RipBot264 v1.18.3 - Simple and easy to use GUI -> IPOD . PSP . CONSOLES . BLURAY

[Atak_Snajpera]

Instead of Lagarith I would recommend using UT Video or MagicYUV.

[DoomBot]

I got this file from someone else so i can't do a new avi from the source, unless you mean change this avi using one of the 2?

[Atak_Snajpera]

It looks like you don't have this codec installed then http://www.videohelp.com/tools/Lagarith-Lossless-Video-Codec

[DoomBot]

Oh man your right! Thank you so much Atak_Snajpera it worked!

I love your software by the way.

[Atak_Snajpera]

@Killuah15
mediainfo report please ;)

[Killuah15]

General
CompleteName : H:\Dxtory\gerendert\Ohne Titel.avi
Format : AVI
Format/Info : Audio Video Interleave
Format_Profile : OpenDML
FileSize/String : 478 GiB
Duration/String : 45mn 49s
OverallBitRate/String : 1 493 Mbps
TCOD : 0
TCDO : 27491464000

Video
ID/String : 0
Format : RGB
CodecID : 0x00000000
CodecID/Info : Basic Windows bitmap format. 1, 4 and 8 bpp versions are palettised. 16, 24 and 32bpp contain raw RGB samples
Duration/String : 45mn 49s
BitRate/String : 1 492 Mbps
Width/String : 1 920 pixels
Height/String : 1 080 pixels
DisplayAspectRatio/String : 16:9
FrameRate/String : 29.970 fps
BitDepth/String : 8 bits
Bits-(Pixel*Frame) : 24.000
StreamSize/String : 477 GiB (100%)

Audio
ID/String : 1
Format : PCM
Format_Settings_Endianness : Little
Format_Settings_Sign : Signed
CodecID : 1
Duration/String : 45mn 49s
BitRate_Mode/String : Constant
BitRate/String : 1 536 Kbps
Channel(s)/String : 2 channels
SamplingRate/String : 48.0 KHz
BitDepth/String : 16 bits
StreamSize/String : 503 MiB (0%)
Alignment/String : Aligned on interleaves
Interleave_Duration/String : 250 ms (7.49 video frames)
Interleave_Preload/String : 250 ms



Hey, sorry for taking so long, i hope this is what you wanted to see...
(you know, I'm not an expert with mediainfo ^^)
i tried to render this video again, but I still get the error.
Another interesting fact: Ripbot264 is always demuxing the audio from the file, but when it wants to "Gather information" i get this error.
In the Job1 folder in the Ripbot264 folder, there is the demuxed audio (fully functional) but no video data (so the demux audio is working perfectly..)

[Atak_Snajpera]

OMG! Why do you use uncompressed format??? Use UT Video or MagicYUV codecs for intermediate files!

[Killuah15]

well I'm just following "orders" in a handbook to create videos xd, i sadly don't know how to use these codecs in vegas 12 :/

http://www.mediafire.com/view/?g6rml0ch392lf2p

allthough i have got enough free space for big files like these ^^ but do you think this is why Ripbot264 can't gather the information?

[Atak_Snajpera]

UT Video has installer. Click exe and then select UT Video codec in vegas.

[Killuah15]

Alright, I'm going to try that, you will get a post from me tomorrow :)
thanks for the help :)

EDIT: hey, i just tried this with a 10 seconds video, without UT video, i got the same error as using the nearly 500 Gb video.
with UT video as compressor, i get no error.

going to try that out tomorrow with my long video :)

EDIT 2: duuuuude, with UT Video, the rendering needed way less time then before, the output file is less 300 GB then before and RipBot264 can now demux audio and video. YOU are my hero :D
thank you very much for your help :)

[damia]

atak.. how to fix this ?? http://i.imgur.com/vi9KQ16.png after i try to re-install ripbot.. and happen this...

[Atak_Snajpera]

Compress content of C:\temp\ripbot264temp\job2 and send to mediafire or mega.co.nz. Do not include heavy files like audio or video.

[damia]

check atak.. already send using PM..

[cypher007]

trying to create some small mkv's from some large ones. basically going from 1080p dts-ma mkv to 720p ac3 stereo. im finding the result is a silent mkv. any ideas?

[damia]

trying to create some small mkv's from some large ones. basically going from 1080p dts-ma mkv to 720p ac3 stereo. im finding the result is a silent mkv. any ideas?

dts-ma to DTS try use Tsmuxer.. for Ac3 don't know.. :D

[8ternity]

Hi Atak,

I just seen your new version. Quite a long time i've not coming in the forum. Just made some computer changes!

Great work! Will test the newer version.

Thanks a lot for your good work.

8ternity

[8ternity]

dts-ma to DTS try use Tsmuxer.. for Ac3 don't know.. :D

You can use EAC3to in Ripbot264\Tools\ and install also EAC3toGUI.exe and you can downgrade DTS-HD MA to AC3 6CH or AC3 2CH.

You need to check 1 or 2 option(s):
- Downmix to 6CH (if you want ac3 6ch)
- Downmix to 2CH (if you want to 2CH)

EAC3to will convert it no problems to AC3 or DTS 6CH.

Check my attached files,.

[PNC]

i have mp4 file. openes in every player, ok!

try to coded in ripbot -
File could not be opened!
(C:\Temp\RipBot264temp\job1\getinfo.avs, line 4)

wtf?

[Atak_Snajpera]

show mediainfo report of that file.

[PNC]

show mediainfo report of that file.

where i can get it?

[soneca]

where i can get it?

MediaInfo (http://mediaarea.net)

[kwangkwang]

Hi
not sure if this had been ask and answered before.

Is there a permanent solution to have eac3to not "-down16" the audio. I would like to convert the audio as is and not down it to 16bit. I have been editing the extractstream cmd file in job to re-extract and convert the audio at its original bit.

Thank you

[Atak_Snajpera]

24 bit very often contains empty 8 bits. Besides 24 bit is useless for end user because even 16 bit is above human hearing range.

[PNC]

try to coded in ripbot 1.17.15 -
File could not be opened!
(C:\Temp\RipBot264temp\job1\getinfo.avs, line 4)

MediaInfo (http://mediaarea.net)

General
Complete name : C:\Temp\asd.mp4
Format : MPEG-4
Format profile : Base Media / Version 2
Codec ID : mp42
File size : 2.15 GiB
Duration : 30mn 14s
Overall bit rate mode : Variable
Overall bit rate : 10.2 Mbps
Encoded date : UTC 2014-01-02 19:13:21
Tagged date : UTC 2014-01-02 19:13:21

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L5.0
Format settings, CABAC : Yes
Format settings, ReFrames : 5 frames
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 30mn 14s
Bit rate mode : Constant
Bit rate : 10 000 Kbps
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 25.000 fps
Standard : PAL
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.193
Stream size : 2.11 GiB (98%)
Language : English
Encoded date : UTC 2014-01-02 19:13:21
Tagged date : UTC 2014-01-02 19:13:21
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709

Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 30mn 14s
Bit rate mode : Variable
Bit rate : 160 Kbps
Maximum bit rate : 326 Kbps
Channel count : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 34.6 MiB (2%)
Language : English
Encoded date : UTC 2014-01-02 19:13:21
Tagged date : UTC 2014-01-02 19:13:21

any ideas?

[george84]

This is a minor docu problem.

After installing RipBot and starting it there pops up a window with missing components.

For FFDshow it refers to http://www.ffdshow.info/files

Opening this link in browser gives only empty page. So I assume I have to find link myself with googling.

[Atak_Snajpera]

http://sourceforge.net/projects/ffdshow-tryout/

[crackinhedz]

Is there a way (like edit the .ini file) to auto select "core" audio before the demux? Sometimes I forget, and end up with the wav audio before I realize the mistake.

Greatly appreciated, thanks!

[8ternity]

@Atak

Did you received my bug regarding subtitles demuxed in "*.aac" and "*.dts" extension?

Thanks

[agressiv]

1) Any chance to use QAAC as an M4A encoder instead of just Nero?
2) Any chance of turning off Demuxing of the audio streams when opening a file (and pushing that back to when you are actually starting a job) ?
3) If #2 is not possible, I'd love to just "turn off audio" altogether since I'll be encoding via QAAC on my own anyways.

FYI, I'm almost always opening files created via MakeMKV.

I love to queue up 10 jobs at once, but when demuxing a ton of audio each time I open a file, it really slows things down...

"Batch Mode" simply has too many limitations, can't crop, etc...

[Atak_Snajpera]

1) Any chance to use QAAC as an M4A encoder instead of just Nero?
RipBot264 does not use Nero

2) Any chance of turning off Demuxing of the audio streams when opening a file (and pushing that back to when you are actually starting a job) ?
No. This is intentional design choice.

3) If #2 is not possible, I'd love to just "turn off audio" altogether since I'll be encoding via QAAC on my own anyways.
Why do you have to use QAAC? CT-AAC (Fraunhofer's aac) offers the same quality.

"Batch Mode" simply has too many limitations, can't crop, etc...
The problems with cropping are:
1) Different movies/videos may require different crop values.
2) Automatic cropping is not always safe. It will fail if source has many dark scenes.

If you can live with those problems then I will add cropping and deinterlacing option in batch mode.

[slalom]

He will have to check cropping at every video added
So it's the same thing

[agressiv]

Guess I didn't look that closely at the AAC encoder. Any way of going higher than 320kbit? This was the main reason I went to QAAC. The Profile folder only seems to have x264 profiles. The QAAC command line I use is --quality 2 --ignorelength -V 100, which gives around a ~512kbps average on a 5-channel track.

In terms of batch encoding, I automatically crop and then manually adjust the values, so it probably wouldn't be that useful if we had to accept the automatic value.

How about this idea on the demuxing - thread it into the background and just grey out the audio section until the demuxing is complete? That way we can at least do our cropping while it's demuxing.

Also, I noticed that if the source video doesn't have an audio track, chapters are not processed, and outside of adding a chapters file in the jobX_muxfiles.cmd manually, there doesn't seem to be way in the GUI to add a chapters file.

[Atak_Snajpera]

512 kbps for 6ch is insane. aac at 256,320 kbps is already transparent. Sorry but i'm not big fan of placebo effect.

[agressiv]

Compared to DTS / Lossless I thought it was fairly tame :(

[agressiv]

So, I went to do some spectrum analyses of audio since I really didn't find must out there on audio tracks from movies; just music files. It's hard to do "blind tests" of a movie soundtrack simply due to length, and the fact that there is more silence in a movie than a music track alone.

So, here's the results what I analyzed:
* Lossless Source (http://i.imgur.com/mGOepYb.png) (flac)
* QAAC at TVBR 100 (http://i.imgur.com/25GkEhs.png) (529kbps) (aac)
* QAAC at TVBR 63 (http://i.imgur.com/iX7nMy2.png) (344kbps) (aac)
* OPUS (http://i.imgur.com/B2cPtUk.png) (320kbps) (opus)
* FH-AAC (http://i.imgur.com/W9y9gNl.png) (318kbps) (aac)

I used RipBot's default settings for 320kbps encodes for OPUS and FH-AAC.

Conclusions
* The original source has a wide frequency response, much higher than 20khz, which most people can't hear. It's no surprise that lossy compression will clip here.
* Both OPUS and QAAC 100 clip around 20khz, QAAC goes a tad higher.
* QAAC 63 clips around 16.5khz. Not uploaded, but QAAC 54 (309kbps) clips around 16.1khz.
* I was impressed with OPUS (via MPC-HC), but even XBMC (Kodi) alpha2 doesn't support it yet, which is my primary platform. (it sees it as an audio track but won't play it, although the "Bleeding edge" nightly builds seem to work)
* FH-AAC clips below 16khz. I'd say, if you are going that low (and that's the highest setting in RipBot), it's not transparent. Of course, that is subjective, but I wouldn't call it placebo.

[Atak_Snajpera]

FH-AAC clips below 16khz. I'd say, if you are going that low (and that's the highest setting in RipBot), it's not transparent. Of course, that is subjective, but I wouldn't call it placebo.
If you can't hear it then it is placebo. Most adult people do not hear anything above 16-18 Khz. You can do simple test in audacity.

You can do additional listening test with 128kbps stereo music (64 kbps per channel). I guarantee that You won't be able to tell whether sound is loosely compressed or not.

Even with 96kbps (48kbps per channel) I have problems to hear difference here
http://forum.doom9.org/showthread.php?t=170938

[agressiv]

What equipment are you using to do these tests?

I have studio monitors on my main workstation, and while my home theater is mid-range, if the original recording is solid, in general, I can tell things apart. Maybe not 320kbps audio from flac, but I use flac to keep a *reference* recording so that I always have the original quality.

I downloaded sample 15 from that sound page.

For one, the recording of that garage band is terrible, and the dynamic range and frequency response is hardly indicative of anything. When a band has distortion effects on both the guitar and vocals, the quality of the lossy audio is fairly pointless, although some of them were extremely obvious. Just listen to the hi-hat.

Samples 17+19 are lone voices. Not worthy of testing for what I care about.

Sample 21 is an orchestra, and is again, primarly midrange.

I tried a few others, but none, to me, were anything above a low-range test, which is, I guess, what they are going after.

My point is, I wouldn't use a music blind test of primarly low-fi recordings as a basis for 5.1 sound from a sci-fi movie - it's apples and oranges.

[Atak_Snajpera]

I use decent headphones and apple's aac sounds very good for me.

320 kbps for 5.1 is the same as 128 kbps for 2.0. Note lfe is very easy to compress therefore we do not count it. In both cases you have 64 kbps per channel.

[soneca]

v1.18.0 BETA 1

Added: Support for x265 (DE mode works as well)
Added: Permanent Subtitles option in Batch Mode (automatically renders subtitles on frame if subtitles have the same name as video file)
Added: AVS meter (You can easily check processing speed of your script)
Added: workaround for crashing encoding client at the very end (while closing form)
Changed: encoded audio is now stored in Jobx folder
Updated: ffmpeg 2014-05-24, mediainfo 0.7.69 , MKVToolnix 6.9.1 , x264 r2431 , x265 1.10.114


http://www.mediafire.com/download/d3913jh3qu2vjfh/RipBot264v1.18.0_BETA1.7z

ToDo
add support for extra audio stream.
Atak, delayed support for additional audio track or is no longer on the list?

[Atak_Snajpera]

Delayed. It will require more work than I initially thought.

[agressiv]

Several bugs with the new version:
1) Distributed Encoding Broken
I'm doing some test HEVC encodes using the new version. There seems to be a bug during the joining of chunks where video stream is getting frozen at a point right where the chunks would be joined. I took an 8 second sample and shrunk it down to 720p. Note that I get the same behavior in x264 mode, so it's nothing with HEVC. (And you get it at every break of the chunks, so it's many times throughout the entire video)

https://www.dropbox.com/s/3ny3zby56wyzgr7/hevc-problem.mkv?dl=1

This pause at the 4-5 second mark of video is right where two chunks are joined, and you only get this with a distributed encode; you don't get it in an standalone encode. The audio never goes out of sync though. This is reproducible every time, even if you clear the job. Original file is tested fine (extracted via makemkv) The pause seems to be 26 frames, so 25 are getting dropped.

2) Cancelling Encoding
If you cancel an encode, it muxes a video anyways. This is new behavior. Its usually the previous video in the queue with the audio of the aborted encode.

3) Killing Encoding kills all processes with that process name
If you abort a job, it kills all processes that match the encoder name (e.g. x265_x64.exe). It should just abort the specific PID. (I sometimes encode using other tools and it is aborting those jobs).

[Atak_Snajpera]

Several bugs with the new version:
1) Distributed Encoding Broken
I'm doing some test HEVC encodes using the new version. There seems to be a bug during the joining of chunks where video stream is getting frozen at a point right where the chunks would be joined. I took an 8 second sample and shrunk it down to 720p. Note that I get the same behavior in x264 mode, so it's nothing with HEVC. (And you get it at every break of the chunks, so it's many times throughout the entire video)

Can you send me original file (5-10 min long) ?

2) Cancelling Encoding
If you cancel an encode, it muxes a video anyways. This is new behavior. Its usually the previous video in the queue with the audio of the aborted encode.
Yep. I've known about that. I basically forgot to do something about that.

3) Killing Encoding kills all processes with that process name
If you abort a job, it kills all processes that match the encoder name (e.g. x265_x64.exe). It should just abort the specific PID. (I sometimes encode using other tools and it is aborting those jobs).
If I find extra time I will change that.

[agressiv]

Can you send me original file (5-10 min long) ?

Well, the total encode is about 3.5gb, so I've taken a 10 minute sample which includes the pause from the small clip:

https://www.dropbox.com/s/xhay366pl9fqt14/test.mkv?dl=1

The pauses are here:

0:13
1:13
2:13
6:13
7:14
9:14

Not sure why there wasn't a pause every single minute, as the chunks all seem to be right around one minute, or ~1460 frames (23.976fps), give or take 10-30 frames. I did see a "-seek" in the batch file for avs2yuv but I honestly don't remember if that was there before.

[sneaker_ger]

That is not the original file, now, is it?

[agressiv]

OK, I guess I misread. Below is the unedited 10 minute clip which should align fairly close to the above 10 minute clip. The only thing I did was change the audio track to opus for size's sake.

It's still huge though, 2.3gb.

https://www.dropbox.com/s/dalgdj8jtxhiqa6/original.mkv?dl=1

[sneaker_ger]

The sample looks like open GOP.

@Atak_Snajpera
How does RipBot split the source segments for distributed encoding? Does it just use Trim() or is there some file splitting using e.g. mkvmerge?

[agressiv]

For what it's worth, I can't get the problem to occur with the 10 minute sample. In the original encode, the first time it appears is at the 12:01 mark.

I see Trim in the source avs chunks. He's using FFMpegSource as a decoder (ffms2.dll).

Here's some examples:
90.avs
Trim(128119,129584)
91.avs
Trim(129562,131036)
92.avs
Trim(131021,132467)

Notice the overlap, guessing that is where the -seek parameters come in for avs2yuv.

Edit: Will extract the first 13 minutes and see if I can get it at the same point. FYI, I'm using ffmpeg and copying the stream using -c copy, so there is no transcoding of video, just splicing. If I can reproduce it, I'll upload the new 13 minute file.

Edit 2: Doesn't happen in the smaller clip. Not sure what to do now :/

[agressiv]

Progress:

so, in the original chunks directory, if you encode 12.avs and 13.avs, you'll exhibit the problem. And to go further, if you look at the output from 13 alone, you'll see the video pause in the first ~20 frames. So, I swapped out the ffmpeg encoder and replaced it with dgdecnv and another with DirectShowSource:

Original:
#VideoSource
LoadPlugin("\\MACHINE\RipBot264temp\Tools\AviSynth plugins\ffms\ffms2.dll")
video=FFVideoSource("\\MACHINE\RipBot264temp\job1\video.mkv",threads=1)

New 1:
#VideoSource
LoadPlugin("D:\Movies\Utilities\Plugins\DGDecodeNV.dll")
video=DGSource("E:\Temp\RipBot264temp\job1\video.dgi")

New 2:
#VideoSource
video=DirectShowSource("E:\Temp\RipBot264temp\job1\video.mkv")


The two chunks with DGSource and DirectShowSource are fluid. So, the problem lies with FFVideoSource. I don't see many options for ffmsindex out there and haven't worked with it much myself.

FWIW, I'm using the latest ffdshow tryout, from 2014-06-28. Hopefully this helps.

[Atak_Snajpera]

1) Download latest FFMS2 (https://github.com/FFMS/ffms2/releases)
2) Delete everything in \Tools\AviSynth plugins\ffms and in X:\temp\RipBot264temp\Tools folder
3) Extract .7z file to \Tools\AviSynth plugins\ffms
4) Copy all 3 files (ffms2.dll,ffms2.lib,ffmsindex.exe) from \Tools\AviSynth plugins\ffms\x86 to \Tools\AviSynth plugins\ffms

FFMS 2.20 change log
Add support for Opus in MKV when ffmpeg/libav are built with libopus (qyot27)
vapoursource: Provide _AbsoluteTime metadata (Daemon404)
Fix a pile of small things found by Coverity (Daemon404)
Add support for HEVC in MKV (qyot27)
Fix infinite loop on garbage data at the beginning of AAC files (Plorkyeran)
Deal with the never-ending bitrot from FFmpeg/Libav API changes (Daemon404, Plorkyeran)
Make indexes not arch-specific and trivially smaller (Plorkyeran)
Fix bug where the first b-frame was replaced with the frame before it with open-gop h.264 (Plorkyeran)
Fix seeking issues with open-gop h.264 (Plorkyeran)
Fix corruption when seeking in interlaced h.264 (Plorkyeran)
Add support for multiple frames per audio packet as FFmpeg seems to have started doing that (Plorkyeran)
Fix seeking in Y4M files (Plorkyeran)
Fix dumping audio while indexing files
Add support for Matroska files with extremely large CodecPrivate sections
Speed up indexing Matroska files a bit


I hope this will fix your issue.

[agressiv]

Thanks for the quick reply. That indeed was the fix to the problem. Guessing this has been there all along, not sure how common Open GOP is.