StaxRip x86 (development for x86 has been stopped)

[Brother John]

Wouldn't reordering be just fine? I think mkvmerge sets the first stream to be default, you can check with the built in MediaInfo feature.
Not it all situations. Consider a multi-language MKV: English audio, German audio, English subs, German subs (streams in that order). No forced subs or anything, just plain old normal full subtitle tracks. Without manual intervention MKVMerge will flag both English audio and subs as default:yes. That means by default players will display the English subtitles.

To prevent the subtitles from showing up automatically (maybe because you understand English well enough and only include them as an optional feature) you need to manually flag the English subs as default:no.

[stax76]

@Lentzeris

Next release fixes some comp check crashes.

Jobs->ADD: Remember last used folder

Which folder?

File batch: Add folder with recursive seek.

Batch processing has low priority because it don't seem to be used by many. The dialog to add files in file batch mode has drag & drop support so you can search for files recursively with windows explorer and then drag them on the dialog.

Because it annoys when adding files to queve it remembers last opened single file folder but not remember the add files dialog last opened folder. Second one is to help adding more files in single step. If recursive folder adding is added, files on different folders are much easier to add.

I'm not sure if I understand it correctly, often I don't remember many details of features added long time ago and rarely used by myself. I'll take a look at it.

Video preview is crashing for most videos in latest beta when I use the next avisynth filter:

I've ass sample I can test, is there a stack trace in the log file or a message box where you can export a stack trace?

@Brother John

Good to see you around and thanks for the explanation, it's added and will be released within the next couple of days.

[davizator]

Hello...

About audio....How can I convert from 6ch AC3 to 6ch AAC? In this 1.2 version automatically convert 6ch AC3 to 2ch WAV and then to 2ch AAC.... but I can do the conversion directly copying the command line to a dos window and changing the wav file name to sourcefilename.ac3...

Thanks for that great program....

[Patman]

Hello community...

I've found a bug in audio conversion. For AC3 conversion i've set the encoder automatic and the bitrate to 448 kbit/s but when i started the process, the bitrate is set to 640 kbit/s. Is this a bug?

Please update the avisynth filters and all applications to the latest and project x to version 0.91.08. So java isn't required for Staxrip.

Nice to see that the development will continued...

Greets Pat

[stax76]

@Patman & davizator

I've overhauled audio processing by much hoping to eliminate some reported problems, it just needs a little bit more testing.

Please update the avisynth filters and all applications to the latest and project x to version 0.91.08. So java isn't required for Staxrip.

9 MB only for this app is way to much, I would like to include MPC 32-Bit for avs preview but it's another 10 MB.

[stax76]

@Lentzeris

I've tried my ASS sample with the code below not getting a crash.

TextSub((LeftStr("%source_file%", StrLen("%source_file%")-4))+ ".ass")

[stax76]

Here is a test build for everybody having a problem with the previous release or anybody wanting to help removing bugs from the next release coming in in 2-3 days.

https://www.dropbox.com/s/flf1s6x622yz1qp/StaxRip_1.2.0.2_2015.01.26.7z?dl=0

I've not taken care migrating settings like in the past so I recommend to delete the content of the settings directory with every build.

[cegy]

Here is a test build for everybody having a problem with the previous release or anybody wanting to help removing bugs from the next release coming in in 2-3 days.

https://www.dropbox.com/s/flf1s6x622yz1qp/StaxRip_1.2.0.2_2015.01.26.7z?dl=0

I've not taken care migrating settings like in the past so I recommend to delete the content of the settings directory with every build.

i've not really used it to encode with if i was to be honest but mainly to see if the changes was made which i asked about here (http://forum.doom9.org/showpost.php?p=1706183&postcount=4626)

however there is one minor thing i've notice which is still nice to be able to see at a quick look then having to blind guess the numbers..

http://i.imgur.com/b42joO5.png

the cropping colours (cropping part) seems much better on how its able to change based on the colour of your windows theme but i do wonder if the classic style will come back via a option (hardcoded maybe?):confused:

the ffmpeg encoding part seems like a nice idea but a suggestion would maybe add a edit box so the user can add there own options if that's possible?

keep up the good work stax76 :cool:

[stax76]

however there is one minor thing i've notice which is still nice to be able to see at a quick look then having to blind guess the numbers..

changed it too now

the cropping colours (cropping part) seems much better on how its able to change based on the colour of your windows theme but i do wonder if the classic style will come back via a option (hardcoded maybe?)

You mean Windows 98/MPC-HC style? If you enable this in Windows then StaxRip should use classic style otherwise it's a bug.

There is a option at Tools/Seetings/System/Menu Style, the menu drawing is from me so I hope it looks OK and works.

the ffmpeg encoding part seems like a nice idea but a suggestion would maybe add a edit box so the user can add there own options if that's possible?

I think I can add it, also for audio encoding.

[stax76]

@Alexander

I tried to answer your private message but for some reason the system don't allow to send it. You can also mail me or use the German doom9 forum, it has a StaxRip support thread (http://forum.gleitz.info/showthread.php?26177-StaxRip-Encoding-Frontend-%28Diskussion%29/page155) too.

[Jeroi]

Huge bug report:

Staxrip delete moves 4gb temp files to recylce bin which fullfills entirely soon C: . Please put force delete to your code to avoid moving to recycle bin. Also could you please add delete log files option also?

Also couple features for audio tracks needed:

- Remove second audio track option. At the moment it encodes automatically second audio track if source have mutliple audio tracks. Please add option to encode only single audio track to mp4 or mkv format.
- Please add more audotracks possibilites. I have few video containing 6 audio tracks and would need quite many tracks to the encoder.

[stax76]

My drive is only 1 GB (2.5" is more silent) and I never had problems with the recycle bin, I don't know what Windows uses as default, I defined it to use 50 GB, default or customized there should be a sane upper limit and Windows shouldn't exceed it, I would like to help but don't understand where the problem is. I might add another option but it's not simple because for such specialized issues I consider to build a settings dialog similar to about:config in Firefox.

[Patman]

Hi Stax ...

I've tested the new version with some audio formats. The result was good in this release. No matter what encoder I have chosen or what format and bit rate, the output file reflected the settings. The only problem was the conversion of some TrueHD audio tracks to AC3. For example, Dolby Atmos can only be converted with ffmpeg. If the selection of the encoder is set to automatic, then the conversion of some TrueHD audio tracks is displayed as an error, but the process goes on and is not interrupted. In TrueHD audio tracks ffmpeg should be preferred as the encoder when the setting is set to automatic.

Greets Pat

[stax76]

@Jeroi

I've added a setting at:

Tools/Settings/System/Use recycle bin when temp files are deleted

Log file will also be deleted by default (using recycle bin).

Remove second audio track option. At the moment it encodes automatically second audio track if source have mutliple audio tracks. Please add option to encode only single audio track to mp4 or mkv format.

There is a audio profile called 'No Audio' for this.

Please add more audotracks possibilites. I have few video containing 6 audio tracks and would need quite many tracks to the encoder.

It don't fit well with the program architecture and I'm unsure how hard it will be. I will try it but I don't know when.

@Patman

Thanks for testing, I got to look for a sample and info on Dolby Atmos then.

[davizator]

When encode again a movie (opening a existing project), now I cannot choose the video file in the "Just mux" option, the program get the source file but I need the output encoded file....

[stax76]

It checks for output, if non existing uses input, I would guess it's probably a incompatibility (encoded with a older build).

[davizator]

Are you sure it checks the output correctly? I am using your last version in post #4657 creating a new project and then open this project and changing some parameters to test different outputs (with x265).. and when Just mux in audio always merge the wav file but I choose the AAC output file...

[stax76]

I thought you mean video muxing, audio might be a bug, there were some audio bugs I fixed meanwhile that possibly could relate.

[stax76]

Here is a build from today: https://www.dropbox.com/s/r1v4fp1kb23r6cg/StaxRip_1.2.0.2_2015.01.27.7z?dl=0

I moved the audio and subtitle demuxing options from the project options dialog to Tools/Settings/Demuxing, it's now global and everything in one place.

Audio had some bugs like when you load a audio profile the stream selection in the context menu was gone, if there are demuxed files the menu now always shows them. The audio textbox never shows a file name but always media info. The audio play feature plays now the correct stream when demuxing is disabled (default), this works only with FFVideoSource because it uses FFAudioSource for stream selection. If you use a audio encoder that handles delay (eac3to or BeSweet) StaxRip removes the delay from the output filename, when you later then open this file for muxing it won't extract the delay from the filename and pass it to the muxer like in previous versions.

[Carpo]

Little crash error using the version from your post Stax, although this same .avs file opens fine in the 1.2.0.1 beta, in 1.2.0.2 it crashes instantly when loading the .avs file

.avs file

SetMTMode(5, 4)
LoadPlugin("D:\dgindex\DGDecode.dll")
DGDecode_mpeg2source("E:\Evanescence - My Immortal.d2v", info=3)
SetMTMode(2)
QTGMC( Preset="Slower", NoiseProcess=1, NoiseRestore=0.0, Denoiser="dfttest", DenoiseMC=true, NoiseTR=2, Sigma=4.0 )
SelectEven()
crop(8, 58, -6, -60)
#resize
#denoise

crash.log

------------------------------------------------------------
Environment
------------------------------------------------------------

StaxRip version: 1.2.0.2
OS Name : Windows 8.1 Pro
OS Version : 6.2.9200.0
OS Type : 64-bit
OS Culture : English (United Kingdom)

------------------------------------------------------------
.NET
------------------------------------------------------------

v2.0.50727 : 2.0.50727.4927
v3.0 : 3.0.30729.4926
v3.5 : 3.5.30729.4926
v4\Client : 4.5.51650
v4\Full : 4.5.51650
v4.0\Client : 4.0.0.0

------------------------------------------------------------
Source file MediaInfo
------------------------------------------------------------

E:\Evanescence - My Immortal.avs

General
Complete name : E:\Evanescence - My Immortal.avs
File size : 436 Bytes


------------------------------------------------------------
Exception
------------------------------------------------------------

System.ArgumentOutOfRangeException: Index was out of range. Must be non-negative and less than the size of the collection.
Parameter name: index
at System.ThrowHelper.ThrowArgumentOutOfRangeException()
at System.Collections.Generic.List`1.get_Item(Int32 index)
at StaxRip.AudioProfile.SetStreamOrLanguage() in C:\Daten\Projekte\VS\VB\StaxRip\General\AudioProfile.vb:line 83
at StaxRip.MainForm.AudioTextChanged(TextBox tb, AudioProfile ap) in C:\Daten\Projekte\VS\VB\StaxRip\Forms\MainForm.vb:line 2672
at StaxRip.MainForm.tbAudioFile0_TextChanged() in C:\Daten\Projekte\VS\VB\StaxRip\Forms\MainForm.vb:line 2723
at StaxRip.MainForm._Lambda$__108(Object a0, EventArgs a1) in C:\Daten\Projekte\VS\VB\StaxRip\Forms\MainForm.vb:line 0
at System.Windows.Forms.Control.OnTextChanged(EventArgs e)
at System.Windows.Forms.TextBoxBase.OnTextChanged(EventArgs e)
at System.Windows.Forms.Control.set_Text(String value)
at System.Windows.Forms.TextBoxBase.set_Text(String value)
at System.Windows.Forms.TextBox.set_Text(String value)
at StaxRip.MainForm.OpenVideoSourceFiles(IEnumerable`1 files, Boolean autoMode) in C:\Daten\Projekte\VS\VB\StaxRip\Forms\MainForm.vb:line 1758

[stax76]

@Carpo

Thanks for posting, it happened with all sources not containing audio.

@all

I fixed Dolby Atmos handling which again resulted in heavy editing of the audio processing, there were bugs with TrueHD, EAC3, various issues with muxing, avs decoding uses now ffmpeg instead of VirtualDubMod. Previously StaxRip would just abort if any audio processing fails, now it tries a alternative method in most cases, it now ignores delay when DirectShowSource is used, the audio text box shows better media info now, there are so many different formats, tools, and things to handle like delay, cutting, many special cases. I feel it's getting into a good shape.

https://www.dropbox.com/s/8tclme8xy6d8i7p/StaxRip_1.2.0.2_2015.01.28.7z?dl=0

[Patman]

Hi Stax...

Is it possible to include the gmkvextract gui?
For now your last release works very fine for me.

[stax76]

I might add it or make a simple demuxing GUI with mkvextract, mp4box and ffmpeg support.

If you like this GUI you might want to forward to the author to use code like this:

Application.EnableVisualStyles()
Application.SetCompatibleTextRenderingDefault(False)
SetProcessDPIAware()
Application.Run(New MainForm())

Right now the app operates in a compatibility mode in High DPI environments making it look blurry. The app wasn't designed for High DPI so the window manager puts it into a compatibility mode doing bitmap based upscaling and this causes the blurry look. The cure for this is declaring the app as High DPI aware and fix the layout if it wasn't built for High DPI to start with, for such a small app it's easy, for large apps it can be work. Some other popular .NET based encoding GUI has room for improvements in this department as well...

[Patman]

Hello,

here is a link to a small update package for some apps and avisynth-plugins.

Apps and Plugins (http://www.4shared.com/zip/CVDpHbQice/Apps_Plugins.html)

[stax76]

Hello Patman,

thanks for the package, except for autocrop, deen and AviSynth I'll include it.

[Patman]

Hello Stax,

I've found another audio format, which makes problems with the conversion to AC3. Eac3to and besweet produce an incorrect audio file. The audio format which causes these problems are DTS-ES (6.1) audio tracks. Only ffmpeg produced a correct audio file (AC3 5.1 448 kbit/s) without a "peep" tone. The easiest way to fix that problem is to set ffmpeg as default for DTS-ES audio. I hope you can check this and change that.

Greets Pat

[stax76]

I don't have a disc to test, did you use MakeMKV or another ripper?

I uploaded a release however, hopefully without too many bugs. :)

1.2.0.2 beta (2015-01-31)


Added new audio cutting method using mkvmerge and made it default for all audio formats.
Added many small improvements in audio processing
Added more x265 switches, there is a GUI option for more then 80 switches now, a search feature searching label, switch and help and a option for additional custom switches
Added feature to the x265 dialog to easily reset numeric values and option values to their default value by double clicking on the label
Added L-SMASH-Works AviSynth source filter, DGAVCDec removed
Added C++/QT based BDSup2Sub++, removed Java based BDSup2Sub
Added latest versions of ffms2 and MP4Box
Added setting to define which source filter will be used for a given source container in case the source filter is automatic
Added option to jobs dialog to either run job processing in the current or a new StaxRip instance, job processing works completely different now
Added ts to mkv remuxing configuration using Haali's dsmux, it works better then using TS directly or remuxing with mkvmerge
Improved GUI and help in various locations
Improved usability in eac3to demuxing dialog
Fixed compressibility check being broken in various configurations
Fixed bug with idx file containing multiple subtitles and fixed a vsrip related crash
Fixed bug Java not being found, if ProjectX is enabled in the settings Java is required.


https://www.dropbox.com/s/0i2zmsovraf3gb3/StaxRip_1.2.0.2_beta.7z?dl=0

https://sourceforge.net/projects/staxmedia/files/StaxRip%20beta

[VfBFan]

I found a folder called "Neuer Ordner" in the MP4Box folder, I think it is unnecessary.

[Patman]

I found a folder called "Neuer Ordner" in the MP4Box folder, I think it is unnecessary.

I think this is a backup folder with the old version. I've deleted this folder and no problems so far.

[cegy]

just to point out support for aac (just mux using aac-lc (adts) in this case) in a mkv/mp4 container is fully supported yet this says other wise... encoding to aac it doesn't kick up a fuss
http://i.imgur.com/E69nJxS.png

also why can't i see nor read the file name of the audio file any more another great option took away which means the user could end up muxing/encoding the incorrect file due to not being able to see/read it :eek: a game of blind guessing anyone ?:mad:
http://i.imgur.com/OGgldYd.png

[stax76]

@VfBFan

Thanks for the hint, Patman is probably right, backup I forgot to remove.

@cegy

The situation with aac is ugly because you never know if it's sbr unless you use a container such as m4a or mka. I should probably enable aac again, it would assume sbr in case the filename contains sbr, that's how a internal routine already work and I would check if it's applied everywhere necessary.

Some useful info is better like a filename especially when the filename is very long and meaningless, when StaxRip demuxes with eac3to it would for instance create a filename like so:

VC1 DTS HRA Core - ID2 - English.dtshr

'VC1 DTS HRA Core' is the m2ts filename, a random file in my test collection, there are three things StaxRip writes to demuxed audio filenames: stream ID, language and delay, in this case there is no delay. Different demuxing routines should use the same naming pattern. When such audio files are later opened StaxRip will detect stream ID, language and delay from the filename and display and handle it. A possible improvement could be instead of removing the complete filename is removing only the obvious part that is identical with the video source file, what do you think?

[cegy]

for me i liked how it was before as at least i'm able to see the file name fully and what you've suggested sure seems like a good idea aswell but maybe give a option to a user to pick between them at least you'll have best of both and the user can pick which option that they like

[Patman]

I don't have a disc to test, did you use MakeMKV or another ripper?

...

That happens during a conversion, wasn't a ripping prozess.

[stax76]

@cegy

Here is a hotfix with some adjustments according your feedback: https://www.dropbox.com/s/evh5c3fwcrss30i/StaxRip_hotfix_2015.02.01.7z?dl=0

@Patman

I used mkvmerge to remux atmos from m2ts to mkv, then opened the mkv with StaxRip and encoded using a AAC encoding profile, it worked.

[Patman]

@cegy

Here is a hotfix with some adjustments according your feedback: https://www.dropbox.com/s/evh5c3fwcrss30i/StaxRip_hotfix_2015.02.01.7z?dl=0

@Patman

I used mkvmerge to remux atmos from m2ts to mkv, then opened the mkv with StaxRip and encoded using a AAC encoding profile, it worked.

Yes, AAC conversion worked but AC3 doesn't work.

[stax76]

AC3 muxing or encoding?

[Patman]

AC3 muxing or encoding?

Encoding, muxing worked. It happens during the encoding process of DTS-ES 6.1 tracks. With forced ffmpeg encoder it worked but with automatic encoder settings the output file was incorrect.

[stax76]

Please let me see your log, mine is here: http://pastebin.com/K6ZX5CPw

[Patman]

Please let me see your log, mine is here: http://pastebin.com/K6ZX5CPw

A log will not be help for that. With eac3to as encoder the process will be done but with a corrupted audio file. There is the whole time a "peep" tone over the real sound. With ffmpeg as encoder the audio file is correct.
And i use Arcsoft DTS Decoder for eac3to.

[cegy]

@cegy

Here is a hotfix with some adjustments according your feedback: https://www.dropbox.com/s/evh5c3fwcrss30i/StaxRip_hotfix_2015.02.01.7z?dl=0

@Patman

I used mkvmerge to remux atmos from m2ts to mkv, then opened the mkv with StaxRip and encoded using a AAC encoding profile, it worked.

i like the change ;)

[stax76]

@Patman

I don't have a special DTS decoder installed. MediaInfo just shows TrueHD for my Atmos stream exactly the same as it would on a normal TrueHD stream so I don't know what else to do, you could ask the MediaInfo author (Zenitram) for help, he has a thread here.

[Patman]

@Patman

I don't have a special DTS decoder installed. MediaInfo just shows TrueHD for my Atmos stream exactly the same as it would on a normal TrueHD stream so I don't know what else to do, you could ask the MediaInfo author (Zenitram) for help, he has a thread here.

DTS-ES 6.1 isn't an atmos stream. Is a dts stream with a back center chanel. I've found the problem with the dts-es streams. Staxrip works very fine but the arcsoft dts decoder package was the malefactor. I've downgrade the version of the decoder and everything is fine. Thanks www for that info ;)

And the hotfix for staxrip is a great solution. Please implement this in the next version.

[mpfiorv]

Hey guys...brand new to stax rip....been using handbrake for some time.....But for some reason, stax rip does not detect Java which is installed on my computer.

I am running windows 8.1 64 bit....can anyone please point me in the right direction??? I have tried running both 32 and 64 bit versions of java and still nothing worked. I am lost for answers.

thanks a ton.

[stax76]

Hi and welcome,

are running the latest StaxRip release? There was a Java detection fix lately, does StaxRip ask for Java or ProjectX or does it simply skip ProjectX processing (it's intended to just ignore ProjectX if Java isn't found), maybe your source can be handled without ProjectX well so you can just disable it.

[dejong12]

The latest StaxRip (and the older ones) aren't loading some of my source files correctly. I've got a Full HD AVC .ts recording and when I load this video into StaxRip, the first ~30 frames equal to the the frame after those ~30 frames. Why this behaviour? When playing the source files, everything is fine.

EDIT: This does not happen when using elementary streams, in this case, .h264 and .mpa. But now arises another problem, the encoded video is slowed down double. The source video is 40 seconds (1000 frames), but now it is 1 min 20 seconds (2000 frames). FPS is still showing 25.

EDIT2: I can correctly use the .h264 elementary stream in StaxRip 1.1.9.0. I don't know if this is because of Helix or DGAVCIndex, but it think it is because the latter one is missing in 1.2.0.2. I tried adding it manually in 1.2.0.2 but it seems to be missing AVCSource (I think)(which is used in 1.1.9.0) and it gives me the "Index does not match the source file" error.

EDIT3: Version 1.1.9.0 also doesn't encode the elementary streams well. It's flickering blocks on all frames and there's no audio attached.

EDIT4: Strange, when remuxing the source file (.wtv) to .mp4, everything works correctly and I can encode the source file fine. So there might be some bugs when reading some containers.

[cegy]

stax76 the new start button sure is nice however would it be possible to keep the "idea" there but if the user does click start it does start whiles clicking on the drop down thing will bring up the other options.. is this possible as i've seen it in other apps ?

[stax76]

The latest StaxRip (and the older ones) aren't loading some of my source files correctly. I've got a Full HD AVC .ts recording and when I load this video into StaxRip, the first ~30 frames equal to the the frame after those ~30 frames. Why this behaviour? When playing the source files, everything is fine.

Full HD AVC TS is with default settings handled with DirectShowSource, your player might not use the same filters because of the following reasons:


StaxRip uses 32-Bit AviSynth, this means DirectShowSource will use 32-Bit DirectShow filters, your player might use 64-Bit filters, depending on which DirectShow filters you have installed this might result in a completely different filter graph. Two useful tools to tweak and diagnostic DirectShow are 'Codec Tweak Tool' and GraphStudio (expert tool).

Your player might prefer internal filters which would be different from system filters used by DirectShowSource, there are various player preferring internal filters most prominent is MPC.

Your player might not be DirectShow based at all (VLC being a popular example).


EDIT: This does not happen when using elementary streams, in this case, .h264 and .mpa. But now arises another problem, the encoded video is slowed down double. The source video is 40 seconds (1000 frames), but now it is 1 min 20 seconds (2000 frames). FPS is still showing 25.

Elementary streams are now handled with the FFVideoSource source filter (ffms2 plugin), I don't have much experience with elementary streams, never versions of FFVideoSource supporting them is one reasons DGAVCDec was released. Seeking/cutting don't seem to work very well with FFVideoSource and elementary streams, have you cut your recordings before? In which application? If possible please upload a 100 mb sample with DropBox's share feature. There are various AviSynth filters regarding frame processing like SelectEven/AssumeFPS etc.

EDIT2: I can correctly use the .h264 elementary stream in StaxRip 1.1.9.0. I don't know if this is because of Helix or DGAVCIndex, but it think it is because the latter one is missing in 1.2.0.2. I tried adding it manually in 1.2.0.2 but it seems to be missing AVCSource (I think)(which is used in 1.1.9.0) and it gives me the "Index does not match the source file" error.

It should be because of DGAVCIndex, adding it manually could be possible but difficult, you would have to add a demuxer at Tools/Settings/Demuxing with proper settings, then you would have to add a source filter by right-clicking the AviSynth filter list and chose 'Profiles' and last you would have to tell StaxRip which source filter profile to use for your demuxers output file, this new feature is at Tools/Settings/Filters.

EDIT3: Version 1.1.9.0 also doesn't encode the elementary streams well. It's flickering blocks on all frames and there's no audio attached.

Didn't you demux audio manually or didn't StaxRip demuxer demux audio? DGAVCDec can demux audio.

EDIT4: Strange, when remuxing the source file (.wtv) to .mp4, everything works correctly and I can encode the source file fine. So there might be some bugs when reading some containers.

From my experience FFVideoSource/ffms2 work best with MP4 and MKV and worst with TS, how did you remux? You can also mux ts to mkv, with mkvmerge these files are playing fine here but when I cut them in StaxRip audio goes async. Muxing with dsmux works better, it seems it does some stream fixing and don't add timestamps tags like mkvmerge. dsmux is enabled by default in StaxRip but it requires Haali's splitter to be installed, Haali's ts splitter seem to be much better then LAV's TS splitter so I recommend it everybody playing ts. To get the best out of Haali and LAV I recommend 'Codec Tweak Tool' and GraphStudio (advanced tool).

stax76 the new start button sure is nice however would it be possible to keep the "idea" there but if the user does click start it does start whiles clicking on the drop down thing will bring up the other options.. is this possible as i've seen it in other apps ?

It would be some work since it's not part of my standard UI elements and I never liked or understood the concept much. There might be other options like a checkbox at the bottom 'Start in separate instance' or a second button, also it might be better to move all buttons to the bottom.

[dejong12]

Elementary streams are now handled with the FFVideoSource source filter (ffms2 plugin), I don't have much experience with elementary streams, never versions of FFVideoSource supporting them is one reasons DGAVCDec was released. Seeking/cutting don't seem to work very well with FFVideoSource and elementary streams, have you cut your recordings before? In which application? If possible please upload a 100 mb sample with DropBox's share feature. There are various AviSynth filters regarding frame processing like SelectEven/AssumeFPS etc.

I record with Windows Media Center, which uses the .wtv container. I use VideoRedo TVSuite 5 to cut and remux, literally the best software to edit .wtv. I can upload you a sample, do you want it in a container (.ts or something else) or in elementary streams?

Also, like mentioned here, remuxing into .mkv solves all problems. No double length or frame hangs. I will probably remux it into .mkv from now on.


It would be some work since it's not part of my standard UI elements and I never liked or understood the concept much. There might be other options like a checkbox at the bottom 'Start in separate instance' or a second button, also it might be better to move all buttons to the bottom.
Yeah, I was wondering if you could put the option for this instance or another instance somewhere else, because honestly I don't like the extra click.

[fabje]

I have a small problem with Staxrip in combination with my TV recordings.

When I open my recordings in Staxrip and are getting processed by DGIndexNV I notice that the audio is always getting a huge delay, think of -600ms until -1500ms.
After the encode is done the start is fine, but at the end I'm always missing the amount of audio that's equal to the delay.

So if I have an audio thats get a delay of -1000ms I know already that I will miss 1000ms of audio at the end.

Is there a way this fix this in the code of Staxrip, to put the actual end cut point in the audio a little bit further so I won't miss any audio at the end?
Just guessing here about a solution...

[Patman]

I have a small problem with Staxrip in combination with my TV recordings.

When I open my recordings in Staxrip and are getting processed by DGIndexNV I notice that the audio is always getting a huge delay, think of -600ms until -1500ms.
After the encode is done the start is fine, but at the end I'm always missing the amount of audio that's equal to the delay.

So if I have an audio thats get a delay of -1000ms I know already that I will miss 1000ms of audio at the end.

Is there a way this fix this in the code of Staxrip, to put the actual end cut point in the audio a little bit further so I won't miss any audio at the end?
Just guessing here about a solution...

Hi,

i use the application delaycut to fix the delay. Works very well.