad=480:320:(ow-iw)/2:(oh-ih)/2:black [scaledpaddedv],
  [scaledpaddedv][scaledicon] overlay=25:25,setsar=1:1 [finalv]\" -map
  \"[finalv]\" -map 0:a" + " -ss " + startCut.ToString() + " -c:v
  libx264 -preset slow -crf 20 -c:a aac -t " + duration.ToString() + " "
  + VideoOut + " -y -report";

Which works perfectly.
and then try to merge these videos with this command:

ffmpeg.StartInfo.Arguments = "/c ffmpeg.exe -f concat -safe 0 -i \"" +
  concatTextDirectory + "\" -c copy -vf \"subtitles=\\'" + videoInSrt +
  "\\':force_style='FontName=DejaVu Serif,FontSize=26'\" -c:v libx264
  -preset slow -crf 20 -c:a aac " + videoOut + " -y -report";

Which again works ok if cutted video parts are from the same type of video, and if they aren't there will be no video and only audio or jittery video on non-similar video parts.
The first cutted video part is dominant on other parts.
Question is most of the time video parts are not from the same video type and output is not playable. How could I cut video parts to be similar in properties to work without problem on concatenation part?
This is -report log output:
https://drive.google.com/open?id=0B_kWzdnSIxuQY09QWWZGVTBPdHM

Show the output of ffprobe for each of the input clips.

https://drive.google.com/open?id=0B_kWzdnSIxuQTGR1M3BnSzN5ak0 ------  https://drive.google.com/open?id=0B_kWzdnSIxuQT3VWR3cwSDJGbUk ------  https://drive.google.com/open?id=0B_kWzdnSIxuQczFJeV9TTzlOUms

End up using filters as this answer guides >> https://video.stackexchange.com/a/15479/10894
ffmpeg.StartInfo.Arguments = "/c ffmpeg.exe -i C:\\Users\\mtst\\Desktop\\Clips\\h\\a1.mp4 -i C:\\Users\\mtst\\Desktop\\Clips\\h\\a2.mp4 -i C:\\Users\\mtst\\Desktop\\Clips\\h\\a3.mp4 -filter_complex \"[0:v]setpts=PTS-STARTPTS[v0]; [0:a]asetpts=PTS-STARTPTS[a0]; [1:v]setpts=PTS-STARTPTS[v1]; [1:a]asetpts=PTS-STARTPTS[a1]; [2:v]setpts=PTS-STARTPTS[v2]; [2:a]asetpts=PTS-STARTPTS[a2]; [v0][a0][v1][a1][v2][a2]concat=n=3:v=1:a=1[v][a]\" -map \"[v]\" -map \"[a]\" -c:v libx264 -preset slow -crf 20 -c:a aac " + videoOut + " -y -report";

Glad you fixed your problem. For anyone else coming across the question...
The problem is that concat demuxer only works when multimedia streams are the same.
As described here, all files must have the same streams (same codecs, same time base, etc.).
So you have to use a filter, forcing a re-encode. See this reference.

I was going to comment on paulzag's comment, but I need 50 reputation (what a stupid system).
For me, the filter and demuxer weren't working at all, but I found out about "Concat protocol" on ffmpeg's site, which worked prefectly for me: https://trac.ffmpeg.org/wiki/Concatenate#protocol
I used the "intermediate files" method, which uses MPEG-2 TS files.
Previous to this, the video was freezing at the first insertion for about 3 seconds, and then playing the rest of the video at half FPS. I think it was due to the videos being encoded using different settings.
Now it perfectly combines both videos together with no issue.
From the site:

While the demuxer works at the stream level, the concat protocol works at the file level. Certain files (MPEG-2 transport streams, possibly others) can be concatenated. This is analogous to using cat on UNIX-like systems or copy on Windows.
Instructions
The following command concatenates three MPEG-2 TS files and concatenates them without re-encoding:
ffmpeg -i "concat:input1.ts|input2.ts|input3.ts" -c copy output.ts

Using intermediate files
If you have MP4 files, these could be losslessly concatenated by first transcoding them to MPEG-2 transport streams. With H.264 video and AAC audio, the following can be used:
ffmpeg -i input1.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts intermediate1.ts
ffmpeg -i input2.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts intermediate2.ts
ffmpeg -i "concat:intermediate1.ts|intermediate2.ts" -c copy -bsf:a aac_adtstoasc output.mp4

Using named pipes to avoid intermediate files
If you're using a system that supports named pipes, you can use those to avoid creating intermediate files. This sends stderr (to which ffmpeg sends all the written data) to /dev/null, to avoid cluttering up the command-line:
mkfifo temp1 temp2
ffmpeg -y -i input1.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts temp1 2> /dev/null & \
ffmpeg -y -i input2.mp4 -c copy -bsf:v h264_mp4toannexb -f mpegts temp2 2> /dev/null & \
ffmpeg -f mpegts -i "concat:temp1|temp2" -c copy -bsf:a aac_adtstoasc output.mp4

The additional -y switch is needed to force ffmpeg to write to existing files temp1 and temp2, which are the named pipes. Without the switch, the first two ffmpeg programs running in the background will not produce any output because they wait for interactive yes/no answers to the questions whether to overwrite existing files.
All MPEG codecs (MPEG-4 Part 10 / AVC, MPEG-4 Part 2, MPEG-2 Video, MPEG-1 Audio Layer II, MPEG-2 Audio Layer III (MP3), MPEG-4 Part III (AAC)) are supported in the MPEG-TS container format, although the commands above would require some alteration (e.g., the -bsf bitstream filters will have to be changed).


text:
Preset/macro for complicated nested effect
I have a few animations that I need to do with some frequency, the steps are as follows:

Generate the image from an external program from a template.
Add the image to the second nest in a doubly-nested sequence.
Determine how long the image should play and apply two animation (motion) presets I made, clamped to the in and out of the image. The animation presets are the same, but since they depend on the in and out I have to determine the length first. (Scaling messes up some of the intermediate keys).
Add some audio at the beginning and end of the image (same audio, but clamped to the in and out).
Add a matte image (same every time) to the first nest, and apply a track matte effect on the second nest referring to it (same every time).

The image is always of the same resolution.
This is very time consuming and I have to do it 5-10 times per video I make. Is there any way I can build some sort of macro that does this (except part 1, obviously)? Preferably it would automate as much as possible. Ideally, I could give it an image and a length parameter and it would generate the whole thing, but even automating just the nesting and track matte would save a lot of time.
Can I make a macro/preset that I can use across projects that does this? I don't have any experience with the Premiere API, but I have programming experience if this would be really straightforward by writing a plugin.

I'd try to export a sample project as XML and then let a script change the necessary parts of the XML project file.

I've done this with Final Cut Pro XML. A good text editor designed for working with XML is helpful (e.g. a programmer's text editor like Atom, sublime text, textWrangler or notepad++).

Unfortunately the Premiere API seems to be almost completely undocumented by Adobe, except for this pdf on Github I don't understand why. There's lots of info out there on scripting After Effects using the Extendscript Toolox (based on javascript), and a lot of it is similar for premiere.
However if you use the ExtendScript ToolBox's data browser you can see all the objects and methods there (you have to have Premiere running and choose Premiere from the drop-down in the top left as your target).

I'm still dipping my toe in the water as far as scripting premiere goes, so that's as far as I can help you.


text:
Nikon D5200 sharpness issue
I've recently been experimenting with the video feature of my D5200 and noticed this issue with the sharpness of the recordings:
2x video crop screenshot:

2x photo crop screenshot

As image width is limited on stackexchange you may want to look at the screenshots in an extra tab for the full resolution. Both shots were taken using the exact same camera settings. I scaled the photo to match the video and I don't think it was sharpened while downscaling. Oh and by the way, I am using the 40mbps firmware patch, but I don't think that should decrease video sharpness.
Has anybody got an idea why this is?

My guess is that it's simply because the photo was shot in a much higher resolution than the video. Even though they were downscaled to the same resolution, a (previously) higher-resolution image will look better in this situation in the same way that a clip shot in 4K will look better on a 1080p monitor than a clip shot in 1080p on the same monitor.


text:
What's wrong with this MP4 file? Is the pixel aspect ratio wrong?
Similar to this post, I've sent some video tapes off to be digitized.  In this case, they are mini-DV tapes which should be 720x480 resolution (more or less).  The resulting MP4 files do claim to be 720x480 but the images don't look right.  I uploaded one such movie here, it's a little hard to tell but even for dancers they aren't shaped quite like that.  The video makes them look more like elves than humans, which is probably better than looking like hobbits, but still not right.  If I play this MP4 using the standard Windows player, pause it, and grab a screenshot, the resulting image is 1118x832 which is an aspect ratio of 1.34 which explains the elvishness.  But why?
I was not familiar with the concept of pixel aspect ratio until I started reading up on DV format but I'm guessing that's what's happening here.  The Wikipedia page on DV says there are "different pixel aspect ratios for fullscreen and widescreen video" but doesn't say what those values are.
Since Windows is stretching this video when it plays it, it makes me think that there is a pixel aspect ratio embedded in the MP4 file, but the standard Windows "Show Properties" doesn't display it.  Or maybe Windows always plays videos using a 1.33 pixel aspect ratio or something?
What I'm thinking, is that since DV is recorded with 480 scan lines, and the resulting image is 480 pixels high, they don't need to re-digitize the video.  I could drop the MP4 into my video editing software (Sony's Vegas Movie Studio) and hit it with a "compress vertically" deform, but it seems like that's going to end up blurring the video.  I have all the right pixels, it's just that my pixels aren't the right shape (right?)  Is the pixel aspect ratio stored in this MP4 file in a place where I could adjust it and get my video to look right?
EDIT
Initially, ffmpeg reports this for my video :
Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 720x480 [SAR 10:11 DAR 15:11], 2886 kb/s, SAR 109:120 DAR 109:80, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
A DAR (display aspect ratio) of 109:80 is 1.36 which is basically correct for full screen DV, so I'm guessing my video is actually wide screen (16:9 = 1.77777).  If I run ffmpeg using this command:
ffmpeg -i input.mp4 -aspect 1.7777777 -c copy output.mp4
I get a video which ffmpeg describes like this:
Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 720x480 [SAR 10:11 DAR 15:11], 2886 kb/s, SAR 32:27 DAR 16:9, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
which looks almost OK - it says DAR 16:9 but then also says DAR 15:11 which is 1.36 and not what I want.  And sure enough, if I play this video using Windows 10 "Movies & TV", the screenshot shows that the video is still at the 1.36 DAR, not the desired 1.77.
Is there an ffmpeg command that will produce what I need?  Or is the Windows 10 Movie Player unable to actually handle the DAR setting?

You are correct. The pixel aspect ratio is wrong. DV has a fixed horizontal resolution, but can display 16:9 frames by stretching out the pixels. The display resolution is set in metadata, so it is possible to make the file play back properly without re-encoding by changing the metadata.
The free, open-source tool ffmpeg can do it. This would be the command you would use:
ffmpeg -i infile.mp4 -c:copy -aspect 16/9 outfile.mp4

Awesome!  Before I make any changes, ffmpeg reports "Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 720x480 [SAR 10:11 DAR 15:11], 2995 kb/s, SAR 109:120 DAR 109:80, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)"

Why are there two values reported for both Display Aspect Ratio and Sample Aspect Ratio?

Sample aspect ratio is the shape of the pixels. Display aspect ratio is the shape of the frame. The two are related, but not the same: DAR = SAR * X / Y where X and Y are the number of pixels across and down.


text:
Sampling and using afftdn with ffmpeg - how with complex filters?
I'm merging together a bunch of videos (30 so far) into a "Brady Bunch" style amalgam using ffmpeg, and I've got that working reasonably well.  The last big problem is the background noise.  Everybody recorded at different locations, and some of those locations had computer fans in the background and some had air conditioners, and when you add them all together, the summed background noise is really noticeable.
I'm doing everything with complex filters, here's a rough idea what I've got going on:
color=size=1920x1080:c=0x000080 [base]
[0:a]volume=0.90,aresample=async=1:first_pts=0[aud0]
[1:a]volume=2.08,aresample=async=1:first_pts=0[aud1]
[2:a]volume=1.48,aresample=async=1:first_pts=0[aud2]
[3:a]volume=2.53,aresample=async=1:first_pts=0[aud3]
[4:a]volume=1.34,aresample=async=1:first_pts=0[aud4]
[5:a]volume=1.81,aresample=async=1:first_pts=0[aud5]
(and a bunch more adjusting the volume of each input video)
[28:a]volume=1.06,aresample=async=1:first_pts=0[aud28]
(and then we start the videos, at first only a subset are visible and then the large group
joins in, so I "zoom in" on the small group and then expand to bring in everyone else
when it's their turn)
[0:v] scale=w=if(lte(n\,1200.00)\,576\,if(lte(n\,1410.00)\,576.0-288.00*(n-1200.00)/210.00\,288)):h=if(lte(n\,1200.00)\,324\,if(lte(n\,1410.00)\,324.0-162.00*(n-1200.00)/210.00\,162)):eval=frame [clip0]
[base][clip0] overlay=shortest=1:x=if(lte(n\,1000.00)\,-608\,if(lte(n\,1175.00)\,-608.0+624.00*(n-1000.00)/175.00\,16)):y=if(lte(n\,1000.00)\,-342\,if(lte(n\,1175.00)\,-342.0+369.00*(n-1000.00)/175.00\,27)) [over0]
[1:v] crop=870:917:553:163,scale=w=if(lte(n\,1198.80)\,307\,if(lte(n\,1408.59)\,307.0-123.00*(n-1198.80)/209.79\,184)):h=if(lte(n\,1198.80)\,324\,if(lte(n\,1408.59)\,324.0-130.00*(n-1198.80)/209.79\,194)):eval=frame [clip1]
[over0][clip1] overlay=shortest=1:x=if(lte(n\,1000.00)\,166\,if(lte(n\,1175.00)\,166.0+222.00*(n-1000.00)/175.00\,388)):y=if(lte(n\,1000.00)\,-342\,if(lte(n\,1175.00)\,-342.0+353.00*(n-1000.00)/175.00\,11)) [over1]
(you get the idea)
(and then I merge all the audio together)
[aud0][aud1][aud2][aud3][aud4][aud5][aud6][aud7][aud8][aud9][aud10][aud11][aud12][aud13][aud14][aud15][aud16][aud17][aud18][aud19][aud20][aud21][aud22][aud23][aud24][aud25][aud26][aud27][aud28]amix=inputs=29[a]
[a]adelay=delays=200:all=1[b]
(I've run a first pass of loudnorm to measure the output levels, and I send them back in)
[b]loudnorm=linear=true:I=-14:LRA=11:tp=-2:measured_I=-30.18:measured_LRA=17.80:measured_tp=-15.23:measured_thresh=-41.39:offset=-0.77:print_format=summary[level]

What I'm thinking, is that most everybody starts out with a few seconds of silence.  It seems like I should be able to measure the audio in that segment, and then use it to filter out the background per stream.
It seems like afftdn will do that, but I cannot for the life of me figure out how to use it.  Apparently I need to send commands to it, as I have read here, but it's not at all clear how I make that work inside my complex filter.  Something to do with ZMQ?  Is ZMQ what's happening with all those "@n"s?
EDIT
The answer that Gyan refers to is this:
ffmpeg -i in -i profile -filter_complex "[1][0]concat=n=2:v=0:a=1,asendcmd=c='0.0 afftdn@n sn start; 1.0 afftdn@n sn stop',afftdn@n,atrim=1,asetpts=PTS-STARTPTS" noiseout.wav

Let me see if I can break this down and understand it.
In that question, [1] is the noise profile and [0] is the input audio.
[1][0]concat=n=2:v=0:a=1

Means concatenate the two audio streams, noise profile first, and generate as an output only an audio stream.
Then we get ready to send commands to afftdn, using the ZMQ syntax:
asendcmd=c='0.0 afftdn@n sn start; 1.0 afftdn@n sn stop'

In ZMQ, you have to give a filter an id so it knows which filter you are sending the command to.  In this case, we've assigned the one and only afftdn the id 'n', and we are telling it to start measuring the noise profile at time 0 (the default so not specified) and stop measuring at time 1.
So now we actually invoke the afftdn filter:
afftdn@n

Then we trim off that first second of the audio stream (which is just the noise profile):
atrim=1

And finally, we want to reset the timestamp of the audio stream so that 'zero' is the new position (after the noise profile):
asetpts=PTS-STARTPTS

SO
For now at least, I'm going to always use the first second of the audio as my noise profile.  Since it's already at the start of the audio, I don't need to isolate it, reset its timestamp, and concatenate it at the start of the audio (right?).  So taking audio stream [aud0], I can do this:
[aud0]asendcmd=c='0.0 afftdn@n sn start; 1.0 afftdn@n sn stop',afftdn@n[faud0]

and then send [faud0] into the 'amix' instead of [aud0]
Yes?
SECOND EDIT
I'm now duplicating the first second of each audio stream and appending it on the front, just like in Gyan's answer.  Here is one such audio stream:
[19:a]volume=31.59,aresample=async=1:first_pts=0[aud19]
[aud19]asplit[aud19a][aud19b]
[aud19a]atrim=0:1[aud19c]
[aud19c][aud19b]concat=n=2:v=0:a=1,asendcmd=c='0.0 afftdn@t sn start; 1.0 afftdn@t sn stop',afftdn@t=nr=90,atrim=1,asetpts=PTS-STARTPTS[aud19d]

Notice that since I've got so many audio streams, and each one has its own afftdn filter, I've given them all separate ids.
And still, the output audio does not sound significantly different from when I started!
THIRD EDIT
So I grabbed one of the videos that has a lot of noise at the front, kind of sounds like lots of air moving, maybe some 60 Hz fan noise too.  I extracted the first 9 seconds into a file I cleverly named "trim.wav".  I then ran ffmpeg twice, once normally, and once to output just the noise.  The "just noise" file is totally devoid of sound, as if it didn't find any noise to work with.  And as expected, the (supposedly) processed file sounds identical to the input.
Here are my command lines:
c:ffmpeg -y -i trim.wav -filter_complex "[0:a]asplit[aud0a][aud0b];[aud0a]atrim=0:1[aud0c];[aud0c][aud0b]concat=n=2:v=0:a=1,asendcmd=c='0.0 afftdn@aa sn start; 1.0 afftdn@aa sn stop',afftdn@aa,atrim=1,asetpts=PTS-STARTPTS[aud0d]" -map "[aud0d]" processed.wav
c:ffmpeg -y -i trim.wav -filter_complex "[0:a]asplit[aud0a][aud0b];[aud0a]atrim=0:1[aud0c];[aud0c][aud0b]concat=n=2:v=0:a=1,asendcmd=c='0.0 afftdn@aa sn start; 1.0 afftdn@aa sn stop',afftdn@aa=om=n,atrim=1,asetpts=PTS-STARTPTS[aud0d]" -map "[aud0d]" noise.wav

Here's what ffmpeg had to say:
ffmpeg version git-2020-08-07-fab00b0 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 10.2.1 (GCC) 20200805
 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libgsm --enable-librav1e --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil      56. 58.100 / 56. 58.100
 libavcodec     58. 99.100 / 58. 99.100
 libavformat    58. 49.100 / 58. 49.100
 libavdevice    58. 11.101 / 58. 11.101
 libavfilter     7. 87.100 /  7. 87.100
 libswscale      5.  8.100 /  5.  8.100
 libswresample   3.  8.100 /  3.  8.100
 libpostproc    55.  8.100 / 55.  8.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'trim.wav':
 Metadata:
 encoder         : Lavf58.49.100
 Duration: 00:00:09.00, bitrate: 1411 kb/s
 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
Stream #0:0 (pcm_s16le) -> asplit
 asetpts -> Stream #0:0 (pcm_s16le)
Output #0, wav, to 'processed.wav':
 Metadata:
  ISFT            : Lavf58.49.100
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  Metadata:
   encoder         : Lavc58.99.100 pcm_s16le
     [afftdn @ 00000261c3b81400] bn=4 -6 -10 -12 -12 -14 -17 -23 -24 -24 -24 -24 -24 -24 -15
     [afftdn @ 00000261c3b81400] bn=5 -4 -8 -10 -11 -14 -18 -23 -24 -24 -24 -24 -24 -24 -15
  size=    1545kB time=00:00:08.97 bitrate=1411.3kbits/s speed= 150x
video:0kB audio:1545kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.004929%

FOURTH EDIT
Based on comments, I tried adjusting the noise floor and also using just a single audio track.  I ran these commands:
c:ffmpeg -y -i trim.wav -filter_complex "[0:a]asendcmd=c='0.0 afftdn@aa sn start; 1.0 afftdn@aa sn stop',afftdn@aa=nr=90[aud0d]" -map "[aud0d]" output-nr90.wav
c:ffmpeg -y -i trim.wav -filter_complex "[0:a]asendcmd=c='0.0 afftdn@aa sn start; 1.0 afftdn@aa sn stop',afftdn@aa=nr=1[aud0d]" -map "[aud0d]" output-nr1.wav
c:ffmpeg -y -i trim.wav -filter_complex "[0:a]asendcmd=c='0.0 afftdn@aa sn start; 1.0 afftdn@aa sn stop',afftdn@aa=nr=90:nf=-20[aud0d]" -map "[aud0d]" output-nr90-nf20.wav
c:ffmpeg -y -i trim.wav -filter_complex "[0:a]asendcmd=c='0.0 afftdn@aa sn start; 1.0 afftdn@aa sn stop',afftdn@aa=nr=1:nf=-20[aud0d]" -map "[aud0d]" output-nr1-nf20.wav
c:ffmpeg -y -i trim.wav -filter_complex "[0:a]asendcmd=c='0.0 afftdn@aa sn start; 1.0 afftdn@aa sn stop',afftdn@aa=nr=90:nf=-80[aud0d]" -map "[aud0d]" output-nr90-nf80.wav
c:ffmpeg -y -i trim.wav -filter_complex "[0:a]asendcmd=c='0.0 afftdn@aa sn start; 1.0 afftdn@aa sn stop',afftdn@aa=nr=1:nf=-80[aud0d]" -map "[aud0d]" output-nr1-nf80.wav

and for the most part I don't hear any noticeable difference.  However, with the filter 'afftdn@aa=nr=90:nf=-20' I hear what sounds like a 1-second reprieve from the noise at the very beginning of the audio, as if that noise was being filtered out while leaving all of the noise in the rest of the track.  Am I just using the wrong filter?  Is my noise not the type of noise that afftdn is effective on?  Or am I somehow misusing the filter so it's only filtering the first second?  For reference, here is the input track.

You would use the 2nd method in my answer. Instead of a separate profile input, you would split each audio stream into two. From one copy, use atrim to isolate a portion useful for profiling, then reset its timestamps using asetpts. Then concat it with the 2nd full copy. Execute commands using asendcmd as shown in the answer, then afftdn. Then use atrim to discard that segment and restore timestamps.

Something doesn't seem quite right.  If I change the final afftdn filter to afftdn@n=om=n (to just hear the noise) it works exactly as expected (gives me just the noise) but I'm not noticing a big difference in audio output even when I use afftdn@n=nr=30

No difference in output even with afftdn@n=nr=80

Try with a minimal command (say, 2 audio streams) and check. Add full command and full logs if it doesn't work.

@Gyan Thanks for your help with this!  I've edited my original question, and uploaded some smaller WAV files to my Google Drive so you have everything that I'm working with.

@PaulB.Mahol I agree that the split/concat seems like extra work in this case, unfortunately the syntax here is so tricky that I have no idea how to make afftdn work without splitting the stream.  Can you help provide the syntax I would need?  Thanks!


text:
Digitized super 8 mm film - got wrong aspect ratio?
I had some old super 8mm film digitized, and the result looks almost right.  I did a little poking around and I'm wondering if they squished the film into a standard (but wrong) aspect ratio.
The resulting video files claim to be 720x480 which would be a ratio of 1.5, but according to Wikipedia, the image portion of a super 8mm frame is 5.79 x 4.01 mm which is a ratio of 1.44.  If they did force it into a ratio of 1.5, that would be off by about 5% which would be noticeable.
The video files are of course too large to upload, but I used "Photos" (comes free with Windows) to snip out a 4-second chunk.  I've got other video editing software, but thought Photos had the best chance of not doing anything to the video except extracting the desired scene.  It's shared on my Google Drive here.
Just for fun, I laid the super 8mm film across my negative scanner and got an image, since the scan was at 2400 dpi I did the math and the images are about 5.58 x 4.27 mm which is approximately what you expect for super 8mm.
Am I overthinking this?  Or should I take the time to drop each video file in my editor and squish it into the right shape?

EDIT
After posting I had a thought - what if they digitized it at the right aspect ratio by trimming?  So I found the image in the video that matched the frame I had scanned on the original film:

At first it looked like I had the answer, as they had clearly trimmed my mother's hair in the digitized video.  However, by zooming in on the encyclopedias on the shelf:

In the scanned film, they are 99x69 = ratio of 1.434; in the digitized video they are 156x94 = ratio of 1.660.  So yes, the aspect ratio was in fact altered during the digitization process, and by about the amount expected from my original calculations.

Pixels are not necessarily square, they are rectangular. For example, both "standard" and "widescreen" formats on DVD have the same frame size in pixels, but widescreen pixels are wider. To complicate the matters, out of 720x480 frame only the 704x480 portion is expected to have content, and it is this portion that should have required display aspect ratio (DAR). In case of old-school "standard" TV it would be 4/3, it case of modern widescreen TV it would be 16/9.
Depending whether you need "standard" or "widescreen" format, pixels may be narror (PAR 0.9 for "standard") or wide (PAR 1.21 for "widescreen").
Super 8 has DAR about 1.36, so when the video is displayed correctly in a 1.33 frame, you should see thin black bars on top and bottom. But really, it is so close to 1.33 that you may want to scale it to full frame.
If you see your video shown wider than it should be, then either the video is missing PAR information, or the video player does not process PAR information properly, or both.

Yes, digitization screws up the aspect ratio.  See the reasons for my conclusion after "EDIT" in the question.


text:
Is HandBrake tune -still image applicable to videos with minimal movement?
I have some educational videos that are mostly PowerPoint slides and a teacher sitting in a chair talking.
Is the -still image tune applicable for videos like that? I would like to make them as small as possible.

Ironically, I find that with no tune specified, I get smaller sizes, all else being equal, for slideshow videos. Although since you do have some live video in your frame, I would suggest skipping the tune.

@Mulvya Make it a answer then!

My experience with handbrake mirrors Mulvya`s comment
I also get smaller file sizes with X264 Tune set to None on the Video Tab.


text:
How can I remove the timer of the preview screen?
How can I remove timer from the lower center of the preview screen (circled in red) of adobe after effects?

AfterEffects does not natively have a timecode display at that place. It is most likely a part of your composition. If it is, you should be able to click on it and determine what layer or effect should be switched off.

Perhaps you have mistakenly enabled "Timecode effect", which must be disabled if you want to get rid of that burn-in window.  You might think this would be a toggle in a top-level menu, but you have to navigate to the effects window and disable it from there.


text:
How to rotate object in After Effects without affecting other values?
I have a motion path of a object already defined within keyframes. Now I would like to adjust the rotation. BUT if the rotation is set, it affects also the X/Y axis of the object, that's why it completely screws up the created keyframes for position coordinates (if the object was moving upwards, I rotate it 90 degrees, it moves to the left).
So my question is, how can I rotate the object without affecting the position, etc. Thanks.

The anchor point defines the "center" around which an object in AE moves, rotates etc. Often, when not taking care of initially setting the anchor point to the right location, rotating the object can lead to surprises. For simple position changes, you probably don't really care where the anchor point is set, once other Params get into play, this can lead to problems since effectively can look as if it affects position, in case of an object with transparency.

Hans is right - make sure the anchor point of the object you wish to rotate is positioned correctly before you make your movement keyframes.  Use the Pan Behind Tool (stupid name) to reposition the anchor point.


text:
How can encode video with images like these [overlay a video over a fixed image] with ffmpeg?
How can I use ffmpeg to composite a video image with a still image to create a final image such as the one below (where the video image "plays" on the large screen and the still image creates the illusion of a theatrical audio environment)?

not a another ? hi

Sorry, don't understand.

Asked and answered : How can encode video with images like these (overlay a video over a fixed image) with ffmpeg?
But you want to add coordinates to the movie file to scale to a specific area.
For the perspective/scale:
https://ffmpeg.org/ffmpeg-filters.html#toc-perspective
For the overlay:
https://ffmpeg.org/ffmpeg-filters.html#toc-overlay-1


text:
MILC or camcorder for ENG?
I'm going to shoot a lot of Electronic News Gathering stuff like demonstrations, reports. But also interviews and longer events like speeches and talks. I can't decide between a camcorder and a MILC (mirrorless interchangeable lens camera) because both have their pros and cons. I like in the MILC that it has a lot of possibilities to develop later.
I have a candidate for both categories. The JVC Everio GC-PX100 as a general camcorder and the Sony Alpha 7 as a MILC. I like about the Sony that it's a full-frame, very serious camera. But I'm afraid that it lacks some important features or not very good at some fields, like OIS or autofocus. I never worked with a DSLR or MILC before.
The JVC would be a safe bet but also a great loss in features due to the fixed lens and small sensor.

You'd probably be best off with a shoulder mount or large size camcorder. The Sony most likely has too shallow depth of field so it will be harder to keep things in focus. If something unexpected happens you want to be able to react quickly. Camcorders are made for this.

If you record speeches that are longer than 30 minutes, the 29m59s limitation of non-video cameras is going to drive you crazy, despite all the other wonderful things they can do.  Interviews can be broken up every 30 minutes.  Speeches not.
If you have ambitions to encompass art projects with your camera, the interchangeable lens feature of the SONY camera opens a world of possibilities (and and bottomless pit of budget requirements).  If you are really focused on ENG then the JVC packs an incredible punch in a single package.  Plus, small sensors are your friend for ENG because of deep depth of field.  It's incredibly annoying to have a speaker in front of a screen and the camera can only focus either on the speaker or their slides.
Based on the things you say you want to do, not the things you say you want, the JVC is the more sensible choice.

If you hope to record ENG footage for TV submission, make sure that the camera can record in a format that is permitted by the stations you hope to be working for. In the case of the station I work for, news footage can only be submitted as interlaced (1080i50) and a lot of photocameras with video function can not shoot interlace video.

My main target is youtube but thanks for the comment because I had an idea to submit some of my material to TV. In that case I would convert the progressive video to interlace during the post production.

Another point that Michael didn't mention is ergonomics. Using a MILC camera means that you will need an external monitor (the viewfinders are never good enough to judge critical focus, and with the short DOF you need to be on top of that), which means some sort of rig to hold it, maybe an external battery, external sound recorder, shoulder mount… so you end up with something that looks like it was built out of meccano. Fine for a studio or film set, atrocious for documentary work.
Mind you, I'm not liking the shape of that JVC camera in this regard. Looks too much like an SLR for my money. Where's the eyepiece?


text:
How can I create floating objects in After Effects?
I want to create a floating object (or objects) when time freezes.
Zach king created a video that throws popcorn in the air, then when the time freezes, the camera moves around.
Search : 'Zachking Popcorn' then see the floating popcorns for more info that what I exactly saying
How can I achieve this effect?

This is accomplished during shooting and is not an editorial effect. An set of DSLRs (digital still cameras) arranged in an arc are triggered to all take a shot at the same instant. The individual frames are strung into a sequence and treated as a motion clip. There are lots of examples of this on the web.
You can do similar things using CGI, but it's not a simple task and it isn't cheap.
Edit:
You asked for a tutorial but I don't know of any, so...
Here's a demonstration
Here's a cheaper alternative

Thank you for your good answer can you give a tutorial link ...


text:
What are the basics of high-speed videography?
I am particularly interested in high-speed videography. Any tips on how to get started, what equipment should I have, understanding about lenses, apertures, what are the laws I need to obey, what should I learn and from where? Clearly I am lacking in basic knowledge.
Perhaps a better question is what books or courses should I take to bring me to the point where I will be able to just Google the questions I have. The problem is, that I have zero background knowledge about video taking and photography and therefore my vocabulary regarding things I need to know about videography is nonexistent.

I just changed the title to focus on high-speed videography, which is a much, much narrower topic than videography (and cinematography) in general.
The basics are these: due to persistence of vision, the eye can be fooled into interpreting a series of still images as if they are continuous motion at around 16 frames per second.  There are a variety of reasons why people shoot at 24, 25, 30, 60, and even 120 frames per second for "normal" speed.
The premise of high-speed video recording is to capture and display temporal details that are normally too fast for human vision to really make sense of: a balloon bursting, the details of a bee's or a hummingbird's wing-beats, the flight of bullet fired from a gun.
A hummingbird flaps its wings approximately 50 times per second.  If sampled at 25 fps with a 180-degree shutter (meaning 1/2 of 1/25th of a second, or 1/50th of a second), then every other wing beat would be fully blurred across the entire range of motion of the flapping.  If the shutter were restricted to 18 degrees (meaning 1/500th of a second), the blur of the wing would be reduced to 1/10th the total range of the flap.  Again, every other wingbeat would be imaged, and what would show up is approximately the same 1/10th of the wing beat every time.  As a movie, it would look like the hummingbird was hovering due to a blurry wing that didn't really move.  That's still not a very good visual explanation of what's happening.
If you want the wings to beat 2x per second, or 120 bpm, you need to slow them down by 25x (from 50 bps down to 2 bps), which, at 25fps would mean shooting 625 fps.  A standard shutter angle for the perception of smooth images at any frame rate is 180 degrees, which means 1/1250th of a second.
There are various cameras that shoot at such high frame rates.  Many have severe restrictions on the resolution they can provide and/or the image compression schemes they use.  At the high end, a Phantom sets the pace, and records up to 1,000,000 frames per second. In the consumer world, SONY sells a camera that will do up to 960 fps (1000 fps in PAL).
Once you have sorted out what frame rate is necessary to expand the event you care about (such as how long it takes for a bullet to pass through an apple) to the time frame you want to observe (a 30 second excerpt of a pop song), you can calculate the slowdown you need, and from that also the shutter speed you will need.
Once you know your shutter speed and the camera's sensitivity, you can calculate how much light you need.  If you are shooting at 1,000,000 frames per second with a 360 degree (1/1,000,000th of a second) shutter, you need a lot of light.  Like 1,000x (or 10 f-stops) more light than when shooting at 1,000th of a second.  Getting a bright enough light that doesn't melt your subject (or turn it to smoke), and that does not appear to flicker or wander when capturing high speed requires research.  The Sun has none of these problems, but it doesn't work at night or inside a studio.
Finally, you have to worry about depth of field.  When shooting things at normal speed, a focus-puller can adjust focus as the subject moves from one depth in the frame to another.  Since nobody can pull focus at super-sonic speed, one must determine how much depth of field is necessary for the crucial moments of the shot, and then compose so that those crucial moments are all in focus from beginning to end.
Now, it may be that you are just interested in 48fps or 96fps, slowed down to 24fps.  That's a lot less extreme than 500 fps or 1,000,000 fps.  But the caveats apply: you need more lighting than when shooting at slower frame rates and shutter speeds, you need lighting that won't flicker, and you need to appreciate that it gets exponentially more difficult to track focus as your multiple of the normal frame rate increases.

Way to render my answer obsolete °ω°

Ok this question should probably be put on hold as it is way too broad for a single answer. If I understand you correctly, you're asking about both the technical and artistic aspects of both photography and videography, topics which people study for years in universities. So if you really want to learn all that stuff, sign up for a course at an university. That said, here are some pointers.
First a disclaimer, you're not gonna 'master' any of this by watching Youtube tutorials, reading blogs or even books. The only way you're gonna become better is by practicing a lot. I know people who know all there is to know about their fancy camera, but couldn't compose a decent shot if their life depended on it. But what sucks even more are those who can't compose a shot and can't handle their fancy camera, so ...
Technical skills
Since you asked about photography, I recommend the reddit photoclass archive. It's a great ressource to start learning the basics of DSLR cameras (most of it will also apply to medium- to high-end video cameras). There's also a list of additional ressources at the end of this post.
Here's an excerpt (Disclaimer: I don't know all of those myself):

Ansel Adams - "The Camera", "The Negative" and "The Print" - The reference. A bit outdated and dealing mostly with print, but still very useful reading for any serious photographer
Scott Kelby - "Digital Photography Vol. 1-3" - very easy to read and quite complete, a good introduction text
1x.com - for inspiration.
/r/photography - reddit's own photography experts.

The photography subreddit has a weekly beginner's question thread, might be useful for you. Also check out /r/askphotography/.
Again, those are only the very basics.
Artistic aspects
I've already written a lengthy post on that, check it out here. This also contains some links concerning the technical aspects of videography (in this field, technical and artistic goes very much hand in hand). Also check out the second answer for some additional links.
Again, practice. It might also help to look at other people's work. For photography, you can check out the following sites:

Instagram (Yes, seriously, there are some real photographers there.)
Gurushots (Photo competitions. Look at all submissions, not only the winners.)
/r/photocritique (Read the comments to learn a lot about how photographers think about photos and what makes a good photo)
500px (Photo sharing site aimed at professional photographers. I'm not using it, but photographers often recommend it.)

For movies, well, just watch good movies. The post above contains a link to a list of movies to watch.

Thanks for comprehensive answer. Will take a look at those links and some university courses. I am actually interested in the technical part of videography, but I imagined that the principles also apply for the photography.