StaxRip x86 (development for x86 has been stopped)

[kikker]

I've just DL'ed the latest version. Now, for whichever source video I try to convert, after the initial demuxing, I get the following error:
"Failed to load source, please ensure you are using a proper source filter and have installed plugins and codecs required by the source filter."

Then, when push OK, the preview pops up and in big red letters, it reads:

"Directshowsource: I can't determine the frame rate of the video, you must use the "fps" parameter (Movie_Source.avs, line 1)

That Movie_source.avs file plays just fine in VirtualDubMod, so I'm not sure which step is failing, but I have a feeling it has to do with Avisynth somewhere. I've tried uninstalling and reinstalling Avisynth, and I still get the same errors.

This is so strange, when 1.0.0.0 worked just fine, but now, even reverting back to 1.0.0.0 results in the same thing. All source video files work just fine in any media player, so all codecs and source filters seem to be working.

I'm sure there is a really simple fix for this, but not sure what else I can try.

Any ideas??


EDIT: OK, I simply uninstalled and re-installed the latest stable beta3 ffdshow, and now things seem to be normal again. Strange how ffdshow worked for playing back files for viewing.

[meltingplastic]

Hey guys,

Im trying to use stax to batch encode a bunch of MPEGs to H.264 MP4s. I created my template file to convert the file and resize it to 480x360. When i go to run a batch encode(either directory or multiple file) the output is never 480x360. it is always much larger. What am i doing wrong or how do i correctly create a template file?

[LRN]

Are you sure your player does not resizes your movies on playback?

[meltingplastic]

Are you sure your player does not resizes your movies on playback?

Yes, when it runs x264 i check the video size and its not the correct size. I noticed it after i did a batch of about 25 vids and thought maybe the zoom was on, started another batch and bam there it was.

As of now to do batch i just do one file, set it the way i want and then repeat the process til ive created all the .rip files for each one... this gets to be very time consuming and i know that theres an easier way!

[kikker]

The answer to this in in this thread. Someone else asked about something similar - I'll try to find it.

[meltingplastic]

The answer to this in in this thread. Someone else asked about something similar - I'll try to find it.

Thanx kikker, i searched through about 25 or so pages and figured it was prolly not covered or quicker to ask...

[kikker]

Try here? http://forum.doom9.org/showthread.php?t=124575

[meltingplastic]

tried that as it instructed... output video from x264 is 576x448... i try doing it with just one file as it instructed and as soon as i select my source file, stax extracts the audio then changes all my settings... it selects crop, cahnges my source, changes the resize to neutral, and also changes the resolution

[jsquare]

tried that as it instructed... output video from x264 is 576x448... i try doing it with just one file as it instructed and as soon as i select my source file, stax extracts the audio then changes all my settings... it selects crop, changes my source, changes the re-size to neutral, and also changes the resolution
Try this: go to Options(F8)-Automation-Auto Resize Image Size = 172800 (480x360), save that setting with your template and next time you do a batch all the encodes will be limited to 480x360, unless the source AR is different than 4:3.

[Lulle]

Which deinterlance filter is good in staxrip?

When I'm deinterlance with "deinterlance" it's bad.
When I deinterlance with "fast deinterlance" I think its bad too.

So can I add some new filter who is better? or make some change with the filter hos allready is in staxrip?

If yes which filter? And how do I do that?

Edit:
And which is the ful script? I can just se this in profiles: Deinterlace = FieldDeinterlace() .

[LRN]

Good filter is IVTC. Great filter is manually tweaked IVTC. Good filter is manually tweaked Deinterlace.

Yes, you can add new (better) filter. TomsMoComp, for example. Yes, you could (and should) change pre-set filters (tweak them).

IVCT and Deinterlace would be the bes...err-r-r... good to start with.

Both "Edit" and "Profiles" shows full script. FieldDeinterlace() is a function call. It calls "FieldDeinterlace" function (which deinterlaces). It looks simple, because no parameters is specified (it uses default parameter values). If you want to tweak these - learn AviSynth ( http://AviSynth.org ). We can give you hints (a lot) and some specific information, but we can't teach you, that is what guides and tutorials are for.

[monks1975]

Hi all,

Great tool Staxrip, making some lovely h264 encodes; just tiny thing-

Is there any way to stop Vista falling back to Aero Basic when importing VOB files (it's in the DGIndex part, so may be more of an issue with that tool)?

After DGIndex finishes, my desktop goes back to Aero Glass again.

--
Jon

[Deinorius]

If you want to know the correct Deinterlacer you should ask in this forum specifically which means for any film type (real, Anime,...) you need another Filter.

Good Filter are TDeint, leakkerneldeint, IVTC, Decomb,...

[contactadam2002]

How do you add avisynth filters to staxrip, I tried dropping the .dll's into the filters folder of avisynth, but I am unsure as to how you get stax rip to recognize it using the script editor. When you download the filters there is no info to tell you how to write the script????

[LRN]

Right click on filters field, Add, add any filter. Mind category: it should be 'misc' for auxiliary script lines and...let's suppose 'field' for video processing. This is because StaxRip drops some lines from filter script when performing some special tasks (like showing video in crop dialog). And because of that StaxRip's AviSynth editor is WICKED :)
Right click on filter field, edit. Find filter you just added. Click on field in 'script' column. Type script here.
For simple things script looks like <function name>(<argument name>=<argument value>,...)
For function names and their arguments read manual/readme/faq/tutorial for particular plugin.
If plugin (dll) is in 'plugins' directory then you do not need to explicitly load it. If it is not, then you have to call LoadPlugin(<path to plugin>), before you could call functions of this plugin.
By the way, symbols '<','>' and '...' is for syntax explanation, they should not be in the script (this may be obvious for someone, but you never know...).
If you need to write a few lines of script, you could 'add' more 'filters' (and thus 'scripts') or write them in one 'script', separating them by spaces (AviSynth permits both line breaks and spaces as separators; line breaks is used more often because they make code more readable, but in StaxRip's wicked script editor you can not use line breaks).
For further information about AviSynth read http://avisynth.org . I explained most (if not all) StaxRip-specific AviSynth things, anything else here would be off topic.

[dvd_maniac]

Seems that about half of my encodes have the last minute or two of audio missing???
I just reinstalled XP and have the latest Staxrip version installed.
Any ideas as to why this could be happening?

[LRN]

First, check audio files produced by encoder. If you have some temporary files (from cutting etc) check them too. If they are OK, then something is probably wrong with muxer.

[dvd_maniac]

The temp audio file is missing the ending as well.
What would cause this to happen so frequently?

[LRN]

You could try to repeat each step (follow log files) manually and see what happens after each stage.

[LRN]

I got the same issue and found that besweet produces trimmed temporary wav file. Maybe it is because it's size is larger than 4Gb...Actually, file itself is not trimmed, but wav header is.
To overcome this, use -ignorelength commandline switch for neroaacenc

2-pass AAC encoding will look like this:

"%application:BeSweet%" -core( -input "%input%" -output "%output%_tmpwav.wav" -2ch -logfile "%output%_tmpwav.wav.log" ) -azid( -c normal -L -3db ) -ota( -d %delay% -g max )
"%application:NeroAACEnc%" -2pass -br 98304 -ignorelength -if "%output%_tmpwav.wav" -of "%output%"
cmd /C del "%output%_tmpwav.wav"

and this

"%application:BeSweet%" -core( -input "%input%" -output "%output%_tmpwav.wav" -6chwav -logfile "%output%_tmpwav.wav.log" ) -azid( -c normal -L -3db ) -ota( -d %delay% -g max )
"%application:NeroAACEnc%" -2pass -br 229376 -ignorelength -if "%output%_tmpwav.wav" -of "%output%"
cmd /C del "%output%_tmpwav.wav"

[dalihrp]

HI,
I try to backup my transformers DVD to h264, using staxrip default x264 profile constant quality and staxrip default 6 chnl AAC. Container is MKV.

Then Staxrip report this:

---------- Audio Encoding Statistics ----------

Start time: 1:55:09 PM
End time: 3:03:16 PM
Duration: 01:08:07
Speed: 50 fps
Size: 287.3 MB
Bitrate: 280.03 kbps
Video Bitrate: 453 -> 398

My question is about the video bitrate that only 398, is that mean staxrip will encode my DVD using that bitrate? isn't that to low?

thanx.

btw>the encoding not yet finished , so I can't tell the quality.

[LRN]

AAC encoding is 1-pass VBR by default. I.e. neroAACenc adapts quality on the fly, based on given threshold (q=0.3, if i remember correctly), thus result audio size is unpredictable. Number you could see in audio profile is just a guess (good guess), and from this log i see that it was around 240 kbps. But encoded audio track size happened to be larger than expected, and StaxRip adjusted video bitrate accordingly.

That is why i am using 2-pass AAC encoding, despite it's relatively low speed, complexity and other issues - it produces exact file size, meaning that StaxRip will use the same video bitrate that i gave, without changes.

By the way, 'constant quality' x264 is 1-pass too, and will give you the same unoptimal quality for video as AAC encoder did for audio.

And the term 'backup' is very innacurate. H.264 with 453 kbps is not enough to backup any DVD without loosing a LOT of quality (either in details or in video resolution), regardless of encoding method (1-pass or 2-pass).

[dalihrp]

How come the picked bit rate is so low by stackrip?

But I've read somewhere that constant quality with x264 is the way to go if you don't care about file size, and it's faster than 2 pass.

Now if i want bitrate around 1000-1500 kbps and single CQ pass , how do i get it with staxrip? Staxrip default for cq=22, do i need to change it to 18 or something?

:thanks:

[burfadel]

How come the picked bit rate is so low by stackrip?

But I've read somewhere that constant quality with x264 is the way to go if you don't care about file size, and it's faster than 2 pass.

Now if i want bitrate around 1000-1500 kbps and single CQ pass , how do i get it with staxrip? Staxrip default for cq=22, do i need to change it to 18 or something?

:thanks:

Thats almost impossible to know without doing a first pass test! If you want 1000-1500, just use 2-pass! Otherwise choose a CRF and see if it meets your requirements. Depending on the source, a crf of 22 could be 800kbps or 2000kbps, just the way it works

[LRN]

Actually, it IS possible to adjust bitrate variation (deviation) in constant quality mode, but i never tried to do that.

How come the picked bit rate is so low by stackrip?
Oh. Good question. I just realized that in constant quality mode StaxRip shouldn't change video bitrate. In fact, there is no pre-calculated video bitrate in this mode at all...Or it is? I don't really know.

constant quality with x264 is the way to go if you don't care about file size, and it's faster than 2 pass.
Yes, 1 pass is always faster than 2, though you can use fast first pass.
Constant quality is not only unprecise (unpredictable file size), but also produces lower quality (on-the-fly bitrate adjustment can not be optimal and leads to quality loss, because codec uses too high bitrate on some simple frames and too low bitrate on some complex frames).
Quality always means 2-pass encoding. I'm not trying to prove that 1-pass is the worst of evils in the world or something. It's just that you said 'backup', and for me it means high quality.

[kobe4rings]

Hello,

I cap basketball games so and ive been using staxrip for 3 months. A regular basketball game audio is MP3128CBR on an XviD.. am doing my cap Mp4 x264 intermediate quality , so i was wondering what AAC audio profile will give me the same quality as the MP3128CBR? is the AACHE VBR 50-70 too low and is the AAC LC VBR110-150kbps too high? whats the difference between AAC LC and AAC HE and is there anyway to customize the bitrate on either profile because i try to but the end result is the same as the default audio profile. Any Help would be appreciated.

[LRN]

all default AAC profiles are 1-pass AAC. 1-pass AAC = constant quality AAC. You cannot specify bitrate, you can only change 'q' value. Default q = 0.3. The 'bitrate' field in profile is only for bitrate precalculation, it does not affects AAC bitrate at all.

[kobe4rings]

all default AAC profiles are 1-pass AAC. 1-pass AAC = constant quality AAC. You cannot specify bitrate, you can only change 'q' value. Default q = 0.3. The 'bitrate' field in profile is only for bitrate precalculation, it does not affects AAC bitrate at all.

ohh i see.. changing the quality level did the trick, thanks :)

[otherdarom]

Dear Stax,
is it poss. to adjust staxrip so that it logs all the working steps to the log file before beginning of the work - I mean Staxrip logs all command lines to run first to the log file and starts then the working. And my second question ist if it's poss. to reuse the stats file of the project. For example after a system crash during 2. pass -> just look if a stats file is there and ask the user (avidemux does it this way) to reuse this file.

[diskborste]

I use the standard MP3 VBR 110-150 kbps option in Staxrip when i encode my recorded tv-shows from mpeg2 to xvid with mp3. I want the audio to be mp3 around 128 kbps average but when i use Staxrip my .avi-files the mp3 only has an average bitrate of 80-90.

What should i do, i want the avg bitrate to be around 128 kbps instead of 90? :O

[LRN]

Click on audio dropdown menu button.
Select MP3->MP3 VBR 110-150 kbps profile.
Click on audio dropdown menu button.
Select Edit
Observe.
Notice -lame( -v --vbr-new -V 5 -b 32 )
This is MP3 encoding parameters.
Refer to Besweet Commandline Reference (http://forum.doom9.org/showthread.php?s=&threadid=91928) for explanation of these parameters.
Tweak parameters accordingly.

[diskborste]

Click on audio dropdown menu button.
Select MP3->MP3 VBR 110-150 kbps profile.
Click on audio dropdown menu button.
Select Edit
Observe.
Notice -lame( -v --vbr-new -V 5 -b 32 )
This is MP3 encoding parameters.
Refer to Besweet Commandline Reference (http://forum.doom9.org/showthread.php?s=&threadid=91928) for explanation of these parameters.
Tweak parameters accordingly.
Yeah, that's how it is at the moment.

What should i change for a highter avg. bitrate?

[swaaye]

I've found that soundtracks usually end up on the low end of the VBR range because they simply don't have that much material playing most of the time. There's a lot of simple sound and silence.

Here is a reference for LAME settings. You see that -V 5 gives you a typical average of 130 kbps, but not with soundtracks.
http://www.hydrogenaudio.org/forums/index.php?showtopic=28124

[duckdown]

Amazing program -- couldn't be happier with it -- just one very important question. Since this is just a frontend to VDUBMOD , How do I access the outputting options? I am trying to get my encodes to output in SEGMENTS/CHUNKS like you would find in the "Save" menu of vdubmod.

I'm sure this option is there I just can't find it

Once again, awesome work, and thanks in advance!

[LRN]

As far as i know, StaxRip only uses VDubMOD for AC3 audio cutting. I.e. StaxRip is NOT a frontend for VDubMOD.

[duckdown]

As far as i know, StaxRip only uses VDubMOD for AC3 audio cutting. I.e. StaxRip is NOT a frontend for VDubMOD.

What does it do all of the xvid encoding in?

And what about the first / second pass options? They're the same ones from vdubmod

[LRN]

H-m-m-m. I looked up at XviD encoding, and it is really goes via VDubMOD. Still
1) StaxRip could use VDub instead of VDubMode
2) This is because XviD encoder is in vfw form (first/second pass dialogs you mentioned - that's vfw) and to deal with it StaxRip needs some AviSynth<->vfw glueing software. Which is VDub(MOD). For commandline encoders with avisynth support (such as x264) VDub is not used.
Anyway, there's no commandline options for using VDub (or i can't find them). You have to hack the source and recompile, or add such options to GUI. This is a good feature request though - lack of commandline support really narrows your options. Someone mentioned 'reusing stat-file from 1st pass' - this is only possible at the moment with commandline encoders (you just copy 2nd pass commandline and launch it manually, then do the same for muxing commandline).

It is possible to encode in xvid via ffmpeg (it has AviSynth support). This would fit better for StaxRip, since it does not uses all VDub filters and options anyway...

[duckdown]

H-m-m-m. I looked up at XviD encoding, and it is really goes via VDubMOD. Still
1) StaxRip could use VDub instead of VDubMode
2) This is because XviD encoder is in vfw form (first/second pass dialogs you mentioned - that's vfw) and to deal with it StaxRip needs some AviSynth<->vfw glueing software. Which is VDub(MOD). For commandline encoders with avisynth support (such as x264) VDub is not used.
Anyway, there's no commandline options for using VDub (or i can't find them). You have to hack the source and recompile, or add such options to GUI. This is a good feature request though - lack of commandline support really narrows your options. Someone mentioned 'reusing stat-file from 1st pass' - this is only possible at the moment with commandline encoders (you just copy 2nd pass commandline and launch it manually, then do the same for muxing commandline).

It is possible to encode in xvid via ffmpeg (it has AviSynth support). This would fit better for StaxRip, since it does not uses all VDub filters and options anyway...

Hi thanks :) It does use vdubmod; i've watched the encoding process carefully

I do appreciate the info but i am convinced that these options are mostly accessable , especially because i know that stax is just an interface to vdb when using it for this form of encoding

hopefully the author can let me know!!! THANKS alot , just trying to access the file-save options so I can save to CHUNKS just like is integrated into vdubmod menu

Thanks fopr all the help!!

[LRN]

Public Overrides Sub Encode()
Dim startTime As DateTime
Dim script As New VirtualDubScript

If Passes > 1 Then
Log.WriteHeader("First Pass Codec Parameters")
SetReg(RegFirstPass)
script.Open(p.TargetAVSDoc.Path)
script.VideoSetModeFastRecompress()
script.VideoSetCompression(VDMSetCompression, p.VideoBitrate / 8)
script.Save(Paths.GetTempDir + p.Name + "_FirstPass.avi")

Log.WriteHeader("First Pass Encoding")
Log.AddLines(script.Text)

script.SaveScript(Paths.GetTempDir + p.Name + "_FirstPass.vcf")
startTime = DateTime.Now

Dim pw As New ProcessWrapper
ProcessForm.ProcessWrapper = pw
p.TargetAVSDoc.Synchronize()

pw.File = GetVirtualDubModPath()
pw.Arguments = "/s""" + Paths.GetTempDir + p.Name + "_FirstPass.vcf"" /x"
pw.Minimized = s.RunAppsMinimized.Value
pw.Wait = True
pw.Priority = s.ProcessPriority.Value
pw.Start()

Point is, VDub script generation is hard-coded and not tweakable. VDub launch commandline is hard-coded and not tweakable.

[mutha88]

Hello ! I have a simple question:)

How can i make the target quallity better that 50% in the x264 configuration?

Thank you all of you, sorry if mistaken something :confused:

[LRN]

Codec configuration->StaxRip->Quality
Of course you understand that this parameter is only for compressibility check, do you?

[mutha88]

Codec configuration->StaxRip->Quality
Of course you understand that this parameter is only for compressibility check, do you?

Oh, i didn't knew that... i have another simple question :)

Which profile is best for this kind of videos?:



Video: MPEG2 Video 720x480 (4:3) 29.97fps 9800Kbps [Video]
Audio: Dolby AC3 48000Hz stereo 256Kbps [Audio]


and:

Video: MPEG2 Video 720x480 (4:3) 29.97fps 7800Kbps [Video]
Audio: PCM 48000Hz stereo 1536Kbps [Audio]

I am using the HQ - Slowest profile, but is this nessecary for a better compresion? I am currently handling music videos (single vob files) :rolleyes:

[diskborste]

I've found that soundtracks usually end up on the low end of the VBR range because they simply don't have that much material playing most of the time. There's a lot of simple sound and silence.

Here is a reference for LAME settings. You see that -V 5 gives you a typical average of 130 kbps, but not with soundtracks.
http://www.hydrogenaudio.org/forums/index.php?showtopic=28124
I'm just trying to encode a regular tv-show. This happens with every show that I encode. When i check the bitrate with Gspot it's always around 90-100 avg, i want it to be around 128. I've never had this problem with Gordian Knot.

Can someone please tell me what to change in Staxrip? :thanks:

[LRN]

HQ - Slowest is always a little better than HQ - Slow. If you got time - use it. The only thing that really need tweaking in codec settings - is matrix. There's a lot of custom matrices, potimized for different content (for example - i saw once special matrix for anime), applying such matrix will improve quality (i think). However, i did not ever tried this, so i can't really tell you anything more. For such advanced topics you should ask in other threads (maybe, x264 main thread at this forum) and look for x264 guides.

Can someone please tell me what to change in Staxrip?
Notice -lame( -v --vbr-new -V 5 -b 32 )
This is MP3 encoding parameters.
Refer to Besweet Commandline Reference (http://forum.doom9.org/showthread.php?s=&threadid=91928) for explanation of these parameters.
Tweak parameters accordingly.
Here is a reference for LAME settings. You see that -V 5 gives you a typical average of 130 kbps, but not with soundtracks.
http://www.hydrogenaudio.org/forums/index.php?showtopic=28124
If LAME thinks 90-100 kbps is enough for such audio at quality=5, then it is so. You could try to raise quality to offset such bitrate lowering.
Your other option is to use 2-pass encoding (always produces exact file size). However, there is no 2-pass mp3 encoder (in 2002 someone asked for that on this forum). You could switch to ogg vorbis though...

[laserfan]

Using 1.0.0.1 and an HD Xvid encoder setup; I want to convert 1080i to 720p but when I select 1280x720 and save as Default Template.rip, I never get 1280x720--staxrip always goes (IIRC) 672xsomething.

As a result I have to open every file individually and select the 1280x720--I can't batch multiple conversions, I have to run DGindex, select the dimension, and save the Job and go on to the next file. What am I doing wrong??? :confused:

[dalihrp]

Hi,

I change the default StaxRip AAC 5.1 profile from q=0.3 to q=0.25.
and the resulting MP4 is about half of the original AC3, that is good.

Then I've found the MPC report the MP4 (with q=0.25) as AAC 24KHZ stereo not as AAC 6 cnhl 48KHZ that I usually get with q=0.3.

Do I loose the 6 cnhl with q=0.25?

Unfortunately my speaker system is 2.1 only, so can't test it.

:thanks:

[swaaye]

I'm just trying to encode a regular tv-show. This happens with every show that I encode. When i check the bitrate with Gspot it's always around 90-100 avg, i want it to be around 128. I've never had this problem with Gordian Knot.

Can someone please tell me what to change in Staxrip? :thanks:

LAME is determining that it doesn't need the higher bitrate for the content of the soundtrack. This is what VBR does. It tries to make intelligent use of the range and quality setting you give it. The average bitrate that they say you will get is a guesstimate based off of music files, probably.

If you really want to make it use more bitrate, set it to -V4 or -V3. Or set it up for ABR or CBR instead of pure VBR. Personally I'd just leave it on VBR (it's best) and accept that TV/movie soundtracks don't need as much bitrate as music.

[LRN]

Using 1.0.0.1 and an HD Xvid encoder setup; I want to convert 1080i to 720p but when I select 1280x720 and save as Default Template.rip, I never get 1280x720--staxrip always goes (IIRC) 672xsomething.
Check StaxRip preferences (somewhere in main menu). There's max size for resize filter, increase it.

dalihrp, first, try foobar2000 for audio information. It reports more accurate technical info than most players/tools. Check both AC3 and AAC info via foobar2000. Maybe MPC is wrong...
Second, post AAC encoding command line (from 'preview' tab). Maybe there's something wrong with it.
Third, you could try to encode with 0.29, 0.28, etc.
By the way, are you playing resulting (muxed) mp4, or audio-only mp4?

[laserfan]

I asked why my 1280x720 preset wasn't working:
Check StaxRip preferences (somewhere in main menu). There's max size for resize filter, increase it.
Thanks LRN, that was it! Actually the setting is under View\Options\Automation-Auto Resize Image Size. I set this to 921600 and saved my Template and I get 1280x720 now!

Appreciate the help; I might never have found this on my own! :thanks:

[avdw]

1st Q) I set the audio bitrate to 112 CBR, yet Staxrip still feeds "audio_rate: 128" to firstpass Xvid.
Also, it always feeds movie length of 1h30m00s to Xvid.

a) Is this normal ?
b) Did I miss something ?
c) Is this a bug ?
d) Whatever the program, it's head-to-head with avi.NET for the BEST encoder tool !!! :)

2nd Q) What's best for normal DVD movie to 700MB Xvid AVI conversion ? CBR or VBR audio ?
Some manuals/guides say CBR, others say VBR ?
I have to add, I want to play the encoded AVI's on a standalone Philips 630, which needs ESS compatibility.