2010-03-10, 05:43 | Link #481 |
Fansubber
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Hmm, I've been stupidly applying deinterlacing BEFORE dot crawl removal. I'll reverse the process, add Bifrost, and report back.
@TheFluff: ColorMatrix was automatically put in by MeGUI, and I assume it's some kind of filter for DVD sources. What's the best filter to replace this with? Thanks. Edit: Downloaded YATTA 7.13b6 from http://ivtc.org/. Documentation is.... underwhelming. Eh, this tool is probably above my skill level, I'll likely just live with having one ugly episode if I can't fix it with notepad. Source: Old settings: New settings (modified order and deinterlace settings): Basically changed the field order from 1 to 0 in tfm, and added Bifrost(). Oh well, I guess there's no way to get rid of all the artifacts. Whack the mole on the rainbowing of the blond guy's shirt, brand new artifacts appear on the front guy's red shirt... I am going to blend the frames and see what happens Last edited by Zergrinch; 2010-03-10 at 20:50. |
2010-03-11, 05:30 | Link #484 |
Excessively jovial fellow
Join Date: Dec 2005
Location: ISDB-T
Age: 37
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the dshow mpeg2 decoder included with cccp defaults to blend deinterlacing
if you want accurate source screenshots use dgmpegdec or mpeg2dec3 in avisynth and screenshot it in vdub or something
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2010-03-11, 18:47 | Link #486 |
Senior Member
Fansubber
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If you just use blend deinterlacing to smooth out the dotcrawl, you might want to set cthresh=-1 so that TFM() will deinterlace every frame. That will avoid flickering between frames that got deinterlaced and ones that didn't, as well as make sure it catches all the dotcrawl.
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2010-04-07, 11:02 | Link #488 |
Junior Member
Join Date: Apr 2010
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Hy guys,
I have the following Problem. I'm unable to load mkv or mp4 files with avysynth, through DirectShowSource. The Thing is that this Situation is only since yesterday, before that it worked fine. Now when i try to load a Mp4 with a script like this: DirectShowSource("D:\[Raws-4U] Kaichou wa Maid-sama! - 01 (TBS 1280x720 H.264 AAC Chap).mp4", fps=23.976, convertfps=true).ConvertToYV12() I get no Audio and depending on what VirtualDub version i use, the Video plays(1.9.0) or not(1.9.8). When i try to load an avi via Directshowsource it works fine. The mp4 and mkv doesn't have a Problem with mpc either. The Things i have done until now: Reinstalling CCCP Trying to reset the Settings of CCCP Reinstalling CoreAVC and the AACparserfilter. The Problem appeared out of the blue, i changed nothing, neither rebooting the system or installing Something, my system runs 24/7^^. I hope you can help me because with the current Situation i'm unable to work on the Projects of my group. //edit Sorry for my bad english. I'm not a native speaker^^ |
2010-04-08, 00:28 | Link #490 |
Junior Member
Join Date: Apr 2010
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Nope that doesn't work.
In Fact that was the first thing i tried^^ But it seems like i must clarify something. It seems that the AAC loading isn't working not only since yesterday, rather it seems like it hasn't worked since i upgraded(new installation)the Win7 Rc1 to the RTM. I have come to this conclusion, because the files that worked yesterday contained vobis audio, and that works. And the other thing is, i talked to our other encoder in the group and the aac loading doesn't work for him either, but he hasn't noticed that because he normaly exctracts the audio from the raw and convert it with an external Program. So it seems it's a Problem with the Windows 7 rtm build. The non Playing video in the new Vd was caused by the audio problem, because if i load the raw with parameter Audio=false. The Video plays. |
2010-05-19, 22:00 | Link #491 |
Junior Member
Join Date: May 2010
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What's the most common method for deinterlacing transport streams in avisynth amongst raw cappers?
Sorry for bumping a thread that's over a month old, but it was only half way down the first page so meh. And to the post above me despite that I'm sure you've moved on by now... run a filter tweaker and set your AAC filters to ffdshow. |
2010-09-02, 16:25 | Link #494 |
Senior Member
Author
Join Date: Jul 2007
Location: Virginia Tech
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Here's a problem I've had, but gotten around, and is coming up again.
I have a Blu-Ray. It's 24 bit PCM audio, 2.0. I want to trim it, and then convert it to FLAC. I load it into eac3to with eac3to "C:\Downloads\S1\BDMV\Stream\00000 PID 1000 delay 0ms.pcm" "C:\Downloads\S1\BDMV\Stream\00000 PID 1000 delay 0ms.wav" -2 -24 -big And I get the WAV file. But when I load it into avisynth (WAVSource("C:\Downloads\S1\BDMV\Stream\00000 PID 1000 delay 0ms.wav") I get the error "No compatible ACM codec to decode 0xFFFE audio stream to PCM (03.avs, line 2)" I usually get that WAV file, put it in audacity, and export it, and then it works. But I can only find how to get 16 bit output. I want 24 bit. How can I do this without having to resort to mkvmerge?
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2010-09-03, 05:44 | Link #495 | |
Excessively jovial fellow
Join Date: Dec 2005
Location: ISDB-T
Age: 37
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For some reason eac3to always sets the AudioCompression field of the fmt chunk of the WAV file to FFFE, which means "use the WAVEFORMATEXTENSIBLE header instead". I suspect it does this because WAVEFORMATEXTENSIBLE supports complex channel layouts (while the old fmt chunk doesn't), and madshi was too lazy to make a special case for plain ol' stereo. Naturally, to make your life more interesting, the old ACM model used by VirtualDub and Avisynth doesn't support WAVEFORMATEXTENSIBLE.
As for the solution, either find a program that supports WAVEFORMATEXTENSIBLE and can save 24-bit stereo WAV's using the old WAVEFORMATEX/fmt chunk header (sox might work), or take the interesting route: hex edit the WAV file produced by eac3to, find the fmt chunk, go to offset 0x04 relative to its start (if the file starts with RIFF, 4 bytes, WAVE and then "fmt ", it should be 20 bytes from the file start), and you should find two bytes with the hex values FE and FF (in that order, remember the endianness). Change them to 01 and 00, respectively, save the file, and see what happens. It may work, or then it may not. See also https://ccrma.stanford.edu/courses/4...ts/WaveFormat/ Edit: or find a wav source plugin for avisynth that supports WAVEFORMATEXTENSIBLE. I think there might be one; at least I know there is one that supports w64. Edit edit: JEEB pointed out to me that eac3to supports W64 too, so ignore all of the above, just write to that instead and use either the W64 source plugin whose name escapes me at the moment or use ffaudiosource. Edit edit edit: or you could let eac3to write FLAC and do the cutting with split_aud.pl which should give the same results as using Avisynth. Edit edit edit edit: found the reason why eac3to does that: Quote:
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Last edited by TheFluff; 2010-09-03 at 05:57. |
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2010-09-03, 18:40 | Link #496 |
Senior Member
Fansubber
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That's odd. I've been encoding some BDs with 24 bit PCM audio and loading the WAV files eac3to spits out into AviSynth 2.58 for trimming just fine.
eac3to CLI: eac3to "00002.m2ts" 2: "00002 - 2 - PCM, 2.0 channels, 24 bits, 48khz.wav" -2 -24 -big -48000 AviSynth script: video = ffvideosource("00002.mkv", fpsnum=24000, fpsden=1001) audio = wavsource("00002 - 2 - PCM, 2.0 channels, 24 bits, 48khz.wav") audiodub(video,audio) But anyway, you could try NicAudio for loading them into AviSynth. |
2010-09-06, 22:29 | Link #498 |
Junior Member
Join Date: Sep 2010
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Hi im Having Problem When Im Inserting A Afx Picture, Heres my Code,
Everytime When i Delete the = Beside the insertsign, it will say no such function. DirectShowSource("C:\Users\Aimbotter33\Desktop\[FFFpeeps]_Kaichou_wa_Maid-sama_23_[720p][B40216DC].mkv") sign1 = ("C:\Users\Aimbotter33\Desktop\[FFFpeeps]_Kaichou_wa_Maid-sama_23_[720p][B40216DC].mkv") logo = imagesource("C:\Users\Aimbotter33\Desktop\images.p ng", pixel_type="RGB32", end=100) insertsign=("last, logo, 0") insertsign=("last, sign1, 563, 900") |
2010-09-06, 22:36 | Link #499 | |
Junior Member
Join Date: Mar 2009
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Check out the very first post of this thread.
Quote:
If you follow the instructions the script will work. |
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