MeGUI: General Questions and Troubleshooting Thread

[Carpo]

I've got ffdshow and haali, but MeGui seems to not see them.

Ok, I unistalled ffdshow, haali and avisynth. Then installed avisynth 2.58 and matroska. M2TS imports, but is still interlaced and analysing type of interlace gives error, that program couldn't recognise this type of interlace.
http://ifotos.pl/img/source_sqhaqs.png

Yesterday, program converted into avi XviD perfectly, but in spite of "Progressive" mode, there was still interlace on video.

I get that from time to time, change the % amount of how much megui will check what the source is.

Options -> Settings -> Configure Source Detector

and its the top most box, some times i have had to put it as high as 5% for it to see what the source is

Well slower is better, but how much is an other question. One thing I can say for sure is that if you use settings faster than the defaults you will lose quality pretty quickly, because you are turning off the psy optimizations. For the slower ones there are diminishing returns, that is you are definitly losing more in speed than you are gaining in encoding efficiency, and if you care about encoding time you might as well stay at the defaults and use a little higher bitrate or lower ratefactor.


I have found no need to go below slow, most of my encodes done on medium. I haven't had a source when tested where I could say slower has been required for me or produced a more notable result than slow.

Thanks both, what crf settings would you advise for dvd to mkv and bluray to mkv? i have googled and found many different settings, or would 2 pass be better?

[nurbs]

CRF and 2 pass are basically the same at the same bitrate. I like CRF better because I don't have space constraints and it makes sure different sources get the appropriate bitrate as needed. I use CRF 21 for 720p and CRF 20 for DVD resolutions.

[Carpo]

ah i remember you suggesting those before, poor memory i have - thanks nurbs :-)

[flebber]

Are you able to play this file in MPC-HC?

[krosswindz]

I am having issues with a particular video file when I encode it to rawavc using megui and then mux it with audio and subtiles using mkvmerge to create a mkv file. I am having issues with the final runtime of my mkv file. Some of the initial discussion can be found in this thread http://forum.doom9.org/showthread.php?t=157545.

MeGUI logs do not show any errors and the right number of frames have been encoded. Once I mux it my runtime is completely off when compared to what the actual runtime should be. Any help on this is appreciated.

The command line used to create the file is as follows


pass 1: "path\to\vfw4x264.exe" --level 4.1 --pass 1 --bitrate 3754 --stats "path\to\video.stats" --deblock -2:-1 --keyint 300 --min-keyint 24 --bframes 4 --ref 6 --weightp 0 --me umh --sar 1:1 --fullrange on --output NUL "path\to\video.avs"

pass 2: "path\to\vfw4x264.exe" --level 4.1 --pass 2 --bitrate 3754 --stats "path\to\video.stats" --deblock -2:-1 --keyint 300 --min-keyint 24 --bframes 4 --ref 6 --weightp 0 --me umh --sar 1:1 --fullrange on --output "path\to\video.264" "path\to\dhoom.avs"

[Sharktooth]

post your avisynth script.

[krosswindz]

post your avisynth script.

Here is the avisynth script file.


SetMemoryMax(2048)
LoadPlugin("path\to\DGAVCDecode.dll")
p1=AVCSource("path\to\video-1.dga")
p2=AVCSource("path\to\video-2.dga")
p1+p2
crop( 0, 130, 0, -134).Lanczos4Resize(1280,544)

[AMED]

try this

SetMemoryMax(2048)
LoadPlugin("path\to\DGAVCDecode.dll")
p1=AVCSource("path\to\video-1.dga")
p2=AVCSource("path\to\video-2.dga")
p1++p2
crop( 0, 130, 0, -134).Lanczos4Resize(1280,544)

[krosswindz]

I changed my avs script. Out of curiosity what is the difference between '+' and '++' in an avs script?

I just noticed something weird, if I manually add the number of frames in both my source dga files and the total frames displayed by megui for the avs file they dont match. The difference is like 8 frames. Any idea what could be causing this? I am wondering could this be the reason why I am facing this issue.

[LigH]

Regarding "+" vs. "++": Read the documentation about "UnalignedSplice" vs. "AlignedSplice".

Surprisingly, this should only be related to clips with both video and audio track. I don't see the relevance either.

[krosswindz]

Chaning '+' to '++' has no effect, I am still facing the same issue. There are no errors reported in MeGUI after the encode, nor does mkvmergeGUI give me any errors when I mux the files. I still have the issue where the total runtime reported by mediainfo or any media player is way off when compared to what the actual runtime is calculated from the number of frames and fps. Any help on this is appreciated.

[poisondeathray]

krosswindz - what happens when you try a different container? e.g. mp4 using mp4box (or yamb) , or .m2ts using tsmuxer ?

what kind of video is the source? native progressive ?

does avs script preview ok?

what happens if you use different source filter? eg. ffvideosoruce()

[tebasuna51]

...
MeGUI logs do not show any errors and the right number of frames have been encoded. Once I mux it my runtime is completely off when compared to what the actual runtime should be. Any help on this is appreciated.
...
And what is the duration of the muxed audio?
If is greater than the video duration the container can show the higher one.

[Sharktooth]

also dgavcdecode may be the culprit.try using ffms2

[krosswindz]

krosswindz - what happens when you try a different container? e.g. mp4 using mp4box (or yamb) , or .m2ts using tsmuxer ?

what kind of video is the source? native progressive ?

does avs script preview ok?

what happens if you use different source filter? eg. ffvideosoruce()

I havent tried other containers, the only reason I havent used mp4box being that afaik it doesnt support DTS. May be I havent figured how to use it with DTS. I will mux it using tsmuxer and let you know the results.

The video source is progressive.

The avs script previews perfectly, except for the total number of frames which for some reason ends up short by 8 as opposed to what it should be.

I have tried it with DGDecodeNV as the source filter and I face the exact same issue, with DGNVtools I dont have the issue with the number of frames being incorrect.

And what is the duration of the muxed audio?
If is greater than the video duration the container can show the higher one.

The duration of the audio before being muxed is 2h 9m 36s after being muxed mediainfo reports the same as that of the video.

also dgavcdecode may be the culprit.try using ffms2

I havent tried ffms2, though like mentioned earlier I have tried DGDecodeNV and ended up with exactly the same issue.

[krosswindz]

I had the same issue with ffms2 as well, I figure the problem was actually with the subtitle stream that I was muxing. The subtitle had a discontinuity which was causing the runtime to go hay wire. Thanks for all the help appreciate it.

[swainjena]

How to change the frame rate of a mp4 video from 25fps to 23.976 fps????

[swainjena]

can anybody help me with this frame rate conversion problem plz....

[zmaster]

swainjena,
See the documentation for AviSynth. Look for information about the function AssumeFPS()/ChangeFPS()/ConvertFPS().

[Lyle_JP]

swainjena,
See the documentation for AviSynth. Look for information about the function AssumeFPS()/ChangeFPS()/ConvertFPS().

Or he could just encode to raw and set framerate in the muxer.

[blah123]

I've got interlaced AVC/h.264 in TS. What i should use instead of DGAVCIndex to demux this file? DAIndex crashes with this:
http://ifotos.pl/img/afterffms_sasqwr.jpg

[poisondeathray]

^This was answered already 2 pages back. Use directshowsource or dss2 + recent ffdshow build + haali media splitter ,or dgnvtools (not free, needs nvidia card)

[blah123]

^This was answered already 2 pages back. Use directshowsource or dss2 + recent ffdshow build + haali media splitter ,or dgnvtools (not free, needs nvidia card)

Ok, but how I should do it? Which app?

I use TsMuxer, but when i remove interlace and convert files after muxing video with audio, the capture in file has a weird freeze of few frames and gives not synchronized audio and video. But whilst watching only video, it's good.

[gorca]

Possibly someone can help me out because am running out of ideas.
Last stable MeGUI (or older/experimental it doesn't matter), last x264 (or few versions older), last Haali, encoding from .m2ts to .mkv and this is what I get: http://img87.imageshack.us/img87/9789/bscap0000tt.jpg Every single time. Different .m2ts files/sources. Always the same. Was fine before. I've compressed dozens of BD's with MeGUI and never had this issue. Tried clean reinstall MeGUI, video decoder, splitter, etc. It didn't change anything. Could this be caused by some x264 setting? I always use the same, more or less, regular settings. Nothing too experimental. I don't know what could be causing these artifacts. Can't compress anything with MeGUI anymore.

[poisondeathray]

Ok, but how I should do it? Which app?

I use TsMuxer, but when i remove interlace and convert files after muxing video with audio, the capture in file has a weird freeze of few frames and gives not synchronized audio and video. But whilst watching only video, it's good.

You can use notepad, avsp, or megui

Just edit the script, it's not that difficult:

Directshowsource("video.m2ts")

or

DSS2("video.m2ts")

[poisondeathray]

Possibly someone can help me out because am running out of ideas.
Last stable MeGUI (or older/experimental it doesn't matter), last x264 (or few versions older), last Haali, encoding from .m2ts to .mkv and this is what I get: http://img87.imageshack.us/img87/9789/bscap0000tt.jpg Every single time. Different .m2ts files/sources. Always the same. Was fine before. I've compressed dozens of BD's with MeGUI and never had this issue. Tried clean reinstall MeGUI, video decoder, splitter, etc. It didn't change anything. Could this be caused by some x264 setting? I always use the same, more or less, regular settings. Nothing too experimental. I don't know what could be causing these artifacts. Can't compress anything with MeGUI anymore.

what source filter?

post your script , settings

what playback software? what decoder?

are artifacts visible in preview before you encode?

[gorca]

As I said, Haali source (last 2 versions), different decoders (including different versions of ffdshow libav or WMV 9), no playback prbolem (only get this when encoding). Script: the usual. Nothing special about it. No MeGUI preview oartifacts.
Here are the x264 settings (although I tried some other settings also):
cabac=1 / ref=6 / deblock=1:0:0 / analyse=0x3:0x113 / me=umh / subme=9 / psy=0 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=2 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=0 / chroma_qp_offset=0 / threads=2 / sliced_threads=0 / nr=0 / decimate=0 / interlaced=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=3 / weightb=1 / open_gop=0 / weightp=2 / keyint=240 / keyint_min=23 / scenecut=40 / intra_refresh=0 / rc_lookahead=50 / rc=2pass / mbtree=1 / bitrate=9500 / ratetol=1.0 / qcomp=0.60 / qpmin=10 / qpmax=51 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / ip_ratio=1.40 / aq=1:1.00

P.S. I just encoded with x264_dp_ Unrestricted (DXVA) profile (didn't touch a thing, MeGUI 3.5.0., 1649 x264) and encoding is OK. This is obviously caused by some x264 setting. The problem is that I always use the same settings and if this is the encoder/settings problem it feels like something is broken.

[Sharktooth]

your player/decoder may be outdated and doesnt support certain x264 encoding features.
however, megui 0.3.5.0 is the latest stable version but not the most up to date.
since we're in the process of releasing a new stable version i suggest you to get the development version since it's pretty stable (probably more the the stable version you're using) and more up to date (tools included).
also update your player, ffdshow AND haali media splitter.

[blah123]

You can use notepad, avsp, or megui

Just edit the script, it's not that difficult:

Directshowsource("video.m2ts")

or

DSS2("video.m2ts")

But I use it in this way! And it isn't working!

[poisondeathray]

But I use it in this way! And it isn't working!

what's not working ? what's the error message when you preview .avs?

do you have haali media splitter, ffdshow installed ?

on win7 you need win7dsfiltertweaker as well

[blah123]

MeGUI 0.3.5.23, ffdshow, haali installed.
Ok, I try way of @poisondeathray again, but with other diffrent options.

[gorca]

Sharktooth, I've clean reinstalled everything (using only the latest meGUI, ffdshow, Haali, etc.) and it didn't change anything. However, I eventually resolved it. It seems that there was something specific with Back to the future (parts I, II, III) BluRay remuxes that I've used as sources. With these sources some x264 settings were not compatible so I got artifacts every time I changed Trellis 1 to 2, for instance. As soon as I used video-only source wrapped in .mkv that has been passed-through eac3to first, encoding artifacts were gone. Can't explain it but happy the issue was only connected to these 3 movies.
Thanks for helping out.

[Sharktooth]

glad you found a way to fix it.

[LigH]

@ blah123:

Also there is FFmpegSource2, which is as independent from the whole DirectShow system and as free of charge as DGAVCDec, supports PAFF as well as MBAFF interlacing, and should be more frame-accurate than DGAVCDec with a replaced libavcodec decoder library (which was the reason why that one got abandoned).

The problem is not the demultiplexing, it is the decoding: There are different types (PAFF / MBAFF), and the old libavcodec decoder library used in DGAVCDec only supported MBAFF, but was the last frame-accurate version. DGDecNV uses Nvidia's PureVideo hardware decoder instead, DGAVCDecDI uses DiAVC.

[blah123]

Ok, thx for support, TsMuxer handles with AVC.
But now I have a problem with audio. Is unsynchronised... I don't know, which delay I should set. tsMuxeR gives "-471", but there is still a little difference between audio and video after muxing...

[GRKNGLR]

Hi,
Will a FLAC encoder and BassAudio be added to MeGUI in the future?

[Sharktooth]

Maybe FLAC. BassAudio is not needed.

[tebasuna51]

BassAudio is not needed.
Maybe only for AAC audio.
DirectShow don't work properly:
[Error] Log
-[Information] Versions
--[Information] MeGUI Version : 0.3.5.22
--[Information] OS : Windows XP Professional x86 SP3 (5.1.196608.2600)
--[Information] Latest .Net Framework installed : unknown
--[Information] Avisynth Version : 2.5.8.5
-[Error] Log for job10 (audio, 6p321.m4a -> 6p321.ac3)
--[Information] [31/10/2010 19:14:32] Started handling job
--[Information] [31/10/2010 19:14:32] Preprocessing
--[Information] Avisynth script
---[NoImage] DirectShowSource("D:\Test\6p321.m4a")
---[NoImage] EnsureVBRMP3Sync()
---[NoImage] return last
--[Information] Commandline used: -readtoeof 1 -b 640 - "{0}"
--[Information] [31/10/2010 19:14:32] Encoding started
--[Information] [31/10/2010 19:14:32] Encode thread started
--[Information] [31/10/2010 19:14:32] Avisynth script environment opened
--[Information] [31/10/2010 19:14:35] Script loaded
--[Information] Output Decoder
---[Information] Channels: 8
---[Information] Bits per sample: 16
---[Information] Sample rate: 48000
--[Information] Commandline: D:\Programa\megui\tools\aften\aften.exe -readtoeof 1 -b 640 - "D:\Test\6p321.ac3"
--[Information] [31/10/2010 19:14:35] Encoder process started
--[Error] An error occurred
---[Error] Exception message: Ha terminado la canalización.
---[Error] Stacktrace
----[NoImage] at System.IO.__Error.WinIOError(Int32 errorCode, String maybeFullPath)
----[NoImage] at System.IO.FileStream.WriteCore(Byte[] buffer, Int32 offset, Int32 count)
----[NoImage] at System.IO.FileStream.Write(Byte[] array, Int32 offset, Int32 count)
----[NoImage] at MeGUI.AviSynthAudioEncoder.encode()
---[Error] Inner exception: null
--[Information] Output from encoder via stderr
---[NoImage] Aften: A/52 audio encoder
---[NoImage] Version SVN
---[NoImage] (c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.
---[NoImage] input format: Microsoft WAVE Signed 16-bit little-endian 48000 Hz multi-channel
---[NoImage] Unsupported # of channels passed to aften_wav_chmask_to_acmod
---[NoImage] mismatch in channels, acmod, and lfe params
--[Information] [31/10/2010 19:14:35] Job completed
My 6p321.m4a is a standard 5.1, not 8 channels.

Without problems using an avs:
BassAudioSource("D:\Test\6p321.m4a")

-[Information] Log for job11 (audio, 6p321_m4a.avs -> 6p321_m4a.ac3)
--[Information] [31/10/2010 19:30:29] Started handling job
--[Information] [31/10/2010 19:30:29] Preprocessing
--[Information] Avisynth script
---[NoImage] Import("D:\Test\6p321_m4a.avs")
---[NoImage] return last
--[Information] Commandline used: -readtoeof 1 -b 640 - "{0}"
--[Information] [31/10/2010 19:30:29] Encoding started
--[Information] [31/10/2010 19:30:29] Encode thread started
--[Information] [31/10/2010 19:30:29] Avisynth script environment opened
--[Information] [31/10/2010 19:30:33] Script loaded
--[Information] Output Decoder
---[Information] Channels: 6
---[Information] Bits per sample: 32
---[Information] Sample rate: 48000
--[Information] Commandline: D:\Programa\megui\tools\aften\aften.exe -readtoeof 1 -b 640 - "D:\Test\6p321_m4a.ac3"
--[Information] [31/10/2010 19:30:33] Encoder process started
--[Information] Output from encoder via stderr
---[NoImage] Aften: A/52 audio encoder
---[NoImage] Version SVN
---[NoImage] (c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.
---[NoImage] input format: Microsoft WAVE Floating-point 32-bit little-endian 48000 Hz 5.1-channel
---[NoImage] output format: 48000 Hz 3/2 + LFE
---[NoImage] SIMD usage: MMX SSE SSE2 SSE3
---[NoImage] Threads: 2
---[NoImage] progress: 100% | q: 534.5 | bw: 58.0 | bitrate: 640.0 kbps
--[Information] [31/10/2010 19:30:35] Postprocessing
---[Information] Deleting intermediate files
--[Information] [31/10/2010 19:30:35] Job completed

[Sharktooth]

ffaudiosource ?

[tebasuna51]

Can work also.

But the problem with ffaudiosource (also with ffmpeg) is the output decoder precission: 16 bits int
---[NoImage] input format: Microsoft WAVE Signed 16-bit little-endian 48000 Hz 5.1-channel

BassAudio and NicAudio decoders output 32 bit float samples, and like this is transferred to encoder.
---[NoImage] input format: Microsoft WAVE Floating-point 32-bit little-endian 48000 Hz 5.1-channel
If there are other functions, like downmix or samplerate change, is better also work with 32 bit float.

[Sharktooth]

32bit float has more or less the same "quality" as 24 bit int. if 32bit INT is going to be implemented in ffmpeg/libavcodec i'd prefer 32bit INT over 32bit float.

[tebasuna51]

... i'd prefer 32bit INT over 32bit float.
Of course.

But the 32 bit float is the internal format of the decoders and you can't improve the quality with a conversion to 32 bit int.

Also many AviSynth functions can only operate with 16 bit int or 32 bit float but not with 32 bit int.

I think the best option is preserve the same sample format througth the whole process.

BTW, you can implement ffaudiosource in MeGUI for AAC (and other formats not covered by NicAudio), is always better than DirectShowSource and you don't need more AviSynth plugins (BassAudio.dll, Bass.dll, Bass_aac.dll).
If a user want use BassAudio always can use the .avs method.

[LigH]

One of the main advantages of float as internal sample format would also be that you don't need to be afraid of overflows until you finally render the sample down to integer. Having a 24 bit mantissa, it won't lose much precision even for quiet scenes or calculation results (the mantissa is independent of the exponent). And some audio functions are easier to implement if they are based on normalized float values (-1.0 .. +1.0), applying them on integers might require forth-and-back conversion to float.

I speak from my experience of e.g. the "Booster" dynamic compressor in BeSweet, implemented as simple functions on normalized float samples.

[Sharktooth]

thanks for the explanation. i think we will go for the ffaudiosource way for coherency since we already have ffms and mencoder/ffmpeg for other tasks.
let's hope developers will increase the resolution in future updates.

[LigH]

:cool: Hooray -- x264 is now also available with integrated audio encoders!

:scared: Sh** -- MeGUI doesn't know that, and encodes my manually created custom AviSynth scripts with audio output without adding "--acodec none", so the "qtaac encoder" gets active, but is unstable, and my AVC encode crashes. And furthermore, I would have preferred NeroAacEnc anyway.

[nurbs]

:cool: Hooray -- x264 is now also available with integrated audio encoders!
When did that happen? I can't see anything on the mailing list or in the git changelog. :confused:

[LigH]

MeGUI 0.3.5.24 -- x264 1745-1 (Jeeb)

x264.exe --fullhelp
x264 core:107 r1745+311 2317302
Syntax: x264 [options] -o outfile infile

Infile can be raw (in which case resolution is required),
or YUV4MPEG (*.y4m),
or Avisynth if compiled with support (yes).
or libav* formats if compiled with lavf support (yes) or ffms support (yes).
Outfile type is selected by filename:
.264 -> Raw bytestream
.mkv -> Matroska
.flv -> Flash Video
.mp4 -> MP4
.3gp -> MP4 (branded '3gp6')
.3g2 -> MP4 (branded '3gp6' and '3g2a')

...

Audio:

Audio options may be used if audio support is compiled in.
Audio is automatically opened from the input file if supported by the demuxer.

--audiofile <filename> Uses audio from the specified file
--ademuxer <string> Demux audio by the specified demuxer [auto]
Supported and compiled in demuxers:
- auto, lavf, avs, lsmash
--atrack <integer> Audio track number [auto]
--acodec <string> Audio codec [auto]
Available settings:
- auto (select muxer default codec and its default encoder)
- copy (copy source audio without transcoding)
- none (disable audio)
Set audio format only and automatically choose encoder
- raw, mp3, aac, ac3, alac, mp2, vorbis, amrnb
Force to use specified audio encoder
'ff' prefix indicate they are supported via libavcodec
- raw, lame, qtaac, (ff)aac, (ff)ac3, (ff)alac, (ff)mp2, (ff)vorbis,
(ff)libopencore_amrnb
--abitrate <float> Enables bitrate mode and set bitrate (kbits/s)
--aquality <float> Quality-based VBR [codec-dependent default]
--asamplerate <integer> Audio samplerate (Hz) [keep source samplerate]
--acodec-quality <float> Codec's internal compression quality [codec specific]
--aextraopt <string> Pass extra option to codec [codec specific]
Should be comma separated "name=value" style

Encoder specific helps:
* raw encoder help
Directly pass the decoded PCM samples (in native endian) to muxer.
All audio options except for --acodec and --audiofile are ignored.

* lame encoder help
--aquality VBR quality [6]
9 (lowest) to 0 (highest)
--abitrate Enables CBR mode. Bitrate should be one of the values below
- for 32000Hz or 44100Hz or 48000Hz
32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320
- for 16000Hz or 22050Hz or 24000Hz
8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160
- for 8000Hz or 11025Hz or 12000Hz
8, 16, 24, 32, 40, 48, 56, 64
--asamplerate Output samplerate. Should be one of the values below
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
--acodec-quality Internal algorithmic complexity [0]
9 (poor quality) to 0 (best quality)

* qtaac encoder help
--aquality VBR quality [63]
Cannot be used for HE-AAC and possible values are:
0, 9 ,18 ,27 ,36 ,45 ,54 ,63 ,73, 82, 91, 100, 109, 118, 127
0 is lowest and 127 is highest.
--abitrate Enables bitrate mode
Bitrate should be one of the discrete preset values depending on
profile, channels count, and samplerate.
Examples for typical configurations
- for 44100Hz or 48000Hz with 1ch
LC: 32, 40, 48, 56, 64, 72, 80, 96, 112, 128, 144, 160, 192, 224, 256
HE: 16, 24, 32, 40
- for 44100Hz or 48000Hz with 2ch
LC: 64, 72, 80, 96, 112, 128, 144, 160, 192, 224, 256, 288, 320
HE: 32, 40, 48, 56, 64, 80
- for 44100Hz or 48000Hz with 5.1ch
LC: 160, 192, 224, 256, 288, 320, 384, 448, 512, 576, 640, 768
HE: 80, 96, 112, 128, 160, 192
The lower samplerate, the lower min/max values are applied
--asamplerate Output samplerate
LC: 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
HE: 32000, 44100, 48000
Samplerate greater than input is not supported
--acodec-quality Encoder's internal complexity [0]
0 (medium), 1 (high), 2 (highest)
--aextraopt Profile and bitrate mode
sbr : enable HE-AAC encoding [0]
mode : bitrate control mode [abr]
"abr", "cbr", "cvbr"

--aquality/--abitrate setting may be changed inside codec due to its
limitations and extreme resampling settings (e.g. 48000->8000) may not work.
If something goes wrong, it will result in a failure of codec initialization.

* (ff)libopencore_amrnb encoder help
Accepts only mono (1ch), 8000Hz audio and not capable of quality based VBR
--abitrate Only one of the values below can be acceptable
4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2, 12.2

...

[nurbs]

So it's a patch, I see. I didn't check if there was a difference between 1745 and 1745-1 and I never noticed it since my scripts don't have audio. Thanks.

[Sharktooth]

add --acodec none in the custom commandline options of the x264 preset config.

[LigH]

... until it gets officially supported by (or "banned" from) MeGUI. ;)
__

P.S.: Added a tracker task (https://sourceforge.net/tracker/?func=detail&aid=3102234&group_id=156112&atid=798476).