I am trying to find out how to pass Quenc the parameters in the pictures below through the command line. I am going to make a batch file and process multiple AVS files.
I do not know what version of Quenc I have, I am using the version that comes with AVSToDVD 2.0.
Also, I need to put a pulldown on the videos as they are 23.976, but Quenc would not let me check it in the GUI.
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Also, has anyone ever experienced problems with CCE when trying to encode a MP4 to MPEG-2 EVEN when the audio is disabled via AviSynth?
Usually CCE is fine working with MP4 files if the audio is disabled but I have some MP4 Files that are causing CCE to crash right when the .vaf file is being made. The script works fine in VirtualModDub.
Here is the script I am using:
<SCRIPT>
# -= AviSynth v2.5.7.0 script by FitCD v1.2.8 =-
DirectShowSource("C:\myvideo.mp4", audio=false)
LanczosResize(640,440)
AddBorders(40,16,40,24)
ConvertToYUY2() # For VirtualDub or CCE
</SCRIPT>
Here are the file specs via MediaInfo:
General #0
Complete name : C:\myvideo.mp4
Format : MPEG-4
Format/Info : ISO 14496-1 Base Media
Format/Family : MPEG-4
File size : 274 MiB
PlayTime : 24mn 256ms
Bit rate : 1598 Kbps
StreamSize/String : 567 KiB
Encoded date : UTC 2007-08-30 06:12:07
Tagged date : UTC 2007-08-30 06:12:07
Video #0
Codec : AVC
Codec/Family : AVC
Codec/Info : Advanced Video Codec
Codec profile : Main@L4.1
Codec settings, CABAC : No
Codec_Settings_RefFrames : 7
PlayTime : 24mn 188ms
Bit rate mode : VBR
Bit rate : 1402 Kbps
Maximum bit rate : 5877 Kbps
Width : 1280 pixels
Height : 720 pixels
Display Aspect ratio : 16/9
Frame rate mode : CFR
Frame rate : 23.976 fps
Chroma : 4:2:0
Interlacement : Progressive
Bits/(Pixel*Frame) : 0.063
StreamSize/String : 241 MiB
Encoded date : UTC 2007-08-30 05:41:19
Tagged date : UTC 2007-08-30 06:12:27
Audio #0
Codec : AAC LC
Codec/Family : AAC
Codec/Info : AAC Low Complexity
PlayTime : 24mn 256ms
Bit rate mode : VBR
Bit rate : 192 Kbps
Maximum bit rate : 250 Kbps
Channel(s) : 2 channels
Channel positions : L R
Sampling rate : 48.0 KHz
Resolution : 16 bits
StreamSize/String : 33.2 MiB
Title : aac audio
Language : Japanese
Encoded date : UTC 2007-08-30 06:12:26
Tagged date : UTC 2007-08-30 06:12:27 -
In the same folder where QuEnc is installed there should be a file named QuEnc_Settings.txt that gives you the commands for command line encoding.
If you don't have that file, go to a command prompt and from the Quenc folder type "Quenc /?" (without the quotes) and it should give you the command list.
Edit: Also according to your screenshot, you're using version 0.75alpha"Shut up Wesley!" -- Captain Jean-Luc Picard
Buy My Books -
Frankly, I'm not sure how QuEnc handles pulldown since I've never used it with pulldown. You can check the box by selecting Interlaced encoding, and interestingly enough you can deselect Interlaced and Pulldown remains checked (although disabled). I don't know whether this means that it applies pulldown to encode to 29.97fps, or that it actually encodes in 23.976 with the pulldown flag set. You could try experimenting with a small segment to see how it works for sure. You can't set pulldown flags in AviSynth, that has to be done in the encoder. You can only have AviSynth convert 23.976 to 29.97 but it will be encoded that way.
Another option is to encode at 23.976 and then use another application to just set the pulldown flags. Unfortunately I can't remember the name of the application that changes just the flags. Maybe someone else can remember for me..."Shut up Wesley!" -- Captain Jean-Luc Picard
Buy My Books -
Originally Posted by Soopafresh
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You could try -pulldown as part of the command line. That would follow the pattern in case the author just didn't update the documentation. But again, I still don't know if that will set flags or encode 29.97.
"Shut up Wesley!" -- Captain Jean-Luc Picard
Buy My Books -
You can run dgpulldown from the command line as well. Just trying to give out some options.
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mickgreen?
to select 'pulldown' in the gui you need to select 'interlaced' too first.
like gadget i thought of using -pulldown too,
but it doesn't work.
so i guess you'd have to set pulldown in the gui, and then set up your command line.
a bit shoddy.
hadn't tried quenc cli before. a bit shoddy.
anyway i don't ever use the encoder, just have it lying around.
but i also wonder about using an alpha.
i remember there being 75alpha and alpha2,
and something about trellis...
can't remember, read through the quenc thread to see what your version does differently.
as for options,
dgpulldown is a good one, i think you went through cli options with soopa already.
and mencoder can do things in a single command.
and of course
try HC
tripp -
Originally Posted by Soopafresh
I understand I can use DGPulldown from the command line, I was actually posting about last week (I think it was you that helped me with the DOS command).
What I was trying to find out is if I could:
1) Pass a pulldown flag to Quenc from the Command Line
2) Call some function in Avisynth to do a pulldown -
Originally Posted by 45tripp
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The only reason I was using Quenc is because when CCE won't work with a particular MP4 file, I automatically go to AVSToDVD as my 2nd option for MP4s and Quenc is one of the Encoders that it uses.
@gadgetguy, I tried putting the -pulldown in Quenc command line but it gave an error when I ran the batch file as it said it was an invalid parameter. When be interesting to see what is passed to Quenc when you check pulldown in the GUI.
Thanks for the help everyone, as I stated I just changed my colorspace in avisynth and CCE is now happy although I have one file that won't work, but it has to do with the audio. I will post that error in the next post. -
I have an FFMpeg Audio Question. I am not sure if this should be in another thread, but I am going to ask it in this thread.
I am getting and error when I try to use FFMPeg to get the audio from a MP4 file. The error is:
Unexpected Channel Configuration Change
Here is what I am passing to FFMpeg:
ffmpeg -i "c:\myfile.mp4" -ab 128K -ar 48000 -ac 2 -acodec ac3 -0 "c:\myaudio.ac3"
I have taken out the "-ac 2" as well and got the same results.
If anyone is familiar with Xilosoft Audio Converter, it got the same error as well. I think I may try Mencoder or something else and see what happens.
Here is the file info from MediaInfo:
<MEDIAINFO>
General #0
Complete name : C:\myvideo.mp4
Format : MPEG-4
Format/Info : ISO 14496-1 Base Media
Format/Family : MPEG-4
File size : 274 MiB
PlayTime : 24mn 42ms
Bit rate : 1597 Kbps
StreamSize/String : 563 KiB
Encoded date : UTC 2008-01-30 17:44:43
Tagged date : UTC 2008-01-30 17:44:43
Video #0
Codec : AVC
Codec/Family : AVC
Codec/Info : Advanced Video Codec
Codec profile : Main@L4.1
Codec settings, CABAC : No
Codec_Settings_RefFrames : 7
PlayTime : 23mn 59s
Bit rate mode : VBR
Bit rate : 1400 Kbps
Maximum bit rate : 5589 Kbps
Width : 1280 pixels
Height : 720 pixels
Display Aspect ratio : 16/9
Frame rate mode : CFR
Frame rate : 23.976 fps
Chroma : 4:2:0
Interlacement : Progressive
Bits/(Pixel*Frame) : 0.063
StreamSize/String : 240 MiB
Encoded date : UTC 2008-01-30 10:51:51
Tagged date : UTC 2008-01-30 17:44:55
Audio #0
Codec : AAC LC
Codec/Family : AAC
Codec/Info : AAC Low Complexity
PlayTime : 24mn 42ms
Bit rate mode : VBR
Bit rate : 192 Kbps
Maximum bit rate : 218 Kbps
Channel(s) : 2 channels
Channel positions : L R
Sampling rate : 48.0 KHz
Resolution : 16 bits
StreamSize/String : 33.2 MiB
Encoded date : UTC 2008-01-30 17:44:53
Tagged date : UTC 2008-01-30 17:44:55
</MEDIAINFO> -
I can't answer directly since I don't use ffmpeg. I know that AAC and AC3 need to encode the channels in a different order, and the error message seems to point to that, but I don't see how it would know or why it would matter with only a two channel audio stream. IMHO the problem is more likely the fact that you are feeding VBR audio. Many applications have trouble with that.
"Shut up Wesley!" -- Captain Jean-Luc Picard
Buy My Books -
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Soopafresh, I wasn't trying to be rude and if I came off that way, I apologize.
I didn't even think of it that way. No probs
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