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3rd August 2017, 09:19 | #5681 | Link |
German doom9/Gleitz SuMo
Join Date: Oct 2001
Location: Germany, rural Altmark
Posts: 6,782
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Please post a complete MeGUI session log of such a case, preferably including a full MediaInfo analysis of the source.
Where does your video source come from? May it have a variable framerate? - AviSynth can only handle constant frame rates. Some source plugins can resample the video to a constant frame rate, with the right parameters. But don't try that if it is not necessary; there are other possible reasons for a change. First we need to know details what you tried so far. |
3rd August 2017, 09:45 | #5682 | Link |
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Source MediaInfo (Frame rate mode: Variable:24.616 FPS)
https://pastebin.com/gDtFQDFG Full Log: https://pastebin.com/PJc3MnX0 My Profile: http://www82.zippyshare.com/v/Tsq80Ji2/file.html Encode MediaInfo (Frame rate mode: Constant 24.617 FPS https://pastebin.com/0LDdm5Fq you right fps has changed to the Constant mode, What are the solutions to this problem? |
3rd August 2017, 10:10 | #5683 | Link |
German doom9/Gleitz SuMo
Join Date: Oct 2001
Location: Germany, rural Altmark
Posts: 6,782
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Two possible solutions:
a) You may manually edit the generated AviSynth script to use the fpsnum/fpsden parameters in LwLibavVideoSource to resample the video to a constant and convenient frame rate. Disadvantage: Choppy playback due to inserted or skipped frames. b) You may extract per-frame timecodes before converting, and restore them in your result. Disadvantage: I don't know details about it, and it is only supported for specific target containers. Please try to find more about this technique on your own (or have patience and wait for links provided by other members). This may require more manual labor outside MeGUI. Last edited by LigH; 3rd August 2017 at 10:12. |
8th August 2017, 21:37 | #5684 | Link |
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Location: Sweden
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FFmpeg v3.3.3 binaries is now available @ Zeranoe's FFmpeg site
http://ffmpeg.zeranoe.com/builds/ |
12th August 2017, 03:00 | #5685 | Link | |
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Quote:
LWLibavVideoSource("D:\video.mkv.lwi", fpsnum=30000, fpsden=1001) TFM().TDecimate() Alternatively, extract the timecodes from the source MKV with gMKVExtractGUI or MKVCleaver, then use the timecodes file to encode in variable frame rate mode.. If the timecodes file you extracted is called "timecodes.txt" (for example) you'd add it to the x264 custom command line section like this: --tcfile-in "D:\timecodes.txt" Save the encoder configuration and add the job to the queue as usual. The frame rate of the video as it's being decoded by LSMash is then irrelevant. |
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20th August 2017, 17:28 | #5690 | Link | |
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Quote:
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28th August 2017, 22:19 | #5691 | Link | |
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Hey guys,
I have a problem. Where is DirectShowSource on Avisynth SC? I want to use Lavfilters. Quote:
PS: Sorry, my english is very bad. |
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28th August 2017, 22:30 | #5692 | Link |
German doom9/Gleitz SuMo
Join Date: Oct 2001
Location: Germany, rural Altmark
Posts: 6,782
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This change seems to make this choice only relevant if you select the One Click Encoder.
AviSource is only suitable if the source is an AVI (and possibly only necessary if it doesn't use a codec LibAV decoders can't handle reliably). And DirectShowSource will probably be offered only if nothing else is obviously more suitable (I guess...). |
31st August 2017, 18:53 | #5693 | Link | |
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Quote:
That change seems to have the potential to be annoying. Other than that, you can create your own DirectShowSource script to open the file in question, and use the script as the input video for the script creator. Last edited by hello_hello; 31st August 2017 at 19:06. |
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3rd September 2017, 06:30 | #5694 | Link |
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Join Date: Mar 2017
Posts: 51
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@Zathor
When transcoding audio using OneClick, it shifts the audio tracks to the wrong position in the output file (even though tracks were in the right order while in the OneClick GUI). I've included a sample file + steps to reproduce: https://mega.nz/#!ZHpjWIBI!VRLZnCWDy...xGVYaedT9rRY1Q |
7th September 2017, 16:29 | #5695 | Link |
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Location: Sweden
Posts: 483
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@ Zathor
New DGDecNV is out.... DGDecNV 2053 slipstream 111 (2017-08-28) - Added an error popup for cases of filenames containing illegal characters (or other cases of file open failure). Previously, it would fail silently. |
9th September 2017, 15:21 | #5698 | Link |
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Join Date: Aug 2002
Location: Sweden
Posts: 65
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There's an issue with the generated .avs file for audio when indexing a video with 6 channel aac audio. In the scripts it says
# detected channels: 2 channels # detected channel positions: 3/2/0.1 regardless of ffmsindex or l-smash being used. MeGUI throw warnings when encoding: --[Warning] [2017-09-08 17:00:32] Channel count mismatch! The input file is reporting 2 channels and the AviSynth script is reporting 6 channels --[Warning] [2017-09-08 17:00:32] Ignoring downmix because of the channel count mismatch and the resulting encode ends up being multi-channel. Trying to encode an mp3 fails with the error message "Unsupported number of channels: 6" Manually changing the .avs files to # detected channels: 6 channels makes everything work fine. Using MeGUI 2768 development server (By the way, is there an easy way of verifying that video/audio files actually are multi-channel without having any matching hardware for playback? A longer-than-normal google session only came up with tools and ways for checking playback and hardware, nothing for verifying the files. Maybe not on topic, but I'm also curious of the big difference in bitrate for the resulting audio file, depending of the l-smash or ffms indexer being used, and other decoding methods. With the l-smash indexer the encoded 2-channel audio had 51 kbps, and with ffms it had 40 kbps. Just encoding the extracted aac file directly in MeGUI resulted in a 2-channel file with 66 kbps, and encoding in foobar2000 with it's downmix filter produced a file with 42 kbps. Encoding settings being the same of course. Is this all good and perfectly normal, especially the difference between l-smash and ffms indexers?) Last edited by j8ee; 9th September 2017 at 15:30. |
9th September 2017, 18:58 | #5700 | Link | |||
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Quote:
Some AAC encoders save 2 channel info to the MP4 header (or something along those lines) for backwards compatibility as MP4 originally only supported 2 channels, so maybe that's confusing MeGUI. As a workaround, you could re-encode as flac, as based on your MP3 test the flac output would have 6 channels, and then you could downmix and re-encode the flac file. Quote:
It's under "Playback Visualisations" in the list of UI Elements. You don't have to let it take up much space if it's at a premium as it has a full screen mode (right click on it). Keep in mind it's an output meter so if you have a DSP in the playback chain downmixing etc, it'll display the DSP's output, but that aside, I tend to trust what it's displaying. Quote:
I'm using an older LSmash and it seemed to be confused by raw HE-AAC and was getting the sample rate wrong. For comparing decoders I'd convert to flac or wave first, or at least mux the raw AAC into an MKA or M4A. Re-encoding raw HE-AAC streams is probably something to avoid. Last edited by hello_hello; 9th September 2017 at 19:01. |
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