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Is it possible to encode video using the same codec and container but different parameters for different scenes?
I know that it's very hard to find out best encoding parameters for a whole video file. Wouldn't it be easier to just go through the video and find out the best parameters for different parts (or scenes).
Does it make sense or probably there is software that already allows that?

No need to use two-passes. See the FFmpeg and x264 Encoding Guide; specifically the section on using a constant rate factor (CRF) for examples and more information. All you need to do is pick your crf value which controls your quality, and your encoding preset which controls your encoding speed/compression efficiency.

What you are describing is effectively what 2-pass VBR does for you.  It makes a first path that calculates the level of change for each particular time in the video and then uses this information to make the best possible use of the available storage space.
It is, however, entirely possible to do the process manually by doing multiple encodings with different parameters.  In fact, many formats even support having multiple encodings with different parameters within the same file.  This is how you are able to change the resolution on the fly when streaming videos on YouTube and how Netflix automatically adjusts the quality on the fly as you watch to try to avoid buffering issues.
Any professional level encoder should allow you to specify multiple formats for the file in queue and many will even let you specify multiple outputs for one instance of the file.  (For example, the render queue in After Effects allows for this.)
In general, I would recommend running a 2-pass VBR and then adjusting the keyframe and total bandwidth accordingly.  If you are using a format that works based on the motion in the scene, then generally keyframes need to be more often if the video gets lots of motion artifacts (where it diverges from where it actually should be too much in high motion).  If the video is simply too low quality and "pixelly", then the bandwidth generally needs to be raised or the keyframes lowered.

Do you know of software that allows scene/regional parameters for MPEG-2 or AVC encoding? I have read that Hollywood DVDs are mastered by a 'compressionist' who uses tools which allow such fine-grained tweaking.

@Mulvya - I have used such a package once when I was using a hardware encoder and Sonic DVD Producer for MPEG-2.  The software was however very complex, unstable and unwieldy and was, in short, the largest waste of $5 grand I have ever spent.  There may be cheaper options that allow that level of control now, but unless you are thoroughly knowledgeable about the algorithms and looking through for particular artifact problems, you will probably have better luck with 2 pass VBR.

Yes, some encoders allow changing parameters (like target quality or bitrate, or psychovisual tuning options (x264's aq and psy-rd options).
x264, the stand-alone command-line frontend for the library, has a --zones parameter to give more bitrate to some parts of the video.  So you could for example reduce the quality for the credits.
The actual x264 library supports changing many parameters on the fly, but I don't know if ffmpeg has a way to specify option changes for ranges of the video.  You could get nearly the same result by segmenting your video, and encoding the different segments with different options, then combining them.
Generally nobody actually does this, because it's a HUGE amount of work, and it's really hard to know which settings are going to help just by looking at a scene.  Usually if you want more quality, you just spend more CPU time on the encode, and that improves everything across the board. (--preset veryslow).  I could imagine getting some benefit out of tweaking AQ settings, though, maybe to address banding in one scene, but changing it for other scenes where that wasn't an issue.
Still, bits are essentially never precious enough for it to be worth paying a human to save them, instead of just electricity and CPU time.  This is more of a hobby thing / thought experiment to see what different codec settings do, IMO.  Maybe when we need to stream video to / from a mars colony, bits will be that precious...

Thanks! My use case is a screen recording of a click dummy. Quite often while I explain on the audio track, the image is still for 1-5 secs, great reduction of frames/filesize with -vf mpdecimate -vsync vfr. Some more dynamic here/then: mouse cursor, UI widgets open/close. All looks great with -crf 28. Just in the intro I shortly slide in/out a portrait and say hello. This lookus ugly at CRF 28. So I looked for a way to compress only that intro scene with a higher quality. No I know: Either separate intro + main video concatenated or find a way for --zones within ffmpeg.

@porg: After I wrote this answer, I tried actually using --zones.  It didn't work well; the bitrate spiked right after a low-bitrate zone IIRC, wasting a lot of the savings, so I think rate-control doesn't interact well with it, at least the way I was doing it where I tried to set a bitrate limit for a short part of a video.  So concatenation with compatible settings is probably good.

Generally you deemed "human encoding optimization not worth it". I overally agree. But an optimization for at most 1 or2 scenes per video (intro or outro/credits) to receive a different quality parameter to me seems a legit optimization effort.
So my 30min video "looks really well enough" at CRF 28. But that first 10secs of the intro really need the CRF 18 to look pleasant and not turn off my viewers right at the beginning, but still I want my viewers (also not blessed with broadband!) to benefit from a video otherwise really well compressed.

Also regarding buffering the UX should be ok if you have the high quality part at the beginning, then followed by average compression.
It will buffer what's necessary for the high quality start. Sort of "overcompensates", may take longer than necessary for the later median bitrate. When reaching the average quality part it will most likely continue without any hickups.
Whereas starting low, the buffer likely was calculated over-optimistically small, and when reaching the main higher quality part will get overwhelmed and may need to rebuffer then.

Regarding concatenation: Could you add to your answer how you did it? Given a concrete input.mov lasting 30m 00s into the target file being a h264 mp4, the audio unprocessed, the video processed. Which command line(s) to create that 10secs intro (from 00:00 to 00:10) at CRF 18. The remainder (00:10-30:00 (ideally not hardcoded but a dynamic "end" syntax) at CRF 28 in which I would like to use -vf bpdecimate -vsync vfr (in that pass or on the final outcome file? And then concat them. Ideally the target h.264 MP4 with a VFR video track + the original audio track passed through from the .MOV.

@porg: It was a few years ago that I tried it, but in my notes I found I'd used ffmpeg -i ...  -x265-params "zones=1500,7000,b=0.7/9100,13600,b=1.6" out.mkv in addition to the usual options.   At that time, it seems the option only overrode bitrate, not modify the CRF like would be much better.

Additionally, my notes said zones only supports q=%d or b=%f sscanf formats.  Others will error and leave qp=0 for that region!! 2pass with zones isn't great.  By the end of a b>1 zone, ratecontrol is back down to below the normal rate. And there's a big spike wasting bits when b increases (from 0.7 back to 1, or from 1 to 1.6) So it seems the bitrate multiplier was being overridden by 2-pass ratecontrol or something.  IDK if I tried it with CRF.  This was I think with libx265, but probably also applies to FFmpeg's -c:v libx264.

I suspect libx265 and libx264 can handle this well, but unless FFmpeg's driver for it has improved (like making CRF an expression involving t or a frame number?), you can't take advantage of the full power.  Or maybe there's some rate-control setting that can make it not suck.

Thanks for sharing your -x265-params "zones=…" out.mkv experiences. Could you be so kind and add concrete instructions for the concat scenario into your existing answer or a new answer? (For this the comments are likely too limited in syntax and length)

@porg: That's not something I've tried recently or maybe ever.  But google finds results like https://ffmpeg.org/faq.html#How-can-I-join-video-files_003f / https://trac.ffmpeg.org/wiki/Concatenate and SO's How to concatenate two MP4 files using FFmpeg?.  I'm not sure how to make sure that two segments are compatible.

Was successful eventually! See my full writeup with detailed instructions at: https://video.stackexchange.com/questions/36646/how-to-encode-scenes-with-different-crf-into-a-h-264-mp4-video-with-a-variable-framerate

Variable bit rate and adaptive codecs are common in the audio world, and work very well. The most obvious example is VBR MP3s. Encoding video is a fundamentally similar problem, so I would be surprised if adaptive codecs don't exist there too.

That would be good to know they are exist. However, I wonder if there are semi-automatic applications for professionals who wish to adjust quality of the output to the best.


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Avid import wmv file, just audio no video
I've got a few wmv files, when I import them into avid it seems that the video channel is not being imported. There's just audio.
Am I missing something in the import stage? Or in the handling of the sequence?

This is likely due to the video codec that you are using. First, an explanation of terms is required: WMV is a container format. These can contain one or more video or audio streams - with video this is normally one video & one audio stream; and these are encoded with various codecs (coder/decoder). I'm not very familiar with the WMV format - for editing I use proper intermediate codecs (Apple ProRes in my case; more in this in a moment), and I prefer MP4 as a delivery codec.
There are broadly three 'types' of video codec: capture codecs, that a video camera will use to record the video; intermediate codecs, that are more suited to post-production editing (they trade larger files for requiring less processing power & the ability to easily cut on any frame); and delivery codecs, that are more compressed (so they can be streamed over the internet, or so you can fit a decent length on a physical medium like a DVD, or for broadcast). There is some overlap between these (many DSLRs use h.264, which is designed to be a delivery codec, to capture video).
This stuff also applies to audio - though there you should use uncompressed PCM audio. However, since audio decompression requires far less processing power, some NLEs will allow you to get away with compressed audio.
I have never heard of the WMV format containing any intermediate formats, as far as I know it's a purely delivery format. That's probably the cause of your problem. I've never used Avid, so I don't know  exactly which intermediate codec you should use. Since it's designed by Avid, the DNxHD codec is a good bet for video (you should convert your audio to PCM audio while you're at it).
Your video editing software probably came with a media conversion tool. You could also use the command-line-based ffmpeg (wiki) to encode to DNxHD. For the following command, you would need to adjust the bit rate - here it is 36 megabits/second - see here for some information on what you should use. Note that the ffmpeg syntax used in that guide is somehat out of date; you should use -c:v instead of -vcodec and -b:v instead of -b. This following command will also convert the audio to uncompressed 16-bit PCM, for maximum compatibility:
ffmpeg -i input.wmv -c:v dnxhd -b:v 36M -c:a pcm_s16le output.mov

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How to Create a High Quality, Small File Size .mp4 from .mov
Recently, I helped a friend produce a short zombie comedy.
He has now asked me to add it to the film's website as a digital download. I'm not having issues with the digital download. My issue is how to create a high-quality file (preferably .mp4) with a small file size.
I currently have a copy of the film in .mov (created in Final Cut). The .mov file is 3.2GB. The film is 29:52 in running time.
From other experiences, I know a 30 minute high quality .mp4 can be 185mb - 250mb. The problem, I can't seem to find any legit tutorials to create a high quality .mp4 file with this type of compression. The closest I have been able to get is 385mb (I believe that was with VLC, but I last converted the file a few months ago, so I'm not positive)
I am on a Windows 7 machine.
I have a wide variety of software at my disposal and would likely be able to purchase software, if needed.

I know this question is old, but thanks for the reply. I have a Windows machine. Also, it's mostly about the setting to use within a program to get a high quality video with a small file size

A really good program you can use is Handbrake. It's a very popular program among so called "pirates" because it's really good at getting a high quality even at tiny file sizes. And this is with good reason, because the main purpose of Handbrake is, indeed, to make the movies small with high quality. Personally I use it when I distribute my short movies on the internet, and it works like a charm. But enough talking, let's get to the settings!
Let's start with the beginning and go through the important settings.
On the right you have a presets panel. Choose Regular -> Normal or Regular -> High Profile.
Now, we start with the Picture settings.
The anamorphic setting is about the aspect ratio. So you've probably heard about the fact that in video, pixels aren't always square. So if you have widescreen footage, if you compress that to be much thinner, like an OLD SD TV, you could tell the video player to make each pixel wider than it actually is. So, you started have a 500 wide picture, you compressed it to 300, then you tell the player to make those 300 as wide as 500 again.
This is just a basic idea about what it is. You can just leave it at Loose with Modulus at 16. But if you want to learn more and squeeze the absolute maximum out of the file, read about it here.
Leave the cropping at Automatic, unless you want to crop.
Then we go over to the Video Filters. You can turn all of these off. They are made to correct bad footage, which you've probably already done in your video composing software..
Now, Video. These are the crucial settings. Use H.264. This is the format that gives you the highest quality compared to file size today. Quality. This is THE setting if you only want to set ONE. Set it to Avg Bitrate. This is the best one because if you have a lot of blacks in the picture (at night), you can optimize for that and get even smaller file sizes, but at other points in the movie you'll need more space, and so average is the best here. Notice that the size is in kbps which means kilo-bits-per-second. One kilo is 1000 (not 1024, that's kibibit.). One bit is 1/8 of a byte. So if you want a 10 second movie that's 1 MB, you want to enter 1 MB = (1024 * 1kB * 8bits) / 10s = 820 kbps.
Remember that the audio will also add up in addition to this. You just have to test some values until you get an acceptable quality that's small enough for you. If you enable 2-pass encoding, it will evaluate the movie twice, which means smaller file sizes compared to the quality, but it also means that you need more CPU (computer power) to watch it. So turn it on if you want to watch it on computers, leave it off if you want to watch it on mobile phones/tablets.
As for the audio, you just set it to what the output of your original file is. The sample rate of most movies is 48 kHz, but if you recorded in 44.1 kHz, set it to 44.1 kHz... As for the bit rate, 160 is acceptable, but if you have high quality mics and music, set it to 192. You can go higher than this, but as you want small file sizes, there is no need. You could also go lower depending on what size you want to end up with. But do a Preview and listen to the result until you are happy with it.
Then, the last thing you need to do is to tick off "Large file size". You need to have this on if the file is going to be larger than 4GB, but it will break compability with a lot of devices if you leave it on. Then you turn on "Web Optimized". It will make your movie better suited for progressive download so you can shuffle in the movie without downloading it all. But if you are planning for your users to download the whole movie as .mp4 and watch it on the computer, leave it off.
Now, set a file name and click Start. You can also Preview a small clip when you're testing the bitrate setting. Make sure that the extension is .mp4 and not .m4v. This isn't important, but you asked for .mp4, not .m4v. It's just the extension that is the difference anyway..
Hope you get your video squeezed down in size and yet keep the quality as high as possible! Good luck :)

2-pass does not require more CPU power to watch, it just varies the bitrate depending on the average rate you are targeting and the relative scene complexity.  Also, usually you'll want to always enable Web Optimized even if most users will download the whole video first; it adds slightly to the encoding time but does not increase the file size and can be a big help for viewers, and is expected for Internet mp4 video.

@FriendOfKim , can you say something about h265 (and also, x265) ??

@mark4o my question is also for you, as you seem to know the things..

The key is to dial in the bitrate - you'll need about 800kbps video with 100kbps audio to hit 200MB. That's low for for SD resolution, but it might be good enough for you. Also look for software that has an option for 2-pass encoding.
Try Expression Encoder, Adobe Media Encoder, or Sorenson Squeeze.

Can you provide more details about the source format, as in resolution, video & audio codec details etc?
My stock answer is to use ffmpeg with Avanti for the frontend. A starter tutorial is here

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Is converting DVD to a format to play on a tablet legal (in the UK)
Simple question, is it legal to take a DVD you already own and convert (rip) it into a format (such as .mp4) so it can be played on a tablet or laptop ?
In the past you had to use various nefarious means to convert CD to .mp3 but now, iTunes & windows media player come with the functionality built in (which in my mind legitimises the practice).
So, assuming DVD -> .mp4 is legal, is there any legitimate software to make the task easy ?

Your question is a bit wonky, in that your title asks if it's legal, and then in the question you assume that it is and ask for tools.  Just sayin

I was just jumping the gun, otherwise I'd have got a 'yes' answer to the first question, then had to ask the second question separately...

This question appears to be off-topic because it is asking for legal advice.

Fair enough - can you suggest an area where it's OK to ask this ?

You could also rephrase your title/question to make it not about the legality of the practice?

But, the question IS mainly about the legality of the practice. I know it can be done and (in the light of the CD example I gave) I was wondering if it was now accepted that the practice is legal or at least, tolerated ?

Ah, I see.  Well, as Dr. Mayhem suggests then, you probably just need to ask for legal advice... from a lawyer (barrister?) :)

Please note this is UK legislation
After some research I found the following in a Government consultation paper:
From MODERNISING COPYRIGHT: A modern, robust and flexible framework
http://www.ipo.gov.uk/response-2011-copyright-final.pdf

There is a clear mismatch between what is permitted by law and the type of private copying that most people think is reasonable. The current law imposes unreasonable restrictions on consumers and may restrict the range of products and services available to them by limiting opportunities for technological innovation. In response to this, the Government will introduce a narrow technology-neutral private copying exception which applies equally to all types of private storage and all types of copyright works. It will allow a lawful owner or buyer of a copy of a work to reproduce that copy for their personal use, but would not permit them to share copies with other people.
This exception would therefore permit someone to copy music from a CD they have bought onto their mobile phone, or to back up an eBook onto their computer hard disk or other private storage, whether held locally or remotely. It would be allowed regardless of contract terms to the contrary. However, it would not allow someone to give copies they have made to friends, family or the wider public, nor would it permit copying of rented, borrowed or streamed content.

And

A permitted act that only allows copies to be made by and for a person who lawfully owns[41] an original copy will remove restrictions on the use of copyright materials that most people consider to be unreasonable and unnecessary, while drawing a clear line between acceptable personal copying and unacceptable sharing of copies. Consumers appear to recognise that sharing copies can harm creators and the Government does not want to dilute this message. Rights holders will therefore still be able to offer family copying and other licences that add value over pure private copying by individuals. They will also remain able to use technological protection measures (TPM) to restrict copying of works as and when they wish to do so.

[41] - This would include downloaded digital content that has been purchased to keep (or licensed in an analogous way) such as eBooks or film downloads, but not rented or streamed materials.
So, as I read it, you can make copies of CDs, eBooks, DVDs/BluRay if and only of they have no TPM on them.

CDs don't have TPM so copying (format shifting) is legal.
eBooks have TPM so copying is illegal.
DVDs/BluRay have TPM so copying is illegal.

But, it's the act of 'breaking' the TPM which is illegal. not the actual copying as long as the copy is for another device (format shifting) or for an archive/backup copy (whether on a local hard disc or a remote storage facility (the cloud)).
So, clear as mud really !

There are different laws regarding ripping DVDs depending on where you live.  Two points to consider though:

Cases have been made (in the US) that you are within your rights as a license holder to make an archival copy for backup.
Having said that, breaking DRM is almost always instantly illegal.

Little bit of a catch-22 there for sure.  Having said all that...
Assuming DVD->.mp4 is legal, a combination of MakeMKV and Handbrake will get you where you want to go.
Both Free.
MakeMKV will do the ripping, and Handbrake will do the transcoding for various formats/devices.  I do believe there are even iPhone/iPad presets.


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How does the source encoding reflect the settings used to export?
I filmed Video during Christmas with my Canon 5D mark II.  My input source is H.264/MPEG-4 (Base Profile @ L5) from my camera.  I have edited the movie and just want to save it to the hard drive for long term storage.
When choosing the profile for encoding for storage, should I use the same H.264/MPEG-4 (Base Profile @ L5) profile to keep the quality, or lower to save space.
How does the source encoding reflect the settings used to export?

It is unclear what you are asking.  What are you trying to accomplish?  Are you trying to shrink the file size for long term storage?  Did you edit it and are trying to make a final output?  Are you asking if there is a reason to use a different profile or higher level even though the source was lower?

I comes from a photo background. If have small size compress gif or very hard compressed jpg (the source) then I open it in photoshop make some changes and I know that it limits
what I can do with the enlarging (interpolate to bigger size), change colorspace to larger one (interpolate to bigger colorspace). To some extent this must be true for video also
change the size from smaller format as 576i to 1080p will give some type of interpolate to bigger size. But how does this reflect in the input media is H.264/MPEG-4 with a give
bitrate selecting output.

The quality you use to export the video for long term storage largely depends on what you will be doing with the footage at a later date.
If you will be mostly using it for personal viewing, a lower quality may be sufficient.  If you will be editing the footage and using it in another video, you will want to encode it a higher quality.
The only scenario I can think of for encoding to a higher quality than the original video is if you were including the clip for use with other videos of higher quality in a project that will ultimately be encoded with the higher quality.
To sum up:
You should encode the video in the highest quality you can without exceeding the original and without going over your space requirements.  Since large hard drives are relatively inexpensive these days, this will likely mean that you encode to the original quality of the video.

For most types of compression, every generation of compression will lose quality.  Any quality you lose can never be recovered in future generations, however encoding at higher quality will reduce the amount of loss.  If you absolutely need to preserve the most detail you can (even out of a low quality video), encoding at a low level of compression (bigger file than you started with) will help avoid losing as much detail.
This is why it is very important to keep video as high quality as possible until you are doing a final output that will be consumed by a viewer.  This is also why it is important to try to minimize the number of times that video needs to be re-encoded when working with it.
There are some types of compression that are very specially designed to not have any additional loss when re-encoded provided that nothing has changed about the video, but they are not used very often and are pretty specialized.  (Generally for images rather than video as well.)


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Windows 7 - multiple programs with conflicting sound output
I have a Windows 7 (32 bit, service pack 1) on my laptop and, whenever I launch my sound editor (Ableton Live 8), all other programs (Firefox, Chrome, Skype etc) stop outputting any sounds. Even when I try switching the order of steps (opening Ableton first and then launching a browser, for example) it doesn't work.
In Control Panel > Sound, I have the laptop speakers as the default output and, as soon as I close Ableton, all other programs are able to make noise again.

What audio devices do you use, and with what driver? You bet it's quite intentional that programs other than your DAW aren't granted access, so there won't be any conflicts over e.g. sample rate. — There are specialised frameworks for bringing the audio from different programs together, perhaps most notably ReWire. You won't be able to use this with programs that aren't professional audio applications, though.

My speakers are the default sound outup - Realtek high definition audio.
To clarify my point, all I want is to be able to have both Ableton and Chrome open and watch tutorials on Youtube and practice the lessons on Ableton (switching from time to time without losing the ability to hear sounds on any of them).

Again, what kind of driver does Ableton use? There are a couple of possibilities, like DirectSound or ASIO4All, which work quite differently in such regards.

Driver type ASIO
Audio device ASIO4ALL v2

Driver type ASIO Audio device ASIO4ALL v2

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

Yes, it works! I changed Driver Type to MME/Direct X and mapped Audio Output Device to my speakers.
Thanks leftaroundabout


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Web Videos via Watch Folders Transcode
What I am looking to do is fairly simple, however seems complicated in theory.
What I need to do is be able to have a folder where all video files placed in the folder, are transcoded to all applicable web video formats (H264, OGG, WEBM, FLV) Then placed into another folder...
I can set this up to a certain extent using Adobe Media Encoder however I need to use source Frame Rate, and Frame Size, as these clips will vary in range from PAL to NTSC to HD...
I am open to using FFmpeg but cant seem to get it work in the Mac Environment correctly doing what I want.  Any software as long as it is mac based would work, but must be able to create all 4 of the above mentioned formats.

If you want to use ffmpeg a bash script would help you a lot to automate the process. Here's an example of a similar bash script using ffmpeg to batch convert video's in iTunes friendly format. If you're a bit code-savvy you could rewrite it to webm, ogg, h264 and flv. Here's a little bit of text on the latter conversion using ffmpeg.

This is what I am working on now, I am in the process of getting ffmpeg to work correctly and then go from there

...FFMPEG worked, it was a steep learning curve in terms of getting it to work in my environment and, took me a few days to learn how to code in Bash Scripts Shell based, but in the end very happy with the results and it's integration with both Macs and my Unix server.

You could create a droplet in compressor.  Unfortunately, your options for destination folder are limited by this method to either source, desktop, cluster storage, or your movie folder.  Since you don't want your watch folder endlessly copying movies, "source" is a bad choice, and you probably don't have cluster storage, so as long as you're ok with your destination being movies or desktop, you could use automator to make a watch folder that performs the Finder->open action using your droplet as the application.
I'm sure someone will chime in here with a better solution soon.
edit  You can change a droplet's destination to any folder you want.  Just open it in the finder after you've created it and change the destination.

I will look into this, I absolutely hate compressor however if it does the job correctly and efficiently I may need to use it...however I do not believe (may be possible just havnt looked into it) that compressor does not support theora (ogg or webm) or flv

I am unable to successfully convert to OGG using several different codec variations, so compressor will not work, I also have yet to find a solution for converting to webM and FLV


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Record on laptop from streaming camera via RCA cable
I have a video camera that only 'streams' the images and has no way of recording them. Currently I'm plugging the RCA into a TV and then recording from that, but I need to be mobile and so need to record onto a laptop.
Will something like this (http://www.amazon.co.uk/tag/av%20to%20usb%20converter/products) work?
Any other ideas?

That seems like an ideal solution.  Converts analog to digital over USB.

EasyCap 2.0 DC60+ is what I use, and they work great. Make sure you get the DC60+ version, you don't want the older versions.
There are a lot of knock-offs floating around--I'd stick with the official one.
There's also a good compilation of capture cards on this thread on the Wirecast Forum.


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Premiere Elements "Debug Event": Keyframe error
I have encountered a probable bug in Premiere Elements 10, and fortunately I've also found a workaround which is why I wrote my own answer below. But since I haven't been able to solve the underlying problem, I'd be very glad for additional ideas on how to fix the problem. Now here's the question:
How can I salvage a Premiere Elements project file (.prel) that can't be opened because of the following error:
Premiere Elements Debug Event
Premiere Elements has encountered an error:
[d:\pre\mediacore\mediafoundation\api\inc\Keyframe/Keyframe.h-142]

This error appears to occur sporadically on non-US versions of Windows. There is no previous indication during the editing session that anything may be wrong (saves and auto-saves work fine), but on the next restart, the project can't be opened. Other projects appear to be unaffected.
According to other users who have seen the same error (Google for "premiere debug keyframe" and you'll find some), there might be some connection with the introduction of DVD menus into the project, but I'm just speculating here.

Analyzing the files that fail to open in contrast to those that do open just fine shows one glaring problem: All lines in the project.prel file (which is just an XML file) where there should be decimal numbers in <StartKeyframe> or <Keyframe> tags are damaged. Example:
Correct version:
<StartKeyframe>-91445760000000000,100.,0,0,0.,0.,0.,0.</StartKeyframe>

Defective version:
<StartKeyframe>-91445760000000000,100,,0,0,0,,0,,0,,0,</StartKeyframe>

So apparently Premiere Elements at some point got confused by European locale settings and decided to save decimal values using the European decimal separator (comma) instead of the US decimal separator (dot). Of course it's then impossible to open that file correctly ever again.
Fortunately, at least in my tests, there are only three kinds of decimal values in .prel files:

integer floats: 1., 0., 100.
the single value 0.5
double precision floats: 0.16666666666666666, 1.000000000000000000000000

This makes it possible to fix the file using a regular expression replace. If your editor supports Perl-style regular expressions, you can search for
,(?:(?=[,<]|\d{10})|(?<=\b0,)(?=5\b))

and replace all with
.

This will replace all erroneous decimal commas (i. e. those that are either followed by another comma, an opening angle bracket, a decimal fraction of at least 10 digits, or the number 5 (but only if that comma is preceded by the single digit 0)) with a decimal point.
After that (be sure to back up your .prel file before you do this!), the file can be opened again correctly. However, as soon as it's saved again, the error is re-introduced.
Therefore, if someone has an idea how to actually fix this problem instead of patching it up with an admittedly wonky regex, I'd be most grateful.

One hint as to what the underlying problem might be: Around the time the error was introduced, Premiere spontaneously opened a little dialog window that said (in German) something like
"Premiere Elements is updating a component. Information is being transmitted..."

While this window was displayed, two Win7 UAC boxes popped up, asking me to allow the Windows Regserver to update my system. The actual commands it needed approval for were
C:\Windows\System32\regsvr32.exe /s
   "C:\Program Files\Adobe\Adobe Premiere Elements 10\mc_dec_mpa_ds.ax"
   "C:\Program Files\Adobe\Adobe Premiere Elements 10\mc_dec_mp2v_ds.ax"

and
C:\Windows\SysWOW64\regsvr32.exe /s
   "C:\Program Files\Adobe\Adobe Premiere Elements 10\32\mc_dec_mpa_ds.ax"
   "C:\Program Files\Adobe\Adobe Premiere Elements 10\32\mc_dec_mp2v_ds.ax"

(line breaks inserted for clarity).
The first time around, I allowed these actions (which in retrospect may have been a bad idea...); now this happens every time I'm trying to open a "patched" project file. It doesn't matter whether I now allow or refuse it - either way, Premiere writes a corrupted project file when I save my project.

That regular expression is brilliant - saved me a lot of time, when I actually had no hope for recovering this project! Altough I've no idea what caused project to be saved in that way, that happened only once for me.

I would give you a huge price for what you did. This is just remarkable.

1. Recover broken files
I found this online application very helpful.
http://www.vibait.com/software/premiere-fix-keyframe.h-142/
Simply upload your Premiere Elements file and you get a fixed version in return. Save it to your disk and open it.
With one file I got error messages like this one a few times:
Parse error on line 1431. Expected 8 values, got 9
Old value: 116050644638,87,5,0,0,25,791250419642971,0,16666666666666666,1,2731148504048133,0,16666666666666666
New value: 116050644638,87,5,0,0,25.791250419642971,0.16666666666666666,1.2731148504048133,0.16666666666666666
Using an editor I searched for some of the mentioned numbers in the XML file and simply cleared the XML element (deleted all numbers).
2. Fix it (i.e. a workaound)
A few web pages mentioned that the number formats of the regional settings (especially , and .) may be involved in this problem. After setting "." as the decimal separator I did not have this problem again.


text:
Rearange order of audio tracks in Premiere
Let's say I have the following arrangement of my audio tracks:
x ----------
y ----------
z ----------

and I want to have them arranged this way instead:
x ----------
z ----------
y ----------

What's the easiest way for me to make this happen?
(As an example, in Reason, I would grab the audio track I wanted to move and then simply place it where I want it. This does not seem to work in Premiere.)

As you have discovered Premiere Pro doesn't let you move the tracks as easily as that (I wish it did!)
Fastest method would be to create 1 or 2 empty tracks (right click on a track label and select Add Tracks) then use the Track Select Tool  ('A' on the Keyboard) to quickly select an entire track:

(Click and hold lets you select and move the track at the same time).
You could also select an entire track, Cut it, move your other tracks around, then Paste it back into a new track (you can choose which track it gets pasted into by clicking on the Track labels, Premiere will paste into the top most highlighted track)

Oh dear! This seems cumbersome but thanks for the "sollution" ;)

@speldosa fingers crossed it has been improved in CS6!

If you need to keep the track settings and move the entire track you can use the Premium version of PlumePack (it's a plugin).
You will be able to change the order of all your tracks without having to create empty tracks and keeping all audio effects and associated key-framing.

On Mac, select footage that you want to move. Hold down ALT, use Up or Down arrows to move the entire track.

This doesn't keep move any of the channel settings, only the footage from track to track

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


text:
What are the minimum camera characteristics required to record a movie for cinema?
What are the minimum characteristics required for a camera to project a movie taken by it on a standard cinema screen? I mean anything that affects the quality of the picture seen on the screen. What sort of camera should be used to shoot the video?
Edit: To be more clear:
If I record a movie with my cellphone's camera, there is no doubt its quality is not good enough to be projected on a cinema. So I should use a more professional camera! My question is about the minimum characteristics this professional camera should have. Like the number of frames it can record per second, the size of picture it records, the number of pixels per picture it records, etc.

I am not clear on what you are asking. Are you interested in recording a projection of a video with another camera? If so why not just make a copy of the the material being projected?

@filzilla Updated the question.

Thanks for the update. A word of caution, unless you have permission to copy the work, you might be violating copyright laws when taking video of other peoples work.

@filzilla I think my question is still ambiguous and you didn't get it. I am not to copy anything. I want to record something from scratch.

If I understood your question correctly, you want to project a movie on a cinema.
This would depend on the projector used by the movie theatre. Most movie theatres use film projectors, for which there really isn't a "quality", but rather you'd have to worry about getting a decent camera with decent film at the correct size. In this case I don't know much about transferring from digital to film.
If the movie theatre uses a digital projector though, they usually come in 2K (2048×1080) at 24 or 48 frames per second, or 4K (4096×2160) at 24 fps if they follow the DCI specification. Basically, get the best camera you can get. 2K should be enough to have a pretty good quality experience at the distance a cinema projection is seen from.
Also remember though, the quality does not only depend on the amount of pixels on screen, but also on the compression used, and the sensor of the camera itself, so be prepared for large files or a lot of processing to compress your movie to a high quality profile.
If you can't get a 2K camera, 1080p should do: it's not too far from 2k: it the same vertical size, just a little less wide.
Links that might help:
http://www.mkpe.com/digital_cinema/faqs/tech_faqs.php#2K+4K
http://en.wikipedia.org/wiki/Digital_cinema
Hope it helps!

Thanks for the answer. I think it's answering the question well. I updated my question to be more clear though.

No problem! Your question is much clear now, btw. Thanks for that.

Depends on the look you're after. Searching For Sugar Man won the 2013 BAFTA Award for Best Documentary, and it was partially shot on an iPhone, the Blair witch Project used a lot of DV video (standard definition heavy compression) and parts of Natural Born Killers were shot on super-8.
If the content is worth it, or if the style of story telling motivates it people will forgive crappy picture quality (but it certainly needs to be good content).


text:
Final Cut: how to playback sequence while capturing video?
When you open "log and capture" window, you can't use other FCP windows anymore. How can I listen to already captured video?
I want to record my solo on top of rhythm which was captured before. I don't want to synchronize videos.  If I just could listen to the first video then everything would be fine.
I would like to do the same thing as in Logic studio, listen to 3 tracks playing and record a 4th one in real time.
How can I achieve this in final cut?

As far as I recall, you can't.  I would tape your solo on your camera while playing your time line and then importing that video back into FCP once it's done.

i don't want to sync audio =( logic studio for video would be great.. =/

Gotcha.  There's still no way to play the timeline while you're also in log-and-capture mode.  Basically, you have to use your camera, or an audio recorder, or something to capture your solo while you play your timeline for your rhythm background track.  Alternatively, you could export that, play it in Quicktime, while importing your live track with Log and Capture.

There is no way of accessing the rest of the FCP program while capturing, because your computer is allocating all of the 2.5GB of Ram to capturing to prevent a smooth capture and no dropped frames, since FCP is software based and not hardware based this is not possible.

what about having 16gb of ram... and use each cpu core for one capturing.. what's the problem to capture god damned webcam which uses not more like 20mb ram =/ what's wrong with this world


text:
What is the best way to clean the outside of audio/video cables
What is the best way to clean a cable that has gotten sticky. I know the best thing to do is to not get them sticky in the first place, but after time they get dirty/sticky all by themselves.
Lets assume that there are no breaks or cuts in the outside of the cable.
How would one best clean the outside of a cable of stickiness without damaging it more.

For delicate cables use dish washing soap and warm water with a clean rag. Do not tug or squeeze on the cable too hard and be extra gentle with anything that has twisted pair.
For not so delicate cables rubbing alcohol on a clean rag then applied should do the trick.

When I was a a corporate photo/video shooter I found the "desk and office cleaner" everyone had worked pretty well with paper towels. Now that I shoot in the third world, it is Lysol wipes when i get back to the studio. And it is terrifying how many it takes until they come out clean...

Goo Gone. Put it on a cloth, rub it on the sticky part of the cable. Let it sit for a minute, then rub off with a dry cloth. Repeat if necessary. For this or any of the other suggested procedures, it would be a good idea to test on a small non-critical part of the cable. I've never had a problem with it though.

can you elaberate? how do you apply the product

Put it on a cloth, rub it on the sticky part of the cable. Let it sit for a minute, then rub off with a dry cloth.  Repeat if necessary. For this or any of the other suggested procedures, it would be a good idea to test on a small part of a non-critical cable. I've never had a problem with it though.

The information from your comment is great, so I edited it into the answer itself.

goo-gone or butane lighter fluid will remove just about any gunk that has built up on your cables.

How is this different from @gauss256's answer?


text:
Is this projector capable of 1080i resolution?
From the specs of the projector:

DLP - 4000 lumen - 1024 x 768 - 480p - 720p - 1080i - 3D Ready - 6.4 pounds

They state the resolution is 1024x768, but then they add 1080i, from wikipedia:

The term 1080i assumes a widescreen aspect ratio of 16:9, implying a
  frame size of 1920×1080 pixels.

I'm a bit confused, what the resolution of this projector?

This question is not exactly on topic here on this site which is about Audio and Video Production, but since projectors are used in some performance scenarios, I've edited it to more closely fit here.

Yes
Most projectors support multiple resolutions.  The specifications list all resolutions supported by the projector.
The projector you are asking about will handle 1024x768, 480p, 720p, and 1080i.

OK, so you changed a little bit the question title. What I meant to ask was: is 1080i similar to 1080X1920 ?

Yes.  1080i refers to a standard resolution which is indeed 1080x1920.


text:
Reading material for technical aspects of timecode and alignment
I'm a sound engineer with extensive live and recorded experience. My electronics and acoustics are pretty strong IMO but I've never had any extensive interaction with video/film production and post timecode hardware. What's the best book/website/whitepaper to learn the nuts-and-bolts technical aspects of that stuff. Yamaha Sound Reinforcement Handbook gives some discussion of it, but it's short and out of date.

This doesn't cover the applications, but it gives a succint view into the nuts and bolts:
Phil Rees


text:
What's the best setup to transfer a digital or analog video over 100 ft?
I'm looking for the best (price/quality) solution to send a digital or analog video signal to an lcd tv or beamer over 80 - 100 ft. The input can be hdmi, vga or dvi (all must be possible). The output (80 - 100 ft. away) doesn't need HD quality.
tv has hdmi, vga, dvi, s-video, composite and coax.
So far i've got this setup in mind for relative low cost and acceptable quality:

80-100 ft. vga cable connected to tv (should cover inputs with vga or dvi)
vga switch 1-4 for switching between inputs
hdmi --> vga converter (to cover hdmi inputs)

Does anyone know a better setup ?
is vga the best option in this case to bridge the 80-100 ft. ?

A long VGA cable with a proper amp/spliter on the TX side, would be the simplest option. You can get good VGA cables up to 150' at modest prices from http://www.pccables.com .
Another good option is a CAT5 extender kit.
http://milestek.com/p-16209-vga-over-cat5ecat6-decora-wall-plate-set.aspx
Without a more detailed description of what you are trying to accomplish, it's hard to give a better answer than that.


text:
Lighting and Camera Angle of the Video
I'm trying myself with my first video. I made the figures with crayon/colored pencils. I have made the video at night with a desktop light.

The pencil drawings seem to glow, how can i prevent this? (Some post processing?)
How is the overall angel of the video? Should it be different?

This is the video:

I don't know how you are lighting this, but generally speaking photographing flat work requires an even light source. Usually two diffused lights above the surface at 45 degree angles should accomplish this task. Your lighting suggests that you are not using a diffused source(s) as witnessed by the hot spot in the middle of your screen hence the halo effect.
Another thing I noticed is that you are using a camera with auto-exposure mode engaged, subsequently every time you move your hand into the frame the light value shifts. If this is a Point & Shoot camera then there is not much you can do about this, but if it's a DSLR you need to lock down the exposure so it is not changing.
There are three manual controls that generally should be locked down before recording:
1) white balance,
2) focus,
3) exposure,
Many Point & Shoot cameras will allow you to set the white balance and focus, but not exposure.
If you are in this situation, then the only way to defeat the exposure changes would be to keep the light even through out the frame and sequence. For instance if you were wearing white gloves this would help keep the auto exposure from going lighter to capture your skin tone, or if you changed the back ground, and made it the same value as your hands--awkward yes, but even light would likely improve the entire scene.
Hope this helps

Wow, thank you very much! I didn't notice the exposure & etc things. This is a iPhone 4 and my only option :)
I will however, look what i can do. Thank you very much for this invaluable information!

You are very welcome. I've updated my answer to address the light source too.

I've downloaded a Camera app which will promise to keep focus and exposure steady. Let's say how it will work out :)
Thank you very much!

I would recommend to use at least one more light source! From the bottom of the image (good luck with that) to avoid the shade of your fingers and paper not flat to the surface. I would also use a warmer light color but that's a preference. I think the angle of the shot is fine. Use bigger pieces of paper so it's easier for you to work with (specially if you set the lighting fixture between you and the table). Please read the faq and find out why you got a down vote on this post (it wasn't me!). I don't think the edges are 'glowing' and I'm not a video guy but to solve that issue I would say that the amount of light is too much. Experiment by dimming the light source down and also with the lens shutter to deny some light to go through. Don't try to fix it afterwords! Good luck!


text:
Whats the difference between a sekonic light meter CINE version vs non-cine version
I'm looking into getting a used Sekonic light meter, my question is what are the advantages of getting 'CINE' version light meters Sekonic makes versus the regular versions. I'm going to be primarily working with video, but some still work to, and a lot of just testing to understand light.

If you're working with video, the cine model is the one you want. It lets you meter for motion, not just ISO, f stops, and shutter speeds. I've had the 758 for years... great meter.

Functionally, they're the same, but the Cine version allows you to select shutter speeds associated with motion picture frame rates - 24fps= 1/48th of a second, etc. I have the 708; I'm not sure if later models included provisions to adjust for various shutter angles...but the 708 does not. Or at least I've not found the feature.


text:
Final Cut Express: Flip a Video Clip
I'm sure this is trivial to do, but I've yet to find the right filter to Flip a video clip in FCE 4.
For example, see what happens at the 0:42 mark in this video  (Night Stand HD Pro)
I see box flips, or page flips, but how can I just flip a single layer around an axis?

I guess I'll answer my own question:
After a lot of searching last night, I found a YouTube video that demonstrated a method to animate the flip of video clips using a free plugin called MultiSpace from idustrial revolution.


text:
Is there any plugin for After Effects to create a 'kinetic typography' video?
I was looking at some kinetic typography videos - like the one below - and wondering whether there is a plugin for Adobe After Effects CS4 that can help with this. If not, then is there any other easy way to do this?

As of 2011, there aren't plugins that assemble a typography video for you, because this kind of work is really a sub-topic of motion graphics.  There are tools that can help you with camera animation and rigging in AE though.
My favorite is Video Copilot's Sure Target 2, which lets you think about the animation in terms of what you're looking at, instead of thinking about it in terms of camera perspective.  This makes the work feel similar to using something like Cinema4D's target camera.
Another noteworthy tutorial: http://ae.tutsplus.com/tutorials/motion-graphics/design-rhythmic-motion-typography-in-after-effects/

If not, then is there any other easy way to do this?

Not really.  Watch lots of videos, take notes on ideas you like, steal the best, and trust in what is difficult.

TypeMonkey will help you a lot when trying to create kinetic text animations. http://aescripts.com/typemonkey/

I don't know of any plugins, but I think it's probably possible to do just with After Effect itself. You may have already seen these, but here's a few articles on the subject.

Crooked Gremlins Kinetic Typography Article (there's also a video version)
BitRebels How To Make Kinetic Typography

you can use some scripts for camery keyframes, to get those type of ease or rushing camera effect, but it realy depends on how good your typography and idea is.


text:
Video codec for progressive enhancement
JPEG has a progressive enhancement mode where a full frame can be downloaded quickly at low quality, and then the quality enhanced as the file is continually loaded.

(Example image from http://promokit.eu/prestahop-blog/image-optimization-to-speed-up-your-site/)
I'm wondering if there is a widely compatible video codec that enables the same functionality.
If the codec were structured this way, it should be technically possible for a player to download a first-pass low-quality stream, begin playback immediately, and then download second-pass and third-pass quality enhancing streams if bandwidth is available to do so, combining them on playback for form progressively higher quality frames.
On the server, there would be segment files for each quality stream, similar to how HLS works today.

(Diagram from http://www.streamingmedia.com/Articles/Editorial/What-Is-.../What-is-HLS-(HTTP-Live-Streaming)-78221.aspx)
The key difference here is that with HLS, entire separate copies of the stream are stored in each stream's segments.  If I want a mid-quality stream, I read from the mid-quality files only.  If I want low-quality, I read from the low-quality files only.
What I'm proposing with progressive enhancement is that all players would always read the low-quality (first pass) stream files, and additionally the mid-quality (second pass) stream files if they wanted that mid-quality stream.  For high quality, they would read all three.  This should greatly reduce the storage overhead of the stream since the total disk usage to support low, mid, and high quality streams should be only slightly more than would be required for just the high quality stream.  This means that the upload bandwidth required on the source end is much less as well, since the stream is effectively uploaded once (plus some overhead) for all three qualities.  I imagine there would be overall a bit less CPU used in the encoding process since I'm only really making one stream (albeit in an unusual structure) rather than having to run three independent codecs simultaneously where much of the codec work is likely duplicated.
I have read about some proprietary codecs in random tech news articles, but never found anything concrete.  I also know that a few companies did audio codecs this way when audio streaming was booming in the mid/late 90s, but that's apples and oranges.
Questions

Is there a codec available that allows for progressive enhancement and segementation?
Is there a way to do this with a widely implemented codec like VP8 or VP9?

I suppose there is some hacky stuff I could do with MJPEG, since MJPEG is basically JPEG progressive enhancement under the hood, but I'd rather find a better codec if it exists.

Not quite the same thing but look up Scalable Video Coding. +

Wouldn't the problem be that the player would have to download each stream entirely before it could play? Though I guess you could download GOPs so that you download GOP1-low, GOP1-mid, GOP1-hi, then GOP2-low, GOP2-mid etc., only downloading what you can in the time available.

@stib Segmentation would be used, similar to how it is used in HLS, so that a few seconds of video would be encoded in each chunk.  You're right, each chunk of each quality stream would have to be downloaded to decode the desired quality for that chunk of time.  And, the client would automatically adjust between downloading 1, 2, or 3 chunks per segment as it has bandwidth to do so.  For HLS, the client decides which bitrate chunk to download, somewhat dynamically.  This would be the same, except that there's 3 requests for the high quality stream instead of one.

Quick! get a patent!

@stib I think that would require me to actually make such a codec.  Math is hard.  :-)  Although, I could probably actually implement it in MJPEG since MJPEG already does the heavy lifting...

@Brad did you ever find any angles on this other than MJPEG?

@awidgery Yes!  It turns out this has been a thing all along.  The keyword here is, "scalability".  See also:  https://en.wikipedia.org/wiki/Scalable_Video_Coding  Also, it's my understanding that SHVC is to be used for Next Gen/ATSC 3.0 TV where some layers will be broadcast, with enhancement layers via internet, to save on the broadcast bandwidth will still providing the higher quality video.

The answer here is "scalability".
https://en.wikipedia.org/wiki/Scalable_Video_Coding
SVC has been available as an extension to H.264 for years.


text:
Does anybody have suggestion to recreate this light trail effect?
Maybe you know this wonderful video:

Can someone help me to understand how the LED trail effect has been done?

There are two ways this can be accomplished.  The easiest is by using a slow shutter speed.  While most DSLRs may be limited to 30FPS and faster, it is possible through the use of third party firmwares to lower that arbitrarily further.  As you can see from the details on Vimeo, they were using GH2s with the EOSHD Vanilla hack.  This removes many of the limitations that would otherwise be on the system.  It is also possible with most Canon hardware using MagicLantern's FPS Override.
The other way to do it is to artificially do it in post production (though the result often isn't as good) by additive combining each frame with previous frames.  This results in a trail that can animate on and animate off and maintain the original frame rate, but it is likely to introduce gaps in the trails and either requires a pre-existing plugin or some creative layering in After Effects.  Layer each video one from apart and then apply blending until you get something that looks like you want.
It is hard to tell which technique is being used for sure without a downloadable copy that I can go through frame by frame, though my initial impression (particularly based on the hacked firmware and the general feel of the video) is that they are using the lower frame rate option in the hacked GH2.
Update: I managed to dig up my old Youtube downloader and got the video in to Premiere.  I can confirm that it is in fact a lowered frame rate (around 1/12 or 1/15 for most of the trail shots).

do you have any idea about software to be used? I have 550D 5D GoPro and EOS-M and NOT a GH2 :-(

@RobertoAlfieri - most of the 5Ds and 550D both support MagicLantern which will do the slow shutter for you.  I highly HIGHLY recommend it if you are doing videography with your DSLRs.  I actually prefer Magic Lantern to most of the GH2 hacks, though the GH2 hack is hard to beat in terms of value for the cost of the camera.  I shoot Canon bodies but a buddy of mine that I work with a couple times a year shoots GH2s and GH3s and we always compare notes.

i actually HAVE MagicLantern on my 550D, but it does not go under 30... #ohwait!

AJ, here is another one for your repertoire: There are specialized frame buffers configured to reiterate images as illustrated in the example. I credit Ed Tannenbaum for developing this device as far back as 1980 when it debuted at the Exploratorium in San Francisco. For more information Google with key words "Ed Tannenbaum Exploratorium". Ed is a colleague of mine and he told me that his inspiration for this device was from watching Norman McLaren's 1968 film "Pas de Deux".

@filzilla - good to know, can this easily be accomplished on non-specialized hardware though?  If not, I'm not sure it is particularly helpful for this answer, though it certainly is interesting information from a historical context.  (Really, it is what the After Effects approach is emulating if I understand you correctly, it just does it with a specialized frame buffer.)

Oh, by no means did I mean to discredit your answer, I was merely trying to enhance it from a historical perspective. It won't be long before many NLE will offer this effect if I am reading the demand correctly.

@filzilla - I didn't take it as trying to discredit my answer, I think it is great perspective, just doesn't help practically unless it can be replicated using standard hardware or software.  I'm not aware of any standard hardware that can do that and I think that my software solution is doing the same thing, unless I am misunderstanding what Ed Tannenbaum did.  Was just explaining why I wasn't updating the answer but I'm glad you commented about it.

I think you covered it well, and AFAIK there is no way to accomplish what Ed's device can do because it can have way more reiterations then any NLE on any platform can handle to date.

@filzilla - you could do it with scripting in AE to blend with previous frames I think, but it would be a pain to write.


text:
Easiest way to create a continuous shot effect using multiple cameras?
Here is the Script:
Lets take the opening scenes of Hugo, of course the film bagged Academy awards, In the scene that starts by following the kid (Hugo), carries on and on for some time with many complex movements and dynamics and finally reaches at a decent close frame.
Yes, in the making it was revealed that it was 3 different boys acting the scene.
What I ask here is an easier way, or maybe easiest among the lot (in the sense of both production cost and human resource/effort) to imitate a continuous shot effect using multiple cameras.
I have heard from my friend (a professional CG artist) that it can be done using 3D techniques. It involves the creation of the entire stage (the area where the scene occurs in real film) in the 3D virtual space. and then intelligently filling up the cuts between cameras using this 3D virtual fillers.
I am really open to that idea. But, are there any real life (no virtual/3D) techniques to establish a continuous looking scene?

CG is probably the best bet.  Motion controlled cameras would be the second best bet, but it would still be pretty tricky as the real world has a remarkable ability to change subtly, but substantially enough to result in a jump cut pretty quickly.  Even if you could manage to stop the camera on something that is static, resuming the movement at the same speed would be difficult at best without a computer controlled camera.
If you are able to have the camera stop movement on scenery in between, that might make a possible stop point, but it would still have to be done pretty quickly to avoid changes in lighting conditions that would be noticeable in the final shot.

hmmm yes, @AJ HEnderson  you are right about the real world inconsistency of marker objects , may be we can use programmable rigs or jibs (but i think the movements will be limited right ? ) in a non natural ( sets ) to some extend. Still without CG its hard to get that natural flow, right ?. Is there any low cost (or low rent ) gadgets with such memory capacity to store the  trajectory , also to continue from some point in between the start and stop of the recorded trajectory.

@Abhijith - someone else would have to comment on that, I have never personally had the priviledge of working with any motion controlled rigs, I just know of their existence and rough capability.  I know they could potentially be helpful in this case, but know basically nothing beyond that.

@Henderson , thank you AJ. :-) seekers finders ;-) i will post here if i am able to get some info :-)

@Abhijith - one other thought, I suppose you might be able to get a rough but workable result with a basic PTU that has a fairly consistent max velocity and the ability to lock an axis.  You could capture with the movement along X at max speed and recapture over the same transition once you make a quick move of your subject.  It might be possible to align and cut the clips (or even just cross fade them) smoothly in post.

link You mean like this clip (an old Indian film ). But the freedom of movement would be limited na ? i didn't makeout what PTU means.

@Abhijith - not able to see the clip right now, but PTU is short for Pan Tilt Unit.  It's a robotic (but not motion controlled) head that you can mount a camera to so that it can be remote controlled.  They are often used with installed cameras or on the ends of cranes.

@Henderson- Alright, i'll check it and get back soon :-)

@henderson: Found this.. Link


text:
2GB vs 4GB Graphics card Memory for playback and encoding of Videos in Adobe Premiere CC
While looking into buying a new graphics card for the editing, live playback and encoding of videos in the new 64bit Premiere Pro CC, I was wondering about:
1) how and when a graphics card's memory is utilised or loaded?
2) when would one benefit from the higher graphics card Memory?
Would I benefit during editing/live playback, or when encoding and exporting a video?
As of summer 2014 the Nvidia GTX 770 seems a sweetspot in terms of price/performance that grasped my attention. This model is abvailable in 2GB and 4GB versions. I will be using some filters and will be using Magic Lantern RAW 1920x840 @ 24 fps footage timeline for short videos of around 4 minutes. Thanks!


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Recommendations for good "bad monitors"?
To make sure the songs still sound OK on crap systems like small radios and computer speakers I've been listening to recordings through my computers speakers, a surprisingly good value €50 Creative Labs set of small speakers + subwoofer (you know, this kind of thing). Although that has been useful, it means I can only listen to the mix after I've mixed it down to my computer (I use a Tascam 2525 mixing desk and midi sequencing a lot).
So I'm looking for a set of extra monitors. They need to be "crap" for a certain value of crap, they must be active, and can't cost more than a couple of hundred dollars and not to big. So yes, another set of creative speakers + subwoofers fits the bill, but maybe somebody has good recommendation for alternatives.

I'm struggling trying to answer this in a way that would enhance what you've got already. Because really, you've got a pretty good representation of "everyman's speaker" there with the Creative set you have already.
So I'm going to take a different approach here and, instead of recommending something to buy, I'm going to recommend some techniques I use to test my mix to see if it'll work on other systems (I mix on a KRK Rokit setup and my big mains are JBLs driven by a McIntosh amp).
Test on a clock radio. They are almost the ubiquitous music listening device. We all have them in our rooms. There are a few, guaranteed, in every office space. Your mechanic has one in his office. They are everywhere. And they sound pretty much like poo. I have a line in on my little Sony clock radio. I'll bounce a mix, drop it on my iPhone, and try it out on the clock radio via the line in.
Test you mix without the sub. Most people don't have one. So click off the sub on your Creative setup and see how it sounds. Is there still a reasonable representation of the low end happening when it's just a small speaker setup?
Test your mix in mono. Or rather: move away away from the sweet spot. I like to throw a mix on in a loop and then walk out of the room. Wander around the house. Listen to it like most people listen to music: while they're going about they're day and it's background to their lives. Few people spend the time to sit down in the proper spot, between two nice speakers, and really enjoy all that hard work you spent placing each instrument in to it's own space. Sad, but true.
Test your mix in the car. Music and cars are just kind of like pie and ice cream. It needs to sound good in the car.
Test your mix in headphones. If only to avoid the "Sgt. Pepper's Effect" where you've panned things to extremes and when you listen to them in headphones it becomes a bit like having your head pulled apart by the ears. I test headphones last because I find if I tune for them I'll end up with a mix that's too bass heavy on regular speakers.

Turning off the sub was a good idea that I haven't done.

license: Creative Commons - Attribution Share-Alike - https://creativecommons.org/licenses/by-sa/2.5/


text:
Utility or method for joining MP4 files
I have two MP4 files that I generated with Camtasia Studio. I need to join them together post rendering with Camtasia (it can't be done during the Camtasia rendering process). Looking for a utility or technique that will work to join these, preferably that doesn't require re-rendering all the video again.
Update: Either Windows (preferred) or Mac. Doesn't need a GUI if it is a good command-line app.

It would help to know what platform/OS you are using.

@filzilla: Either Windows (preferred) or Mac.

See http://superuser.com/q/51047/31983

I think the simplest way to this is to use MP4Box.
./MP4Box -add file1.mp4 -cat file2.mp4 -cat file3.mp4 output.mp4

I was encountering audio sync issues when trying to do this with Avidemux, but I came upon a two-step process that keeps the audio in sync:

Concatenate files using MKVToolNix (I used version 7.4.0 for Mac). This results in a Matroska video (.mkv) file.
Remux the .mkv file to MP4 format using Avidemux (I used verison 2.6 for Mac). I chose the MP4v2muxer option in the Output Format section. This results in an MP4 file with audio and video in sync.

I would suggest using ffmpeg, and the -vcopy switch.  Usually this will work, but it depends on codec support.
Various builds of it have various codecs.  If one version of it doesn't work, try downloading a copy from a different site.

i think MPEG Streamclip can do this.  It's free on mac and pc.

Avidemux should do the work.

Load File
Go to File -> Append and select File
To the left of the video window,

set video mode to Copy;
same for audio mode;
set format to MP4.

Go to File -> Save -> Video.

This does what I am looking for, but it is causing the audio to go out of sync at the join point. I am going to try and join and mix the audio from a separate channel.


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How do I apply effects to a layer at its current resolution (scaled up by the layer's scale property) instead of its original resolution?
I'm a sprite animator in Adobe After Effects.
For example, I want to be able to skew a layer, but because this image is very small on its own, when I scale up its scale property and then skew it using one of a few possible effects, it does the skew based on its original resolution, meaning the pixels trying to show the skew are very large.
The only way I know of to do this is to use the transform property, and to scale it up that way, then use its scaling property. Trouble is that this means that the bounds of the layer are much smaller than the visible portion, and it means I have to screw with anchors and positions again to get everything back to normal, which is very tedious.
I basically want to be able to skew just like in Flash. Is there any way to do it that is much quicker and smoother than the method I described?
Here's an example image:

In the first and second sprite, it's scaled up by playing with the layer's scale property. On the third one, it's scaled up using the effect called "Transform", and then I use CC Slant to skew it. For a few reasons, this is tedious and a bit unwieldy, so is there a better way to do this without using Transform? I know I can do something similar by precomposing the sprite, but this is also unwieldy.
I should also mention that I only own AE CS5.

i find your question incredibly hard to follow, pictures illustrating each step/process would be extremely beneficial. side note, throwing it out there, maybe try using the continuous rasterize option? (i probably dont understand the question adequately)

Here's an image: http://puu.sh/sWmWO/79e1b713da.png
In the first and second sprite, it's scaled up by playing with the layer's scale property. On the third one, it's scaled up using the effect called "Transform", and then I use CC Slant to skew it. For a few reasons, this is tedious and a bit unwieldy, so is there a better way to do this without using Transform? I know I can do something similar by precomposing the sprite, but this is also unwieldy.

precomps are kind of required in ae, otherwise things differently get unwieldy. perhaps ae isnt the right tool for your job.

AE gives me a lot more possibilities, and allows these kinds of animations to achieve a certain quality, which is why I use it. Precomps aren't so bad, but the main issue is trying to edit things that you need to see multiple comp layers for, and being stuck in only one frame. And I heard from someone else that if I use a precomp, then I need to animate the 3D camera in each precomp (I use 3D space for parallax).

The final point isn't a 100% accurate. You can click the continuous rasterize button (little sun) on the precomp layer and 3d from the precomp will be brought in to the comp with the camera. If that makes sense? Otherwise I can try and explain better.

I just played with it, and it works beautifully. I'd prefer the simplicity of a plugin that applies effects at the current resolution, but if this is all I can do, then this is pretty damn close, and I'll be satisfied.

Yea it's a little known feature that most people only seem to use for vector graphics but it works nicely for organizing 3d comps. Glad I could help

Well, the answer to the question in your title is to apply the CC Slant effect to an adjustment layer above your (scaled-up) sprite layer. The adjustment layer will have the same resolution as the comp.
But based on the description below the title, I'm not 100% confident this is what you want. Let me know.

This is almost what I want! This is so close! Now I just need a reliable way to apply this only to the layer or layers I want. Precomps would help, but then I could just apply the effect to the precomp itself, and I want to avoid precomps as the additional levels of comps within comps means going in and out of stuff, which can make multi-level editing more difficult.

Well, I think that not allowing precomps is an odd constraint. Maybe After Effects is not the right tool for the job. But good luck!


text:
Premiere: Renaming cameras in a multi-camera sequence
Is it possible to rename cameras in a multi-camera sequence?
Why I need this:
I have a multicam sequence that shows footage that has to be synced with audio for certain people. However it is hard to tell which person's camera it is from what can be seen on the footage. In order to be able to quickly figure out who's view is being shown and what audio to sync with it, it would be nice to be able to rename those cameras to something more descriptive than "camera 1, camera 2, etc ..."

When you create a Multi-Cam sequence, the prompt will give you the option to name the cameras based on the track or clip name.

I don't believe there is a way to rename the cameras after setting up Multi-Cam.
One other option would be to put each clip into a sequence with a text layer as an overlay. After you have finished arranging your Multi-Cam composition, the text layer can be hidden or deleted.

ah cool ... thanks. Should've spent more time in the creation dialogue ... I really hope Adobe will let us do changes like that after the fact some day. Had to do the whole procedure three times because I forgot enable audio switch lol


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What's the difference between Behringer XENYX 1202FX and the Behringer Eurorack UB1202FX?
Behringer makes two mixers with preamps that look the same. One of them is Eurorack lineup.

XENYX 1202FX (sometimes marketed as premium)
EURORACK UB1202FX

Here are the pictures:

Could someone help me discern the difference between the two lineups. Feel free to remove the association with 1202 in the Question if the answer applies to the Eurorack and Xenyx lineups in general.
Even the discreptions are vague:

Ultra-low noise design, highest possible headroom, ultra-transparent audio

vs:

Premium ultra-low noise, high headroom analog mixer

Other oddities make it really hard to compare. With the 1202FX +48V phantom power is listed as a bullet. With the UB1202FX, it's only listed in the description.

This is a very good question and I'm having a look for you. It's not painfully obvious if there are any differences at all but if I find out anything you will be the first to know :)

@Jazibobs If you think it's a good question, do upvote it. Questions need votes too :)

have done friend :D

The telling part is in the name. If you are not aware, Behringer has had a longstanding feud with Mackie (among other manufacturers as well) about copyright and trademark violations. Behringer has lost at least one of these lawsuits and were forced to change their cosmetic design on many of their mixers.
Their Xenyx series is a not so subtle copy of Mackie's Onyx series preamps. The Onyx series were a noticeable step up from the vLz and vLz Pro preamps. Behringer decided to create their own premium line and call them Xenix. The problem is, I have yet to read a review that states a discernible difference between the UB and Xenyx lines. My presumption is that at this basement price point, it hardly matters.
If you could audition both (many online retailers have a return policy), I wouldn't bother trying to find qualitative differences...I'd just listen for noise floor differences.

I researched both, and though the specs have the same features and the same sound quality, XENYX beats EURORACK because of the preamps.  XENYX mic pre is a bit better than the EURORACK.  You should purchase good quality microphone cables in either case to avoid excess noise.
In the end both are good desks.  XENYX only wins because of preamps.  Otherwise, the two have almost the same quality.
I have helped good luck hug!  :)

Perhaps you could offer some support for your saying "XENYX mic pre is a bit better than the EURORACK."  :)

I know this is old, but just in case anyone else comes along looking for differences like I did, there's a big one that no one touched upon - the Xenyx has a direct USB connection. This makes a difference to me as I'd like to do some recording. Hope this helps!

This is not true. the 1202FX does not have a USB connection.

Captain - Evan is correct. The spec sheet from Behringer shows no USB connection on the 1202FX.

You can see from the photos that the XENYX 1202FX has an LED screen towards the top right side—absent on the XENYX 1202. That screen displays the various onboard FX processor settings that've been added to the FX version—hence, the name. A nice added feature, considering for a mere $20 more ($99 street price) you get 100 user selectable "real-world" presets that include, reverb, chorus and delay effects!

Question wasn't about 1202 vs 1202FX

To make matters more confusing, I've just taken apart my Xenyx1202fx for routine cleaning and we can see printed on the circuit board are the words "Behringer Eurorack UB1202FX."

This doesn't necessarily imply they're exactly the same product, however, as it could be that only the PCB is of the same design between these mixers, and for the XENYX, new components could have been used. For example the better preamps, as discussed.

I have several of each mixer. The Q1202USB has a compressor. The 1202FX does not. The Q1202USB has a digital USB interface. The 1202FX does not. The compressor is a useful feature and makes a difference, allowing things to cut through the mix. The Q1102USB is a notch above the 1202FX.

The question is about 1202FX and UB1202FX, however you are comparing against Q1202USB. Is this an alternative name for the product or have you mixed up something?


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How to trim / edit a video and save with original settings, without encoding
I have several videos from which I want to trim the end. Usually it takes too long time – so I need software that does not change the original settings and does not re-encode.
For example if my video 720×576, encoded with H.264  at 25 fps and a bitrate of 8000, these settings should remain – also for sound – and it should export as quickly as possible.

also, here is a very useful article on the subject: http://www.echeng.com/journal/2016/10/2/trimming-h264-mp4-video-files-without-re-encoding

MPEG Streamclip does exactly that


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How to show number of available streams using ffmpeg?
I just want to see how many streams are inside my video file.
I looked at the official documentation and I cannot see what option I can use. Perhaps another command/terminal app can do something similar?
Any ideas how it can be done?

If you have ffprobe installed:
ffprobe <input> -show_entries format=nb_streams -v 0 -of compact=p=0:nk=1

You can also filter for video or audio streams by adding -select_streams v or -select_streams a, respectively. See the manual for more details.
If you do not have ffprobe, ffmpeg can be used too, but in a slightly less robust way (i.e., this may break if the output format changes or something contains the words "Stream #" in the output):
ffmpeg -i <input> 2>&1 | grep "Stream #" | wc -l

More direct is ffprobe input -show_entries format=nb_streams -v 0 -of compact=p=0:nk=1

Thanks, post updated, I sometimes forget that there's a format class as well.

ffprobe.exe -i "vid.mkv" -show_entries format gets the job done, I don't know about all those options you listed.

@Shayan The options I've used ensure that you'll only get one number as output. Your command shows everything.

For a "smaller" command, we could use another writer: -of csv=p=0 instead of -of compact=p=0:nk=1

To count "number of available streams using ffmpeg'":
ffmpeg.exe -i %input% 2>&1 | find /c "Stream"

Tested in Win 10 CMD


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Audio discontinuities when generating HLS segments in different processes
I am creating MPEG-TS segments for HLS playback from multiple ffmpeg processes (it will be used for parallel encoding at a later stage). The commands are as follows — they can be run in sequence for testing purposes:
#!/usr/bin/env bash
rm seg*.ts master.m3u8

ffmpeg \
  -f lavfi -i testsrc=s=320x240:r=30 -f lavfi -i sine=440 \
  -vf 'drawtext=text=%{n}:fontsize=72:r=60:x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=1: boxcolor=0x00000099' \
  -pix_fmt yuv420p \
  -preset ultrafast -c:v:0 libx264 -x264-params "nal-hrd=cbr:force-cfr=1" -b:v:0 4M -maxrate:v:0 4M -minrate:v:0 4M -bufsize:v:0 8M -g 60 -sc_threshold 0 -keyint_min 60 \
  -f hls -hls_time 2 -hls_playlist_type event \
  -hls_flags independent_segments+append_list+omit_endlist \
  -hls_segment_type mpegts -hls_list_size 0 \
  -hls_segment_filename seg_01_%02d.ts -master_pl_name master.m3u8 -start_number 0 \
  -muxdelay 0 \
  -muxpreload 0 \
  -output_ts_offset 0 \
  -t 10 \
  master.m3u8

ffmpeg \
  -f lavfi -i testsrc=s=320x240:r=30 -f lavfi -i sine=440 \
  -vf 'drawtext=text=%{n}:fontsize=72:r=60:x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=1: boxcolor=0x00000099' \
  -pix_fmt yuv420p \
  -preset ultrafast -c:v:0 libx264 -x264-params "nal-hrd=cbr:force-cfr=1" -b:v:0 4M -maxrate:v:0 4M -minrate:v:0 4M -bufsize:v:0 8M -g 60 -sc_threshold 0 -keyint_min 60 \
  -f hls -hls_time 2 -hls_playlist_type event \
  -hls_flags independent_segments+append_list \
  -hls_segment_type mpegts -hls_list_size 0 \
  -hls_segment_filename seg_02_%02d.ts -master_pl_name master.m3u8 -start_number 0 \
  -muxdelay 0 \
  -muxpreload 0 \
  -output_ts_offset 10 \
  -ss 10 \
  -to 20 \
  master.m3u8

You can load this in your browser if you simply run a web server like python3 -m http.server with:
<html>
  <head></head>
  <body>
    <script src="https://cdn.jsdelivr.net/npm/hls.js@latest"></script>
    <video id="video" controls autoplay></video>
    <script>
      var video = document.getElementById("video");
      var videoSrc = "master.m3u8";
      var hls = new Hls({
        debug: true,
      });
      hls.loadSource(videoSrc);
      hls.attachMedia(video);
    </script>
  </body>
</html>

Now, the video seems to play fine. Looking at the frame counter, I see that no frames are lost (although it seems like the very last frame of segment set 1 is displayed a little bit shorter).
The audio, however, has a small dropout at the switch from segment set 1 to 2.
The generated M3U8 has an #EXT-X-DISCONTINUITY tag before switching to the second segment set.
I noted that the start times of the segments don't quite correspond when comparing the audio and video streams. To test this, I run ffprobe and look at the start_time and start_pts values:
for s in *.ts; do ffprobe -loglevel error -hide_banner -select_streams v -show_streams -of json "$s" | jq --arg input "$s" '.streams[0] | {$input, duration, start_time, start_pts}'; done | jq -s .

This is the output:
[
  {
    "input": "seg_01_00.ts",
    "duration": "2.000000",
    "start_time": "0.023222",
    "start_pts": 2090
  },
  // ...
  {
    "input": "seg_01_04.ts",
    "duration": "2.000000",
    "start_time": "8.023222",
    "start_pts": 722090
  },
  // ...
  {
    "input": "seg_02_05.ts",
    "duration": "2.000000",
    "start_time": "10.000000",
    "start_pts": 900000
  },
  // ...
]

The second set is forced to start at 10 seconds, while the last segment of the first set actually extends to 8.023222 + 2 = 10.023222 seconds. This is due to the non-negative offset of the first segment of the first set.
For audio, the timestamps and durations are completely different:
[
  {
    "input": "seg_01_00.ts",
    "duration": "2.043344",
    "start_time": "0.000000",
    "start_pts": 0
  },
  // ...
  {
    "input": "seg_01_04.ts",
    "duration": "1.996900",
    "start_time": "8.034111",
    "start_pts": 723070
  },
  {
    "input": "seg_02_05.ts",
    "duration": "2.043344",
    "start_time": "9.976778",
    "start_pts": 897910
  },
  // ...
]

Is there any way to fix this, so that the segments generated by the second ffmpeg command can be stitched during playback, without choppy audio?

I have tried another approach, which consists of generating the audio stream separately before starting to encode the video. The audio and video playlists are written into dedicated .m3u8 files which I am simply referencing from a master.m3u8 playlist. This one I have to  manually generate, because otherwise ffmpeg would overwrite it. The whole approach is shown here.
The problem is that the video won't start in hls.js:

[log] > Unknown video PTS for cc 0, waiting for video PTS before demuxing audio frag 1 of [0 ,10],track 0

This can be demoed here.
Once I remux everything into new sets of segments, as suggested by @Gyan, it works again. Demo is here.
However, this is not a workable solution here, since I need to be able to generate the HLS segments on the fly, and append to the playlists later.

I was able to make it work with the individual audio and video generation by ensuring that audio is generated with ffmpeg's discont_start flag set.
-hls_flags independent_segments+append_list+discont_start

I noticed that when looking at the CC counters in the hls.js debug logs, these were not aligned. Video started with 1, audio witht 0.
Looking at the playlists, for some reason, video began with #EXT-X-DISCONTINUITY, so I had to put an extra one into the audio stream to have the continuity counters align (i.e., start with 1 for both streams).

You have to render the audio in one command.
MDCT-based codecs like AAC or MP3 carry out their transforms over multiple adjoining frames. When doing a segmented encode of a single stream, the encoder 'supplies' silent frames at the top and tail of each segment.
From Apple:

AAC requires data beyond the source PCM audio samples in order to
correctly encode and decode audio samples due to the nature of the
encoding algorithm. AAC encoding uses a transform over consecutive
sets of 2048 audio samples, applied every 1024 audio samples
(overlapped). For correct audio to be decoded, both transforms for any
period of 1024 audio samples are needed. For this reason, encoders add
at least 1024 samples of silence before the first ‘true’ audio sample,
and often add more. This is called variously “priming”, “priming
samples”, or “encoder delay”.

During muxing, these priming samples can either be discarded or kept. If kept, there will be a gap at the seams. If discarded, the decoder will use the tail frame of the previous segment for overlap purposes and the result will be a glitch.

So, in essence, there could be a dedicated process rendering the .ts chunks for audio completely separately — but how would you then combine them back into a single .m3u8 master playlist?

See option var_stream_map in the hls muxer docs. You will have two inputs in ffmpeg: -f concat -i video_segs_list -i audio_stream -map 0 -map 1 .... You can also concat the video segs into one TS file and use that.

That indeed produces a playable stream. Thanks! But due to the live requirement and wanting to be able to append video segments to the playlist as they are being generated, I can't do a single remux command, because that will rewrite all the segments and playlist files. I can generate the audio in a separate thread before, and only then start outputting the video segments, but that approach yields a stream that never starts playing in HLS.js. It seems to wait for an initial video PTS but doesn't get any: https://pastebin.com/raw/MAYJyHs5 — any idea?

I'll check out the details of your process tomorrow.

Much appreciated. In case this becomes more complex, if you are interested, we can discuss consulting offline.

I think I was able to make it work with the individual audio and video generation by ensuring that audio is generated with ffmpeg's discont_start, and video is not. For some reason, video still begins with #EXT-X-DISCONTINUITY, so I had to put an extra one into the audio stream to have the continuity counters align.

@slhck I am curious to know if your results would be helpful to me in solving https://sound.stackexchange.com/questions/51209/possible-to-ffmpeg-encode-an-mp3-media-segment-for-hls-to-follow-existing-mp3 also let's get a beer sometime :)

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