My Premiere Pro exports give me little black borders on the top and bottom

[budwzr]

Well, if upsampling does nothing, why does Ken Burns bother to do it?

No, it does not really improve native resolution, but after using an Unsharp Mask it does improve "apparent" resolution. It's a cheat, a con, a flimflam, whatever, but it does trick the eye away from the otherwise softness introduced by zooming after stabilizing.

I don't want to detract from the topic, so I will pull back on the upsampling, and retract my comment. I use it frequently when I need to do Edge Detection.

[poisondeathray]

Well, if upsampling does nothing, why does Ken Burns bother to do it?

No, it does not really improve native resolution, but after using an Unsharp Mask it does improve "apparent" resolution. It's a cheat, a con, a flimflam, whatever, but it does trick the eye away from the otherwise softness introduced by zooming after stabilizing.

I don't want to detract from the topic, so I will pull back on the upsampling, and retract my comment.

If you mean upsampling, as in upsampling the chroma - that happens as soon as you import into vegas, it's converted to RGB. RGB is full color. But there are different methods / algorithms of upsampling - you might choose one over another in some circumstances. And Adobe screwed the pooch because it used the wrong chroma upsampling algorithm (progressive instead of interlaced)

You cannot create details that were never there. You can sharpen , create false details (even aliasing can look like "details") , and it may improve subjectively the look , but it might also worsen it (halos, oversharpen artifacts)

But definitely, some people might sharpen it a bit after stabilization, because the zoom will make it softer. But the proportion remains the same, it doesn't affect how much you have to crop if you upscale

[budwzr]

Here's a sample.

http://files.videohelp.com/u/135518/UnSharp%20Mask%20Comparison.mp4

Second image is 300ppi.

[CZbwoi]

Waking up this thread for some help as I got Vegas. I'm dragging a bunch of files into Vegas now and for every file I insert it places these 2 files inside of my source folder where the files are: .scc and .sfk files. What are these and why are they appearing? No such thing appears for PP or anything else I've ever used, this is odd.



Then after manually erasing them, the .sfk ones seem to come back once I move around in Vegas. I even closed the new project, didn't save anything whatsoever and they're still there; they didn't even disappear from getting rid of an unsaved project.


Also this is very miniscule but in the Vegas timeline, even though I'm zooming in it's not showing frame by frame representation of the video, it's showing one frame every however many seconds followed by a grey streak. Is this normal or is there a way to get rid of the grey streak and have actual frame representation from the videos appear?




edit: Another thing, I loaded a basic script into MeGUI with the frameserving going on, only 1 video in the Vegas timeline:

SetMTMode()
AVISource("k:\untitled.avi")
AssumeTFF()
ConvertToYV12(interlaced=true)
QTGMC()
Tweak(hue=0.0, sat=1.6, bright=4, cont=1.0, coring=True, sse=False, startHue=0, endHue=360, maxSat=150, minSat=0, interp=16)

Opens up fine in MeGUI, preview works, but every time I encode it never completes when it gets to the video part. It stayed stuck at 97.69% and after I aborted that and tried it a second time it stuck at 97.78%. Is this because I'm frameserving wrong, something wrong with the script, something wrong with the encoding settings?

edit 2: Well this is weird, it finally finished after I randomly decided to move around towards the end of the timeline in AvsPmod as the script was still open in there (not sure if there's a correlation). The muxed final file it gave me has the green flash at the start of it (really fast this time), there are 4:3-like black borders on both sides even though it says it's still 720x480 (I think this is because of how Vegas is frameserving it, in the wrong aspect?), and there is no audio.

edit 3: Running it a third time, no correlation, now paused at 97.81% with AvsPmod closed. It's been like that for an extra 20 minutes now, stuck. Not sure how moving through the timeline in AvsP that one time made it finish, but now with it not even open since the beginning...it's just been staying stuck at 97.81%. If it means anything, the frameserver format settings is YUY2, not sure it does.

[smrpix]

.sfk are the drawings of your audio waveforms. .scc is subtitles - most likely your time and date stamps. Premiere does exactly the same thing, it just stores them in a different folder unless you specify otherwise.

[budwzr]

There's only a handful of people that understand this Frame-serving stuff. It goes back to the old DOS days.

[poisondeathray]

Also this is very miniscule but in the Vegas timeline, even though I'm zooming in it's not showing frame by frame representation of the video, it's showing one frame every however many seconds followed by a grey streak. Is this normal or is there a way to get rid of the grey streak and have actual frame representation from the videos appear?

Did you mean thumbnails on the timeline itself on the video layer ? You can enable it in the options=>preferences => thumbnails to show in video events and check which option you want (e.g. none, or all, head/tail etc..)

edit: Another thing, I loaded a basic script into MeGUI with the frameserving going on, only 1 video in the Vegas timeline:

SetMTMode()
AVISource("k:\untitled.avi")
AssumeTFF()
ConvertToYV12(interlaced=true)
QTGMC()
Tweak(hue=0.0, sat=1.6, bright=4, cont=1.0, coring=True, sse=False, startHue=0, endHue=360, maxSat=150, minSat=0, interp=16)

Opens up fine in MeGUI, preview works, but every time I encode it never completes when it gets to the video part. It stayed stuck at 97.69% and after I aborted that and tried it a second time it stuck at 97.78%. Is this because I'm frameserving wrong, something wrong with the script, something wrong with the encoding settings?

edit 2: Well this is weird, it finally finished after I randomly decided to move around towards the end of the timeline in AvsPmod as the script was still open in there (not sure if there's a correlation). The muxed final file it gave me has the green flash at the start of it (really fast this time), there are 4:3-like black borders on both sides even though it says it's still 720x480 (I think this is because of how Vegas is frameserving it, in the wrong aspect?), and there is no audio.

edit 3: Running it a third time, no correlation, now paused at 97.81% with AvsPmod closed. It's been like that for an extra 20 minutes now, stuck. Not sure how moving through the timeline in AvsP that one time made it finish, but now with it not even open since the beginning...it's just been staying stuck at 97.81%. If it means anything, the frameserver format settings is YUY2, not sure it does.

In the dmfs configuration, frameserve RGB, not YUY2. Vegas is working in studio RGB with your native files. Make sure audio is checkmarked if you want audio in the signpost. Use ConvertToYV12(interlaced=true, matrix="pc.601") to essentially "undo" the studio RGB conversion. I'm pretty sure I mentioned this earlier

Try swapping out the DLL, since you are running "vanilla" avisynth . For me I just copy/paste, but Groucho2004 has a utility for swapping versions (if you can't copy/paste for some reason, maybe due to permissions)
http://forum.doom9.org/showthread.php?t=172124

You need to set the --sar in the encoding settings if you are not resizing with sqaure pixels. Recall that's what started this thread ? Recall ITU vs. non ITU aspect ratios?
--sar 32:27

Or resize to a square pixel format, like 854x480, and keep --sar 1:1 or undefined. You can resize in your script if you wanted to go that route

Personally I don't like using megui for something this because it does audio & video separately. Essentially you have to frameserve twice, once for audio, once for video, and a separate step for muxing. If you have lots of stuff, filters going on in vegas, it can be very slow unless you go back and toggle stuff off manually for the audio step. Using ffmpeg you can encode audio & video and mux in 1 step

It works for me, I'm able to frameserve your test video and complete it. But if you can't get it working despite all those suggestions, you can still use the lossless intermediate option.

[CZbwoi]

.sfk are the drawings of your audio waveforms. .scc is subtitles - most likely your time and date stamps. Premiere does exactly the same thing, it just stores them in a different folder unless you specify otherwise.

Is there a way to make them not appear at all or in a separate default Vegas folder instead like in it's Program Files? I want this folder I have with all this source cam footage to be strictly that, I don't want 2x more the amount of meaningless data files in there with the bunch getting in the way of everything.

Did you mean thumbnails on the timeline itself on the video layer ? You can enable it in the options=>preferences => thumbnails to show in video events and check which option you want (e.g. none, or all, head/tail etc..)

Thanks for that, it was in Options > Preferences > Video > Thumbnails to show in video events > All

In the dmfs configuration, frameserve RGB, not YUY2. Vegas is working in studio RGB with your native files.

Got it, but which RGB do I choose? In Vegas there's an option for RGB24 and RGB32. Is there a specific one I should always choose or does it vary on the file?

Or resize to a square pixel format, like 854x480, and keep --sar 1:1 or undefined. You can resize in your script if you wanted to go that route

Forgot about trying to insert that into avisynth like how I did in PP, thanks, it makes it a little simpler than figuring out what SAR to use. Question about resizing in the script, I always use the Spline36Resize(854,480) code to resize but is this right to use all the time? I see people mention one called Spline64Resize and so on. Is one generally better than the other? I read that 36 is the "middle-ground" but for different source materials it can vary.

Personally I don't like using megui for something this because it does audio & video separately. Essentially you have to frameserve twice, once for audio, once for video, and a separate step for muxing.

No, there's an auto-encode button in the bottom right that can do it all for you in one step as well. Let me showcase to you the odd thing that's happening though, I ran it once more and it stopped at 97.81% again. I went back into AvsPmod like I did that one time, moved in it's timeline further to the end, and all of a sudden the percentage in MeGUI went up a bit. Okay, weird. Then I went all the way to the end of it and the encoding finished. AvsPmod then goes into (Not Responding) mode for all eternity until I decide to force shut it down.

I play the file and it's all cool until the final couple seconds where it repeats, stutters and freezes on a frame. I'll upload it for you to see.

SetMTMode()
AVISource("k:\untitled.avi")
AssumeTFF()
ConvertToYV12(interlaced=true, matrix="pc.601")
QTGMC()
Tweak(hue=0.0, sat=1.6, bright=4, cont=1.0, coring=True, sse=False, startHue=0, endHue=360, maxSat=150, minSat=0, interp=16)
Spline36Resize(854,480)

Also do you think it looks good? I started typing tweak in AvsPmod and then it autofilled to that if I pressed enter. I don't know how that happened but it ended up looking pretty good to me so I kept it.

But here's the video:

[_Al_]

It is a bug, frame server has to be re-run again to get an audio. Audio works only for first run. Vegas 8 does not have this bug but later versions might have. That my script goes around that bug, where muxes pre-rendered audio into MP4. But for some reason you do not want to try it. Just drop signpost.INI and signpost.avi (export RGB32) onto _AVI to MP4.BAT.


your signpost.INI file:

path_destination      :C:\Destination
path_temp             :C:\Temp
x264_variables        :--crf 18 --vbv-bufsize 20000 --vbv-maxrate 20000
mux_external_audio    :yes

write avisynth script between lines, no loading clip line in there!
--------------------------------------------------------------
AssumeTFF()
ConvertToYV12(interlaced=true, matrix="pc.601")
QTGMC()
Tweak(hue=0.0, sat=1.6, bright=4, cont=1.0, coring=True, sse=False, startHue=0, endHue=360, maxSat=150, minSat=0, interp=16)
Spline36Resize(854,480)
--------------------------------------------------------------

so this batch script goes around this audio bug, it needs to have prerendered signpost.m4a in that batch script folder, you do it in Vegas before, mainconcept mpeg4 render, uncheck video, so just m4aaudio will be created, audio is rendered pretty fast

[poisondeathray]

.sfk are the drawings of your audio waveforms. .scc is subtitles - most likely your time and date stamps. Premiere does exactly the same thing, it just stores them in a different folder unless you specify otherwise.

Is there a way to make them not appear at all or in a separate default Vegas folder instead like in it's Program Files? I want this folder I have with all this source cam footage to be strictly that, I don't want 2x more the amount of meaningless data files in there with the bunch getting in the way of everything.

I don't know offhand, not sure if you can change the folder, but you can have a look in the options

In the dmfs configuration, frameserve RGB, not YUY2. Vegas is working in studio RGB with your native files.

Got it, but which RGB do I choose? In Vegas there's an option for RGB24 and RGB32. Is there a specific one I should always choose or does it vary on the file?

Usually RGB24. RGB32 would be for when you need to keep an alpha channel (transparency) in the frameserved project (for example, for compositing in other applications)

Or resize to a square pixel format, like 854x480, and keep --sar 1:1 or undefined. You can resize in your script if you wanted to go that route

Forgot about trying to insert that into avisynth like how I did in PP, thanks, it makes it a little simpler than figuring out what SAR to use. Question about resizing in the script, I always use the Spline36Resize(854,480) code to resize but is this right to use all the time? I see people mention one called Spline64Resize and so on. Is one generally better than the other? I read that 36 is the "middle-ground" but for different source materials it can vary.

You can read about the various resizing algorithms when you have time, but spline36 is fairly sharp, spline36 is even sharper. "Sharp" isn't necessarily good, because you get ringing/edge sharpening artifacts. So it really depends on the situation and personal taste what you might use

Here is a HTML5 visualization on some resizing kernals
http://svn.int64.org/viewvc/int64/resamplehq/doc/kernels.html

Personally I don't like using megui for something this because it does audio & video separately. Essentially you have to frameserve twice, once for audio, once for video, and a separate step for muxing.

No, there's an auto-encode button in the bottom right that can do it all for you in one step as well. Let me showcase to you the odd thing that's happening though, I ran it once more and it stopped at 97.81% again. I went back into AvsPmod like I did that one time, moved in it's timeline further to the end, and all of a sudden the percentage in MeGUI went up a bit. Okay, weird. Then I went all the way to the end of it and the encoding finished. AvsPmod then goes into (Not Responding) mode for all eternity until I decide to force shut it down.

I play the file and it's all cool until the final couple seconds where it repeats, stutters and freezes on a frame. I'll upload it for you to see.

SetMTMode()
AVISource("k:\untitled.avi")
AssumeTFF()
ConvertToYV12(interlaced=true, matrix="pc.601")
QTGMC()
Tweak(hue=0.0, sat=1.6, bright=4, cont=1.0, coring=True, sse=False, startHue=0, endHue=360, maxSat=150, minSat=0, interp=16)
Spline36Resize(854,480)

Also do you think it looks good? I started typing tweak in AvsPmod and then it autofilled to that if I pressed enter. I don't know how that happened but it ended up looking pretty good to me so I kept it.

But here's the video:

I think auto encode in megui just queues all the passes. It's not the same thing. It's going to be either slower (or much slower depending on the project) .

For sure audio is frameserved in dmfs for vegas 13 - I just double checked. I don' t know of any "bugs" here that _Al_ is referring to, and i don't know why you're getting this "bugginess" - it might have to do with your setmtmode() - recall avsmeter reported a non zero mt mode for you. I would swap out the .dll

If you're going to do multiple simultaneous encodes (because single threaded QTGMC) , then it might make more sense to export lossless intermediates. You might be able to run multiple frameserver instances with multiple vegas instances, not sure. I know you can have multiple instances of vegas open, but I've never frameserved from them all at once

I don't have time too look at your file yet but I personally would do the color work in vegas

[CZbwoi]

_Al_: It's sort of hard to understand what you're saying as English probably isn't your first language, but when I try to do what you say I get this error:

but spline36 is fairly sharp, spline36 is even sharper

I'm assuming you didn't mean to put the same thing twice, do you mean 16 for the latter? And the wiki says that 64 is the most accurate, what does that even mean lol? One is not as accurate but more sharp, one is more accurate but less sharp...ai ai ai. Even the wiki explaining it is confusing, but it looks well as is.

I think auto encode in megui just queues all the passes. It's not the same thing. It's going to be either slower (or much slower depending on the project) .

For sure audio is frameserved in dmfs for vegas 13 - I just double checked. I don' t know of any "bugs" here that _Al_ is referring to, and i don't know why you're getting this "bugginess" - it might have to do with your setmtmode() - recall avsmeter reported a non zero mt mode for you. I would swap out the .dll

I did so, uninstalled avisynth and reinstalled, in turn it put the original .dll back. When encoding the fps dropped and it took about 1/3 longer of the time but it actually finished without any freezing at 97%! All went good until I got to playing the end of the file, same stutter noise effect effect. I'm now looking at the original file and it is 58 seconds long, these new frameserved encodes are 59 seconds long, so for some reason it's adding a second and then it's looping one of the last sounds a couple times to make up for it. What could be causing this?

I set my audio encoder settings to Nero AAC and the extension MP4-AAC. When it was on the default FFmpeg AC-3 there'd be no audio when played in WMP and in VLC it'd be buggy and go mute for half a second if I skipped through it. Do you think this 1 second added at the end w/ weird audio bug is because of the audio encoder settings? If you know, what's the best thing to set it at or the standard for best quality?

[poisondeathray]

There shouldn't be any speed change if you were running single threaded (you probably weren't because had an active MT mode) . That might be the reason for you hanging. For some reason , some people just can't get MT to work

Spline64 is sharper, you can test it yourself in avspmod on a test image like patterns. Or have a look at the link, just look at the bar chart. (Sharper isn't necessarily "better", often it's a lot worse)

I think that's what _Al_ was talking about, besides the repeat audio glitch at the end , it also resamples to 44.1Khz - those were the reasons for prerendering audio
https://forum.videohelp.com/threads/365574-Sound-reverting-at-the-last-second-when-Fram...ing-from-Vegas

Or you can use the lossless intermediate route. Or go back to PP if you're more comfortable with PP. But the frameserver there is even more buggy, definitely recommend using intermediate there.

[CZbwoi]

I don't think the seconds bit matters, but the problem still does and I think its something with Vegas... When I frameserved out of PP and encoded it the sound glitch error at the end wasn't there...but the chroma problem is (if you recall that problem from PP with the red car freshener). Vegas works perfectly but gives me that sound repeat blip at the end with a paused frame, PP doesn't give me that but it still has the bad chroma effect. This is legit like a nightmare, every time I'm right there something else comes up, this sound blip pause frame thing now.

I've never been more frustrated with something in my life than with all this in the last 2 weeks...

If I were to export the audio from out of Vegas before frameserving, how should I do it as? There's .wav but about 16 options, 6 for 44,100, 5 for 48,000, .ac3 w/ stereo DVD and then one with "AGC" on, many .aa3 choices, .mp3. Flac? I just want a direct audio copy/rip.

I'm encoding right now with a random .wav selection export just to see if the glitch at the end remains.

[_Al_]

when I try to do what you say I get this error

Virtual Dub crashes when Avisynth script is loaded into VirtualDub because of your Avisynth problems. You use different Avisynth flavors etc. I use just the one downloaded from this website.
Btw. I deprecated that VirtualDub function, loading lately and replaced it with avsmeter. VirtualDub is not needed anymore in that script. You can try to download AVI to MP4-DVD.zip again and try it.

I think that's what _Al_ was talking about, besides the repeat audio glitch at the end , it also resamples to 44.1Khz - those were the reasons for prerendering audio

There are possibly three bugs for audio in total, they can occur or not, God knows what it depends on. Vegas version etc.

1. That audio frequency is one.

2. Second is silent audio. If frame server is ran second time (after encoding or even after testing avs script by software, audio is silent). So frame server has to start to generate audio as a absolute first thing, which is sometimes not possible. But doable if done manually. Like command line or even if software that does not do anything with avisynth script before.

3. Possible audio glitch at the end of audio track.

It is not a problem to pre-render that audio, Vegas generates AAC audio (in m4a container) or AC3 audio (192kbps without DVD Architect licence) without any problem. And I even adjusted my script for that, it can take that AAC or AC3 prerendered and create MP4 or VIDEO_TS

[_Al_]

If I were to export the audio from out of Vegas before frameserving, how should I do it as? There's .wav but about 16 options, 6 for 44,100, 5 for 48,000, .ac3 w/ stereo DVD and then one with "AGC" on, many .aa3 choices, .mp3. Flac? I just want a direct audio copy/rip.

-direct audio copy, export "Save as a Type" would be Wave (Microsoft), it is a PCM audio, so you'd ger audio rendered with wav extension, make sure you have set 48000Hz 16bit and stereo. You can use that audio for DVD but it is large, bitrate is 1500kbps.

-exporting AAC for MP4 "Save as a Type" would be MainConcept Mpeg4 with unchecked video, you'd get m4a container with aac in it. Bitrate is up to you. 224 should be enough, or max 384kbps. Higher values would make little sense. Frequency set the same as original audio is. AAC audio is good for MP4 container.

-exporting AC3 you'd export using Dolby Digital Ac3 Studio. That would get you 192kbps bitrate. For VHS? Perhaps more than enough. If you'd want more bitrate you'd need licence for DVD Architect and using Dolby Digital Ac3 Pro. Ac3 is good for DVD or MKV container.

[CZbwoi]

If I were to export the audio from out of Vegas before frameserving, how should I do it as? There's .wav but about 16 options, 6 for 44,100, 5 for 48,000, .ac3 w/ stereo DVD and then one with "AGC" on, many .aa3 choices, .mp3. Flac? I just want a direct audio copy/rip.

-direct audio copy, export "Save as a Type" would be Wave (Microsoft), it is a PCM audio, so you'd ger audio rendered with wav extension, make sure you have set 48000Hz 16bit and stereo. You can use that audio for DVD but it is large, bitrate is 1500kbps.

-exporting AAC for MP4 "Save as a Type" would be MainConcept Mpeg4 with unchecked video, you'd get m4a container with aac in it. This audio is good for MP4 container.

-exporting AC3 you'd export using Dolby Digital Ac3 Studio. That would get you 192kbps bitrate. For VHS? Perhaps more than enough. If you'd want more bitrate you'd need licence for DVD Architect. Ac3 is good for DVD or MKV container.

No DVDs btw, just .mp4 files.

Are you still talking about in Vegas? If you go to File > Render As, I don't see an option for MainConcept Mpeg4 and to uncheck the video.

But if a direct audio copy is possible with Wav then that's what I want, there are 4 options for Wav though if I select the 3 checkboxes: Audio templates only, Match project audio channel count, Match project audio sample rate. Those options are:
- Default Template
- 48,000 Hz, 16 Bit, Stereo, PCM
- 48,000 Hz, 24 Bit, Stereo, PCM
- 48,000 Hz, 32 Bit (IEEE Float), Stereo, PCM (Float)

Which one should be my option? Is default template the actual default source or is that a default Vegas setting for Wav files?
If I get this right and all this works, the sound glitch should be gone.


Btw do you know how to get rid of or move where these .scc and .sfk files get placed by Vegas so that they're not next to my original files anymore? I can't find anything that says this in the preferences. (I do not even understand why it's making closed captioning files when there are no closed captions or subtitles in them)

[_Al_]

Yes Vegas,
I just looked - precisely, It is called MainConcept avc (mp4) , whatever not MainConcept mpeg4, I have a box there that says Include video so I uncheck thta box, you have to have something like that in there in custom setting

That wave choose 48000 16 bit stereo.

I vaguely remember that location of those files cannot be changed. But just set folder priorities to type, they would pile up separately, click "type" is windows explorer. It is really no big deal then. On the contrary. After editing you just select them easily and delete. Or grabbing just avi's you select them all easily as well.

[poisondeathray]

I think the audio glitch for dmfs/vegas is only because of dmfs in some versions. I don't think intermediate export is affected

RE: PP and the chroma upsampling error - that only affects 4:2:0 MPEG2 when it's in a program stream (ie. .mpg). If you batch re-wrap them into a transport stream...poof the error disappears. Still a major goofup on Adobe's part. I see some posts from Adobe forum reporting this a few years ago and still not fixed... I guess nobody cares about importing generic MPEG2 or things like DVD

But the Adobe frameserver, either dmfs or advanced frameserver is like a crap shoot for CS6 or CC. Maybe a handful of people have it working consistently for whatever reason. Otherwise audio glitches, crashes, black/green frames, you name it. Earlier versions seemed to work ok

[poisondeathray]

RE: PP and the chroma upsampling error - that only affects 4:2:0 MPEG2 when it's in a program stream (ie. .mpg). If you batch re-wrap them into a transport stream...poof the error disappears. Still a major goofup on Adobe's part. I see some posts from Adobe forum reporting this a few years ago and still not fixed... I guess nobody cares about importing generic MPEG2 or things like DVD

Instead of doing a proper rewrap - even if you rename the extension from .mpg to .m2ts, it works for this ....

Hilarious.

[smrpix]

FWIW, I have no problems with dmfs and Adobe CC. I can't give you any special tips. I just installed it and it works with a variety of sources.

[CZbwoi]

That wave choose 48000 16 bit stereo.

I vaguely remember that location of those files cannot be changed. But just set folder priorities to type, they would pile up separately, click "type" is windows explorer. It is really no big deal then. On the contrary. After editing you just select them easily and delete. Or grabbing just avi's you select them all easily as well.

Thank you.

PDR: I think I'll stick with Vegas for now and just do the audio separately as I'm slowly but surely learning how to do that. Too many work-arounds to get it to work right in PP it seems (because I don't even know what it means to re-wrap something and would rather not inquire about it at this point xD).

smrpix: PP, right? If you can get it to work without that chroma problem (without those weird workarounds PDR mentioned) and audio glitch I'd love to know how.


Another mini-question for you guys, I encoded-muxed the Wav audio file I exported from Vegas with the frameserved script via MeGUI. When I selected the audio input as the Wav file it then makes me select the encoder, etc. I went to Nero AAC as the regular FFmpeg AC-3 is glitchy for me for some reason. When it finished I went to the file's details and the Bit Rate is 180 kbps, the original files are at 256 kbps. This means the encoded muxed file's audio is worse, right? Is this because of the Nero AAC or some settings? How can I match it with the original? I don't know which audio encoding to pick and what settings. Also the Wav file's bit rate is 1536 kbps which isn't like the original files' either, I thought that was supposed to be a direct copy, is it bigger because it's total combined files?

If this question is better suited for another topic let me know.

[poisondeathray]

Those PP workarounds aren't really "weird" . Rewrapping is just swapping containers. Think of it as taking the audio & video out of a "box" and putting it into another "box". But for some reason just renaming the extension works

And if you're one of the people that can get DMFS or Advanced FS working in PP properly, by all means use it. Same with avisynth MT - it works perfectly for some people with linear encoding. (But random access, random seeks will almost always screw things up for MT)

For audio - it's decoded and decompressed to uncompressed WAV in vegas, that's what you're frameserving. Even if you use the same bitrate , or even a higher bitrate than the original - there will be some quality loss because you're re-compressing it with lossy compression. Recall the FLAC vs. MP3 analogy. It's similar to that, but uncompressed PCM wav is lossless, and uncompressed, whereas FLAC is lossless but compressed. AAC is lossy compression. For the nero settings, you can use ABR or CBR encoding instead of quality based if you're fixated on a number (you shouldn't be)

[CZbwoi]

Those PP workarounds aren't really "weird" . Rewrapping is just swapping containers. Think of it as taking the audio & video out of a "box" and putting it into another "box". But for some reason just renaming the extension works

I'll have to uninstall dbfs and reinstall the other one again to try to do what you're saying, but I will to see if it works. But addressing what you say later on in the post, will there be audio quality lost if I frameserve the audio out too? Would it be better to export the audio losslessly as a Wav/Flac and mux that with the frameserved video? Or are they the same? (if all goes well and there's no audio bug at the end of the files again)

For audio - it's decoded and decompressed to uncompressed WAV in vegas, that's what you're frameserving. Even if you use the same bitrate , or even a higher bitrate than the original - there will be some quality loss because you're re-compressing it with lossy compression.

Oh no, I'm not talking about frameserving out audio out of Vegas, we already know that gives me the audio loop glitch at the end of files that's why I'm not f-serving the audio anymore. I'm talking about exporting the audio out of Vegas as a standalone file before I start the video frameserving, I was wondering which of the Wav options in File > Render As to select, but I think that the one simply labeled default is correct and lossless, I hope. Or if I should just go with .flac out of Vegas and simply select the .flac encoder in MeGUI and therefore it wouldn't re-encode it but simply mux it?

For the nero settings, you can use ABR or CBR encoding instead of quality based if you're fixated on a number (you shouldn't be)

Should I use Nero AAC though, no quality lost converting the lossless Wav to this? And are you saying I should be on Variable Bitrate instead of the other 2? What exactly does that bar mean? It's a Q slider that goes from 0 to 1, I'm guessing the Q stands for quality but I don't know which way to go for lossless if that's possible, it's all decimals until 1. And then there's also an option for AAC profile set to Automatic, but I'm guessing that's okay as it is.

If you get a chance to look at that file I posted earlier let me know what you think of it and if you would change/add anything. I think it looks great from my eye test, colors actually pop out.

[poisondeathray]

Those PP workarounds aren't really "weird" . Rewrapping is just swapping containers. Think of it as taking the audio & video out of a "box" and putting it into another "box". But for some reason just renaming the extension works

I'll have to uninstall dbfs and reinstall the other one again to try to do what you're saying, but I will to see if it works. But addressing what you say later on in the post, will there be audio quality lost if I frameserve the audio out too? Would it be better to export the audio losslessly as a Wav/Flac and mux that with the frameserved video? Or are they the same? (if all goes well and there's no audio bug at the end of the files again)

Frameserving doesn't work for me in newer PP versions. It "works" but too many problems with glitches. But it works for some people. So do some tests if you're planning on using it

For audio - it's decoded and decompressed to uncompressed WAV in vegas, that's what you're frameserving. Even if you use the same bitrate , or even a higher bitrate than the original - there will be some quality loss because you're re-compressing it with lossy compression.

Oh no, I'm not talking about frameserving out audio out of Vegas, we already know that gives me the audio loop glitch at the end of files that's why I'm not f-serving the audio anymore. I'm talking about exporting the audio out of Vegas as a standalone file before I start the video frameserving, I was wondering which of the Wav options in File > Render As to select, but I think that the one simply labeled default is correct and lossless, I hope. Or if I should just go with .flac out of Vegas and simply select the .flac encoder in MeGUI and therefore it wouldn't re-encode it but simply mux it?

I said it's decoded and decompressed. From there you either go lossy or lossless. Lossy, uncompressed 16bit stereo wav is going to be 1536kb/s. So many times larger than the original because it's decompressed from the original compressed audio. But not compatible in all containers (for example it's difficult to mux into MP4 with open source muxers). Flac is lossless and compressed. It will give you something smaller but less compatible with things like devices and some targets, not compatible in some containers too. Or something lossy and common. Such as AAC, MP3 etc..those are more compatible in a variety of containers. You'll never get back the original quality with the same filesize, because it has been decoded (decompressed).

Should I use Nero AAC though, no quality lost converting the lossless Wav to this? And are you saying I should be on Variable Bitrate instead of the other 2? What exactly does that bar mean? It's a Q slider that goes from 0 to 1, I'm guessing the Q stands for quality but I don't know which way to go for lossless if that's possible, it's all decimals until 1. And then there's also an option for AAC profile set to Automatic, but I'm guessing that's okay as it is.

Lossless isn't possible with AAC. AAC is a lossy format. But at high bitrates, you won't be able to hear the difference.

If you get a chance to look at that file I posted earlier let me know what you think of it and if you would change/add anything. I think it looks great from my eye test, colors actually pop out.

I'll have a look later today

[_Al_]

Also the Wav file's bit rate is 1536 kbps which isn't like the original files' either, I thought that was supposed to be a direct copy

For some reason I thought your avi capture that you loaded into Vegas had uncompressed audio hence that wav export would be basically export as "copy".

Vegas cannot export direct copy, same codec, original audio codec, not sure what your's is, if your audio was already encoded into "something".

Sometimes , if using frame server in Vegas, and there is problem with audio, while exporting signpost.avi there is a checkbox "Write audio as PCM samples in signpost.avi". That is self-explanatory. What it does, it render's uncompressed audio into that signpost.avi first. If video is long it appears as if frameserver froze for a time, because it is creating that signpost.avi. That file gets pretty larger then, because for every second of video it ads 1536kilo bits of uncompressed audio into that file. The longer signpost.avi, the larger it gets. doing so, you might get rid of a audio problem, not sure if that glitch of yours, but I was lucky in couple of cases, but not remembering exactly what was the reason, I think I was importing that signpost avi into Ripbot264 encoder, it was years back. With that box checked I had no problem then. Or that one bug with "silent" audio might be solved. But not sure because I use Vagas's version where simply everything works.

[acheter]

It's a bug , even in the CC 2015 version (from your screenshots it looks like you're using CS6 ?

[CZbwoi]

I said it's decoded and decompressed. From there you either go lossy or lossless. Lossy, uncompressed 16bit stereo wav is going to be 1536kb/s. So many times larger than the original because it's decompressed from the original compressed audio. But not compatible in all containers (for example it's difficult to mux into MP4 with open source muxers). Flac is lossless and compressed. It will give you something smaller but less compatible with things like devices and some targets, not compatible in some containers too. Or something lossy and common. Such as AAC, MP3 etc..those are more compatible in a variety of containers. You'll never get back the original quality with the same filesize, because it has been decoded (decompressed).

Hm. So what's your choice for audio encoding when you want the best possible "near-lossless" sound while still being in a compatible container? For your projects, family tapes, etc.

I'll have a look later today

K, looking forward to it.

PDR: I tried to do that PP workaround for switching the containers and extensions to fix the problem but I don't seem to know what you mean. It's not something you do in the export menu before frameserving, right? Do you mean to change the filetypes of all of my source files to .m2ts, outside of PP?

edit: answered my own question, only now I can't even export to frameserve anymore since I did it once and closed it and am now running into this bug: https://groups.google.com/forum/#!topic/frameserver-discuss/7M8q3NVUkw8

Yes, I've closed and opened it multiple times too. Incredible, Adobe. Wtf.


_AL_ I don't know what the deal is with frameserving these files out of Vegas 13 but if I "write audio as PCM samples" with it when I encode it gives me that audio glitch bug in the last second of the video. With both MeGUI and WinFF (basically ffmpeg), so the solution is to just do the audio separately I guess instead of frameserving it, but I don't know which audio encoder to select and with what settings for the best possible or exact quality (while being compatible with most players).

It's a bug , even in the CC 2015 version (from your screenshots it looks like you're using CS6 ?

CC7, so 2013 I think. Are you referring to the chroma problem?

[poisondeathray]

The color looks more saturated, it looks better - but it's really a subjective thing. There is no "right or wrong" , except if you clip something or really screw something up. I might have done it a bit differently , but that's just me. I already explained earlier how I would have done it , along with a bit of stabilization. And for the outdoor shots I would have lowered the black level a bit (it looks a bit washed out)

I said it's decoded and decompressed. From there you either go lossy or lossless. Lossy, uncompressed 16bit stereo wav is going to be 1536kb/s. So many times larger than the original because it's decompressed from the original compressed audio. But not compatible in all containers (for example it's difficult to mux into MP4 with open source muxers). Flac is lossless and compressed. It will give you something smaller but less compatible with things like devices and some targets, not compatible in some containers too. Or something lossy and common. Such as AAC, MP3 etc..those are more compatible in a variety of containers. You'll never get back the original quality with the same filesize, because it has been decoded (decompressed).

Hm. So what's your choice for audio encoding when you want the best possible "near-lossless" sound while still being in a compatible container? For your projects, family tapes, etc.

It depends what the target is going to be. For portable devices, generic use age - MP4 container, AAC audio. For something like optical disc, BD, AC3 audio. The higher the bitrate, the less loss . You're not going to be able to tell the difference for this type of audio source at 384kbps stereo AC3, or 160-220kbps stereo AAC . You're going to be archiving the original footage - so it shouldn't matter what you do.

PDR: I tried to do that PP workaround for switching the containers and extensions to fix the problem but I don't seem to know what you mean. It's not something you do in the export menu before frameserving, right? Do you mean to change the filetypes of all of my source files to .m2ts, outside of PP?

edit: answered my own question, only now I can't even export to frameserve anymore since I did it once and closed it and am now running into this bug: https://groups.google.com/forum/#!topic/frameserver-discuss/7M8q3NVUkw8

Yes, I've closed and opened it multiple times too. Incredible, Adobe. Wtf.

Yes, I mean outside of PP. Even renaming it works

eg.
file.mpg becomes file.m2ts

And for the frameserver issue, I have the run once problem too ( among other problems.) But for some people everything just works, apparently . For some people advanced fs works better than dmfs. Typically I use a lossless intermediate, especially if I have multiple exports/destinations or multipass encoding. It ends up being much faster, definitely more stable

[CZbwoi]

The color looks more saturated, it looks better - but it's really a subjective thing. There is no "right or wrong" , except if you clip something or really screw something up. I might have done it a bit differently , but that's just me. I already explained earlier how I would have done it , along with a bit of stabilization. And for the outdoor shots I would have lowered the black level a bit (it looks a bit washed out)

Thanks. Yeah, I'm not sure if I'm going to stabilize now that I think about it since it would take away from the "home movie" feel and get rid of picture or screen data by zooming in. But at the same time it would be a good idea to do some so that it's visually pleasing and easier to watch...agh. You can't have one with the other, this is a tough choice.

It depends what the target is going to be. For portable devices, generic use age - MP4 container, AAC audio. For something like optical disc, BD, AC3 audio. The higher the bitrate, the less loss . You're not going to be able to tell the difference for this type of audio source at 384kbps stereo AC3, or 160-220kbps stereo AAC . You're going to be archiving the original footage - so it shouldn't matter what you do.

These videos will go on a like, permanent family hard drive/cloud drive for years to come and these will be the only way people will see these videos and tapes, they won't be seeing archival footage to hear the best sound, so I'd prefer to have the best audio possibly or a direct clone source of it for that reason. Like these will be the FINAL files and what everyone will see til they're old, so I'd want to have the best audio quality as well, not the best portable quality or DVD quality, just like exactly the same sound.

The default is FFmpeg AC-3, but whatever I do to it, there's no sound in WMP - and in VLC it mutes for a half second before playing if you skip through it. Then there's LAME MP3, Flac, OPUS, Vorbis. I downloaded "Nero AAC" from somewhere after someone mentioned it, but I don't know which is best something I didn't mention?

Going into them individually there's a "Preferred Decoder" with options like NicAudio, FFAudioSource, DirectShow, etc. Which one do I select there? I've googled things like NicAudio vs. FFAudioSource to no avail.

I just want the highest quality and most compatible (with all players, systems) audio you can get possible, not something good enough for phones or whatever. I'm inputting that lossless decompressed Wav file into it...but when it comes to outputting it I don't know what's what.

And for the frameserver issue, I have the run once problem too ( among other problems.) But for some people everything just works, apparently . For some people advanced fs works better than dmfs. Typically I use a lossless intermediate, especially if I have multiple exports/destinations or multipass encoding. It ends up being much faster, definitely more stable

You've mentioned this a few times but have never explained how to do it, I went back through the thread and looked. How do I do this with PP and Vegas? Googling it takes me to a foreign website lol. What do I download and how do I attempt this? I have UT Video and MagicYUV codecs installed already if I would need those.

[_Al_]

I just want the highest quality and most compatible (with all players, systems) audio you can get possible

We all do, but always make audio that is widely accepted in particular container so to speak.

if you have MP4 audio should be AAC. Bitrate? 384kbps is a value that is most likely too much and certainly enough. It is up to you. You have 2 channels, stereo. 192kbps per channel is considered good enough, look it up. You have only two channels, you might see audios with 640kbps but 5.1

if you have MKV container, perhaps AC3. Bitrate same thing. Flac? go ahead, but most players would return a silence to you while playing that video back. You'd have to be aware of that limitation. But I can tell you someone might be surpriced after some time, hearing nothing and shaking his head.

if DVD, then you might choose AC3 (max 640kbps) or PCM (~1500kbps). That AC3 bitrate is maximum allowed, but again for stereo and having 192kbps per channel is enough.

Those codecs in particular container guarantee play-ability, if you deviate you might be limited in a way. Like you decide to go lossless and putting Flac into MKV etc.
I'd stick with specifications, expectations.

Do not look for some extraordinary good codec, with compression the way it works, if you have enough bitrate, it is alright anyway. That goes for video and audio as well. So that is your case. You give it enough bitrate, that is 384kbps for sure, same frequency as original the better, so use whatever there is in Megui, audio compressor or whatever you use.