AutoRV10/Oagmachine conversion

[Backflip]

Hiya, glad to be posting for the first time, hope things go well :)

Now, what I'm doing is attempting to create RV10 + 6CH AC3 *MKV* files and RV10 + 6CH AAC *MKV* files.

Source DVD is Face/Off, PAL @ 25 fps and is interlaced.

My main question is what *Field Filter in AutoRV10 should I be using so as to do the conversion from interlaced to progressive for best results? Is it correct to use a De-Interlace filter?

Currently I follow this guide (http://dark.pluridis.org/autorv10_guide/AutoRv10.htm):

1. Rip with latest DVD Decrypter, load into DGIndex, leave Field Operation as None, save out d2v file.

2. Load d2v file into AutoRV10, crop, choose Bicubic Precision, resize to 696 x 296 (just so AR error is 0%), tick Use Anamorphic Resize, *use DeComb De-Interlace filter, use Light FluxSmooth Denoise Filter. Pass Mode is Two Pass VBR & Video Mode is Normal Motion. E.H.Q Level is 100 - Insane. Also have credits encoded at 30%.

My target file size is around 1034MB video-only.

Also I thought I should ask this too, as my bitrate is around 1000 kbps for video would it be okay to use Lanczos Precision as the resize movie filter?

One last thing - AC3 to AAC conversion. I believe I've got everything correctly set up, but am wondering why I get an AAC file with a sampling rate of 44100 Hz. I would like to keep the 48000 Hz sampling rate. I've unticked Set Sampling Rate to etc. Is it possible to keep 48000 using Oagmachine/Nero or is it essential to be lowered to 44100 Hz?

This is a log of a small sample:
BeSweet v1.5b29 by DSPguru.
--------------------------
Using azid.dll v1.9 (b922) by Midas (midas@egon.gyaloglo.hu).
Using Shibatch.dll v0.24 by Naoki Shibata & DSPguru (shibatch.sourceforge.net).
Using bsn.dll v0.24 by DPeshev,Richard,E-Male,DSPguru (DSPguru.Doom9.org).

Logging start : 05/12/05 , 21:18:55.

C:\Program Files\OagMachine\BeSweet.exe -core( -input C:\FACE_OFF\VIDEO_TS\FO AC3 T01 3_2ch 384Kbps DELAY 0ms.ac3 -output C:\FACE_OFF\VIDEO_TS\FO AC3 T01 3_2ch 384Kbps DELAY 0ms.mp4 -logfilea C:\Program Files\OagMachine\BeSweet.log ) -azid( -g max -L -3db -c normal ) -ota( -d 0 ) -split( -start 0 -end 60 ) -bsn( -6chnew -config ) -profile( The OagMachine v0.11 )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : C:\FACE_OFF\VIDEO_TS\FO AC3 T01 3_2ch 384Kbps DELAY 0ms.ac3
[00:00:00:000] | Output: C:\FACE_OFF\VIDEO_TS\FO AC3 T01 3_2ch 384Kbps DELAY 0ms.mp4
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Input Channels Mode: 3/2, Bitrate: 384kbps
[00:00:00:000] | Total Gain: 13.045dB, Compression: Normal
[00:00:00:000] | LFE levels: To LR -3.0dB, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: BSI
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] +------ Shibatch -----
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Dest. Sample-Rate: 44.1KHz
[00:00:00:000] | Attenuation : 0.0db
[00:00:00:000] +-------- BSN --------
[00:00:00:000] | bitrate: vbr streaming
[00:00:00:000] | LC-aac high quality encoding
[00:00:00:000] +---------------------
[00:01:00:032] Conversion Completed !
[00:00:56:000] <-- Transcoding Duration

Logging ends : 05/12/05 , 21:19:51.

Gives a file like:

Video
Complete name : C:\FACE_OFF\VIDEO_TS\FO AC3 T01 3_2ch 384Kbps DELAY 0ms.mp4
File size : 2.85 MiB
Format : MPEG 4
Overal bitrate : 399 Kbps
PlayTime : 1mn

Audio #0
Codec : MPEG-4 AAC LC
Bit rate : 390 Kbps
Sampling rate : 44 KHz
PlayTime : 1mn

Thanks for any help :)

[Dark-Cracker]

hi,

>My main question is what *Field Filter in AutoRV10 should I be using so as to do the conversion from interlaced to progressive for best results?

all the deinterlacer give you best result (it only depend of the source and the desired output).

i suggest you to use decomb + bicubic resize.

if you want a more sharp result (it's first better to use a slightly highter bitrate because more detail mean more bitrate needed to encode them) use kerneldeint + lanczos.

hope this help.

++

[Backflip]

Thanks for the help :)

Now, regarding deinterlacer and resize filter, what target bitrate is good to use in conjunction with kerneldeint + lanczos in general. 1500 kbps? 2000 kbps?

Also, should I be using the FluxSmooth denoise filter on a modern DVD source? If used, does FluxSmooth have either a good effect, or no effect at all?

[EDIT] Just going back to AAC, it's strange but I think I'm getting well oversized 6CH files. Even at the Streaming (Medium), High quality LC setting I'm getting a file which is larger than the original 6CH AC3 file. I'm following this guide:

http://www.therealbmd.com/guides/cronus/

with every encoded AAC file, I'm not getting that much of a reduction in size, even at the Streaming (Medium setting).

[EDIT2] found the reason audio must be downsampled from 48 to 44.1, it's because Nero AAC encoder doesn't support 48 properly. Now just looking for the reason I get oversized AAC output :/

[Dark-Cracker]

a bitrate around 1200 - 1500 kbps should be good enought with the current resolution if you want to use lanczos.

fluxsmooth in light or medium preset made minor change so you can use without fear on a modern dvd :)

Ps : for your oversized nero AAC file you could always try to open a thred in the audio section.

++

[Backflip]

I was a bit afraid to open a thread in the Audio encoding sections, it might be too simple a question. I believe I found the answer though, just went through tests and found only settings:

Variable bit rate - Streaming :: Medium
Encoder Quality - High
AAC profile - HE (High Efficiency AAC)

gives me a smaller 6 channel AAC file than the original 6 channel AC3 file.

Any higher setting and I get a AAC file larger than the original 6 channel AC3 file. Even if I try using the LC (Low Complexity) setting at the same Streaming/High encoder quality level I get a larger AAC file than the AC3 file.

May be incorrect, but from the settings and such is it fair to say that it's not possible to use any LC setting in Nero AAC Encoder to get an 6 channel AAC file smaller than the original 6 channel AC3 file?

Also, I read somewhere that High Efficiency is recommended for stereo sound, while 5.1/6 channel encoding should only be done using Low Complexity.

In the AAC guide here (http://www.therealbmd.com/guides/cronus/) it's got (about half-way down the page):

The only setting you really want to worry about here is the AAC Profile, the rest should be pretty self-explanatory. Leave the Encoder Quality on HIGH!

LC and HE are two different profiles indicating how hard the encoder should compress the file. HE-AAC (High-Efficiency) isn't supported on digital music players such as the iPod, but LC-AAC (Low Complexity) is. HE is better used on less complex audio tracks, such as commentary, to squeeze out the most data per bit while maintaining reasonable listening quality. For a 5.1 track, this kind of compression isn't a good idea, so use LC Profile.

To get an idea of filesize, using the VBR "Normal" setting will yield about a 20% reduction in filesize. While encoding the Pulp Fiction AC3, it started as about 500 MB in size, but the AAC file, at VBR Normal would come out to be 400 MB. At the setting "Internet - Medium" I would get about 340 MB.

At VBR "Normal" setting I don't get a 20% reduction, i only get an increase in filesize -- it becomes larger than the original AC3 file, dunno why this is.

I might start another thread in the audio encoding section.

[Backflip]

Hi again :)

One quick question - I noticed in the guide (http://dark.pluridis.org/autorv10_guide/AutoRv10.htm) that Smart Crop All is not selected, which I can understand because if you did select it you'd just be cutting away more picture than is required. However, I've seen in this guide (http://www.everwicked.com/content/XviD_Guide/content-05.php) that Smart Crop All is something which is recommended. Is Smart Crop All necessary?

Also the Everwicked guide suggests that DVD2AVI should be set to '64-bit Floating Point' (http://www.everwicked.com/content/XviD_Guide/content-03.php) before making a project. Why is this? I just leave it at default usually.

[Dark-Cracker]

hi,

the smartcrop all option is not always needed especially since the avisynth crop need some modulo 4 values. it's just to keep aspect ratio error the most near 0%.
the fact to select 64bit floating point is not needed it theoricaly improve speed but it depend of your CPU optimisation.

++

[Backflip]

Thanks Dark-Cracker you've been a great help. Very awesome :)

edit - Cool! I finally found the solution to the AAC being resampled. I needed to disable Show Configuration Dialog in Oagmachine. This is something that wasn't covered in the guide I was following, but that's understandable because the guide is quite old.