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

[Atak_Snajpera]

forget about a copyrights. you are not the first user who sends me sample for debuging.
besides i asked only for first 11 min of the movie.

like i said before open video.mkv in mmg.exe and prepare proper sample. without that forget about any help here. your choice.

[Lehm]

After quite a bit of investigating I think I may have narrowed down what the problem is. Its not directly with your tool. The problem appears to be with Avisynth. I used virtualdub to load in the video.mkv file (and other test videos) with an avs script. On certain types of files when using the trim command if you trim on certain frames you will get three repeated frames at the beginning. On one file I noticed that it was every fourteenth frame. I was thinking this might be on keyframes, but that was just a guess. I was able to reproduce the error on a 1 minute segment. I don't mind sending a short clip like that. But I won't have time to send until tomorrow.

[Atak_Snajpera]

so send me 5 min chunk instead of 11min then. I will have to create bug report for ffmpegSource team.

[Gabes]

this is my profile:


--level 4.1 --aud --nal-hrd vbr --vbv-bufsize 25000 --vbv-maxrate 25000 --filter 0,0 --ref 5 --bframes 8 --b-adapt 1 --b-pyramid none --subme 8 --aq-mode 1 --trellis 1 --partitions all --me esa

I want to add deblock -3 , -3 and aq-strength 0.5-0.8 values but don't know where or after what. Please someone rewrite the good script I have to copy on my profile file.

PS: If someone finds inaccurate settings, just let me know.

[Atak_Snajpera]

this is my profile:


--level 4.1 --aud --nal-hrd vbr --vbv-bufsize 25000 --vbv-maxrate 25000 --filter 0,0 --ref 5 --bframes 8 --b-adapt 1 --b-pyramid none --subme 8 --aq-mode 1 --trellis 1 --partitions all --me esa

I want to add deblock -3 , -3 and aq-strength 0.5-0.8 values but don't know where or after what. Please someone rewrite the good script I have to copy on my profile file.

PS: If someone finds inaccurate settings, just let me know.

--level 4.0 --aud --nal-hrd vbr --vbv-bufsize 25000 --vbv-maxrate 25000 --filter -3,-3 --ref 5 --bframes 8 --b-adapt 1 --b-pyramid none --subme 8 --aq-mode 1 --aq-strength 0.X --trellis 1 --partitions all --me esa

[zanaitoryoushi]

encoded 850 frames, 16.96 fps, 2576.09 kb/s


C:\>"C:\Documents and Settings\Administrator\Desktop\tools\mp4box\mp4box .exe" -tmp "C:\Temp\RipBot264temp" -add "C:\Temp\RipBot264temp\video.264":name="The Avengers (1)-122" -fps 23.976 -add "C:\Temp\RipBot264temp\job1\1_audio_Undetermined.a ac":lang=und:delay=0 -mpeg4 -chap "C:\Temp\RipBot264temp\job1\chapters.txt" -new "C:\Documents and Settings\Administrator\Desktop\Encoder\The Avengers (1)-122.mp4"

The system cannot execute the specified program.
-------------------------

Elapsed Time: 00h:00m:53s


it has this error on the log and it doesnt show the output file...anyone knows how to fix it?

[Atak_Snajpera]

The system cannot execute the specified program.
this should explain everything.

[poolshark2014]

Hello Atak I was just wondering if higher audio bit rate for LC 2 channel will be added higher than 128kbps for MP4 encoding. The reason I ask is because when converting MKV files into MP4 format for Xbox360 the audio sounds low compared to the audio of 320Kbps on the PS3 and have to higher the volume drastically because Voice parts sound low while the action scenes sound louder than expected. Reason I use Xbox is because some movies contain Cinavia and therefore have to watch them on Xbox instead of PS3.

[Atak_Snajpera]

128 kbps AAC is transparent even for HydrogenAudio forum users. If you hear difference between 128 and 320 so it has to be something wrong with your equipment. Bitrate has nothing to do with low volume. Most likely dialog normalization has been removed.

[slalom]

A small bug,
in DE in the lower part of the window, displays fps & eta, which do not correspond to the sum of fps of the servers (or the server if there is one left) or the eta displayed in encoding chunks (or chunk)

[Atak_Snajpera]

basicly at the bottom you see average fps for all chunks. also eta is based on average fps.

[slalom]

I realised that, shouldn't it be better to show the current sum of the servers??

[Lehm]

I realised that, shouldn't it be better to show the current sum of the servers??

Maybe as an addition, but I think what it shows now is more important.

[Peter_mov]

Hi all,

I have a little problem with this great piece of software.....

A encoding server on one of my PC's switching between encoding and Idle . In the textbx the server displays :

"Encoding started" and then "encoding terminated" nothing else. The encoding client shows the IP "192.168.0.7:1000 ->Starting...." and then "Queue" without transfering the chunk. Share on the server pc is created (manually) and te network access is granted. The access from explorer to tempfile on the server pc is possible, but the folder is empty.

Any suggestions what I did wrong or how to solve this issue? Thx in advance.

[Atak_Snajpera]

see tutorial again maybe. it clearly shows step by step what has to be done.

[Shrekage]

Hi all,

I have a little problem with this great piece of software.....

A encoding server on one of my PC's switching between encoding and Idle . In the textbx the server displays :

"Encoding started" and then "encoding terminated" nothing else. The encoding client shows the IP "192.168.0.7:1000 ->Starting...." and then "Queue" without transfering the chunk. Share on the server pc is created (manually) and te network access is granted. The access from explorer to tempfile on the server pc is possible, but the folder is empty.

Any suggestions what I did wrong or how to solve this issue? Thx in advance.

Start an encoding session with that server as the only one...

then...

from the server that won't connect run the following from a command prompt window \\[servername]\ripbot264temp\job[x]\chunks\2.cmd

post the result if you can't determine the source of the problem...

[Peter_mov]

@Atak :
thx for response - believe me, I watched the Tutorial mo0re then 1 time :-).... and distributed encoding works on 4 PC's without trouble - :thanks:

@Shrekage : I tried this on a PC where the encoding server is encoding - This folder/file is not existing on that PC which is not working as a encoding server - sadly. :(

What I wonder - when is the shared folder created with it's content normally (is it created automatically? I think normally yes..) - means when is the content transfered and what makes it impossible to transfer it?

Or did I have to run that command from the PC with the encoding client which distributes the chunks to encoding servers?

[Atak_Snajpera]

make sure that all pcs are in the same workgroup.

[Shrekage]

@Peter_mov

I think you are making this more difficult than it has to be...

I will use the following to identify the machines
Server - the PC that runs the encodingserver.exe program
client - the PC that runs ripbot.

The client machine should have a share as described in the turorial \\client\ripbot264temp should be accessable from the server. As Atak pointed out, they should be on the same workgroup, but you could be able to type in \\[client]\ripbot264temp from the server machine and be able to access the temp folder on the client PC. once you have that accomplished, you should be able to use DE assuming firewalls, etc... are configured correctly.

[rc71]

--- Help Please ----

I need some help here. I'm trying to encode a BluRay and the audio (DTS) will NOT cooperate. If I use AAC, the sound is always out of sync . If I encode it in AC3 or Ogg, it's synced up just fine. Does anyone know what is the problem or how to fix it? I suspect it's the encoder, but that's just a guess.

[hoju3508]

@rc71, log files plz. What's your source?

[slalom]

--- Help Please ----

I need some help here. I'm trying to encode a BluRay and the audio (DTS) will NOT cooperate. If I use AAC, the sound is always out of sync . If I encode it in AC3 or Ogg, it's synced up just fine. Does anyone know what is the problem or how to fix it? I suspect it's the encoder, but that's just a guess.
Use the "no audio" option, and then remux the final mkv with the oroginal audio

[rc71]

Use the "no audio" option, and then remux the final mkv with the oroginal audio

That would leave me with a 1.8 GB audio file. What's the point of shrinking the video if I did that?

[rc71]

@rc71, log files plz. What's your source?

Format : DTS
Source is a BluRay Disc

[slalom]

That would leave me with a 1.8 GB audio file. What's the point of shrinking the video if I did that?
Beg your pardon??
Don't you want to encode the video?

[GMJCZP]

Hello Atak_Snajpera:

I was converting an AVI video with MP3 audio with this setup (from mediainfo):

General
Complete name : C:\video.mp4
Format : MPEG-4
Format profile : Base Media
Codec ID : isom
File size : 49,2MB
Duration : 21min.
Overall bit rate mode : Variable
Overall bit rate : 314Kbps
Encoded date : UTC 2012-09-22 02:31:14
Tagged date : UTC 2012-09-22 02:31:14
Writing application : Yamb 2.1.0.0 [http://yamb.unite-video.com]

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Main@L2.0
Format settings, CABAC : Si
Format settings, ReFrames : 3marcos
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 21min.
Bit rate mode : Variable
Bit rate : 246Kbps
Maximum bit rate : 260Kbps
Width : 512pixeles
Height : 384pixeles
Display aspect ratio : 4:3
Frame rate mode : Constante
Frame rate : 23,976fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8bits
Scan type : Progresivo
Bits/(Pixel*Frame) : 0.052
Stream size : 38,6MB (79%)
Title : video
Writing library : x264 core 125 r2208 d9d2288
Encoding settings : cabac=1 / ref=3 / deblock=1:1:1 / analyse=0x1:0x111 / me=umh / subme=8 / psy=1 / psy_rd=0.40:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=0 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=3 / lookahead_threads=1 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=5 / b_pyramid=0 / b_adapt=2 / b_bias=0 / direct=3 / weightb=1 / open_gop=0 / weightp=2 / keyint=240 / keyint_min=24 / scenecut=40 / intra_refresh=0 / rc_lookahead=50 / rc=crf / mbtree=1 / crf=20.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / vbv_maxrate=260 / vbv_bufsize=260 / crf_max=0.0 / nal_hrd=vbr / ip_ratio=1.40 / aq=1:0.60
Encoded date : UTC 2012-09-22 21:47:31
Tagged date : UTC 2012-09-22 02:31:21

Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : HE-AAC / LC
Codec ID : 40
Duration : 21min.
Bit rate mode : Variable
Bit rate : 64,5Kbps
Maximum bit rate : 72,2Kbps
Channel(s) : 2canales
Channel positions : Front: L R
Sampling rate : 44,1KHz / 22,05KHz
Stream size : 10,1MB (21%)
Encoded date : UTC 2012-09-22 02:31:20
Tagged date : UTC 2012-09-22 02:31:21

And I noticed that the audio was cut at the end a bit. After several tests I used Yamb to separate and rejoin the tracks and the problem was solved. Please check if in the process of joining tracks (muxing) at the end of the video processing is the problem. Thank you.

[slalom]

When combining chunks, a progress bar or a "%" or a sum, would be nice to show that the program is running OK

[Pratyatosa]

I'm trying to figure out how RipBot264 works it's magic so that I can do exactly the same thing using Perl scripts. Apparently, it uses other people's command line programs to do the actual conversions, compression etc. Therefore, it should be possible.

Here is what I've been able to figure out so far, using the compression of an exact copy (minus the copy protection) of a typical movie DVD as an example:

RipBot264: written in Delphi by Atak Snajpera

"Combining VOBs...": vjoin.exe (CombineVOBs.cmd) & eac3to.exe (VOBgetinfo.cmd)
"Indexing video stream...": DGindex.exe (No .cmd file created.)
"Demuxing subtitles...": VSRip_Launcher.exe & VSRip.exe (No .cmd file created.)
"Gathering information...": ? (No .cmd file created.)
"Extracting frame...": ? (No .cmd file created.)
----- User input -----
"Pass1...": avs2yuv.exe & x264_x64.exe (job1_EncodeVideoPass1.cmd)
"Muxing...": mkvmerge.exe (job1_MuxFiles.cmd)

The "Gathering information..." & "Extracting frame..." steps happen too fast for me to see what program is executing by looking at the Windows Task Manager list, but it could also be that no external program is used for those steps.

[Pratyatosa]

RipBot264 is a brilliant program, but it could be improved as follows:

1. Allow the video track to be simply copied. Maybe all the user wants to do is to convert the audio to a different codec.

2. Add an option to convert the audio to 2 channel AC3 if the input is, for example, 2 channel AAC.

3. Add an option to skip all user interaction and simply use the defaults. Perhaps more settings could be added to the .ini file.

4. Display what the bit rate is for the audio input, so an intelligent choice may be made for the output bit rate. Either that or don't allow illogical choices. It doesn't make sense to offer the user the option of 640kbps AC3 audio if, for example, the source file has 448kbps AAC audio. Why even offer such an option?

5. Stop stripping the dots (periods) out of the target file name.

[slalom]

1. Set your parameters, then stop the program when audio conversion is done. Mux new audio and video.

[Atak_Snajpera]

1. Allow the video track to be simply copied. Maybe all the user wants to do is to convert the audio to a different codec.
This is only available for HDDVD/blu-ray sources

2. Add an option to convert the audio to 2 channel AC3 if the input is, for example, 2 channel AAC.
AC3 2.0 is only available for AVCHD. For other containers AAC is better than ancient AC3.

3. Add an option to skip all user interaction and simply use the defaults. Perhaps more settings could be added to the .ini file.
Batch mode maybe?

4. Display what the bit rate is for the audio input, so an intelligent choice may be made for the output bit rate. Either that or don't allow illogical choices. It doesn't make sense to offer the user the option of 640kbps AC3 audio if, for example, the source file has 448kbps AAC audio. Why even offer such an option?
WRONG! 640kbps AC3 makes sense because AC3 is older and less efficient than AAC. So in order to limit further degradation in sound quality you need to use more bits!

5. Stop stripping the dots (periods) out of the target file name.
not all muxers like multiple periods in file name!

[soneca]

Atak,

The sharpen filter applied in this case makes no sense.

#Resize
video=Spline36Resize(video,1920,1038).Sharpen(0.2)

Please create an option to enable/disable via .ini file.

[rc71]

5. Stop stripping the dots (periods) out of the target file name.

I completely Disagree with that one. I love that it does that. It also makes it easy to id files that have been converted when the source was mkv as well as the target.

[rc71]

This is only available for HDDVD/blu-ray sources


AC3 2.0 is only available for AVCHD. For other containers AAC is better than ancient AC3.


Batch mode maybe?


WRONG! 640kbps AC3 makes sense because AC3 is older and less efficient than AAC. So in order to limit further degradation in sound quality you need to use more bits!


not all muxers like multiple periods in file name!


Okay Atak.

But here is the problem I have, I can't ALWAYS use AAC. I love it, but with certain source I ALWAYS get sync problems. So my options are use something else, NeroAAC seems to work for me, or use one of the other audio codecs like AC3 or Vorbis.

[Pratyatosa]

1. Set your parameters, then stop the program when audio conversion is done. Mux new audio and video.

Yes. Step-by-step instructions on how to do this are given on <Tutorial: How to Convert DTS/AAC 5.1 Audio to 5.1 Dolby Digital (AC3) Within a Video File (http://llbest.com/?P=79)>.

However, there's a better way: Free Download & Tutorial: Batch/Multiple Subfolder Conversion of MKV/MP4/3GP/AVI/DIVX/M2TS/MOV/MPG/MPEG/OGV/TS/VOB/WEBM-DTS/AAC/MP3/OGG Video Files to MKV-AC3 (Dolby Digital), Preserving All Tracks! (http://llbest.com/?P=7i)

[Pratyatosa]

I completely Disagree with that one. I love that it does that. It also makes it easy to id files that have been converted when the source was mkv as well as the target.

Why not simply note the "Date modified" or which folder the newly created mkv file is in?

[Pratyatosa]

not all muxers like multiple periods in file name!

But RipBot264 uses MKVmerge to do the muxing, which has no problem with multiple periods in files.

[Pratyatosa]

Batch mode maybe?

Maybe. But I haven't tried it yet. Where may I find some explanation of how it works?

[Pratyatosa]

WRONG! 640kbps AC3 makes sense because AC3 is older and less efficient than AAC. So in order to limit further degradation in sound quality you need to use more bits!

Even if this is true, the RipBot264 software should be able to easily calculate what the ideal target file audio track bitrate is, depending upon what the source file bitrate is. So, why not have an INI file option to have the program choose the best bitrate to use?

Another thing is that one user is always going to want AAC audio in the target file. Another is going to always want AC3 audio, etc. So why not have the output audio track codec an INI file default setting, same as MP4 vs. MKV?

[Pratyatosa]

AC3 2.0 is only available for AVCHD. For other containers AAC is better than ancient AC3.

If I try to play a file with AAC audio on my home theater system, I get nothing but silence, because the "ancient" receiver that I use can only decode AC3 and DTS. Therefore, my standard is 100% MKV with AC3 audio, even if it's only 2 channels.

[flebber]

Even if this is true, the RipBot264 software should be able to easily calculate what the ideal target file audio track bitrate is, depending upon what the source file bitrate is. So, why not have an INI file option to have the program choose the best bitrate to use?

Another thing is that one user is always going to want AAC audio in the target file. Another is going to always want AC3 audio, etc. So why not have the output audio track codec an INI file default setting, same as MP4 vs. MKV?

The "best" option is the highest. You can choose to lower it but they would be based on personal preference and who can create s formula to calculate personal preference?

[GoldenEye4ever]

I've been using RipBot264 for the last couple years and really appreciate the great work.

I have 2 feature requests:
Multiple Audio Streams
When I'm encoding a foreign film, I like to encode the movie into an MKV file and including the English audio track, the native audio track and the English subtitle track.
Currently I have to encode the movie with one of the audio files, then extract the 2nd audio file manually and then add it to the MKV file using MKVMerge.
I'd really appreciate it if I could select more than 1 audio track into a build.

Stopping Initial Scan
When I first click the Add button, RipBot begins analyzing the 1st video stream; especially when it comes to tv shows, the 1st stream may not be the stream I want.
I'd appreciate it if I could cancel the initial analysis, or have to click a button to initialize the scan.


Thanks for the great app :)

[Chetwood]

Multiple Audio Streams

+1. It's also interesting for people who want Director's Comments.

[GMJCZP]

Hello Atak_Snajpera:

I was converting an AVI video with MP3 audio with this setup (from mediainfo):

General
Complete name : C:\video.mp4
Format : MPEG-4
Format profile : Base Media
Codec ID : isom
File size : 49,2MB
Duration : 21min.
Overall bit rate mode : Variable
Overall bit rate : 314Kbps
Encoded date : UTC 2012-09-22 02:31:14
Tagged date : UTC 2012-09-22 02:31:14
Writing application : Yamb 2.1.0.0 [http://yamb.unite-video.com]

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Main@L2.0
Format settings, CABAC : Si
Format settings, ReFrames : 3marcos
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 21min.
Bit rate mode : Variable
Bit rate : 246Kbps
Maximum bit rate : 260Kbps
Width : 512pixeles
Height : 384pixeles
Display aspect ratio : 4:3
Frame rate mode : Constante
Frame rate : 23,976fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8bits
Scan type : Progresivo
Bits/(Pixel*Frame) : 0.052
Stream size : 38,6MB (79%)
Title : video
Writing library : x264 core 125 r2208 d9d2288
Encoding settings : cabac=1 / ref=3 / deblock=1:1:1 / analyse=0x1:0x111 / me=umh / subme=8 / psy=1 / psy_rd=0.40:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=0 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=3 / lookahead_threads=1 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=5 / b_pyramid=0 / b_adapt=2 / b_bias=0 / direct=3 / weightb=1 / open_gop=0 / weightp=2 / keyint=240 / keyint_min=24 / scenecut=40 / intra_refresh=0 / rc_lookahead=50 / rc=crf / mbtree=1 / crf=20.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / vbv_maxrate=260 / vbv_bufsize=260 / crf_max=0.0 / nal_hrd=vbr / ip_ratio=1.40 / aq=1:0.60
Encoded date : UTC 2012-09-22 21:47:31
Tagged date : UTC 2012-09-22 02:31:21

Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : HE-AAC / LC
Codec ID : 40
Duration : 21min.
Bit rate mode : Variable
Bit rate : 64,5Kbps
Maximum bit rate : 72,2Kbps
Channel(s) : 2canales
Channel positions : Front: L R
Sampling rate : 44,1KHz / 22,05KHz
Stream size : 10,1MB (21%)
Encoded date : UTC 2012-09-22 02:31:20
Tagged date : UTC 2012-09-22 02:31:21

And I noticed that the audio was cut at the end a bit. After several tests I used Yamb to separate and rejoin the tracks and the problem was solved. Please check if in the process of joining tracks (muxing) at the end of the video processing is the problem. Thank you.

Any response Atak_Snajpera?

[Pratyatosa]

WRONG! 640kbps AC3 makes sense because AC3 is older and less efficient than AAC. So in order to limit further degradation in sound quality you need to use more bits!

Free Download & Tutorial: Batch/Multiple Subfolder Conversion of YouTube (or Any 2-Track) Video Files to MKV-AC3 (Dolby Digital) (http://llbest.com/?P=7j) lets ffmpeg decide what the target file's bitrate is. For example, if the source is MP4-192 kbps AAC audio, then it chooses MKV-192 kbps AC3 for the target file.

So, according to you, the author of ffmpeg doesn't know what he's doing, right?

[Pratyatosa]

+1. It's also interesting for people who want Director's Comments.

Asus O!Play media players cannot handle multiple audio tracks. Is there a stand-alone USB media player that can?

[Atak_Snajpera]

So, according to you, the author of ffmpeg doesn't know what he's doing, right?
Imagine that you have music encoded in HE-AAC@64kbps and for some reason you have to re-encode that file to MP3 format (older less efficient). What bitrate would you choose for MP3? 64kbps or 128kbps+?

[Ryushin]

If I try to play a file with AAC audio on my home theater system, I get nothing but silence, because the "ancient" receiver that I use can only decode AC3 and DTS. Therefore, my standard is 100% MKV with AC3 audio, even if it's only 2 channels.

Ancient or not, very, very, very, very few receivers can decode AAC. I spent better part of a day look at receivers and I only found a couple of very high end models that supported it. At that point, I switched all my encoded to AC3. Though for 2.0 sources, it's still AAC, and I'm hoping it will just stream PCM at that point.

[73ChargerFan]

Ancient or not, very, very, very, very few receivers can decode AAC. I spent better part of a day look at receivers and I only found a couple of very high end models that supported it.
Wait - lots of receivers have ipod / aac support (Pioneer VSX-522-K, $250) or are they limited to 2.0 sources?

[Chetwood]

Asus O!Play media players cannot handle multiple audio tracks. Is there a stand-alone USB media player that can?
Are you sure about this? I mean, it's been a while since I tested the O!Play but it's a Realtek player and they can handle multiple audio tracks, as do all WDTVs (Sigma chipset).